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authorMatthias P. Braendli <matthias.braendli@mpb.li>2019-11-11 11:38:02 +0100
committerMatthias P. Braendli <matthias.braendli@mpb.li>2019-11-11 11:38:02 +0100
commit0e5af65c467b2423a0b857ae3ad98c91acc1e190 (patch)
treed07f69550d8886271e44fe79c4dcfb299cafbd38 /fdk-aac/libAACdec/src/usacdec_lpc.cpp
parentefe406d9724f959c8bc2a31802559ca6d41fd897 (diff)
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Include patched FDK-AAC in the repository
The initial idea was to get the DAB+ patch into upstream, but since that follows the android source releases, there is no place for a custom DAB+ patch there. So instead of having to maintain a patched fdk-aac that has to have the same .so version as the distribution package on which it is installed, we prefer having a separate fdk-aac-dab library to avoid collision. At that point, there's no reason to keep fdk-aac in a separate repository, as odr-audioenc is the only tool that needs DAB+ encoding support. Including it here simplifies installation, and makes it consistent with toolame-dab, also shipped in this repository. DAB+ decoding support (needed by ODR-SourceCompanion, dablin, etisnoop, welle.io and others) can be done using upstream FDK-AAC.
Diffstat (limited to 'fdk-aac/libAACdec/src/usacdec_lpc.cpp')
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+/* -----------------------------------------------------------------------------
+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, Manuel Jander
+
+ Description: USAC LPC/AVQ decode
+
+*******************************************************************************/
+
+#include "usacdec_lpc.h"
+
+#include "usacdec_rom.h"
+#include "FDK_trigFcts.h"
+
+#define NQ_MAX 36
+
+/*
+ * Helper functions.
+ */
+
+/**
+ * \brief Read unary code.
+ * \param hBs bitstream handle as data source.
+ * \return decoded value.
+ */
+static int get_vlclbf(HANDLE_FDK_BITSTREAM hBs) {
+ int result = 0;
+
+ while (FDKreadBits(hBs, 1) && result <= NQ_MAX) {
+ result++;
+ }
+ return result;
+}
+
+/**
+ * \brief Read bit count limited unary code.
+ * \param hBs bitstream handle as data source
+ * \param n max amount of bits to be read.
+ * \return decoded value.
+ */
+static int get_vlclbf_n(HANDLE_FDK_BITSTREAM hBs, int n) {
+ int result = 0;
+
+ while (FDKreadBits(hBs, 1)) {
+ result++;
+ n--;
+ if (n <= 0) {
+ break;
+ }
+ }
+
+ return result;
+}
+
+/*
+ * Algebraic Vector Quantizer
+ */
+
+/* ZF_SCALE must be greater than (number of FIXP_ZF)/2
+ because the loss of precision caused by fPow2Div2 in RE8_PPV() */
+//#define ZF_SCALE ((NQ_MAX-3)>>1)
+#define ZF_SCALE ((DFRACT_BITS / 2))
+#define FIXP_ZF FIXP_DBL
+#define INT2ZF(x, s) (FIXP_ZF)((x) << (ZF_SCALE - (s)))
+#define ZF2INT(x) (INT)((x) >> ZF_SCALE)
+
+/* 1.0 in ZF format format */
+#define ONEZF ((FIXP_ZF)INT2ZF(1, 0))
+
+/* static */
+void nearest_neighbor_2D8(FIXP_ZF x[8], int y[8]) {
+ FIXP_ZF s, em, e[8];
+ int i, j, sum;
+
+ /* round x into 2Z^8 i.e. compute y=(y1,...,y8) such that yi = 2[xi/2]
+ where [.] is the nearest integer operator
+ in the mean time, compute sum = y1+...+y8
+ */
+ sum = 0;
+ for (i = 0; i < 8; i++) {
+ FIXP_ZF tmp;
+ /* round to ..., -2, 0, 2, ... ([-1..1[ --> 0) */
+ if (x[i] < (FIXP_ZF)0) {
+ tmp = ONEZF - x[i];
+ y[i] = -2 * ((ZF2INT(tmp)) >> 1);
+ } else {
+ tmp = ONEZF + x[i];
+ y[i] = 2 * ((ZF2INT(tmp)) >> 1);
+ }
+ sum += y[i];
+ }
+ /* check if y1+...+y8 is a multiple of 4
+ if not, y is not round xj in the wrong way where j is defined by
+ j = arg max_i | xi -yi|
+ (this is called the Wagner rule)
+ */
+ if (sum % 4) {
+ /* find j = arg max_i | xi -yi| */
+ em = (FIXP_SGL)0;
+ j = 0;
+ for (i = 0; i < 8; i++) {
+ /* compute ei = xi-yi */
+ e[i] = x[i] - INT2ZF(y[i], 0);
+ }
+ for (i = 0; i < 8; i++) {
+ /* compute |ei| = | xi-yi | */
