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| author | Matthias P. Braendli <matthias.braendli@mpb.li> | 2019-11-11 11:38:02 +0100 |
|---|---|---|
| committer | Matthias P. Braendli <matthias.braendli@mpb.li> | 2019-11-11 11:38:02 +0100 |
| commit | 0e5af65c467b2423a0b857ae3ad98c91acc1e190 (patch) | |
| tree | d07f69550d8886271e44fe79c4dcfb299cafbd38 /fdk-aac/libSBRdec/src | |
| parent | efe406d9724f959c8bc2a31802559ca6d41fd897 (diff) | |
| download | ODR-AudioEnc-0e5af65c467b2423a0b857ae3ad98c91acc1e190.tar.gz ODR-AudioEnc-0e5af65c467b2423a0b857ae3ad98c91acc1e190.tar.bz2 ODR-AudioEnc-0e5af65c467b2423a0b857ae3ad98c91acc1e190.zip | |
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/libSBRdec/src')
40 files changed, 23743 insertions, 0 deletions
diff --git a/fdk-aac/libSBRdec/src/HFgen_preFlat.cpp b/fdk-aac/libSBRdec/src/HFgen_preFlat.cpp new file mode 100644 index 0000000..96adbb9 --- /dev/null +++ b/fdk-aac/libSBRdec/src/HFgen_preFlat.cpp @@ -0,0 +1,993 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): Oliver Moser, Manuel Jander, Matthias Hildenbrand + + Description: QMF frequency pre-whitening for SBR. + In the documentation the terms "scale factor" and "exponent" + mean the same. Variables containing such information have + the suffix "_sf". + +*******************************************************************************/ + +#include "HFgen_preFlat.h" + +#define POLY_ORDER 3 +#define MAXLOWBANDS 32 +#define LOG10FAC 0.752574989159953f /* == 10/log2(10) * 2^-2 */ +#define LOG10FAC_INV 0.664385618977472f /* == log2(10)/20 * 2^2 */ + +#define FIXP_CHB FIXP_SGL /* STB sinus Tab used in transformation */ +#define CHC(a) (FX_DBL2FXCONST_SGL(a)) +#define FX_CHB2FX_DBL(a) FX_SGL2FX_DBL(a) + +typedef struct backsubst_data { + FIXP_CHB Lnorm1d[3]; /*!< Normalized L matrix */ + SCHAR Lnorm1d_sf[3]; + FIXP_CHB Lnormii + [3]; /*!< The diagonal data points [i][i] of the normalized L matrix */ + SCHAR Lnormii_sf[3]; + FIXP_CHB Bmul0 + [4]; /*!< To normalize L*x=b, Bmul0 is what we need to multiply b with. */ + SCHAR Bmul0_sf[4]; + FIXP_CHB LnormInv1d[6]; /*!< Normalized inverted L matrix (L') */ + SCHAR LnormInv1d_sf[6]; + FIXP_CHB + Bmul1[4]; /*!< To normalize L'*x=b, Bmul1 is what we need to multiply b + with. */ + SCHAR Bmul1_sf[4]; +} backsubst_data; + +/* for each element n do, f(n) = trunc(log2(n))+1 */ +const UCHAR getLog2[32] = {0, 1, 2, 2, 3, 3, 3, 3, 4, 4, 4, 4, 4, 4, 4, 4, + 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5}; + +/** \def BSD_IDX_OFFSET + * + * bsd[] begins at index 0 with data for numBands=5. The correct bsd[] is + * indexed like bsd[numBands-BSD_IDX_OFFSET]. + */ +#define BSD_IDX_OFFSET 5 + +#define N_NUMBANDS \ + MAXLOWBANDS - BSD_IDX_OFFSET + \ + 1 /*!< Number of backsubst_data elements in bsd */ + +const backsubst_data bsd[N_NUMBANDS] = { + { + /* numBands=5 */ + {CHC(0x66c85a52), CHC(0x4278e587), CHC(0x697dcaff)}, + {-1, 0, 0}, + {CHC(0x66a61789), CHC(0x5253b8e3), CHC(0x5addad81)}, + {3, 4, 1}, + {CHC(0x7525ee90), CHC(0x6e2a1210), CHC(0x6523bb40), CHC(0x59822ead)}, + {-6, -4, -2, 0}, + {CHC(0x609e4cad), CHC(0x59c7e312), CHC(0x681eecac), CHC(0x440ea893), + CHC(0x4a214bb3), CHC(0x53c345a1)}, + {1, 0, -1, -1, -3, -5}, + {CHC(0x7525ee90), CHC(0x58587936), CHC(0x410d0b38), CHC(0x7f1519d6)}, + {-6, -1, 2, 0}, + }, + { + /* numBands=6 */ + {CHC(0x68943285), CHC(0x4841d2c3), CHC(0x6a6214c7)}, + {-1, 0, 0}, + {CHC(0x63c5923e), CHC(0x4e906e18), CHC(0x6285af8a)}, + {3, 4, 1}, + {CHC(0x7263940b), CHC(0x424a69a5), CHC(0x4ae8383a), CHC(0x517b7730)}, + {-7, -4, -2, 0}, + {CHC(0x518aee5f), CHC(0x4823a096), CHC(0x43764a39), CHC(0x6e6faf23), + CHC(0x61bba44f), CHC(0x59d8b132)}, + {1, 0, -1, -2, -4, -6}, + {CHC(0x7263940b), CHC(0x6757bff2), CHC(0x5bf40fe0), CHC(0x7d6f4292)}, + {-7, -2, 1, 0}, + }, + { + /* numBands=7 */ + {CHC(0x699b4c3c), CHC(0x4b8b702f), CHC(0x6ae51a4f)}, + {-1, 0, 0}, + {CHC(0x623a7f49), CHC(0x4ccc91fc), CHC(0x68f048dd)}, + {3, 4, 1}, + {CHC(0x7e6ebe18), CHC(0x5701daf2), CHC(0x74a8198b), CHC(0x4b399aa1)}, + {-8, -5, -3, 0}, + {CHC(0x464a64a6), CHC(0x78e42633), CHC(0x5ee174ba), CHC(0x5d0008c8), + CHC(0x455cff0f), CHC(0x6b9100e7)}, + {1, -1, -2, -2, -4, -7}, + {CHC(0x7e6ebe18), CHC(0x42c52efe), CHC(0x45fe401f), CHC(0x7b5808ef)}, + {-8, -2, 1, 0}, + }, + { + /* numBands=8 */ + {CHC(0x6a3fd9b4), CHC(0x4d99823f), CHC(0x6b372a94)}, + {-1, 0, 0}, + {CHC(0x614c6ef7), CHC(0x4bd06699), CHC(0x6e59cfca)}, + {3, 4, 1}, + {CHC(0x4c389cc5), CHC(0x79686681), CHC(0x5e2544c2), CHC(0x46305b43)}, + {-8, -6, -3, 0}, + {CHC(0x7b4ca7c6), CHC(0x68270ac5), CHC(0x467c644c), CHC(0x505c1b0f), + CHC(0x67a14778), CHC(0x45801767)}, + {0, -1, -2, -2, -5, -7}, + {CHC(0x4c389cc5), CHC(0x5c499ceb), CHC(0x6f863c9f), CHC(0x79059bfc)}, + {-8, -3, 0, 0}, + }, + { + /* numBands=9 */ + {CHC(0x6aad9988), CHC(0x4ef8ac18), CHC(0x6b6df116)}, + {-1, 0, 0}, + {CHC(0x60b159b0), CHC(0x4b33f772), CHC(0x72f5573d)}, + {3, 4, 1}, + {CHC(0x6206cb18), CHC(0x58a7d8dc), CHC(0x4e0b2d0b), CHC(0x4207ad84)}, + {-9, -6, -3, 0}, + {CHC(0x6dadadae), CHC(0x5b8b2cfc), CHC(0x6cf61db2), CHC(0x46c3c90b), + CHC(0x506314ea), CHC(0x5f034acd)}, + {0, -1, -3, -2, -5, -8}, + {CHC(0x6206cb18), CHC(0x42f8b8de), CHC(0x5bb4776f), CHC(0x769acc79)}, + {-9, -3, 0, 0}, + }, + { + /* numBands=10 */ + {CHC(0x6afa7252), CHC(0x4feed3ed), CHC(0x6b94504d)}, + {-1, 0, 0}, + {CHC(0x60467899), CHC(0x4acbafba), CHC(0x76eb327f)}, + {3, 4, 1}, + {CHC(0x42415b15), CHC(0x431080da), CHC(0x420f1c32), CHC(0x7d0c1aeb)}, + {-9, -6, -3, -1}, + {CHC(0x62b2c7a4), CHC(0x51b040a6), CHC(0x56caddb4), CHC(0x7e74a2c8), + CHC(0x4030adf5), CHC(0x43d1dc4f)}, + {0, -1, -3, -3, -5, -8}, + {CHC(0x42415b15), CHC(0x64e299b3), CHC(0x4d33b5e8), CHC(0x742cee5f)}, + {-9, -4, 0, 0}, + }, + { + /* numBands=11 */ + {CHC(0x6b3258bb), CHC(0x50a21233), CHC(0x6bb03c19)}, + {-1, 0, 0}, + {CHC(0x5ff997c6), CHC(0x4a82706e), CHC(0x7a5aae36)}, + {3, 4, 1}, + {CHC(0x5d2fb4fb), CHC(0x685bddd8), CHC(0x71b5e983), CHC(0x7708c90b)}, + {-10, -7, -4, -1}, + {CHC(0x59aceea2), CHC(0x49c428a0), CHC(0x46ca5527), CHC(0x724be884), + CHC(0x68e586da), CHC(0x643485b6)}, + {0, -1, -3, -3, -6, -9}, + {CHC(0x5d2fb4fb), CHC(0x4e3fad1a), CHC(0x42310ba2), CHC(0x71c8b3ce)}, + {-10, -4, 0, 0}, + }, + { + /* numBands=12 */ + {CHC(0x6b5c4726), CHC(0x5128a4a8), CHC(0x6bc52ee1)}, + {-1, 0, 0}, + {CHC(0x5fc06618), CHC(0x4a4ce559), CHC(0x7d5c16e9)}, + {3, 4, 1}, + {CHC(0x43af8342), CHC(0x531533d3), CHC(0x633660a6), CHC(0x71ce6052)}, + {-10, -7, -4, -1}, + {CHC(0x522373d7), CHC(0x434150cb), CHC(0x75b58afc), CHC(0x68474f2d), + CHC(0x575348a5), CHC(0x4c20973f)}, + {0, -1, -4, -3, -6, -9}, + {CHC(0x43af8342), CHC(0x7c4d3d11), CHC(0x732e13db), CHC(0x6f756ac4)}, + {-10, -5, -1, 0}, + }, + { + /* numBands=13 */ + {CHC(0x6b7c8953), CHC(0x51903fcd), CHC(0x6bd54d2e)}, + {-1, 0, 0}, + {CHC(0x5f94abf0), CHC(0x4a2480fa), CHC(0x40013553)}, + {3, 4, 2}, + {CHC(0x6501236e), CHC(0x436b9c4e), CHC(0x578d7881), CHC(0x6d34f92e)}, + {-11, -7, -4, -1}, + {CHC(0x4bc0e2b2), CHC(0x7b9d12ac), CHC(0x636c1c1b), CHC(0x5fe15c2b), + CHC(0x49d54879), CHC(0x7662cfa5)}, + {0, -2, -4, -3, -6, -10}, + {CHC(0x6501236e), CHC(0x64b059fe), CHC(0x656d8359), CHC(0x6d370900)}, + {-11, -5, -1, 0}, + }, + { + /* numBands=14 */ + {CHC(0x6b95e276), CHC(0x51e1b637), CHC(0x6be1f7ed)}, + {-1, 0, 0}, + {CHC(0x5f727a1c), CHC(0x4a053e9c), CHC(0x412e528c)}, + {3, 4, 2}, + {CHC(0x4d178bd4), CHC(0x6f33b4e8), CHC(0x4e028f7f), CHC(0x691ee104)}, + {-11, -8, -4, -1}, + {CHC(0x46473d3f), CHC(0x725bd0a6), CHC(0x55199885), CHC(0x58bcc56b), + CHC(0x7e7e6288), CHC(0x5ddef6eb)}, + {0, -2, -4, -3, -7, -10}, + {CHC(0x4d178bd4), CHC(0x52ebd467), CHC(0x5a395a6e), CHC(0x6b0f724f)}, + {-11, -5, -1, 0}, + }, + { + /* numBands=15 */ + {CHC(0x6baa2a22), CHC(0x5222eb91), CHC(0x6bec1a86)}, + {-1, 0, 0}, + {CHC(0x5f57393b), CHC(0x49ec8934), CHC(0x423b5b58)}, + {3, 4, 2}, + {CHC(0x77fd2486), CHC(0x5cfbdf2c), CHC(0x46153bd1), CHC(0x65757ed9)}, + {-12, -8, -4, -1}, + {CHC(0x41888ee6), CHC(0x6a661db3), CHC(0x49abc8c8), CHC(0x52965848), + CHC(0x6d9301b7), CHC(0x4bb04721)}, + {0, -2, -4, -3, -7, -10}, + {CHC(0x77fd2486), CHC(0x45424c68), CHC(0x50f33cc6), CHC(0x68ff43f0)}, + {-12, -5, -1, 0}, + }, + { + /* numBands=16 */ + {CHC(0x6bbaa499), CHC(0x5257ed94), CHC(0x6bf456e4)}, + {-1, 0, 0}, + {CHC(0x5f412594), CHC(0x49d8a766), CHC(0x432d1dbd)}, + {3, 4, 2}, + {CHC(0x5ef5cfde), CHC(0x4eafcd2d), CHC(0x7ed36893), CHC(0x62274b45)}, + {-12, -8, -5, -1}, + {CHC(0x7ac438f5), CHC(0x637aab21), CHC(0x4067617a), CHC(0x4d3c6ec7), + CHC(0x5fd6e0dd), CHC(0x7bd5f024)}, + {-1, -2, -4, -3, -7, -11}, + {CHC(0x5ef5cfde), CHC(0x751d0d4f), CHC(0x492b3c41), CHC(0x67065409)}, + {-12, -6, -1, 0}, + }, + { + /* numBands=17 */ + {CHC(0x6bc836c9), CHC(0x5283997e), CHC(0x6bfb1f5e)}, + {-1, 0, 0}, + {CHC(0x5f2f02b6), CHC(0x49c868e9), CHC(0x44078151)}, + {3, 4, 2}, + {CHC(0x4c43b65a), CHC(0x4349dcf6), CHC(0x73799e2d), CHC(0x5f267274)}, + {-12, -8, -5, -1}, + {CHC(0x73726394), CHC(0x5d68511a), CHC(0x7191bbcc), CHC(0x48898c70), + CHC(0x548956e1), CHC(0x66981ce8)}, + {-1, -2, -5, -3, -7, -11}, + {CHC(0x4c43b65a), CHC(0x64131116), CHC(0x429028e2), CHC(0x65240211)}, + {-12, -6, -1, 0}, + }, + { + /* numBands=18 */ + {CHC(0x6bd3860d), CHC(0x52a80156), CHC(0x6c00c68d)}, + {-1, 0, 0}, + {CHC(0x5f1fed86), CHC(0x49baf636), CHC(0x44cdb9dc)}, + {3, 4, 2}, + {CHC(0x7c189389), CHC(0x742666d8), CHC(0x69b8c776), CHC(0x5c67e27d)}, + {-13, -9, -5, -1}, + {CHC(0x6cf1ea76), CHC(0x58095703), CHC(0x64e351a9), CHC(0x4460da90), + CHC(0x4b1f8083), CHC(0x55f2d3e1)}, + {-1, -2, -5, -3, -7, -11}, + {CHC(0x7c189389), CHC(0x5651792a), CHC(0x79cb9b3d), CHC(0x635769c0)}, + {-13, -6, -2, 0}, + }, + { + /* numBands=19 */ + {CHC(0x6bdd0c40), CHC(0x52c6abf6), CHC(0x6c058950)}, + {-1, 0, 0}, + {CHC(0x5f133f88), CHC(0x49afb305), CHC(0x45826d73)}, + {3, 4, 2}, + {CHC(0x6621a164), CHC(0x6512528e), CHC(0x61449fc8), CHC(0x59e2a0c0)}, + {-13, -9, -5, -1}, + {CHC(0x6721cadb), CHC(0x53404cd4), CHC(0x5a389e91), CHC(0x40abcbd2), + CHC(0x43332f01), CHC(0x48b82e46)}, + {-1, -2, -5, -3, -7, -11}, + {CHC(0x6621a164), CHC(0x4b12cc28), CHC(0x6ffd4df8), CHC(0x619f835e)}, + {-13, -6, -2, 0}, + }, + { + /* numBands=20 */ + {CHC(0x6be524c5), CHC(0x52e0beb3), CHC(0x6c099552)}, + {-1, 0, 0}, + {CHC(0x5f087c68), CHC(0x49a62bb5), CHC(0x4627d175)}, + {3, 4, 2}, + {CHC(0x54ec6afe), CHC(0x58991a42), CHC(0x59e23e8c), CHC(0x578f4ef4)}, + {-13, -9, -5, -1}, + {CHC(0x61e78f6f), CHC(0x4ef5e1e9), CHC(0x5129c3b8), CHC(0x7ab0f7b2), + CHC(0x78efb076), CHC(0x7c2567ea)}, + {-1, -2, -5, -4, -8, -12}, + {CHC(0x54ec6afe), CHC(0x41c7812c), CHC(0x676f6f8d), CHC(0x5ffb383f)}, + {-13, -6, -2, 0}, + }, + { + /* numBands=21 */ + {CHC(0x6bec1542), CHC(0x52f71929), CHC(0x6c0d0d5e)}, + {-1, 0, 0}, + {CHC(0x5eff45c5), CHC(0x499e092d), CHC(0x46bfc0c9)}, + {3, 4, 2}, + {CHC(0x47457a78), CHC(0x4e2d99b3), CHC(0x53637ea5), CHC(0x5567d0e9)}, + {-13, -9, -5, -1}, + {CHC(0x5d2dc61b), CHC(0x4b1760c8), CHC(0x4967cf39), CHC(0x74b113d8), + CHC(0x6d6676b6), CHC(0x6ad114e9)}, + {-1, -2, -5, -4, -8, -12}, + {CHC(0x47457a78), CHC(0x740accaa), CHC(0x5feb6609), CHC(0x5e696f95)}, + {-13, -7, -2, 0}, + }, + { + /* numBands=22 */ + {CHC(0x6bf21387), CHC(0x530a683c), CHC(0x6c100c59)}, + {-1, 0, 0}, + {CHC(0x5ef752ea), CHC(0x499708c6), CHC(0x474bcd1b)}, + {3, 4, 2}, + {CHC(0x78a21ab7), CHC(0x45658aec), CHC(0x4da3c4fe), CHC(0x5367094b)}, + {-14, -9, -5, -1}, + {CHC(0x58e2df6a), CHC(0x4795990e), CHC(0x42b5e0f7), CHC(0x6f408c64), + CHC(0x6370bebf), CHC(0x5c91ca85)}, + {-1, -2, -5, -4, -8, -12}, + {CHC(0x78a21ab7), CHC(0x66f951d6), CHC(0x594605bb), CHC(0x5ce91657)}, + {-14, -7, -2, 0}, + }, + { + /* numBands=23 */ + {CHC(0x6bf749b2), CHC(0x531b3348), CHC(0x6c12a750)}, + {-1, 0, 0}, + {CHC(0x5ef06b17), CHC(0x4990f6c9), CHC(0x47cd4c5b)}, + {3, 4, 2}, + {CHC(0x66dede36), CHC(0x7bdf90a9), CHC(0x4885b2b9), CHC(0x5188a6b7)}, + {-14, -10, -5, -1}, + {CHC(0x54f85812), CHC(0x446414ae), CHC(0x79c8d519), CHC(0x6a4c2f31), + CHC(0x5ac8325f), CHC(0x50bf9200)}, + {-1, -2, -6, -4, -8, -12}, + {CHC(0x66dede36), CHC(0x5be0d90e), CHC(0x535cc453), CHC(0x5b7923f0)}, + {-14, -7, -2, 0}, + }, + { + /* numBands=24 */ + {CHC(0x6bfbd91d), CHC(0x5329e580), CHC(0x6c14eeed)}, + {-1, 0, 0}, + {CHC(0x5eea6179), CHC(0x498baa90), CHC(0x4845635d)}, + {3, 4, 2}, + {CHC(0x58559b7e), CHC(0x6f1b231f), CHC(0x43f1789b), CHC(0x4fc8fcb8)}, + {-14, -10, -5, -1}, + {CHC(0x51621775), CHC(0x417881a3), CHC(0x6f9ba9b6), CHC(0x65c412b2), + CHC(0x53352c61), CHC(0x46db9caf)}, + {-1, -2, -6, -4, -8, -12}, + {CHC(0x58559b7e), CHC(0x52636003), CHC(0x4e13b316), CHC(0x5a189cdf)}, + {-14, -7, -2, 0}, + }, + { + /* numBands=25 */ + {CHC(0x6bffdc73), CHC(0x5336d4af), CHC(0x6c16f084)}, + {-1, 0, 0}, + {CHC(0x5ee51249), CHC(0x498703cc), CHC(0x48b50e4f)}, + {3, 4, 2}, + {CHC(0x4c5616cf), CHC(0x641b9fad), CHC(0x7fa735e0), CHC(0x4e24e57a)}, + {-14, -10, -6, -1}, + {CHC(0x4e15f47a), CHC(0x7d9481d6), CHC(0x66a82f8a), CHC(0x619ae971), + CHC(0x4c8b2f5f), CHC(0x7d09ec11)}, + {-1, -3, -6, -4, -8, -13}, + {CHC(0x4c5616cf), CHC(0x4a3770fb), CHC(0x495402de), CHC(0x58c693fa)}, + {-14, -7, -2, 0}, + }, + { + /* numBands=26 */ + {CHC(0x6c036943), CHC(0x53424625), CHC(0x6c18b6dc)}, + {-1, 0, 0}, + {CHC(0x5ee060aa), CHC(0x4982e88a), CHC(0x491d277f)}, + {3, 4, 2}, + {CHC(0x425ada5b), CHC(0x5a9368ac), CHC(0x78380a42), CHC(0x4c99aa05)}, + {-14, -10, -6, -1}, + {CHC(0x4b0b569c), CHC(0x78a420da), CHC(0x5ebdf203), CHC(0x5dc57e63), + CHC(0x46a650ff), CHC(0x6ee13fb8)}, + {-1, -3, -6, -4, -8, -13}, + {CHC(0x425ada5b), CHC(0x4323073c), CHC(0x450ae92b), CHC(0x57822ad5)}, + {-14, -7, -2, 0}, + }, + { + /* numBands=27 */ + {CHC(0x6c06911a), CHC(0x534c7261), CHC(0x6c1a4aba)}, + {-1, 0, 0}, + {CHC(0x5edc3524), CHC(0x497f43c0), CHC(0x497e6cd8)}, + {3, 4, 2}, + {CHC(0x73fb550e), CHC(0x5244894f), CHC(0x717aad78), CHC(0x4b24ef6c)}, + {-15, -10, -6, -1}, + {CHC(0x483aebe4), CHC(0x74139116), CHC(0x57b58037), CHC(0x5a3a4f3c), + CHC(0x416950fe), CHC(0x62c7f4f2)}, + {-1, -3, -6, -4, -8, -13}, + {CHC(0x73fb550e), CHC(0x79efb994), CHC(0x4128cab7), CHC(0x564a919a)}, + {-15, -8, -2, 0}, + }, + { + /* numBands=28 */ + {CHC(0x6c096264), CHC(0x535587cd), CHC(0x6c1bb355)}, + {-1, 0, 0}, + {CHC(0x5ed87c76), CHC(0x497c0439), CHC(0x49d98452)}, + {3, 4, 2}, + {CHC(0x65dec5bf), CHC(0x4afd1ba3), CHC(0x6b58b4b3), CHC(0x49c4a7b0)}, + {-15, -10, -6, -1}, + {CHC(0x459e6eb1), CHC(0x6fd850b7), CHC(0x516e7be9), CHC(0x56f13d05), + CHC(0x79785594), CHC(0x58617de7)}, + {-1, -3, -6, -4, -9, -13}, + {CHC(0x65dec5bf), CHC(0x6f2168aa), CHC(0x7b41310f), CHC(0x551f0692)}, + {-15, -8, -3, 0}, + }, + { + /* numBands=29 */ + {CHC(0x6c0be913), CHC(0x535dacd5), CHC(0x6c1cf6a3)}, + {-1, 0, 0}, + {CHC(0x5ed526b4), CHC(0x49791bc5), CHC(0x4a2eff99)}, + {3, 4, 2}, + {CHC(0x59e44afe), CHC(0x44949ada), CHC(0x65bf36f5), CHC(0x487705a0)}, + {-15, -10, -6, -1}, + {CHC(0x43307779), CHC(0x6be959c4), CHC(0x4bce2122), CHC(0x53e34d89), + CHC(0x7115ff82), CHC(0x4f6421a1)}, + {-1, -3, -6, -4, -9, -13}, + {CHC(0x59e44afe), CHC(0x659eab7d), CHC(0x74cea459), CHC(0x53fed574)}, + {-15, -8, -3, 0}, + }, + { + /* numBands=30 */ + {CHC(0x6c0e2f17), CHC(0x53650181), CHC(0x6c1e199d)}, + {-1, 0, 0}, + {CHC(0x5ed2269f), CHC(0x49767e9e), CHC(0x4a7f5f0b)}, + {3, 4, 2}, + {CHC(0x4faa4ae6), CHC(0x7dd3bf11), CHC(0x609e2732), CHC(0x473a72e9)}, + {-15, -11, -6, -1}, + {CHC(0x40ec57c6), CHC(0x683ee147), CHC(0x46be261d), CHC(0x510a7983), + CHC(0x698a84cb), CHC(0x4794a927)}, + {-1, -3, -6, -4, -9, -13}, + {CHC(0x4faa4ae6), CHC(0x5d3615ad), CHC(0x6ee74773), CHC(0x52e956a1)}, + {-15, -8, -3, 0}, + }, + { + /* numBands=31 */ + {CHC(0x6c103cc9), CHC(0x536ba0ac), CHC(0x6c1f2070)}, + {-1, 0, 0}, + {CHC(0x5ecf711e), CHC(0x497422ea), CHC(0x4acb1438)}, + {3, 4, 2}, + {CHC(0x46e322ad), CHC(0x73c32f3c), CHC(0x5be7d172), CHC(0x460d8800)}, + {-15, -11, -6, -1}, + {CHC(0x7d9bf8ad), CHC(0x64d22351), CHC(0x422bdc81), CHC(0x4e6184aa), + CHC(0x62ba2375), CHC(0x40c325de)}, + {-2, -3, -6, -4, -9, -13}, + {CHC(0x46e322ad), CHC(0x55bef2a3), CHC(0x697b3135), CHC(0x51ddee4d)}, + {-15, -8, -3, 0}, + }, + { + // numBands=32 + {CHC(0x6c121933), CHC(0x5371a104), CHC(0x6c200ea0)}, + {-1, 0, 0}, + {CHC(0x5eccfcd3), CHC(0x49720060), CHC(0x4b1283f0)}, + {3, 4, 2}, + {CHC(0x7ea12a52), CHC(0x6aca3303), CHC(0x579072bf), CHC(0x44ef056e)}, + {-16, -11, -6, -1}, + {CHC(0x79a3a9ab), CHC(0x619d38fc), CHC(0x7c0f0734), CHC(0x4be3dd5d), + CHC(0x5c8d7163), CHC(0x7591065f)}, + {-2, -3, -7, -4, -9, -14}, + {CHC(0x7ea12a52), CHC(0x4f1782a6), CHC(0x647cbcb2), CHC(0x50dc0bb1)}, + {-16, -8, -3, 0}, + }, +}; + +/** \def SUM_SAFETY + * + * SUM_SAFTEY defines the bits needed to right-shift every summand in + * order to be overflow-safe. In the two backsubst functions we sum up 4 + * values. Since one of which is definitely not MAXVAL_DBL (the L[x][y]), + * we spare just 2 safety bits instead of 3. + */ +#define SUM_SAFETY 2 + +/** + * \brief Solves L*x=b via backsubstitution according to the following + * structure: + * + * x[0] = b[0]; + * x[1] = (b[1] - x[0]) / L[1][1]; + * x[2] = (b[2] - x[1]*L[2][1] - x[0]) / L[2][2]; + * x[3] = (b[3] - x[2]*L[3][2] - x[1]*L[3][1] - x[0]) / L[3][3]; + * + * \param[in] numBands SBR crossover band index + * \param[in] b the b in L*x=b (one-dimensional) + * \param[out] x output polynomial coefficients (mantissa) + * \param[out] x_sf exponents of x[] + */ +static void backsubst_fw(const int numBands, const FIXP_DBL *const b, + FIXP_DBL *RESTRICT x, int *RESTRICT x_sf) { + int i, k; + int m; /* the trip counter that indexes incrementally through Lnorm1d[] */ + + const FIXP_CHB *RESTRICT pLnorm1d = bsd[numBands - BSD_IDX_OFFSET].Lnorm1d; + const SCHAR *RESTRICT pLnorm1d_sf = bsd[numBands - BSD_IDX_OFFSET].Lnorm1d_sf; + const FIXP_CHB *RESTRICT pLnormii = bsd[numBands - BSD_IDX_OFFSET].Lnormii; + const SCHAR *RESTRICT pLnormii_sf = bsd[numBands - BSD_IDX_OFFSET].Lnormii_sf; + + x[0] = b[0]; + + for (i = 1, m = 0; i <= POLY_ORDER; ++i) { + FIXP_DBL sum = b[i] >> SUM_SAFETY; + int sum_sf = x_sf[i]; + for (k = i - 1; k > 0; --k, ++m) { + int e; + FIXP_DBL mult = fMultNorm(FX_CHB2FX_DBL(pLnorm1d[m]), x[k], &e); + int mult_sf = pLnorm1d_sf[m] + x_sf[k] + e; + + /* check if the new summand mult has a different sf than the sum currently + * has */ + int diff = mult_sf - sum_sf; + + if (diff > 0) { + /* yes, and it requires the sum to be adjusted (scaled down) */ + sum >>= diff; + sum_sf = mult_sf; + } else if (diff < 0) { + /* yes, but here mult needs to be scaled down */ + mult >>= -diff; + } + sum -= (mult >> SUM_SAFETY); + } + + /* - x[0] */ + if (x_sf[0] > sum_sf) { + sum >>= (x_sf[0] - sum_sf); + sum_sf = x_sf[0]; + } + sum -= (x[0] >> (sum_sf - x_sf[0] + SUM_SAFETY)); + + /* instead of the division /L[i][i], we multiply by the inverse */ + int e; + x[i] = fMultNorm(sum, FX_CHB2FX_DBL(pLnormii[i - 1]), &e); + x_sf[i] = sum_sf + pLnormii_sf[i - 1] + e + SUM_SAFETY; + } +} + +/** + * \brief Solves L*x=b via backsubstitution according to the following + * structure: + * + * x[3] = b[3]; + * x[2] = b[2] - L[2][3]*x[3]; + * x[1] = b[1] - L[1][2]*x[2] - L[1][3]*x[3]; + * x[0] = b[0] - L[0][1]*x[1] - L[0][2]*x[2] - L[0][3]*x[3]; + * + * \param[in] numBands SBR crossover band index + * \param[in] b the b in L*x=b (one-dimensional) + * \param[out] x solution vector + * \param[out] x_sf exponents of x[] + */ +static void backsubst_bw(const int numBands, const FIXP_DBL *const b, + FIXP_DBL *RESTRICT x, int *RESTRICT x_sf) { + int i, k; + int m; /* the trip counter that indexes incrementally through LnormInv1d[] */ + + const FIXP_CHB *RESTRICT pLnormInv1d = + bsd[numBands - BSD_IDX_OFFSET].LnormInv1d; + const SCHAR *RESTRICT pLnormInv1d_sf = + bsd[numBands - BSD_IDX_OFFSET].LnormInv1d_sf; + + x[POLY_ORDER] = b[POLY_ORDER]; + + for (i = POLY_ORDER - 1, m = 0; i >= 0; i--) { + FIXP_DBL sum = b[i] >> SUM_SAFETY; + int sum_sf = x_sf[i]; /* sum's sf but disregarding SUM_SAFETY (added at the + iteration's end) */ + + for (k = i + 1; k <= POLY_ORDER; ++k, ++m) { + int e; + FIXP_DBL mult = fMultNorm(FX_CHB2FX_DBL(pLnormInv1d[m]), x[k], &e); + int mult_sf = pLnormInv1d_sf[m] + x_sf[k] + e; + + /* check if the new summand mult has a different sf than sum currently has + */ + int diff = mult_sf - sum_sf; + + if (diff > 0) { + /* yes, and it requires the sum v to be adjusted (scaled down) */ + sum >>= diff; + sum_sf = mult_sf; + } else if (diff < 0) { + /* yes, but here mult needs to be scaled down */ + mult >>= -diff; + } + + /* mult has now the same sf than what it is about to be added to. */ + /* scale mult down additionally so that building the sum is overflow-safe. + */ + sum -= (mult >> SUM_SAFETY); + } + + x_sf[i] = sum_sf + SUM_SAFETY; + x[i] = sum; + } +} + +/** + * \brief Solves a system of linear equations (L*x=b) with the Cholesky + * algorithm. + * + * \param[in] numBands SBR crossover band index + * \param[in,out] b input: vector b, output: solution vector p. + * \param[in,out] b_sf input: exponent of b; output: exponent of solution + * p. + */ +static void choleskySolve(const int numBands, FIXP_DBL *RESTRICT b, + int *RESTRICT b_sf) { + int i, e; + + const FIXP_CHB *RESTRICT pBmul0 = bsd[numBands - BSD_IDX_OFFSET].Bmul0; + const SCHAR *RESTRICT pBmul0_sf = bsd[numBands - BSD_IDX_OFFSET].Bmul0_sf; + const FIXP_CHB *RESTRICT pBmul1 = bsd[numBands - BSD_IDX_OFFSET].Bmul1; + const SCHAR *RESTRICT pBmul1_sf = bsd[numBands - BSD_IDX_OFFSET].Bmul1_sf; + + /* normalize b */ + FIXP_DBL bnormed[POLY_ORDER + 1]; + for (i = 0; i <= POLY_ORDER; ++i) { + bnormed[i] = fMultNorm(b[i], FX_CHB2FX_DBL(pBmul0[i]), &e); + b_sf[i] += pBmul0_sf[i] + e; + } + + backsubst_fw(numBands, bnormed, b, b_sf); + + /* normalize b again */ + for (i = 0; i <= POLY_ORDER; ++i) { + bnormed[i] = fMultNorm(b[i], FX_CHB2FX_DBL(pBmul1[i]), &e); + b_sf[i] += pBmul1_sf[i] + e; + } + + backsubst_bw(numBands, bnormed, b, b_sf); +} + +/** + * \brief Find polynomial approximation of vector y with implicit abscisas + * x=0,1,2,3..n-1 + * + * The problem (V^T * V * p = V^T * y) is solved with Cholesky. + * V is the Vandermode Matrix constructed with x = 0...n-1; + * A = V^T * V; b = V^T * y; + * + * \param[in] numBands SBR crossover band index (BSD_IDX_OFFSET <= numBands <= + * MAXLOWBANDS) + * \param[in] y input vector (mantissa) + * \param[in] y_sf exponents of y[] + * \param[out] p output polynomial coefficients (mantissa) + * \param[out] p_sf exponents of p[] + */ +static void polyfit(const int numBands, const FIXP_DBL *const y, const int y_sf, + FIXP_DBL *RESTRICT p, int *RESTRICT p_sf) { + int i, k; + LONG v[POLY_ORDER + 1]; + int sum_saftey = getLog2[numBands - 1]; + + FDK_ASSERT((numBands >= BSD_IDX_OFFSET) && (numBands <= MAXLOWBANDS)); + + /* construct vector b[] temporarily stored in array p[] */ + FDKmemclear(p, (POLY_ORDER + 1) * sizeof(FIXP_DBL)); + + /* p[] are the sums over n values and each p[i] has its own sf */ + for (i = 0; i <= POLY_ORDER; ++i) p_sf[i] = 1 - DFRACT_BITS; + + for (k = 0; k < numBands; k++) { + v[0] = (LONG)1; + for (i = 1; i <= POLY_ORDER; i++) { + v[i] = k * v[i - 1]; + } + + for (i = 0; i <= POLY_ORDER; i++) { + if (v[POLY_ORDER - i] != 0 && y[k] != FIXP_DBL(0)) { + int e; + FIXP_DBL mult = fMultNorm((FIXP_DBL)v[POLY_ORDER - i], y[k], &e); + int sf = DFRACT_BITS - 1 + y_sf + e; + + /* check if the new summand has a different sf than the sum p[i] + * currently has */ + int diff = sf - p_sf[i]; + + if (diff > 0) { + /* yes, and it requires the sum p[i] to be adjusted (scaled down) */ + p[i] >>= fMin(DFRACT_BITS - 1, diff); + p_sf[i] = sf; + } else if (diff < 0) { + /* yes, but here mult needs to be scaled down */ + mult >>= -diff; + } + + /* mult has now the same sf than what it is about to be added to. + scale mult down additionally so that building the sum is + overflow-safe. */ + p[i] += mult >> sum_saftey; + } + } + } + + p_sf[0] += sum_saftey; + p_sf[1] += sum_saftey; + p_sf[2] += sum_saftey; + p_sf[3] += sum_saftey; + + choleskySolve(numBands, p, p_sf); +} + +/** + * \brief Calculates the output of a POLY_ORDER-degree polynomial function + * with Horner scheme: + * + * y(x) = p3 + p2*x + p1*x^2 + p0*x^3 + * = p3 + x*(p2 + x*(p1 + x*p0)) + * + * The for loop iterates through the mult/add parts in y(x) as above, + * during which regular upscaling ensures a stable exponent of the + * result. + * + * \param[in] p coefficients as in y(x) + * \param[in] p_sf exponents of p[] + * \param[in] x_int non-fractional integer representation of x as in y(x) + * \param[out] out_sf exponent of return value + * + * \return result y(x) + */ +static FIXP_DBL polyval(const FIXP_DBL *const p, const int *const p_sf, + const int x_int, int *out_sf) { + FDK_ASSERT(x_int <= 31); /* otherwise getLog2[] needs more elements */ + + int k, x_sf; + int result_sf; /* working space to compute return value *out_sf */ + FIXP_DBL x; /* fractional value of x_int */ + FIXP_DBL result; /* return value */ + + /* if x == 0, then y(x) is just p3 */ + if (x_int != 0) { + x_sf = getLog2[x_int]; + x = (FIXP_DBL)x_int << (DFRACT_BITS - 1 - x_sf); + } else { + *out_sf = p_sf[3]; + return p[3]; + } + + result = p[0]; + result_sf = p_sf[0]; + + for (k = 1; k <= POLY_ORDER; ++k) { + FIXP_DBL mult = fMult(x, result); + int mult_sf = x_sf + result_sf; + + int room = CountLeadingBits(mult); + mult <<= room; + mult_sf -= room; + + FIXP_DBL pp = p[k]; + int pp_sf = p_sf[k]; + + /* equalize the shift factors of pp and mult so that we can sum them up */ + int diff = pp_sf - mult_sf; + + if (diff > 0) { + diff = fMin(diff, DFRACT_BITS - 1); + mult >>= diff; + } else if (diff < 0) { + diff = fMax(diff, 1 - DFRACT_BITS); + pp >>= -diff; + } + + /* downshift by 1 to ensure safe summation */ + mult >>= 1; + mult_sf++; + pp >>= 1; + pp_sf++; + + result_sf = fMax(pp_sf, mult_sf); + + result = mult + pp; + /* rarely, mult and pp happen to be almost equal except their sign, + and then upon summation, result becomes so small, that it is within + the inaccuracy range of a few bits, and then the relative error + produced by this function may become HUGE */ + } + + *out_sf = result_sf; + return result; +} + +void sbrDecoder_calculateGainVec(FIXP_DBL **sourceBufferReal, + FIXP_DBL **sourceBufferImag, + int sourceBuf_e_overlap, + int sourceBuf_e_current, int overlap, + FIXP_DBL *RESTRICT GainVec, int *GainVec_exp, + int numBands, const int startSample, + const int stopSample) { + FIXP_DBL p[POLY_ORDER + 1]; + FIXP_DBL meanNrg; + FIXP_DBL LowEnv[MAXLOWBANDS]; + FIXP_DBL invNumBands = GetInvInt(numBands); + FIXP_DBL invNumSlots = GetInvInt(stopSample - startSample); + int i, loBand, exp, scale_nrg, scale_nrg_ov; + int sum_scale = 5, sum_scale_ov = 3; + + if (overlap > 8) { + FDK_ASSERT(overlap <= 16); + sum_scale_ov += 1; + sum_scale += 1; + } + + /* exponents of energy values */ + sourceBuf_e_overlap = sourceBuf_e_overlap * 2 + sum_scale_ov; + sourceBuf_e_current = sourceBuf_e_current * 2 + sum_scale; + exp = fMax(sourceBuf_e_overlap, sourceBuf_e_current); + scale_nrg = sourceBuf_e_current - exp; + scale_nrg_ov = sourceBuf_e_overlap - exp; + + meanNrg = (FIXP_DBL)0; + /* Calculate the spectral envelope in dB over the current copy-up frame. */ + for (loBand = 0; loBand < numBands; loBand++) { + FIXP_DBL nrg_ov, nrg; + INT reserve = 0, exp_new; + FIXP_DBL maxVal = FL2FX_DBL(0.0f); + + for (i = startSample; i < stopSample; i++) { + maxVal |= + (FIXP_DBL)((LONG)(sourceBufferReal[i][loBand]) ^ + ((LONG)sourceBufferReal[i][loBand] >> (SAMPLE_BITS - 1))); + maxVal |= + (FIXP_DBL)((LONG)(sourceBufferImag[i][loBand]) ^ + ((LONG)sourceBufferImag[i][loBand] >> (SAMPLE_BITS - 1))); + } + + if (maxVal != FL2FX_DBL(0.0f)) { + reserve = fixMax(0, CntLeadingZeros(maxVal) - 2); + } + + nrg_ov = nrg = (FIXP_DBL)0; + if (scale_nrg_ov > -31) { + for (i = startSample; i < overlap; i++) { + nrg_ov += (fPow2Div2(sourceBufferReal[i][loBand] << reserve) + + fPow2Div2(sourceBufferImag[i][loBand] << reserve)) >> + sum_scale_ov; + } + } else { + scale_nrg_ov = 0; + } + if (scale_nrg > -31) { + for (i = overlap; i < stopSample; i++) { + nrg += (fPow2Div2(sourceBufferReal[i][loBand] << reserve) + + fPow2Div2(sourceBufferImag[i][loBand] << reserve)) >> + sum_scale; + } + } else { + scale_nrg = 0; + } + + nrg = (scaleValue(nrg_ov, scale_nrg_ov) >> 1) + + (scaleValue(nrg, scale_nrg) >> 1); + nrg = fMult(nrg, invNumSlots); + + exp_new = + exp - (2 * reserve) + + 2; /* +1 for addition directly above, +1 for fPow2Div2 in loops above */ + + /* LowEnv = 10*log10(nrg) = log2(nrg) * 10/log2(10) */ + /* exponent of logarithmic energy is 8 */ + if (nrg > (FIXP_DBL)0) { + int exp_log2; + nrg = CalcLog2(nrg, exp_new, &exp_log2); + nrg = scaleValue(nrg, exp_log2 - 6); + nrg = fMult(FL2FXCONST_SGL(LOG10FAC), nrg); + } else { + nrg = (FIXP_DBL)0; + } + LowEnv[loBand] = nrg; + meanNrg += fMult(nrg, invNumBands); + } + exp = 6 + 2; /* exponent of LowEnv: +2 is exponent of LOG10FAC */ + + /* subtract mean before polynomial approximation to reduce dynamic of p[] */ + for (loBand = 0; loBand < numBands; loBand++) { + LowEnv[loBand] = meanNrg - LowEnv[loBand]; + } + + /* For numBands < BSD_IDX_OFFSET (== POLY_ORDER+2) we dont get an + overdetermined equation system. The calculated polynomial will exactly fit + the input data and evaluating the polynomial will lead to the same vector + than the original input vector: lowEnvSlope[] == lowEnv[] + */ + if (numBands > POLY_ORDER + 1) { + /* Find polynomial approximation of LowEnv */ + int p_sf[POLY_ORDER + 1]; + + polyfit(numBands, LowEnv, exp, p, p_sf); + + for (i = 0; i < numBands; i++) { + int sf; + + /* lowBandEnvSlope[i] = tmp; */ + FIXP_DBL tmp = polyval(p, p_sf, i, &sf); + + /* GainVec = 10^((mean(y)-y)/20) = 2^( (mean(y)-y) * log2(10)/20 ) */ + tmp = fMult(tmp, FL2FXCONST_SGL(LOG10FAC_INV)); + GainVec[i] = f2Pow(tmp, sf - 2, + &GainVec_exp[i]); /* -2 is exponent of LOG10FAC_INV */ + } + } else { /* numBands <= POLY_ORDER+1 */ + for (i = 0; i < numBands; i++) { + int sf = exp; /* exponent of LowEnv[] */ + + /* lowBandEnvSlope[i] = LowEnv[i]; */ + FIXP_DBL tmp = LowEnv[i]; + + /* GainVec = 10^((mean(y)-y)/20) = 2^( (mean(y)-y) * log2(10)/20 ) */ + tmp = fMult(tmp, FL2FXCONST_SGL(LOG10FAC_INV)); + GainVec[i] = f2Pow(tmp, sf - 2, + &GainVec_exp[i]); /* -2 is exponent of LOG10FAC_INV */ + } + } +} diff --git a/fdk-aac/libSBRdec/src/HFgen_preFlat.h b/fdk-aac/libSBRdec/src/HFgen_preFlat.h new file mode 100644 index 0000000..c1fc49d --- /dev/null +++ b/fdk-aac/libSBRdec/src/HFgen_preFlat.h @@ -0,0 +1,132 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): Manuel Jander, Matthias Hildenbrand + + Description: QMF frequency pre whitening for SBR + +*******************************************************************************/ + +#include "common_fix.h" + +#ifndef HFGEN_PREFLAT_H +#define HFGEN_PREFLAT_H + +#define GAIN_VEC_EXP 6 /* exponent of GainVec[] */ + +/** + * \brief Find gain vector to flatten the QMF frequency bands whithout loosing + * the fine structure. + * \param[in] sourceBufferReal real part of QMF domain data. + * \param[in] sourceBufferImag imaginary part of QMF domain data. + * \param[in] sourceBuffer_e_overlap exponent of sourceBufferReal. + * \param[in] sourceBuffer_e_current exponent of sourceBufferImag. + * \param[in] overlap number of overlap samples. + * \param[out] GainVec array of gain values (one for each QMF band). + * \param[out] GainVec_exp exponents of GainVec (one for each QMF band). + * \param[in] numBands number of low bands (k_0). + * \param[in] startSample time slot start. + * \param[in] stopSample time slot stop. + */ +void sbrDecoder_calculateGainVec(FIXP_DBL **sourceBufferReal, + FIXP_DBL **sourceBufferImag, + int sourceBuffer_e_overlap, + int sourceBuffer_e_current, int overlap, + FIXP_DBL GainVec[], int GainVec_exp[], + const int numBands, const int startSample, + const int stopSample); + +#endif /* __HFGEN_PREFLAT_H */ diff --git a/fdk-aac/libSBRdec/src/arm/lpp_tran_arm.cpp b/fdk-aac/libSBRdec/src/arm/lpp_tran_arm.cpp new file mode 100644 index 0000000..db1948f --- /dev/null +++ b/fdk-aac/libSBRdec/src/arm/lpp_tran_arm.cpp @@ -0,0 +1,159 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): Arthur Tritthart + + Description: (ARM optimised) LPP transposer subroutines + +*******************************************************************************/ + +#if defined(__arm__) + +#define FUNCTION_LPPTRANSPOSER_func1 + +#ifdef FUNCTION_LPPTRANSPOSER_func1 + +/* Note: This code requires only 43 cycles per iteration instead of 61 on + * ARM926EJ-S */ +static void lppTransposer_func1(FIXP_DBL *lowBandReal, FIXP_DBL *lowBandImag, + FIXP_DBL **qmfBufferReal, + FIXP_DBL **qmfBufferImag, int loops, int hiBand, + int dynamicScale, int descale, FIXP_SGL a0r, + FIXP_SGL a0i, FIXP_SGL a1r, FIXP_SGL a1i, + const int fPreWhitening, + FIXP_DBL preWhiteningGain, + int preWhiteningGains_sf) { + FIXP_DBL real1, real2, imag1, imag2, accu1, accu2; + + real2 = lowBandReal[-2]; + real1 = lowBandReal[-1]; + imag2 = lowBandImag[-2]; + imag1 = lowBandImag[-1]; + for (int i = 0; i < loops; i++) { + accu1 = fMultDiv2(a0r, real1); + accu2 = fMultDiv2(a0i, imag1); + accu1 = fMultAddDiv2(accu1, a1r, real2); + accu2 = fMultAddDiv2(accu2, a1i, imag2); + real2 = fMultDiv2(a1i, real2); + accu1 = accu1 - accu2; + accu1 = accu1 >> dynamicScale; + + accu2 = fMultAddDiv2(real2, a1r, imag2); + real2 = real1; + imag2 = imag1; + accu2 = fMultAddDiv2(accu2, a0i, real1); + real1 = lowBandReal[i]; + accu2 = fMultAddDiv2(accu2, a0r, imag1); + imag1 = lowBandImag[i]; + accu2 = accu2 >> dynamicScale; + + accu1 <<= 1; + accu2 <<= 1; + accu1 += (real1 >> descale); + accu2 += (imag1 >> descale); + if (fPreWhitening) { + accu1 = scaleValueSaturate(fMultDiv2(accu1, preWhiteningGain), + preWhiteningGains_sf); + accu2 = scaleValueSaturate(fMultDiv2(accu2, preWhiteningGain), + preWhiteningGains_sf); + } + qmfBufferReal[i][hiBand] = accu1; + qmfBufferImag[i][hiBand] = accu2; + } +} +#endif /* #ifdef FUNCTION_LPPTRANSPOSER_func1 */ + +#endif /* __arm__ */ diff --git a/fdk-aac/libSBRdec/src/env_calc.cpp b/fdk-aac/libSBRdec/src/env_calc.cpp new file mode 100644 index 0000000..cb1474f --- /dev/null +++ b/fdk-aac/libSBRdec/src/env_calc.cpp @@ -0,0 +1,3158 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief Envelope calculation + + The envelope adjustor compares the energies present in the transposed + highband to the reference energies conveyed with the bitstream. + The highband is amplified (sometimes) or attenuated (mostly) to the + desired level. + + The spectral shape of the reference energies can be changed several times per + frame if necessary. Each set of energy values corresponding to a certain range + in time will be called an <em>envelope</em> here. + The bitstream supports several frequency scales and two resolutions. Normally, + one or more QMF-subbands are grouped to one SBR-band. An envelope contains + reference energies for each SBR-band. + In addition to the energy envelopes, noise envelopes are transmitted that + define the ratio of energy which is generated by adding noise instead of + transposing the lowband. The noise envelopes are given in a coarser time + and frequency resolution. + If a signal contains strong tonal components, synthetic sines can be + generated in individual SBR bands. + + An overlap buffer of 6 QMF-timeslots is used to allow a more + flexible alignment of the envelopes in time that is not restricted to the + core codec's frame borders. + Therefore the envelope adjustor has access to the spectral data of the + current frame as well as the last 6 QMF-timeslots of the previous frame. + However, in average only the data of 1 frame is being processed as + the adjustor is called once per frame. + + Depending on the frequency range set in the bitstream, only QMF-subbands + between <em>lowSubband</em> and <em>highSubband</em> are adjusted. + + Scaling of spectral data to maximize SNR (see #QMF_SCALE_FACTOR) as well as a + special Mantissa-Exponent format ( see calculateSbrEnvelope() ) are being + used. The main entry point for this modules is calculateSbrEnvelope(). + + \sa sbr_scale.h, #QMF_SCALE_FACTOR, calculateSbrEnvelope(), \ref + documentationOverview +*/ + +#include "env_calc.h" + +#include "sbrdec_freq_sca.h" +#include "env_extr.h" +#include "transcendent.h" +#include "sbr_ram.h" +#include "sbr_rom.h" + +#include "genericStds.h" /* need FDKpow() for debug outputs */ + +typedef struct { + FIXP_DBL nrgRef[MAX_FREQ_COEFFS]; + FIXP_DBL nrgEst[MAX_FREQ_COEFFS]; + FIXP_DBL nrgGain[MAX_FREQ_COEFFS]; + FIXP_DBL noiseLevel[MAX_FREQ_COEFFS]; + FIXP_DBL nrgSine[MAX_FREQ_COEFFS]; + + SCHAR nrgRef_e[MAX_FREQ_COEFFS]; + SCHAR nrgEst_e[MAX_FREQ_COEFFS]; + SCHAR nrgGain_e[MAX_FREQ_COEFFS]; + SCHAR noiseLevel_e[MAX_FREQ_COEFFS]; + SCHAR nrgSine_e[MAX_FREQ_COEFFS]; + /* yet another exponent [0]: for ts < no_cols; [1]: for ts >= no_cols */ + SCHAR exponent[2]; +} ENV_CALC_NRGS; + +static void equalizeFiltBufferExp(FIXP_DBL *filtBuffer, SCHAR *filtBuffer_e, + FIXP_DBL *NrgGain, SCHAR *NrgGain_e, + int subbands); + +static void calcNrgPerSubband(FIXP_DBL **analysBufferReal, + FIXP_DBL **analysBufferImag, int lowSubband, + int highSubband, int start_pos, int next_pos, + SCHAR frameExp, FIXP_DBL *nrgEst, + SCHAR *nrgEst_e); + +static void calcNrgPerSfb(FIXP_DBL **analysBufferReal, + FIXP_DBL **analysBufferImag, int nSfb, + UCHAR *freqBandTable, int start_pos, int next_pos, + SCHAR input_e, FIXP_DBL *nrg_est, SCHAR *nrg_est_e); + +static void calcSubbandGain(FIXP_DBL nrgRef, SCHAR nrgRef_e, + ENV_CALC_NRGS *nrgs, int c, FIXP_DBL tmpNoise, + SCHAR tmpNoise_e, UCHAR sinePresentFlag, + UCHAR sineMapped, int noNoiseFlag); + +static void calcAvgGain(ENV_CALC_NRGS *nrgs, int lowSubband, int highSubband, + FIXP_DBL *sumRef_m, SCHAR *sumRef_e, + FIXP_DBL *ptrAvgGain_m, SCHAR *ptrAvgGain_e); + +static void adjustTimeSlot_EldGrid(FIXP_DBL *ptrReal, ENV_CALC_NRGS *nrgs, + UCHAR *ptrHarmIndex, int lowSubbands, + int noSubbands, int scale_change, + int noNoiseFlag, int *ptrPhaseIndex, + int scale_diff_low); + +static void adjustTimeSlotLC(FIXP_DBL *ptrReal, ENV_CALC_NRGS *nrgs, + UCHAR *ptrHarmIndex, int lowSubbands, + int noSubbands, int scale_change, int noNoiseFlag, + int *ptrPhaseIndex); + +/** + * \brief Variant of adjustTimeSlotHQ() which only regards gain and noise but no + * additional harmonics + */ +static void adjustTimeSlotHQ_GainAndNoise( + FIXP_DBL *ptrReal, FIXP_DBL *ptrImag, + HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, ENV_CALC_NRGS *nrgs, + int lowSubbands, int noSubbands, int scale_change, FIXP_SGL smooth_ratio, + int noNoiseFlag, int filtBufferNoiseShift); +/** + * \brief Variant of adjustTimeSlotHQ() which only adds the additional harmonics + */ +static void adjustTimeSlotHQ_AddHarmonics( + FIXP_DBL *ptrReal, FIXP_DBL *ptrImag, + HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, ENV_CALC_NRGS *nrgs, + int lowSubbands, int noSubbands, int scale_change); + +static void adjustTimeSlotHQ(FIXP_DBL *ptrReal, FIXP_DBL *ptrImag, + HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, + ENV_CALC_NRGS *nrgs, int lowSubbands, + int noSubbands, int scale_change, + FIXP_SGL smooth_ratio, int noNoiseFlag, + int filtBufferNoiseShift); + +/*! + \brief Map sine flags from bitstream to QMF bands + + The bitstream carries only 1 sine flag per band (Sfb) and frame. + This function maps every sine flag from the bitstream to a specific QMF + subband and to a specific envelope where the sine shall start. The result is + stored in the vector sineMapped which contains one entry per QMF subband. The + value of an entry specifies the envelope where a sine shall start. A value of + 32 indicates that no sine is present in the subband. The missing harmonics + flags from the previous frame (harmFlagsPrev) determine if a sine starts at + the beginning of the frame or at the transient position. Additionally, the + flags in harmFlagsPrev are being updated by this function for the next frame. +*/ +static void mapSineFlags( + UCHAR *freqBandTable, /*!