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diff --git a/libAACdec/src/usacdec_acelp.cpp b/libAACdec/src/usacdec_acelp.cpp new file mode 100644 index 0000000..ec4437f --- /dev/null +++ b/libAACdec/src/usacdec_acelp.cpp @@ -0,0 +1,1289 @@ +/* ----------------------------------------------------------------------------- +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): Matthias Hildenbrand + + Description: USAC ACELP frame decoder + +*******************************************************************************/ + +#include "usacdec_acelp.h" + +#include "usacdec_ace_d4t64.h" +#include "usacdec_ace_ltp.h" +#include "usacdec_rom.h" +#include "usacdec_lpc.h" +#include "genericStds.h" + +#define PIT_FR2_12k8 128 /* Minimum pitch lag with resolution 1/2 */ +#define PIT_FR1_12k8 160 /* Minimum pitch lag with resolution 1 */ +#define TILT_CODE2 \ + FL2FXCONST_SGL(0.3f * 2.0f) /* ACELP code pre-emphasis factor ( *2 ) */ +#define PIT_SHARP \ + FL2FXCONST_SGL(0.85f) /* pitch sharpening factor */ +#define PREEMPH_FAC \ + FL2FXCONST_SGL(0.68f) /* ACELP synth pre-emphasis factor */ + +#define ACELP_HEADROOM 1 +#define ACELP_OUTSCALE (MDCT_OUT_HEADROOM - ACELP_HEADROOM) + +/** + * \brief Calculate pre-emphasis (1 - mu z^-1) on input signal. + * \param[in] in pointer to input signal; in[-1] is also needed. + * \param[out] out pointer to output signal. + * \param[in] L length of filtering. + */ +/* static */ +void E_UTIL_preemph(const FIXP_DBL *in, FIXP_DBL *out, INT L) { + int i; + + for (i = 0; i < L; i++) { + out[i] = in[i] - fMult(PREEMPH_FAC, in[i - 1]); + } + + return; +} + +/** + * \brief Calculate de-emphasis 1/(1 - TILT_CODE z^-1) on innovative codebook + * vector. + * \param[in,out] x innovative codebook vector. + */ +static void Preemph_code( + FIXP_COD x[] /* (i/o) : input signal overwritten by the output */ +) { + int i; + FIXP_DBL L_tmp; + + /* ARM926: 12 cycles per sample */ + for (i = L_SUBFR - 1; i > 0; i--) { + L_tmp = FX_COD2FX_DBL(x[i]); + L_tmp -= fMultDiv2(x[i - 1], TILT_CODE2); + x[i] = FX_DBL2FX_COD(L_tmp); + } +} + +/** + * \brief Apply pitch sharpener to the innovative codebook vector. + * \param[in,out] x innovative codebook vector. + * \param[in] pit_lag decoded pitch lag. + */ +static void Pit_shrp( + FIXP_COD x[], /* in/out: impulse response (or algebraic code) */ + int pit_lag /* input : pitch lag */ +) { + int i; + FIXP_DBL L_tmp; + + for (i = pit_lag; i < L_SUBFR; i++) { + L_tmp = FX_COD2FX_DBL(x[i]); + L_tmp += fMult(x[i - pit_lag], PIT_SHARP); + x[i] = FX_DBL2FX_COD(L_tmp); + } + + return; +} + + /** + * \brief Calculate Quantized codebook gain, Quantized pitch gain and unbiased + * Innovative code vector energy. + * \param[in] index index of quantizer. + * \param[in] code innovative code vector with exponent = SF_CODE. + * \param[out] gain_pit Quantized pitch gain g_p with exponent = SF_GAIN_P. + * \param[out] gain_code Quantized codebook gain g_c. + * \param[in] mean_ener mean_ener defined in open-loop (2 bits), exponent = 7. + * \param[out] E_code unbiased innovative code vector energy. + * \param[out] E_code_e exponent of unbiased innovative code vector energy. + */ + +#define SF_MEAN_ENER_LG10 9 + +/* pow(10.0, {18, 30, 42, 54}/20.0) /(float)(1<<SF_MEAN_ENER_LG10) */ +static const FIXP_DBL pow_10_mean_energy[4] = {0x01fc5ebd, 0x07e7db92, + 0x1f791f65, 0x7d4bfba3}; + +static void D_gain2_plus(int index, FIXP_COD code[], FIXP_SGL *gain_pit, + FIXP_DBL *gain_code, int mean_ener_bits, int bfi, + FIXP_SGL *past_gpit, FIXP_DBL *past_gcode, + FIXP_DBL *pEner_code, int *pEner_code_e) { + FIXP_DBL Ltmp; + FIXP_DBL gcode0, gcode_inov; + INT gcode0_e, gcode_inov_e; + int i; + + FIXP_DBL ener_code; + INT ener_code_e; + + /* ener_code = sum(code[]^2) */ + ener_code = FIXP_DBL(0); + for (i = 0; i < L_SUBFR; i++) { + ener_code += fPow2Div2(code[i]); + } + + ener_code_e = fMax(fNorm(ener_code) - 1, 0); + ener_code <<= ener_code_e; + ener_code_e = 2 * SF_CODE + 1 - ener_code_e; + + /* export energy of code for calc_period_factor() */ + *pEner_code = ener_code; + *pEner_code_e = ener_code_e; + + ener_code += scaleValue(FL2FXCONST_DBL(0.01f), -ener_code_e); + + /* ener_code *= 1/L_SUBFR, and make exponent even (because of square root + * below). */ + if (ener_code_e & 1) { + ener_code_e -= 5; + ener_code >>= 1; + } else { + ener_code_e -= 6; + } + gcode_inov = invSqrtNorm2(ener_code, &gcode0_e); + gcode_inov_e = gcode0_e - (ener_code_e >> 1); + + if (bfi) { + FIXP_DBL tgcode; + FIXP_SGL tgpit; + + tgpit = *past_gpit; + + if (tgpit > FL2FXCONST_SGL(0.95f / (1 << SF_GAIN_P))) { + tgpit = FL2FXCONST_SGL(0.95f / (1 << SF_GAIN_P)); + } else if (tgpit < FL2FXCONST_SGL(0.5f / (1 << SF_GAIN_P))) { + tgpit = FL2FXCONST_SGL(0.5f / (1 << SF_GAIN_P)); + } + *gain_pit = tgpit; + tgpit = FX_DBL2FX_SGL(fMult(tgpit, FL2FXCONST_DBL(0.95f))); + *past_gpit = tgpit; + + tgpit = FL2FXCONST_SGL(1.4f / (1 << SF_GAIN_P)) - tgpit; + tgcode = fMult(*past_gcode, tgpit) << SF_GAIN_P; + *gain_code = scaleValue(fMult(tgcode, gcode_inov), gcode_inov_e); + *past_gcode = tgcode; + + return; + } + + /*-------------- Decode gains ---------------*/ + /* + gcode0 = pow(10.0, (float)mean_ener/20.0); + gcode0 = gcode0 / sqrt(ener_code/L_SUBFR); + */ + gcode0 = pow_10_mean_energy[mean_ener_bits]; + gcode0 = fMultDiv2(gcode0, gcode_inov); + gcode0_e = gcode0_e + SF_MEAN_ENER_LG10 - (ener_code_e >> 1) + 1; + + i = index << 1; + *gain_pit = t_qua_gain7b[i]; /* adaptive codebook gain */ + /* t_qua_gain[ind2p1] : fixed codebook gain correction factor */ + Ltmp = fMult(t_qua_gain7b[i + 1], gcode0); + *gain_code = scaleValue(Ltmp, gcode0_e - SF_GAIN_C + SF_QUA_GAIN7B); + + /* update bad frame handler */ + *past_gpit = *gain_pit; + + /*-------------------------------------------------------- + past_gcode = gain_code/gcode_inov + --------------------------------------------------------*/ + { + FIXP_DBL gcode_m; + INT gcode_e; + + gcode_m = fDivNormHighPrec(Ltmp, gcode_inov, &gcode_e); + gcode_e += (gcode0_e - SF_GAIN_C + SF_QUA_GAIN7B) - (gcode_inov_e); + *past_gcode = scaleValue(gcode_m, gcode_e); + } +} + +/** + * \brief Calculate period/voicing factor r_v + * \param[in] exc pitch excitation. + * \param[in] gain_pit gain of pitch g_p. + * \param[in] gain_code gain of code g_c. + * \param[in] gain_code_e exponent of gain of code. + * \param[in] ener_code unbiased innovative code vector energy. + * \param[in] ener_code_e exponent of unbiased innovative code vector energy. + * \return period/voice factor r_v (-1=unvoiced to 1=voiced), exponent SF_PFAC. + */ +static FIXP_DBL calc_period_factor(FIXP_DBL exc[], FIXP_SGL gain_pit, + FIXP_DBL gain_code, FIXP_DBL ener_code, + int ener_code_e) { + int ener_exc_e, L_tmp_e, s = 0; + FIXP_DBL ener_exc, L_tmp; + FIXP_DBL period_fac; + + /* energy of pitch excitation */ + ener_exc = (FIXP_DBL)0; + for (int i = 0; i < L_SUBFR; i++) { + ener_exc += fPow2Div2(exc[i]) >> s; + if (ener_exc > FL2FXCONST_DBL(0.5f)) { + ener_exc >>= 1; + s++; + } + } + + ener_exc_e = fNorm(ener_exc); + ener_exc = fMult(ener_exc << ener_exc_e, fPow2(gain_pit)); + if (ener_exc != (FIXP_DBL)0) { + ener_exc_e = 2 * SF_EXC + 1 + 2 * SF_GAIN_P - ener_exc_e + s; + } else { + ener_exc_e = 0; + } + + /* energy of innovative code excitation */ + /* L_tmp = ener_code * gain_code*gain_code; */ + L_tmp_e = fNorm(gain_code); + L_tmp = fPow2(gain_code << L_tmp_e); + L_tmp = fMult(ener_code, L_tmp); + L_tmp_e = 2 * SF_GAIN_C + ener_code_e - 2 * L_tmp_e; + + /* Find common exponent */ + { + FIXP_DBL num, den; + int exp_diff; + + exp_diff = ener_exc_e - L_tmp_e; + if (exp_diff >= 0) { + ener_exc >>= 1; + if (exp_diff <= DFRACT_BITS - 2) { + L_tmp >>= exp_diff + 1; + } else { + L_tmp = (FIXP_DBL)0; + } + den = ener_exc + L_tmp; + if (ener_exc_e < DFRACT_BITS - 1) { + den += scaleValue(FL2FXCONST_DBL(0.01f), -ener_exc_e - 1); + } + } else { + if (exp_diff >= -(DFRACT_BITS - 2)) { + ener_exc >>= 1 - exp_diff; + } else { + ener_exc = (FIXP_DBL)0; + } + L_tmp >>= 1; + den = ener_exc + L_tmp; + if (L_tmp_e < DFRACT_BITS - 1) { + den += scaleValue(FL2FXCONST_DBL(0.01f), -L_tmp_e - 1); + } + } + num = (ener_exc - L_tmp); + num >>= SF_PFAC; + + if (den > (FIXP_DBL)0) { + if (ener_exc > L_tmp) { + period_fac = schur_div(num, den, 16); + } else { + period_fac = -schur_div(-num, den, 16); + } + } else { + period_fac = (FIXP_DBL)MAXVAL_DBL; + } + } + + /* exponent = SF_PFAC */ + return period_fac; +} + +/*------------------------------------------------------------* + * noise enhancer * + * ~~~~~~~~~~~~~~ * + * - Enhance excitation on noise. (modify gain of code) * + * If signal is noisy and LPC filter is stable, move gain * + * of code 1.5 dB toward gain of code threshold. * + * This decrease by 3 dB noise energy variation. * + *------------------------------------------------------------*/ +/** + * \brief Enhance excitation on noise. (modify gain of code) + * \param[in] gain_code Quantized codebook gain g_c, exponent = SF_GAIN_C. + * \param[in] period_fac periodicity factor, exponent = SF_PFAC. + * \param[in] stab_fac stability factor, exponent = SF_STAB. + * \param[in,out] p_gc_threshold modified gain of previous subframe. + * \return gain_code smoothed gain of code g_sc, exponent = SF_GAIN_C. + */ +static FIXP_DBL +noise_enhancer(/* (o) : smoothed gain g_sc SF_GAIN_C */ + FIXP_DBL gain_code, /* (i) : Quantized codebook gain SF_GAIN_C */ + FIXP_DBL period_fac, /* (i) : periodicity factor (-1=unvoiced to + 1=voiced), SF_PFAC */ + FIXP_SGL stab_fac, /* (i) : stability factor (0 <= ... < 1.0) + SF_STAB */ + FIXP_DBL + *p_gc_threshold) /* (io): gain of code threshold SF_GAIN_C */ +{ + FIXP_DBL fac, L_tmp, gc_thres; + + gc_thres = *p_gc_threshold; + + L_tmp = gain_code; + if (L_tmp < gc_thres) { + L_tmp += fMultDiv2(gain_code, + FL2FXCONST_SGL(2.0 * 0.19f)); /* +1.5dB => *(1.0+0.19) */ + if (L_tmp > gc_thres) { + L_tmp = gc_thres; + } + } else { + L_tmp = fMult(gain_code, + FL2FXCONST_SGL(1.0f / 1.19f)); /* -1.5dB => *10^(-1.5/20) */ + if (L_tmp < gc_thres) { + L_tmp = gc_thres; + } + } + *p_gc_threshold = L_tmp; + + /* voicing factor lambda = 0.5*(1-period_fac) */ + /* gain smoothing factor S_m = lambda*stab_fac (=fac) + = 0.5(stab_fac - stab_fac * period_fac) */ + fac = (FX_SGL2FX_DBL(stab_fac) >> (SF_PFAC + 1)) - + fMultDiv2(stab_fac, period_fac); + /* fac_e = SF_PFAC + SF_STAB */ + FDK_ASSERT(fac >= (FIXP_DBL)0); + + /* gain_code = (float)((fac*tmp) + ((1.0-fac)*gain_code)); */ + gain_code = fMult(fac, L_tmp) - + fMult(FL2FXCONST_DBL(-1.0f / (1 << (SF_PFAC + SF_STAB))) + fac, + gain_code); + gain_code <<= (SF_PFAC + SF_STAB); + + return gain_code; +} + +/** + * \brief Update adaptive codebook u'(n) (exc) + * Enhance pitch of c(n) and build post-processed excitation u(n) (exc2) + * \param[in] code innovative codevector c(n), exponent = SF_CODE. + * \param[in,out] exc filtered adaptive codebook v(n), exponent = SF_EXC. + * \param[in] gain_pit adaptive codebook gain, exponent = SF_GAIN_P. + * \param[in] gain_code innovative codebook gain g_c, exponent = SF_GAIN_C. + * \param[in] gain_code_smoothed smoothed innov. codebook gain g_sc, exponent = + * SF_GAIN_C. + * \param[in] period_fac periodicity factor r_v, exponent = SF_PFAC. + * \param[out] exc2 post-processed excitation u(n), exponent = SF_EXC. + */ +void BuildAdaptiveExcitation( + FIXP_COD code[], /* (i) : algebraic codevector c(n) Q9 */ + FIXP_DBL exc[], /* (io): filtered adaptive codebook v(n) Q15 */ + FIXP_SGL gain_pit, /* (i) : adaptive codebook gain g_p Q14 */ + FIXP_DBL gain_code, /* (i) : innovative codebook gain g_c Q16 */ + FIXP_DBL gain_code_smoothed, /* (i) : smoothed innov. codebook gain g_sc + Q16 */ + FIXP_DBL period_fac, /* (i) : periodicity factor r_v Q15 */ + FIXP_DBL exc2[] /* (o) : post-processed excitation u(n) Q15 */ +) { +/* Note: code[L_SUBFR] and exc2[L_SUBFR] share the same memory! + If exc2[i] is written, code[i] will be destroyed! +*/ +#define SF (SF_CODE + SF_GAIN_C + 1 - SF_EXC) + + int i; + FIXP_DBL tmp, cpe, code_smooth_prev, code_smooth; + + FIXP_COD code_i; + FIXP_DBL cpe_code_smooth, cpe_code_smooth_prev; + + /* cpe = (1+r_v)/8 * 2 ; ( SF = -1) */ + cpe = (period_fac >> (2 - SF_PFAC)) + FL2FXCONST_DBL(0.25f); + + /* u'(n) */ + tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P + 1); /* v(0)*g_p */ + *exc++ = tmp + (fMultDiv2(code[0], gain_code) << SF); + + /* u(n) */ + code_smooth_prev = fMultDiv2(*code++, gain_code_smoothed) + << SF; /* c(0) * g_sc */ + code_i = *code++; + code_smooth = fMultDiv2(code_i, gain_code_smoothed) << SF; /* c(1) * g_sc */ + tmp += code_smooth_prev; /* tmp = v(0)*g_p + c(0)*g_sc */ + cpe_code_smooth = fMultDiv2(cpe, code_smooth); + *exc2++ = tmp - cpe_code_smooth; + cpe_code_smooth_prev = fMultDiv2(cpe, code_smooth_prev); + + i = L_SUBFR - 2; + do /* ARM926: 22 cycles per iteration */ + { + /* u'(n) */ + tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P + 1); + *exc++ = tmp + (fMultDiv2(code_i, gain_code) << SF); + /* u(n) */ + tmp += code_smooth; /* += g_sc * c(i) */ + tmp -= cpe_code_smooth_prev; + cpe_code_smooth_prev = cpe_code_smooth; + code_i = *code++; + code_smooth = fMultDiv2(code_i, gain_code_smoothed) << SF; + cpe_code_smooth = fMultDiv2(cpe, code_smooth); + *exc2++ = tmp - cpe_code_smooth; /* tmp - c_pe * g_sc * c(i+1) */ + } while (--i != 0); + + /* u'(n) */ + tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P + 1); + *exc = tmp + (fMultDiv2(code_i, gain_code) << SF); + /* u(n) */ + tmp += code_smooth; + tmp -= cpe_code_smooth_prev; + *exc2++ = tmp; + + return; +} + +/** + * \brief Interpolate LPC vector in LSP domain for current subframe and convert + * to LP domain + * \param[in] lsp_old LPC vector (LSP domain) corresponding to the beginning of + * current ACELP frame. + * \param[in] lsp_new LPC vector (LSP domain) corresponding to the end of + * current ACELP frame. + * \param[in] subfr_nr number of current ACELP subframe 0..3. + * \param[in] nb_subfr total number of ACELP subframes in this frame. + * \param[out] A LP filter coefficients for current ACELP subframe, exponent = + * SF_A_COEFFS. + */ +/* static */ +void int_lpc_acelp( + const FIXP_LPC lsp_old[], /* input : LSPs from past frame */ + const FIXP_LPC lsp_new[], /* input : LSPs from present frame */ + int subfr_nr, int nb_subfr, + FIXP_LPC + A[], /* output: interpolated LP coefficients for current subframe */ + INT *A_exp) { + int i; + FIXP_LPC lsp_interpol[M_LP_FILTER_ORDER]; + FIXP_SGL fac_old, fac_new; + + FDK_ASSERT((nb_subfr == 3) || (nb_subfr == 4)); + + fac_old = lsp_interpol_factor[nb_subfr & 0x1][(nb_subfr - 1) - subfr_nr]; + fac_new = lsp_interpol_factor[nb_subfr & 0x1][subfr_nr]; + for (i = 0; i < M_LP_FILTER_ORDER; i++) { + lsp_interpol[i] = FX_DBL2FX_LPC( + (fMultDiv2(lsp_old[i], fac_old) + fMultDiv2(lsp_new[i], fac_new)) << 1); + } + + E_LPC_f_lsp_a_conversion(lsp_interpol, A, A_exp); + + return; +} + +/** + * \brief Perform LP synthesis by filtering the post-processed excitation u(n) + * through the LP synthesis filter 1/A(z) + * \param[in] a LP filter coefficients, exponent = SF_A_COEFFS. + * \param[in] length length of input/output signal. + * \param[in] x post-processed excitation u(n). + * \param[in,out] y LP synthesis signal and filter memory + * y[-M_LP_FILTER_ORDER..