+ if (e[i] < (FIXP_ZF)0) {
+ s = -e[i];
+ } else {
+ s = e[i];
+ }
+ /* check if |ei| is maximal, if so, set j=i */
+ if (em < s) {
+ em = s;
+ j = i;
+ }
+ }
+ /* round xj in the "wrong way" */
+ if (e[j] < (FIXP_ZF)0) {
+ y[j] -= 2;
+ } else {
+ y[j] += 2;
+ }
+ }
+}
+
+/*--------------------------------------------------------------
+ RE8_PPV(x,y)
+ NEAREST NEIGHBOR SEARCH IN INFINITE LATTICE RE8
+ the algorithm is based on the definition of RE8 as
+ RE8 = (2D8) U (2D8+[1,1,1,1,1,1,1,1])
+ it applies the coset decoding of Sloane and Conway
+ (i) x: point in R^8 in 32-ZF_SCALE.ZF_SCALE format
+ (o) y: point in RE8 (8-dimensional integer vector)
+ --------------------------------------------------------------
+*/
+/* static */
+void RE8_PPV(FIXP_ZF x[], SHORT y[], int r) {
+ int i, y0[8], y1[8];
+ FIXP_ZF x1[8], tmp;
+ FIXP_DBL e;
+
+ /* find the nearest neighbor y0 of x in 2D8 */
+ nearest_neighbor_2D8(x, y0);
+ /* find the nearest neighbor y1 of x in 2D8+(1,...,1) (by coset decoding) */
+ for (i = 0; i < 8; i++) {
+ x1[i] = x[i] - ONEZF;
+ }
+ nearest_neighbor_2D8(x1, y1);
+ for (i = 0; i < 8; i++) {
+ y1[i] += 1;
+ }
+
+ /* compute e0=||x-y0||^2 and e1=||x-y1||^2 */
+ e = (FIXP_DBL)0;
+ for (i = 0; i < 8; i++) {
+ tmp = x[i] - INT2ZF(y0[i], 0);
+ e += fPow2Div2(
+ tmp << r); /* shift left to ensure that no fract part bits get lost. */
+ tmp = x[i] - INT2ZF(y1[i], 0);
+ e -= fPow2Div2(tmp << r);
+ }
+ /* select best candidate y0 or y1 to minimize distortion */
+ if (e < (FIXP_DBL)0) {
+ for (i = 0; i < 8; i++) {
+ y[i] = y0[i];
+ }
+ } else {
+ for (i = 0; i < 8; i++) {
+ y[i] = y1[i];
+ }
+ }
+}
+
+/* table look-up of unsigned value: find i where index >= table[i]
+ Note: range must be >= 2, index must be >= table[0] */
+static int table_lookup(const USHORT *table, unsigned int index, int range) {
+ int i;
+
+ for (i = 4; i < range; i += 4) {
+ if (index < table[i]) {
+ break;
+ }
+ }
+ if (i > range) {
+ i = range;
+ }
+
+ if (index < table[i - 2]) {
+ i -= 2;
+ }
+ if (index < table[i - 1]) {
+ i--;
+ }
+ i--;
+
+ return (i); /* index >= table[i] */
+}
+
+/*--------------------------------------------------------------------------
+ re8_decode_rank_of_permutation(rank, xs, x)
+ DECODING OF THE RANK OF THE PERMUTATION OF xs
+ (i) rank: index (rank) of a permutation
+ (i) xs: signed leader in RE8 (8-dimensional integer vector)
+ (o) x: point in RE8 (8-dimensional integer vector)
+ --------------------------------------------------------------------------
+ */
+static void re8_decode_rank_of_permutation(int rank, int *xs, SHORT x[8]) {
+ INT a[8], w[8], B, fac, fac_B, target;
+ int i, j;
+
+ /* --- pre-processing based on the signed leader xs ---
+ - compute the alphabet a=[a[0] ... a[q-1]] of x (q elements)
+ such that a[0]!=...!=a[q-1]
+ it is assumed that xs is sorted in the form of a signed leader
+ which can be summarized in 2 requirements:
+ a) |xs[0]| >= |xs[1]| >= |xs[2]| >= ... >= |xs[7]|
+ b) if |xs[i]|=|xs[i-1]|, xs[i]>=xs[i+1]
+ where |.| indicates the absolute value operator
+ - compute q (the number of symbols in the alphabet)
+ - compute w[0..q-1] where w[j] counts the number of occurences of
+ the symbol a[j] in xs
+ - compute B = prod_j=0..q-1 (w[j]!) where .! is the factorial */
+ /* xs[i], xs[i-1] and ptr_w/a*/
+ j = 0;
+ w[j] = 1;
+ a[j] = xs[0];
+ B = 1;
+ for (i = 1; i < 8; i++) {
+ if (xs[i] != xs[i - 1]) {
+ j++;
+ w[j] = 1;
+ a[j] = xs[i];
+ } else {
+ w[j]++;
+ B *= w[j];
+ }
+ }
+
+ /* --- actual rank decoding ---
+ the rank of x (where x is a permutation of xs) is based on
+ Schalkwijk's formula
+ it is given by rank=sum_{k=0..7} (A_k * fac_k/B_k)
+ the decoding of this rank is sequential and reconstructs x[0..7]
+ element by element from x[0] to x[7]
+ [the tricky part is the inference of A_k for each k...]