< Band borders (there's only 1 flag per band) */ + int nSfb, /*!< Number of bands in the table */ + ULONG *addHarmonics, /*!< Packed addHarmonics of current frame (aligned to + the MSB) */ + ULONG *harmFlagsPrev, /*!< Packed addHarmonics of previous frame (aligned to + the LSB) */ + ULONG *harmFlagsPrevActive, /*!< Packed sineMapped of previous frame + (aligned to the LSB) */ + int tranEnv, /*!< Transient position */ + SCHAR *sineMapped) /*!< Resulting vector of sine start positions for each + QMF band */ + +{ + int i; + int bitcount = 31; + ULONG harmFlagsQmfBands[ADD_HARMONICS_FLAGS_SIZE] = {0}; + ULONG *curFlags = addHarmonics; + + /* + Format of addHarmonics (aligned to MSB): + + Up to MAX_FREQ_COEFFS sfb bands can be flagged for a sign. + first word = flags for lowest 32 sfb bands in use + second word = flags for higest 32 sfb bands (if present) + + Format of harmFlagsPrev (aligned to LSB): + + Index is absolute (not relative to lsb) so it is correct even if lsb + changes first word = flags for lowest 32 qmf bands (0...31) second word = + flags for next higher 32 qmf bands (32...63) + + */ + + /* Reset the output vector first */ + FDKmemset(sineMapped, 32, + MAX_FREQ_COEFFS * sizeof(SCHAR)); /* 32 means 'no sine' */ + FDKmemclear(harmFlagsPrevActive, ADD_HARMONICS_FLAGS_SIZE * sizeof(ULONG)); + for (i = 0; i < nSfb; i++) { + ULONG maskSfb = + 1 << bitcount; /* mask to extract addHarmonics flag of current Sfb */ + + if (*curFlags & maskSfb) { /* There is a sine in this band */ + const int lsb = freqBandTable[0]; /* start of sbr range */ + /* qmf band to which sine should be added */ + const int qmfBand = (freqBandTable[i] + freqBandTable[i + 1]) >> 1; + const int qmfBandDiv32 = qmfBand >> 5; + const int maskQmfBand = + 1 << (qmfBand & + 31); /* mask to extract harmonic flag from prevFlags */ + + /* mapping of sfb with sine to a certain qmf band -> for harmFlagsPrev */ + harmFlagsQmfBands[qmfBandDiv32] |= maskQmfBand; + + /* + If there was a sine in the last frame, let it continue from the first + envelope on else start at the transient position. Indexing of sineMapped + starts relative to lsb. + */ + sineMapped[qmfBand - lsb] = + (harmFlagsPrev[qmfBandDiv32] & maskQmfBand) ? 0 : tranEnv; + if (sineMapped[qmfBand - lsb] < PVC_NTIMESLOT) { + harmFlagsPrevActive[qmfBandDiv32] |= maskQmfBand; + } + } + + if (bitcount-- == 0) { + bitcount = 31; + curFlags++; + } + } + FDKmemcpy(harmFlagsPrev, harmFlagsQmfBands, + sizeof(ULONG) * ADD_HARMONICS_FLAGS_SIZE); +} + +/*! + \brief Restore sineMapped of previous frame + + For PVC it might happen that the PVC framing (always 0) is out of sync with + the SBR framing. The adding of additional harmonics is done based on the SBR + framing. If the SBR framing is trailing the PVC framing the sine mapping of + the previous SBR frame needs to be used for the overlapping time slots. +*/ +/*static*/ void mapSineFlagsPvc( + UCHAR *freqBandTable, /*!< Band borders (there's only 1 flag per + band) */ + int nSfb, /*!< Number of bands in the table */ + ULONG *harmFlagsPrev, /*!< Packed addHarmonics of previous frame + (aligned to the MSB) */ + ULONG *harmFlagsPrevActive, /*!< Packed sineMapped of previous + frame (aligned to the LSB) */ + SCHAR *sineMapped, /*!< Resulting vector of sine start positions + for each QMF band */ + int sinusoidalPos, /*!< sinusoidal position */ + SCHAR *sinusoidalPosPrev, /*!< sinusoidal position of previous + frame */ + int trailingSbrFrame) /*!< indication if the SBR framing is + trailing the PVC framing */ +{ + /* Reset the output vector first */ + FDKmemset(sineMapped, 32, MAX_FREQ_COEFFS); /* 32 means 'no sine' */ + + if (trailingSbrFrame) { + /* restore sineMapped[] of previous frame */ + int i; + const int lsb = freqBandTable[0]; + const int usb = freqBandTable[nSfb]; + for (i = lsb; i < usb; i++) { + const int qmfBandDiv32 = i >> 5; + const int maskQmfBand = + 1 << (i & 31); /* mask to extract harmonic flag from prevFlags */ + + /* Two cases need to be distinguished ... */ + if (harmFlagsPrevActive[qmfBandDiv32] & maskQmfBand) { + /* the sine mapping already started last PVC frame -> seamlessly + * continue */ + sineMapped[i - lsb] = 0; + } else if (harmFlagsPrev[qmfBandDiv32] & maskQmfBand) { + /* sinusoidalPos of prev PVC frame was >= PVC_NTIMESLOT -> sine starts + * in this frame */ + sineMapped[i - lsb] = + *sinusoidalPosPrev - PVC_NTIMESLOT; /* we are 16 sbr time slots + ahead of last frame now */ + } + } + } + *sinusoidalPosPrev = sinusoidalPos; +} + +/*! + \brief Reduce gain-adjustment induced aliasing for real valued filterbank. +*/ +/*static*/ void aliasingReduction( + FIXP_DBL *degreeAlias, /*!< estimated aliasing for each QMF + channel */ + ENV_CALC_NRGS *nrgs, + UCHAR *useAliasReduction, /*!< synthetic sine energy for each + subband, used as flag */ + int noSubbands) /*!< number of QMF channels to process */ +{ + FIXP_DBL *nrgGain = nrgs->nrgGain; /*!< subband gains to be modified */ + SCHAR *nrgGain_e = + nrgs->nrgGain_e; /*!< subband gains to be modified (exponents) */ + FIXP_DBL *nrgEst = nrgs->nrgEst; /*!< subband energy before amplification */ + SCHAR *nrgEst_e = + nrgs->nrgEst_e; /*!< subband energy before amplification (exponents) */ + int grouping = 0, index = 0, noGroups, k; + int groupVector[MAX_FREQ_COEFFS]; + + /* Calculate grouping*/ + for (k = 0; k < noSubbands - 1; k++) { + if ((degreeAlias[k + 1] != FL2FXCONST_DBL(0.0f)) && useAliasReduction[k]) { + if (grouping == 0) { + groupVector[index++] = k; + grouping = 1; + } else { + if (groupVector[index - 1] + 3 == k) { + groupVector[index++] = k + 1; + grouping = 0; + } + } + } else { + if (grouping) { + if (useAliasReduction[k]) + groupVector[index++] = k + 1; + else + groupVector[index++] = k; + grouping = 0; + } + } + } + + if (grouping) { + groupVector[index++] = noSubbands; + } + noGroups = index >> 1; + + /*Calculate new gain*/ + for (int group = 0; group < noGroups; group++) { + FIXP_DBL nrgOrig = FL2FXCONST_DBL( + 0.0f); /* Original signal energy in current group of bands */ + SCHAR nrgOrig_e = 0; + FIXP_DBL nrgAmp = FL2FXCONST_DBL( + 0.0f); /* Amplified signal energy in group (using current gains) */ + SCHAR nrgAmp_e = 0; + FIXP_DBL nrgMod = FL2FXCONST_DBL( + 0.0f); /* Signal energy in group when applying modified gains */ + SCHAR nrgMod_e = 0; + FIXP_DBL groupGain; /* Total energy gain in group */ + SCHAR groupGain_e; + FIXP_DBL compensation; /* Compensation factor for the energy change when + applying modified gains */ + SCHAR compensation_e; + + int startGroup = groupVector[2 * group]; + int stopGroup = groupVector[2 * group + 1]; + + /* Calculate total energy in group before and after amplification with + * current gains: */ + for (k = startGroup; k < stopGroup; k++) { + /* Get original band energy */ + FIXP_DBL tmp = nrgEst[k]; + SCHAR tmp_e = nrgEst_e[k]; + + FDK_add_MantExp(tmp, tmp_e, nrgOrig, nrgOrig_e, &nrgOrig, &nrgOrig_e); + + /* Multiply band energy with current gain */ + tmp = fMult(tmp, nrgGain[k]); + tmp_e = tmp_e + nrgGain_e[k]; + + FDK_add_MantExp(tmp, tmp_e, nrgAmp, nrgAmp_e, &nrgAmp, &nrgAmp_e); + } + + /* Calculate total energy gain in group */ + FDK_divide_MantExp(nrgAmp, nrgAmp_e, nrgOrig, nrgOrig_e, &groupGain, + &groupGain_e); + + for (k = startGroup; k < stopGroup; k++) { + FIXP_DBL tmp; + SCHAR tmp_e; + + FIXP_DBL alpha = degreeAlias[k]; + if (k < noSubbands - 1) { + if (degreeAlias[k + 1] > alpha) alpha = degreeAlias[k + 1]; + } + + /* Modify gain depending on the degree of aliasing */ + FDK_add_MantExp( + fMult(alpha, groupGain), groupGain_e, + fMult(/*FL2FXCONST_DBL(1.0f)*/ (FIXP_DBL)MAXVAL_DBL - alpha, + nrgGain[k]), + nrgGain_e[k], &nrgGain[k], &nrgGain_e[k]); + + /* Apply modified gain to original energy */ + tmp = fMult(nrgGain[k], nrgEst[k]); + tmp_e = nrgGain_e[k] + nrgEst_e[k]; + + /* Accumulate energy with modified gains applied */ + FDK_add_MantExp(tmp, tmp_e, nrgMod, nrgMod_e, &nrgMod, &nrgMod_e); + } + + /* Calculate compensation factor to retain the energy of the amplified + * signal */ + FDK_divide_MantExp(nrgAmp, nrgAmp_e, nrgMod, nrgMod_e, &compensation, + &compensation_e); + + /* Apply compensation factor to all gains of the group */ + for (k = startGroup; k < stopGroup; k++) { + nrgGain[k] = fMult(nrgGain[k], compensation); + nrgGain_e[k] = nrgGain_e[k] + compensation_e; + } + } +} + +#define INTER_TES_SF_CHANGE 3 + +typedef struct { + FIXP_DBL subsample_power_low[(((1024) / (32) * (4) / 2) + (3 * (4)))]; + FIXP_DBL subsample_power_high[(((1024) / (32) * (4) / 2) + (3 * (4)))]; + FIXP_DBL gain[(((1024) / (32) * (4) / 2) + (3 * (4)))]; + SCHAR subsample_power_low_sf[(((1024) / (32) * (4) / 2) + (3 * (4)))]; + SCHAR subsample_power_high_sf[(((1024) / (32) * (4) / 2) + (3 * (4)))]; +} ITES_TEMP; + +static void apply_inter_tes(FIXP_DBL **qmfReal, FIXP_DBL **qmfImag, + const QMF_SCALE_FACTOR *sbrScaleFactor, + const SCHAR exp[2], const int RATE, + const int startPos, const int stopPos, + const int lowSubband, const int nbSubband, + const UCHAR gamma_idx) { + int highSubband = lowSubband + nbSubband; + FIXP_DBL *subsample_power_high, *subsample_power_low; + SCHAR *subsample_power_high_sf, *subsample_power_low_sf; + FIXP_DBL total_power_high = (FIXP_DBL)0; + FIXP_DBL total_power_low = (FIXP_DBL)0; + FIXP_DBL *gain; + int gain_sf[(((1024) / (32) * (4) / 2) + (3 * (4)))]; + + /* gamma[gamma_idx] = {0.0f, 1.0f, 2.0f, 4.0f} */ + int gamma_sf = + (int)gamma_idx - 1; /* perhaps +1 to save one bit? (0.99999f vs 1.f) */ + + int nbSubsample = stopPos - startPos; + int i, j; + + C_ALLOC_SCRATCH_START(pTmp, ITES_TEMP, 1); + subsample_power_high = pTmp->subsample_power_high; + subsample_power_low = pTmp->subsample_power_low; + subsample_power_high_sf = pTmp->subsample_power_high_sf; + subsample_power_low_sf = pTmp->subsample_power_low_sf; + gain = pTmp->gain; + + if (gamma_idx > 0) { + int preShift2 = 32 - fNormz((FIXP_DBL)nbSubsample); + int total_power_low_sf = 1 - DFRACT_BITS; + int total_power_high_sf = 1 - DFRACT_BITS; + + for (i = 0; i < nbSubsample; ++i) { + FIXP_DBL bufferReal[(((1024) / (32) * (4) / 2) + (3 * (4)))]; + FIXP_DBL bufferImag[(((1024) / (32) * (4) / 2) + (3 * (4)))]; + FIXP_DBL maxVal = (FIXP_DBL)0; + + int ts = startPos + i; + + int low_sf = (ts < 3 * RATE) ? sbrScaleFactor->ov_lb_scale + : sbrScaleFactor->lb_scale; + low_sf = 15 - low_sf; + + for (j = 0; j < lowSubband; ++j) { + bufferImag[j] = qmfImag[startPos + i][j]; + maxVal |= (FIXP_DBL)((LONG)(bufferImag[j]) ^ + ((LONG)bufferImag[j] >> (DFRACT_BITS - 1))); + bufferReal[j] = qmfReal[startPos + i][j]; + maxVal |= (FIXP_DBL)((LONG)(bufferReal[j]) ^ + ((LONG)bufferReal[j] >> (DFRACT_BITS - 1))); + } + + subsample_power_low[i] = (FIXP_DBL)0; + subsample_power_low_sf[i] = 0; + + if (maxVal != FL2FXCONST_DBL(0.f)) { + /* multiply first, then shift for safe summation */ + int preShift = 1 - CntLeadingZeros(maxVal); + int postShift = 32 - fNormz((FIXP_DBL)lowSubband); + + /* reduce preShift because otherwise we risk to square -1.f */ + if (preShift != 0) preShift++; + + subsample_power_low_sf[i] += (low_sf + preShift) * 2 + postShift + 1; + + scaleValues(bufferReal, lowSubband, -preShift); + scaleValues(bufferImag, lowSubband, -preShift); + for (j = 0; j < lowSubband; ++j) { + FIXP_DBL addme; + addme = fPow2Div2(bufferReal[j]); + subsample_power_low[i] += addme >> postShift; + addme = fPow2Div2(bufferImag[j]); + subsample_power_low[i] += addme >> postShift; + } + } + + /* now get high */ + + maxVal = (FIXP_DBL)0; + + int high_sf = exp[(ts < 16 * RATE) ? 0 : 1]; + + for (j = lowSubband; j < highSubband; ++j) { + bufferImag[j] = qmfImag[startPos + i][j]; + maxVal |= (FIXP_DBL)((LONG)(bufferImag[j]) ^ + ((LONG)bufferImag[j] >> (DFRACT_BITS - 1))); + bufferReal[j] = qmfReal[startPos + i][j]; + maxVal |= (FIXP_DBL)((LONG)(bufferReal[j]) ^ + ((LONG)bufferReal[j] >> (DFRACT_BITS - 1))); + } + + subsample_power_high[i] = (FIXP_DBL)0; + subsample_power_high_sf[i] = 0; + + if (maxVal != FL2FXCONST_DBL(0.f)) { + int preShift = 1 - CntLeadingZeros(maxVal); + /* reduce preShift because otherwise we risk to square -1.f */ + if (preShift != 0) preShift++; + + int postShift = 32 - fNormz((FIXP_DBL)(highSubband - lowSubband)); + subsample_power_high_sf[i] += (high_sf + preShift) * 2 + postShift + 1; + + scaleValues(&bufferReal[lowSubband], highSubband - lowSubband, + -preShift); + scaleValues(&bufferImag[lowSubband], highSubband - lowSubband, + -preShift); + for (j = lowSubband; j < highSubband; j++) { + subsample_power_high[i] += fPow2Div2(bufferReal[j]) >> postShift; + subsample_power_high[i] += fPow2Div2(bufferImag[j]) >> postShift; + } + } + + /* sum all together */ + FIXP_DBL new_summand = subsample_power_low[i]; + int new_summand_sf = subsample_power_low_sf[i]; + + /* make sure the current sum, and the new summand have the same SF */ + if (new_summand_sf > total_power_low_sf) { + int diff = fMin(DFRACT_BITS - 1, new_summand_sf - total_power_low_sf); + total_power_low >>= diff; + total_power_low_sf = new_summand_sf; + } else if (new_summand_sf < total_power_low_sf) { + new_summand >>= + fMin(DFRACT_BITS - 1, total_power_low_sf - new_summand_sf); + } + + total_power_low += (new_summand >> preShift2); + + new_summand = subsample_power_high[i]; + new_summand_sf = subsample_power_high_sf[i]; + if (new_summand_sf > total_power_high_sf) { + total_power_high >>= + fMin(DFRACT_BITS - 1, new_summand_sf - total_power_high_sf); + total_power_high_sf = new_summand_sf; + } else if (new_summand_sf < total_power_high_sf) { + new_summand >>= + fMin(DFRACT_BITS - 1, total_power_high_sf - new_summand_sf); + } + + total_power_high += (new_summand >> preShift2); + } + + total_power_low_sf += preShift2; + total_power_high_sf += preShift2; + + /* gain[i] = e_LOW[i] */ + for (i = 0; i < nbSubsample; ++i) { + int sf2; + FIXP_DBL mult = + fMultNorm(subsample_power_low[i], (FIXP_DBL)nbSubsample, &sf2); + int mult_sf = subsample_power_low_sf[i] + DFRACT_BITS - 1 + sf2; + + if (total_power_low != FIXP_DBL(0)) { + gain[i] = fDivNorm(mult, total_power_low, &sf2); + gain_sf[i] = mult_sf - total_power_low_sf + sf2; + gain[i] = sqrtFixp_lookup(gain[i], &gain_sf[i]); + if (gain_sf[i] < 0) { + gain[i] >>= -gain_sf[i]; + gain_sf[i] = 0; + } + } else { + if (mult == FIXP_DBL(0)) { + gain[i] = FIXP_DBL(0); + gain_sf[i] = 0; + } else { + gain[i] = (FIXP_DBL)MAXVAL_DBL; + gain_sf[i] = 0; + } + } + } + + FIXP_DBL total_power_high_after = (FIXP_DBL)0; + int total_power_high_after_sf = 1 - DFRACT_BITS; + + /* gain[i] = g_inter[i] */ + for (i = 0; i < nbSubsample; ++i) { + if (gain_sf[i] < 0) { + gain[i] >>= -gain_sf[i]; + gain_sf[i] = 0; + } + + /* calculate: gain[i] = 1.0f + gamma * (gain[i] - 1.0f); */ + FIXP_DBL one = (FIXP_DBL)MAXVAL_DBL >> + gain_sf[i]; /* to substract this from gain[i] */ + + /* gamma is actually always 1 according to the table, so skip the + * fMultDiv2 */ + FIXP_DBL mult = (gain[i] - one) >> 1; + int mult_sf = gain_sf[i] + gamma_sf; + + one = FL2FXCONST_DBL(0.5f) >> mult_sf; + gain[i] = one + mult; + gain_sf[i] += gamma_sf + 1; /* +1 because of fMultDiv2() */ + + /* set gain to at least 0.2f */ + FIXP_DBL point_two = FL2FXCONST_DBL(0.8f); /* scaled up by 2 */ + int point_two_sf = -2; + + FIXP_DBL tmp = gain[i]; + if (point_two_sf < gain_sf[i]) { + point_two >>= gain_sf[i] - point_two_sf; + } else { + tmp >>= point_two_sf - gain_sf[i]; + } + + /* limit and calculate gain[i]^2 too */ + FIXP_DBL gain_pow2; + int gain_pow2_sf; + if (tmp < point_two) { + gain[i] = FL2FXCONST_DBL(0.8f); + gain_sf[i] = -2; + gain_pow2 = FL2FXCONST_DBL(0.64f); + gain_pow2_sf = -4; + } else { + /* this upscaling seems quite important */ + int r = CountLeadingBits(gain[i]); + gain[i] <<= r; + gain_sf[i] -= r; + + gain_pow2 = fPow2(gain[i]); + gain_pow2_sf = gain_sf[i] << 1; + } + + int room; + subsample_power_high[i] = + fMultNorm(subsample_power_high[i], gain_pow2, &room); + subsample_power_high_sf[i] = + subsample_power_high_sf[i] + gain_pow2_sf + room; + + int new_summand_sf = subsample_power_high_sf[i]; /* + gain_pow2_sf; */ + if (new_summand_sf > total_power_high_after_sf) { + total_power_high_after >>= + fMin(DFRACT_BITS - 1, new_summand_sf - total_power_high_after_sf); + total_power_high_after_sf = new_summand_sf; + } else if (new_summand_sf < total_power_high_after_sf) { + subsample_power_high[i] >>= total_power_high_after_sf - new_summand_sf; + } + total_power_high_after += subsample_power_high[i] >> preShift2; + } + + total_power_high_after_sf += preShift2; + + int sf2 = 0; + FIXP_DBL gain_adj_2 = FL2FX_DBL(0.5f); + int gain_adj_2_sf = 1; + + if ((total_power_high != (FIXP_DBL)0) && + (total_power_high_after != (FIXP_DBL)0)) { + gain_adj_2 = fDivNorm(total_power_high, total_power_high_after, &sf2); + gain_adj_2_sf = total_power_high_sf - total_power_high_after_sf + sf2; + } + + FIXP_DBL gain_adj = sqrtFixp_lookup(gain_adj_2, &gain_adj_2_sf); + int gain_adj_sf = gain_adj_2_sf; + + for (i = 0; i < nbSubsample; ++i) { + gain[i] = fMult(gain[i], gain_adj); + gain_sf[i] += gain_adj_sf; + + /* limit gain */ + if (gain_sf[i] > INTER_TES_SF_CHANGE) { + gain[i] = (FIXP_DBL)MAXVAL_DBL; + gain_sf[i] = INTER_TES_SF_CHANGE; + } + } + + for (i = 0; i < nbSubsample; ++i) { + /* equalize gain[]'s scale factors */ + gain[i] >>= INTER_TES_SF_CHANGE - gain_sf[i]; + + for (j = lowSubband; j < highSubband; j++) { + qmfReal[startPos + i][j] = fMult(qmfReal[startPos + i][j], gain[i]); + qmfImag[startPos + i][j] = fMult(qmfImag[startPos + i][j], gain[i]); + } + } + } else { /* gamma_idx == 0 */ + /* Inter-TES is not active. Still perform the scale change to have a + * consistent scaling for all envelopes of this frame. */ + for (i = 0; i < nbSubsample; ++i) { + for (j = lowSubband; j < highSubband; j++) { + qmfReal[startPos + i][j] >>= INTER_TES_SF_CHANGE; + qmfImag[startPos + i][j] >>= INTER_TES_SF_CHANGE; + } + } + } + C_ALLOC_SCRATCH_END(pTmp, ITES_TEMP, 1); +} + +/*! + \brief Apply spectral envelope to subband samples + + This function is called from sbr_dec.cpp in each frame. + + To enhance accuracy and due to the usage of tables for squareroots and + inverse, some calculations are performed with the operands being split + into mantissa and exponent. The variable names in the source code carry + the suffixes <em>_m</em> and <em>_e</em> respectively. The control data + in #hFrameData containts envelope data which is represented by this format but + stored in single words. (See requantizeEnvelopeData() for details). This data + is unpacked within calculateSbrEnvelope() to follow the described suffix + convention. + + The actual value (comparable to the corresponding float-variable in the + research-implementation) of a mantissa/exponent-pair can be calculated as + + \f$ value = value\_m * 2^{value\_e} \f$ + + All energies and noise levels decoded from the bitstream suit for an + original signal magnitude of \f$\pm 32768 \f$ rather than \f$ \pm 1\f$. + Therefore, the scale factor <em>hb_scale</em> passed into this function will + be converted to an 'input exponent' (#input_e), which fits the internal + representation. + + Before the actual processing, an exponent #adj_e for resulting adjusted + samples is derived from the maximum reference energy. + + Then, for each envelope, the following steps are performed: + + \li Calculate energy in the signal to be adjusted. Depending on the the value + of #interpolFreq (interpolation mode), this is either done seperately for each + QMF-subband or for each SBR-band. The resulting energies are stored in + #nrgEst_m[#MAX_FREQ_COEFFS] (mantissas) and #nrgEst_e[#MAX_FREQ_COEFFS] + (exponents). \li Calculate gain and noise level for each subband:<br> \f$ gain + = \sqrt{ \frac{nrgRef}{nrgEst} \cdot (1 - noiseRatio) } \hspace{2cm} noise = + \sqrt{ nrgRef \cdot noiseRatio } \f$<br> where <em>noiseRatio</em> and + <em>nrgRef</em> are extracted from the bitstream and <em>nrgEst</em> is the + subband energy before adjustment. The resulting gains are stored in + #nrgGain_m[#MAX_FREQ_COEFFS] (mantissas) and #nrgGain_e[#MAX_FREQ_COEFFS] + (exponents), the noise levels are stored in #noiseLevel_m[#MAX_FREQ_COEFFS] + and #noiseLevel_e[#MAX_FREQ_COEFFS] (exponents). The sine levels are stored in + #nrgSine_m[#MAX_FREQ_COEFFS] and #nrgSine_e[#MAX_FREQ_COEFFS]. \li Noise + limiting: The gain for each subband is limited both absolutely and relatively + compared to the total gain over all subbands. \li Boost gain: Calculate and + apply boost factor for each limiter band in order to compensate for the energy + loss imposed by the limiting. \li Apply gains and add noise: The gains and + noise levels are applied to all timeslots of the current envelope. A short + FIR-filter (length 4 QMF-timeslots) can be used to smooth the sudden change at + the envelope borders. Each complex subband sample of the current timeslot is + multiplied by the smoothed gain, then random noise with the calculated level + is added. + + \note + To reduce the stack size, some of the local arrays could be located within + the time output buffer. Of the 512 samples temporarily available there, + about half the size is already used by #SBR_FRAME_DATA. A pointer to the + remaining free memory could be supplied by an additional argument to + calculateSbrEnvelope() in sbr_dec: + + \par + \code + calculateSbrEnvelope (&hSbrDec->sbrScaleFactor, + &hSbrDec->SbrCalculateEnvelope, + hHeaderData, + hFrameData, + QmfBufferReal, + QmfBufferImag, + timeOutPtr + sizeof(SBR_FRAME_DATA)/sizeof(Float) + + 1); \endcode + + \par + Within calculateSbrEnvelope(), some pointers could be defined instead of the + arrays #nrgRef_m, #nrgRef_e, #nrgEst_m, #nrgEst_e, #noiseLevel_m: + + \par + \code + fract* nrgRef_m = timeOutPtr; + SCHAR* nrgRef_e = nrgRef_m + MAX_FREQ_COEFFS; + fract* nrgEst_m = nrgRef_e + MAX_FREQ_COEFFS; + SCHAR* nrgEst_e = nrgEst_m + MAX_FREQ_COEFFS; + fract* noiseLevel_m = nrgEst_e + MAX_FREQ_COEFFS; + \endcode + + <br> +*/ +void calculateSbrEnvelope( + QMF_SCALE_FACTOR *sbrScaleFactor, /*!< Scaling factors */ + HANDLE_SBR_CALCULATE_ENVELOPE + h_sbr_cal_env, /*!< Handle to struct filled by the create-function */ + HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ + HANDLE_SBR_FRAME_DATA hFrameData, /*!< Control data of current frame */ + PVC_DYNAMIC_DATA *pPvcDynamicData, + FIXP_DBL * + *analysBufferReal, /*!< Real part of subband samples to be processed */ + FIXP_DBL * + *analysBufferImag, /*!< Imag part of subband samples to be processed */ + const int useLP, + FIXP_DBL *degreeAlias, /*!< Estimated aliasing for each QMF channel */ + const UINT flags, const int frameErrorFlag) { + int c, i, i_stop, j, envNoise = 0; + UCHAR *borders = hFrameData->frameInfo.borders; + UCHAR *bordersPvc = hFrameData->frameInfo.pvcBorders; + int pvc_mode = pPvcDynamicData->pvc_mode; + int first_start = + ((pvc_mode > 0) ? bordersPvc[0] : borders[0]) * hHeaderData->timeStep; + FIXP_SGL *noiseLevels = hFrameData->sbrNoiseFloorLevel; + HANDLE_FREQ_BAND_DATA hFreq = &hHeaderData->freqBandData; + UCHAR **pFreqBandTable = hFreq->freqBandTable; + UCHAR *pFreqBandTableNoise = hFreq->freqBandTableNoise; + + int lowSubband = hFreq->lowSubband; + int highSubband = hFreq->highSubband; + int noSubbands = highSubband - lowSubband; + + /* old high subband before headerchange + we asume no headerchange here */ + int ov_highSubband = hFreq->highSubband; + + int noNoiseBands = hFreq->nNfb; + UCHAR *noSubFrameBands = hFreq->nSfb; + int no_cols = hHeaderData->numberTimeSlots * hHeaderData->timeStep; + + SCHAR sineMapped[MAX_FREQ_COEFFS]; + SCHAR ov_adj_e = SCALE2EXP(sbrScaleFactor->ov_hb_scale); + SCHAR adj_e = 0; + SCHAR output_e; + SCHAR final_e = 0; + /* inter-TES is active in one or more envelopes of the current SBR frame */ + const int iTES_enable = hFrameData->iTESactive; + const int iTES_scale_change = (iTES_enable) ? INTER_TES_SF_CHANGE : 0; + SCHAR maxGainLimit_e = (frameErrorFlag) ? MAX_GAIN_CONCEAL_EXP : MAX_GAIN_EXP; + + UCHAR smooth_length = 0; + + FIXP_SGL *pIenv = hFrameData->iEnvelope; + + C_ALLOC_SCRATCH_START(useAliasReduction, UCHAR, 64) + + /* if values differ we had a headerchange; if old highband is bigger then new + one we need to patch overlap-highband-scaling for this frame (see use of + ov_highSubband) as overlap contains higher frequency components which would + get lost */ + if (hFreq->highSubband < hFreq->ov_highSubband) { + ov_highSubband = hFreq->ov_highSubband; + } + + if (pvc_mode > 0) { + if (hFrameData->frameInfo.bordersNoise[0] > bordersPvc[0]) { + /* noise envelope of previous frame is trailing into current PVC frame */ + envNoise = -1; + noiseLevels = h_sbr_cal_env->prevSbrNoiseFloorLevel; + noNoiseBands = h_sbr_cal_env->prevNNfb; + noSubFrameBands = h_sbr_cal_env->prevNSfb; + lowSubband = h_sbr_cal_env->prevLoSubband; + highSubband = h_sbr_cal_env->prevHiSubband; + + noSubbands = highSubband - lowSubband; + ov_highSubband = highSubband; + if (highSubband < h_sbr_cal_env->prev_ov_highSubband) { + ov_highSubband = h_sbr_cal_env->prev_ov_highSubband; + } + + pFreqBandTable[0] = h_sbr_cal_env->prevFreqBandTableLo; + pFreqBandTable[1] = h_sbr_cal_env->prevFreqBandTableHi; + pFreqBandTableNoise = h_sbr_cal_env->prevFreqBandTableNoise; + } + + mapSineFlagsPvc(pFreqBandTable[1], noSubFrameBands[1], + h_sbr_cal_env->harmFlagsPrev, + h_sbr_cal_env->harmFlagsPrevActive, sineMapped, + hFrameData->sinusoidal_position, + &h_sbr_cal_env->sinusoidal_positionPrev, + (borders[0] > bordersPvc[0]) ? 1 : 0); + } else { + /* + Extract sine flags for all QMF bands + */ + mapSineFlags(pFreqBandTable[1], noSubFrameBands[1], + hFrameData->addHarmonics, h_sbr_cal_env->harmFlagsPrev, + h_sbr_cal_env->harmFlagsPrevActive, + hFrameData->frameInfo.tranEnv, sineMapped); + } + + /* + Scan for maximum in bufferd noise levels. + This is needed in case that we had strong noise in the previous frame + which is smoothed into the current frame. + The resulting exponent is used as start value for the maximum search + in reference energies + */ + if (!useLP) + adj_e = h_sbr_cal_env->filtBufferNoise_e - + getScalefactor(h_sbr_cal_env->filtBufferNoise, noSubbands); + + /* + Scan for maximum reference energy to be able + to select appropriate values for adj_e and final_e. + */ + if (pvc_mode > 0) { + INT maxSfbNrg_e = pPvcDynamicData->predEsg_expMax; + + /* Energy -> magnitude (sqrt halfens exponent) */ + maxSfbNrg_e = + (maxSfbNrg_e + 1) >> 1; /* +1 to go safe (round to next higher int) */ + + /* Some safety margin is needed for 2 reasons: + - The signal energy is not equally spread over all subband samples in + a specific sfb of an envelope (Nrg could be too high by a factor of + envWidth * sfbWidth) + - Smoothing can smear high gains of the previous envelope into the + current + */ + maxSfbNrg_e += 6; + + adj_e = maxSfbNrg_e; + // final_e should not exist for PVC fixfix framing + } else { + for (i = 0; i < hFrameData->frameInfo.nEnvelopes; i++) { + INT maxSfbNrg_e = + -FRACT_BITS + NRG_EXP_OFFSET; /* start value for maximum search */ + + /* Fetch frequency resolution for current envelope: */ + for (j = noSubFrameBands[hFrameData->frameInfo.freqRes[i]]; j != 0; j--) { + maxSfbNrg_e = fixMax(maxSfbNrg_e, (INT)((LONG)(*pIenv++) & MASK_E)); + } + maxSfbNrg_e -= NRG_EXP_OFFSET; + + /* Energy -> magnitude (sqrt halfens exponent) */ + maxSfbNrg_e = + (maxSfbNrg_e + 1) >> 1; /* +1 to go safe (round to next higher int) */ + + /* Some safety margin is needed for 2 reasons: + - The signal energy is not equally spread over all subband samples in + a specific sfb of an envelope (Nrg could be too high by a factor of + envWidth * sfbWidth) + - Smoothing can smear high gains of the previous envelope into the + current + */ + maxSfbNrg_e += 6; + + if (borders[i] < hHeaderData->numberTimeSlots) + /* This envelope affects timeslots that belong to the output frame */ + adj_e = fMax(maxSfbNrg_e, adj_e); + + if (borders[i + 1] > hHeaderData->numberTimeSlots) + /* This envelope affects timeslots after the output frame */ + final_e = fMax(maxSfbNrg_e, final_e); + } + } + /* + Calculate adjustment factors and apply them for every envelope. + */ + pIenv = hFrameData->iEnvelope; + + if (pvc_mode > 0) { + /* iterate over SBR time slots starting with bordersPvc[i] */ + i = bordersPvc[0]; /* usually 0; can be >0 if switching from legacy SBR to + PVC */ + i_stop = PVC_NTIMESLOT; + FDK_ASSERT(bordersPvc[hFrameData->frameInfo.nEnvelopes] == PVC_NTIMESLOT); + } else { + /* iterate over SBR envelopes starting with 0 */ + i = 0; + i_stop = hFrameData->frameInfo.nEnvelopes; + } + for (; i < i_stop; i++) { + int k, noNoiseFlag; + SCHAR noise_e, input_e = SCALE2EXP(sbrScaleFactor->hb_scale); + C_ALLOC_SCRATCH_START(pNrgs, ENV_CALC_NRGS, 1); + + /* + Helper variables. + */ + int start_pos, stop_pos, freq_res; + if (pvc_mode > 0) { + start_pos = + hHeaderData->timeStep * + i; /* Start-position in time (subband sample) for current envelope. */ + stop_pos = hHeaderData->timeStep * (i + 1); /* Stop-position in time + (subband sample) for + current envelope. */ + freq_res = + hFrameData->frameInfo + .freqRes[0]; /* Frequency resolution for current envelope. */ + FDK_ASSERT( + freq_res == + hFrameData->frameInfo.freqRes[hFrameData->frameInfo.nEnvelopes - 1]); + } else { + start_pos = hHeaderData->timeStep * + borders[i]; /* Start-position in time (subband sample) for + current envelope. */ + stop_pos = hHeaderData->timeStep * + borders[i + 1]; /* Stop-position in time (subband sample) for + current envelope. */ + freq_res = + hFrameData->frameInfo + .freqRes[i]; /* Frequency resolution for current envelope. */ + } + + /* Always fully initialize the temporary energy table. This prevents + negative energies and extreme gain factors in cases where the number of + limiter bands exceeds the number of subbands. The latter can be caused by + undetected bit errors and is tested by some streams from the + certification set. */ + FDKmemclear(pNrgs, sizeof(ENV_CALC_NRGS)); + + if (pvc_mode > 0) { + /* get predicted energy values from PVC module */ + expandPredEsg(pPvcDynamicData, i, (int)MAX_FREQ_COEFFS, pNrgs->nrgRef, + pNrgs->nrgRef_e); + + if (i == borders[0]) { + mapSineFlags(pFreqBandTable[1], noSubFrameBands[1], + hFrameData->addHarmonics, h_sbr_cal_env->harmFlagsPrev, + h_sbr_cal_env->harmFlagsPrevActive, + hFrameData->sinusoidal_position, sineMapped); + } + + if (i >= hFrameData->frameInfo.bordersNoise[envNoise + 1]) { + if (envNoise >= 0) { + noiseLevels += noNoiseBands; /* The noise floor data is stored in a + row [noiseFloor1 noiseFloor2...].*/ + } else { + /* leave trailing noise envelope of past frame */ + noNoiseBands = hFreq->nNfb; + noSubFrameBands = hFreq->nSfb; + noiseLevels = hFrameData->sbrNoiseFloorLevel; + + lowSubband = hFreq->lowSubband; + highSubband = hFreq->highSubband; + + noSubbands = highSubband - lowSubband; + ov_highSubband = highSubband; + if (highSubband < hFreq->ov_highSubband) { + ov_highSubband = hFreq->ov_highSubband; + } + + pFreqBandTable[0] = hFreq->freqBandTableLo; + pFreqBandTable[1] = hFreq->freqBandTableHi; + pFreqBandTableNoise = hFreq->freqBandTableNoise; + } + envNoise++; + } + } else { + /* If the start-pos of the current envelope equals the stop pos of the + current noise envelope, increase the pointer (i.e. choose the next + noise-floor).*/ + if (borders[i] == hFrameData->frameInfo.bordersNoise[envNoise + 1]) { + noiseLevels += noNoiseBands; /* The noise floor data is stored in a row + [noiseFloor1 noiseFloor2...].