-1]. + */ + +/* static */ +void Syn_filt(const FIXP_LPC a[], /* (i) : a[m] prediction coefficients Q12 */ + const INT a_exp, + INT length, /* (i) : length of input/output signal (64|128) */ + FIXP_DBL x[], /* (i) : input signal Qx */ + FIXP_DBL y[] /* (i/o) : filter states / output signal Qx-s*/ +) { + int i, j; + FIXP_DBL L_tmp; + + for (i = 0; i < length; i++) { + L_tmp = (FIXP_DBL)0; + + for (j = 0; j < M_LP_FILTER_ORDER; j++) { + L_tmp -= fMultDiv2(a[j], y[i - (j + 1)]); + } + + L_tmp = scaleValue(L_tmp, a_exp + 1); + y[i] = L_tmp + x[i]; + } + + return; +} + +/** + * \brief Calculate de-emphasis 1/(1 - mu z^-1) on input signal. + * \param[in] x input signal. + * \param[out] y output signal. + * \param[in] L length of signal. + * \param[in,out] mem memory (signal[-1]). + */ +/* static */ +void Deemph(FIXP_DBL *x, FIXP_DBL *y, int L, FIXP_DBL *mem) { + int i; + FIXP_DBL yi = *mem; + + for (i = 0; i < L; i++) { + FIXP_DBL xi = x[i] >> 1; + xi = fMultAddDiv2(xi, PREEMPH_FAC, yi); + yi = SATURATE_LEFT_SHIFT(xi, 1, 32); + y[i] = yi; + } + *mem = yi; + return; +} + +/** + * \brief Compute the LP residual by filtering the input speech through the + * analysis filter A(z). + * \param[in] a LP filter coefficients, exponent = SF_A_COEFFS + * \param[in] x input signal (note that values x[-m..-1] are needed), exponent = + * SF_SYNTH + * \param[out] y output signal (residual), exponent = SF_EXC + * \param[in] l length of filtering + */ +/* static */ +void E_UTIL_residu(const FIXP_LPC *a, const INT a_exp, FIXP_DBL *x, FIXP_DBL *y, + INT l) { + FIXP_DBL s; + INT i, j; + + /* (note that values x[-m..-1] are needed) */ + for (i = 0; i < l; i++) { + s = (FIXP_DBL)0; + + for (j = 0; j < M_LP_FILTER_ORDER; j++) { + s += fMultDiv2(a[j], x[i - j - 1]); + } + + s = scaleValue(s, a_exp + 1); + y[i] = fAddSaturate(s, x[i]); + } + + return; +} + +/* use to map subfr number to number of bits used for acb_index */ +static const UCHAR num_acb_idx_bits_table[2][NB_SUBFR] = { + {9, 6, 9, 6}, /* coreCoderFrameLength == 1024 */ + {9, 6, 6, 0} /* coreCoderFrameLength == 768 */ +}; + +static int DecodePitchLag(HANDLE_FDK_BITSTREAM hBs, + const UCHAR num_acb_idx_bits, + const int PIT_MIN, /* TMIN */ + const int PIT_FR2, /* TFR2 */ + const int PIT_FR1, /* TFR1 */ + const int PIT_MAX, /* TMAX */ + int *pT0, int *pT0_frac, int *pT0_min, int *pT0_max) { + int acb_idx; + int error = 0; + int T0, T0_frac; + + FDK_ASSERT((num_acb_idx_bits == 9) || (num_acb_idx_bits == 6)); + + acb_idx = FDKreadBits(hBs, num_acb_idx_bits); + + if (num_acb_idx_bits == 6) { + /* When the pitch value is encoded on 6 bits, a pitch resolution of 1/4 is + always used in the range [T1-8, T1+7.75], where T1 is nearest integer to + the fractional pitch lag of the previous subframe. + */ + T0 = *pT0_min + acb_idx / 4; + T0_frac = acb_idx & 0x3; + } else { /* num_acb_idx_bits == 9 */ + /* When the pitch value is encoded on 9 bits, a fractional pitch delay is + used with resolutions 0.25 in the range [TMIN, TFR2-0.25], resolutions + 0.5 in the range [TFR2, TFR1-0.5], and integers only in the range [TFR1, + TMAX]. NOTE: for small sampling rates TMAX can get smaller than TFR1. + */ + int T0_min, T0_max; + + if (acb_idx < (PIT_FR2 - PIT_MIN) * 4) { + /* first interval with 0.25 pitch resolution */ + T0 = PIT_MIN + (acb_idx / 4); + T0_frac = acb_idx & 0x3; + } else if (acb_idx < ((PIT_FR2 - PIT_MIN) * 4 + (PIT_FR1 - PIT_FR2) * 2)) { + /* second interval with 0.5 pitch resolution */ + acb_idx -= (PIT_FR2 - PIT_MIN) * 4; + T0 = PIT_FR2 + (acb_idx / 2); + T0_frac = (acb_idx & 0x1) * 2; + } else { + /* third interval with 1.0 pitch resolution */ + T0 = acb_idx + PIT_FR1 - ((PIT_FR2 - PIT_MIN) * 4) - + ((PIT_FR1 - PIT_FR2) * 2); + T0_frac = 0; + } + /* find T0_min and T0_max for subframe 1 or 3 */ + T0_min = T0 - 8; + if (T0_min < PIT_MIN) { + T0_min = PIT_MIN; + } + T0_max = T0_min + 15; + if (T0_max > PIT_MAX) { + T0_max = PIT_MAX; + T0_min = T0_max - 15; + } + *pT0_min = T0_min; + *pT0_max = T0_max; + } + *pT0 = T0; + *pT0_frac = T0_frac; + + return error; +} +static void ConcealPitchLag(CAcelpStaticMem *acelp_mem, const int PIT_MAX, + int *pT0, int *pT0_frac) { + USHORT *pold_T0 = &acelp_mem->old_T0; + UCHAR *pold_T0_frac = &acelp_mem->old_T0_frac; + + if ((int)*pold_T0 >= PIT_MAX) { + *pold_T0 = (UCHAR)(PIT_MAX - 5); + } + *pT0 = (int)*pold_T0; + *pT0_frac = (int)*pold_T0_frac; +} + +static UCHAR tab_coremode2nbits[8] = {20, 28, 36, 44, 52, 64, 12, 16}; + +static int MapCoreMode2NBits(int core_mode) { + return (int)tab_coremode2nbits[core_mode]; +} + +void CLpd_AcelpDecode(CAcelpStaticMem *acelp_mem, INT i_offset, + const FIXP_LPC lsp_old[M_LP_FILTER_ORDER], + const FIXP_LPC lsp_new[M_LP_FILTER_ORDER], + FIXP_SGL stab_fac, CAcelpChannelData *pAcelpData, + INT numLostSubframes, int lastLpcLost, int frameCnt, + FIXP_DBL synth[], int pT[], FIXP_DBL *pit_gain, + INT coreCoderFrameLength) { + int i_subfr, subfr_nr, l_div, T; + int T0 = -1, T0_frac = -1; /* mark invalid */ + + int pit_gain_index = 0; + + const int PIT_MAX = PIT_MAX_12k8 + (6 * i_offset); /* maximum pitch lag */ + + FIXP_COD *code; + FIXP_DBL *exc2; + FIXP_DBL *syn; + FIXP_DBL *exc; + FIXP_LPC A[M_LP_FILTER_ORDER]; + INT A_exp; + + FIXP_DBL period_fac; + FIXP_SGL gain_pit; + FIXP_DBL gain_code, gain_code_smooth, Ener_code; + int Ener_code_e; + int n; + int bfi = (numLostSubframes > 0) ? 1 : 0; + + C_ALLOC_SCRATCH_START( + exc_buf, FIXP_DBL, + PIT_MAX_MAX + L_INTERPOL + L_DIV + 1); /* 411 + 17 + 256 + 1 = 685 */ + C_ALLOC_SCRATCH_START(syn_buf, FIXP_DBL, + M_LP_FILTER_ORDER + L_DIV); /* 16 + 256 = 272 */ + /* use same memory for code[L_SUBFR] and exc2[L_SUBFR] */ + C_ALLOC_SCRATCH_START(tmp_buf, FIXP_DBL, L_SUBFR); /* 64 */ + /* make sure they don't overlap if they are accessed alternatingly in + * BuildAdaptiveExcitation() */ +#if (COD_BITS == FRACT_BITS) + code = (FIXP_COD *)(tmp_buf + L_SUBFR / 2); +#elif (COD_BITS == DFRACT_BITS) + code = (FIXP_COD *)tmp_buf; +#endif + exc2 = (FIXP_DBL *)tmp_buf; + + syn = syn_buf + M_LP_FILTER_ORDER; + exc = exc_buf + PIT_MAX_MAX + L_INTERPOL; + + FDKmemcpy(syn_buf, acelp_mem->old_syn_mem, + M_LP_FILTER_ORDER * sizeof(FIXP_DBL)); + FDKmemcpy(exc_buf, acelp_mem->old_exc_mem, + (PIT_MAX_MAX + L_INTERPOL) * sizeof(FIXP_DBL)); + + FDKmemclear(exc_buf + (PIT_MAX_MAX + L_INTERPOL), + (L_DIV + 1) * sizeof(FIXP_DBL)); + + l_div = coreCoderFrameLength / NB_DIV; + + for (i_subfr = 0, subfr_nr = 0; i_subfr < l_div; + i_subfr += L_SUBFR, subfr_nr++) { + /*-------------------------------------------------* + * - Decode pitch lag (T0 and T0_frac) * + *-------------------------------------------------*/ + if (bfi) { + ConcealPitchLag(acelp_mem, PIT_MAX, &T0, &T0_frac); + } else { + T0 = (int)pAcelpData->T0[subfr_nr]; + T0_frac = (int)pAcelpData->T0_frac[subfr_nr]; + } + + /*-------------------------------------------------* + * - Find the pitch gain, the interpolation filter * + * and the adaptive codebook vector. * + *-------------------------------------------------*/ + Pred_lt4(&exc[i_subfr], T0, T0_frac); + + if ((!bfi && pAcelpData->ltp_filtering_flag[subfr_nr] == 0) || + (bfi && numLostSubframes == 1 && stab_fac < FL2FXCONST_SGL(0.25f))) { + /* find pitch excitation with lp filter: v'(n) => v(n) */ + Pred_lt4_postfilter(&exc[i_subfr]); + } + + /*-------------------------------------------------------* + * - Decode innovative codebook. * + * - Add the fixed-gain pitch contribution to code[]. * + *-------------------------------------------------------*/ + if (bfi) { + for (n = 0; n < L_SUBFR; n++) { + code[n] = + FX_SGL2FX_COD((FIXP_SGL)E_UTIL_random(&acelp_mem->seed_ace)) >> 4; + } + } else { + int nbits = MapCoreMode2NBits((int)pAcelpData->acelp_core_mode); + D_ACELP_decode_4t64(pAcelpData->icb_index[subfr_nr], nbits, &code[0]); + } + + T = T0; + if (T0_frac > 2) { + T += 1; + } + + Preemph_code(code); + Pit_shrp(code, T); + + /* Output pitch lag for bass post-filter */ + if (T > PIT_MAX) { + pT[subfr_nr] = PIT_MAX; + } else { + pT[subfr_nr] = T; + } + D_gain2_plus( + pAcelpData->gains[subfr_nr], + code, /* (i) : Innovative code vector, exponent = SF_CODE */ + &gain_pit, /* (o) : Quantized pitch gain, exponent = SF_GAIN_P */ + &gain_code, /* (o) : Quantized codebook gain */ + pAcelpData + ->mean_energy, /* (i) : mean_ener defined in open-loop (2 bits) */ + bfi, &acelp_mem->past_gpit, &acelp_mem->past_gcode, + &Ener_code, /* (o) : Innovative code vector energy */ + &Ener_code_e); /* (o) : Innovative code vector energy exponent */ + + pit_gain[pit_gain_index++] = FX_SGL2FX_DBL(gain_pit); + + /* calc periodicity factor r_v */ + period_fac = + calc_period_factor(/* (o) : factor (-1=unvoiced to 1=voiced) */ + &exc[i_subfr], /* (i) : pitch excitation, exponent = + SF_EXC */ + gain_pit, /* (i) : gain of pitch, exponent = + SF_GAIN_P */ + gain_code, /* (i) : gain of code */ + Ener_code, /* (i) : Energy of code[] */ + Ener_code_e); /* (i) : Exponent of energy of code[] + */ + + if (lastLpcLost && frameCnt == 0) { + if (gain_pit > FL2FXCONST_SGL(1.0f / (1 << SF_GAIN_P))) { + gain_pit = FL2FXCONST_SGL(1.0f / (1 << SF_GAIN_P)); + } + } + + gain_code_smooth = + noise_enhancer(/* (o) : smoothed gain g_sc exponent = SF_GAIN_C */ + gain_code, /* (i) : Quantized codebook gain */ + period_fac, /* (i) : periodicity factor (-1=unvoiced to + 1=voiced) */ + stab_fac, /* (i) : stability factor (0 <= ... < 1), + exponent = 1 */ + &acelp_mem->gc_threshold); + + /* Compute adaptive codebook update u'(n), pitch enhancement c'(n) and + * post-processed excitation u(n). */ + BuildAdaptiveExcitation(code, exc + i_subfr, gain_pit, gain_code, + gain_code_smooth, period_fac, exc2); + + /* Interpolate filter coeffs for current subframe in lsp domain and convert + * to LP domain */ + int_lpc_acelp(lsp_old, /* input : LSPs from past frame */ + lsp_new, /* input : LSPs from present frame */ + subfr_nr, /* input : ACELP subframe index */ + coreCoderFrameLength / L_DIV, + A, /* output: LP coefficients of this subframe */ + &A_exp); + + Syn_filt(A, /* (i) : a[m] prediction coefficients */ + A_exp, L_SUBFR, /* (i) : length */ + exc2, /* (i) : input signal */ + &syn[i_subfr] /* (i/o) : filter states / output signal */ + ); + + } /* end of subframe loop */ + + /* update pitch value for bfi procedure */ + acelp_mem->old_T0_frac = T0_frac; + acelp_mem->old_T0 = T0; + + /* save old excitation and old synthesis memory for next ACELP frame */ + FDKmemcpy(acelp_mem->old_exc_mem, exc + l_div - (PIT_MAX_MAX + L_INTERPOL), + sizeof(FIXP_DBL) * (PIT_MAX_MAX + L_INTERPOL)); + FDKmemcpy(acelp_mem->old_syn_mem, syn_buf + l_div, + sizeof(FIXP_DBL) * M_LP_FILTER_ORDER); + + Deemph(syn, synth, l_div, + &acelp_mem->de_emph_mem); /* ref soft: mem = synth[-1] */ + + scaleValues(synth, l_div, -ACELP_OUTSCALE); + acelp_mem->deemph_mem_wsyn = acelp_mem->de_emph_mem; + + C_ALLOC_SCRATCH_END(tmp_buf, FIXP_DBL, L_SUBFR); + C_ALLOC_SCRATCH_END(syn_buf, FIXP_DBL, M_LP_FILTER_ORDER + L_DIV); + C_ALLOC_SCRATCH_END(exc_buf, FIXP_DBL, PIT_MAX_MAX + L_INTERPOL + L_DIV + 1); + return; +} + +void CLpd_AcelpReset(CAcelpStaticMem *acelp) { + acelp->gc_threshold = (FIXP_DBL)0; + + acelp->past_gpit = (FIXP_SGL)0; + acelp->past_gcode = (FIXP_DBL)0; + acelp->old_T0 = 64; + acelp->old_T0_frac = 0; + acelp->deemph_mem_wsyn = (FIXP_DBL)0; + acelp->wsyn_rms = (FIXP_DBL)0; + acelp->seed_ace = 0; +} + +/* TCX time domain concealment */ +/* Compare to figure 13a on page 54 in 3GPP TS 26.290 */ +void CLpd_TcxTDConceal(CAcelpStaticMem *acelp_mem, SHORT *pitch, + const FIXP_LPC lsp_old[M_LP_FILTER_ORDER], + const FIXP_LPC lsp_new[M_LP_FILTER_ORDER], + const FIXP_SGL stab_fac, INT nLostSf, FIXP_DBL synth[], + INT coreCoderFrameLength, UCHAR last_tcx_noise_factor) { + /* repeat past excitation with pitch from previous decoded TCX frame */ + C_ALLOC_SCRATCH_START( + exc_buf, FIXP_DBL, + PIT_MAX_MAX + L_INTERPOL + L_DIV); /* 411 + 17 + 256 + 1 = */ + C_ALLOC_SCRATCH_START(syn_buf, FIXP_DBL, + M_LP_FILTER_ORDER + L_DIV); /* 256 + 16 = */ + /* += */ + FIXP_DBL ns_buf[L_DIV + 1]; + FIXP_DBL *syn = syn_buf + M_LP_FILTER_ORDER; + FIXP_DBL *exc = exc_buf + PIT_MAX_MAX + L_INTERPOL; + FIXP_DBL *ns = ns_buf + 1; + FIXP_DBL tmp, fact_exc; + INT T = fMin(*pitch, (SHORT)PIT_MAX_MAX); + int i, i_subfr, subfr_nr; + int lDiv = coreCoderFrameLength / NB_DIV; + + FDKmemcpy(syn_buf, acelp_mem->old_syn_mem, + M_LP_FILTER_ORDER * sizeof(FIXP_DBL)); + FDKmemcpy(exc_buf, acelp_mem->old_exc_mem, + (PIT_MAX_MAX + L_INTERPOL) * sizeof(FIXP_DBL)); + + /* if we lost all packets (i.e. 1 packet of TCX-20 ms, 2 packets of + the TCX-40 ms or 4 packets of the TCX-80ms), we lost the whole + coded frame extrapolation strategy: repeat lost excitation and + use extrapolated LSFs */ + + /* AMR-WB+ like TCX TD concealment */ + + /* number of lost frame cmpt */ + if (nLostSf < 2) { + fact_exc = FL2FXCONST_DBL(0.8f); + } else { + fact_exc = FL2FXCONST_DBL(0.4f); + } + + /* repeat past excitation */ + for (i = 0; i < lDiv; i++) { + exc[i] = fMult(fact_exc, exc[i - T]); + } + + tmp = fMult(fact_exc, acelp_mem->wsyn_rms); + acelp_mem->wsyn_rms = tmp; + + /* init deemph_mem_wsyn */ + acelp_mem->deemph_mem_wsyn = exc[-1]; + + ns[-1] = acelp_mem->deemph_mem_wsyn; + + for (i_subfr = 0, subfr_nr = 0; i_subfr < lDiv; + i_subfr += L_SUBFR, subfr_nr++) { + FIXP_DBL tRes[L_SUBFR]; + FIXP_LPC A[M_LP_FILTER_ORDER]; + INT A_exp; + + /* interpolate LPC coefficients */ + int_lpc_acelp(lsp_old, lsp_new, subfr_nr, lDiv / L_SUBFR, A, &A_exp); + + Syn_filt(A, /* (i) : a[m] prediction coefficients */ + A_exp, L_SUBFR, /* (i) : length */ + &exc[i_subfr], /* (i) : input signal */ + &syn[i_subfr] /* (i/o) : filter states / output signal */ + ); + + E_LPC_a_weight( + A, A, + M_LP_FILTER_ORDER); /* overwrite A as it is not needed any longer */ + + E_UTIL_residu(A, A_exp, &syn[i_subfr], tRes, L_SUBFR); + + Deemph(tRes, &ns[i_subfr], L_SUBFR, &acelp_mem->deemph_mem_wsyn); + + /* Amplitude limiter (saturate at wsyn_rms) */ + for (i = i_subfr; i < i_subfr + L_SUBFR; i++) { + if (ns[i] > tmp) { + ns[i] = tmp; + } else { + if (ns[i] < -tmp) { + ns[i] = -tmp; + } + } + } + + E_UTIL_preemph(&ns[i_subfr], tRes, L_SUBFR); + + Syn_filt(A, /* (i) : a[m] prediction coefficients */ + A_exp, L_SUBFR, /* (i) : length */ + tRes, /* (i) : input signal */ + &syn[i_subfr] /* (i/o) : filter states / output signal */ + ); + + FDKmemmove(&synth[i_subfr], &syn[i_subfr], L_SUBFR * sizeof(FIXP_DBL)); + } + + /* save old excitation and old synthesis memory for next ACELP frame */ + FDKmemcpy(acelp_mem->old_exc_mem, exc + lDiv - (PIT_MAX_MAX + L_INTERPOL), + sizeof(FIXP_DBL) * (PIT_MAX_MAX + L_INTERPOL)); + FDKmemcpy(acelp_mem->old_syn_mem, syn_buf + lDiv, + sizeof(FIXP_DBL) * M_LP_FILTER_ORDER); + acelp_mem->de_emph_mem = acelp_mem->deemph_mem_wsyn; + + C_ALLOC_SCRATCH_END(syn_buf, FIXP_DBL, M_LP_FILTER_ORDER + L_DIV); + C_ALLOC_SCRATCH_END(exc_buf, FIXP_DBL, PIT_MAX_MAX + L_INTERPOL + L_DIV); +} + +void Acelp_PreProcessing(FIXP_DBL *synth_buf, FIXP_DBL *old_synth, INT *pitch, + INT *old_T_pf, FIXP_DBL *pit_gain, + FIXP_DBL *old_gain_pf, INT samplingRate, INT *i_offset, + INT coreCoderFrameLength, INT synSfd, + INT nbSubfrSuperfr) { + int n; + + /* init beginning of synth_buf with old synthesis from previous frame */ + FDKmemcpy(synth_buf, old_synth, sizeof(FIXP_DBL) * (PIT_MAX_MAX - BPF_DELAY)); + + /* calculate pitch lag offset for ACELP decoder */ + *i_offset = + (samplingRate * PIT_MIN_12k8 + (FSCALE_DENOM / 2)) / FSCALE_DENOM - + PIT_MIN_12k8; + + /* for bass postfilter */ + for (n = 0; n < synSfd; n++) { + pitch[n] = old_T_pf[n]; + pit_gain[n] = old_gain_pf[n]; + } + for (n = 0; n < nbSubfrSuperfr; n++) { + pitch[n + synSfd] = L_SUBFR; + pit_gain[n + synSfd] = (FIXP_DBL)0; + } +} + +void Acelp_PostProcessing(FIXP_DBL *synth_buf, FIXP_DBL *old_synth, INT *pitch, + INT *old_T_pf, INT coreCoderFrameLength, INT synSfd, + INT nbSubfrSuperfr) { + int n; + + /* store last part of synth_buf (which is not handled by the IMDCT overlap) + * for next frame */ + FDKmemcpy(old_synth, synth_buf + coreCoderFrameLength, + sizeof(FIXP_DBL) * (PIT_MAX_MAX - BPF_DELAY)); + + /* for bass postfilter */ + for (n = 0; n < synSfd; n++) { + old_T_pf[n] = pitch[nbSubfrSuperfr + n]; + } +} + +#define L_FAC_ZIR (LFAC) + +void CLpd_Acelp_Zir(const FIXP_LPC A[], const INT A_exp, + CAcelpStaticMem *acelp_mem, const INT length, + FIXP_DBL zir[], int doDeemph) { + C_ALLOC_SCRATCH_START(tmp_buf, FIXP_DBL, L_FAC_ZIR + M_LP_FILTER_ORDER); + FDK_ASSERT(length <= L_FAC_ZIR); + + FDKmemcpy(tmp_buf, acelp_mem->old_syn_mem, + M_LP_FILTER_ORDER * sizeof(FIXP_DBL)); + FDKmemset(tmp_buf + M_LP_FILTER_ORDER, 0, L_FAC_ZIR * sizeof(FIXP_DBL)); + + Syn_filt(A, A_exp, length, &tmp_buf[M_LP_FILTER_ORDER], + &tmp_buf[M_LP_FILTER_ORDER]); + if (!doDeemph) { + /* if last lpd mode was TD concealment, then bypass deemph */ + FDKmemcpy(zir, tmp_buf, length * sizeof(*zir)); + } else { + Deemph(&tmp_buf[M_LP_FILTER_ORDER], &zir[0], length, + &acelp_mem->de_emph_mem); + scaleValues(zir, length, -ACELP_OUTSCALE); + } + C_ALLOC_SCRATCH_END(tmp_buf, FIXP_DBL, L_FAC_ZIR + M_LP_FILTER_ORDER); +} + +void CLpd_AcelpPrepareInternalMem(const FIXP_DBL *synth, UCHAR last_lpd_mode, + UCHAR last_last_lpd_mode, + const FIXP_LPC *A_new, const INT A_new_exp, + const FIXP_LPC *A_old, const INT A_old_exp, + CAcelpStaticMem *acelp_mem, + INT coreCoderFrameLength, UCHAR lpd_mode) { + int l_div = + coreCoderFrameLength / NB_DIV; /* length of one ACELP/TCX20 frame */ + int l_div_partial; + FIXP_DBL *syn, *old_exc_mem; + + C_ALLOC_SCRATCH_START(synth_buf, FIXP_DBL, + PIT_MAX_MAX + L_INTERPOL + M_LP_FILTER_ORDER); + syn = &synth_buf[M_LP_FILTER_ORDER]; + + l_div_partial = PIT_MAX_MAX + L_INTERPOL - l_div; + old_exc_mem = acelp_mem->old_exc_mem; + + if (lpd_mode == 4) { + /* Bypass Domain conversion. TCXTD Concealment does no deemphasis in the + * end. */ + FDKmemcpy( + synth_buf, &synth[-(PIT_MAX_MAX + L_INTERPOL + M_LP_FILTER_ORDER)], + (PIT_MAX_MAX + L_INTERPOL + M_LP_FILTER_ORDER) * sizeof(FIXP_DBL)); + /* Set deemphasis memory state for TD concealment */ + acelp_mem->deemph_mem_wsyn = scaleValueSaturate(synth[-1], ACELP_OUTSCALE); + } else { + /* convert past [PIT_MAX_MAX+L_INTERPOL+M_LP_FILTER_ORDER] synthesis to + * preemph domain */ + E_UTIL_preemph(&synth[-(PIT_MAX_MAX + L_INTERPOL + M_LP_FILTER_ORDER)], + synth_buf, PIT_MAX_MAX + L_INTERPOL + M_LP_FILTER_ORDER); + scaleValuesSaturate(synth_buf, PIT_MAX_MAX + L_INTERPOL + M_LP_FILTER_ORDER, + ACELP_OUTSCALE); + } + + /* Set deemphasis memory state */ + acelp_mem->de_emph_mem = scaleValueSaturate(synth[-1], ACELP_OUTSCALE); + + /* update acelp synth filter memory */ + FDKmemcpy(acelp_mem->old_syn_mem, + &syn[PIT_MAX_MAX + L_INTERPOL - M_LP_FILTER_ORDER], + M_LP_FILTER_ORDER * sizeof(FIXP_DBL)); + + /* update past [PIT_MAX_MAX+L_INTERPOL] samples of exc memory */ + if (last_lpd_mode == 1) { /* last frame was TCX20 */ + if (last_last_lpd_mode == 0) { /* ACELP -> TCX20 -> ACELP transition */ + /* Delay valid part of excitation buffer (from previous ACELP frame) by + * l_div samples */ + FDKmemmove(old_exc_mem, old_exc_mem + l_div, + sizeof(FIXP_DBL) * l_div_partial); + } else if (last_last_lpd_mode > 0) { /* TCX -> TCX20 -> ACELP transition */ + E_UTIL_residu(A_old, A_old_exp, syn, old_exc_mem, l_div_partial); + } + E_UTIL_residu(A_new, A_new_exp, syn + l_div_partial, + old_exc_mem + l_div_partial, l_div); + } else { /* prev frame was FD, TCX40 or TCX80 */ + int exc_A_new_length = (coreCoderFrameLength / 2 > PIT_MAX_MAX + L_INTERPOL) + ? PIT_MAX_MAX + L_INTERPOL + : coreCoderFrameLength / 2; + + int exc_A_old_length = PIT_MAX_MAX + L_INTERPOL - exc_A_new_length; + E_UTIL_residu(A_old, A_old_exp, syn, old_exc_mem, exc_A_old_length); + E_UTIL_residu(A_new, A_new_exp, &syn[exc_A_old_length], + &old_exc_mem[exc_A_old_length], exc_A_new_length); + } + C_ALLOC_SCRATCH_END(synth_buf, FIXP_DBL, + PIT_MAX_MAX + L_INTERPOL + M_LP_FILTER_ORDER); + + return; +} + +FIXP_DBL *CLpd_ACELP_GetFreeExcMem(CAcelpStaticMem *acelp_mem, INT length) { + FDK_ASSERT(length <= PIT_MAX_MAX + L_INTERPOL); + return acelp_mem->old_exc_mem; +} + +INT CLpd_AcelpRead(HANDLE_FDK_BITSTREAM hBs, CAcelpChannelData *acelp, + INT acelp_core_mode, INT coreCoderFrameLength, + INT i_offset) { + int nb_subfr = coreCoderFrameLength / L_DIV; + const UCHAR *num_acb_index_bits = + (nb_subfr == 4) ? num_acb_idx_bits_table[0] : num_acb_idx_bits_table[1]; + int nbits; + int error = 0; + + const int PIT_MIN = PIT_MIN_12k8 + i_offset; + const int PIT_FR2 = PIT_FR2_12k8 - i_offset; + const int PIT_FR1 = PIT_FR1_12k8; + const int PIT_MAX = PIT_MAX_12k8 + (6 * i_offset); + int T0, T0_frac, T0_min = 0, T0_max; + + if (PIT_MAX > PIT_MAX_MAX) { + error = AAC_DEC_DECODE_FRAME_ERROR; + goto bail; + } + + acelp->acelp_core_mode = acelp_core_mode; + + nbits = MapCoreMode2NBits(acelp_core_mode); + + /* decode mean energy with 2 bits : 18, 30, 42 or 54 dB */ + acelp->mean_energy = FDKreadBits(hBs, 2); + + for (int sfr = 0; sfr < nb_subfr; sfr++) { + /* read ACB index and store T0 and T0_frac for each ACELP subframe. */ + error = DecodePitchLag(hBs, num_acb_index_bits[sfr], PIT_MIN, PIT_FR2, + PIT_FR1, PIT_MAX, &T0, &T0_frac, &T0_min, &T0_max); + if (error) { + goto bail; + } + acelp->T0[sfr] = (USHORT)T0; + acelp->T0_frac[sfr] = (UCHAR)T0_frac; + acelp->ltp_filtering_flag[sfr] = FDKreadBits(hBs, 1); + switch (nbits) { + case 12: /* 12 bits AMR-WB codebook is used */ + acelp->icb_index[sfr][0] = FDKreadBits(hBs, 1); + acelp->icb_index[sfr][1] = FDKreadBits(hBs, 5); + acelp->icb_index[sfr][2] = FDKreadBits(hBs, 1); + acelp->icb_index[sfr][3] = FDKreadBits(hBs, 5); + break; + case 16: /* 16 bits AMR-WB codebook is used */ + acelp->icb_index[sfr][0] = FDKreadBits(hBs, 1); + acelp->icb_index[sfr][1] = FDKreadBits(hBs, 5); + acelp->icb_index[sfr][2] = FDKreadBits(hBs, 5); + acelp->icb_index[sfr][3] = FDKreadBits(hBs, 5); + break; + case 20: /* 20 bits AMR-WB codebook is used */ + acelp->icb_index[sfr][0] = FDKreadBits(hBs, 5); + acelp->icb_index[sfr][1] = FDKreadBits(hBs, 5); + acelp->icb_index[sfr][2] = FDKreadBits(hBs, 5); + acelp->icb_index[sfr][3] = FDKreadBits(hBs, 5); + break; + case 28: /* 28 bits AMR-WB codebook is used */ + acelp->icb_index[sfr][0] = FDKreadBits(hBs, 9); + acelp->icb_index[sfr][1] = FDKreadBits(hBs, 9); + acelp->icb_index[sfr][2] = FDKreadBits(hBs, 5); + acelp->icb_index[sfr][3] = FDKreadBits(hBs, 5); + break; + case 36: /* 36 bits AMR-WB codebook is used */ + acelp->icb_index[sfr][0] = FDKreadBits(hBs, 9); + acelp->icb_index[sfr][1] = FDKreadBits(hBs, 9); + acelp->icb_index[sfr][2] = FDKreadBits(hBs, 9); + acelp->icb_index[sfr][3] = FDKreadBits(hBs, 9); + break; + case 44: /* 44 bits AMR-WB codebook is used */ + acelp->icb_index[sfr][0] = FDKreadBits(hBs, 13); + acelp->icb_index[sfr][1] = FDKreadBits(hBs, 13); + acelp->icb_index[sfr][2] = FDKreadBits(hBs, 9); + acelp->icb_index[sfr][3] = FDKreadBits(hBs, 9); + break; + case 52: /* 52 bits AMR-WB codebook is used */ + acelp->icb_index[sfr][0] = FDKreadBits(hBs, 13); + acelp->icb_index[sfr][1] = FDKreadBits(hBs, 13); + acelp->icb_index[sfr][2] = FDKreadBits(hBs, 13); + acelp->icb_index[sfr][3] = FDKreadBits(hBs, 13); + break; + case 64: /* 64 bits AMR-WB codebook is used */ + acelp->icb_index[sfr][0] = FDKreadBits(hBs, 2); + acelp->icb_index[sfr][1] = FDKreadBits(hBs, 2); + acelp->icb_index[sfr][2] = FDKreadBits(hBs, 2); + acelp->icb_index[sfr][3] = FDKreadBits(hBs, 2); + acelp->icb_index[sfr][4] = FDKreadBits(hBs, 14); + acelp->icb_index[sfr][5] = FDKreadBits(hBs, 14); + acelp->icb_index[sfr][6] = FDKreadBits(hBs, 14); + acelp->icb_index[sfr][7] = FDKreadBits(hBs, 14); + break; + default: + FDK_ASSERT(0); + break; + } + acelp->gains[sfr] = FDKreadBits(hBs, 7); + } + +bail: + return error; +} |