+ */
+
+ if (w[0] == 8) {
+ for (i = 0; i < 8; i++) {
+ x[i] = a[0]; /* avoid fac of 40320 */
+ }
+ } else {
+ target = rank * B;
+ fac_B = 1;
+ /* decode x element by element */
+ for (i = 0; i < 8; i++) {
+ fac = fac_B * fdk_dec_tab_factorial[i]; /* fac = 1..5040 */
+ j = -1;
+ do {
+ target -= w[++j] * fac;
+ } while (target >= 0); /* max of 30 tests / SV */
+ x[i] = a[j];
+ /* update rank, denominator B (B_k) and counter w[j] */
+ target += w[j] * fac; /* target = fac_B*B*rank */
+ fac_B *= w[j];
+ w[j]--;
+ }
+ }
+}
+
+/*--------------------------------------------------------------------------
+ re8_decode_base_index(n, I, y)
+ DECODING OF AN INDEX IN Qn (n=0,2,3 or 4)
+ (i) n: codebook number (*n is an integer defined in {0,2,3,4})
+ (i) I: index of c (pointer to unsigned 16-bit word)
+ (o) y: point in RE8 (8-dimensional integer vector)
+ note: the index I is defined as a 32-bit word, but only
+ 16 bits are required (long can be replaced by unsigned integer)
+ --------------------------------------------------------------------------
+ */
+static void re8_decode_base_index(int *n, UINT index, SHORT y[8]) {
+ int i, im, t, sign_code, ka, ks, rank, leader[8];
+
+ if (*n < 2) {
+ for (i = 0; i < 8; i++) {
+ y[i] = 0;
+ }
+ } else {
+ // index = (unsigned int)*I;
+ /* search for the identifier ka of the absolute leader (table-lookup)
+ Q2 is a subset of Q3 - the two cases are considered in the same branch
+ */
+ switch (*n) {
+ case 2:
+ case 3:
+ i = table_lookup(fdk_dec_I3, index, NB_LDQ3);
+ ka = fdk_dec_A3[i];
+ break;
+ case 4:
+ i = table_lookup(fdk_dec_I4, index, NB_LDQ4);
+ ka = fdk_dec_A4[i];
+ break;
+ default:
+ FDK_ASSERT(0);
+ return;
+ }
+ /* reconstruct the absolute leader */
+ for (i = 0; i < 8; i++) {
+ leader[i] = fdk_dec_Da[ka][i];
+ }
+ /* search for the identifier ks of the signed leader (table look-up)
+ (this search is focused based on the identifier ka of the absolute
+ leader)*/
+ t = fdk_dec_Ia[ka];
+ im = fdk_dec_Ns[ka];
+ ks = table_lookup(fdk_dec_Is + t, index, im);
+
+ /* reconstruct the signed leader from its sign code */
+ sign_code = 2 * fdk_dec_Ds[t + ks];
+ for (i = 7; i >= 0; i--) {
+ leader[i] *= (1 - (sign_code & 2));
+ sign_code >>= 1;
+ }
+
+ /* compute and decode the rank of the permutation */
+ rank = index - fdk_dec_Is[t + ks]; /* rank = index - cardinality offset */
+
+ re8_decode_rank_of_permutation(rank, leader, y);
+ }
+ return;
+}
+
+/* re8_y2k(y,m,k)
+ VORONOI INDEXING (INDEX DECODING) k -> y
+ (i) k: Voronoi index k[0..7]
+ (i) m: Voronoi modulo (m = 2^r = 1<<r, where r is integer >=2)
+ (i) r: Voronoi order (m = 2^r = 1<<r, where r is integer >=2)
+ (o) y: 8-dimensional point y[0..7] in RE8
+ */
+static void re8_k2y(int *k, int r, SHORT *y) {
+ int i, tmp, sum;
+ SHORT v[8];
+ FIXP_ZF zf[8];
+
+ FDK_ASSERT(r <= ZF_SCALE);
+
+ /* compute y = k M and z=(y-a)/m, where
+ M = [4 ]
+ [2 2 ]
+ [| \ ]
+ [2 2 ]
+ [1 1 _ 1 1]
+ a=(2,0,...,0)
+ m = 1<<r
+ */
+ for (i = 0; i < 8; i++) {
+ y[i] = k[7];
+ }
+ zf[7] = INT2ZF(y[7], r);
+ sum = 0;
+ for (i = 6; i >= 1; i--) {
+ tmp = 2 * k[i];
+ sum += tmp;
+ y[i] += tmp;
+ zf[i] = INT2ZF(y[i], r);
+ }
+ y[0] += (4 * k[0] + sum);
+ zf[0] = INT2ZF(y[0] - 2, r);
+ /* find nearest neighbor v of z in infinite RE8 */
+ RE8_PPV(zf, v, r);
+ /* compute y -= m v */
+ for (i = 0; i < 8; i++) {
+ y[i] -= (SHORT)(v[i] << r);
+ }
+}
+
+/*--------------------------------------------------------------------------
+ RE8_dec(n, I, k, y)
+ MULTI-RATE INDEXING OF A POINT y in THE LATTICE RE8 (INDEX DECODING)
+ (i) n: codebook number (*n is an integer defined in {0,2,3,4,..,n_max}). n_max
+ = 36 (i) I: index of c (pointer to unsigned 16-bit word) (i) k: index of v
+ (8-dimensional vector of binary indices) = Voronoi index (o) y: point in RE8
+ (8-dimensional integer vector) note: the index I is defined as a 32-bit word,
+ but only 16 bits are required (long can be replaced by unsigned integer)
+
+ return 0 on success, -1 on error.
+ --------------------------------------------------------------------------
+ */
+static int RE8_dec(int n, int I, int *k, FIXP_DBL *y) {
+ SHORT v[8];
+ SHORT _y[8];
+ UINT r;
+ int i;
+
+ /* Check bound of codebook qn */
+ if (n > NQ_MAX) {
+ return -1;
+ }
+
+ /* decode the sub-indices I and kv[] according to the codebook number n:
+ if n=0,2,3,4, decode I (no Voronoi extension)
+ if n>4, Voronoi extension is used, decode I and kv[] */
+ if (n <= 4) {
+ re8_decode_base_index(&n, I, _y);
+ for (i = 0; i < 8; i++) {
+ y[i] = (LONG)_y[i];
+ }
+ } else {
+ /* compute the Voronoi modulo m = 2^r where r is extension order */
+ r = ((n - 3) >> 1);
+
+ while (n > 4) {
+ n -= 2;
+ }
+ /* decode base codebook index I into c (c is an element of Q3 or Q4)
+ [here c is stored in y to save memory] */
+ re8_decode_base_index(&n, I, _y);
+ /* decode Voronoi index k[] into v */
+ re8_k2y(k, r, v);
+ /* reconstruct y as y = m c + v (with m=2^r, r integer >=1) */
+ for (i = 0; i < 8; i++) {
+ y[i] = (LONG)((_y[i] << r) + v[i]);
+ }
+ }
+ return 0;
+}
+
+/**************************/
+/* start LPC decode stuff */
+/**************************/
+//#define M 16
+#define FREQ_MAX 6400.0f
+#define FREQ_DIV 400.0f
+#define LSF_GAP 50.0f
+
+/**
+ * \brief calculate inverse weighting factor and add non-weighted residual
+ * LSF vector to first stage LSF approximation
+ * \param lsfq first stage LSF approximation values.