*/ + envNoise++; + } + } + if (i == hFrameData->frameInfo.tranEnv || + i == h_sbr_cal_env->prevTranEnv) /* attack */ + { + noNoiseFlag = 1; + if (!useLP) smooth_length = 0; /* No smoothing on attacks! */ + } else { + noNoiseFlag = 0; + if (!useLP) + smooth_length = (1 - hHeaderData->bs_data.smoothingLength) + << 2; /* can become either 0 or 4 */ + } + + /* + Energy estimation in transposed highband. + */ + if (hHeaderData->bs_data.interpolFreq) + calcNrgPerSubband(analysBufferReal, (useLP) ? NULL : analysBufferImag, + lowSubband, highSubband, start_pos, stop_pos, input_e, + pNrgs->nrgEst, pNrgs->nrgEst_e); + else + calcNrgPerSfb(analysBufferReal, (useLP) ? NULL : analysBufferImag, + noSubFrameBands[freq_res], pFreqBandTable[freq_res], + start_pos, stop_pos, input_e, pNrgs->nrgEst, + pNrgs->nrgEst_e); + + /* + Calculate subband gains + */ + { + UCHAR *table = pFreqBandTable[freq_res]; + UCHAR *pUiNoise = + &pFreqBandTableNoise[1]; /*! Upper limit of the current noise floor + band. */ + + FIXP_SGL *pNoiseLevels = noiseLevels; + + FIXP_DBL tmpNoise = + FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pNoiseLevels) & MASK_M)); + SCHAR tmpNoise_e = + (UCHAR)((LONG)(*pNoiseLevels++) & MASK_E) - NOISE_EXP_OFFSET; + + int cc = 0; + c = 0; + if (pvc_mode > 0) { + for (j = 0; j < noSubFrameBands[freq_res]; j++) { + UCHAR sinePresentFlag = 0; + int li = table[j]; + int ui = table[j + 1]; + + for (k = li; k < ui; k++) { + sinePresentFlag |= (i >= sineMapped[cc]); + cc++; + } + + for (k = li; k < ui; k++) { + FIXP_DBL refNrg = pNrgs->nrgRef[k - lowSubband]; + SCHAR refNrg_e = pNrgs->nrgRef_e[k - lowSubband]; + + if (k >= *pUiNoise) { + tmpNoise = + FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pNoiseLevels) & MASK_M)); + tmpNoise_e = + (SCHAR)((LONG)(*pNoiseLevels++) & MASK_E) - NOISE_EXP_OFFSET; + + pUiNoise++; + } + + FDK_ASSERT(k >= lowSubband); + + if (useLP) useAliasReduction[k - lowSubband] = !sinePresentFlag; + + pNrgs->nrgSine[c] = FL2FXCONST_DBL(0.0f); + pNrgs->nrgSine_e[c] = 0; + + calcSubbandGain(refNrg, refNrg_e, pNrgs, c, tmpNoise, tmpNoise_e, + sinePresentFlag, i >= sineMapped[c], noNoiseFlag); + + c++; + } + } + } else { + for (j = 0; j < noSubFrameBands[freq_res]; j++) { + FIXP_DBL refNrg = FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pIenv) & MASK_M)); + SCHAR refNrg_e = (SCHAR)((LONG)(*pIenv) & MASK_E) - NRG_EXP_OFFSET; + + UCHAR sinePresentFlag = 0; + int li = table[j]; + int ui = table[j + 1]; + + for (k = li; k < ui; k++) { + sinePresentFlag |= (i >= sineMapped[cc]); + cc++; + } + + for (k = li; k < ui; k++) { + if (k >= *pUiNoise) { + tmpNoise = + FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pNoiseLevels) & MASK_M)); + tmpNoise_e = + (SCHAR)((LONG)(*pNoiseLevels++) & MASK_E) - NOISE_EXP_OFFSET; + + pUiNoise++; + } + + FDK_ASSERT(k >= lowSubband); + + if (useLP) useAliasReduction[k - lowSubband] = !sinePresentFlag; + + pNrgs->nrgSine[c] = FL2FXCONST_DBL(0.0f); + pNrgs->nrgSine_e[c] = 0; + + calcSubbandGain(refNrg, refNrg_e, pNrgs, c, tmpNoise, tmpNoise_e, + sinePresentFlag, i >= sineMapped[c], noNoiseFlag); + + pNrgs->nrgRef[c] = refNrg; + pNrgs->nrgRef_e[c] = refNrg_e; + + c++; + } + pIenv++; + } + } + } + + /* + Noise limiting + */ + + for (c = 0; c < hFreq->noLimiterBands; c++) { + FIXP_DBL sumRef, boostGain, maxGain; + FIXP_DBL accu = FL2FXCONST_DBL(0.0f); + SCHAR sumRef_e, boostGain_e, maxGain_e, accu_e = 0; + int maxGainLimGainSum_e = 0; + + calcAvgGain(pNrgs, hFreq->limiterBandTable[c], + hFreq->limiterBandTable[c + 1], &sumRef, &sumRef_e, &maxGain, + &maxGain_e); + + /* Multiply maxGain with limiterGain: */ + maxGain = fMult( + maxGain, + FDK_sbrDecoder_sbr_limGains_m[hHeaderData->bs_data.limiterGains]); + /* maxGain_e += + * FDK_sbrDecoder_sbr_limGains_e[hHeaderData->bs_data.limiterGains]; */ + /* The addition of maxGain_e and FDK_sbrDecoder_sbr_limGains_e[3] might + yield values greater than 127 which doesn't fit into an SCHAR! In these + rare situations limit maxGain_e to 127. + */ + maxGainLimGainSum_e = + maxGain_e + + FDK_sbrDecoder_sbr_limGains_e[hHeaderData->bs_data.limiterGains]; + maxGain_e = + (maxGainLimGainSum_e > 127) ? (SCHAR)127 : (SCHAR)maxGainLimGainSum_e; + + /* Scale mantissa of MaxGain into range between 0.5 and 1: */ + if (maxGain == FL2FXCONST_DBL(0.0f)) + maxGain_e = -FRACT_BITS; + else { + SCHAR charTemp = CountLeadingBits(maxGain); + maxGain_e -= charTemp; + maxGain <<= (int)charTemp; + } + + if (maxGain_e >= maxGainLimit_e) { /* upper limit (e.g. 96 dB) */ + maxGain = FL2FXCONST_DBL(0.5f); + maxGain_e = maxGainLimit_e; + } + + /* Every subband gain is compared to the scaled "average gain" + and limited if necessary: */ + for (k = hFreq->limiterBandTable[c]; k < hFreq->limiterBandTable[c + 1]; + k++) { + if ((pNrgs->nrgGain_e[k] > maxGain_e) || + (pNrgs->nrgGain_e[k] == maxGain_e && pNrgs->nrgGain[k] > maxGain)) { + FIXP_DBL noiseAmp; + SCHAR noiseAmp_e; + + FDK_divide_MantExp(maxGain, maxGain_e, pNrgs->nrgGain[k], + pNrgs->nrgGain_e[k], &noiseAmp, &noiseAmp_e); + pNrgs->noiseLevel[k] = fMult(pNrgs->noiseLevel[k], noiseAmp); + pNrgs->noiseLevel_e[k] += noiseAmp_e; + pNrgs->nrgGain[k] = maxGain; + pNrgs->nrgGain_e[k] = maxGain_e; + } + } + + /* -- Boost gain + Calculate and apply boost factor for each limiter band: + 1. Check how much energy would be present when using the limited gain + 2. Calculate boost factor by comparison with reference energy + 3. Apply boost factor to compensate for the energy loss due to limiting + */ + for (k = hFreq->limiterBandTable[c]; k < hFreq->limiterBandTable[c + 1]; + k++) { + /* 1.a Add energy of adjusted signal (using preliminary gain) */ + FIXP_DBL tmp = fMult(pNrgs->nrgGain[k], pNrgs->nrgEst[k]); + SCHAR tmp_e = pNrgs->nrgGain_e[k] + pNrgs->nrgEst_e[k]; + FDK_add_MantExp(tmp, tmp_e, accu, accu_e, &accu, &accu_e); + + /* 1.b Add sine energy (if present) */ + if (pNrgs->nrgSine[k] != FL2FXCONST_DBL(0.0f)) { + FDK_add_MantExp(pNrgs->nrgSine[k], pNrgs->nrgSine_e[k], accu, accu_e, + &accu, &accu_e); + } else { + /* 1.c Add noise energy (if present) */ + if (noNoiseFlag == 0) { + FDK_add_MantExp(pNrgs->noiseLevel[k], pNrgs->noiseLevel_e[k], accu, + accu_e, &accu, &accu_e); + } + } + } + + /* 2.a Calculate ratio of wanted energy and accumulated energy */ + if (accu == (FIXP_DBL)0) { /* If divisor is 0, limit quotient to +4 dB */ + boostGain = FL2FXCONST_DBL(0.6279716f); + boostGain_e = 2; + } else { + INT div_e; + boostGain = fDivNorm(sumRef, accu, &div_e); + boostGain_e = sumRef_e - accu_e + div_e; + } + + /* 2.b Result too high? --> Limit the boost factor to +4 dB */ + if ((boostGain_e > 3) || + (boostGain_e == 2 && boostGain > FL2FXCONST_DBL(0.6279716f)) || + (boostGain_e == 3 && boostGain > FL2FXCONST_DBL(0.3139858f))) { + boostGain = FL2FXCONST_DBL(0.6279716f); + boostGain_e = 2; + } + /* 3. Multiply all signal components with the boost factor */ + for (k = hFreq->limiterBandTable[c]; k < hFreq->limiterBandTable[c + 1]; + k++) { + pNrgs->nrgGain[k] = fMultDiv2(pNrgs->nrgGain[k], boostGain); + pNrgs->nrgGain_e[k] = pNrgs->nrgGain_e[k] + boostGain_e + 1; + + pNrgs->nrgSine[k] = fMultDiv2(pNrgs->nrgSine[k], boostGain); + pNrgs->nrgSine_e[k] = pNrgs->nrgSine_e[k] + boostGain_e + 1; + + pNrgs->noiseLevel[k] = fMultDiv2(pNrgs->noiseLevel[k], boostGain); + pNrgs->noiseLevel_e[k] = pNrgs->noiseLevel_e[k] + boostGain_e + 1; + } + } + /* End of noise limiting */ + + if (useLP) + aliasingReduction(degreeAlias + lowSubband, pNrgs, useAliasReduction, + noSubbands); + + /* For the timeslots within the range for the output frame, + use the same scale for the noise levels. + Drawback: If the envelope exceeds the frame border, the noise levels + will have to be rescaled later to fit final_e of + the gain-values. + */ + noise_e = (start_pos < no_cols) ? adj_e : final_e; + + /* + Convert energies to amplitude levels + */ + for (k = 0; k < noSubbands; k++) { + FDK_sqrt_MantExp(&pNrgs->nrgSine[k], &pNrgs->nrgSine_e[k], &noise_e); + FDK_sqrt_MantExp(&pNrgs->nrgGain[k], &pNrgs->nrgGain_e[k], + &pNrgs->nrgGain_e[k]); + FDK_sqrt_MantExp(&pNrgs->noiseLevel[k], &pNrgs->noiseLevel_e[k], + &noise_e); + } + + /* + Apply calculated gains and adaptive noise + */ + + /* assembleHfSignals() */ + { + int scale_change, sc_change; + FIXP_SGL smooth_ratio; + int filtBufferNoiseShift = 0; + + /* Initialize smoothing buffers with the first valid values */ + if (h_sbr_cal_env->startUp) { + if (!useLP) { + h_sbr_cal_env->filtBufferNoise_e = noise_e; + + FDKmemcpy(h_sbr_cal_env->filtBuffer_e, pNrgs->nrgGain_e, + noSubbands * sizeof(SCHAR)); + FDKmemcpy(h_sbr_cal_env->filtBufferNoise, pNrgs->noiseLevel, + noSubbands * sizeof(FIXP_DBL)); + FDKmemcpy(h_sbr_cal_env->filtBuffer, pNrgs->nrgGain, + noSubbands * sizeof(FIXP_DBL)); + } + h_sbr_cal_env->startUp = 0; + } + + if (!useLP) { + equalizeFiltBufferExp(h_sbr_cal_env->filtBuffer, /* buffered */ + h_sbr_cal_env->filtBuffer_e, /* buffered */ + pNrgs->nrgGain, /* current */ + pNrgs->nrgGain_e, /* current */ + noSubbands); + + /* Adapt exponent of buffered noise levels to the current exponent + so they can easily be smoothed */ + if ((h_sbr_cal_env->filtBufferNoise_e - noise_e) >= 0) { + int shift = fixMin(DFRACT_BITS - 1, + (int)(h_sbr_cal_env->filtBufferNoise_e - noise_e)); + for (k = 0; k < noSubbands; k++) + h_sbr_cal_env->filtBufferNoise[k] <<= shift; + } else { + int shift = + fixMin(DFRACT_BITS - 1, + -(int)(h_sbr_cal_env->filtBufferNoise_e - noise_e)); + for (k = 0; k < noSubbands; k++) + h_sbr_cal_env->filtBufferNoise[k] >>= shift; + } + + h_sbr_cal_env->filtBufferNoise_e = noise_e; + } + + /* find best scaling! */ + scale_change = -(DFRACT_BITS - 1); + for (k = 0; k < noSubbands; k++) { + scale_change = fixMax(scale_change, (int)pNrgs->nrgGain_e[k]); + } + sc_change = (start_pos < no_cols) ? adj_e - input_e : final_e - input_e; + + if ((scale_change - sc_change + 1) < 0) + scale_change -= (scale_change - sc_change + 1); + + scale_change = (scale_change - sc_change) + 1; + + for (k = 0; k < noSubbands; k++) { + int sc = scale_change - pNrgs->nrgGain_e[k] + (sc_change - 1); + pNrgs->nrgGain[k] >>= sc; + pNrgs->nrgGain_e[k] += sc; + } + + if (!useLP) { + for (k = 0; k < noSubbands; k++) { + int sc = + scale_change - h_sbr_cal_env->filtBuffer_e[k] + (sc_change - 1); + h_sbr_cal_env->filtBuffer[k] >>= sc; + } + } + + for (j = start_pos; j < stop_pos; j++) { + /* This timeslot is located within the first part of the processing + buffer and will be fed into the QMF-synthesis for the current frame. + adj_e - input_e + This timeslot will not yet be fed into the QMF so we do not care + about the adj_e. + sc_change = final_e - input_e + */ + if ((j == no_cols) && (start_pos < no_cols)) { + int shift = (int)(noise_e - final_e); + if (!useLP) + filtBufferNoiseShift = shift; /* shifting of + h_sbr_cal_env->filtBufferNoise[k] + will be applied in function + adjustTimeSlotHQ() */ + if (shift >= 0) { + shift = fixMin(DFRACT_BITS - 1, shift); + for (k = 0; k < noSubbands; k++) { + pNrgs->nrgSine[k] <<= shift; + pNrgs->noiseLevel[k] <<= shift; + /* + if (!useLP) + h_sbr_cal_env->filtBufferNoise[k] <<= shift; + */ + } + } else { + shift = fixMin(DFRACT_BITS - 1, -shift); + for (k = 0; k < noSubbands; k++) { + pNrgs->nrgSine[k] >>= shift; + pNrgs->noiseLevel[k] >>= shift; + /* + if (!useLP) + h_sbr_cal_env->filtBufferNoise[k] >>= shift; + */ + } + } + + /* update noise scaling */ + noise_e = final_e; + if (!useLP) + h_sbr_cal_env->filtBufferNoise_e = + noise_e; /* scaling value unused! */ + + /* update gain buffer*/ + sc_change -= (final_e - input_e); + + if (sc_change < 0) { + for (k = 0; k < noSubbands; k++) { + pNrgs->nrgGain[k] >>= -sc_change; + pNrgs->nrgGain_e[k] += -sc_change; + } + if (!useLP) { + for (k = 0; k < noSubbands; k++) { + h_sbr_cal_env->filtBuffer[k] >>= -sc_change; + } + } + } else { + scale_change += sc_change; + } + + } /* if */ + + if (!useLP) { + /* Prevent the smoothing filter from running on constant levels */ + if (j - start_pos < smooth_length) + smooth_ratio = FDK_sbrDecoder_sbr_smoothFilter[j - start_pos]; + else + smooth_ratio = FL2FXCONST_SGL(0.0f); + + if (iTES_enable) { + /* adjustTimeSlotHQ() without adding of additional harmonics */ + adjustTimeSlotHQ_GainAndNoise( + &analysBufferReal[j][lowSubband], + &analysBufferImag[j][lowSubband], h_sbr_cal_env, pNrgs, + lowSubband, noSubbands, fMin(scale_change, DFRACT_BITS - 1), + smooth_ratio, noNoiseFlag, filtBufferNoiseShift); + } else { + adjustTimeSlotHQ(&analysBufferReal[j][lowSubband], + &analysBufferImag[j][lowSubband], h_sbr_cal_env, + pNrgs, lowSubband, noSubbands, + fMin(scale_change, DFRACT_BITS - 1), smooth_ratio, + noNoiseFlag, filtBufferNoiseShift); + } + } else { + FDK_ASSERT(!iTES_enable); /* not supported */ + if (flags & SBRDEC_ELD_GRID) { + /* FDKmemset(analysBufferReal[j], 0, 64 * sizeof(FIXP_DBL)); */ + adjustTimeSlot_EldGrid(&analysBufferReal[j][lowSubband], pNrgs, + &h_sbr_cal_env->harmIndex, lowSubband, + noSubbands, + fMin(scale_change, DFRACT_BITS - 1), + noNoiseFlag, &h_sbr_cal_env->phaseIndex, + EXP2SCALE(adj_e) - sbrScaleFactor->lb_scale); + } else { + adjustTimeSlotLC(&analysBufferReal[j][lowSubband], pNrgs, + &h_sbr_cal_env->harmIndex, lowSubband, noSubbands, + fMin(scale_change, DFRACT_BITS - 1), noNoiseFlag, + &h_sbr_cal_env->phaseIndex); + } + } + /* In case the envelope spans accross the no_cols border both exponents + * are needed. */ + /* nrgGain_e[0...(noSubbands-1)] are equalized by + * equalizeFiltBufferExp() */ + pNrgs->exponent[(j < no_cols) ? 0 : 1] = + (SCHAR)((15 - sbrScaleFactor->hb_scale) + pNrgs->nrgGain_e[0] + 1 - + scale_change); + } /* for */ + + if (iTES_enable) { + apply_inter_tes( + analysBufferReal, /* pABufR, */ + analysBufferImag, /* pABufI, */ + sbrScaleFactor, pNrgs->exponent, hHeaderData->timeStep, start_pos, + stop_pos, lowSubband, noSubbands, + hFrameData + ->interTempShapeMode[i] /* frameData->interTempShapeMode[env] */ + ); + + /* add additional harmonics */ + for (j = start_pos; j < stop_pos; j++) { + /* match exponent of additional harmonics to scale change of QMF data + * caused by apply_inter_tes() */ + scale_change = 0; + + if ((start_pos <= no_cols) && (stop_pos > no_cols)) { + /* Scaling of analysBuffers was potentially changed within this + envelope. The pNrgs->nrgSine_e match the second part of the + envelope. For (j<=no_cols) the exponent of the sine energies has + to be adapted. */ + scale_change = pNrgs->exponent[1] - pNrgs->exponent[0]; + } + + adjustTimeSlotHQ_AddHarmonics( + &analysBufferReal[j][lowSubband], + &analysBufferImag[j][lowSubband], h_sbr_cal_env, pNrgs, + lowSubband, noSubbands, + -iTES_scale_change + ((j < no_cols) ? scale_change : 0)); + } + } + + if (!useLP) { + /* Update time-smoothing-buffers for gains and noise levels + The gains and the noise values of the current envelope are copied + into the buffer. This has to be done at the end of each envelope as + the values are required for a smooth transition to the next envelope. + */ + FDKmemcpy(h_sbr_cal_env->filtBuffer, pNrgs->nrgGain, + noSubbands * sizeof(FIXP_DBL)); + FDKmemcpy(h_sbr_cal_env->filtBuffer_e, pNrgs->nrgGain_e, + noSubbands * sizeof(SCHAR)); + FDKmemcpy(h_sbr_cal_env->filtBufferNoise, pNrgs->noiseLevel, + noSubbands * sizeof(FIXP_DBL)); + } + } + C_ALLOC_SCRATCH_END(pNrgs, ENV_CALC_NRGS, 1); + } + + /* adapt adj_e to the scale change caused by apply_inter_tes() */ + adj_e += iTES_scale_change; + + /* Rescale output samples */ + { + FIXP_DBL maxVal; + int ov_reserve, reserve; + + /* Determine headroom in old adjusted samples */ + maxVal = + maxSubbandSample(analysBufferReal, (useLP) ? NULL : analysBufferImag, + lowSubband, ov_highSubband, 0, first_start); + + ov_reserve = fNorm(maxVal); + + /* Determine headroom in new adjusted samples */ + maxVal = + maxSubbandSample(analysBufferReal, (useLP) ? NULL : analysBufferImag, + lowSubband, highSubband, first_start, no_cols); + + reserve = fNorm(maxVal); + + /* Determine common output exponent */ + output_e = fMax(ov_adj_e - ov_reserve, adj_e - reserve); + + /* Rescale old samples */ + rescaleSubbandSamples(analysBufferReal, (useLP) ? NULL : analysBufferImag, + lowSubband, ov_highSubband, 0, first_start, + ov_adj_e - output_e); + + /* Rescale new samples */ + rescaleSubbandSamples(analysBufferReal, (useLP) ? NULL : analysBufferImag, + lowSubband, highSubband, first_start, no_cols, + adj_e - output_e); + } + + /* Update hb_scale */ + sbrScaleFactor->hb_scale = EXP2SCALE(output_e); + + /* Save the current final exponent for the next frame: */ + /* adapt final_e to the scale change caused by apply_inter_tes() */ + sbrScaleFactor->ov_hb_scale = EXP2SCALE(final_e + iTES_scale_change); + + /* We need to remember to the next frame that the transient + will occur in the first envelope (if tranEnv == nEnvelopes). */ + if (hFrameData->frameInfo.tranEnv == hFrameData->frameInfo.nEnvelopes) + h_sbr_cal_env->prevTranEnv = 0; + else + h_sbr_cal_env->prevTranEnv = -1; + + if (pvc_mode > 0) { + /* Not more than just the last noise envelope reaches into the next PVC + frame! This should be true because bs_noise_position is <= 15 */ + FDK_ASSERT(hFrameData->frameInfo + .bordersNoise[hFrameData->frameInfo.nNoiseEnvelopes - 1] < + PVC_NTIMESLOT); + if (hFrameData->frameInfo + .bordersNoise[hFrameData->frameInfo.nNoiseEnvelopes] > + PVC_NTIMESLOT) { + FDK_ASSERT(noiseLevels == + (hFrameData->sbrNoiseFloorLevel + + (hFrameData->frameInfo.nNoiseEnvelopes - 1) * noNoiseBands)); + h_sbr_cal_env->prevNNfb = noNoiseBands; + + h_sbr_cal_env->prevNSfb[0] = noSubFrameBands[0]; + h_sbr_cal_env->prevNSfb[1] = noSubFrameBands[1]; + + h_sbr_cal_env->prevLoSubband = lowSubband; + h_sbr_cal_env->prevHiSubband = highSubband; + h_sbr_cal_env->prev_ov_highSubband = ov_highSubband; + + FDKmemcpy(h_sbr_cal_env->prevFreqBandTableLo, pFreqBandTable[0], + noSubFrameBands[0] + 1); + FDKmemcpy(h_sbr_cal_env->prevFreqBandTableHi, pFreqBandTable[1], + noSubFrameBands[1] + 1); + FDKmemcpy(h_sbr_cal_env->prevFreqBandTableNoise, + hFreq->freqBandTableNoise, sizeof(hFreq->freqBandTableNoise)); + + FDKmemcpy(h_sbr_cal_env->prevSbrNoiseFloorLevel, noiseLevels, + MAX_NOISE_COEFFS * sizeof(FIXP_SGL)); + } + } + + C_ALLOC_SCRATCH_END(useAliasReduction, UCHAR, 64) +} + +/*! + \brief Create envelope instance + + Must be called once for each channel before calculateSbrEnvelope() can be + used. + + \return errorCode, 0 if successful +*/ +SBR_ERROR +createSbrEnvelopeCalc( + HANDLE_SBR_CALCULATE_ENVELOPE hs, /*!< pointer to envelope instance */ + HANDLE_SBR_HEADER_DATA + hHeaderData, /*!< static SBR control data, initialized with defaults */ + const int chan, /*!< Channel for which to assign buffers */ + const UINT flags) { + SBR_ERROR err = SBRDEC_OK; + int i; + + /* Clear previous missing harmonics flags */ + for (i = 0; i < ADD_HARMONICS_FLAGS_SIZE; i++) { + hs->harmFlagsPrev[i] = 0; + hs->harmFlagsPrevActive[i] = 0; + } + hs->harmIndex = 0; + + FDKmemclear(hs->prevSbrNoiseFloorLevel, sizeof(hs->prevSbrNoiseFloorLevel)); + hs->prevNNfb = 0; + FDKmemclear(hs->prevFreqBandTableNoise, sizeof(hs->prevFreqBandTableNoise)); + hs->sinusoidal_positionPrev = 0; + + /* + Setup pointers for time smoothing. + The buffer itself will be initialized later triggered by the startUp-flag. + */ + hs->prevTranEnv = -1; + + /* initialization */ + resetSbrEnvelopeCalc(hs); + + if (chan == 0) { /* do this only once */ + err = resetFreqBandTables(hHeaderData, flags); + } + + return err; +} + +/*! + \brief Create envelope instance + + Must be called once for each channel before calculateSbrEnvelope() can be + used. + + \return errorCode, 0 if successful +*/ +int deleteSbrEnvelopeCalc(HANDLE_SBR_CALCULATE_ENVELOPE hs) { return 0; } + +/*! + \brief Reset envelope instance + + This function must be called for each channel on a change of configuration. + Note that resetFreqBandTables should also be called in this case. + + \return errorCode, 0 if successful +*/ +void resetSbrEnvelopeCalc( + HANDLE_SBR_CALCULATE_ENVELOPE hCalEnv) /*!< pointer to envelope instance */ +{ + hCalEnv->phaseIndex = 0; + + /* Noise exponent needs to be reset because the output exponent for the next + * frame depends on it */ + hCalEnv->filtBufferNoise_e = 0; + + hCalEnv->startUp = 1; +} + +/*! + \brief Equalize exponents of the buffered gain values and the new ones + + After equalization of exponents, the FIR-filter addition for smoothing + can be performed. + This function is called once for each envelope before adjusting. +*/ +static void equalizeFiltBufferExp( + FIXP_DBL *filtBuffer, /*!< bufferd gains */ + SCHAR *filtBuffer_e, /*!< exponents of bufferd gains */ + FIXP_DBL *nrgGain, /*!< gains for current envelope */ + SCHAR *nrgGain_e, /*!< exponents of gains for current envelope */ + int subbands) /*!< Number of QMF subbands */ +{ + int band; + int diff; + + for (band = 0; band < subbands; band++) { + diff = (int)(nrgGain_e[band] - filtBuffer_e[band]); + if (diff > 0) { + filtBuffer[band] >>= + diff; /* Compensate for the scale change by shifting the mantissa. */ + filtBuffer_e[band] += diff; /* New gain is bigger, use its exponent */ + } else if (diff < 0) { + /* The buffered gains seem to be larger, but maybe there + are some unused bits left in the mantissa */ + + int reserve = CntLeadingZeros(fixp_abs(filtBuffer[band])) - 1; + + if ((-diff) <= reserve) { + /* There is enough space in the buffered mantissa so + that we can take the new exponent as common. + */ + filtBuffer[band] <<= (-diff); + filtBuffer_e[band] += diff; /* becomes equal to *ptrNewExp */ + } else { + filtBuffer[band] <<= + reserve; /* Shift the mantissa as far as possible: */ + filtBuffer_e[band] -= reserve; /* Compensate in the exponent: */ + + /* For the remaining difference, change the new gain value */ + diff = fixMin(-(reserve + diff), DFRACT_BITS - 1); + nrgGain[band] >>= diff; + nrgGain_e[band] += diff; + } + } + } +} + +/*! + \brief Shift left the mantissas of all subband samples + in the giventime and frequency range by the specified number of bits. + + This function is used to rescale the audio data in the overlap buffer + which has already been envelope adjusted with the last frame. +*/ +void rescaleSubbandSamples( + FIXP_DBL **re, /*!< Real part of input and output subband samples */ + FIXP_DBL **im, /*!< Imaginary part of input and output subband samples */ + int lowSubband, /*!< Begin of frequency range to process */ + int highSubband, /*!< End of frequency range to process */ + int start_pos, /*!< Begin of time rage (QMF-timeslot) */ + int next_pos, /*!< End of time rage (QMF-timeslot) */ + int shift) /*!< number of bits to shift */ +{ + int width = highSubband - lowSubband; + + if ((width > 0) && (shift != 0)) { + if (im != NULL) { + for (int l = start_pos; l < next_pos; l++) { + scaleValues(&re[l][lowSubband], width, shift); + scaleValues(&im[l][lowSubband], width, shift); + } + } else { + for (int l = start_pos; l < next_pos; l++) { + scaleValues(&re[l][lowSubband], width, shift); + } + } + } +} + +static inline FIXP_DBL FDK_get_maxval_real(FIXP_DBL maxVal, FIXP_DBL *reTmp, + INT width) { + maxVal = (FIXP_DBL)0; + while (width-- != 0) { + FIXP_DBL tmp = *(reTmp++); + maxVal |= (FIXP_DBL)((LONG)(tmp) ^ ((LONG)tmp >> (DFRACT_BITS - 1))); + } + + return maxVal; +} + +/*! + \brief Determine headroom for shifting + + Determine by how much the spectrum can be shifted left + for better accuracy in later processing. + + \return Number of free bits in the biggest spectral value +*/ + +FIXP_DBL maxSubbandSample( + FIXP_DBL **re, /*!< Real part of input and output subband samples */ + FIXP_DBL **im, /*!< Real part of input and output subband samples */ + int lowSubband, /*!< Begin of frequency range to process */ + int highSubband, /*!< Number of QMF bands to process */ + int start_pos, /*!< Begin of time rage (QMF-timeslot) */ + int next_pos /*!< End of time rage (QMF-timeslot) */ +) { + FIXP_DBL maxVal = FL2FX_DBL(0.0f); + unsigned int width = highSubband - lowSubband; + + FDK_ASSERT(width <= (64)); + + if (width > 0) { + if (im != NULL) { + for (int l = start_pos; l < next_pos; l++) { + int k = width; + FIXP_DBL *reTmp = &re[l][lowSubband]; + FIXP_DBL *imTmp = &im[l][lowSubband]; + do { + FIXP_DBL tmp1 = *(reTmp++); + FIXP_DBL tmp2 = *(imTmp++); + maxVal |= + (FIXP_DBL)((LONG)(tmp1) ^ ((LONG)tmp1 >> (DFRACT_BITS - 1))); + maxVal |= + (FIXP_DBL)((LONG)(tmp2) ^ ((LONG)tmp2 >> (DFRACT_BITS - 1))); + } while (--k != 0); + } + } else { + for (int l = start_pos; l < next_pos; l++) { + maxVal |= FDK_get_maxval_real(maxVal, &re[l][lowSubband], width); + } + } + } + + if (maxVal > (FIXP_DBL)0) { + /* For negative input values, maxVal is too small by 1. Add 1 only when + * necessary: if maxVal is a power of 2 */ + FIXP_DBL lowerPow2 = + (FIXP_DBL)(1 << (DFRACT_BITS - 1 - CntLeadingZeros(maxVal))); + if (maxVal == lowerPow2) maxVal += (FIXP_DBL)1; + } + + return (maxVal); +} + +/* #define SHIFT_BEFORE_SQUARE (3) */ /* (7/2) */ +/* Avoid assertion failures triggerd by overflows which occured in robustness + tests. Setting the SHIFT_BEFORE_SQUARE to 4 has negligible effect on (USAC) + conformance results. */ +#define SHIFT_BEFORE_SQUARE (4) /* ((8 - 0) / 2) */ + +/*!< + If the accumulator does not provide enough overflow bits or + does not provide a high dynamic range, the below energy calculation + requires an additional shift operation for each sample. + On the other hand, doing the shift allows using a single-precision + multiplication for the square (at least 16bit x 16bit). + For even values of OVRFLW_BITS (0, 2, 4, 6), saturated arithmetic + is required for the energy accumulation. + Theoretically, the sample-squares can sum up to a value of 76, + requiring 7 overflow bits. However since such situations are *very* + rare, accu can be limited to 64. + In case native saturated arithmetic is not available, overflows + can be prevented by replacing the above #define by + #define SHIFT_BEFORE_SQUARE ((8 - OVRFLW_BITS) / 2) + which will result in slightly reduced accuracy. +*/ + +/*! + \brief Estimates the mean energy of each filter-bank channel for the + duration of the current envelope + + This function is used when interpolFreq is true. +*/ +static void calcNrgPerSubband( + FIXP_DBL **analysBufferReal, /*!< Real part of subband samples */ + FIXP_DBL **analysBufferImag, /*!< Imaginary part of subband samples */ + int lowSubband, /*!< Begin of the SBR frequency range */ + int highSubband, /*!< High end of the SBR frequency range */ + int start_pos, /*!< First QMF-slot of current envelope */ + int next_pos, /*!< Last QMF-slot of current envelope + 1 */ + SCHAR frameExp, /*!< Common exponent for all input samples */ + FIXP_DBL *nrgEst, /*!< resulting Energy (0..1) */ + SCHAR *nrgEst_e) /*!< Exponent of resulting Energy */ +{ + FIXP_SGL invWidth; + SCHAR preShift; + SCHAR shift; + FIXP_DBL sum; + int k; + + /* Divide by width of envelope later: */ + invWidth = FX_DBL2FX_SGL(GetInvInt(next_pos - start_pos)); + /* The common exponent needs to be doubled because all mantissas are squared: + */ + frameExp = frameExp << 1; + + for (k = lowSubband; k < highSubband; k++) { + FIXP_DBL bufferReal[(((1024) / (32) * (4) / 2) + (3 * (4)))]; + FIXP_DBL bufferImag[(((1024) / (32) * (4) / 2) + (3 * (4)))]; + FIXP_DBL maxVal; + + if (analysBufferImag != NULL) { + int l; + maxVal = FL2FX_DBL(0.0f); + for (l = start_pos; l < next_pos; l++) { + bufferImag[l] = analysBufferImag[l][k]; + maxVal |= (FIXP_DBL)((LONG)(bufferImag[l]) ^ + ((LONG)bufferImag[l] >> (DFRACT_BITS - 1))); + bufferReal[l] = analysBufferReal[l][k]; + maxVal |= (FIXP_DBL)((LONG)(bufferReal[l]) ^ + ((LONG)bufferReal[l] >> (DFRACT_BITS - 1))); + } + } else { + int l; + maxVal = FL2FX_DBL(0.0f); + for (l = start_pos; l < next_pos; l++) { + bufferReal[l] = analysBufferReal[l][k]; + maxVal |= (FIXP_DBL)((LONG)(bufferReal[l]) ^ + ((LONG)bufferReal[l] >> (DFRACT_BITS - 1))); + } + } + + if (maxVal != FL2FXCONST_DBL(0.f)) { + /* If the accu does not provide enough overflow bits, we cannot + shift the samples up to the limit. + Instead, keep up to 3 free bits in each sample, i.e. up to + 6 bits after calculation of square. + Please note the comment on saturated arithmetic above! + */ + FIXP_DBL accu; + preShift = CntLeadingZeros(maxVal) - 1; + preShift -= SHIFT_BEFORE_SQUARE; + + /* Limit preShift to a maximum value to prevent accumulator overflow in + exceptional situations where the signal in the analysis-buffer is very + small (small maxVal). + */ + preShift = fMin(preShift, (SCHAR)25); + + accu = FL2FXCONST_DBL(0.0f); + if (preShift >= 0) { + int l; + if (analysBufferImag != NULL) { + for (l = start_pos; l < next_pos; l++) { + FIXP_DBL temp1 = bufferReal[l] << (int)preShift; + FIXP_DBL temp2 = bufferImag[l] << (int)preShift; + accu = fPow2AddDiv2(accu, temp1); + accu = fPow2AddDiv2(accu, temp2); + } + } else { + for (l = start_pos; l < next_pos; l++) { + FIXP_DBL temp = bufferReal[l] << (int)preShift; + accu = fPow2AddDiv2(accu, temp); + } + } + } else { /* if negative shift value */ + int l; + int negpreShift = -preShift; + if (analysBufferImag != NULL) { + for (l = start_pos; l < next_pos; l++) { + FIXP_DBL temp1 = bufferReal[l] >> (int)negpreShift; + FIXP_DBL temp2 = bufferImag[l] >> (int)negpreShift; + accu = fPow2AddDiv2(accu, temp1); + accu = fPow2AddDiv2(accu, temp2); + } + } else { + for (l = start_pos; l < next_pos; l++) { + FIXP_DBL temp = bufferReal[l] >> (int)negpreShift; + accu = fPow2AddDiv2(accu, temp); + } + } + } + accu <<= 1; + + /* Convert double precision to Mantissa/Exponent: */ + shift = fNorm(accu); + sum = accu << (int)shift; + + /* Divide by width of envelope and apply frame scale: */ + *nrgEst++ = fMult(sum, invWidth); + shift += 2 * preShift; + if (analysBufferImag != NULL) + *nrgEst_e++ = frameExp - shift; + else + *nrgEst_e++ = frameExp - shift + 1; /* +1 due to missing imag. part */ + } /* maxVal!=0 */ + else { + /* Prevent a zero-mantissa-number from being misinterpreted + due to its exponent. */ + *nrgEst++ = FL2FXCONST_DBL(0.0f); + *nrgEst_e++ = 0; + } + } +} + +/*! + \brief Estimates the mean energy of each Scale factor band for the + duration of the current envelope. + + This function is used when interpolFreq is false. +*/ +static void calcNrgPerSfb( + FIXP_DBL **analysBufferReal, /*!< Real part of subband samples */ + FIXP_DBL **analysBufferImag, /*!< Imaginary part of subband samples */ + int nSfb, /*!< Number of scale factor bands */ + UCHAR *freqBandTable, /*!< First Subband for each Sfb */ + int start_pos, /*!< First QMF-slot of current envelope */ + int next_pos, /*!< Last QMF-slot of current envelope + 1 */ + SCHAR input_e, /*!< Common exponent for all input samples */ + FIXP_DBL *nrgEst, /*!< resulting Energy (0..1) */ + SCHAR *nrgEst_e) /*!< Exponent of resulting Energy */ +{ + FIXP_SGL invWidth; + FIXP_DBL temp; + SCHAR preShift; + SCHAR shift, sum_e; + FIXP_DBL sum; + + int j, k, l, li, ui; + FIXP_DBL sumAll, sumLine; /* Single precision would be sufficient, + but overflow bits are required for accumulation */ + + /* Divide by width of envelope later: */ + invWidth = FX_DBL2FX_SGL(GetInvInt(next_pos - start_pos)); + /* The common exponent needs to be doubled because all mantissas are squared: + */ + input_e = input_e << 1; + + for (j = 0; j < nSfb; j++) { + li = freqBandTable[j]; + ui = freqBandTable[j + 1]; + + FIXP_DBL maxVal = maxSubbandSample(analysBufferReal, analysBufferImag, li, + ui, start_pos, next_pos); + + if (maxVal != FL2FXCONST_DBL(0.f)) { + preShift = CntLeadingZeros(maxVal) - 1; + + /* If the accu does not provide enough overflow bits, we cannot + shift the samples up to the limit. + Instead, keep up to 3 free bits in each sample, i.e. up to + 6 bits after calculation of square. + Please note the comment on saturated arithmetic above! + */ + preShift -= SHIFT_BEFORE_SQUARE; + + sumAll = FL2FXCONST_DBL(0.0f); + + for (k = li; k < ui; k++) { + sumLine = FL2FXCONST_DBL(0.0f); + + if (analysBufferImag != NULL) { + if (preShift >= 0) { + for (l = start_pos; l < next_pos; l++) { + temp = analysBufferReal[l][k] << (int)preShift; + sumLine += fPow2Div2(temp); + temp = analysBufferImag[l][k] << (int)preShift; + sumLine += fPow2Div2(temp); + } + } else { + for (l = start_pos; l < next_pos; l++) { + temp = analysBufferReal[l][k] >> -(int)preShift; + sumLine += fPow2Div2(temp); + temp = analysBufferImag[l][k] >> -(int)preShift; + sumLine += fPow2Div2(temp); + } + } + } else { + if (preShift >= 0) { + for (l = start_pos; l < next_pos; l++) { + temp = analysBufferReal[l][k] << (int)preShift; + sumLine += fPow2Div2(temp); + } + } else { + for (l = start_pos; l < next_pos; l++) { + temp = analysBufferReal[l][k] >> -(int)preShift; + sumLine += fPow2Div2(temp); + } + } + } + + /* The number of QMF-channels per SBR bands may be up to 15. + Shift right to avoid overflows in sum over all channels. */ + sumLine = sumLine >> (4 - 1); + sumAll += sumLine; + } + + /* Convert double precision to Mantissa/Exponent: */ + shift = fNorm(sumAll); + sum = sumAll << (int)shift; + + /* Divide by width of envelope: */ + sum = fMult(sum, invWidth); + + /* Divide by width of Sfb: */ + sum = fMult(sum, FX_DBL2FX_SGL(GetInvInt(ui - li))); + + /* Set all Subband energies in the Sfb to the average energy: */ + if (analysBufferImag != NULL) + sum_e = input_e + 4 - shift; /* -4 to compensate right-shift */ + else + sum_e = input_e + 4 + 1 - + shift; /* -4 to compensate right-shift; +1 due to missing + imag. part */ + + sum_e -= 2 * preShift; + } /* maxVal!=0 */ + else { + /* Prevent a zero-mantissa-number from being misinterpreted + due to its exponent. */ + sum = FL2FXCONST_DBL(0.0f); + sum_e = 0; + } + + for (k = li; k < ui; k++) { + *nrgEst++ = sum; + *nrgEst_e++ = sum_e; + } + } +} + +/*! + \brief Calculate gain, noise, and additional sine level for one subband. + + The resulting energy gain is given by mantissa and exponent. +*/ +static void calcSubbandGain( + FIXP_DBL nrgRef, /*!< Reference Energy according to envelope data */ + SCHAR + nrgRef_e, /*!< Reference Energy according to envelope data (exponent) */ + ENV_CALC_NRGS *nrgs, int i, FIXP_DBL tmpNoise, /*!< Relative noise level */ + SCHAR tmpNoise_e, /*!< Relative noise level (exponent) */ + UCHAR sinePresentFlag, /*!< Indicates if sine is present on band */ + UCHAR sineMapped, /*!< Indicates if sine must be added */ + int noNoiseFlag) /*!< Flag to suppress noise addition */ +{ + FIXP_DBL nrgEst = nrgs->nrgEst[i]; /*!< Energy in transposed signal */ + SCHAR nrgEst_e = + nrgs->nrgEst_e[i]; /*!< Energy in transposed signal (exponent) */ + FIXP_DBL *ptrNrgGain = &nrgs->nrgGain[i]; /*!< Resulting energy gain */ + SCHAR *ptrNrgGain_e = + &nrgs->nrgGain_e[i]; /*!< Resulting energy gain (exponent) */ + FIXP_DBL *ptrNoiseLevel = + &nrgs->noiseLevel[i]; /*!< Resulting absolute noise energy */ + SCHAR *ptrNoiseLevel_e = + &nrgs->noiseLevel_e[i]; /*!< Resulting absolute noise energy (exponent) */ + FIXP_DBL *ptrNrgSine = &nrgs->nrgSine[i]; /*!< Additional sine energy */ + SCHAR *ptrNrgSine_e = + &nrgs->nrgSine_e[i]; /*!< Additional sine energy (exponent) */ + + FIXP_DBL a, b, c; + SCHAR a_e, b_e, c_e; + + /* + This addition of 1 prevents divisions by zero in the reference code. + For very small energies in nrgEst, it prevents the gains from becoming + very high which could cause some trouble due to the smoothing. + */ + b_e = (int)(nrgEst_e - 1); + if (b_e >= 0) { + nrgEst = (FL2FXCONST_DBL(0.5f) >> (INT)fixMin(b_e + 1, DFRACT_BITS - 1)) + + (nrgEst >> 1); + nrgEst_e += 1; /* shift by 1 bit to avoid overflow */ + + } else { + nrgEst = (nrgEst >> (INT)(fixMin(-b_e + 1, DFRACT_BITS - 1))) + + (FL2FXCONST_DBL(0.5f) >> 1); + nrgEst_e = 2; /* shift by 1 bit to avoid overflow */ + } + + /* A = NrgRef * TmpNoise */ + a = fMult(nrgRef, tmpNoise); + a_e = nrgRef_e + tmpNoise_e; + + /* B = 1 + TmpNoise */ + b_e = (int)(tmpNoise_e - 1); + if (b_e >= 0) { + b = (FL2FXCONST_DBL(0.5f) >> (INT)fixMin(b_e + 1, DFRACT_BITS - 1)) + + (tmpNoise >> 1); + b_e = tmpNoise_e + 1; /* shift by 1 bit to avoid overflow */ + } else { + b = (tmpNoise >> (INT)(fixMin(-b_e + 1, DFRACT_BITS - 1))) + + (FL2FXCONST_DBL(0.5f) >> 1); + b_e = 2; /* shift by 1 bit to avoid overflow */ + } + + /* noiseLevel = A / B = (NrgRef * TmpNoise) / (1 + TmpNoise) */ + FDK_divide_MantExp(a, a_e, b, b_e, ptrNoiseLevel, ptrNoiseLevel_e); + + if (sinePresentFlag) { + /* C = (1 + TmpNoise) * NrgEst */ + c = fMult(b, nrgEst); + c_e = b_e + nrgEst_e; + + /* gain = A / C = (NrgRef * TmpNoise) / (1 + TmpNoise) * NrgEst */ + FDK_divide_MantExp(a, a_e, c, c_e, ptrNrgGain, ptrNrgGain_e); + + if (sineMapped) { + /* sineLevel = nrgRef/ (1 + TmpNoise) */ + FDK_divide_MantExp(nrgRef, nrgRef_e, b, b_e, ptrNrgSine, ptrNrgSine_e); + } + } else { + if (noNoiseFlag) { + /* B = NrgEst */ + b = nrgEst; + b_e = nrgEst_e; + } else { + /* B = NrgEst * (1 + TmpNoise) */ + b = fMult(b, nrgEst); + b_e = b_e + nrgEst_e; + } + + /* gain = nrgRef / B */ + FDK_divide_MantExp(nrgRef, nrgRef_e, b, b_e, ptrNrgGain, ptrNrgGain_e); + } +} + +/*! + \brief Calculate "average gain" for the specified subband range. + + This is rather a gain of the average magnitude than the average + of gains! + The result is used as a relative limit for all gains within the + current "limiter band" (a certain frequency range). +*/ +static void calcAvgGain( + ENV_CALC_NRGS *nrgs, int lowSubband, /*!< Begin of the limiter band */ + int highSubband, /*!< High end of the limiter band */ + FIXP_DBL *ptrSumRef, SCHAR *ptrSumRef_e, + FIXP_DBL *ptrAvgGain, /*!< Resulting overall gain (mantissa) */ + SCHAR *ptrAvgGain_e) /*!< Resulting overall gain (exponent) */ +{ + FIXP_DBL *nrgRef = + nrgs->nrgRef; /*!< Reference Energy according to envelope data */ + SCHAR *nrgRef_e = + nrgs->nrgRef_e; /*!< Reference Energy according to envelope data + (exponent) */ + FIXP_DBL *nrgEst = nrgs->nrgEst; /*!< Energy in transposed signal */ + SCHAR *nrgEst_e = + nrgs->nrgEst_e; /*!< Energy in transposed signal (exponent) */ + + FIXP_DBL sumRef = 1; + FIXP_DBL sumEst = 1; + SCHAR sumRef_e = -FRACT_BITS; + SCHAR sumEst_e = -FRACT_BITS; + int k; + + for (k = lowSubband; k < highSubband; k++) { + /* Add nrgRef[k] to sumRef: */ + FDK_add_MantExp(sumRef, sumRef_e, nrgRef[k], nrgRef_e[k], &sumRef, + &sumRef_e); + + /* Add nrgEst[k] to sumEst: */ + FDK_add_MantExp(sumEst, sumEst_e, nrgEst[k], nrgEst_e[k], &sumEst, + &sumEst_e); + } + + FDK_divide_MantExp(sumRef, sumRef_e, sumEst, sumEst_e, ptrAvgGain, + ptrAvgGain_e); + + *ptrSumRef = sumRef; + *ptrSumRef_e = sumRef_e; +} + +static void adjustTimeSlot_EldGrid( + FIXP_DBL *RESTRICT + ptrReal, /*!< Subband samples to be adjusted, real part */ + ENV_CALC_NRGS *nrgs, UCHAR *ptrHarmIndex, /*!< Harmonic index */ + int lowSubband, /*!< Lowest QMF-channel in the currently used SBR range. */ + int noSubbands, /*!< Number of QMF subbands */ + int scale_change, /*!< Number of bits to shift adjusted samples */ + int noNoiseFlag, /*!< Flag to suppress noise addition */ + int *ptrPhaseIndex, /*!< Start index to random number array */ + int scale_diff_low) /*!< */ + +{ + int k; + FIXP_DBL signalReal, sbNoise; + int tone_count = 0; + + FIXP_DBL *pGain = nrgs->nrgGain; /*!< Gains of current envelope */ + FIXP_DBL *RESTRICT pNoiseLevel = + nrgs->noiseLevel; /*!< Noise levels of current envelope */ + FIXP_DBL *RESTRICT pSineLevel = nrgs->nrgSine; /*!< Sine levels */ + + int phaseIndex = *ptrPhaseIndex; + UCHAR harmIndex = *ptrHarmIndex; + + static const INT harmonicPhase[4][2] = {{1, 0}, {0, 1}, {-1, 0}, {0, -1}}; + + static const FIXP_DBL harmonicPhaseX[4][2] = { + {FL2FXCONST_DBL(2.0 * 1.245183154539139e-001), + FL2FXCONST_DBL(2.0 * 1.245183154539139e-001)}, + {FL2FXCONST_DBL(2.0 * -1.123767859325028e-001), + FL2FXCONST_DBL(2.0 * 1.123767859325028e-001)}, + {FL2FXCONST_DBL(2.0 * -1.245183154539139e-001), + FL2FXCONST_DBL(2.0 * -1.245183154539139e-001)}, + {FL2FXCONST_DBL(2.0 * 1.123767859325028e-001), + FL2FXCONST_DBL(2.0 * -1.123767859325028e-001)}}; + + const FIXP_DBL *p_harmonicPhaseX = &harmonicPhaseX[harmIndex][0]; + const INT *p_harmonicPhase = &harmonicPhase[harmIndex][0]; + + *(ptrReal - 1) = fAddSaturate( + *(ptrReal - 1), + SATURATE_SHIFT(fMultDiv2(p_harmonicPhaseX[lowSubband & 1], pSineLevel[0]), + scale_diff_low, DFRACT_BITS)); + FIXP_DBL pSineLevel_prev = (FIXP_DBL)0; + + int idx_k = lowSubband & 1; + + for (k = 0; k < noSubbands; k++) { + FIXP_DBL sineLevel_curr = *pSineLevel++; + phaseIndex = (phaseIndex + 1) & (SBR_NF_NO_RANDOM_VAL - 1); + + signalReal = fMultDiv2(*ptrReal, *pGain++) << ((int)scale_change); + sbNoise = *pNoiseLevel++; + if (((INT)sineLevel_curr | noNoiseFlag) == 0) { + signalReal += + (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[phaseIndex][0], sbNoise) + << 4); + } + signalReal += sineLevel_curr * p_harmonicPhase[0]; + signalReal = + fMultAddDiv2(signalReal, pSineLevel_prev, p_harmonicPhaseX[idx_k]); + pSineLevel_prev = sineLevel_curr; + idx_k = !idx_k; + if (k < noSubbands - 1) { + signalReal = + fMultAddDiv2(signalReal, pSineLevel[0], p_harmonicPhaseX[idx_k]); + } else /* (k == noSubbands - 1) */ + { + if (k + lowSubband + 1 < 63) { + *(ptrReal + 1) += fMultDiv2(pSineLevel_prev, p_harmonicPhaseX[idx_k]); + } + } + *ptrReal++ = signalReal; + + if (pSineLevel_prev != FL2FXCONST_DBL(0.0f)) { + if (++tone_count == 16) { + k++; + break; + } + } + } + /* Run again, if previous loop got breaked with tone_count = 16 */ + for (; k < noSubbands; k++) { + FIXP_DBL sineLevel_curr = *pSineLevel++; + phaseIndex = (phaseIndex + 1) & (SBR_NF_NO_RANDOM_VAL - 1); + + signalReal = fMultDiv2(*ptrReal, *pGain++) << ((int)scale_change); + sbNoise = *pNoiseLevel++; + if (((INT)sineLevel_curr | noNoiseFlag) == 0) { + signalReal += + (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[phaseIndex][0], sbNoise) + << 4); + } + signalReal += sineLevel_curr * p_harmonicPhase[0]; + *ptrReal++ = signalReal; + } + + *ptrHarmIndex = (harmIndex + 1) & 3; + *ptrPhaseIndex = phaseIndex & (SBR_NF_NO_RANDOM_VAL - 1); +} + +/*! + \brief Amplify one timeslot of the signal with the calculated gains + and add the noisefloor. +*/ + +static void adjustTimeSlotLC( + FIXP_DBL *ptrReal, /*!< Subband samples to be adjusted, real part */ + ENV_CALC_NRGS *nrgs, UCHAR *ptrHarmIndex, /*!< Harmonic index */ + int lowSubband, /*!< Lowest QMF-channel in the currently used SBR range. */ + int noSubbands, /*!< Number of QMF subbands */ + int scale_change, /*!< Number of bits to shift adjusted samples */ + int noNoiseFlag, /*!< Flag to suppress noise addition */ + int *ptrPhaseIndex) /*!< Start index to random number array */ +{ + FIXP_DBL *pGain = nrgs->nrgGain; /*!< Gains of current envelope */ + FIXP_DBL *pNoiseLevel = + nrgs->noiseLevel; /*!< Noise levels of current envelope */ + FIXP_DBL *pSineLevel = nrgs->nrgSine; /*!< Sine levels */ + + int k; + int index = *ptrPhaseIndex; + UCHAR harmIndex = *ptrHarmIndex; + UCHAR freqInvFlag = (lowSubband & 1); + FIXP_DBL signalReal, sineLevel, sineLevelNext, sineLevelPrev; + int tone_count = 0; + int sineSign = 1; + +#define C1 ((FIXP_SGL)FL2FXCONST_SGL(2.f * 0.00815f)) +#define C1_CLDFB ((FIXP_SGL)FL2FXCONST_SGL(2.f * 0.16773f)) + + /* + First pass for k=0 pulled out of the loop: + */ + + index = (index + 1) & (SBR_NF_NO_RANDOM_VAL - 1); + + /* + The next multiplication constitutes the actual envelope adjustment + of the signal and should be carried out with full accuracy + (supplying #FRACT_BITS valid bits). + */ + signalReal = fMultDiv2(*ptrReal, *pGain++) << ((int)scale_change); + sineLevel = *pSineLevel++; + sineLevelNext = (noSubbands > 1) ? pSineLevel[0] : FL2FXCONST_DBL(0.0f); + + if (sineLevel != FL2FXCONST_DBL(0.0f)) + tone_count++; + else if (!noNoiseFlag) + /* Add noisefloor to the amplified signal */ + signalReal += + (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0]) + << 4); + + { + if (!(harmIndex & 0x1)) { + /* harmIndex 0,2 */ + signalReal += (harmIndex & 0x2) ? -sineLevel : sineLevel; + *ptrReal++ = signalReal; + } else { + /* harmIndex 1,3 in combination with freqInvFlag */ + int shift = (int)(scale_change + 1); + shift = (shift >= 0) ? fixMin(DFRACT_BITS - 1, shift) + : fixMax(-(DFRACT_BITS - 1), shift); + + FIXP_DBL tmp1 = (shift >= 0) ? (fMultDiv2(C1, sineLevel) >> shift) + : (fMultDiv2(C1, sineLevel) << (-shift)); + FIXP_DBL tmp2 = fMultDiv2(C1, sineLevelNext); + + /* save switch and compare operations and reduce to XOR statement */ + if (((harmIndex >> 1) & 0x1) ^ freqInvFlag) { + *(ptrReal - 1) += tmp1; + signalReal -= tmp2; + } else { + *(ptrReal - 1) -= tmp1; + signalReal += tmp2; + } + *ptrReal++ = signalReal; + freqInvFlag = !freqInvFlag; + } + } + + pNoiseLevel++; + + if (noSubbands > 2) { + if (!(harmIndex & 0x1)) { + /* harmIndex 0,2 */ + if (!harmIndex) { + sineSign = 0; + } + + for (k = noSubbands - 2; k != 0; k--) { + FIXP_DBL sinelevel = *pSineLevel++; + index++; + if (((signalReal = (sineSign ? -sinelevel : sinelevel)) == + FL2FXCONST_DBL(0.0f)) && + !noNoiseFlag) { + /* Add noisefloor to the amplified signal */ + index &= (SBR_NF_NO_RANDOM_VAL - 1); + signalReal += (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], + pNoiseLevel[0]) + << 4); + } + + /* The next multiplication constitutes the actual envelope adjustment of + * the signal. */ + signalReal += fMultDiv2(*ptrReal, *pGain++) << ((int)scale_change); + + pNoiseLevel++; + *ptrReal++ = signalReal; + } /* for ... */ + } else { + /* harmIndex 1,3 in combination with freqInvFlag */ + if (harmIndex == 1) freqInvFlag = !freqInvFlag; + + for (k = noSubbands - 2; k != 0; k--) { + index++; + /* The next multiplication constitutes the actual envelope adjustment of + * the signal. */ + signalReal = fMultDiv2(*ptrReal, *pGain++) << ((int)scale_change); + + if (*pSineLevel++ != FL2FXCONST_DBL(0.0f)) + tone_count++; + else if (!noNoiseFlag) { + /* Add noisefloor to the amplified signal */ + index &= (SBR_NF_NO_RANDOM_VAL - 1); + signalReal += (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], + pNoiseLevel[0]) + << 4); + } + + pNoiseLevel++; + + if (tone_count <= 16) { + FIXP_DBL addSine = fMultDiv2((pSineLevel[-2] - pSineLevel[0]), C1); + signalReal += (freqInvFlag) ? (-addSine) : (addSine); + } + + *ptrReal++ = signalReal; + freqInvFlag = !freqInvFlag; + } /* for ... */ + } + } + + if (noSubbands > -1) { + index++; + /* The next multiplication constitutes the actual envelope adjustment of the + * signal. */ + signalReal = fMultDiv2(*ptrReal, *pGain) << ((int)scale_change); + sineLevelPrev = fMultDiv2(pSineLevel[-1], FL2FX_SGL(0.0163f)); + sineLevel = pSineLevel[0]; + + if (pSineLevel[0] != FL2FXCONST_DBL(0.0f)) + tone_count++; + else if (!noNoiseFlag) { + /* Add noisefloor to the amplified signal */ + index &= (SBR_NF_NO_RANDOM_VAL - 1); + signalReal = + signalReal + + (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0]) + << 4); + } + + if (!(harmIndex & 0x1)) { + /* harmIndex 0,2 */ + *ptrReal = signalReal + ((sineSign) ? -sineLevel : sineLevel); + } else { + /* harmIndex 1,3 in combination with freqInvFlag */ + if (tone_count <= 16) { + if (freqInvFlag) { + *ptrReal++ = signalReal - sineLevelPrev; + if (noSubbands + lowSubband < 63) + *ptrReal = *ptrReal + fMultDiv2(C1, sineLevel); + } else { + *ptrReal++ = signalReal + sineLevelPrev; + if (noSubbands + lowSubband < 63) + *ptrReal = *ptrReal - fMultDiv2(C1, sineLevel); + } + } else + *ptrReal = signalReal; + } + } + *ptrHarmIndex = (harmIndex + 1) & 3; + *ptrPhaseIndex = index & (SBR_NF_NO_RANDOM_VAL - 1); +} + +static void adjustTimeSlotHQ_GainAndNoise( + FIXP_DBL *RESTRICT + ptrReal, /*!< Subband samples to be adjusted, real part */ + FIXP_DBL *RESTRICT + ptrImag, /*!< Subband samples to be adjusted, imag part */ + HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, ENV_CALC_NRGS *nrgs, + int lowSubband, /*!< Lowest QMF-channel in the currently used SBR range. */ + int noSubbands, /*!< Number of QMF subbands */ + int scale_change, /*!< Number of bits to shift adjusted samples */ + FIXP_SGL smooth_ratio, /*!< Impact of last envelope */ + int noNoiseFlag, /*!< Start index to random number array */ + int filtBufferNoiseShift) /*!< Shift factor of filtBufferNoise */ +{ + FIXP_DBL *RESTRICT gain = nrgs->nrgGain; /*!< Gains of current envelope */ + FIXP_DBL *RESTRICT noiseLevel = + nrgs->noiseLevel; /*!< Noise levels of current envelope */ + FIXP_DBL *RESTRICT pSineLevel = nrgs->nrgSine; /*!< Sine levels */ + + FIXP_DBL *RESTRICT filtBuffer = + h_sbr_cal_env->filtBuffer; /*!< Gains of last envelope */ + FIXP_DBL *RESTRICT filtBufferNoise = + h_sbr_cal_env->filtBufferNoise; /*!< Noise levels of last envelope */ + int *RESTRICT ptrPhaseIndex = + &h_sbr_cal_env->phaseIndex; /*!< Start index to random number array */ + + int k; + FIXP_DBL signalReal, signalImag; + FIXP_DBL noiseReal, noiseImag; + FIXP_DBL smoothedGain, smoothedNoise; + FIXP_SGL direct_ratio = + /*FL2FXCONST_SGL(1.0f) */ (FIXP_SGL)MAXVAL_SGL - smooth_ratio; + int index = *ptrPhaseIndex; + int shift; + + *ptrPhaseIndex = (index + noSubbands) & (SBR_NF_NO_RANDOM_VAL - 1); + + filtBufferNoiseShift += + 1; /* due to later use of fMultDiv2 instead of fMult */ + if (filtBufferNoiseShift < 0) { + shift = fixMin(DFRACT_BITS - 1, -filtBufferNoiseShift); + } else { + shift = fixMin(DFRACT_BITS - 1, filtBufferNoiseShift); + } + + if (smooth_ratio > FL2FXCONST_SGL(0.0f)) { + for (k = 0; k < noSubbands; k++) { + /* + Smoothing: The old envelope has been bufferd and a certain ratio + of the old gains and noise levels is used. + */ + smoothedGain = + fMult(smooth_ratio, filtBuffer[k]) + fMult(direct_ratio, gain[k]); + + if (filtBufferNoiseShift < 0) { + smoothedNoise = (fMultDiv2(smooth_ratio, filtBufferNoise[k]) >> shift) + + fMult(direct_ratio, noiseLevel[k]); + } else { + smoothedNoise = (fMultDiv2(smooth_ratio, filtBufferNoise[k]) << shift) + + fMult(direct_ratio, noiseLevel[k]); + } + + /* + The next 2 multiplications constitute the actual envelope adjustment + of the signal and should be carried out with full accuracy + (supplying #DFRACT_BITS valid bits). + */ + signalReal = fMultDiv2(*ptrReal, smoothedGain) << ((int)scale_change); + signalImag = fMultDiv2(*ptrImag, smoothedGain) << ((int)scale_change); + + index++; + + if ((pSineLevel[k] != FL2FXCONST_DBL(0.0f)) || noNoiseFlag) { + /* Just the amplified signal is saved */ + *ptrReal++ = signalReal; + *ptrImag++ = signalImag; + } else { + /* Add noisefloor to the amplified signal */ + index &= (SBR_NF_NO_RANDOM_VAL - 1); + noiseReal = + fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], smoothedNoise) + << 4; + noiseImag = + fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][1], smoothedNoise) + << 4; + *ptrReal++ = (signalReal + noiseReal); + *ptrImag++ = (signalImag + noiseImag); + } + } + } else { + for (k = 0; k < noSubbands; k++) { + smoothedGain = gain[k]; + signalReal = fMultDiv2(*ptrReal, smoothedGain) << scale_change; + signalImag = fMultDiv2(*ptrImag, smoothedGain) << scale_change; + + index++; + + if ((pSineLevel[k] == FL2FXCONST_DBL(0.0f)) && (noNoiseFlag == 0)) { + /* Add noisefloor to the amplified signal */ + smoothedNoise = noiseLevel[k]; + index &= (SBR_NF_NO_RANDOM_VAL - 1); + noiseReal = + fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], smoothedNoise); + noiseImag = + fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][1], smoothedNoise); + + /* FDK_sbrDecoder_sbr_randomPhase is downscaled by 2^3 */ + signalReal += noiseReal << 4; + signalImag += noiseImag << 4; + } + *ptrReal++ = signalReal; + *ptrImag++ = signalImag; + } + } +} + +static void adjustTimeSlotHQ_AddHarmonics( + FIXP_DBL *RESTRICT + ptrReal, /*!< Subband samples to be adjusted, real part */ + FIXP_DBL *RESTRICT + ptrImag, /*!< Subband samples to be adjusted, imag part */ + HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, ENV_CALC_NRGS *nrgs, + int lowSubband, /*!< Lowest QMF-channel in the currently used SBR range. */ + int noSubbands, /*!< Number of QMF subbands */ + int scale_change /*!< Scale mismatch between QMF input and sineLevel + exponent. */ +) { + FIXP_DBL *RESTRICT pSineLevel = nrgs->nrgSine; /*!< Sine levels */ + UCHAR *RESTRICT ptrHarmIndex = + &h_sbr_cal_env->harmIndex; /*!< Harmonic index */ + + int k; + FIXP_DBL signalReal, signalImag; + UCHAR harmIndex = *ptrHarmIndex; + int freqInvFlag = (lowSubband & 1); + FIXP_DBL sineLevel; + + *ptrHarmIndex = (harmIndex + 1) & 3; + + for (k = 0; k < noSubbands; k++) { + sineLevel = pSineLevel[k]; + freqInvFlag ^= 1; + if (sineLevel != FL2FXCONST_DBL(0.f)) { + signalReal = ptrReal[k]; + signalImag = ptrImag[k]; + sineLevel = scaleValue(sineLevel, scale_change); + if (harmIndex & 2) { + /* case 2,3 */ + sineLevel = -sineLevel; + } + if (!(harmIndex & 1)) { + /* case 0,2: */ + ptrReal[k] = signalReal + sineLevel; + } else { + /* case 1,3 */ + if (!freqInvFlag) sineLevel = -sineLevel; + ptrImag[k] = signalImag + sineLevel; + } + } + } +} + +static void adjustTimeSlotHQ( + FIXP_DBL *RESTRICT + ptrReal, /*!< Subband samples to be adjusted, real part */ + FIXP_DBL *RESTRICT + ptrImag, /*!< Subband samples to be adjusted, imag part */ + HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, ENV_CALC_NRGS *nrgs, + int lowSubband, /*!< Lowest QMF-channel in the currently used SBR range. */ + int noSubbands, /*!< Number of QMF subbands */ + int scale_change, /*!< Number of bits to shift adjusted samples */ + FIXP_SGL smooth_ratio, /*!< Impact of last envelope */ + int noNoiseFlag, /*!< Start index to random number array */ + int filtBufferNoiseShift) /*!< Shift factor of filtBufferNoise */ +{ + FIXP_DBL *RESTRICT gain = nrgs->nrgGain; /*!< Gains of current envelope */ + FIXP_DBL *RESTRICT noiseLevel = + nrgs->noiseLevel; /*!< Noise levels of current envelope */ + FIXP_DBL *RESTRICT pSineLevel = nrgs->nrgSine; /*!< Sine levels */ + + FIXP_DBL *RESTRICT filtBuffer = + h_sbr_cal_env->filtBuffer; /*!< Gains of last envelope */ + FIXP_DBL *RESTRICT filtBufferNoise = + h_sbr_cal_env->filtBufferNoise; /*!< Noise levels of last envelope */ + UCHAR *RESTRICT ptrHarmIndex = + &h_sbr_cal_env->harmIndex; /*!< Harmonic index */ + int *RESTRICT ptrPhaseIndex = + &h_sbr_cal_env->phaseIndex; /*!< Start index to random number array */ + + int k; + FIXP_DBL signalReal, signalImag; + FIXP_DBL noiseReal, noiseImag; + FIXP_DBL smoothedGain, smoothedNoise; + FIXP_SGL direct_ratio = + /*FL2FXCONST_SGL(1.0f) */ (FIXP_SGL)MAXVAL_SGL - smooth_ratio; + int index = *ptrPhaseIndex; + UCHAR harmIndex = *ptrHarmIndex; + int freqInvFlag = (lowSubband & 1); + FIXP_DBL sineLevel; + int shift; + + *ptrPhaseIndex = (index + noSubbands) & (SBR_NF_NO_RANDOM_VAL - 1); + *ptrHarmIndex = (harmIndex + 1) & 3; + + /* + Possible optimization: + smooth_ratio and harmIndex stay constant during the loop. + It might be faster to include a separate loop in each path. + + the check for smooth_ratio is now outside the loop and the workload + of the whole function decreased by about 20 % + */ + + filtBufferNoiseShift += + 1; /* due to later use of fMultDiv2 instead of fMult */ + if (filtBufferNoiseShift < 0) + shift = fixMin(DFRACT_BITS - 1, -filtBufferNoiseShift); + else + shift = fixMin(DFRACT_BITS - 1, filtBufferNoiseShift); + + if (smooth_ratio > FL2FXCONST_SGL(0.0f)) { + for (k = 0; k < noSubbands; k++) { + /* + Smoothing: The old envelope has been bufferd and a certain ratio + of the old gains and noise levels is used. + */ + + smoothedGain = + fMult(smooth_ratio, filtBuffer[k]) + fMult(direct_ratio, gain[k]); + + if (filtBufferNoiseShift < 0) { + smoothedNoise = (fMultDiv2(smooth_ratio, filtBufferNoise[k]) >> shift) + + fMult(direct_ratio, noiseLevel[k]); + } else { + smoothedNoise = (fMultDiv2(smooth_ratio, filtBufferNoise[k]) << shift) + + fMult(direct_ratio, noiseLevel[k]); + } + + /* + The next 2 multiplications constitute the actual envelope adjustment + of the signal and should be carried out with full accuracy + (supplying #DFRACT_BITS valid bits). + */ + signalReal = fMultDiv2(*ptrReal, smoothedGain) << ((int)scale_change); + signalImag = fMultDiv2(*ptrImag, smoothedGain) << ((int)scale_change); + + index++; + + if (pSineLevel[k] != FL2FXCONST_DBL(0.0f)) { + sineLevel = pSineLevel[k]; + + switch (harmIndex) { + case 0: + *ptrReal++ = (signalReal + sineLevel); + *ptrImag++ = (signalImag); + break; + case 2: + *ptrReal++ = (signalReal - sineLevel); + *ptrImag++ = (signalImag); + break; + case 1: + *ptrReal++ = (signalReal); + if (freqInvFlag) + *ptrImag++ = (signalImag - sineLevel); + else + *ptrImag++ = (signalImag + sineLevel); + break; + case 3: + *ptrReal++ = signalReal; + if (freqInvFlag) + *ptrImag++ = (signalImag + sineLevel); + else + *ptrImag++ = (signalImag - sineLevel); + break; + } + } else { + if (noNoiseFlag) { + /* Just the amplified signal is saved */ + *ptrReal++ = (signalReal); + *ptrImag++ = (signalImag); + } else { + /* Add noisefloor to the amplified signal */ + index &= (SBR_NF_NO_RANDOM_VAL - 1); + /* FDK_sbrDecoder_sbr_randomPhase is downscaled by 2^3 */ + noiseReal = + fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], smoothedNoise) + << 4; + noiseImag = + fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][1], smoothedNoise) + << 4; + *ptrReal++ = (signalReal + noiseReal); + *ptrImag++ = (signalImag + noiseImag); + } + } + freqInvFlag ^= 1; + } + + } else { + for (k = 0; k < noSubbands; k++) { + smoothedGain = gain[k]; + signalReal = fMultDiv2(*ptrReal, smoothedGain) << scale_change; + signalImag = fMultDiv2(*ptrImag, smoothedGain) << scale_change; + + index++; + + if ((sineLevel = pSineLevel[k]) != FL2FXCONST_DBL(0.0f)) { + switch (harmIndex) { + case 0: + signalReal += sineLevel; + break; + case 1: + if (freqInvFlag) + signalImag -= sineLevel; + else + signalImag += sineLevel; + break; + case 2: + signalReal -= sineLevel; + break; + case 3: + if (freqInvFlag) + signalImag += sineLevel; + else + signalImag -= sineLevel; + break; + } + } else { + if (noNoiseFlag == 0) { + /* Add noisefloor to the amplified signal */ + smoothedNoise = noiseLevel[k]; + index &= (SBR_NF_NO_RANDOM_VAL - 1); + noiseReal = fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], + smoothedNoise); + noiseImag = fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][1], + smoothedNoise); + + /* FDK_sbrDecoder_sbr_randomPhase is downscaled by 2^3 */ + signalReal += noiseReal << 4; + signalImag += noiseImag << 4; + } + } + *ptrReal++ = signalReal; + *ptrImag++ = signalImag; + + freqInvFlag ^= 1; + } + } +} + +/*! + \brief Reset limiter bands. + + Build frequency band table for the gain limiter dependent on + the previously generated transposer patch areas. + + \return SBRDEC_OK if ok, SBRDEC_UNSUPPORTED_CONFIG on error +*/ +SBR_ERROR +ResetLimiterBands( + UCHAR *limiterBandTable, /*!< Resulting band borders in QMF channels */ + UCHAR *noLimiterBands, /*!< Resulting number of limiter band */ + UCHAR *freqBandTable, /*!< Table with possible band borders */ + int noFreqBands, /*!< Number of bands in freqBandTable */ + const PATCH_PARAM *patchParam, /*!< Transposer patch parameters */ + int noPatches, /*!< Number of transposer patches */ + int limiterBands, /*!< Selected 'band density' from bitstream */ + UCHAR sbrPatchingMode, int xOverQmf[MAX_NUM_PATCHES], int b41Sbr) { + int i, k, isPatchBorder[2], loLimIndex, hiLimIndex, tempNoLim, nBands; + UCHAR workLimiterBandTable[MAX_FREQ_COEFFS / 2 + MAX_NUM_PATCHES + 1]; + int patchBorders[MAX_NUM_PATCHES + 1]; + int kx, k2; + + int lowSubband = freqBandTable[0]; + int highSubband = freqBandTable[noFreqBands]; + + /* 1 limiter band. */ + if (limiterBands == 0) { + limiterBandTable[0] = 0; + limiterBandTable[1] = highSubband - lowSubband; + nBands = 1; + } else { + if (!sbrPatchingMode && xOverQmf != NULL) { + noPatches = 0; + + if (b41Sbr == 1) { + for (i = 1; i < MAX_NUM_PATCHES_HBE; i++) + if (xOverQmf[i] != 0) noPatches++; + } else { + for (i = 1; i < MAX_STRETCH_HBE; i++) + if (xOverQmf[i] != 0) noPatches++; + } + for (i = 0; i < noPatches; i++) { + patchBorders[i] = xOverQmf[i] - lowSubband; + } + } else { + for (i = 0; i < noPatches; i++) { + patchBorders[i] = patchParam[i].guardStartBand - lowSubband; + } + } + patchBorders[i] = highSubband - lowSubband; + + /* 1.2, 2, or 3 limiter bands/octave plus bandborders at patchborders. */ + for (k = 0; k <= noFreqBands; k++) { + workLimiterBandTable[k] = freqBandTable[k] - lowSubband; + } + for (k = 1; k < noPatches; k++) { + workLimiterBandTable[noFreqBands + k] = patchBorders[k]; + } + + tempNoLim = nBands = noFreqBands + noPatches - 1; + shellsort(workLimiterBandTable, tempNoLim + 1); + + loLimIndex = 0; + hiLimIndex = 1; + + while (hiLimIndex <= tempNoLim) { + FIXP_DBL div_m, oct_m, temp; + INT div_e = 0, oct_e = 0, temp_e = 0; + + k2 = workLimiterBandTable[hiLimIndex] + lowSubband; + kx = workLimiterBandTable[loLimIndex] + lowSubband; + + div_m = fDivNorm(k2, kx, &div_e); + + /* calculate number of octaves */ + oct_m = fLog2(div_m, div_e, &oct_e); + + /* multiply with limiterbands per octave */ + /* values 1, 1.2, 2, 3 -> scale factor of 2 */ + temp = fMultNorm( + oct_m, FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4_DBL[limiterBands], + &temp_e); + + /* overall scale factor of temp ist addition of scalefactors from log2 + calculation, limiter bands scalefactor (2) and limiter bands + multiplication */ + temp_e += oct_e + 2; + + /* div can be a maximum of 64 (k2 = 64 and kx = 1) + -> oct can be a maximum of 6 + -> temp can be a maximum of 18 (as limiterBandsPerOctoave is a maximum + factor of 3) + -> we need a scale factor of 5 for comparisson + */ + if (temp >> (5 - temp_e) < FL2FXCONST_DBL(0.49f) >> 5) { + if (workLimiterBandTable[hiLimIndex] == + workLimiterBandTable[loLimIndex]) { + workLimiterBandTable[hiLimIndex] = highSubband; + nBands--; + hiLimIndex++; + continue; + } + isPatchBorder[0] = isPatchBorder[1] = 0; + for (k = 0; k <= noPatches; k++) { + if (workLimiterBandTable[hiLimIndex] == patchBorders[k]) { + isPatchBorder[1] = 1; + break; + } + } + if (!isPatchBorder[1]) { + workLimiterBandTable[hiLimIndex] = highSubband; + nBands--; + hiLimIndex++; + continue; + } + for (k = 0; k <= noPatches; k++) { + if (workLimiterBandTable[loLimIndex] == patchBorders[k]) { + isPatchBorder[0] = 1; + break; + } + } + if (!isPatchBorder[0]) { + workLimiterBandTable[loLimIndex] = highSubband; + nBands--; + } + } + loLimIndex = hiLimIndex; + hiLimIndex++; + } + shellsort(workLimiterBandTable, tempNoLim + 1); + + /* Test if algorithm exceeded maximum allowed limiterbands */ + if (nBands > MAX_NUM_LIMITERS || nBands <= 0) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + + /* Copy limiterbands from working buffer into final destination */ + for (k = 0; k <= nBands; k++) { + limiterBandTable[k] = workLimiterBandTable[k]; + } + } + *noLimiterBands = nBands; + + return SBRDEC_OK; +} diff --git a/fdk-aac/libSBRdec/src/env_calc.h b/fdk-aac/libSBRdec/src/env_calc.h new file mode 100644 index 0000000..cff365d --- /dev/null +++ b/fdk-aac/libSBRdec/src/env_calc.h @@ -0,0 +1,182 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief Envelope calculation prototypes +*/ +#ifndef ENV_CALC_H +#define ENV_CALC_H + +#include "sbrdecoder.h" +#include "env_extr.h" /* for HANDLE_SBR_HEADER_DATA */ + +typedef struct { + FIXP_DBL filtBuffer[MAX_FREQ_COEFFS]; /*!< previous gains (required for + smoothing) */ + FIXP_DBL filtBufferNoise[MAX_FREQ_COEFFS]; /*!< previous noise levels + (required for smoothing) */ + SCHAR filtBuffer_e[MAX_FREQ_COEFFS]; /*!< Exponents of previous gains */ + SCHAR filtBufferNoise_e; /*!< Common exponent of previous noise levels */ + + int startUp; /*!< flag to signal initial conditions in buffers */ + int phaseIndex; /*!< Index for randomPase array */ + int prevTranEnv; /*!< The transient envelope of the previous frame. */ + + ULONG harmFlagsPrev[ADD_HARMONICS_FLAGS_SIZE]; + /*!< Words with 16 flags each indicating where a sine was added in the + * previous frame.*/ + UCHAR harmIndex; /*!< Current phase of synthetic sine */ + int sbrPatchingMode; /*!< Current patching mode */ + + FIXP_SGL prevSbrNoiseFloorLevel[MAX_NOISE_COEFFS]; + UCHAR prevNNfb; + UCHAR prevNSfb[2]; + UCHAR prevLoSubband; + UCHAR prevHiSubband; + UCHAR prev_ov_highSubband; + UCHAR *prevFreqBandTable[2]; + UCHAR prevFreqBandTableLo[MAX_FREQ_COEFFS / 2 + 1]; + UCHAR prevFreqBandTableHi[MAX_FREQ_COEFFS + 1]; + UCHAR prevFreqBandTableNoise[MAX_NOISE_COEFFS + 1]; + SCHAR sinusoidal_positionPrev; + ULONG harmFlagsPrevActive[ADD_HARMONICS_FLAGS_SIZE]; +} SBR_CALCULATE_ENVELOPE; + +typedef SBR_CALCULATE_ENVELOPE *HANDLE_SBR_CALCULATE_ENVELOPE; + +void calculateSbrEnvelope( + QMF_SCALE_FACTOR *sbrScaleFactor, + HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, + HANDLE_SBR_HEADER_DATA hHeaderData, HANDLE_SBR_FRAME_DATA hFrameData, + PVC_DYNAMIC_DATA *pPvcDynamicData, FIXP_DBL **analysBufferReal, + FIXP_DBL * + *analysBufferImag, /*!< Imag part of subband samples to be processed */ + const int useLP, + FIXP_DBL *degreeAlias, /*!< Estimated aliasing for each QMF channel */ + const UINT flags, const int frameErrorFlag); + +SBR_ERROR +createSbrEnvelopeCalc(HANDLE_SBR_CALCULATE_ENVELOPE hSbrCalculateEnvelope, + HANDLE_SBR_HEADER_DATA hHeaderData, const int chan, + const UINT flags); + +int deleteSbrEnvelopeCalc(HANDLE_SBR_CALCULATE_ENVELOPE hSbrCalculateEnvelope); + +void resetSbrEnvelopeCalc(HANDLE_SBR_CALCULATE_ENVELOPE hCalEnv); + +SBR_ERROR +ResetLimiterBands(UCHAR *limiterBandTable, UCHAR *noLimiterBands, + UCHAR *freqBandTable, int noFreqBands, + const PATCH_PARAM *patchParam, int noPatches, + int limiterBands, UCHAR sbrPatchingMode, + int xOverQmf[MAX_NUM_PATCHES], int sbrRatio); + +void rescaleSubbandSamples(FIXP_DBL **re, FIXP_DBL **im, int lowSubband, + int noSubbands, int start_pos, int next_pos, + int shift); + +FIXP_DBL maxSubbandSample(FIXP_DBL **analysBufferReal_m, + FIXP_DBL **analysBufferImag_m, int lowSubband, + int highSubband, int start_pos, int stop_pos); + +#endif // ENV_CALC_H diff --git a/fdk-aac/libSBRdec/src/env_dec.cpp b/fdk-aac/libSBRdec/src/env_dec.cpp new file mode 100644 index 0000000..95807c9 --- /dev/null +++ b/fdk-aac/libSBRdec/src/env_dec.cpp @@ -0,0 +1,873 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief envelope decoding + This module provides envelope decoding and error concealment algorithms. The + main entry point is decodeSbrData(). + + \sa decodeSbrData(),\ref documentationOverview +*/ + +#include "env_dec.h" + +#include "env_extr.h" +#include "transcendent.h" + +#include "genericStds.h" + +static void decodeEnvelope(HANDLE_SBR_HEADER_DATA hHeaderData, + HANDLE_SBR_FRAME_DATA h_sbr_data, + HANDLE_SBR_PREV_FRAME_DATA h_prev_data, + HANDLE_SBR_PREV_FRAME_DATA h_prev_data_otherChannel); +static void sbr_envelope_unmapping(HANDLE_SBR_HEADER_DATA hHeaderData, + HANDLE_SBR_FRAME_DATA h_data_left, + HANDLE_SBR_FRAME_DATA h_data_right); +static void requantizeEnvelopeData(HANDLE_SBR_FRAME_DATA h_sbr_data, + int ampResolution); +static void deltaToLinearPcmEnvelopeDecoding( + HANDLE_SBR_HEADER_DATA hHeaderData, HANDLE_SBR_FRAME_DATA h_sbr_data, + HANDLE_SBR_PREV_FRAME_DATA h_prev_data); +static void decodeNoiseFloorlevels(HANDLE_SBR_HEADER_DATA hHeaderData, + HANDLE_SBR_FRAME_DATA h_sbr_data, + HANDLE_SBR_PREV_FRAME_DATA h_prev_data); +static void timeCompensateFirstEnvelope(HANDLE_SBR_HEADER_DATA hHeaderData, + HANDLE_SBR_FRAME_DATA h_sbr_data, + HANDLE_SBR_PREV_FRAME_DATA h_prev_data); +static int checkEnvelopeData(HANDLE_SBR_HEADER_DATA hHeaderData, + HANDLE_SBR_FRAME_DATA h_sbr_data, + HANDLE_SBR_PREV_FRAME_DATA h_prev_data); + +#define SBR_ENERGY_PAN_OFFSET (12 << ENV_EXP_FRACT) +#define SBR_MAX_ENERGY (35 << ENV_EXP_FRACT) + +#define DECAY (1 << ENV_EXP_FRACT) + +#if ENV_EXP_FRACT +#define DECAY_COUPLING \ + (1 << (ENV_EXP_FRACT - 1)) /*!< corresponds to a value of 0.5 */ +#else +#define DECAY_COUPLING \ + 1 /*!< If the energy data is not shifted, use 1 instead of 0.5 */ +#endif + +/*! + \brief Convert table index +*/ +static int indexLow2High(int offset, /*!< mapping factor */ + int index, /*!< index to scalefactor band */ + int res) /*!< frequency resolution */ +{ + if (res == 0) { + if (offset >= 0) { + if (index < offset) + return (index); + else + return (2 * index - offset); + } else { + offset = -offset; + if (index < offset) + return (2 * index + index); + else + return (2 * index + offset); + } + } else + return (index); +} + +/*! + \brief Update previous envelope value for delta-coding + + The current envelope values needs to be stored for delta-coding + in the next frame. The stored envelope is always represented with + the high frequency resolution. If the current envelope uses the + low frequency resolution, the energy value will be mapped to the + corresponding high-res bands. +*/ +static void mapLowResEnergyVal( + FIXP_SGL currVal, /*!< current energy value */ + FIXP_SGL *prevData, /*!< pointer to previous data vector */ + int offset, /*!< mapping factor */ + int index, /*!< index to scalefactor band */ + int res) /*!< frequeny resolution */ +{ + if (res == 0) { + if (offset >= 0) { + if (index < offset) + prevData[index] = currVal; + else { + prevData[2 * index - offset] = currVal; + prevData[2 * index + 1 - offset] = currVal; + } + } else { + offset = -offset; + if (index < offset) { + prevData[3 * index] = currVal; + prevData[3 * index + 1] = currVal; + prevData[3 * index + 2] = currVal; + } else { + prevData[2 * index + offset] = currVal; + prevData[2 * index + 1 + offset] = currVal; + } + } + } else + prevData[index] = currVal; +} + +/*! + \brief Convert raw envelope and noisefloor data to energy levels + + This function is being called by sbrDecoder_ParseElement() and provides two + important algorithms: + + First the function decodes envelopes and noise floor levels as described in + requantizeEnvelopeData() and sbr_envelope_unmapping(). The function also + implements concealment algorithms in case there are errors within the sbr + data. For both operations fractional arithmetic is used. Therefore you might + encounter different output values on your target system compared to the + reference implementation. +*/ +void decodeSbrData( + HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ + HANDLE_SBR_FRAME_DATA + h_data_left, /*!< pointer to left channel frame data */ + HANDLE_SBR_PREV_FRAME_DATA + h_prev_data_left, /*!< pointer to left channel previous frame data */ + HANDLE_SBR_FRAME_DATA + h_data_right, /*!< pointer to right channel frame data */ + HANDLE_SBR_PREV_FRAME_DATA + h_prev_data_right) /*!< pointer to right channel previous frame data */ +{ + FIXP_SGL tempSfbNrgPrev[MAX_FREQ_COEFFS]; + int errLeft; + + /* Save previous energy values to be able to reuse them later for concealment. + */ + FDKmemcpy(tempSfbNrgPrev, h_prev_data_left->sfb_nrg_prev, + MAX_FREQ_COEFFS * sizeof(FIXP_SGL)); + + if (hHeaderData->frameErrorFlag || hHeaderData->bs_info.pvc_mode == 0) { + decodeEnvelope(hHeaderData, h_data_left, h_prev_data_left, + h_prev_data_right); + } else { + FDK_ASSERT(h_data_right == NULL); + } + decodeNoiseFloorlevels(hHeaderData, h_data_left, h_prev_data_left); + + if (h_data_right != NULL) { + errLeft = hHeaderData->frameErrorFlag; + decodeEnvelope(hHeaderData, h_data_right, h_prev_data_right, + h_prev_data_left); + decodeNoiseFloorlevels(hHeaderData, h_data_right, h_prev_data_right); + + if (!errLeft && hHeaderData->frameErrorFlag) { + /* If an error occurs in the right channel where the left channel seemed + ok, we apply concealment also on the left channel. This ensures that + the coupling modes of both channels match and that we have the same + number of envelopes in coupling mode. However, as the left channel has + already been processed before, the resulting energy levels are not the + same as if the left channel had been concealed during the first call of + decodeEnvelope(). + */ + /* Restore previous energy values for concealment, because the values have + been overwritten by the first call of decodeEnvelope(). */ + FDKmemcpy(h_prev_data_left->sfb_nrg_prev, tempSfbNrgPrev, + MAX_FREQ_COEFFS * sizeof(FIXP_SGL)); + /* Do concealment */ + decodeEnvelope(hHeaderData, h_data_left, h_prev_data_left, + h_prev_data_right); + } + + if (h_data_left->coupling) { + sbr_envelope_unmapping(hHeaderData, h_data_left, h_data_right); + } + } + + /* Display the data for debugging: */ +} + +/*! + \brief Convert from coupled channels to independent L/R data +*/ +static void sbr_envelope_unmapping( + HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ + HANDLE_SBR_FRAME_DATA h_data_left, /*!< pointer to left channel */ + HANDLE_SBR_FRAME_DATA h_data_right) /*!< pointer to right channel */ +{ + int i; + FIXP_SGL tempL_m, tempR_m, tempRplus1_m, newL_m, newR_m; + SCHAR tempL_e, tempR_e, tempRplus1_e, newL_e, newR_e; + + /* 1. Unmap (already dequantized) coupled envelope energies */ + + for (i = 0; i < h_data_left->nScaleFactors; i++) { + tempR_m = (FIXP_SGL)((LONG)h_data_right->iEnvelope[i] & MASK_M); + tempR_e = (SCHAR)((LONG)h_data_right->iEnvelope[i] & MASK_E); + + tempR_e -= (18 + NRG_EXP_OFFSET); /* -18 = ld(UNMAPPING_SCALE / + h_data_right->nChannels) */ + tempL_m = (FIXP_SGL)((LONG)h_data_left->iEnvelope[i] & MASK_M); + tempL_e = (SCHAR)((LONG)h_data_left->iEnvelope[i] & MASK_E); + + tempL_e -= NRG_EXP_OFFSET; + + /* Calculate tempRight+1 */ + FDK_add_MantExp(tempR_m, tempR_e, FL2FXCONST_SGL(0.5f), 1, /* 1.0 */ + &tempRplus1_m, &tempRplus1_e); + + FDK_divide_MantExp(tempL_m, tempL_e + 1, /* 2 * tempLeft */ + tempRplus1_m, tempRplus1_e, &newR_m, &newR_e); + + if (newR_m >= ((FIXP_SGL)MAXVAL_SGL - ROUNDING)) { + newR_m >>= 1; + newR_e += 1; + } + + newL_m = FX_DBL2FX_SGL(fMult(tempR_m, newR_m)); + newL_e = tempR_e + newR_e; + + h_data_right->iEnvelope[i] = + ((FIXP_SGL)((SHORT)(FIXP_SGL)(newR_m + ROUNDING) & MASK_M)) + + (FIXP_SGL)((SHORT)(FIXP_SGL)(newR_e + NRG_EXP_OFFSET) & MASK_E); + h_data_left->iEnvelope[i] = + ((FIXP_SGL)((SHORT)(FIXP_SGL)(newL_m + ROUNDING) & MASK_M)) + + (FIXP_SGL)((SHORT)(FIXP_SGL)(newL_e + NRG_EXP_OFFSET) & MASK_E); + } + + /* 2. Dequantize and unmap coupled noise floor levels */ + + for (i = 0; i < hHeaderData->freqBandData.nNfb * + h_data_left->frameInfo.nNoiseEnvelopes; + i++) { + tempL_e = (SCHAR)(6 - (LONG)h_data_left->sbrNoiseFloorLevel[i]); + tempR_e = (SCHAR)((LONG)h_data_right->sbrNoiseFloorLevel[i] - + 12) /*SBR_ENERGY_PAN_OFFSET*/; + + /* Calculate tempR+1 */ + FDK_add_MantExp(FL2FXCONST_SGL(0.5f), 1 + tempR_e, /* tempR */ + FL2FXCONST_SGL(0.5f), 1, /* 1.0 */ + &tempRplus1_m, &tempRplus1_e); + + /* Calculate 2*tempLeft/(tempR+1) */ + FDK_divide_MantExp(FL2FXCONST_SGL(0.5f), tempL_e + 2, /* 2 * tempLeft */ + tempRplus1_m, tempRplus1_e, &newR_m, &newR_e); + + /* if (newR_m >= ((FIXP_SGL)MAXVAL_SGL - ROUNDING)) { + newR_m >>= 1; + newR_e += 1; + } */ + + /* L = tempR * R */ + newL_m = newR_m; + newL_e = newR_e + tempR_e; + h_data_right->sbrNoiseFloorLevel[i] = + ((FIXP_SGL)((SHORT)(FIXP_SGL)(newR_m + ROUNDING) & MASK_M)) + + (FIXP_SGL)((SHORT)(FIXP_SGL)(newR_e + NOISE_EXP_OFFSET) & MASK_E); + h_data_left->sbrNoiseFloorLevel[i] = + ((FIXP_SGL)((SHORT)(FIXP_SGL)(newL_m + ROUNDING) & MASK_M)) + + (FIXP_SGL)((SHORT)(FIXP_SGL)(newL_e + NOISE_EXP_OFFSET) & MASK_E); + } +} + +/*! + \brief Simple alternative to the real SBR concealment + + If the real frameInfo is not available due to a frame loss, a replacement will + be constructed with 1 envelope spanning the whole frame (FIX-FIX). + The delta-coded energies are set to negative values, resulting in a fade-down. + In case of coupling, the balance-channel will move towards the center. +*/ +static void leanSbrConcealment( + HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ + HANDLE_SBR_FRAME_DATA h_sbr_data, /*!< pointer to current data */ + HANDLE_SBR_PREV_FRAME_DATA h_prev_data /*!