+ * \param xq weighted residual LSF vector
+ * \param nk_mode code book number coding mode.
+ */
+static void lsf_weight_2st(FIXP_LPC *lsfq, FIXP_DBL *xq, int nk_mode) {
+ FIXP_LPC d[M_LP_FILTER_ORDER + 1];
+ FIXP_SGL factor;
+ LONG w; /* inverse weight factor */
+ int i;
+
+ /* compute lsf distance */
+ d[0] = lsfq[0];
+ d[M_LP_FILTER_ORDER] =
+ FL2FXCONST_LPC(FREQ_MAX / (1 << LSF_SCALE)) - lsfq[M_LP_FILTER_ORDER - 1];
+ for (i = 1; i < M_LP_FILTER_ORDER; i++) {
+ d[i] = lsfq[i] - lsfq[i - 1];
+ }
+
+ switch (nk_mode) {
+ case 0:
+ factor = FL2FXCONST_SGL(2.0f * 60.0f / FREQ_DIV);
+ break; /* abs */
+ case 1:
+ factor = FL2FXCONST_SGL(2.0f * 65.0f / FREQ_DIV);
+ break; /* mid */
+ case 2:
+ factor = FL2FXCONST_SGL(2.0f * 64.0f / FREQ_DIV);
+ break; /* rel1 */
+ default:
+ factor = FL2FXCONST_SGL(2.0f * 63.0f / FREQ_DIV);
+ break; /* rel2 */
+ }
+ /* add non-weighted residual LSF vector to LSF1st */
+ for (i = 0; i < M_LP_FILTER_ORDER; i++) {
+ w = (LONG)fMultDiv2(factor, sqrtFixp(fMult(d[i], d[i + 1])));
+ lsfq[i] = fAddSaturate(lsfq[i], FX_DBL2FX_LPC((FIXP_DBL)(w * (LONG)xq[i])));
+ }
+
+ return;
+}
+
+/**
+ * \brief decode nqn amount of code book numbers. These values determine the
+ * amount of following bits for nqn AVQ RE8 vectors.
+ * \param nk_mode quantization mode.
+ * \param nqn amount code book number to read.
+ * \param qn pointer to output buffer to hold decoded code book numbers qn.
+ */
+static void decode_qn(HANDLE_FDK_BITSTREAM hBs, int nk_mode, int nqn,
+ int qn[]) {
+ int n;
+
+ if (nk_mode == 1) { /* nk mode 1 */
+ /* Unary code for mid LPC1/LPC3 */
+ /* Q0=0, Q2=10, Q3=110, ... */
+ for (n = 0; n < nqn; n++) {
+ qn[n] = get_vlclbf(hBs);
+ if (qn[n] > 0) {
+ qn[n]++;
+ }
+ }
+ } else { /* nk_mode 0, 3 and 2 */
+ /* 2 bits to specify Q2,Q3,Q4,ext */
+ for (n = 0; n < nqn; n++) {
+ qn[n] = 2 + FDKreadBits(hBs, 2);
+ }
+ if (nk_mode == 2) {
+ /* Unary code for rel LPC1/LPC3 */
+ /* Q0 = 0, Q5=10, Q6=110, ... */
+ for (n = 0; n < nqn; n++) {
+ if (qn[n] > 4) {
+ qn[n] = get_vlclbf(hBs);
+ if (qn[n] > 0) qn[n] += 4;
+ }
+ }
+ } else { /* nk_mode == (0 and 3) */
+ /* Unary code for abs and rel LPC0/LPC2 */
+ /* Q5 = 0, Q6=10, Q0=110, Q7=1110, ... */
+ for (n = 0; n < nqn; n++) {
+ if (qn[n] > 4) {
+ qn[n] = get_vlclbf(hBs);
+ switch (qn[n]) {
+ case 0:
+ qn[n] = 5;
+ break;
+ case 1:
+ qn[n] = 6;
+ break;
+ case 2:
+ qn[n] = 0;
+ break;
+ default:
+ qn[n] += 4;
+ break;
+ }
+ }
+ }
+ }
+ }
+}
+
+/**
+ * \brief reorder LSF coefficients to minimum distance.
+ * \param lsf pointer to buffer containing LSF coefficients and where reordered
+ * LSF coefficients will be stored into, scaled by LSF_SCALE.
+ * \param min_dist min distance scaled by LSF_SCALE
+ * \param n number of LSF/LSP coefficients.