< pointer to data of last frame */ +) { + FIXP_SGL target; /* targeted level for sfb_nrg_prev during fade-down */ + FIXP_SGL step; /* speed of fade */ + int i; + + int currentStartPos = + fMax(0, h_prev_data->stopPos - hHeaderData->numberTimeSlots); + int currentStopPos = hHeaderData->numberTimeSlots; + + /* Use some settings of the previous frame */ + h_sbr_data->ampResolutionCurrentFrame = h_prev_data->ampRes; + h_sbr_data->coupling = h_prev_data->coupling; + for (i = 0; i < MAX_INVF_BANDS; i++) + h_sbr_data->sbr_invf_mode[i] = h_prev_data->sbr_invf_mode[i]; + + /* Generate concealing control data */ + + h_sbr_data->frameInfo.nEnvelopes = 1; + h_sbr_data->frameInfo.borders[0] = currentStartPos; + h_sbr_data->frameInfo.borders[1] = currentStopPos; + h_sbr_data->frameInfo.freqRes[0] = 1; + h_sbr_data->frameInfo.tranEnv = -1; /* no transient */ + h_sbr_data->frameInfo.nNoiseEnvelopes = 1; + h_sbr_data->frameInfo.bordersNoise[0] = currentStartPos; + h_sbr_data->frameInfo.bordersNoise[1] = currentStopPos; + + h_sbr_data->nScaleFactors = hHeaderData->freqBandData.nSfb[1]; + + /* Generate fake envelope data */ + + h_sbr_data->domain_vec[0] = 1; + + if (h_sbr_data->coupling == COUPLING_BAL) { + target = (FIXP_SGL)SBR_ENERGY_PAN_OFFSET; + step = (FIXP_SGL)DECAY_COUPLING; + } else { + target = FL2FXCONST_SGL(0.0f); + step = (FIXP_SGL)DECAY; + } + if (hHeaderData->bs_info.ampResolution == 0) { + target <<= 1; + step <<= 1; + } + + for (i = 0; i < h_sbr_data->nScaleFactors; i++) { + if (h_prev_data->sfb_nrg_prev[i] > target) + h_sbr_data->iEnvelope[i] = -step; + else + h_sbr_data->iEnvelope[i] = step; + } + + /* Noisefloor levels are always cleared ... */ + + h_sbr_data->domain_vec_noise[0] = 1; + FDKmemclear(h_sbr_data->sbrNoiseFloorLevel, + sizeof(h_sbr_data->sbrNoiseFloorLevel)); + + /* ... and so are the sines */ + FDKmemclear(h_sbr_data->addHarmonics, + sizeof(ULONG) * ADD_HARMONICS_FLAGS_SIZE); +} + +/*! + \brief Build reference energies and noise levels from bitstream elements +*/ +static void decodeEnvelope( + HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ + HANDLE_SBR_FRAME_DATA h_sbr_data, /*!< pointer to current data */ + HANDLE_SBR_PREV_FRAME_DATA + h_prev_data, /*!< pointer to data of last frame */ + HANDLE_SBR_PREV_FRAME_DATA + otherChannel /*!< other channel's last frame data */ +) { + int i; + int fFrameError = hHeaderData->frameErrorFlag; + FIXP_SGL tempSfbNrgPrev[MAX_FREQ_COEFFS]; + + if (!fFrameError) { + /* + To avoid distortions after bad frames, set the error flag if delta coding + in time occurs. However, SBR can take a little longer to come up again. + */ + if (h_prev_data->frameErrorFlag) { + if (h_sbr_data->domain_vec[0] != 0) { + fFrameError = 1; + } + } else { + /* Check that the previous stop position and the current start position + match. (Could be done in checkFrameInfo(), but the previous frame data + is not available there) */ + if (h_sbr_data->frameInfo.borders[0] != + h_prev_data->stopPos - hHeaderData->numberTimeSlots) { + /* Both the previous as well as the current frame are flagged to be ok, + * but they do not match! */ + if (h_sbr_data->domain_vec[0] == 1) { + /* Prefer concealment over delta-time coding between the mismatching + * frames */ + fFrameError = 1; + } else { + /* Close the gap in time by triggering timeCompensateFirstEnvelope() + */ + fFrameError = 1; + } + } + } + } + + if (fFrameError) /* Error is detected */ + { + leanSbrConcealment(hHeaderData, h_sbr_data, h_prev_data); + + /* decode the envelope data to linear PCM */ + deltaToLinearPcmEnvelopeDecoding(hHeaderData, h_sbr_data, h_prev_data); + } else /*Do a temporary dummy decoding and check that the envelope values are + within limits */ + { + if (h_prev_data->frameErrorFlag) { + timeCompensateFirstEnvelope(hHeaderData, h_sbr_data, h_prev_data); + if (h_sbr_data->coupling != h_prev_data->coupling) { + /* + Coupling mode has changed during concealment. + The stored energy levels need to be converted. + */ + for (i = 0; i < hHeaderData->freqBandData.nSfb[1]; i++) { + /* Former Level-Channel will be used for both channels */ + if (h_prev_data->coupling == COUPLING_BAL) { + h_prev_data->sfb_nrg_prev[i] = + (otherChannel != NULL) ? otherChannel->sfb_nrg_prev[i] + : (FIXP_SGL)SBR_ENERGY_PAN_OFFSET; + } + /* Former L/R will be combined as the new Level-Channel */ + else if (h_sbr_data->coupling == COUPLING_LEVEL && + otherChannel != NULL) { + h_prev_data->sfb_nrg_prev[i] = (h_prev_data->sfb_nrg_prev[i] + + otherChannel->sfb_nrg_prev[i]) >> + 1; + } else if (h_sbr_data->coupling == COUPLING_BAL) { + h_prev_data->sfb_nrg_prev[i] = (FIXP_SGL)SBR_ENERGY_PAN_OFFSET; + } + } + } + } + FDKmemcpy(tempSfbNrgPrev, h_prev_data->sfb_nrg_prev, + MAX_FREQ_COEFFS * sizeof(FIXP_SGL)); + + deltaToLinearPcmEnvelopeDecoding(hHeaderData, h_sbr_data, h_prev_data); + + fFrameError = checkEnvelopeData(hHeaderData, h_sbr_data, h_prev_data); + + if (fFrameError) { + hHeaderData->frameErrorFlag = 1; + FDKmemcpy(h_prev_data->sfb_nrg_prev, tempSfbNrgPrev, + MAX_FREQ_COEFFS * sizeof(FIXP_SGL)); + decodeEnvelope(hHeaderData, h_sbr_data, h_prev_data, otherChannel); + return; + } + } + + requantizeEnvelopeData(h_sbr_data, h_sbr_data->ampResolutionCurrentFrame); + + hHeaderData->frameErrorFlag = fFrameError; +} + +/*! + \brief Verify that envelope energies are within the allowed range + \return 0 if all is fine, 1 if an envelope value was too high +*/ +static int checkEnvelopeData( + HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ + HANDLE_SBR_FRAME_DATA h_sbr_data, /*!< pointer to current data */ + HANDLE_SBR_PREV_FRAME_DATA h_prev_data /*!< pointer to data of last frame */ +) { + FIXP_SGL *iEnvelope = h_sbr_data->iEnvelope; + FIXP_SGL *sfb_nrg_prev = h_prev_data->sfb_nrg_prev; + int i = 0, errorFlag = 0; + FIXP_SGL sbr_max_energy = (h_sbr_data->ampResolutionCurrentFrame == 1) + ? SBR_MAX_ENERGY + : (SBR_MAX_ENERGY << 1); + + /* + Range check for current energies + */ + for (i = 0; i < h_sbr_data->nScaleFactors; i++) { + if (iEnvelope[i] > sbr_max_energy) { + errorFlag = 1; + } + if (iEnvelope[i] < FL2FXCONST_SGL(0.0f)) { + errorFlag = 1; + /* iEnvelope[i] = FL2FXCONST_SGL(0.0f); */ + } + } + + /* + Range check for previous energies + */ + for (i = 0; i < hHeaderData->freqBandData.nSfb[1]; i++) { + sfb_nrg_prev[i] = fixMax(sfb_nrg_prev[i], FL2FXCONST_SGL(0.0f)); + sfb_nrg_prev[i] = fixMin(sfb_nrg_prev[i], sbr_max_energy); + } + + return (errorFlag); +} + +/*! + \brief Verify that the noise levels are within the allowed range + + The function is equivalent to checkEnvelopeData(). + When the noise-levels are being decoded, it is already too late for + concealment. Therefore the noise levels are simply limited here. +*/ +static void limitNoiseLevels( + HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ + HANDLE_SBR_FRAME_DATA h_sbr_data) /*!< pointer to current data */ +{ + int i; + int nNfb = hHeaderData->freqBandData.nNfb; + +/* + Set range limits. The exact values depend on the coupling mode. + However this limitation is primarily intended to avoid unlimited + accumulation of the delta-coded noise levels. +*/ +#define lowerLimit \ + ((FIXP_SGL)0) /* lowerLimit actually refers to the _highest_ noise energy */ +#define upperLimit \ + ((FIXP_SGL)35) /* upperLimit actually refers to the _lowest_ noise energy */ + + /* + Range check for current noise levels + */ + for (i = 0; i < h_sbr_data->frameInfo.nNoiseEnvelopes * nNfb; i++) { + h_sbr_data->sbrNoiseFloorLevel[i] = + fixMin(h_sbr_data->sbrNoiseFloorLevel[i], upperLimit); + h_sbr_data->sbrNoiseFloorLevel[i] = + fixMax(h_sbr_data->sbrNoiseFloorLevel[i], lowerLimit); + } +} + +/*! + \brief Compensate for the wrong timing that might occur after a frame error. +*/ +static void timeCompensateFirstEnvelope( + HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ + HANDLE_SBR_FRAME_DATA h_sbr_data, /*!< pointer to actual data */ + HANDLE_SBR_PREV_FRAME_DATA + h_prev_data) /*!< pointer to data of last frame */ +{ + int i, nScalefactors; + FRAME_INFO *pFrameInfo = &h_sbr_data->frameInfo; + UCHAR *nSfb = hHeaderData->freqBandData.nSfb; + int estimatedStartPos = + fMax(0, h_prev_data->stopPos - hHeaderData->numberTimeSlots); + int refLen, newLen, shift; + FIXP_SGL deltaExp; + + /* Original length of first envelope according to bitstream */ + refLen = pFrameInfo->borders[1] - pFrameInfo->borders[0]; + /* Corrected length of first envelope (concealing can make the first envelope + * longer) */ + newLen = pFrameInfo->borders[1] - estimatedStartPos; + + if (newLen <= 0) { + /* An envelope length of <= 0 would not work, so we don't use it. + May occur if the previous frame was flagged bad due to a mismatch + of the old and new frame infos. */ + newLen = refLen; + estimatedStartPos = pFrameInfo->borders[0]; + } + + deltaExp = FDK_getNumOctavesDiv8(newLen, refLen); + + /* Shift by -3 to rescale ld-table, ampRes-1 to enable coarser steps */ + shift = (FRACT_BITS - 1 - ENV_EXP_FRACT - 1 + + h_sbr_data->ampResolutionCurrentFrame - 3); + deltaExp = deltaExp >> shift; + pFrameInfo->borders[0] = estimatedStartPos; + pFrameInfo->bordersNoise[0] = estimatedStartPos; + + if (h_sbr_data->coupling != COUPLING_BAL) { + nScalefactors = (pFrameInfo->freqRes[0]) ? nSfb[1] : nSfb[0]; + + for (i = 0; i < nScalefactors; i++) + h_sbr_data->iEnvelope[i] = h_sbr_data->iEnvelope[i] + deltaExp; + } +} + +/*! + \brief Convert each envelope value from logarithmic to linear domain + + Energy levels are transmitted in powers of 2, i.e. only the exponent + is extracted from the bitstream. + Therefore, normally only integer exponents can occur. However during + fading (in case of a corrupt bitstream), a fractional part can also + occur. The data in the array iEnvelope is shifted left by ENV_EXP_FRACT + compared to an integer representation so that numbers smaller than 1 + can be represented. + + This function calculates a mantissa corresponding to the fractional + part of the exponent for each reference energy. The array iEnvelope + is converted in place to save memory. Input and output data must + be interpreted differently, as shown in the below figure: + + \image html EnvelopeData.png + + The data is then used in calculateSbrEnvelope(). +*/ +static void requantizeEnvelopeData(HANDLE_SBR_FRAME_DATA h_sbr_data, + int ampResolution) { + int i; + FIXP_SGL mantissa; + int ampShift = 1 - ampResolution; + int exponent; + + /* In case that ENV_EXP_FRACT is changed to something else but 0 or 8, + the initialization of this array has to be adapted! + */ +#if ENV_EXP_FRACT + static const FIXP_SGL pow2[ENV_EXP_FRACT] = { + FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 1))), /* 0.7071 */ + FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 2))), /* 0.5946 */ + FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 3))), + FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 4))), + FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 5))), + FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 6))), + FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 7))), + FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 8))) /* 0.5013 */ + }; + + int bit, mask; +#endif + + for (i = 0; i < h_sbr_data->nScaleFactors; i++) { + exponent = (LONG)h_sbr_data->iEnvelope[i]; + +#if ENV_EXP_FRACT + + exponent = exponent >> ampShift; + mantissa = 0.5f; + + /* Amplify mantissa according to the fractional part of the + exponent (result will be between 0.500000 and 0.999999) + */ + mask = 1; /* begin with lowest bit of exponent */ + + for (bit = ENV_EXP_FRACT - 1; bit >= 0; bit--) { + if (exponent & mask) { + /* The current bit of the exponent is set, + multiply mantissa with the corresponding factor: */ + mantissa = (FIXP_SGL)((mantissa * pow2[bit]) << 1); + } + /* Advance to next bit */ + mask = mask << 1; + } + + /* Make integer part of exponent right aligned */ + exponent = exponent >> ENV_EXP_FRACT; + +#else + /* In case of the high amplitude resolution, 1 bit of the exponent gets lost + by the shift. This will be compensated by a mantissa of 0.5*sqrt(2) + instead of 0.5 if that bit is 1. */ + mantissa = (exponent & ampShift) ? FL2FXCONST_SGL(0.707106781186548f) + : FL2FXCONST_SGL(0.5f); + exponent = exponent >> ampShift; +#endif + + /* + Mantissa was set to 0.5 (instead of 1.0, therefore increase exponent by + 1). Multiply by L=nChannels=64 by increasing exponent by another 6. + => Increase exponent by 7 + */ + exponent += 7 + NRG_EXP_OFFSET; + + /* Combine mantissa and exponent and write back the result */ + h_sbr_data->iEnvelope[i] = + ((FIXP_SGL)((SHORT)(FIXP_SGL)mantissa & MASK_M)) + + (FIXP_SGL)((SHORT)(FIXP_SGL)exponent & MASK_E); + } +} + +/*! + \brief Build new reference energies from old ones and delta coded data +*/ +static void deltaToLinearPcmEnvelopeDecoding( + HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ + HANDLE_SBR_FRAME_DATA h_sbr_data, /*!< pointer to current data */ + HANDLE_SBR_PREV_FRAME_DATA h_prev_data) /*!< pointer to previous data */ +{ + int i, domain, no_of_bands, band, freqRes; + + FIXP_SGL *sfb_nrg_prev = h_prev_data->sfb_nrg_prev; + FIXP_SGL *ptr_nrg = h_sbr_data->iEnvelope; + + int offset = + 2 * hHeaderData->freqBandData.nSfb[0] - hHeaderData->freqBandData.nSfb[1]; + + for (i = 0; i < h_sbr_data->frameInfo.nEnvelopes; i++) { + domain = h_sbr_data->domain_vec[i]; + freqRes = h_sbr_data->frameInfo.freqRes[i]; + + FDK_ASSERT(freqRes >= 0 && freqRes <= 1); + + no_of_bands = hHeaderData->freqBandData.nSfb[freqRes]; + + FDK_ASSERT(no_of_bands < (64)); + + if (domain == 0) { + mapLowResEnergyVal(*ptr_nrg, sfb_nrg_prev, offset, 0, freqRes); + ptr_nrg++; + for (band = 1; band < no_of_bands; band++) { + *ptr_nrg = *ptr_nrg + *(ptr_nrg - 1); + mapLowResEnergyVal(*ptr_nrg, sfb_nrg_prev, offset, band, freqRes); + ptr_nrg++; + } + } else { + for (band = 0; band < no_of_bands; band++) { + *ptr_nrg = + *ptr_nrg + sfb_nrg_prev[indexLow2High(offset, band, freqRes)]; + mapLowResEnergyVal(*ptr_nrg, sfb_nrg_prev, offset, band, freqRes); + ptr_nrg++; + } + } + } +} + +/*! + \brief Build new noise levels from old ones and delta coded data +*/ +static void decodeNoiseFloorlevels( + HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ + HANDLE_SBR_FRAME_DATA h_sbr_data, /*!< pointer to current data */ + HANDLE_SBR_PREV_FRAME_DATA h_prev_data) /*!< pointer to previous data */ +{ + int i; + int nNfb = hHeaderData->freqBandData.nNfb; + int nNoiseFloorEnvelopes = h_sbr_data->frameInfo.nNoiseEnvelopes; + + /* Decode first noise envelope */ + + if (h_sbr_data->domain_vec_noise[0] == 0) { + FIXP_SGL noiseLevel = h_sbr_data->sbrNoiseFloorLevel[0]; + for (i = 1; i < nNfb; i++) { + noiseLevel += h_sbr_data->sbrNoiseFloorLevel[i]; + h_sbr_data->sbrNoiseFloorLevel[i] = noiseLevel; + } + } else { + for (i = 0; i < nNfb; i++) { + h_sbr_data->sbrNoiseFloorLevel[i] += h_prev_data->prevNoiseLevel[i]; + } + } + + /* If present, decode the second noise envelope + Note: nNoiseFloorEnvelopes can only be 1 or 2 */ + + if (nNoiseFloorEnvelopes > 1) { + if (h_sbr_data->domain_vec_noise[1] == 0) { + FIXP_SGL noiseLevel = h_sbr_data->sbrNoiseFloorLevel[nNfb]; + for (i = nNfb + 1; i < 2 * nNfb; i++) { + noiseLevel += h_sbr_data->sbrNoiseFloorLevel[i]; + h_sbr_data->sbrNoiseFloorLevel[i] = noiseLevel; + } + } else { + for (i = 0; i < nNfb; i++) { + h_sbr_data->sbrNoiseFloorLevel[i + nNfb] += + h_sbr_data->sbrNoiseFloorLevel[i]; + } + } + } + + limitNoiseLevels(hHeaderData, h_sbr_data); + + /* Update prevNoiseLevel with the last noise envelope */ + for (i = 0; i < nNfb; i++) + h_prev_data->prevNoiseLevel[i] = + h_sbr_data->sbrNoiseFloorLevel[i + nNfb * (nNoiseFloorEnvelopes - 1)]; + + /* Requantize the noise floor levels in COUPLING_OFF-mode */ + if (!h_sbr_data->coupling) { + int nf_e; + + for (i = 0; i < nNoiseFloorEnvelopes * nNfb; i++) { + nf_e = 6 - (LONG)h_sbr_data->sbrNoiseFloorLevel[i] + 1 + NOISE_EXP_OFFSET; + /* +1 to compensate for a mantissa of 0.5 instead of 1.0 */ + + h_sbr_data->sbrNoiseFloorLevel[i] = + (FIXP_SGL)(((LONG)FL2FXCONST_SGL(0.5f)) + /* mantissa */ + (nf_e & MASK_E)); /* exponent */ + } + } +} diff --git a/fdk-aac/libSBRdec/src/env_dec.h b/fdk-aac/libSBRdec/src/env_dec.h new file mode 100644 index 0000000..0b11ce1 --- /dev/null +++ b/fdk-aac/libSBRdec/src/env_dec.h @@ -0,0 +1,119 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief Envelope decoding +*/ +#ifndef ENV_DEC_H +#define ENV_DEC_H + +#include "sbrdecoder.h" +#include "env_extr.h" + +void decodeSbrData(HANDLE_SBR_HEADER_DATA hHeaderData, + HANDLE_SBR_FRAME_DATA h_data_left, + HANDLE_SBR_PREV_FRAME_DATA h_prev_data_left, + HANDLE_SBR_FRAME_DATA h_data_right, + HANDLE_SBR_PREV_FRAME_DATA h_prev_data_right); + +#endif diff --git a/fdk-aac/libSBRdec/src/env_extr.cpp b/fdk-aac/libSBRdec/src/env_extr.cpp new file mode 100644 index 0000000..c72a7b6 --- /dev/null +++ b/fdk-aac/libSBRdec/src/env_extr.cpp @@ -0,0 +1,1728 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief Envelope extraction + The functions provided by this module are mostly called by applySBR(). After + it is determined that there is valid SBR data, sbrGetHeaderData() might be + called if the current SBR data contains an \ref SBR_HEADER_ELEMENT as opposed + to a \ref SBR_STANDARD_ELEMENT. This function may return various error codes + as defined in #SBR_HEADER_STATUS . Most importantly it returns HEADER_RESET + when decoder settings need to be recalculated according to the SBR + specifications. In that case applySBR() will initiatite the required + re-configuration. + + The header data is stored in a #SBR_HEADER_DATA structure. + + The actual SBR data for the current frame is decoded into SBR_FRAME_DATA + stuctures by sbrGetChannelPairElement() [for stereo streams] and + sbrGetSingleChannelElement() [for mono streams]. There is no fractional + arithmetic involved. + + Once the information is extracted, the data needs to be further prepared + before the actual decoding process. This is done in decodeSbrData(). + + \sa Description of buffer management in applySBR(). \ref documentationOverview + + <h1>About the SBR data format:</h1> + + Each frame includes SBR data (side chain information), and can be either the + \ref SBR_HEADER_ELEMENT or the \ref SBR_STANDARD_ELEMENT. Parts of the data + can be protected by a CRC checksum. + + \anchor SBR_HEADER_ELEMENT <h2>The SBR_HEADER_ELEMENT</h2> + + The SBR_HEADER_ELEMENT can be transmitted with every frame, however, it + typically is send every second or so. It contains fundamental information such + as SBR sampling frequency and frequency range as well as control signals that + do not require frequent changes. It also includes the \ref + SBR_STANDARD_ELEMENT. + + Depending on the changes between the information in a current + SBR_HEADER_ELEMENT and the previous SBR_HEADER_ELEMENT, the SBR decoder might + need to be reset and reconfigured (e.g. new tables need to be calculated). + + \anchor SBR_STANDARD_ELEMENT <h2>The SBR_STANDARD_ELEMENT</h2> + + This data can be subdivided into "side info" and "raw data", where side info + is defined as signals needed to decode the raw data and some decoder tuning + signals. Raw data is referred to as PCM and Huffman coded envelope and noise + floor estimates. The side info also includes information about the + time-frequency grid for the current frame. + + \sa \ref documentationOverview +*/ + +#include "env_extr.h" + +#include "sbr_ram.h" +#include "sbr_rom.h" +#include "huff_dec.h" + +#include "psbitdec.h" + +#define DRM_PARAMETRIC_STEREO 0 +#define EXTENSION_ID_PS_CODING 2 + +static int extractPvcFrameInfo( + HANDLE_FDK_BITSTREAM hBs, /*!< bitbuffer handle */ + HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ + HANDLE_SBR_FRAME_DATA h_frame_data, /*!< pointer to memory where the + frame-info will be stored */ + HANDLE_SBR_PREV_FRAME_DATA h_prev_frame_data, /*!< pointer to memory where + the previous frame-info + will be stored */ + UCHAR pvc_mode_last, /**< PVC mode of last frame */ + const UINT flags); +static int extractFrameInfo(HANDLE_FDK_BITSTREAM hBs, + HANDLE_SBR_HEADER_DATA hHeaderData, + HANDLE_SBR_FRAME_DATA h_frame_data, + const UINT nrOfChannels, const UINT flags); + +static int sbrGetPvcEnvelope(HANDLE_SBR_HEADER_DATA hHeaderData, + HANDLE_SBR_FRAME_DATA h_frame_data, + HANDLE_FDK_BITSTREAM hBs, const UINT flags, + const UINT pvcMode); +static int sbrGetEnvelope(HANDLE_SBR_HEADER_DATA hHeaderData, + HANDLE_SBR_FRAME_DATA h_frame_data, + HANDLE_FDK_BITSTREAM hBs, const UINT flags); + +static void sbrGetDirectionControlData(HANDLE_SBR_FRAME_DATA hFrameData, + HANDLE_FDK_BITSTREAM hBs, + const UINT flags, const int bs_pvc_mode); + +static void sbrGetNoiseFloorData(HANDLE_SBR_HEADER_DATA hHeaderData, + HANDLE_SBR_FRAME_DATA h_frame_data, + HANDLE_FDK_BITSTREAM hBs); + +static int checkFrameInfo(FRAME_INFO *pFrameInfo, int numberOfTimeSlots, + int overlap, int timeStep); + +/* Mapping to std samplerate table according to 14496-3 (4.6.18.2.6) */ +typedef struct SR_MAPPING { + UINT fsRangeLo; /* If fsRangeLo(n+1)>fs>=fsRangeLo(n), it will be mapped to... + */ + UINT fsMapped; /* fsMapped. */ +} SR_MAPPING; + +static const SR_MAPPING stdSampleRatesMapping[] = { + {0, 8000}, {9391, 11025}, {11502, 12000}, {13856, 16000}, + {18783, 22050}, {23004, 24000}, {27713, 32000}, {37566, 44100}, + {46009, 48000}, {55426, 64000}, {75132, 88200}, {92017, 96000}}; +static const SR_MAPPING stdSampleRatesMappingUsac[] = { + {0, 16000}, {18783, 22050}, {23004, 24000}, {27713, 32000}, + {35777, 40000}, {42000, 44100}, {46009, 48000}, {55426, 64000}, + {75132, 88200}, {92017, 96000}}; + +UINT sbrdec_mapToStdSampleRate(UINT fs, + UINT isUsac) /*!< Output sampling frequency */ +{ + UINT fsMapped = fs, tableSize = 0; + const SR_MAPPING *mappingTable; + int i; + + if (!isUsac) { + mappingTable = stdSampleRatesMapping; + tableSize = sizeof(stdSampleRatesMapping) / sizeof(SR_MAPPING); + } else { + mappingTable = stdSampleRatesMappingUsac; + tableSize = sizeof(stdSampleRatesMappingUsac) / sizeof(SR_MAPPING); + } + + for (i = tableSize - 1; i >= 0; i--) { + if (fs >= mappingTable[i].fsRangeLo) { + fsMapped = mappingTable[i].fsMapped; + break; + } + } + + return (fsMapped); +} + +SBR_ERROR +initHeaderData(HANDLE_SBR_HEADER_DATA hHeaderData, const int sampleRateIn, + const int sampleRateOut, const INT downscaleFactor, + const int samplesPerFrame, const UINT flags, + const int setDefaultHdr) { + HANDLE_FREQ_BAND_DATA hFreq = &hHeaderData->freqBandData; + SBR_ERROR sbrError = SBRDEC_OK; + int numAnalysisBands; + int sampleRateProc; + + if (!(flags & (SBRDEC_SYNTAX_USAC | SBRDEC_SYNTAX_RSVD50))) { + sampleRateProc = + sbrdec_mapToStdSampleRate(sampleRateOut * downscaleFactor, 0); + } else { + sampleRateProc = sampleRateOut * downscaleFactor; + } + + if (sampleRateIn == sampleRateOut) { + hHeaderData->sbrProcSmplRate = sampleRateProc << 1; + numAnalysisBands = 32; + } else { + hHeaderData->sbrProcSmplRate = sampleRateProc; + if ((sampleRateOut >> 1) == sampleRateIn) { + /* 1:2 */ + numAnalysisBands = 32; + } else if ((sampleRateOut >> 2) == sampleRateIn) { + /* 1:4 */ + numAnalysisBands = 16; + } else if ((sampleRateOut * 3) >> 3 == (sampleRateIn * 8) >> 3) { + /* 3:8, 3/4 core frame length */ + numAnalysisBands = 24; + } else { + sbrError = SBRDEC_UNSUPPORTED_CONFIG; + goto bail; + } + } + numAnalysisBands /= downscaleFactor; + + if (setDefaultHdr) { + /* Fill in default values first */ + hHeaderData->syncState = SBR_NOT_INITIALIZED; + hHeaderData->status = 0; + hHeaderData->frameErrorFlag = 0; + + hHeaderData->bs_info.ampResolution = 1; + hHeaderData->bs_info.xover_band = 0; + hHeaderData->bs_info.sbr_preprocessing = 0; + hHeaderData->bs_info.pvc_mode = 0; + + hHeaderData->bs_data.startFreq = 5; + hHeaderData->bs_data.stopFreq = 0; + hHeaderData->bs_data.freqScale = + 0; /* previously 2; for ELD reduced delay bitstreams + /samplerates initializing of the sbr decoder instance fails if + freqScale is set to 2 because no master table can be generated; in + ELD reduced delay bitstreams this value is always 0; gets overwritten + when header is read */ + hHeaderData->bs_data.alterScale = 1; + hHeaderData->bs_data.noise_bands = 2; + hHeaderData->bs_data.limiterBands = 2; + hHeaderData->bs_data.limiterGains = 2; + hHeaderData->bs_data.interpolFreq = 1; + hHeaderData->bs_data.smoothingLength = 1; + + /* Patch some entries */ + if (sampleRateOut * downscaleFactor >= 96000) { + hHeaderData->bs_data.startFreq = + 4; /* having read these frequency values from bit stream before. */ + hHeaderData->bs_data.stopFreq = 3; + } else if (sampleRateOut * downscaleFactor > + 24000) { /* Trigger an error if SBR is going to be processed + without */ + hHeaderData->bs_data.startFreq = + 7; /* having read these frequency values from bit stream before. */ + hHeaderData->bs_data.stopFreq = 3; + } + } + + if ((sampleRateOut >> 2) == sampleRateIn) { + hHeaderData->timeStep = 4; + } else { + hHeaderData->timeStep = (flags & SBRDEC_ELD_GRID) ? 1 : 2; + } + + /* Setup pointers to frequency band tables */ + hFreq->freqBandTable[0] = hFreq->freqBandTableLo; + hFreq->freqBandTable[1] = hFreq->freqBandTableHi; + + /* One SBR timeslot corresponds to the amount of samples equal to the amount + * of analysis bands, divided by the timestep. */ + hHeaderData->numberTimeSlots = + (samplesPerFrame / numAnalysisBands) >> (hHeaderData->timeStep - 1); + if (hHeaderData->numberTimeSlots > (16)) { + sbrError = SBRDEC_UNSUPPORTED_CONFIG; + } + + hHeaderData->numberOfAnalysisBands = numAnalysisBands; + if ((sampleRateOut >> 2) == sampleRateIn) { + hHeaderData->numberTimeSlots <<= 1; + } + +bail: + return sbrError; +} + +/*! + \brief Initialize the SBR_PREV_FRAME_DATA struct +*/ +void initSbrPrevFrameData( + HANDLE_SBR_PREV_FRAME_DATA + h_prev_data, /*!< handle to struct SBR_PREV_FRAME_DATA */ + int timeSlots) /*!< Framelength in SBR-timeslots */ +{ + int i; + + /* Set previous energy and noise levels to 0 for the case + that decoding starts in the middle of a bitstream */ + for (i = 0; i < MAX_FREQ_COEFFS; i++) + h_prev_data->sfb_nrg_prev[i] = (FIXP_DBL)0; + for (i = 0; i < MAX_NOISE_COEFFS; i++) + h_prev_data->prevNoiseLevel[i] = (FIXP_DBL)0; + for (i = 0; i < MAX_INVF_BANDS; i++) h_prev_data->sbr_invf_mode[i] = INVF_OFF; + + h_prev_data->stopPos = timeSlots; + h_prev_data->coupling = COUPLING_OFF; + h_prev_data->ampRes = 0; + + FDKmemclear(&h_prev_data->prevFrameInfo, sizeof(h_prev_data->prevFrameInfo)); +} + +/*! + \brief Read header data from bitstream + + \return error status - 0 if ok +*/ +SBR_HEADER_STATUS +sbrGetHeaderData(HANDLE_SBR_HEADER_DATA hHeaderData, HANDLE_FDK_BITSTREAM hBs, + const UINT flags, const int fIsSbrData, + const UCHAR configMode) { + SBR_HEADER_DATA_BS *pBsData; + SBR_HEADER_DATA_BS lastHeader; + SBR_HEADER_DATA_BS_INFO lastInfo; + int headerExtra1 = 0, headerExtra2 = 0; + + /* Read and discard new header in config change detection mode */ + if (configMode & AC_CM_DET_CFG_CHANGE) { + if (!(flags & (SBRDEC_SYNTAX_RSVD50 | SBRDEC_SYNTAX_USAC))) { + /* ampResolution */ + FDKreadBits(hBs, 1); + } + /* startFreq, stopFreq */ + FDKpushFor(hBs, 8); + if (!(flags & (SBRDEC_SYNTAX_RSVD50 | SBRDEC_SYNTAX_USAC))) { + /* xover_band */ + FDKreadBits(hBs, 3); + /* reserved bits */ + FDKreadBits(hBs, 2); + } + headerExtra1 = FDKreadBit(hBs); + headerExtra2 = FDKreadBit(hBs); + FDKpushFor(hBs, 5 * headerExtra1 + 6 * headerExtra2); + + return HEADER_OK; + } + + /* Copy SBR bit stream header to temporary header */ + lastHeader = hHeaderData->bs_data; + lastInfo = hHeaderData->bs_info; + + /* Read new header from bitstream */ + if ((flags & (SBRDEC_SYNTAX_RSVD50 | SBRDEC_SYNTAX_USAC)) && !fIsSbrData) { + pBsData = &hHeaderData->bs_dflt; + } else { + pBsData = &hHeaderData->bs_data; + } + + if (!(flags & (SBRDEC_SYNTAX_RSVD50 | SBRDEC_SYNTAX_USAC))) { + hHeaderData->bs_info.ampResolution = FDKreadBits(hBs, 1); + } + + pBsData->startFreq = FDKreadBits(hBs, 4); + pBsData->stopFreq = FDKreadBits(hBs, 4); + + if (!(flags & (SBRDEC_SYNTAX_RSVD50 | SBRDEC_SYNTAX_USAC))) { + hHeaderData->bs_info.xover_band = FDKreadBits(hBs, 3); + FDKreadBits(hBs, 2); + } + + headerExtra1 = FDKreadBits(hBs, 1); + headerExtra2 = FDKreadBits(hBs, 1); + + /* Handle extra header information */ + if (headerExtra1) { + pBsData->freqScale = FDKreadBits(hBs, 2); + pBsData->alterScale = FDKreadBits(hBs, 1); + pBsData->noise_bands = FDKreadBits(hBs, 2); + } else { + pBsData->freqScale = 2; + pBsData->alterScale = 1; + pBsData->noise_bands = 2; + } + + if (headerExtra2) { + pBsData->limiterBands = FDKreadBits(hBs, 2); + pBsData->limiterGains = FDKreadBits(hBs, 2); + pBsData->interpolFreq = FDKreadBits(hBs, 1); + pBsData->smoothingLength = FDKreadBits(hBs, 1); + } else { + pBsData->limiterBands = 2; + pBsData->limiterGains = 2; + pBsData->interpolFreq = 1; + pBsData->smoothingLength = 1; + } + + /* Look for new settings. IEC 14496-3, 4.6.18.3.1 */ + if (hHeaderData->syncState < SBR_HEADER || + lastHeader.startFreq != pBsData->startFreq || + lastHeader.stopFreq != pBsData->stopFreq || + lastHeader.freqScale != pBsData->freqScale || + lastHeader.alterScale != pBsData->alterScale || + lastHeader.noise_bands != pBsData->noise_bands || + lastInfo.xover_band != hHeaderData->bs_info.xover_band) { + return HEADER_RESET; /* New settings */ + } + + return HEADER_OK; +} + +/*! + \brief Get missing harmonics parameters (only used for AAC+SBR) + + \return error status - 0 if ok +*/ +int sbrGetSyntheticCodedData(HANDLE_SBR_HEADER_DATA hHeaderData, + HANDLE_SBR_FRAME_DATA hFrameData, + HANDLE_FDK_BITSTREAM hBs, const UINT flags) { + int i, bitsRead = 0; + + int add_harmonic_flag = FDKreadBits(hBs, 1); + bitsRead++; + + if (add_harmonic_flag) { + int nSfb = hHeaderData->freqBandData.nSfb[1]; + for (i = 0; i < ADD_HARMONICS_FLAGS_SIZE; i++) { + /* read maximum 32 bits and align them to the MSB */ + int readBits = fMin(32, nSfb); + nSfb -= readBits; + if (readBits > 0) { + hFrameData->addHarmonics[i] = FDKreadBits(hBs, readBits) + << (32 - readBits); + } else { + hFrameData->addHarmonics[i] = 0; + } + + bitsRead += readBits; + } + /* bs_pvc_mode = 0 for Rsvd50 */ + if (flags & SBRDEC_SYNTAX_USAC) { + if (hHeaderData->bs_info.pvc_mode) { + int bs_sinusoidal_position = 31; + if (FDKreadBit(hBs) /* bs_sinusoidal_position_flag */) { + bs_sinusoidal_position = FDKreadBits(hBs, 5); + } + hFrameData->sinusoidal_position = bs_sinusoidal_position; + } + } + } else { + for (i = 0; i < ADD_HARMONICS_FLAGS_SIZE; i++) + hFrameData->addHarmonics[i] = 0; + } + + return (bitsRead); +} + +/*! + \brief Reads extension data from the bitstream + + The bitstream format allows up to 4 kinds of extended data element. + Extended data may contain several elements, each identified by a 2-bit-ID. + So far, no extended data elements are defined hence the first 2 parameters + are unused. The data should be skipped in order to update the number + of read bits for the consistency check in applySBR(). +*/ +static int extractExtendedData( + HANDLE_SBR_HEADER_DATA hHeaderData, /*!< handle to SBR header */ + HANDLE_FDK_BITSTREAM hBs /*!< Handle to the bit buffer */ + , + HANDLE_PS_DEC hParametricStereoDec /*!< Parametric Stereo Decoder */ +) { + INT nBitsLeft; + int extended_data; + int i, frameOk = 1; + + extended_data = FDKreadBits(hBs, 1); + + if (extended_data) { + int cnt; + int bPsRead = 0; + + cnt = FDKreadBits(hBs, 4); + if (cnt == (1 << 4) - 1) cnt += FDKreadBits(hBs, 8); + + nBitsLeft = 8 * cnt; + + /* sanity check for cnt */ + if (nBitsLeft > (INT)FDKgetValidBits(hBs)) { + /* limit nBitsLeft */ + nBitsLeft = (INT)FDKgetValidBits(hBs); + /* set frame error */ + frameOk = 0; + } + + while (nBitsLeft > 7) { + int extension_id = FDKreadBits(hBs, 2); + nBitsLeft -= 2; + + switch (extension_id) { + case EXTENSION_ID_PS_CODING: + + /* Read PS data from bitstream */ + + if (hParametricStereoDec != NULL) { + if (bPsRead && + !hParametricStereoDec->bsData[hParametricStereoDec->bsReadSlot] + .mpeg.bPsHeaderValid) { + cnt = nBitsLeft >> 3; /* number of remaining bytes */ + for (i = 0; i < cnt; i++) FDKreadBits(hBs, 8); + nBitsLeft -= cnt * 8; + } else { + nBitsLeft -= + (INT)ReadPsData(hParametricStereoDec, hBs, nBitsLeft); + bPsRead = 1; + } + } + + /* parametric stereo detected, could set channelMode accordingly here + */ + /* */ + /* "The usage of this parametric stereo extension to HE-AAC is */ + /* signalled implicitly in the bitstream. Hence, if an sbr_extension() + */ + /* with bs_extension_id==EXTENSION_ID_PS is found in the SBR part of + */ + /* the bitstream, a decoder supporting the combination of SBR and PS + */ + /* shall operate the PS tool to generate a stereo output signal." */ + /* source: ISO/IEC 14496-3:2001/FDAM 2:2004(E) */ + + break; + + default: + cnt = nBitsLeft >> 3; /* number of remaining bytes */ + for (i = 0; i < cnt; i++) FDKreadBits(hBs, 8); + nBitsLeft -= cnt * 8; + break; + } + } + + if (nBitsLeft < 0) { + frameOk = 0; + goto bail; + } else { + /* Read fill bits for byte alignment */ + FDKreadBits(hBs, nBitsLeft); + } + } + +bail: + return (frameOk); +} + +/*! + \brief Read bitstream elements of a SBR channel element + \return SbrFrameOK +*/ +int sbrGetChannelElement(HANDLE_SBR_HEADER_DATA hHeaderData, + HANDLE_SBR_FRAME_DATA hFrameDataLeft, + HANDLE_SBR_FRAME_DATA hFrameDataRight, + HANDLE_SBR_PREV_FRAME_DATA hFrameDataLeftPrev, + UCHAR pvc_mode_last, HANDLE_FDK_BITSTREAM hBs, + HANDLE_PS_DEC hParametricStereoDec, const UINT flags, + const int overlap) { + int i, bs_coupling = COUPLING_OFF; + const int nCh = (hFrameDataRight == NULL) ? 1 : 2; + + if (!(flags & (SBRDEC_SYNTAX_USAC | SBRDEC_SYNTAX_RSVD50))) { + /* Reserved bits */ + if (FDKreadBits(hBs, 1)) { /* bs_data_extra */ + FDKreadBits(hBs, 4); + if ((flags & SBRDEC_SYNTAX_SCAL) || (nCh == 2)) { + FDKreadBits(hBs, 4); + } + } + } + + if (nCh == 2) { + /* Read coupling flag */ + bs_coupling = FDKreadBits(hBs, 1); + if (bs_coupling) { + hFrameDataLeft->coupling = COUPLING_LEVEL; + hFrameDataRight->coupling = COUPLING_BAL; + } else { + hFrameDataLeft->coupling = COUPLING_OFF; + hFrameDataRight->coupling = COUPLING_OFF; + } + } else { + if (flags & SBRDEC_SYNTAX_SCAL) { + FDKreadBits(hBs, 1); /* bs_coupling */ + } + hFrameDataLeft->coupling = COUPLING_OFF; + } + + if (flags & (SBRDEC_SYNTAX_USAC | SBRDEC_SYNTAX_RSVD50)) { + if (flags & SBRDEC_USAC_HARMONICSBR) { + hFrameDataLeft->sbrPatchingMode = FDKreadBit(hBs); + if (hFrameDataLeft->sbrPatchingMode == 0) { + hFrameDataLeft->sbrOversamplingFlag = FDKreadBit(hBs); + if (FDKreadBit(hBs)) { /* sbrPitchInBinsFlag */ + hFrameDataLeft->sbrPitchInBins = FDKreadBits(hBs, 7); + } else { + hFrameDataLeft->sbrPitchInBins = 0; + } + } else { + hFrameDataLeft->sbrOversamplingFlag = 0; + hFrameDataLeft->sbrPitchInBins = 0; + } + + if (nCh == 2) { + if (bs_coupling) { + hFrameDataRight->sbrPatchingMode = hFrameDataLeft->sbrPatchingMode; + hFrameDataRight->sbrOversamplingFlag = + hFrameDataLeft->sbrOversamplingFlag; + hFrameDataRight->sbrPitchInBins = hFrameDataLeft->sbrPitchInBins; + } else { + hFrameDataRight->sbrPatchingMode = FDKreadBit(hBs); + if (hFrameDataRight->sbrPatchingMode == 0) { + hFrameDataRight->sbrOversamplingFlag = FDKreadBit(hBs); + if (FDKreadBit(hBs)) { /* sbrPitchInBinsFlag */ + hFrameDataRight->sbrPitchInBins = FDKreadBits(hBs, 7); + } else { + hFrameDataRight->sbrPitchInBins = 0; + } + } else { + hFrameDataRight->sbrOversamplingFlag = 0; + hFrameDataRight->sbrPitchInBins = 0; + } + } + } + } else { + if (nCh == 2) { + hFrameDataRight->sbrPatchingMode = 1; + hFrameDataRight->sbrOversamplingFlag = 0; + hFrameDataRight->sbrPitchInBins = 0; + } + + hFrameDataLeft->sbrPatchingMode = 1; + hFrameDataLeft->sbrOversamplingFlag = 0; + hFrameDataLeft->sbrPitchInBins = 0; + } + } else { + if (nCh == 2) { + hFrameDataRight->sbrPatchingMode = 1; + hFrameDataRight->sbrOversamplingFlag = 0; + hFrameDataRight->sbrPitchInBins = 0; + } + + hFrameDataLeft->sbrPatchingMode = 1; + hFrameDataLeft->sbrOversamplingFlag = 0; + hFrameDataLeft->sbrPitchInBins = 0; + } + + /* + sbr_grid(): Grid control + */ + if (hHeaderData->bs_info.pvc_mode) { + FDK_ASSERT(nCh == 1); /* PVC not possible for CPE */ + if (!extractPvcFrameInfo(hBs, hHeaderData, hFrameDataLeft, + hFrameDataLeftPrev, pvc_mode_last, flags)) + return 0; + + if (!checkFrameInfo(&hFrameDataLeft->frameInfo, + hHeaderData->numberTimeSlots, overlap, + hHeaderData->timeStep)) + return 0; + } else { + if (!extractFrameInfo(hBs, hHeaderData, hFrameDataLeft, 1, flags)) return 0; + + if (!checkFrameInfo(&hFrameDataLeft->frameInfo, + hHeaderData->numberTimeSlots, overlap, + hHeaderData->timeStep)) + return 0; + } + if (nCh == 2) { + if (hFrameDataLeft->coupling) { + FDKmemcpy(&hFrameDataRight->frameInfo, &hFrameDataLeft->frameInfo, + sizeof(FRAME_INFO)); + hFrameDataRight->ampResolutionCurrentFrame = + hFrameDataLeft->ampResolutionCurrentFrame; + } else { + if (!extractFrameInfo(hBs, hHeaderData, hFrameDataRight, 2, flags)) + return 0; + + if (!checkFrameInfo(&hFrameDataRight->frameInfo, + hHeaderData->numberTimeSlots, overlap, + hHeaderData->timeStep)) + return 0; + } + } + + /* + sbr_dtdf(): Fetch domain vectors (time or frequency direction for + delta-coding) + */ + sbrGetDirectionControlData(hFrameDataLeft, hBs, flags, + hHeaderData->bs_info.pvc_mode); + if (nCh == 2) { + sbrGetDirectionControlData(hFrameDataRight, hBs, flags, 0); + } + + /* sbr_invf() */ + for (i = 0; i < hHeaderData->freqBandData.nInvfBands; i++) { + hFrameDataLeft->sbr_invf_mode[i] = (INVF_MODE)FDKreadBits(hBs, 2); + } + if (nCh == 2) { + if (hFrameDataLeft->coupling) { + for (i = 0; i < hHeaderData->freqBandData.nInvfBands; i++) { + hFrameDataRight->sbr_invf_mode[i] = hFrameDataLeft->sbr_invf_mode[i]; + } + } else { + for (i = 0; i < hHeaderData->freqBandData.nInvfBands; i++) { + hFrameDataRight->sbr_invf_mode[i] = (INVF_MODE)FDKreadBits(hBs, 2); + } + } + } + + if (nCh == 1) { + if (hHeaderData->bs_info.pvc_mode) { + if (!sbrGetPvcEnvelope(hHeaderData, hFrameDataLeft, hBs, flags, + hHeaderData->bs_info.pvc_mode)) + return 0; + } else if (!sbrGetEnvelope(hHeaderData, hFrameDataLeft, hBs, flags)) + return 0; + + sbrGetNoiseFloorData(hHeaderData, hFrameDataLeft, hBs); + } else if (hFrameDataLeft->coupling) { + if (!sbrGetEnvelope(hHeaderData, hFrameDataLeft, hBs, flags)) { + return 0; + } + + sbrGetNoiseFloorData(hHeaderData, hFrameDataLeft, hBs); + + if (!sbrGetEnvelope(hHeaderData, hFrameDataRight, hBs, flags)) { + return 0; + } + sbrGetNoiseFloorData(hHeaderData, hFrameDataRight, hBs); + } else { /* nCh == 2 && no coupling */ + + if (!sbrGetEnvelope(hHeaderData, hFrameDataLeft, hBs, flags)) return 0; + + if (!sbrGetEnvelope(hHeaderData, hFrameDataRight, hBs, flags)) return 0; + + sbrGetNoiseFloorData(hHeaderData, hFrameDataLeft, hBs); + + sbrGetNoiseFloorData(hHeaderData, hFrameDataRight, hBs); + } + + sbrGetSyntheticCodedData(hHeaderData, hFrameDataLeft, hBs, flags); + if (nCh == 2) { + sbrGetSyntheticCodedData(hHeaderData, hFrameDataRight, hBs, flags); + } + + if (!