+ */
+static void reorder_lsf(FIXP_LPC *lsf, FIXP_LPC min_dist, int n) {
+ FIXP_LPC lsf_min;
+ int i;
+
+ lsf_min = min_dist;
+ for (i = 0; i < n; i++) {
+ if (lsf[i] < lsf_min) {
+ lsf[i] = lsf_min;
+ }
+ lsf_min = fAddSaturate(lsf[i], min_dist);
+ }
+
+ /* reverse */
+ lsf_min = FL2FXCONST_LPC(FREQ_MAX / (1 << LSF_SCALE)) - min_dist;
+ for (i = n - 1; i >= 0; i--) {
+ if (lsf[i] > lsf_min) {
+ lsf[i] = lsf_min;
+ }
+
+ lsf_min = lsf[i] - min_dist;
+ }
+}
+
+/**
+ * \brief First stage approximation
+ * \param hBs bitstream handle as data source
+ * \param lsfq pointer to output buffer to hold LPC coefficients scaled by
+ * LSF_SCALE.
+ */
+static void vlpc_1st_dec(
+ HANDLE_FDK_BITSTREAM hBs, /* input: codebook index */
+ FIXP_LPC *lsfq /* i/o: i:prediction o:quantized lsf */
+) {
+ const FIXP_LPC *p_dico;
+ int i, index;
+
+ index = FDKreadBits(hBs, 8);
+ p_dico = &fdk_dec_dico_lsf_abs_8b[index * M_LP_FILTER_ORDER];
+ for (i = 0; i < M_LP_FILTER_ORDER; i++) {
+ lsfq[i] = p_dico[i];
+ }
+}
+
+/**
+ * \brief Do first stage approximation weighting and multiply with AVQ
+ * refinement.
+ * \param hBs bitstream handle data ssource.
+ * \param lsfq buffer holding 1st stage approx, 2nd stage approx is added to
+ * this values.
+ * \param nk_mode quantization mode.
+ * \return 0 on success, -1 on error.
+ */
+static int vlpc_2st_dec(
+ HANDLE_FDK_BITSTREAM hBs,
+ FIXP_LPC *lsfq, /* i/o: i:1st stage o:1st+2nd stage */
+ int nk_mode /* input: 0=abs, >0=rel */
+) {
+ int err;
+ FIXP_DBL xq[M_LP_FILTER_ORDER]; /* weighted residual LSF vector */
+
+ /* Decode AVQ refinement */
+ { err = CLpc_DecodeAVQ(hBs, xq, nk_mode, 2, 8); }
+ if (err != 0) {
+ return -1;
+ }
+
+ /* add non-weighted residual LSF vector to LSF1st */
+ lsf_weight_2st(lsfq, xq, nk_mode);
+
+ /* reorder */
+ reorder_lsf(lsfq, FL2FXCONST_LPC(LSF_GAP / (1 << LSF_SCALE)),
+ M_LP_FILTER_ORDER);
+
+ return 0;
+}
+
+/*
+ * Externally visible functions
+ */
+
+int CLpc_DecodeAVQ(HANDLE_FDK_BITSTREAM hBs, FIXP_DBL *pOutput, int nk_mode,
+ int no_qn, int length) {
+ int i, l;
+
+ for (i = 0; i < length; i += 8 * no_qn) {
+ int qn[2], nk, n, I;
+ int kv[8] = {0};
+
+ decode_qn(hBs, nk_mode, no_qn, qn);
+
+ for (l = 0; l < no_qn; l++) {
+ if (qn[l] == 0) {
+ FDKmemclear(&pOutput[i + l * 8], 8 * sizeof(FIXP_DBL));
+ }
+
+ /* Voronoi extension order ( nk ) */
+ nk = 0;
+ n = qn[l];
+ if (qn[l] > 4) {
+ nk = (qn[l] - 3) >> 1;
+ n = qn[l] - nk * 2;
+ }
+
+ /* Base codebook index, in reverse bit group order (!) */
+ I = FDKreadBits(hBs, 4 * n);
+
+ if (nk > 0) {
+ int j;
+
+ for (j = 0; j < 8; j++) {
+ kv[j] = FDKreadBits(hBs, nk);
+ }
+ }
+
+ if (RE8_dec(qn[l], I, kv, &pOutput[i + l * 8]) != 0) {
+ return -1;
+ }
+ }
+ }
+ return 0;
+}
+
+int CLpc_Read(HANDLE_FDK_BITSTREAM hBs, FIXP_LPC lsp[][M_LP_FILTER_ORDER],
+ FIXP_LPC lpc4_lsf[M_LP_FILTER_ORDER],
+ FIXP_LPC lsf_adaptive_mean_cand[M_LP_FILTER_ORDER],
+ FIXP_SGL pStability[], UCHAR *mod, int first_lpd_flag,
+ int last_lpc_lost, int last_frame_ok) {
+ int i, k, err;
+ int mode_lpc_bin = 0; /* mode_lpc bitstream representation */
+ int lpc_present[5] = {0, 0, 0, 0, 0};
+ int lpc0_available = 1;
+ int s = 0;
+ int l = 3;
+ const int nbDiv = NB_DIV;
+
+ lpc_present[4 >> s] = 1; /* LPC4 */
+
+ /* Decode LPC filters in the following order: LPC 4,0,2,1,3 */