(flags & (SBRDEC_SYNTAX_USAC | SBRDEC_SYNTAX_RSVD50))) { + if (!extractExtendedData(hHeaderData, hBs, hParametricStereoDec)) { + return 0; + } + } + + return 1; +} + +/*! + \brief Read direction control data from bitstream +*/ +void sbrGetDirectionControlData( + HANDLE_SBR_FRAME_DATA h_frame_data, /*!< handle to struct SBR_FRAME_DATA */ + HANDLE_FDK_BITSTREAM hBs, /*!< handle to struct BIT_BUF */ + const UINT flags, const int bs_pvc_mode) + +{ + int i; + int indepFlag = 0; + + if (flags & (SBRDEC_SYNTAX_USAC | SBRDEC_SYNTAX_RSVD50)) { + indepFlag = flags & SBRDEC_USAC_INDEP; + } + + if (bs_pvc_mode == 0) { + i = 0; + if (indepFlag) { + h_frame_data->domain_vec[i++] = 0; + } + for (; i < h_frame_data->frameInfo.nEnvelopes; i++) { + h_frame_data->domain_vec[i] = FDKreadBits(hBs, 1); + } + } + + i = 0; + if (indepFlag) { + h_frame_data->domain_vec_noise[i++] = 0; + } + for (; i < h_frame_data->frameInfo.nNoiseEnvelopes; i++) { + h_frame_data->domain_vec_noise[i] = FDKreadBits(hBs, 1); + } +} + +/*! + \brief Read noise-floor-level data from bitstream +*/ +void sbrGetNoiseFloorData( + HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ + HANDLE_SBR_FRAME_DATA h_frame_data, /*!< handle to struct SBR_FRAME_DATA */ + HANDLE_FDK_BITSTREAM hBs) /*!< handle to struct BIT_BUF */ +{ + int i, j; + int delta; + COUPLING_MODE coupling; + int noNoiseBands = hHeaderData->freqBandData.nNfb; + + Huffman hcb_noiseF; + Huffman hcb_noise; + int envDataTableCompFactor; + + coupling = h_frame_data->coupling; + + /* + Select huffman codebook depending on coupling mode + */ + if (coupling == COUPLING_BAL) { + hcb_noise = (Huffman)&FDK_sbrDecoder_sbr_huffBook_NoiseBalance11T; + hcb_noiseF = + (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvBalance11F; /* "sbr_huffBook_NoiseBalance11F" + */ + envDataTableCompFactor = 1; + } else { + hcb_noise = (Huffman)&FDK_sbrDecoder_sbr_huffBook_NoiseLevel11T; + hcb_noiseF = + (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvLevel11F; /* "sbr_huffBook_NoiseLevel11F" + */ + envDataTableCompFactor = 0; + } + + /* + Read raw noise-envelope data + */ + for (i = 0; i < h_frame_data->frameInfo.nNoiseEnvelopes; i++) { + if (h_frame_data->domain_vec_noise[i] == 0) { + if (coupling == COUPLING_BAL) { + h_frame_data->sbrNoiseFloorLevel[i * noNoiseBands] = + (FIXP_SGL)(((int)FDKreadBits(hBs, 5)) << envDataTableCompFactor); + } else { + h_frame_data->sbrNoiseFloorLevel[i * noNoiseBands] = + (FIXP_SGL)(int)FDKreadBits(hBs, 5); + } + + for (j = 1; j < noNoiseBands; j++) { + delta = DecodeHuffmanCW(hcb_noiseF, hBs); + h_frame_data->sbrNoiseFloorLevel[i * noNoiseBands + j] = + (FIXP_SGL)(delta << envDataTableCompFactor); + } + } else { + for (j = 0; j < noNoiseBands; j++) { + delta = DecodeHuffmanCW(hcb_noise, hBs); + h_frame_data->sbrNoiseFloorLevel[i * noNoiseBands + j] = + (FIXP_SGL)(delta << envDataTableCompFactor); + } + } + } +} + +/* ns = mapNsMode2ns[pvcMode-1][nsMode] */ +static const UCHAR mapNsMode2ns[2][2] = { + {16, 4}, /* pvcMode = 1 */ + {12, 3} /* pvcMode = 2 */ +}; + +static int sbrGetPvcEnvelope( + HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ + HANDLE_SBR_FRAME_DATA h_frame_data, /*!< handle to struct SBR_FRAME_DATA */ + HANDLE_FDK_BITSTREAM hBs, /*!< handle to struct BIT_BUF */ + const UINT flags, const UINT pvcMode) { + int divMode, nsMode; + int indepFlag = flags & SBRDEC_USAC_INDEP; + UCHAR *pvcID = h_frame_data->pvcID; + + divMode = FDKreadBits(hBs, PVC_DIVMODE_BITS); + nsMode = FDKreadBit(hBs); + FDK_ASSERT((pvcMode == 1) || (pvcMode == 2)); + h_frame_data->ns = mapNsMode2ns[pvcMode - 1][nsMode]; + + if (divMode <= 3) { + int i, k = 1, sum_length = 0, reuse_pcvID; + + /* special treatment for first time slot k=0 */ + indepFlag ? (reuse_pcvID = 0) : (reuse_pcvID = FDKreadBit(hBs)); + if (reuse_pcvID) { + pvcID[0] = hHeaderData->pvcIDprev; + } else { + pvcID[0] = FDKreadBits(hBs, PVC_PVCID_BITS); + } + + /* other time slots k>0 */ + for (i = 0; i < divMode; i++) { + int length, numBits = 4; + + if (sum_length >= 13) { + numBits = 1; + } else if (sum_length >= 11) { + numBits = 2; + } else if (sum_length >= 7) { + numBits = 3; + } + + length = FDKreadBits(hBs, numBits); + sum_length += length + 1; + if (sum_length >= PVC_NTIMESLOT) { + return 0; /* parse error */ + } + for (; length--; k++) { + pvcID[k] = pvcID[k - 1]; + } + pvcID[k++] = FDKreadBits(hBs, PVC_PVCID_BITS); + } + for (; k < 16; k++) { + pvcID[k] = pvcID[k - 1]; + } + } else { /* divMode >= 4 */ + int num_grid_info, fixed_length, grid_info, j, k = 0; + + divMode -= 4; + num_grid_info = 2 << divMode; + fixed_length = 8 >> divMode; + FDK_ASSERT(num_grid_info * fixed_length == PVC_NTIMESLOT); + + /* special treatment for first time slot k=0 */ + indepFlag ? (grid_info = 1) : (grid_info = FDKreadBit(hBs)); + if (grid_info) { + pvcID[k++] = FDKreadBits(hBs, PVC_PVCID_BITS); + } else { + pvcID[k++] = hHeaderData->pvcIDprev; + } + j = fixed_length - 1; + for (; j--; k++) { + pvcID[k] = pvcID[k - 1]; + } + num_grid_info--; + + /* other time slots k>0 */ + for (; num_grid_info--;) { + j = fixed_length; + grid_info = FDKreadBit(hBs); + if (grid_info) { + pvcID[k++] = FDKreadBits(hBs, PVC_PVCID_BITS); + j--; + } + for (; j--; k++) { + pvcID[k] = pvcID[k - 1]; + } + } + } + + hHeaderData->pvcIDprev = pvcID[PVC_NTIMESLOT - 1]; + + /* usage of PVC excludes inter-TES tool */ + h_frame_data->iTESactive = (UCHAR)0; + + return 1; +} +/*! + \brief Read envelope data from bitstream +*/ +static int sbrGetEnvelope( + HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ + HANDLE_SBR_FRAME_DATA h_frame_data, /*!< handle to struct SBR_FRAME_DATA */ + HANDLE_FDK_BITSTREAM hBs, /*!< handle to struct BIT_BUF */ + const UINT flags) { + int i, j; + UCHAR no_band[MAX_ENVELOPES]; + int delta = 0; + int offset = 0; + COUPLING_MODE coupling = h_frame_data->coupling; + int ampRes = hHeaderData->bs_info.ampResolution; + int nEnvelopes = h_frame_data->frameInfo.nEnvelopes; + int envDataTableCompFactor; + int start_bits, start_bits_balance; + Huffman hcb_t, hcb_f; + + h_frame_data->nScaleFactors = 0; + + if ((h_frame_data->frameInfo.frameClass == 0) && (nEnvelopes == 1)) { + if (flags & SBRDEC_ELD_GRID) + ampRes = h_frame_data->ampResolutionCurrentFrame; + else + ampRes = 0; + } + h_frame_data->ampResolutionCurrentFrame = ampRes; + + /* + Set number of bits for first value depending on amplitude resolution + */ + if (ampRes == 1) { + start_bits = 6; + start_bits_balance = 5; + } else { + start_bits = 7; + start_bits_balance = 6; + } + + /* + Calculate number of values for each envelope and alltogether + */ + for (i = 0; i < nEnvelopes; i++) { + no_band[i] = + hHeaderData->freqBandData.nSfb[h_frame_data->frameInfo.freqRes[i]]; + h_frame_data->nScaleFactors += no_band[i]; + } + if (h_frame_data->nScaleFactors > MAX_NUM_ENVELOPE_VALUES) return 0; + + /* + Select Huffman codebook depending on coupling mode and amplitude resolution + */ + if (coupling == COUPLING_BAL) { + envDataTableCompFactor = 1; + if (ampRes == 0) { + hcb_t = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvBalance10T; + hcb_f = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvBalance10F; + } else { + hcb_t = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvBalance11T; + hcb_f = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvBalance11F; + } + } else { + envDataTableCompFactor = 0; + if (ampRes == 0) { + hcb_t = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvLevel10T; + hcb_f = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvLevel10F; + } else { + hcb_t = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvLevel11T; + hcb_f = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvLevel11F; + } + } + + h_frame_data->iTESactive = (UCHAR)0; /* disable inter-TES by default */ + /* + Now read raw envelope data + */ + for (j = 0, offset = 0; j < nEnvelopes; j++) { + if (h_frame_data->domain_vec[j] == 0) { + if (coupling == COUPLING_BAL) { + h_frame_data->iEnvelope[offset] = + (FIXP_SGL)(((int)FDKreadBits(hBs, start_bits_balance)) + << envDataTableCompFactor); + } else { + h_frame_data->iEnvelope[offset] = + (FIXP_SGL)(int)FDKreadBits(hBs, start_bits); + } + } + + for (i = (1 - h_frame_data->domain_vec[j]); i < no_band[j]; i++) { + if (h_frame_data->domain_vec[j] == 0) { + delta = DecodeHuffmanCW(hcb_f, hBs); + } else { + delta = DecodeHuffmanCW(hcb_t, hBs); + } + + h_frame_data->iEnvelope[offset + i] = + (FIXP_SGL)(delta << envDataTableCompFactor); + } + if ((flags & SBRDEC_SYNTAX_USAC) && (flags & SBRDEC_USAC_ITES)) { + int bs_temp_shape = FDKreadBit(hBs); + FDK_ASSERT(j < 8); + h_frame_data->iTESactive |= (UCHAR)(bs_temp_shape << j); + if (bs_temp_shape) { + h_frame_data->interTempShapeMode[j] = + FDKread2Bits(hBs); /* bs_inter_temp_shape_mode */ + } else { + h_frame_data->interTempShapeMode[j] = 0; + } + } + offset += no_band[j]; + } + +#if ENV_EXP_FRACT + /* Convert from int to scaled fract (ENV_EXP_FRACT bits for the fractional + * part) */ + for (i = 0; i < h_frame_data->nScaleFactors; i++) { + h_frame_data->iEnvelope[i] <<= ENV_EXP_FRACT; + } +#endif + + return 1; +} + +/***************************************************************************/ +/*! + \brief Generates frame info for FIXFIXonly frame class used for low delay + version + + \return zero for error, one for correct. + ****************************************************************************/ +static int generateFixFixOnly(FRAME_INFO *hSbrFrameInfo, int tranPosInternal, + int numberTimeSlots, const UINT flags) { + int nEnv, i, tranIdx; + const int *pTable; + + switch (numberTimeSlots) { + case 8: + pTable = FDK_sbrDecoder_envelopeTable_8[tranPosInternal]; + break; + case 15: + pTable = FDK_sbrDecoder_envelopeTable_15[tranPosInternal]; + break; + case 16: + pTable = FDK_sbrDecoder_envelopeTable_16[tranPosInternal]; + break; + default: + return 0; + } + + /* look number of envelopes in table */ + nEnv = pTable[0]; + /* look up envelope distribution in table */ + for (i = 1; i < nEnv; i++) hSbrFrameInfo->borders[i] = pTable[i + 2]; + /* open and close frame border */ + hSbrFrameInfo->borders[0] = 0; + hSbrFrameInfo->borders[nEnv] = numberTimeSlots; + hSbrFrameInfo->nEnvelopes = nEnv; + + /* transient idx */ + tranIdx = hSbrFrameInfo->tranEnv = pTable[1]; + + /* add noise floors */ + hSbrFrameInfo->bordersNoise[0] = 0; + hSbrFrameInfo->bordersNoise[1] = + hSbrFrameInfo->borders[tranIdx ? tranIdx : 1]; + hSbrFrameInfo->bordersNoise[2] = numberTimeSlots; + /* nEnv is always > 1, so nNoiseEnvelopes is always 2 (IEC 14496-3 4.6.19.3.2) + */ + hSbrFrameInfo->nNoiseEnvelopes = 2; + + return 1; +} + +/*! + \brief Extracts LowDelaySBR control data from the bitstream. + + \return zero for bitstream error, one for correct. +*/ +static int extractLowDelayGrid( + HANDLE_FDK_BITSTREAM hBitBuf, /*!< bitbuffer handle */ + HANDLE_SBR_HEADER_DATA hHeaderData, + HANDLE_SBR_FRAME_DATA + h_frame_data, /*!< contains the FRAME_INFO struct to be filled */ + int timeSlots, const UINT flags) { + FRAME_INFO *pFrameInfo = &h_frame_data->frameInfo; + INT numberTimeSlots = hHeaderData->numberTimeSlots; + INT temp = 0, k; + + /* FIXFIXonly framing case */ + h_frame_data->frameInfo.frameClass = 0; + + /* get the transient position from the bitstream */ + switch (timeSlots) { + case 8: + /* 3bit transient position (temp={0;..;7}) */ + temp = FDKreadBits(hBitBuf, 3); + break; + + case 16: + case 15: + /* 4bit transient position (temp={0;..;15}) */ + temp = FDKreadBits(hBitBuf, 4); + break; + + default: + return 0; + } + + /* For "case 15" only*/ + if (temp >= timeSlots) { + return 0; + } + + /* calculate borders according to the transient position */ + if (!generateFixFixOnly(pFrameInfo, temp, numberTimeSlots, flags)) { + return 0; + } + + /* decode freq res: */ + for (k = 0; k < pFrameInfo->nEnvelopes; k++) { + pFrameInfo->freqRes[k] = + (UCHAR)FDKreadBits(hBitBuf, 1); /* f = F [1 bits] */ + } + + return 1; +} + +/*! + \brief Extract the PVC frame information (structure FRAME_INFO) from the + bitstream \return Zero for bitstream error, one for correct. +*/ +int extractPvcFrameInfo( + HANDLE_FDK_BITSTREAM hBs, /*!< bitbuffer handle */ + HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ + HANDLE_SBR_FRAME_DATA h_frame_data, /*!< pointer to memory where the + frame-info will be stored */ + HANDLE_SBR_PREV_FRAME_DATA h_prev_frame_data, /*!< pointer to memory where + the previous frame-info + will be stored */ + UCHAR pvc_mode_last, /**< PVC mode of last frame */ + const UINT flags) { + FRAME_INFO *pFrameInfo = &h_frame_data->frameInfo; + FRAME_INFO *pPrevFrameInfo = &h_prev_frame_data->prevFrameInfo; + int bs_var_len_hf, bs_noise_position; + bs_noise_position = FDKreadBits(hBs, 4); /* SBR_PVC_NOISEPOSITION_BITS 4 */ + bs_var_len_hf = FDKreadBit(hBs); + pFrameInfo->noisePosition = bs_noise_position; + pFrameInfo->tranEnv = -1; + + /* Init for bs_noise_position == 0 in case a parse error is found below. */ + pFrameInfo->nEnvelopes = 1; + pFrameInfo->nNoiseEnvelopes = 1; + pFrameInfo->freqRes[0] = 0; + + if (bs_var_len_hf) { /* 1 or 3 Bits */ + pFrameInfo->varLength = FDKreadBits(hBs, 2) + 1; + if (pFrameInfo->varLength > 3) { + pFrameInfo->varLength = + 0; /* assume bs_var_len_hf == 0 in case of error */ + return 0; /* reserved value -> parse error */ + } + } else { + pFrameInfo->varLength = 0; + } + + if (bs_noise_position) { + pFrameInfo->nEnvelopes = 2; + pFrameInfo->nNoiseEnvelopes = 2; + FDKmemclear(pFrameInfo->freqRes, sizeof(pFrameInfo->freqRes)); + } + + /* frame border calculation */ + if (hHeaderData->bs_info.pvc_mode > 0) { + /* See "7.5.1.4 HF adjustment of SBR envelope scalefactors" for reference. + */ + + FDK_ASSERT((pFrameInfo->nEnvelopes == 1) || (pFrameInfo->nEnvelopes == 2)); + + /* left timeborder-offset: use the timeborder of prev SBR frame */ + if (pPrevFrameInfo->nEnvelopes > 0) { + pFrameInfo->borders[0] = + pPrevFrameInfo->borders[pPrevFrameInfo->nEnvelopes] - PVC_NTIMESLOT; + FDK_ASSERT(pFrameInfo->borders[0] <= 3); + } else { + pFrameInfo->borders[0] = 0; + } + + /* right timeborder-offset: */ + pFrameInfo->borders[pFrameInfo->nEnvelopes] = 16 + pFrameInfo->varLength; + + if (pFrameInfo->nEnvelopes == 2) { + pFrameInfo->borders[1] = pFrameInfo->noisePosition; + } + + /* Calculation of PVC time borders t_EPVC */ + if (pvc_mode_last == 0) { + /* there was a legacy SBR frame before this frame => use bs_var_len' for + * first PVC timeslot */ + pFrameInfo->pvcBorders[0] = pFrameInfo->borders[0]; + } else { + pFrameInfo->pvcBorders[0] = 0; + } + if (pFrameInfo->nEnvelopes == 2) { + pFrameInfo->pvcBorders[1] = pFrameInfo->borders[1]; + } + pFrameInfo->pvcBorders[pFrameInfo->nEnvelopes] = 16; + + /* calculation of SBR noise-floor time-border vector: */ + for (INT i = 0; i <= pFrameInfo->nNoiseEnvelopes; i++) { + pFrameInfo->bordersNoise[i] = pFrameInfo->borders[i]; + } + + pFrameInfo->tranEnv = -1; /* tranEnv not used */ + } + return 1; +} + +/*! + \brief Extract the frame information (structure FRAME_INFO) from the + bitstream \return Zero for bitstream error, one for correct. +*/ +int extractFrameInfo( + HANDLE_FDK_BITSTREAM hBs, /*!< bitbuffer handle */ + HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ + HANDLE_SBR_FRAME_DATA h_frame_data, /*!< pointer to memory where the + frame-info will be stored */ + const UINT nrOfChannels, const UINT flags) { + FRAME_INFO *pFrameInfo = &h_frame_data->frameInfo; + int numberTimeSlots = hHeaderData->numberTimeSlots; + int pointer_bits = 0, nEnv = 0, b = 0, border, i, n = 0, k, p, aL, aR, nL, nR, + temp = 0, staticFreqRes; + UCHAR frameClass; + + if (flags & SBRDEC_ELD_GRID) { + /* CODEC_AACLD (LD+SBR) only uses the normal 0 Grid for non-transient Frames + * and the LowDelayGrid for transient Frames */ + frameClass = FDKreadBits(hBs, 1); /* frameClass = [1 bit] */ + if (frameClass == 1) { + /* if frameClass == 1, extract LowDelaySbrGrid, otherwise extract normal + * SBR-Grid for FIXIFX */ + /* extract the AACLD-Sbr-Grid */ + pFrameInfo->frameClass = frameClass; + int err = 1; + err = extractLowDelayGrid(hBs, hHeaderData, h_frame_data, numberTimeSlots, + flags); + return err; + } + } else { + frameClass = FDKreadBits(hBs, 2); /* frameClass = C [2 bits] */ + } + + switch (frameClass) { + case 0: + temp = FDKreadBits(hBs, 2); /* E [2 bits ] */ + nEnv = (int)(1 << temp); /* E -> e */ + + if ((flags & SBRDEC_ELD_GRID) && (nEnv == 1)) + h_frame_data->ampResolutionCurrentFrame = + FDKreadBits(hBs, 1); /* new ELD Syntax 07-11-09 */ + + staticFreqRes = FDKreadBits(hBs, 1); + + if (flags & (SBRDEC_SYNTAX_USAC | SBRDEC_SYNTAX_RSVD50)) { + if (nEnv > MAX_ENVELOPES_USAC) return 0; + } else + + b = nEnv + 1; + switch (nEnv) { + case 1: + switch (numberTimeSlots) { + case 15: + FDKmemcpy(pFrameInfo, &FDK_sbrDecoder_sbr_frame_info1_15, + sizeof(FRAME_INFO)); + break; + case 16: + FDKmemcpy(pFrameInfo, &FDK_sbrDecoder_sbr_frame_info1_16, + sizeof(FRAME_INFO)); + break; + default: + FDK_ASSERT(0); + } + break; + case 2: + switch (numberTimeSlots) { + case 15: + FDKmemcpy(pFrameInfo, &FDK_sbrDecoder_sbr_frame_info2_15, + sizeof(FRAME_INFO)); + break; + case 16: + FDKmemcpy(pFrameInfo, &FDK_sbrDecoder_sbr_frame_info2_16, + sizeof(FRAME_INFO)); + break; + default: + FDK_ASSERT(0); + } + break; + case 4: + switch (numberTimeSlots) { + case 15: + FDKmemcpy(pFrameInfo, &FDK_sbrDecoder_sbr_frame_info4_15, + sizeof(FRAME_INFO)); + break; + case 16: + FDKmemcpy(pFrameInfo, &FDK_sbrDecoder_sbr_frame_info4_16, + sizeof(FRAME_INFO)); + break; + default: + FDK_ASSERT(0); + } + break; + case 8: +#if (MAX_ENVELOPES >= 8) + switch (numberTimeSlots) { + case 15: + FDKmemcpy(pFrameInfo, &FDK_sbrDecoder_sbr_frame_info8_15, + sizeof(FRAME_INFO)); + break; + case 16: + FDKmemcpy(pFrameInfo, &FDK_sbrDecoder_sbr_frame_info8_16, + sizeof(FRAME_INFO)); + break; + default: + FDK_ASSERT(0); + } + break; +#else + return 0; +#endif + } + /* Apply correct freqRes (High is default) */ + if (!staticFreqRes) { + for (i = 0; i < nEnv; i++) pFrameInfo->freqRes[i] = 0; + } + + break; + case 1: + case 2: + temp = FDKreadBits(hBs, 2); /* A [2 bits] */ + + n = FDKreadBits(hBs, 2); /* n = N [2 bits] */ + + nEnv = n + 1; /* # envelopes */ + b = nEnv + 1; /* # borders */ + + break; + } + + switch (frameClass) { + case 1: + /* Decode borders: */ + pFrameInfo->borders[0] = 0; /* first border */ + border = temp + numberTimeSlots; /* A -> aR */ + i = b - 1; /* frame info index for last border */ + pFrameInfo->borders[i] = border; /* last border */ + + for (k = 0; k < n; k++) { + temp = FDKreadBits(hBs, 2); /* R [2 bits] */ + border -= (2 * temp + 2); /* R -> r */ + pFrameInfo->borders[--i] = border; + } + + /* Decode pointer: */ + pointer_bits = DFRACT_BITS - 1 - CountLeadingBits((FIXP_DBL)(n + 1)); + p = FDKreadBits(hBs, pointer_bits); /* p = P [pointer_bits bits] */ + + if (p > n + 1) return 0; + + pFrameInfo->tranEnv = p ? n + 2 - p : -1; + + /* Decode freq res: */ + for (k = n; k >= 0; k--) { + pFrameInfo->freqRes[k] = FDKreadBits(hBs, 1); /* f = F [1 bits] */ + } + + /* Calculate noise floor middle border: */ + if (p == 0 || p == 1) + pFrameInfo->bordersNoise[1] = pFrameInfo->borders[n]; + else + pFrameInfo->bordersNoise[1] = pFrameInfo->borders[pFrameInfo->tranEnv]; + + break; + + case 2: + /* Decode borders: */ + border = temp; /* A -> aL */ + pFrameInfo->borders[0] = border; /* first border */ + + for (k = 1; k <= n; k++) { + temp = FDKreadBits(hBs, 2); /* R [2 bits] */ + border += (2 * temp + 2); /* R -> r */ + pFrameInfo->borders[k] = border; + } + pFrameInfo->borders[k] = numberTimeSlots; /* last border */ + + /* Decode pointer: */ + pointer_bits = DFRACT_BITS - 1 - CountLeadingBits((FIXP_DBL)(n + 1)); + p = FDKreadBits(hBs, pointer_bits); /* p = P [pointer_bits bits] */ + if (p > n + 1) return 0; + + if (p == 0 || p == 1) + pFrameInfo->tranEnv = -1; + else + pFrameInfo->tranEnv = p - 1; + + /* Decode freq res: */ + for (k = 0; k <= n; k++) { + pFrameInfo->freqRes[k] = FDKreadBits(hBs, 1); /* f = F [1 bits] */ + } + + /* Calculate noise floor middle border: */ + switch (p) { + case 0: + pFrameInfo->bordersNoise[1] = pFrameInfo->borders[1]; + break; + case 1: + pFrameInfo->bordersNoise[1] = pFrameInfo->borders[n]; + break; + default: + pFrameInfo->bordersNoise[1] = + pFrameInfo->borders[pFrameInfo->tranEnv]; + break; + } + + break; + + case 3: + /* v_ctrlSignal = [frameClass,aL,aR,nL,nR,v_rL,v_rR,p,v_fLR]; */ + + aL = FDKreadBits(hBs, 2); /* AL [2 bits], AL -> aL */ + + aR = FDKreadBits(hBs, 2) + numberTimeSlots; /* AR [2 bits], AR -> aR */ + + nL = FDKreadBits(hBs, 2); /* nL = NL [2 bits] */ + + nR = FDKreadBits(hBs, 2); /* nR = NR [2 bits] */ + + /*------------------------------------------------------------------------- + Calculate help variables + --------------------------------------------------------------------------*/ + + /* general: */ + nEnv = nL + nR + 1; /* # envelopes */ + if (nEnv > MAX_ENVELOPES) return 0; + b = nEnv + 1; /* # borders */ + + /*------------------------------------------------------------------------- + Decode envelopes + --------------------------------------------------------------------------*/ + + /* L-borders: */ + border = aL; /* first border */ + pFrameInfo->borders[0] = border; + + for (k = 1; k <= nL; k++) { + temp = FDKreadBits(hBs, 2); /* R [2 bits] */ + border += (2 * temp + 2); /* R -> r */ + pFrameInfo->borders[k] = border; + } + + /* R-borders: */ + border = aR; /* last border */ + i = nEnv; + + pFrameInfo->borders[i] = border; + + for (k = 0; k < nR; k++) { + temp = FDKreadBits(hBs, 2); /* R [2 bits] */ + border -= (2 * temp + 2); /* R -> r */ + pFrameInfo->borders[--i] = border; + } + + /* decode pointer: */ + pointer_bits = + DFRACT_BITS - 1 - CountLeadingBits((FIXP_DBL)(nL + nR + 1)); + p = FDKreadBits(hBs, pointer_bits); /* p = P [pointer_bits bits] */ + + if (p > nL + nR + 1) return 0; + + pFrameInfo->tranEnv = p ? b - p : -1; + + /* decode freq res: */ + for (k = 0; k < nEnv; k++) { + pFrameInfo->freqRes[k] = FDKreadBits(hBs, 1); /* f = F [1 bits] */ + } + + /*------------------------------------------------------------------------- + Decode noise floors + --------------------------------------------------------------------------*/ + pFrameInfo->bordersNoise[0] = aL; + + if (nEnv == 1) { + /* 1 noise floor envelope: */ + pFrameInfo->bordersNoise[1] = aR; + } else { + /* 2 noise floor envelopes */ + if (p == 0 || p == 1) + pFrameInfo->bordersNoise[1] = pFrameInfo->borders[nEnv - 1]; + else + pFrameInfo->bordersNoise[1] = + pFrameInfo->borders[pFrameInfo->tranEnv]; + pFrameInfo->bordersNoise[2] = aR; + } + break; + } + + /* + Store number of envelopes, noise floor envelopes and frame class + */ + pFrameInfo->nEnvelopes = nEnv; + + if (nEnv == 1) + pFrameInfo->nNoiseEnvelopes = 1; + else + pFrameInfo->nNoiseEnvelopes = 2; + + pFrameInfo->frameClass = frameClass; + + if (pFrameInfo->frameClass == 2 || pFrameInfo->frameClass == 1) { + /* calculate noise floor first and last borders: */ + pFrameInfo->bordersNoise[0] = pFrameInfo->borders[0]; + pFrameInfo->bordersNoise[pFrameInfo->nNoiseEnvelopes] = + pFrameInfo->borders[nEnv]; + } + + return 1; +} + +/*! + \brief Check if the frameInfo vector has reasonable values. + \return Zero for error, one for correct +*/ +static int checkFrameInfo( + FRAME_INFO *pFrameInfo, /*!< pointer to frameInfo */ + int numberOfTimeSlots, /*!< QMF time slots per frame */ + int overlap, /*!< Amount of overlap QMF time slots */ + int timeStep) /*!< QMF slots to SBR slots step factor */ +{ + int maxPos, i, j; + int startPos; + int stopPos; + int tranEnv; + int startPosNoise; + int stopPosNoise; + int nEnvelopes = pFrameInfo->nEnvelopes; + int nNoiseEnvelopes = pFrameInfo->nNoiseEnvelopes; + + if (nEnvelopes < 1 || nEnvelopes > MAX_ENVELOPES) return 0; + + if (nNoiseEnvelopes > MAX_NOISE_ENVELOPES) return 0; + + startPos = pFrameInfo->borders[0]; + stopPos = pFrameInfo->borders[nEnvelopes]; + tranEnv = pFrameInfo->tranEnv; + startPosNoise = pFrameInfo->bordersNoise[0]; + stopPosNoise = pFrameInfo->bordersNoise[nNoiseEnvelopes]; + + if (overlap < 0 || overlap > (3 * (4))) { + return 0; + } + if (timeStep < 1 || timeStep > (4)) { + return 0; + } + maxPos = numberOfTimeSlots + (overlap / timeStep); + + /* Check that the start and stop positions of the frame are reasonable values. + */ + if ((startPos < 0) || (startPos >= stopPos)) return 0; + if (startPos > maxPos - numberOfTimeSlots) /* First env. must start in or + directly after the overlap + buffer */ + return 0; + if (stopPos < numberOfTimeSlots) /* One complete frame must be ready for + output after processing */ + return 0; + if (stopPos > maxPos) return 0; + + /* Check that the start border for every envelope is strictly later in time + */ + for (i = 0; i < nEnvelopes; i++) { + if (pFrameInfo->borders[i] >= pFrameInfo->borders[i + 1]) return 0; + } + + /* Check that the envelope to be shortened is actually among the envelopes */ + if (tranEnv > nEnvelopes) return 0; + + /* Check the noise borders */ + if (nEnvelopes == 1 && nNoiseEnvelopes > 1) return 0; + + if (startPos != startPosNoise || stopPos != stopPosNoise) return 0; + + /* Check that the start border for every noise-envelope is strictly later in + * time*/ + for (i = 0; i < nNoiseEnvelopes; i++) { + if (pFrameInfo->bordersNoise[i] >= pFrameInfo->bordersNoise[i + 1]) + return 0; + } + + /* Check that every noise border is the same as an envelope border*/ + for (i = 0; i < nNoiseEnvelopes; i++) { + startPosNoise = pFrameInfo->bordersNoise[i]; + + for (j = 0; j < nEnvelopes; j++) { + if (pFrameInfo->borders[j] == startPosNoise) break; + } + if (j == nEnvelopes) return 0; + } + + return 1; +} diff --git a/fdk-aac/libSBRdec/src/env_extr.h b/fdk-aac/libSBRdec/src/env_extr.h new file mode 100644 index 0000000..38c04a3 --- /dev/null +++ b/fdk-aac/libSBRdec/src/env_extr.h @@ -0,0 +1,415 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief Envelope extraction prototypes +*/ + +#ifndef ENV_EXTR_H +#define ENV_EXTR_H + +#include "sbrdecoder.h" + +#include "FDK_bitstream.h" +#include "lpp_tran.h" + +#include "psdec.h" +#include "pvc_dec.h" + +#define ENV_EXP_FRACT 0 +/*!< Shift raw envelope data to support fractional numbers. + Can be set to 8 instead of 0 to enhance accuracy during concealment. + This is not required for conformance and #requantizeEnvelopeData() will + become more expensive. +*/ + +#define EXP_BITS 6 +/*!< Size of exponent-part of a pseudo float envelope value (should be at least + 6). The remaining bits in each word are used for the mantissa (should be at + least 10). This format is used in the arrays iEnvelope[] and + sbrNoiseFloorLevel[] in the FRAME_DATA struct which must fit in a certain part + of the output buffer (See buffer management in sbr_dec.cpp). Exponents and + mantissas could also be stored in separate arrays. Accessing the exponent or + the mantissa would be simplified and the masks #MASK_E resp. #MASK_M would + no longer be required. +*/ + +#define MASK_M \ + (((1 << (FRACT_BITS - EXP_BITS)) - 1) \ + << EXP_BITS) /*!< Mask for extracting the mantissa of a pseudo float \ + envelope value */ +#define MASK_E \ + ((1 << EXP_BITS) - 1) /*!< Mask for extracting the exponent of a pseudo \ + float envelope value */ + +#define SIGN_EXT \ + (((SCHAR)-1) ^ \ + MASK_E) /*!< a CHAR-constant with all bits above our sign-bit set */ +#define ROUNDING \ + ((FIXP_SGL)( \ + 1 << (EXP_BITS - 1))) /*!< 0.5-offset for rounding the mantissa of a \ + pseudo-float envelope value */ +#define NRG_EXP_OFFSET \ + 16 /*!< Will be added to the reference energy's exponent to prevent negative \ + numbers */ +#define NOISE_EXP_OFFSET \ + 38 /*!< Will be added to the noise level exponent to prevent negative \ + numbers */ + +#define ADD_HARMONICS_FLAGS_SIZE 2 /* ceil(MAX_FREQ_COEFFS/32) */ + +typedef enum { + HEADER_NOT_PRESENT, + HEADER_ERROR, + HEADER_OK, + HEADER_RESET +} SBR_HEADER_STATUS; + +typedef enum { + SBR_NOT_INITIALIZED = 0, + UPSAMPLING = 1, + SBR_HEADER = 2, + SBR_ACTIVE = 3 +} SBR_SYNC_STATE; + +typedef enum { COUPLING_OFF = 0, COUPLING_LEVEL, COUPLING_BAL } COUPLING_MODE; + +typedef struct { + UCHAR nSfb[2]; /*!< Number of SBR-bands for low and high freq-resolution */ + UCHAR nNfb; /*!< Actual number of noise bands to read from the bitstream*/ + UCHAR numMaster; /*!< Number of SBR-bands in v_k_master */ + UCHAR lowSubband; /*!< QMF-band where SBR frequency range starts */ + UCHAR highSubband; /*!< QMF-band where SBR frequency range ends */ + UCHAR ov_highSubband; /*!< if headerchange applies this value holds the old + highband value -> highband value of overlap area; + required for overlap in usac when headerchange + occurs between XVAR and VARX frame */ + UCHAR limiterBandTable[MAX_NUM_LIMITERS + 1]; /*!< Limiter band table. */ + UCHAR noLimiterBands; /*!< Number of limiter bands. */ + UCHAR nInvfBands; /*!< Number of bands for inverse filtering */ + UCHAR + *freqBandTable[2]; /*!< Pointers to freqBandTableLo and freqBandTableHi */ + UCHAR freqBandTableLo[MAX_FREQ_COEFFS / 2 + 1]; + /*!< Mapping of SBR bands to QMF bands for low frequency resolution */ + UCHAR freqBandTableHi[MAX_FREQ_COEFFS + 1]; + /*!< Mapping of SBR bands to QMF bands for high frequency resolution */ + UCHAR freqBandTableNoise[MAX_NOISE_COEFFS + 1]; + /*!< Mapping of SBR noise bands to QMF bands */ + UCHAR v_k_master[MAX_FREQ_COEFFS + 1]; + /*!< Master BandTable which freqBandTable is derived from */ +} FREQ_BAND_DATA; + +typedef FREQ_BAND_DATA *HANDLE_FREQ_BAND_DATA; + +#define SBRDEC_ELD_GRID 1 +#define SBRDEC_SYNTAX_SCAL 2 +#define SBRDEC_SYNTAX_USAC 4 +#define SBRDEC_SYNTAX_RSVD50 8 +#define SBRDEC_USAC_INDEP \ + 16 /* Flag indicating that USAC global independency flag is active. */ +#define SBRDEC_LOW_POWER \ + 32 /* Flag indicating that Low Power QMF mode shall be used. */ +#define SBRDEC_PS_DECODED \ + 64 /* Flag indicating that PS was decoded and rendered. */ +#define SBRDEC_QUAD_RATE \ + 128 /* Flag indicating that USAC SBR 4:1 is active. \ + */ +#define SBRDEC_USAC_HARMONICSBR \ + 256 /* Flag indicating that USAC HBE tool is active. */ +#define SBRDEC_LD_MPS_QMF \ + 512 /* Flag indicating that the LD-MPS QMF shall be used. */ +#define SBRDEC_USAC_ITES \ + 1024 /* Flag indicating that USAC inter TES tool is active. */ +#define SBRDEC_SYNTAX_DRM \ + 2048 /* Flag indicating that DRM30/DRM+ reverse syntax is being used. */ +#define SBRDEC_ELD_DOWNSCALE \ + 4096 /* Flag indicating that ELD downscaled mode decoding is used */ +#define SBRDEC_DOWNSAMPLE \ + 8192 /* Flag indicating that the downsampling mode is used. */ +#define SBRDEC_FLUSH 16384 /* Flag is used to flush all elements in use. */ +#define SBRDEC_FORCE_RESET \ + 32768 /* Flag is used to force a reset of all elements in use. */ +#define SBRDEC_SKIP_QMF_ANA \ + (1 << 21) /* Flag indicating that the input data is provided in the QMF \ + domain. */ +#define SBRDEC_SKIP_QMF_SYN \ + (1 << 22) /* Flag indicating that the output data is exported in the QMF \ + domain. */ + +#define SBRDEC_HDR_STAT_RESET 1 +#define SBRDEC_HDR_STAT_UPDATE 2 + +typedef struct { + UCHAR ampResolution; /*!< Amplitude resolution of envelope values (0: 1.5dB, + 1: 3dB) */ + UCHAR + xover_band; /*!< Start index in #v_k_master[] used for dynamic crossover + frequency */ + UCHAR sbr_preprocessing; /*!< SBR prewhitening flag. */ + UCHAR pvc_mode; /*!< Predictive vector coding mode */ +} SBR_HEADER_DATA_BS_INFO; + +typedef struct { + /* Changes in these variables causes a reset of the decoder */ + UCHAR startFreq; /*!< Index for SBR start frequency */ + UCHAR stopFreq; /*!< Index for SBR highest frequency */ + UCHAR freqScale; /*!< 0: linear scale, 1-3 logarithmic scales */ + UCHAR alterScale; /*!< Flag for coarser frequency resolution */ + UCHAR noise_bands; /*!< Noise bands per octave, read from bitstream*/ + + /* don't require reset */ + UCHAR limiterBands; /*!< Index for number of limiter bands per octave */ + UCHAR limiterGains; /*!< Index to select gain limit */ + UCHAR interpolFreq; /*!< Select gain calculation method (1: per QMF channel, + 0: per SBR band) */ + UCHAR smoothingLength; /*!< Smoothing of gains over time (0: on 1: off) */ + +} SBR_HEADER_DATA_BS; + +typedef struct { + SBR_SYNC_STATE + syncState; /*!< The current initialization status of the header */ + + UCHAR status; /*!< Flags field used for signaling a reset right before the + processing starts and an update from config (e.g. ASC). */ + UCHAR + frameErrorFlag; /*!< Frame data valid flag. CAUTION: This variable will be + overwritten by the flag stored in the element + structure. This is necessary because of the frame + delay. There it might happen that different slots use + the same header. */ + UCHAR numberTimeSlots; /*!< AAC: 16,15 */ + UCHAR numberOfAnalysisBands; /*!< Number of QMF analysis bands */ + UCHAR timeStep; /*!< Time resolution of SBR in QMF-slots */ + UINT + sbrProcSmplRate; /*!< SBR processing sampling frequency (!= + OutputSamplingRate) (always: CoreSamplingRate * + UpSamplingFactor; even in single rate mode) */ + + SBR_HEADER_DATA_BS bs_data; /*!< current SBR header. */ + SBR_HEADER_DATA_BS bs_dflt; /*!< Default sbr header. */ + SBR_HEADER_DATA_BS_INFO bs_info; /*!< SBR info. */ + + FREQ_BAND_DATA freqBandData; /*!< Pointer to struct #FREQ_BAND_DATA */ + UCHAR pvcIDprev; +} SBR_HEADER_DATA; + +typedef SBR_HEADER_DATA *HANDLE_SBR_HEADER_DATA; + +typedef struct { + UCHAR frameClass; /*!< Select grid type */ + UCHAR nEnvelopes; /*!< Number of envelopes */ + UCHAR borders[MAX_ENVELOPES + 1]; /*!< Envelope borders (in SBR-timeslots, + e.g. mp3PRO: 0..11) */ + UCHAR freqRes[MAX_ENVELOPES]; /*!< Frequency resolution for each envelope + (0=low, 1=high) */ + SCHAR tranEnv; /*!< Transient envelope, -1 if none */ + UCHAR nNoiseEnvelopes; /*!< Number of noise envelopes */ + UCHAR + bordersNoise[MAX_NOISE_ENVELOPES + 1]; /*!< borders of noise envelopes */ + UCHAR pvcBorders[MAX_PVC_ENVELOPES + 1]; + UCHAR noisePosition; + UCHAR varLength; +} FRAME_INFO; + +typedef struct { + FIXP_SGL sfb_nrg_prev[MAX_FREQ_COEFFS]; /*!< Previous envelope (required for + differential-coded values) */ + FIXP_SGL + prevNoiseLevel[MAX_NOISE_COEFFS]; /*!< Previous noise envelope (required + for differential-coded values) */ + COUPLING_MODE coupling; /*!< Stereo-mode of previous frame */ + INVF_MODE sbr_invf_mode[MAX_INVF_BANDS]; /*!< Previous strength of filtering + in transposer */ + UCHAR ampRes; /*!< Previous amplitude resolution (0: 1.5dB, 1: 3dB) */ + UCHAR stopPos; /*!< Position in time where last envelope ended */ + UCHAR frameErrorFlag; /*!< Previous frame status */ + UCHAR prevSbrPitchInBins; /*!< Previous frame pitchInBins */ + FRAME_INFO prevFrameInfo; +} SBR_PREV_FRAME_DATA; + +typedef SBR_PREV_FRAME_DATA *HANDLE_SBR_PREV_FRAME_DATA; + +typedef struct { + int nScaleFactors; /*!< total number of scalefactors in frame */ + + FRAME_INFO frameInfo; /*!< time grid for current frame */ + UCHAR domain_vec[MAX_ENVELOPES]; /*!< Bitfield containing direction of + delta-coding for each envelope + (0:frequency, 1:time) */ + UCHAR domain_vec_noise + [MAX_NOISE_ENVELOPES]; /*!< Same as above, but for noise envelopes */ + + INVF_MODE + sbr_invf_mode[MAX_INVF_BANDS]; /*!< Strength of filtering in transposer */ + COUPLING_MODE coupling; /*!< Stereo-mode */ + int ampResolutionCurrentFrame; /*!< Amplitude resolution of envelope values + (0: 1.5dB, 1: 3dB) */ + + ULONG addHarmonics[ADD_HARMONICS_FLAGS_SIZE]; /*!< Flags for synthetic sine + addition (aligned to MSB) */ + + FIXP_SGL iEnvelope[MAX_NUM_ENVELOPE_VALUES]; /*!< Envelope data */ + FIXP_SGL sbrNoiseFloorLevel[MAX_NUM_NOISE_VALUES]; /*!< Noise envelope data */ + UCHAR iTESactive; /*!