+
+ /*** Decode LPC4 ***/
+ vlpc_1st_dec(hBs, lsp[4 >> s]);
+ err = vlpc_2st_dec(hBs, lsp[4 >> s], 0); /* nk_mode = 0 */
+ if (err != 0) {
+ return err;
+ }
+
+ /*** Decode LPC0 and LPC2 ***/
+ k = 0;
+ if (!first_lpd_flag) {
+ lpc_present[0] = 1;
+ lpc0_available = !last_lpc_lost;
+ /* old LPC4 is new LPC0 */
+ for (i = 0; i < M_LP_FILTER_ORDER; i++) {
+ lsp[0][i] = lpc4_lsf[i];
+ }
+ /* skip LPC0 and continue with LPC2 */
+ k = 2;
+ }
+
+ for (; k < l; k += 2) {
+ int nk_mode = 0;
+
+ if ((k == 2) && (mod[0] == 3)) {
+ break; /* skip LPC2 */
+ }
+
+ lpc_present[k >> s] = 1;
+
+ mode_lpc_bin = FDKreadBit(hBs);
+
+ if (mode_lpc_bin == 0) {
+ /* LPC0/LPC2: Abs */
+ vlpc_1st_dec(hBs, lsp[k >> s]);
+ } else {
+ /* LPC0/LPC2: RelR */
+ for (i = 0; i < M_LP_FILTER_ORDER; i++) {
+ lsp[k >> s][i] = lsp[4 >> s][i];
+ }
+ nk_mode = 3;
+ }
+
+ err = vlpc_2st_dec(hBs, lsp[k >> s], nk_mode);
+ if (err != 0) {
+ return err;
+ }
+ }
+
+ /*** Decode LPC1 ***/
+ if (mod[0] < 2) { /* else: skip LPC1 */
+ lpc_present[1] = 1;
+ mode_lpc_bin = get_vlclbf_n(hBs, 2);
+
+ switch (mode_lpc_bin) {
+ case 1:
+ /* LPC1: abs */
+ vlpc_1st_dec(hBs, lsp[1]);
+ err = vlpc_2st_dec(hBs, lsp[1], 0);
+ if (err != 0) {
+ return err;
+ }
+ break;
+ case 2:
+ /* LPC1: mid0 (no second stage AVQ quantizer in this case) */
+ if (lpc0_available) { /* LPC0/lsf[0] might be zero some times */
+ for (i = 0; i < M_LP_FILTER_ORDER; i++) {
+ lsp[1][i] = (lsp[0][i] >> 1) + (lsp[2][i] >> 1);
+ }
+ } else {
+ for (i = 0; i < M_LP_FILTER_ORDER; i++) {
+ lsp[1][i] = lsp[2][i];
+ }
+ }
+ break;
+ case 0:
+ /* LPC1: RelR */
+ for (i = 0; i < M_LP_FILTER_ORDER; i++) {
+ lsp[1][i] = lsp[2][i];
+ }
+ err = vlpc_2st_dec(hBs, lsp[1], 2 << s);
+ if (err != 0) {
+ return err;
+ }
+ break;
+ }
+ }
+
+ /*** Decode LPC3 ***/
+ if ((mod[2] < 2)) { /* else: skip LPC3 */
+ int nk_mode = 0;
+ lpc_present[3] = 1;
+
+ mode_lpc_bin = get_vlclbf_n(hBs, 3);
+
+ switch (mode_lpc_bin) {
+ case 1:
+ /* LPC3: abs */
+ vlpc_1st_dec(hBs, lsp[3]);
+ break;
+ case 0:
+ /* LPC3: mid */
+ for (i = 0; i < M_LP_FILTER_ORDER; i++) {
+ lsp[3][i] = (lsp[2][i] >> 1) + (lsp[4][i] >> 1);
+ }
+ nk_mode = 1;
+ break;
+ case 2:
+ /* LPC3: relL */
+ for (i = 0; i < M_LP_FILTER_ORDER; i++) {
+ lsp[3][i] = lsp[2][i];
+ }
+ nk_mode = 2;
+ break;
+ case 3:
+ /* LPC3: relR */
+ for (i = 0; i < M_LP_FILTER_ORDER; i++) {
+ lsp[3][i] = lsp[4][i];
+ }
+ nk_mode = 2;
+ break;
+ }
+ err = vlpc_2st_dec(hBs, lsp[3], nk_mode);
+ if (err != 0) {
+ return err;
+ }
+ }
+
+ if (!lpc0_available && !last_frame_ok) {
+ /* LPC(0) was lost. Use next available LPC(k) instead */
+ for (k = 1; k < (nbDiv + 1); k++) {
+ if (lpc_present[k]) {
+ for (i = 0; i < M_LP_FILTER_ORDER; i++) {
+#define LSF_INIT_TILT (0.25f)
+ if (mod[0] > 0) {
+ lsp[0][i] = FX_DBL2FX_LPC(
+ fMult(lsp[k][i], FL2FXCONST_SGL(1.0f - LSF_INIT_TILT)) +
+ fMult(fdk_dec_lsf_init[i], FL2FXCONST_SGL(LSF_INIT_TILT)));
+ } else {
+ lsp[0][i] = lsp[k][i];
+ }
+ }
+ break;
+ }
+ }
+ }
+
+ for (i = 0; i < M_LP_FILTER_ORDER; i++) {
+ lpc4_lsf[i] = lsp[4 >> s][i];
+ }
+
+ {
+ FIXP_DBL divFac;
+ int last, numLpc = 0;
+
+ i = nbDiv;
+ do {
+ numLpc += lpc_present[i--];
+ } while (i >= 0 && numLpc < 3);
+
+ last = i;
+
+ switch (numLpc) {
+ case 3:
+ divFac = FL2FXCONST_DBL(1.0f / 3.0f);
+ break;
+ case 2:
+ divFac = FL2FXCONST_DBL(1.0f / 2.0f);
+ break;
+ default:
+ divFac = FL2FXCONST_DBL(1.0f);
+ break;
+ }
+
+ /* get the adaptive mean for the next (bad) frame */
+ for (k = 0; k < M_LP_FILTER_ORDER; k++) {
+ FIXP_DBL tmp = (FIXP_DBL)0;
+ for (i = nbDiv; i > last; i--) {
+ if (lpc_present[i]) {
+ tmp = fMultAdd(tmp >> 1, lsp[i][k], divFac);
+ }
+ }
+ lsf_adaptive_mean_cand[k] = FX_DBL2FX_LPC(tmp);
+ }
+ }
+
+ /* calculate stability factor Theta. Needed for ACELP decoder and concealment
+ */
+ {
+ FIXP_LPC *lsf_prev, *lsf_curr;
+ k = 0;
+
+ FDK_ASSERT(lpc_present[0] == 1 && lpc_present[4 >> s] == 1);
+ lsf_prev = lsp[0];
+ for (i = 1; i < (nbDiv + 1); i++) {
+ if (lpc_present[i]) {
+ FIXP_DBL tmp = (FIXP_DBL)0;
+ int j;
+ lsf_curr = lsp[i];
+
+ /* sum = tmp * 2^(LSF_SCALE*2 + 4) */
+ for (j = 0; j < M_LP_FILTER_ORDER; j++) {
+ tmp += fPow2Div2((FIXP_SGL)(lsf_curr[j] - lsf_prev[j])) >> 3;
+ }
+
+ /* tmp = (float)(FL2FXCONST_DBL(1.25f) - fMult(tmp,
+ * FL2FXCONST_DBL(1/400000.0f))); */
+ tmp = FL2FXCONST_DBL(1.25f / (1 << LSF_SCALE)) -
+ fMult(tmp, FL2FXCONST_DBL((1 << (LSF_SCALE + 4)) / 400000.0f));
+ if (tmp >= FL2FXCONST_DBL(1.0f / (1 << LSF_SCALE))) {
+ pStability[k] = FL2FXCONST_SGL(1.0f / 2.0f);
+ } else if (tmp < FL2FXCONST_DBL(0.0f)) {
+ pStability[k] = FL2FXCONST_SGL(0.0f);
+ } else {
+ pStability[k] = FX_DBL2FX_SGL(tmp << (LSF_SCALE - 1));
+ }
+
+ lsf_prev = lsf_curr;
+ k = i;
+ } else {
+ /* Mark stability value as undefined. */
+ pStability[i] = (FIXP_SGL)-1;
+ }
+ }
+ }
+
+ /* convert into LSP domain */
+ for (i = 0; i < (nbDiv + 1); i++) {
+ if (lpc_present[i]) {
+ for (k = 0; k < M_LP_FILTER_ORDER; k++) {
+ lsp[i][k] = FX_DBL2FX_LPC(
+ fixp_cos(fMult(lsp[i][k],
+ FL2FXCONST_SGL((1 << LSPARG_SCALE) * M_PI / 6400.0)),
+ LSF_SCALE - LSPARG_SCALE));
+ }
+ }
+ }
+
+ return 0;
+}
+
+void CLpc_Conceal(FIXP_LPC lsp[][M_LP_FILTER_ORDER],
+ FIXP_LPC lpc4_lsf[M_LP_FILTER_ORDER],
+ FIXP_LPC lsf_adaptive_mean[M_LP_FILTER_ORDER],
+ const int first_lpd_flag) {
+ int i, j;
+
+#define BETA (FL2FXCONST_SGL(0.25f))
+#define ONE_BETA (FL2FXCONST_SGL(0.75f))
+#define BFI_FAC (FL2FXCONST_SGL(0.90f))
+#define ONE_BFI_FAC (FL2FXCONST_SGL(0.10f))
+
+ /* Frame loss concealment (could be improved) */
+
+ if (first_lpd_flag) {
+ /* Reset past LSF values */
+ for (i = 0; i < M_LP_FILTER_ORDER; i++) {
+ lsp[0][i] = lpc4_lsf[i] = fdk_dec_lsf_init[i];
+ }
+ } else {
+ /* old LPC4 is new LPC0 */
+ for (i = 0; i < M_LP_FILTER_ORDER; i++) {
+ lsp[0][i] = lpc4_lsf[i];
+ }
+ }
+
+ /* LPC1 */
+ for (i = 0; i < M_LP_FILTER_ORDER; i++) {
+ FIXP_LPC lsf_mean = FX_DBL2FX_LPC(fMult(BETA, fdk_dec_lsf_init[i]) +
+ fMult(ONE_BETA, lsf_adaptive_mean[i]));
+
+ lsp[1][i] = FX_DBL2FX_LPC(fMult(BFI_FAC, lpc4_lsf[i]) +
+ fMult(ONE_BFI_FAC, lsf_mean));
+ }
+
+ /* LPC2 - LPC4 */
+ for (j = 2; j <= 4; j++) {
+ for (i = 0; i < M_LP_FILTER_ORDER; i++) {
+ /* lsf_mean[i] = FX_DBL2FX_LPC(fMult((FIXP_LPC)(BETA + j *
+ FL2FXCONST_LPC(0.1f)), fdk_dec_lsf_init[i])
+ + fMult((FIXP_LPC)(ONE_BETA - j *
+ FL2FXCONST_LPC(0.1f)), lsf_adaptive_mean[i])); */
+
+ FIXP_LPC lsf_mean = FX_DBL2FX_LPC(
+ fMult((FIXP_SGL)(BETA + (FIXP_SGL)(j * (INT)FL2FXCONST_SGL(0.1f))),
+ (FIXP_SGL)fdk_dec_lsf_init[i]) +
+ fMult(
+ (FIXP_SGL)(ONE_BETA - (FIXP_SGL)(j * (INT)FL2FXCONST_SGL(0.1f))),
+ lsf_adaptive_mean[i]));