< One flag for each envelope to enable USAC inter-TES */ + UCHAR + interTempShapeMode[MAX_ENVELOPES]; /*!< USAC inter-TES: + bs_inter_temp_shape_mode[ch][env] + value */ + UCHAR pvcID[PVC_NTIMESLOT]; /*!< One PVC ID value for each time slot */ + UCHAR ns; + UCHAR sinusoidal_position; + + UCHAR sbrPatchingMode; + UCHAR sbrOversamplingFlag; + UCHAR sbrPitchInBins; +} SBR_FRAME_DATA; + +typedef SBR_FRAME_DATA *HANDLE_SBR_FRAME_DATA; + +/*! +\brief Maps sampling frequencies to frequencies for which setup tables are +available + +Maps arbitary sampling frequency to nearest neighbors for which setup tables +are available (e.g. 25600 -> 24000). +Used for startFreq calculation. +The mapping is defined in 14496-3 (4.6.18.2.6), fs(SBR), and table 4.82 + +\return mapped sampling frequency +*/ +UINT sbrdec_mapToStdSampleRate(UINT fs, + UINT isUsac); /*!< Output sampling frequency */ + +void initSbrPrevFrameData(HANDLE_SBR_PREV_FRAME_DATA h_prev_data, + int timeSlots); + +int sbrGetChannelElement(HANDLE_SBR_HEADER_DATA hHeaderData, + HANDLE_SBR_FRAME_DATA hFrameDataLeft, + HANDLE_SBR_FRAME_DATA hFrameDataRight, + HANDLE_SBR_PREV_FRAME_DATA hFrameDataLeftPrev, + UCHAR pvc_mode_last, HANDLE_FDK_BITSTREAM hBitBuf, + HANDLE_PS_DEC hParametricStereoDec, const UINT flags, + const int overlap); + +SBR_HEADER_STATUS +sbrGetHeaderData(HANDLE_SBR_HEADER_DATA headerData, + HANDLE_FDK_BITSTREAM hBitBuf, const UINT flags, + const int fIsSbrData, const UCHAR configMode); + +/*! + \brief Initialize SBR header data + + Copy default values to the header data struct and patch some entries + depending on the core codec. +*/ +SBR_ERROR +initHeaderData(HANDLE_SBR_HEADER_DATA hHeaderData, const int sampleRateIn, + const int sampleRateOut, const INT downscaleFactor, + const int samplesPerFrame, const UINT flags, + const int setDefaultHdr); +#endif + +/* Convert headroom bits to exponent */ +#define SCALE2EXP(s) (15 - (s)) +#define EXP2SCALE(e) (15 - (e)) diff --git a/fdk-aac/libSBRdec/src/hbe.cpp b/fdk-aac/libSBRdec/src/hbe.cpp new file mode 100644 index 0000000..3310dcd --- /dev/null +++ b/fdk-aac/libSBRdec/src/hbe.cpp @@ -0,0 +1,2202 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief Fast FFT routines prototypes + \author Fabian Haussel +*/ + +#include "hbe.h" +#include "qmf.h" +#include "env_extr.h" + +#define HBE_MAX_QMF_BANDS (40) + +#define HBE_MAX_OUT_SLOTS (11) + +#define QMF_WIN_LEN \ + (12 + 6 - 4 - 1) /* 6 subband slots extra delay to align with HQ - 4 slots \ + to compensate for critical sampling delay - 1 slot to \ + align critical sampling exactly (w additional time \ + domain delay)*/ + +#ifndef PI +#define PI 3.14159265358979323846 +#endif + +static const int xProducts[MAX_STRETCH_HBE - 1] = { + 1, 1, 1}; /* Cross products on(1)/off(0) for T=2,3,4. */ +static const int startSubband2kL[33] = { + 0, 0, 0, 0, 0, 0, 0, 2, 2, 2, 4, 4, 4, 4, 4, 6, 6, + 6, 8, 8, 8, 8, 8, 10, 10, 10, 12, 12, 12, 12, 12, 12, 12}; + +static const int pmin = 12; + +static const FIXP_DBL hintReal_F[4][3] = { + {FL2FXCONST_DBL(0.39840335f), FL2FXCONST_DBL(0.39840335f), + FL2FXCONST_DBL(-0.39840335f)}, + {FL2FXCONST_DBL(0.39840335f), FL2FXCONST_DBL(-0.39840335f), + FL2FXCONST_DBL(-0.39840335f)}, + {FL2FXCONST_DBL(-0.39840335f), FL2FXCONST_DBL(-0.39840335f), + FL2FXCONST_DBL(0.39840335f)}, + {FL2FXCONST_DBL(-0.39840335f), FL2FXCONST_DBL(0.39840335f), + FL2FXCONST_DBL(0.39840335f)}}; + +static const FIXP_DBL factors[4] = { + FL2FXCONST_DBL(0.39840335f), FL2FXCONST_DBL(-0.39840335f), + FL2FXCONST_DBL(-0.39840335f), FL2FXCONST_DBL(0.39840335f)}; + +#define PSCALE 32 + +static const FIXP_DBL p_F[128] = {FL2FXCONST_DBL(0.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(1.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(2.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(3.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(4.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(5.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(6.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(7.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(8.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(9.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(10.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(11.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(12.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(13.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(14.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(15.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(16.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(17.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(18.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(19.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(20.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(21.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(22.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(23.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(24.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(25.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(26.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(27.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(28.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(29.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(30.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(31.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(32.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(33.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(34.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(35.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(36.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(37.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(38.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(39.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(40.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(41.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(42.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(43.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(44.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(45.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(46.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(47.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(48.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(49.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(50.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(51.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(52.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(53.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(54.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(55.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(56.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(57.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(58.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(59.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(60.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(61.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(62.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(63.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(64.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(65.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(66.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(67.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(68.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(69.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(70.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(71.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(72.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(73.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(74.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(75.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(76.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(77.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(78.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(79.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(80.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(81.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(82.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(83.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(84.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(85.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(86.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(87.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(88.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(89.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(90.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(91.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(92.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(93.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(94.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(95.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(96.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(97.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(98.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(99.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(100.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(101.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(102.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(103.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(104.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(105.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(106.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(107.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(108.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(109.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(110.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(111.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(112.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(113.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(114.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(115.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(116.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(117.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(118.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(119.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(120.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(121.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(122.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(123.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(124.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(125.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(126.f / (PSCALE * 12.f)), + FL2FXCONST_DBL(127.f / (PSCALE * 12.f))}; + +static const FIXP_DBL band_F[64] = { + FL2FXCONST_DBL((0.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((1.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((2.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((3.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((4.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((5.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((6.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((7.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((8.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((9.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((10.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((11.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((12.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((13.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((14.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((15.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((16.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((17.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((18.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((19.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((20.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((21.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((22.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((23.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((24.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((25.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((26.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((27.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((28.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((29.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((30.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((31.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((32.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((33.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((34.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((35.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((36.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((37.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((38.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((39.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((40.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((41.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((42.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((43.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((44.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((45.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((46.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((47.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((48.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((49.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((50.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((51.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((52.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((53.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((54.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((55.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((56.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((57.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((58.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((59.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((60.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((61.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((62.f * 2.f + 1) / (PSCALE << 2)), + FL2FXCONST_DBL((63.f * 2.f + 1) / (PSCALE << 2))}; + +static const FIXP_DBL tr_str[3] = {FL2FXCONST_DBL(1.f / 4.f), + FL2FXCONST_DBL(2.f / 4.f), + FL2FXCONST_DBL(3.f / 4.f)}; + +static const FIXP_DBL stretchfac[3] = {FL2FXCONST_DBL(1.f / 2.f), + FL2FXCONST_DBL(1.f / 3.f), + FL2FXCONST_DBL(1.f / 4.f)}; + +static const FIXP_DBL cos_F[64] = { + 26353028, -79043208, 131685776, -184244944, 236697216, -289006912, + 341142496, -393072608, 444773984, -496191392, 547325824, -598114752, + 648559104, -698597248, 748230016, -797411904, 846083200, -894275136, + 941928192, -989013760, 1035474624, -1081340672, 1126555136, -1171063296, + 1214893696, -1257992192, 1300332544, -1341889408, 1382612736, -1422503808, + 1461586944, -1499741440, 1537039104, -1573364864, 1608743808, -1643196672, + 1676617344, -1709028992, 1740450560, -1770784896, 1800089472, -1828273536, + 1855357440, -1881356288, 1906190080, -1929876608, 1952428928, -1973777664, + 1993962880, -2012922240, 2030670208, -2047216000, 2062508288, -2076559488, + 2089376128, -2100932224, 2111196800, -2120214784, 2127953792, -2134394368, + 2139565056, -2143444864, 2146026624, -2147321856}; + +static const FIXP_DBL twiddle[121] = {1073741824, + 1071442860, + 1064555814, + 1053110176, + 1037154959, + 1016758484, + 992008094, + 963009773, + 929887697, + 892783698, + 851856663, + 807281846, + 759250125, + 707967178, + 653652607, + 596538995, + 536870912, + 474903865, + 410903207, + 345142998, + 277904834, + 209476638, + 140151432, + 70226075, + 0, + -70226075, + -140151432, + -209476638, + -277904834, + -345142998, + -410903207, + -474903865, + -536870912, + -596538995, + -653652607, + -707967178, + -759250125, + -807281846, + -851856663, + -892783698, + -929887697, + -963009773, + -992008094, + -1016758484, + -1037154959, + -1053110176, + -1064555814, + -1071442860, + -1073741824, + -1071442860, + -1064555814, + -1053110176, + -1037154959, + -1016758484, + -992008094, + -963009773, + -929887697, + -892783698, + -851856663, + -807281846, + -759250125, + -707967178, + -653652607, + -596538995, + -536870912, + -474903865, + -410903207, + -345142998, + -277904834, + -209476638, + -140151432, + -70226075, + 0, + 70226075, + 140151432, + 209476638, + 277904834, + 345142998, + 410903207, + 474903865, + 536870912, + 596538995, + 653652607, + 707967178, + 759250125, + 807281846, + 851856663, + 892783698, + 929887697, + 963009773, + 992008094, + 1016758484, + 1037154959, + 1053110176, + 1064555814, + 1071442860, + 1073741824, + 1071442860, + 1064555814, + 1053110176, + 1037154959, + 1016758484, + 992008094, + 963009773, + 929887697, + 892783698, + 851856663, + 807281846, + 759250125, + 707967178, + 653652607, + 596538995, + 536870912, + 474903865, + 410903207, + 345142998, + 277904834, + 209476638, + 140151432, + 70226075, + 0}; + +#if FIXP_QTW == FIXP_SGL +#define HTW(x) (x) +#else +#define HTW(x) FX_DBL2FX_QTW(FX_SGL2FX_DBL((const FIXP_SGL)x)) +#endif + +static const FIXP_QTW post_twiddle_cos_8[8] = { + HTW(-1606), HTW(4756), HTW(-7723), HTW(10394), + HTW(-12665), HTW(14449), HTW(-15679), HTW(16305)}; + +static const FIXP_QTW post_twiddle_cos_16[16] = { + HTW(-804), HTW(2404), HTW(-3981), HTW(5520), HTW(-7005), HTW(8423), + HTW(-9760), HTW(11003), HTW(-12140), HTW(13160), HTW(-14053), HTW(14811), + HTW(-15426), HTW(15893), HTW(-16207), HTW(16364)}; + +static const FIXP_QTW post_twiddle_cos_24[24] = { + HTW(-536), HTW(1606), HTW(-2669), HTW(3720), HTW(-4756), HTW(5771), + HTW(-6762), HTW(7723), HTW(-8652), HTW(9543), HTW(-10394), HTW(11200), + HTW(-11958), HTW(12665), HTW(-13318), HTW(13913), HTW(-14449), HTW(14924), + HTW(-15334), HTW(15679), HTW(-15956), HTW(16165), HTW(-16305), HTW(16375)}; + +static const FIXP_QTW post_twiddle_cos_32[32] = { + HTW(-402), HTW(1205), HTW(-2006), HTW(2801), HTW(-3590), HTW(4370), + HTW(-5139), HTW(5897), HTW(-6639), HTW(7366), HTW(-8076), HTW(8765), + HTW(-9434), HTW(10080), HTW(-10702), HTW(11297), HTW(-11866), HTW(12406), + HTW(-12916), HTW(13395), HTW(-13842), HTW(14256), HTW(-14635), HTW(14978), + HTW(-15286), HTW(15557), HTW(-15791), HTW(15986), HTW(-16143), HTW(16261), + HTW(-16340), HTW(16379)}; + +static const FIXP_QTW post_twiddle_cos_40[40] = { + HTW(-322), HTW(965), HTW(-1606), HTW(2245), HTW(-2880), HTW(3511), + HTW(-4137), HTW(4756), HTW(-5368), HTW(5971), HTW(-6566), HTW(7150), + HTW(-7723), HTW(8285), HTW(-8833), HTW(9368), HTW(-9889), HTW(10394), + HTW(-10883), HTW(11356), HTW(-11810), HTW(12247), HTW(-12665), HTW(13063), + HTW(-13441), HTW(13799), HTW(-14135), HTW(14449), HTW(-14741), HTW(15011), + HTW(-15257), HTW(15480), HTW(-15679), HTW(15853), HTW(-16003), HTW(16129), + HTW(-16229), HTW(16305), HTW(-16356), HTW(16381)}; + +static const FIXP_QTW post_twiddle_sin_8[8] = { + HTW(16305), HTW(-15679), HTW(14449), HTW(-12665), + HTW(10394), HTW(-7723), HTW(4756), HTW(-1606)}; + +static const FIXP_QTW post_twiddle_sin_16[16] = { + HTW(16364), HTW(-16207), HTW(15893), HTW(-15426), HTW(14811), HTW(-14053), + HTW(13160), HTW(-12140), HTW(11003), HTW(-9760), HTW(8423), HTW(-7005), + HTW(5520), HTW(-3981), HTW(2404), HTW(-804)}; + +static const FIXP_QTW post_twiddle_sin_24[24] = { + HTW(16375), HTW(-16305), HTW(16165), HTW(-15956), HTW(15679), HTW(-15334), + HTW(14924), HTW(-14449), HTW(13913), HTW(-13318), HTW(12665), HTW(-11958), + HTW(11200), HTW(-10394), HTW(9543), HTW(-8652), HTW(7723), HTW(-6762), + HTW(5771), HTW(-4756), HTW(3720), HTW(-2669), HTW(1606), HTW(-536)}; + +static const FIXP_QTW post_twiddle_sin_32[32] = { + HTW(16379), HTW(-16340), HTW(16261), HTW(-16143), HTW(15986), HTW(-15791), + HTW(15557), HTW(-15286), HTW(14978), HTW(-14635), HTW(14256), HTW(-13842), + HTW(13395), HTW(-12916), HTW(12406), HTW(-11866), HTW(11297), HTW(-10702), + HTW(10080), HTW(-9434), HTW(8765), HTW(-8076), HTW(7366), HTW(-6639), + HTW(5897), HTW(-5139), HTW(4370), HTW(-3590), HTW(2801), HTW(-2006), + HTW(1205), HTW(-402)}; + +static const FIXP_QTW post_twiddle_sin_40[40] = { + HTW(16381), HTW(-16356), HTW(16305), HTW(-16229), HTW(16129), HTW(-16003), + HTW(15853), HTW(-15679), HTW(15480), HTW(-15257), HTW(15011), HTW(-14741), + HTW(14449), HTW(-14135), HTW(13799), HTW(-13441), HTW(13063), HTW(-12665), + HTW(12247), HTW(-11810), HTW(11356), HTW(-10883), HTW(10394), HTW(-9889), + HTW(9368), HTW(-8833), HTW(8285), HTW(-7723), HTW(7150), HTW(-6566), + HTW(5971), HTW(-5368), HTW(4756), HTW(-4137), HTW(3511), HTW(-2880), + HTW(2245), HTW(-1606), HTW(965), HTW(-322)}; + +static const FIXP_DBL preModCos[32] = { + -749875776, 786681536, 711263552, -821592064, -670937792, 854523392, + 628995648, -885396032, -585538240, 914135680, 540670208, -940673088, + -494499680, 964944384, 447137824, -986891008, -398698816, 1006460096, + 349299264, -1023604544, -299058240, 1038283072, 248096752, -1050460288, + -196537584, 1060106816, 144504928, -1067199488, -92124160, 1071721152, + 39521456, -1073660992}; + +static const FIXP_DBL preModSin[32] = { + 768510144, 730789760, -804379072, -691308864, 838310208, 650162560, + -870221760, -607449920, 900036928, 563273856, -927683776, -517740896, + 953095808, 470960608, -976211712, -423045728, 996975808, 374111712, + -1015338112, -324276416, 1031254400, 273659904, -1044686336, -222384144, + 1055601472, 170572640, -1063973632, -118350192, 1069782528, 65842640, + -1073014208, -13176464}; + +/* The cube root function */ +/***************************************************************************** + + functionname: invCubeRootNorm2 + description: delivers 1/cuberoot(op) in Q1.31 format and modified exponent + +*****************************************************************************/ +#define CUBE_ROOT_BITS 7 +#define CUBE_ROOT_VALUES (128 + 2) +#define CUBE_ROOT_BITS_MASK 0x7f +#define CUBE_ROOT_FRACT_BITS_MASK 0x007FFFFF +/* Inverse cube root table for operands running from 0.5 to 1.0 */ +/* (INT) (1.0/cuberoot((op))); */ +/* Implicit exponent is 1. */ + +LNK_SECTION_CONSTDATA +static const FIXP_DBL invCubeRootTab[CUBE_ROOT_VALUES] = { + (0x50a28be6), (0x506d1172), (0x503823c4), (0x5003c05a), (0x4fcfe4c0), + (0x4f9c8e92), (0x4f69bb7d), (0x4f37693b), (0x4f059594), (0x4ed43e5f), + (0x4ea36181), (0x4e72fcea), (0x4e430e98), (0x4e139495), (0x4de48cf5), + (0x4db5f5db), (0x4d87cd73), (0x4d5a11f2), (0x4d2cc19c), (0x4cffdabb), + (0x4cd35ba4), (0x4ca742b7), (0x4c7b8e5c), (0x4c503d05), (0x4c254d2a), + (0x4bfabd50), (0x4bd08c00), (0x4ba6b7cd), (0x4b7d3f53), (0x4b542134), + (0x4b2b5c18), (0x4b02eeb1), (0x4adad7b8), (0x4ab315ea), (0x4a8ba80d), + (0x4a648cec), (0x4a3dc35b), (0x4a174a30), (0x49f1204a), (0x49cb448d), + (0x49a5b5e2), (0x49807339), (0x495b7b86), (0x4936cdc2), (0x491268ec), + (0x48ee4c08), (0x48ca761f), (0x48a6e63e), (0x48839b76), (0x486094de), + (0x483dd190), (0x481b50ad), (0x47f91156), (0x47d712b3), (0x47b553f0), + (0x4793d43c), (0x477292c9), (0x47518ece), (0x4730c785), (0x47103c2d), + (0x46efec06), (0x46cfd655), (0x46affa61), (0x46905777), (0x4670ece4), + (0x4651b9f9), (0x4632be0b), (0x4613f871), (0x45f56885), (0x45d70da5), + (0x45b8e72f), (0x459af487), (0x457d3511), (0x455fa835), (0x45424d5d), + (0x452523f6), (0x45082b6e), (0x44eb6337), (0x44cecac5), (0x44b2618d), + (0x44962708), (0x447a1ab1), (0x445e3c02), (0x44428a7c), (0x4427059e), + (0x440bacec), (0x43f07fe9), (0x43d57e1c), (0x43baa70e), (0x439ffa48), + (0x43857757), (0x436b1dc8), (0x4350ed2b), (0x4336e511), (0x431d050c), + (0x43034cb2), (0x42e9bb98), (0x42d05156), (0x42b70d85), (0x429defc0), + (0x4284f7a2), (0x426c24cb), (0x425376d8), (0x423aed6a), (0x42228823), + (0x420a46a6), (0x41f22898), (0x41da2d9f), (0x41c25561), (0x41aa9f86), + (0x41930bba), (0x417b99a5), (0x416448f5), (0x414d1956), (0x41360a76), + (0x411f1c06), (0x41084db5), (0x40f19f35), (0x40db1039), (0x40c4a074), + (0x40ae4f9b), (0x40981d64), (0x40820985), (0x406c13b6), (0x40563bb1), + (0x4040812e), (0x402ae3e7), (0x40156399), (0x40000000), (0x3FEAB8D9)}; +/* n.a. */ +static const FIXP_DBL invCubeRootCorrection[3] = {0x40000000, 0x50A28BE6, + 0x6597FA95}; + +/***************************************************************************** + * \brief calculate 1.0/cube_root(op), op contains mantissa and exponent + * \param op_m: (i) mantissa of operand, must not be zero (0x0000.0000) or + * negative + * \param op_e: (i) pointer to the exponent of the operand (must be initialized) + * and .. (o) pointer to the exponent of the result + * \return: (o) mantissa of the result + * \description: + * This routine calculates the cube root of the input operand, that is + * given with its mantissa in Q31 format (FIXP_DBL) and its exponent (INT). + * The resulting mantissa is returned in format Q31. The exponent (*op_e) + * is modified accordingly. It is not assured, that the result is fully + * left-aligned but assumed to have not more than 2 bits headroom. There is one + * macro to activate the use of this algorithm: FUNCTION_invCubeRootNorm2 By + * means of activating the macro INVCUBEROOTNORM2_LINEAR_INTERPOLATE_HQ, a + * slightly higher precision is reachable (by default, not active). For DEBUG + * purpose only: a FDK_ASSERT macro validates, if the input mantissa is greater + * zero. + * + */ +static +#ifdef __arm__ + FIXP_DBL FDK_FORCEINLINE + invCubeRootNorm2(FIXP_DBL op_m, INT* op_e) +#else + FIXP_DBL + invCubeRootNorm2(FIXP_DBL op_m, INT* op_e) +#endif +{ + FDK_ASSERT(op_m > FIXP_DBL(0)); + + /* normalize input, calculate shift value */ + INT exponent = (INT)fNormz(op_m) - 1; + op_m <<= exponent; + + INT index = (INT)(op_m >> (DFRACT_BITS - 1 - (CUBE_ROOT_BITS + 1))) & + CUBE_ROOT_BITS_MASK; + FIXP_DBL fract = (FIXP_DBL)(((INT)op_m & CUBE_ROOT_FRACT_BITS_MASK) + << (CUBE_ROOT_BITS + 1)); + FIXP_DBL diff = invCubeRootTab[index + 1] - invCubeRootTab[index]; + op_m = fMultAddDiv2(invCubeRootTab[index], diff << 1, fract); +#if defined(INVCUBEROOTNORM2_LINEAR_INTERPOLATE_HQ) + /* reg1 = t[i] + (t[i+1]-t[i])*fract ... already computed ... + + * (1-fract)fract*(t[i+2]-t[i+1])/2 */ + if (fract != (FIXP_DBL)0) { + /* fract = fract * (1 - fract) */ + fract = fMultDiv2(fract, (FIXP_DBL)((LONG)0x80000000 - (LONG)fract)) << 1; + diff = diff - (invCubeRootTab[index + 2] - invCubeRootTab[index + 1]); + op_m = fMultAddDiv2(op_m, fract, diff); + } +#endif /* INVCUBEROOTNORM2_LINEAR_INTERPOLATE_HQ */ + + /* calculate the output exponent = input * exp/3 = cubicroot(m)*2^(exp/3) + * where 2^(exp/3) = 2^k'*2 or 2^k'*2^(1/3) or 2^k'*2^(2/3) */ + exponent = exponent - *op_e + 3; + INT shift_tmp = + ((INT)fMultDiv2((FIXP_SGL)fAbs(exponent), (FIXP_SGL)0x5556)) >> 16; + if (exponent < 0) { + shift_tmp = -shift_tmp; + } + INT rem = exponent - 3 * shift_tmp; + if (rem < 0) { + rem += 3; + shift_tmp--; + } + + *op_e = shift_tmp; + op_m = fMultDiv2(op_m, invCubeRootCorrection[rem]) << 2; + + return (op_m); +} + + /***************************************************************************** + + functionname: invFourthRootNorm2 + description: delivers 1/FourthRoot(op) in Q1.31 format and modified + exponent + + *****************************************************************************/ + +#define FOURTHROOT_BITS 7 +#define FOURTHROOT_VALUES (128 + 2) +#define FOURTHROOT_BITS_MASK 0x7f +#define FOURTHROOT_FRACT_BITS_MASK 0x007FFFFF + +LNK_SECTION_CONSTDATA +static const FIXP_DBL invFourthRootTab[FOURTHROOT_VALUES] = { + (0x4c1bf829), (0x4bf61977), (0x4bd09843), (0x4bab72ef), (0x4b86a7eb), + (0x4b6235ac), (0x4b3e1ab6), (0x4b1a5592), (0x4af6e4d4), (0x4ad3c718), + (0x4ab0fb03), (0x4a8e7f42), (0x4a6c5288), (0x4a4a7393), (0x4a28e126), + (0x4a079a0c), (0x49e69d16), (0x49c5e91f), (0x49a57d04), (0x498557ac), + (0x49657802), (0x4945dcf9), (0x49268588), (0x490770ac), (0x48e89d6a), + (0x48ca0ac9), (0x48abb7d6), (0x488da3a6), (0x486fcd4f), (0x485233ed), + (0x4834d6a3), (0x4817b496), (0x47faccf0), (0x47de1ee0), (0x47c1a999), + (0x47a56c51), (0x47896643), (0x476d96af), (0x4751fcd6), (0x473697ff), + (0x471b6773), (0x47006a81), (0x46e5a079), (0x46cb08ae), (0x46b0a279), + (0x46966d34), (0x467c683d), (0x466292f4), (0x4648ecbc), (0x462f74fe), + (0x46162b20), (0x45fd0e91), (0x45e41ebe), (0x45cb5b19), (0x45b2c315), + (0x459a562a), (0x458213cf), (0x4569fb81), (0x45520cbc), (0x453a4701), + (0x4522a9d1), (0x450b34b0), (0x44f3e726), (0x44dcc0ba), (0x44c5c0f7), + (0x44aee768), (0x4498339e), (0x4481a527), (0x446b3b96), (0x4454f67e), + (0x443ed576), (0x4428d815), (0x4412fdf3), (0x43fd46ad), (0x43e7b1de), + (0x43d23f23), (0x43bcee1e), (0x43a7be6f), (0x4392afb8), (0x437dc19d), + (0x4368f3c5), (0x435445d6), (0x433fb779), (0x432b4856), (0x4316f81a), + (0x4302c66f), (0x42eeb305), (0x42dabd8a), (0x42c6e5ad), (0x42b32b21), + (0x429f8d96), (0x428c0cc2), (0x4278a859), (0x42656010), (0x4252339e), + (0x423f22bc), (0x422c2d23), (0x4219528b), (0x420692b2), (0x41f3ed51), + (0x41e16228), (0x41cef0f2), (0x41bc9971), (0x41aa5b62), (0x41983687), + (0x41862aa2), (0x41743775), (0x41625cc3), (0x41509a50), (0x413eefe2), + (0x412d5d3e), (0x411be22b), (0x410a7e70), (0x40f931d5), (0x40e7fc23), + (0x40d6dd24), (0x40c5d4a2), (0x40b4e268), (0x40a40642), (0x40933ffc), + (0x40828f64), (0x4071f447), (0x40616e73), (0x4050fdb9), (0x4040a1e6), + (0x40305acc), (0x4020283c), (0x40100a08), (0x40000000), (0x3ff009f9), +}; + +static const FIXP_DBL invFourthRootCorrection[4] = {0x40000000, 0x4C1BF829, + 0x5A82799A, 0x6BA27E65}; + +/* The fourth root function */ +/***************************************************************************** + * \brief calculate 1.0/fourth_root(op), op contains mantissa and exponent + * \param op_m: (i) mantissa of operand, must not be zero (0x0000.0000) or + * negative + * \param op_e: (i) pointer to the exponent of the operand (must be initialized) + * and .. (o) pointer to the exponent of the result + * \return: (o) mantissa of the result + * \description: + * This routine calculates the cube root of the input operand, that is + * given with its mantissa in Q31 format (FIXP_DBL) and its exponent (INT). + * The resulting mantissa is returned in format Q31. The exponent (*op_e) + * is modified accordingly. It is not assured, that the result is fully + * left-aligned but assumed to have not more than 2 bits headroom. There is one + * macro to activate the use of this algorithm: FUNCTION_invFourthRootNorm2 By + * means of activating the macro INVFOURTHROOTNORM2_LINEAR_INTERPOLATE_HQ, a + * slightly higher precision is reachable (by default, not active). For DEBUG + * purpose only: a FDK_ASSERT macro validates, if the input mantissa is greater + * zero. + * + */ + +/* #define INVFOURTHROOTNORM2_LINEAR_INTERPOLATE_HQ */ + +static +#ifdef __arm__ + FIXP_DBL FDK_FORCEINLINE + invFourthRootNorm2(FIXP_DBL op_m, INT* op_e) +#else + FIXP_DBL + invFourthRootNorm2(FIXP_DBL op_m, INT* op_e) +#endif +{ + FDK_ASSERT(op_m > FL2FXCONST_DBL(0.0)); + + /* normalize input, calculate shift value */ + INT exponent = (INT)fNormz(op_m) - 1; + op_m <<= exponent; + + INT index = (INT)(op_m >> (DFRACT_BITS - 1 - (FOURTHROOT_BITS + 1))) & + FOURTHROOT_BITS_MASK; + FIXP_DBL fract = (FIXP_DBL)(((INT)op_m & FOURTHROOT_FRACT_BITS_MASK) + << (FOURTHROOT_BITS + 1)); + FIXP_DBL diff = invFourthRootTab[index + 1] - invFourthRootTab[index]; + op_m = invFourthRootTab[index] + (fMultDiv2(diff, fract) << 1); + +#if defined(INVFOURTHROOTNORM2_LINEAR_INTERPOLATE_HQ) + /* reg1 = t[i] + (t[i+1]-t[i])*fract ... already computed ... + + * (1-fract)fract*(t[i+2]-t[i+1])/2 */ + if (fract != (FIXP_DBL)0) { + /* fract = fract * (1 - fract) */ + fract = fMultDiv2(fract, (FIXP_DBL)((LONG)0x80000000 - (LONG)fract)) << 1; + diff = diff - (invFourthRootTab[index + 2] - invFourthRootTab[index + 1]); + op_m = fMultAddDiv2(op_m, fract, diff); + } +#endif /* INVFOURTHROOTNORM2_LINEAR_INTERPOLATE_HQ */ + + exponent = exponent - *op_e + 4; + INT rem = exponent & 0x00000003; + INT shift_tmp = (exponent >> 2); + + *op_e = shift_tmp; + op_m = fMultDiv2(op_m, invFourthRootCorrection[rem]) << 2; + + return (op_m); +} + +/***************************************************************************** + + functionname: inv3EigthRootNorm2 + description: delivers 1/cubert(op) normalized to .5...1 and the shift value +of the OUTPUT + +*****************************************************************************/ +#define THREEIGTHROOT_BITS 7 +#define THREEIGTHROOT_VALUES (128 + 2) +#define THREEIGTHROOT_BITS_MASK 0x7f +#define THREEIGTHROOT_FRACT_BITS_MASK 0x007FFFFF + +LNK_SECTION_CONSTDATA +static const FIXP_DBL inv3EigthRootTab[THREEIGTHROOT_VALUES] = { + (0x45cae0f2), (0x45b981bf), (0x45a8492a), (0x45973691), (0x45864959), + (0x457580e6), (0x4564dca4), (0x45545c00), (0x4543fe6b), (0x4533c35a), + (0x4523aa44), (0x4513b2a4), (0x4503dbf7), (0x44f425be), (0x44e48f7b), + (0x44d518b6), (0x44c5c0f7), (0x44b687c8), (0x44a76cb8), (0x44986f58), + (0x44898f38), (0x447acbef), (0x446c2514), (0x445d9a3f), (0x444f2b0d), + (0x4440d71a), (0x44329e07), (0x44247f73), (0x44167b04), (0x4408905e), + (0x43fabf28), (0x43ed070b), (0x43df67b0), (0x43d1e0c5), (0x43c471f7), + (0x43b71af6), (0x43a9db71), (0x439cb31c), (0x438fa1ab), (0x4382a6d2), + (0x4375c248), (0x4368f3c5), (0x435c3b03), (0x434f97bc), (0x434309ac), + (0x43369091), (0x432a2c28), (0x431ddc30), (0x4311a06c), (0x4305789c), + (0x42f96483), (0x42ed63e5), (0x42e17688), (0x42d59c30), (0x42c9d4a6), + (0x42be1fb1), (0x42b27d1a), (0x42a6ecac), (0x429b6e2f), (0x42900172), + (0x4284a63f), (0x42795c64), (0x426e23b0), (0x4262fbf2), (0x4257e4f9), + (0x424cde96), (0x4241e89a), (0x423702d8), (0x422c2d23), (0x4221674d), + (0x4216b12c), (0x420c0a94), (0x4201735b), (0x41f6eb57), (0x41ec725f), + (0x41e2084b), (0x41d7acf3), (0x41cd6030), (0x41c321db), (0x41b8f1ce), + (0x41aecfe5), (0x41a4bbf8), (0x419ab5e6), (0x4190bd89), (0x4186d2bf), + (0x417cf565), (0x41732558), (0x41696277), (0x415faca1), (0x415603b4), + (0x414c6792), (0x4142d818), (0x4139552a), (0x412fdea6), (0x41267470), + (0x411d1668), (0x4113c472), (0x410a7e70), (0x41014445), (0x40f815d4), + (0x40eef302), (0x40e5dbb4), (0x40dccfcd), (0x40d3cf33), (0x40cad9cb), + (0x40c1ef7b), (0x40b9102a), (0x40b03bbd), (0x40a7721c), (0x409eb32e), + (0x4095feda), (0x408d5508), (0x4084b5a0), (0x407c208b), (0x407395b2), + (0x406b14fd), (0x40629e56), (0x405a31a6), (0x4051ced8), (0x404975d5), + (0x40412689), (0x4038e0dd), (0x4030a4bd), (0x40287215), (0x402048cf), + (0x401828d7), (0x4010121a), (0x40080483), (0x40000000), (0x3ff8047d), +}; + +/* The last value is rounded in order to avoid any overflow due to the values + * range of the root table */ +static const FIXP_DBL inv3EigthRootCorrection[8] = { + 0x40000000, 0x45CAE0F2, 0x4C1BF829, 0x52FF6B55, + 0x5A82799A, 0x62B39509, 0x6BA27E65, 0x75606373}; + +/* The 3/8 root function */ +/***************************************************************************** + * \brief calculate 1.0/3Eigth_root(op) = 1.0/(x)^(3/8), op contains mantissa + * and exponent + * \param op_m: (i) mantissa of operand, must not be zero (0x0000.0000) or + * negative + * \param op_e: (i) pointer to the exponent of the operand (must be initialized) + * and .. (o) pointer to the exponent of the result + * \return: (o) mantissa of the result + * \description: + * This routine calculates the cube root of the input operand, that is + * given with its mantissa in Q31 format (FIXP_DBL) and its exponent (INT). + * The resulting mantissa is returned in format Q31. The exponent (*op_e) + * is modified accordingly. It is not assured, that the result is fully + * left-aligned but assumed to have not more than 2 bits headroom. There is one + * macro to activate the use of this algorithm: FUNCTION_inv3EigthRootNorm2 By + * means of activating the macro INVTHREEIGTHROOTNORM2_LINEAR_INTERPOLATE_HQ, a + * slightly higher precision is reachable (by default, not active). For DEBUG + * purpose only: a FDK_ASSERT macro validates, if the input mantissa is greater + * zero. + * + */ + +/* #define INVTHREEIGTHROOTNORM2_LINEAR_INTERPOLATE_HQ */ + +static +#ifdef __arm__ + FIXP_DBL FDK_FORCEINLINE + inv3EigthRootNorm2(FIXP_DBL op_m, INT* op_e) +#else + FIXP_DBL + inv3EigthRootNorm2(FIXP_DBL op_m, INT* op_e) +#endif +{ + FDK_ASSERT(op_m > FL2FXCONST_DBL(0.0)); + + /* normalize input, calculate shift op_mue */ + INT exponent = (INT)fNormz(op_m) - 1; + op_m <<= exponent; + + INT index = (INT)(op_m >> (DFRACT_BITS - 1 - (THREEIGTHROOT_BITS + 1))) & + THREEIGTHROOT_BITS_MASK; + FIXP_DBL fract = (FIXP_DBL)(((INT)op_m & THREEIGTHROOT_FRACT_BITS_MASK) + << (THREEIGTHROOT_BITS + 1)); + FIXP_DBL diff = inv3EigthRootTab[index + 1] - inv3EigthRootTab[index]; + op_m = inv3EigthRootTab[index] + (fMultDiv2(diff, fract) << 1); + +#if defined(INVTHREEIGTHROOTNORM2_LINEAR_INTERPOLATE_HQ) + /* op_m = t[i] + (t[i+1]-t[i])*fract ... already computed ... + + * (1-fract)fract*(t[i+2]-t[i+1])/2 */ + if (fract != (FIXP_DBL)0) { + /* fract = fract * (1 - fract) */ + fract = fMultDiv2(fract, (FIXP_DBL)((LONG)0x80000000 - (LONG)fract)) << 1; + diff = diff - (inv3EigthRootTab[index + 2] - inv3EigthRootTab[index + 1]); + op_m = fMultAddDiv2(op_m, fract, diff); + } +#endif /* INVTHREEIGTHROOTNORM2_LINEAR_INTERPOLATE_HQ */ + + exponent = exponent - *op_e + 8; + INT rem = exponent & 0x00000007; + INT shift_tmp = (exponent >> 3); + + *op_e = shift_tmp * 3; + op_m = fMultDiv2(op_m, inv3EigthRootCorrection[rem]) << 2; + + return (fMult(op_m, fMult(op_m, op_m))); +} + +SBR_ERROR +QmfTransposerCreate(HANDLE_HBE_TRANSPOSER* hQmfTransposer, const int frameSize, + int bDisableCrossProducts, int bSbr41) { + HANDLE_HBE_TRANSPOSER hQmfTran = NULL; + + int i; + + if (hQmfTransposer != NULL) { + /* Memory allocation */ + /*--------------------------------------------------------------------------------------------*/ + hQmfTran = + (HANDLE_HBE_TRANSPOSER)FDKcalloc(1, sizeof(struct hbeTransposer)); + if (hQmfTran == NULL) { + return SBRDEC_MEM_ALLOC_FAILED; + } + + for (i = 0; i < MAX_STRETCH_HBE - 1; i++) { + hQmfTran->bXProducts[i] = (bDisableCrossProducts ? 