+
+ lsp[j][i] = FX_DBL2FX_LPC(fMult(BFI_FAC, lsp[j - 1][i]) +
+ fMult(ONE_BFI_FAC, lsf_mean));
+ }
+ }
+
+ /* Update past values for the future */
+ for (i = 0; i < M_LP_FILTER_ORDER; i++) {
+ lpc4_lsf[i] = lsp[4][i];
+ }
+
+ /* convert into LSP domain */
+ for (j = 0; j < 5; j++) {
+ for (i = 0; i < M_LP_FILTER_ORDER; i++) {
+ lsp[j][i] = FX_DBL2FX_LPC(fixp_cos(
+ fMult(lsp[j][i], FL2FXCONST_SGL((1 << LSPARG_SCALE) * M_PI / 6400.0)),
+ LSF_SCALE - LSPARG_SCALE));
+ }
+ }
+}
+
+void E_LPC_a_weight(FIXP_LPC *wA, const FIXP_LPC *A, int m) {
+ FIXP_DBL f;
+ int i;
+
+ f = FL2FXCONST_DBL(0.92f);
+ for (i = 0; i < m; i++) {
+ wA[i] = FX_DBL2FX_LPC(fMult(A[i], f));
+ f = fMult(f, FL2FXCONST_DBL(0.92f));
+ }
+}
+
+void CLpd_DecodeGain(FIXP_DBL *gain, INT *gain_e, int gain_code) {
+ /* gain * 2^(gain_e) = 10^(gain_code/28) */
+ *gain = fLdPow(
+ FL2FXCONST_DBL(3.3219280948873623478703194294894 / 4.0), /* log2(10)*/
+ 2,
+ fMultDiv2((FIXP_DBL)gain_code << (DFRACT_BITS - 1 - 7),
+ FL2FXCONST_DBL(2.0f / 28.0f)),
+ 7, gain_e);
+}
+
+ /**
+ * \brief * Find the polynomial F1(z) or F2(z) from the LSPs.
+ * This is performed by expanding the product polynomials:
+ *
+ * F1(z) = product ( 1 - 2 LSP_i z^-1 + z^-2 )
+ * i=0,2,4,6,8
+ * F2(z) = product ( 1 - 2 LSP_i z^-1 + z^-2 )
+ * i=1,3,5,7,9
+ *
+ * where LSP_i are the LSPs in the cosine domain.
+ * R.A.Salami October 1990
+ * \param lsp input, line spectral freq. (cosine domain)
+ * \param f output, the coefficients of F1 or F2, scaled by 8 bits
+ * \param n no of coefficients (m/2)
+ * \param flag 1 : F1(z) ; 2 : F2(z)
+ */
+
+#define SF_F 8
+
+static void get_lsppol(FIXP_LPC lsp[], FIXP_DBL f[], int n, int flag) {
+ FIXP_DBL b;
+ FIXP_LPC *plsp;
+ int i, j;
+
+ plsp = lsp + flag - 1;
+ f[0] = FL2FXCONST_DBL(1.0f / (1 << SF_F));
+ b = -FX_LPC2FX_DBL(*plsp);
+ f[1] = b >> (SF_F - 1);
+ for (i = 2; i <= n; i++) {
+ plsp += 2;
+ b = -FX_LPC2FX_DBL(*plsp);
+ f[i] = ((fMultDiv2(b, f[i - 1]) << 1) + (f[i - 2])) << 1;
+ for (j = i - 1; j > 1; j--) {
+ f[j] = f[j] + (fMultDiv2(b, f[j - 1]) << 2) + f[j - 2];
+ }
+ f[1] = f[1] + (b >> (SF_F - 1));
+ }
+ return;
+}
+
+#define NC M_LP_FILTER_ORDER / 2
+
+/**
+ * \brief lsp input LSP vector
+ * \brief a output LP filter coefficient vector scaled by SF_A_COEFFS.
+ */
+void E_LPC_f_lsp_a_conversion(FIXP_LPC *lsp, FIXP_LPC *a, INT *a_exp) {
+ FIXP_DBL f1[NC + 1], f2[NC + 1];
+ int i, k;
+
+ /*-----------------------------------------------------*
+ * Find the polynomials F1(z) and F2(z) *
+ *-----------------------------------------------------*/
+
+ get_lsppol(lsp, f1, NC, 1);
+ get_lsppol(lsp, f2, NC, 2);
+
+ /*-----------------------------------------------------*
+ * Multiply F1(z) by (1+z^-1) and F2(z) by (1-z^-1) *
+ *-----------------------------------------------------*/
+ for (i = NC; i > 0; i--) {
+ f1[i] += f1[i - 1];
+ f2[i] -= f2[i - 1];
+ }
+
+ FIXP_DBL aDBL[M_LP_FILTER_ORDER];
+
+ for (i = 1, k = M_LP_FILTER_ORDER - 1; i <= NC; i++, k--) {
+ FIXP_DBL tmp1, tmp2;
+
+ tmp1 = f1[i] >> 1;
+ tmp2 = f2[i] >> 1;
+
+ aDBL[i - 1] = (tmp1 + tmp2);
+ aDBL[k] = (tmp1 - tmp2);
+ }
+
+ int headroom_a = getScalefactor(aDBL, M_LP_FILTER_ORDER);
+
+ for (i = 0; i < M_LP_FILTER_ORDER; i++) {
+ a[i] = FX_DBL2FX_LPC(aDBL[i] << headroom_a);
+ }
+
+ *a_exp = 8 - headroom_a;
+}