0 : xProducts[i]); + } + + hQmfTran->timeDomainWinLen = frameSize; + if (frameSize == 768) { + hQmfTran->noCols = + (8 * frameSize / 3) / QMF_SYNTH_CHANNELS; /* 32 for 24:64 */ + } else { + hQmfTran->noCols = + (bSbr41 + 1) * 2 * frameSize / + QMF_SYNTH_CHANNELS; /* 32 for 32:64 and 64 for 16:64 -> identical to + sbrdec->no_cols */ + } + + hQmfTran->noChannels = frameSize / hQmfTran->noCols; + + hQmfTran->qmfInBufSize = QMF_WIN_LEN; + hQmfTran->qmfOutBufSize = 2 * (hQmfTran->noCols / 2 + QMF_WIN_LEN - 1); + + hQmfTran->inBuf_F = + (INT_PCM*)FDKcalloc(QMF_SYNTH_CHANNELS + 20 + 1, sizeof(INT_PCM)); + /* buffered time signal needs to be delayed by synthesis_size; max + * synthesis_size = 20; */ + if (hQmfTran->inBuf_F == NULL) { + QmfTransposerClose(hQmfTran); + return SBRDEC_MEM_ALLOC_FAILED; + } + + hQmfTran->qmfInBufReal_F = + (FIXP_DBL**)FDKcalloc(hQmfTran->qmfInBufSize, sizeof(FIXP_DBL*)); + hQmfTran->qmfInBufImag_F = + (FIXP_DBL**)FDKcalloc(hQmfTran->qmfInBufSize, sizeof(FIXP_DBL*)); + + if (hQmfTran->qmfInBufReal_F == NULL) { + QmfTransposerClose(hQmfTran); + return SBRDEC_MEM_ALLOC_FAILED; + } + if (hQmfTran->qmfInBufImag_F == NULL) { + QmfTransposerClose(hQmfTran); + return SBRDEC_MEM_ALLOC_FAILED; + } + + for (i = 0; i < hQmfTran->qmfInBufSize; i++) { + hQmfTran->qmfInBufReal_F[i] = (FIXP_DBL*)FDKaalloc( + QMF_SYNTH_CHANNELS * sizeof(FIXP_DBL), ALIGNMENT_DEFAULT); + hQmfTran->qmfInBufImag_F[i] = (FIXP_DBL*)FDKaalloc( + QMF_SYNTH_CHANNELS * sizeof(FIXP_DBL), ALIGNMENT_DEFAULT); + if (hQmfTran->qmfInBufReal_F[i] == NULL) { + QmfTransposerClose(hQmfTran); + return SBRDEC_MEM_ALLOC_FAILED; + } + if (hQmfTran->qmfInBufImag_F[i] == NULL) { + QmfTransposerClose(hQmfTran); + return SBRDEC_MEM_ALLOC_FAILED; + } + } + + hQmfTran->qmfHBEBufReal_F = + (FIXP_DBL**)FDKcalloc(HBE_MAX_OUT_SLOTS, sizeof(FIXP_DBL*)); + hQmfTran->qmfHBEBufImag_F = + (FIXP_DBL**)FDKcalloc(HBE_MAX_OUT_SLOTS, sizeof(FIXP_DBL*)); + + if (hQmfTran->qmfHBEBufReal_F == NULL) { + QmfTransposerClose(hQmfTran); + return SBRDEC_MEM_ALLOC_FAILED; + } + if (hQmfTran->qmfHBEBufImag_F == NULL) { + QmfTransposerClose(hQmfTran); + return SBRDEC_MEM_ALLOC_FAILED; + } + + for (i = 0; i < HBE_MAX_OUT_SLOTS; i++) { + hQmfTran->qmfHBEBufReal_F[i] = + (FIXP_DBL*)FDKcalloc(QMF_SYNTH_CHANNELS, sizeof(FIXP_DBL)); + hQmfTran->qmfHBEBufImag_F[i] = + (FIXP_DBL*)FDKcalloc(QMF_SYNTH_CHANNELS, sizeof(FIXP_DBL)); + if (hQmfTran->qmfHBEBufReal_F[i] == NULL) { + QmfTransposerClose(hQmfTran); + return SBRDEC_MEM_ALLOC_FAILED; + } + if (hQmfTran->qmfHBEBufImag_F[i] == NULL) { + QmfTransposerClose(hQmfTran); + return SBRDEC_MEM_ALLOC_FAILED; + } + } + + hQmfTran->qmfBufferCodecTempSlot_F = + (FIXP_DBL*)FDKcalloc(QMF_SYNTH_CHANNELS / 2, sizeof(FIXP_DBL)); + if (hQmfTran->qmfBufferCodecTempSlot_F == NULL) { + QmfTransposerClose(hQmfTran); + return SBRDEC_MEM_ALLOC_FAILED; + } + + hQmfTran->bSbr41 = bSbr41; + + hQmfTran->highband_exp[0] = 0; + hQmfTran->highband_exp[1] = 0; + hQmfTran->target_exp[0] = 0; + hQmfTran->target_exp[1] = 0; + + *hQmfTransposer = hQmfTran; + } + + return SBRDEC_OK; +} + +SBR_ERROR QmfTransposerReInit(HANDLE_HBE_TRANSPOSER hQmfTransposer, + UCHAR* FreqBandTable[2], UCHAR NSfb[2]) +/* removed bSbr41 from parameterlist: + don't know where to get this value from + at call-side */ +{ + int L, sfb, patch, stopPatch, qmfErr; + + if (hQmfTransposer != NULL) { + const FIXP_QTW* tmp_t_cos; + const FIXP_QTW* tmp_t_sin; + + hQmfTransposer->startBand = FreqBandTable[0][0]; + FDK_ASSERT((!hQmfTransposer->bSbr41 && hQmfTransposer->startBand <= 32) || + (hQmfTransposer->bSbr41 && + hQmfTransposer->startBand <= + 16)); /* is checked by resetFreqBandTables() */ + hQmfTransposer->stopBand = FreqBandTable[0][NSfb[0]]; + + hQmfTransposer->synthSize = + 4 * ((hQmfTransposer->startBand + 4) / 8 + 1); /* 8, 12, 16, 20 */ + hQmfTransposer->kstart = startSubband2kL[hQmfTransposer->startBand]; + + /* don't know where to take this information from */ + /* hQmfTransposer->bSbr41 = bSbr41; */ + + if (hQmfTransposer->bSbr41) { + if (hQmfTransposer->kstart + hQmfTransposer->synthSize > 16) + hQmfTransposer->kstart = 16 - hQmfTransposer->synthSize; + } else if (hQmfTransposer->timeDomainWinLen == 768) { + if (hQmfTransposer->kstart + hQmfTransposer->synthSize > 24) + hQmfTransposer->kstart = 24 - hQmfTransposer->synthSize; + } + + hQmfTransposer->synthesisQmfPreModCos_F = + &preModCos[hQmfTransposer->kstart]; + hQmfTransposer->synthesisQmfPreModSin_F = + &preModSin[hQmfTransposer->kstart]; + + L = 2 * hQmfTransposer->synthSize; /* 8, 16, 24, 32, 40 */ + /* Change analysis post twiddles */ + + switch (L) { + case 8: + tmp_t_cos = post_twiddle_cos_8; + tmp_t_sin = post_twiddle_sin_8; + break; + case 16: + tmp_t_cos = post_twiddle_cos_16; + tmp_t_sin = post_twiddle_sin_16; + break; + case 24: + tmp_t_cos = post_twiddle_cos_24; + tmp_t_sin = post_twiddle_sin_24; + break; + case 32: + tmp_t_cos = post_twiddle_cos_32; + tmp_t_sin = post_twiddle_sin_32; + break; + case 40: + tmp_t_cos = post_twiddle_cos_40; + tmp_t_sin = post_twiddle_sin_40; + break; + default: + return SBRDEC_UNSUPPORTED_CONFIG; + } + + qmfErr = qmfInitSynthesisFilterBank( + &hQmfTransposer->HBESynthesisQMF, hQmfTransposer->synQmfStates, + hQmfTransposer->noCols, 0, hQmfTransposer->synthSize, + hQmfTransposer->synthSize, 1); + if (qmfErr != 0) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + + qmfErr = qmfInitAnalysisFilterBank( + &hQmfTransposer->HBEAnalysiscQMF, hQmfTransposer->anaQmfStates, + hQmfTransposer->noCols / 2, 0, 2 * hQmfTransposer->synthSize, + 2 * hQmfTransposer->synthSize, 0); + + if (qmfErr != 0) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + + hQmfTransposer->HBEAnalysiscQMF.t_cos = tmp_t_cos; + hQmfTransposer->HBEAnalysiscQMF.t_sin = tmp_t_sin; + + FDKmemset(hQmfTransposer->xOverQmf, 0, + MAX_NUM_PATCHES * sizeof(int)); /* global */ + sfb = 0; + if (hQmfTransposer->bSbr41) { + stopPatch = MAX_NUM_PATCHES; + hQmfTransposer->maxStretch = MAX_STRETCH_HBE; + } else { + stopPatch = MAX_STRETCH_HBE; + } + + for (patch = 1; patch <= stopPatch; patch++) { + while (sfb <= NSfb[0] && + FreqBandTable[0][sfb] <= patch * hQmfTransposer->startBand) + sfb++; + if (sfb <= NSfb[0]) { + /* If the distance is larger than three QMF bands - try aligning to high + * resolution frequency bands instead. */ + if ((patch * hQmfTransposer->startBand - FreqBandTable[0][sfb - 1]) <= + 3) { + hQmfTransposer->xOverQmf[patch - 1] = FreqBandTable[0][sfb - 1]; + } else { + int sfb_tmp = 0; + while (sfb_tmp <= NSfb[1] && + FreqBandTable[1][sfb_tmp] <= patch * hQmfTransposer->startBand) + sfb_tmp++; + hQmfTransposer->xOverQmf[patch - 1] = FreqBandTable[1][sfb_tmp - 1]; + } + } else { + hQmfTransposer->xOverQmf[patch - 1] = hQmfTransposer->stopBand; + hQmfTransposer->maxStretch = fMin(patch, MAX_STRETCH_HBE); + break; + } + } + + hQmfTransposer->highband_exp[0] = 0; + hQmfTransposer->highband_exp[1] = 0; + hQmfTransposer->target_exp[0] = 0; + hQmfTransposer->target_exp[1] = 0; + } + + return SBRDEC_OK; +} + +void QmfTransposerClose(HANDLE_HBE_TRANSPOSER hQmfTransposer) { + int i; + + if (hQmfTransposer != NULL) { + if (hQmfTransposer->inBuf_F) FDKfree(hQmfTransposer->inBuf_F); + + if (hQmfTransposer->qmfInBufReal_F) { + for (i = 0; i < hQmfTransposer->qmfInBufSize; i++) { + FDKafree(hQmfTransposer->qmfInBufReal_F[i]); + } + FDKfree(hQmfTransposer->qmfInBufReal_F); + } + + if (hQmfTransposer->qmfInBufImag_F) { + for (i = 0; i < hQmfTransposer->qmfInBufSize; i++) { + FDKafree(hQmfTransposer->qmfInBufImag_F[i]); + } + FDKfree(hQmfTransposer->qmfInBufImag_F); + } + + if (hQmfTransposer->qmfHBEBufReal_F) { + for (i = 0; i < HBE_MAX_OUT_SLOTS; i++) { + FDKfree(hQmfTransposer->qmfHBEBufReal_F[i]); + } + FDKfree(hQmfTransposer->qmfHBEBufReal_F); + } + + if (hQmfTransposer->qmfHBEBufImag_F) { + for (i = 0; i < HBE_MAX_OUT_SLOTS; i++) { + FDKfree(hQmfTransposer->qmfHBEBufImag_F[i]); + } + FDKfree(hQmfTransposer->qmfHBEBufImag_F); + } + + FDKfree(hQmfTransposer->qmfBufferCodecTempSlot_F); + + FDKfree(hQmfTransposer); + } +} + +inline void scaleUp(FIXP_DBL* real_m, FIXP_DBL* imag_m, INT* _e) { + INT reserve; + /* shift gc_r and gc_i up if possible */ + reserve = CntLeadingZeros((INT(*real_m) ^ INT((*real_m >> 31))) | + (INT(*imag_m) ^ INT((*imag_m >> 31)))) - + 1; + reserve = fMax(reserve - 1, + 0); /* Leave one bit headroom such that (real_m^2 + imag_m^2) + does not overflow later if both are 0x80000000. */ + reserve = fMin(reserve, *_e); + FDK_ASSERT(reserve >= 0); + *real_m <<= reserve; + *imag_m <<= reserve; + *_e -= reserve; +} + +static void calculateCenterFIXP(FIXP_DBL gammaVecReal, FIXP_DBL gammaVecImag, + FIXP_DBL* centerReal, FIXP_DBL* centerImag, + INT* exponent, int stretch, int mult) { + scaleUp(&gammaVecReal, &gammaVecImag, exponent); + FIXP_DBL energy = fPow2Div2(gammaVecReal) + fPow2Div2(gammaVecImag); + + if (energy != FL2FXCONST_DBL(0.f)) { + FIXP_DBL gc_r_m, gc_i_m, factor_m = (FIXP_DBL)0; + INT factor_e, gc_e; + factor_e = 2 * (*exponent) + 1; + + switch (stretch) { + case 2: + factor_m = invFourthRootNorm2(energy, &factor_e); + break; + case 3: + factor_m = invCubeRootNorm2(energy, &factor_e); + break; + case 4: + factor_m = inv3EigthRootNorm2(energy, &factor_e); + break; + } + + gc_r_m = fMultDiv2(gammaVecReal, + factor_m); /* exponent = HBE_SCALE + factor_e + 1 */ + gc_i_m = fMultDiv2(gammaVecImag, + factor_m); /* exponent = HBE_SCALE + factor_e + 1*/ + gc_e = *exponent + factor_e + 1; + + scaleUp(&gc_r_m, &gc_i_m, &gc_e); + + switch (mult) { + case 0: + *centerReal = gc_r_m; + *centerImag = gc_i_m; + break; + case 1: + *centerReal = fPow2Div2(gc_r_m) - fPow2Div2(gc_i_m); + *centerImag = fMult(gc_r_m, gc_i_m); + gc_e = 2 * gc_e + 1; + break; + case 2: + FIXP_DBL tmp_r = gc_r_m; + FIXP_DBL tmp_i = gc_i_m; + gc_r_m = fPow2Div2(gc_r_m) - fPow2Div2(gc_i_m); + gc_i_m = fMult(tmp_r, gc_i_m); + gc_e = 3 * gc_e + 1 + 1; + cplxMultDiv2(¢erReal[0], ¢erImag[0], gc_r_m, gc_i_m, tmp_r, + tmp_i); + break; + } + + scaleUp(centerReal, centerImag, &gc_e); + + FDK_ASSERT(gc_e >= 0); + *exponent = gc_e; + } else { + *centerReal = energy; /* energy = 0 */ + *centerImag = energy; /* energy = 0 */ + *exponent = (INT)energy; + } +} + +static int getHBEScaleFactorFrame(const int bSbr41, const int maxStretch, + const int pitchInBins) { + if (pitchInBins >= pmin * (1 + bSbr41)) { + /* crossproducts enabled */ + return 26; + } else { + return (maxStretch == 2) ? 24 : 25; + } +} + +static void addHighBandPart(FIXP_DBL g_r_m, FIXP_DBL g_i_m, INT g_e, + FIXP_DBL mult, FIXP_DBL gammaCenterReal_m, + FIXP_DBL gammaCenterImag_m, INT gammaCenter_e, + INT stretch, INT scale_factor_hbe, + FIXP_DBL* qmfHBEBufReal_F, + FIXP_DBL* qmfHBEBufImag_F) { + if ((g_r_m | g_i_m) != FL2FXCONST_DBL(0.f)) { + FIXP_DBL factor_m = (FIXP_DBL)0; + INT factor_e; + INT add = (stretch == 4) ? 1 : 0; + INT shift = (stretch == 4) ? 1 : 2; + + scaleUp(&g_r_m, &g_i_m, &g_e); + FIXP_DBL energy = fPow2AddDiv2(fPow2Div2(g_r_m), g_i_m); + factor_e = 2 * g_e + 1; + + switch (stretch) { + case 2: + factor_m = invFourthRootNorm2(energy, &factor_e); + break; + case 3: + factor_m = invCubeRootNorm2(energy, &factor_e); + break; + case 4: + factor_m = inv3EigthRootNorm2(energy, &factor_e); + break; + } + + factor_m = fMult(factor_m, mult); + + FIXP_DBL tmp_r, tmp_i; + cplxMultDiv2(&tmp_r, &tmp_i, g_r_m, g_i_m, gammaCenterReal_m, + gammaCenterImag_m); + + g_r_m = fMultDiv2(tmp_r, factor_m) << shift; + g_i_m = fMultDiv2(tmp_i, factor_m) << shift; + g_e = scale_factor_hbe - (g_e + factor_e + gammaCenter_e + add); + fMax((INT)0, g_e); + *qmfHBEBufReal_F += g_r_m >> g_e; + *qmfHBEBufImag_F += g_i_m >> g_e; + } +} + +void QmfTransposerApply(HANDLE_HBE_TRANSPOSER hQmfTransposer, + FIXP_DBL** qmfBufferCodecReal, + FIXP_DBL** qmfBufferCodecImag, int nColsIn, + FIXP_DBL** ppQmfBufferOutReal_F, + FIXP_DBL** ppQmfBufferOutImag_F, + FIXP_DBL lpcFilterStatesReal[2 + (3 * (4))][(64)], + FIXP_DBL lpcFilterStatesImag[2 + (3 * (4))][(64)], + int pitchInBins, int scale_lb, int scale_hbe, + int* scale_hb, int timeStep, int firstSlotOffsset, + int ov_len, + KEEP_STATES_SYNCED_MODE keepStatesSyncedMode) { + int i, j, stretch, band, sourceband, r, s; + int qmfVocoderColsIn = hQmfTransposer->noCols / 2; + int bSbr41 = hQmfTransposer->bSbr41; + + const int winLength[3] = {10, 8, 6}; + const int slotOffset = 6; /* hQmfTransposer->winLen-6; */ + + int qmfOffset = 2 * hQmfTransposer->kstart; + int scale_border = (nColsIn == 64) ? 32 : nColsIn; + + INT slot_stretch4[9] = {0, 0, 0, 0, 2, 4, 6, 8, 10}; + INT slot_stretch2[11] = {0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10}; + INT slot_stretch3[10] = {0, 0, 0, 1, 3, 4, 6, 7, 9, 10}; + INT filt_stretch3[10] = {0, 0, 0, 1, 0, 1, 0, 1, 0, 1}; + INT filt_dummy[11] = {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}; + INT* pSlotStretch; + INT* pFilt; + + int offset = 0; /* where to take QmfTransposer data */ + + int signPreMod = + (hQmfTransposer->synthesisQmfPreModCos_F[0] < FL2FXCONST_DBL(0.f)) ? 1 + : -1; + + int scale_factor_hbe = + getHBEScaleFactorFrame(bSbr41, hQmfTransposer->maxStretch, pitchInBins); + + if (keepStatesSyncedMode != KEEP_STATES_SYNCED_OFF) { + offset = hQmfTransposer->noCols - ov_len - LPC_ORDER; + } + + hQmfTransposer->highband_exp[0] = hQmfTransposer->highband_exp[1]; + hQmfTransposer->target_exp[0] = hQmfTransposer->target_exp[1]; + + hQmfTransposer->highband_exp[1] = scale_factor_hbe; + hQmfTransposer->target_exp[1] = + fixMax(hQmfTransposer->highband_exp[1], hQmfTransposer->highband_exp[0]); + + scale_factor_hbe = hQmfTransposer->target_exp[1]; + + int shift_ov = hQmfTransposer->target_exp[0] - hQmfTransposer->target_exp[1]; + + if (shift_ov != 0) { + for (i = 0; i < HBE_MAX_OUT_SLOTS; i++) { + for (band = 0; band < QMF_SYNTH_CHANNELS; band++) { + if (shift_ov >= 0) { + hQmfTransposer->qmfHBEBufReal_F[i][band] <<= shift_ov; + hQmfTransposer->qmfHBEBufImag_F[i][band] <<= shift_ov; + } else { + hQmfTransposer->qmfHBEBufReal_F[i][band] >>= (-shift_ov); + hQmfTransposer->qmfHBEBufImag_F[i][band] >>= (-shift_ov); + } + } + } + } + + if ((keepStatesSyncedMode == KEEP_STATES_SYNCED_OFF) && shift_ov != 0) { + for (i = timeStep * firstSlotOffsset; i < ov_len; i++) { + for (band = hQmfTransposer->startBand; band < hQmfTransposer->stopBand; + band++) { + if (shift_ov >= 0) { + ppQmfBufferOutReal_F[i][band] <<= shift_ov; + ppQmfBufferOutImag_F[i][band] <<= shift_ov; + } else { + ppQmfBufferOutReal_F[i][band] >>= (-shift_ov); + ppQmfBufferOutImag_F[i][band] >>= (-shift_ov); + } + } + } + + /* shift lpc filterstates */ + for (i = 0; i < timeStep * firstSlotOffsset + LPC_ORDER; i++) { + for (band = 0; band < (64); band++) { + if (shift_ov >= 0) { + lpcFilterStatesReal[i][band] <<= shift_ov; + lpcFilterStatesImag[i][band] <<= shift_ov; + } else { + lpcFilterStatesReal[i][band] >>= (-shift_ov); + lpcFilterStatesImag[i][band] >>= (-shift_ov); + } + } + } + } + + FIXP_DBL twid_m_new[3][2]; /* [stretch][cos/sin] */ + INT stepsize = 1 + !bSbr41, sine_offset = 24, mod = 96; + INT mult[3] = {1, 2, 3}; + + for (s = 0; s <= MAX_STRETCH_HBE - 2; s++) { + twid_m_new[s][0] = twiddle[(mult[s] * (stepsize * pitchInBins)) % mod]; + twid_m_new[s][1] = + twiddle[((mult[s] * (stepsize * pitchInBins)) + sine_offset) % mod]; + } + + /* Time-stretch */ + for (j = 0; j < qmfVocoderColsIn; j++) { + int sign = -1, k, z, addrshift, codecTemp_e; + /* update inbuf */ + for (i = 0; i < hQmfTransposer->synthSize; i++) { + hQmfTransposer->inBuf_F[i] = + hQmfTransposer->inBuf_F[i + 2 * hQmfTransposer->synthSize]; + } + + /* run synthesis for two sbr slots as transposer uses + half slots double bands representation */ + for (z = 0; z < 2; z++) { + int scale_factor = ((nColsIn == 64) && ((2 * j + z) < scale_border)) + ? scale_lb + : scale_hbe; + codecTemp_e = scale_factor - 1; /* -2 for Div2 and cos/sin scale of 1 */ + + for (k = 0; k < hQmfTransposer->synthSize; k++) { + int ki = hQmfTransposer->kstart + k; + hQmfTransposer->qmfBufferCodecTempSlot_F[k] = + fMultDiv2(signPreMod * hQmfTransposer->synthesisQmfPreModCos_F[k], + qmfBufferCodecReal[2 * j + z][ki]); + hQmfTransposer->qmfBufferCodecTempSlot_F[k] += + fMultDiv2(signPreMod * hQmfTransposer->synthesisQmfPreModSin_F[k], + qmfBufferCodecImag[2 * j + z][ki]); + } + + C_AALLOC_SCRATCH_START(pWorkBuffer, FIXP_DBL, (HBE_MAX_QMF_BANDS << 1)); + + qmfSynthesisFilteringSlot( + &hQmfTransposer->HBESynthesisQMF, + hQmfTransposer->qmfBufferCodecTempSlot_F, NULL, 0, + -7 - hQmfTransposer->HBESynthesisQMF.filterScale - codecTemp_e + 1, + hQmfTransposer->inBuf_F + hQmfTransposer->synthSize * (z + 1), 1, + pWorkBuffer); + + C_AALLOC_SCRATCH_END(pWorkBuffer, FIXP_DBL, (HBE_MAX_QMF_BANDS << 1)); + } + + C_AALLOC_SCRATCH_START(pWorkBuffer, FIXP_DBL, (HBE_MAX_QMF_BANDS << 1)); + + qmfAnalysisFilteringSlot(&hQmfTransposer->HBEAnalysiscQMF, + hQmfTransposer->qmfInBufReal_F[QMF_WIN_LEN - 1], + hQmfTransposer->qmfInBufImag_F[QMF_WIN_LEN - 1], + hQmfTransposer->inBuf_F + 1, 1, pWorkBuffer); + + C_AALLOC_SCRATCH_END(pWorkBuffer, FIXP_DBL, (HBE_MAX_QMF_BANDS << 1)); + + if ((keepStatesSyncedMode == KEEP_STATES_SYNCED_NORMAL) && + j <= qmfVocoderColsIn - ((LPC_ORDER + ov_len + QMF_WIN_LEN - 1) >> 1)) { + /* update in buffer */ + for (i = 0; i < QMF_WIN_LEN - 1; i++) { + FDKmemcpy( + hQmfTransposer->qmfInBufReal_F[i], + hQmfTransposer->qmfInBufReal_F[i + 1], + sizeof(FIXP_DBL) * hQmfTransposer->HBEAnalysiscQMF.no_channels); + FDKmemcpy( + hQmfTransposer->qmfInBufImag_F[i], + hQmfTransposer->qmfInBufImag_F[i + 1], + sizeof(FIXP_DBL) * hQmfTransposer->HBEAnalysiscQMF.no_channels); + } + continue; + } + + for (stretch = 2; stretch <= hQmfTransposer->maxStretch; stretch++) { + int start = slotOffset - winLength[stretch - 2] / 2; + int stop = slotOffset + winLength[stretch - 2] / 2; + + FIXP_DBL factor = FL2FXCONST_DBL(1.f / 3.f); + + for (band = hQmfTransposer->xOverQmf[stretch - 2]; + band < hQmfTransposer->xOverQmf[stretch - 1]; band++) { + FIXP_DBL gammaCenterReal_m[2] = {(FIXP_DBL)0, (FIXP_DBL)0}, + gammaCenterImag_m[2] = {(FIXP_DBL)0, (FIXP_DBL)0}; + INT gammaCenter_e[2] = {0, 0}; + + FIXP_DBL gammaVecReal_m[2] = {(FIXP_DBL)0, (FIXP_DBL)0}, + gammaVecImag_m[2] = {(FIXP_DBL)0, (FIXP_DBL)0}; + INT gammaVec_e[2] = {0, 0}; + + FIXP_DBL wingain = (FIXP_DBL)0; + + gammaCenter_e[0] = + SCALE2EXP(-hQmfTransposer->HBEAnalysiscQMF.outScalefactor); + gammaCenter_e[1] = + SCALE2EXP(-hQmfTransposer->HBEAnalysiscQMF.outScalefactor); + + /* interpolation filters for 3rd order */ + sourceband = 2 * band / stretch - qmfOffset; + FDK_ASSERT(sourceband >= 0); + + /* maximum gammaCenter_e == 20 */ + calculateCenterFIXP( + hQmfTransposer->qmfInBufReal_F[slotOffset][sourceband], + hQmfTransposer->qmfInBufImag_F[slotOffset][sourceband], + &gammaCenterReal_m[0], &gammaCenterImag_m[0], &gammaCenter_e[0], + stretch, stretch - 2); + + if (stretch == 4) { + r = band - 2 * (band / 2); + sourceband += (r == 0) ? -1 : 1; + pSlotStretch = slot_stretch4; + factor = FL2FXCONST_DBL(2.f / 3.f); + pFilt = filt_dummy; + } else if (stretch == 2) { + r = 0; + sourceband = 2 * band / stretch - qmfOffset; + pSlotStretch = slot_stretch2; + factor = FL2FXCONST_DBL(1.f / 3.f); + pFilt = filt_dummy; + } else { + r = 2 * band - 3 * (2 * band / 3); + sourceband = 2 * band / stretch - qmfOffset; + pSlotStretch = slot_stretch3; + factor = FL2FXCONST_DBL(1.4142f / 3.0f); + pFilt = filt_stretch3; + } + + if (r == 2) { + calculateCenterFIXP( + hQmfTransposer->qmfInBufReal_F[slotOffset][sourceband + 1], + hQmfTransposer->qmfInBufImag_F[slotOffset][sourceband + 1], + &gammaCenterReal_m[1], &gammaCenterImag_m[1], &gammaCenter_e[1], + stretch, stretch - 2); + + factor = FL2FXCONST_DBL(1.4142f / 6.0f); + } + + if (r == 2) { + for (k = start; k < stop; k++) { + gammaVecReal_m[0] = + hQmfTransposer->qmfInBufReal_F[pSlotStretch[k]][sourceband]; + gammaVecReal_m[1] = + hQmfTransposer->qmfInBufReal_F[pSlotStretch[k]][sourceband + 1]; + gammaVecImag_m[0] = + hQmfTransposer->qmfInBufImag_F[pSlotStretch[k]][sourceband]; + gammaVecImag_m[1] = + hQmfTransposer->qmfInBufImag_F[pSlotStretch[k]][sourceband + 1]; + gammaVec_e[0] = gammaVec_e[1] = + SCALE2EXP(-hQmfTransposer->HBEAnalysiscQMF.outScalefactor); + + if (pFilt[k] == 1) { + FIXP_DBL tmpRealF = gammaVecReal_m[0], tmpImagF; + gammaVecReal_m[0] = + (fMult(gammaVecReal_m[0], hintReal_F[sourceband % 4][1]) - + fMult(gammaVecImag_m[0], + hintReal_F[(sourceband + 3) % 4][1])) >> + 1; /* sum should be <= 1 because of sin/cos multiplication */ + gammaVecImag_m[0] = + (fMult(tmpRealF, hintReal_F[(sourceband + 3) % 4][1]) + + fMult(gammaVecImag_m[0], hintReal_F[sourceband % 4][1])) >> + 1; /* sum should be <= 1 because of sin/cos multiplication */ + + tmpRealF = hQmfTransposer + ->qmfInBufReal_F[pSlotStretch[k] + 1][sourceband]; + tmpImagF = hQmfTransposer + ->qmfInBufImag_F[pSlotStretch[k] + 1][sourceband]; + + gammaVecReal_m[0] += + (fMult(tmpRealF, hintReal_F[sourceband % 4][1]) - + fMult(tmpImagF, hintReal_F[(sourceband + 1) % 4][1])) >> + 1; /* sum should be <= 1 because of sin/cos multiplication */ + gammaVecImag_m[0] += + (fMult(tmpRealF, hintReal_F[(sourceband + 1) % 4][1]) + + fMult(tmpImagF, hintReal_F[sourceband % 4][1])) >> + 1; /* sum should be <= 1 because of sin/cos multiplication */ + gammaVec_e[0]++; + + tmpRealF = gammaVecReal_m[1]; + + gammaVecReal_m[1] = + (fMult(gammaVecReal_m[1], hintReal_F[sourceband % 4][2]) - + fMult(gammaVecImag_m[1], + hintReal_F[(sourceband + 3) % 4][2])) >> + 1; + gammaVecImag_m[1] = + (fMult(tmpRealF, hintReal_F[(sourceband + 3) % 4][2]) + + fMult(gammaVecImag_m[1], hintReal_F[sourceband % 4][2])) >> + 1; + + tmpRealF = + hQmfTransposer + ->qmfInBufReal_F[pSlotStretch[k] + 1][sourceband + 1]; + tmpImagF = + hQmfTransposer + ->qmfInBufImag_F[pSlotStretch[k] + 1][sourceband + 1]; + + gammaVecReal_m[1] += + (fMult(tmpRealF, hintReal_F[sourceband % 4][2]) - + fMult(tmpImagF, hintReal_F[(sourceband + 1) % 4][2])) >> + 1; + gammaVecImag_m[1] += + (fMult(tmpRealF, hintReal_F[(sourceband + 1) % 4][2]) + + fMult(tmpImagF, hintReal_F[sourceband % 4][2])) >> + 1; + gammaVec_e[1]++; + } + + addHighBandPart(gammaVecReal_m[1], gammaVecImag_m[1], gammaVec_e[1], + factor, gammaCenterReal_m[0], gammaCenterImag_m[0], + gammaCenter_e[0], stretch, scale_factor_hbe, + &hQmfTransposer->qmfHBEBufReal_F[k][band], + &hQmfTransposer->qmfHBEBufImag_F[k][band]); + + addHighBandPart(gammaVecReal_m[0], gammaVecImag_m[0], gammaVec_e[0], + factor, gammaCenterReal_m[1], gammaCenterImag_m[1], + gammaCenter_e[1], stretch, scale_factor_hbe, + &hQmfTransposer->qmfHBEBufReal_F[k][band], + &hQmfTransposer->qmfHBEBufImag_F[k][band]); + } + } else { + for (k = start; k < stop; k++) { + gammaVecReal_m[0] = + hQmfTransposer->qmfInBufReal_F[pSlotStretch[k]][sourceband]; + gammaVecImag_m[0] = + hQmfTransposer->qmfInBufImag_F[pSlotStretch[k]][sourceband]; + gammaVec_e[0] = + SCALE2EXP(-hQmfTransposer->HBEAnalysiscQMF.outScalefactor); + + if (pFilt[k] == 1) { + FIXP_DBL tmpRealF = gammaVecReal_m[0], tmpImagF; + gammaVecReal_m[0] = + (fMult(gammaVecReal_m[0], hintReal_F[sourceband % 4][1]) - + fMult(gammaVecImag_m[0], + hintReal_F[(sourceband + 3) % 4][1])) >> + 1; /* sum should be <= 1 because of sin/cos multiplication */ + gammaVecImag_m[0] = + (fMult(tmpRealF, hintReal_F[(sourceband + 3) % 4][1]) + + fMult(gammaVecImag_m[0], hintReal_F[sourceband % 4][1])) >> + 1; /* sum should be <= 1 because of sin/cos multiplication */ + + tmpRealF = hQmfTransposer + ->qmfInBufReal_F[pSlotStretch[k] + 1][sourceband]; + tmpImagF = hQmfTransposer + ->qmfInBufImag_F[pSlotStretch[k] + 1][sourceband]; + + gammaVecReal_m[0] += + (fMult(tmpRealF, hintReal_F[sourceband % 4][1]) - + fMult(tmpImagF, hintReal_F[(sourceband + 1) % 4][1])) >> + 1; /* sum should be <= 1 because of sin/cos multiplication */ + gammaVecImag_m[0] += + (fMult(tmpRealF, hintReal_F[(sourceband + 1) % 4][1]) + + fMult(tmpImagF, hintReal_F[sourceband % 4][1])) >> + 1; /* sum should be <= 1 because of sin/cos multiplication */ + gammaVec_e[0]++; + } + + addHighBandPart(gammaVecReal_m[0], gammaVecImag_m[0], gammaVec_e[0], + factor, gammaCenterReal_m[0], gammaCenterImag_m[0], + gammaCenter_e[0], stretch, scale_factor_hbe, + &hQmfTransposer->qmfHBEBufReal_F[k][band], + &hQmfTransposer->qmfHBEBufImag_F[k][band]); + } + } + + /* pitchInBins is given with the resolution of a 768 bins FFT and we + * need 64 QMF units so factor 768/64 = 12 */ + if (pitchInBins >= pmin * (1 + bSbr41)) { + int tr, ti1, ti2, mTr = 0, ts1 = 0, ts2 = 0, mVal_e = 0, temp_e = 0; + int sqmag0_e = + SCALE2EXP(-hQmfTransposer->HBEAnalysiscQMF.outScalefactor); + + FIXP_DBL mVal_F = FL2FXCONST_DBL(0.f), sqmag0_F, sqmag1_F, sqmag2_F, + temp_F, f1_F; /* all equal exponent */ + sign = -1; + + sourceband = 2 * band / stretch - qmfOffset; /* consistent with the + already computed for + stretch = 3,4. */ + FDK_ASSERT(sourceband >= 0); + + FIXP_DBL sqmag0R_F = + hQmfTransposer->qmfInBufReal_F[slotOffset][sourceband]; + FIXP_DBL sqmag0I_F = + hQmfTransposer->qmfInBufImag_F[slotOffset][sourceband]; + scaleUp(&sqmag0R_F, &sqmag0I_F, &sqmag0_e); + + sqmag0_F = fPow2Div2(sqmag0R_F); + sqmag0_F += fPow2Div2(sqmag0I_F); + sqmag0_e = 2 * sqmag0_e + 1; + + for (tr = 1; tr < stretch; tr++) { + int sqmag1_e = + SCALE2EXP(-hQmfTransposer->HBEAnalysiscQMF.outScalefactor); + int sqmag2_e = + SCALE2EXP(-hQmfTransposer->HBEAnalysiscQMF.outScalefactor); + + FIXP_DBL tmp_band = band_F[band]; + FIXP_DBL tr_p = + fMult(p_F[pitchInBins] >> bSbr41, tr_str[tr - 1]); /* scale 7 */ + f1_F = + fMult(tmp_band - tr_p, stretchfac[stretch - 2]); /* scale 7 */ + ti1 = (INT)(f1_F >> (DFRACT_BITS - 1 - 7)) - qmfOffset; + ti2 = (INT)(((f1_F) + ((p_F[pitchInBins] >> bSbr41) >> 2)) >> + (DFRACT_BITS - 1 - 7)) - + qmfOffset; + + if (ti1 >= 0 && ti2 < 2 * hQmfTransposer->synthSize) { + FIXP_DBL sqmag1R_F = + hQmfTransposer->qmfInBufReal_F[slotOffset][ti1]; + FIXP_DBL sqmag1I_F = + hQmfTransposer->qmfInBufImag_F[slotOffset][ti1]; + scaleUp(&sqmag1R_F, &sqmag1I_F, &sqmag1_e); + sqmag1_F = fPow2Div2(sqmag1R_F); + sqmag1_F += fPow2Div2(sqmag1I_F); + sqmag1_e = 2 * sqmag1_e + 1; + + FIXP_DBL sqmag2R_F = + hQmfTransposer->qmfInBufReal_F[slotOffset][ti2]; + FIXP_DBL sqmag2I_F = + hQmfTransposer->qmfInBufImag_F[slotOffset][ti2]; + scaleUp(&sqmag2R_F, &sqmag2I_F, &sqmag2_e); + sqmag2_F = fPow2Div2(sqmag2R_F); + sqmag2_F += fPow2Div2(sqmag2I_F); + sqmag2_e = 2 * sqmag2_e + 1; + + int shift1 = fMin(fMax(sqmag1_e, sqmag2_e) - sqmag1_e, 31); + int shift2 = fMin(fMax(sqmag1_e, sqmag2_e) - sqmag2_e, 31); + + temp_F = fMin((sqmag1_F >> shift1), (sqmag2_F >> shift2)); + temp_e = fMax(sqmag1_e, sqmag2_e); + + int shift3 = fMin(fMax(temp_e, mVal_e) - temp_e, 31); + int shift4 = fMin(fMax(temp_e, mVal_e) - mVal_e, 31); + + if ((temp_F >> shift3) > (mVal_F >> shift4)) { + mVal_F = temp_F; + mVal_e = temp_e; /* equals sqmag2_e + shift2 */ + mTr = tr; + ts1 = ti1; + ts2 = ti2; + } + } + } + + int shift1 = fMin(fMax(sqmag0_e, mVal_e) - sqmag0_e, 31); + int shift2 = fMin(fMax(sqmag0_e, mVal_e) - mVal_e, 31); + + if ((mVal_F >> shift2) > (sqmag0_F >> shift1) && ts1 >= 0 && + ts2 < 2 * hQmfTransposer->synthSize) { + INT gammaOut_e[2]; + FIXP_DBL gammaOutReal_m[2], gammaOutImag_m[2]; + FIXP_DBL tmpReal_m = (FIXP_DBL)0, tmpImag_m = (FIXP_DBL)0; + + int Tcenter, Tvec; + + Tcenter = stretch - mTr; /* default phase power parameters */ + Tvec = mTr; + switch (stretch) /* 2 tap block creation design depends on stretch + order */ + { + case 2: + wingain = + FL2FXCONST_DBL(5.f / 12.f); /* sum of taps divided by two */ + + if (hQmfTransposer->bXProducts[0]) { + gammaCenterReal_m[0] = + hQmfTransposer->qmfInBufReal_F[slotOffset][ts1]; + gammaCenterImag_m[0] = + hQmfTransposer->qmfInBufImag_F[slotOffset][ts1]; + + for (k = 0; k < 2; k++) { + gammaVecReal_m[k] = + hQmfTransposer->qmfInBufReal_F[slotOffset - 1 + k][ts2]; + gammaVecImag_m[k] = + hQmfTransposer->qmfInBufImag_F[slotOffset - 1 + k][ts2]; + } + + gammaCenter_e[0] = SCALE2EXP( + -hQmfTransposer->HBEAnalysiscQMF.outScalefactor); + gammaVec_e[0] = gammaVec_e[1] = SCALE2EXP( + -hQmfTransposer->HBEAnalysiscQMF.outScalefactor); + } + break; + + case 4: + wingain = + FL2FXCONST_DBL(6.f / 12.f); /* sum of taps divided by two */ + if (hQmfTransposer->bXProducts[2]) { + if (mTr == 1) { + gammaCenterReal_m[0] = + hQmfTransposer->qmfInBufReal_F[slotOffset][ts1]; + gammaCenterImag_m[0] = + hQmfTransposer->qmfInBufImag_F[slotOffset][ts1]; + + for (k = 0; k < 2; k++) { + gammaVecReal_m[k] = + hQmfTransposer + ->qmfInBufReal_F[slotOffset + 2 * (k - 1)][ts2]; + gammaVecImag_m[k] = + hQmfTransposer + ->qmfInBufImag_F[slotOffset + 2 * (k - 1)][ts2]; + } + } else if (mTr == 2) { + gammaCenterReal_m[0] = + hQmfTransposer->qmfInBufReal_F[slotOffset][ts1]; + gammaCenterImag_m[0] = + hQmfTransposer->qmfInBufImag_F[slotOffset][ts1]; + + for (k = 0; k < 2; k++) { + gammaVecReal_m[k] = + hQmfTransposer + ->qmfInBufReal_F[slotOffset + (k - 1)][ts2]; + gammaVecImag_m[k] = + hQmfTransposer + ->qmfInBufImag_F[slotOffset + (k - 1)][ts2]; + } + } else /* (mTr == 3) */ + { + sign = 1; + Tcenter = mTr; /* opposite phase power parameters as ts2 is + center */ + Tvec = stretch - mTr; + + gammaCenterReal_m[0] = + hQmfTransposer->qmfInBufReal_F[slotOffset][ts2]; + gammaCenterImag_m[0] = + hQmfTransposer->qmfInBufImag_F[slotOffset][ts2]; + + for (k = 0; k < 2; k++) { + gammaVecReal_m[k] = + hQmfTransposer + ->qmfInBufReal_F[slotOffset + 2 * (k - 1)][ts1]; + gammaVecImag_m[k] = + hQmfTransposer + ->qmfInBufImag_F[slotOffset + 2 * (k - 1)][ts1]; + } + } + + gammaCenter_e[0] = SCALE2EXP( + -hQmfTransposer->HBEAnalysiscQMF.outScalefactor); + gammaVec_e[0] = gammaVec_e[1] = SCALE2EXP( + -hQmfTransposer->HBEAnalysiscQMF.outScalefactor); + } + break; + + case 3: + wingain = FL2FXCONST_DBL(5.6568f / + 12.f); /* sum of taps divided by two */ + + if (hQmfTransposer->bXProducts[1]) { + FIXP_DBL tmpReal_F, tmpImag_F; + if (mTr == 1) { + gammaCenterReal_m[0] = + hQmfTransposer->qmfInBufReal_F[slotOffset][ts1]; + gammaCenterImag_m[0] = + hQmfTransposer->qmfInBufImag_F[slotOffset][ts1]; + gammaVecReal_m[1] = + hQmfTransposer->qmfInBufReal_F[slotOffset][ts2]; + gammaVecImag_m[1] = + hQmfTransposer->qmfInBufImag_F[slotOffset][ts2]; + + addrshift = -2; + tmpReal_F = + hQmfTransposer + ->qmfInBufReal_F[addrshift + slotOffset][ts2]; + tmpImag_F = + hQmfTransposer + ->qmfInBufImag_F[addrshift + slotOffset][ts2]; + + gammaVecReal_m[0] = + (fMult(factors[ts2 % 4], tmpReal_F) - + fMult(factors[(ts2 + 3) % 4], tmpImag_F)) >> + 1; + gammaVecImag_m[0] = + (fMult(factors[(ts2 + 3) % 4], tmpReal_F) + + fMult(factors[ts2 % 4], tmpImag_F)) >> + 1; + + tmpReal_F = + hQmfTransposer + ->qmfInBufReal_F[addrshift + 1 + slotOffset][ts2]; + tmpImag_F = + hQmfTransposer + ->qmfInBufImag_F[addrshift + 1 + slotOffset][ts2]; + + gammaVecReal_m[0] += + (fMult(factors[ts2 % 4], tmpReal_F) - + fMult(factors[(ts2 + 1) % 4], tmpImag_F)) >> + 1; + gammaVecImag_m[0] += + (fMult(factors[(ts2 + 1) % 4], tmpReal_F) + + fMult(factors[ts2 % 4], tmpImag_F)) >> + 1; + + } else /* (mTr == 2) */ + { + sign = 1; + Tcenter = mTr; /* opposite phase power parameters as ts2 is + center */ + Tvec = stretch - mTr; + + gammaCenterReal_m[0] = + hQmfTransposer->qmfInBufReal_F[slotOffset][ts2]; + gammaCenterImag_m[0] = + hQmfTransposer->qmfInBufImag_F[slotOffset][ts2]; + gammaVecReal_m[1] = + hQmfTransposer->qmfInBufReal_F[slotOffset][ts1]; + gammaVecImag_m[1] = + hQmfTransposer->qmfInBufImag_F[slotOffset][ts1]; + + addrshift = -2; + tmpReal_F = + hQmfTransposer + ->qmfInBufReal_F[addrshift + slotOffset][ts1]; + tmpImag_F = + hQmfTransposer + ->qmfInBufImag_F[addrshift + slotOffset][ts1]; + + gammaVecReal_m[0] = + (fMult(factors[ts1 % 4], tmpReal_F) - + fMult(factors[(ts1 + 3) % 4], tmpImag_F)) >> + 1; + gammaVecImag_m[0] = + (fMult(factors[(ts1 + 3) % 4], tmpReal_F) + + fMult(factors[ts1 % 4], tmpImag_F)) >> + 1; + + tmpReal_F = + hQmfTransposer + ->qmfInBufReal_F[addrshift + 1 + slotOffset][ts1]; + tmpImag_F = + hQmfTransposer + ->qmfInBufImag_F[addrshift + 1 + slotOffset][ts1]; + + gammaVecReal_m[0] += + (fMult(factors[ts1 % 4], tmpReal_F) - + fMult(factors[(ts1 + 1) % 4], tmpImag_F)) >> + 1; + gammaVecImag_m[0] += + (fMult(factors[(ts1 + 1) % 4], tmpReal_F) + + fMult(factors[ts1 % 4], tmpImag_F)) >> + 1; + } + + gammaCenter_e[0] = gammaVec_e[1] = SCALE2EXP( + -hQmfTransposer->HBEAnalysiscQMF.outScalefactor); + gammaVec_e[0] = + SCALE2EXP( + -hQmfTransposer->HBEAnalysiscQMF.outScalefactor) + + 1; + } + break; + default: + FDK_ASSERT(0); + break; + } /* stretch cases */ + + /* parameter controlled phase modification parts */ + /* maximum *_e == 20 */ + calculateCenterFIXP(gammaCenterReal_m[0], gammaCenterImag_m[0], + &gammaCenterReal_m[0], &gammaCenterImag_m[0], + &gammaCenter_e[0], stretch, Tcenter - 1); + calculateCenterFIXP(gammaVecReal_m[0], gammaVecImag_m[0], + &gammaVecReal_m[0], &gammaVecImag_m[0], + &gammaVec_e[0], stretch, Tvec - 1); + calculateCenterFIXP(gammaVecReal_m[1], gammaVecImag_m[1], + &gammaVecReal_m[1], &gammaVecImag_m[1], + &gammaVec_e[1], stretch, Tvec - 1); + + /* Final multiplication of prepared parts */ + for (k = 0; k < 2; k++) { + gammaOutReal_m[k] = + fMultDiv2(gammaVecReal_m[k], gammaCenterReal_m[0]) - + fMultDiv2(gammaVecImag_m[k], gammaCenterImag_m[0]); + gammaOutImag_m[k] = + fMultDiv2(gammaVecReal_m[k], gammaCenterImag_m[0]) + + fMultDiv2(gammaVecImag_m[k], gammaCenterReal_m[0]); + gammaOut_e[k] = gammaCenter_e[0] + gammaVec_e[k] + 1; + } + + scaleUp(&gammaOutReal_m[0], &gammaOutImag_m[0], &gammaOut_e[0]); + scaleUp(&gammaOutReal_m[1], &gammaOutImag_m[1], &gammaOut_e[1]); + FDK_ASSERT(gammaOut_e[0] >= 0); + FDK_ASSERT(gammaOut_e[0] < 32); + + tmpReal_m = gammaOutReal_m[0]; + tmpImag_m = gammaOutImag_m[0]; + + INT modstretch4 = ((stretch == 4) && (mTr == 2)); + + FIXP_DBL cos_twid = twid_m_new[stretch - 2 - modstretch4][0]; + FIXP_DBL sin_twid = sign * twid_m_new[stretch - 2 - modstretch4][1]; + + gammaOutReal_m[0] = + fMult(tmpReal_m, cos_twid) - + fMult(tmpImag_m, sin_twid); /* sum should be <= 1 because of + sin/cos multiplication */ + gammaOutImag_m[0] = + fMult(tmpImag_m, cos_twid) + + fMult(tmpReal_m, sin_twid); /* sum should be <= 1 because of + sin/cos multiplication */ + + /* wingain */ + for (k = 0; k < 2; k++) { + gammaOutReal_m[k] = (fMult(gammaOutReal_m[k], wingain) << 1); + gammaOutImag_m[k] = (fMult(gammaOutImag_m[k], wingain) << 1); + } + + gammaOutReal_m[1] >>= 1; + gammaOutImag_m[1] >>= 1; + gammaOut_e[0] += 2; + gammaOut_e[1] += 2; + + /* OLA including window scaling by wingain/3 */ + for (k = 0; k < 2; k++) /* need k=1 to correspond to + grainModImag[slotOffset] -> out to + j*2+(slotOffset-offset) */ + { + hQmfTransposer->qmfHBEBufReal_F[(k + slotOffset - 1)][band] += + gammaOutReal_m[k] >> (scale_factor_hbe - gammaOut_e[k]); + hQmfTransposer->qmfHBEBufImag_F[(k + slotOffset - 1)][band] += + gammaOutImag_m[k] >> (scale_factor_hbe - gammaOut_e[k]); + } + } /* mVal > qThrQMF * qThrQMF * sqmag0 && ts1 > 0 && ts2 < 64 */ + } /* p >= pmin */ + } /* for band */ + } /* for stretch */ + + for (i = 0; i < QMF_WIN_LEN - 1; i++) { + FDKmemcpy(hQmfTransposer->qmfInBufReal_F[i], + hQmfTransposer->qmfInBufReal_F[i + 1], + sizeof(FIXP_DBL) * hQmfTransposer->HBEAnalysiscQMF.no_channels); + FDKmemcpy(hQmfTransposer->qmfInBufImag_F[i], + hQmfTransposer->qmfInBufImag_F[i + 1], + sizeof(FIXP_DBL) * hQmfTransposer->HBEAnalysiscQMF.no_channels); + } + + if (keepStatesSyncedMode != KEEP_STATES_SYNCED_NOOUT) { + if (2 * j >= offset) { + /* copy first two slots of internal buffer to output */ + if (keepStatesSyncedMode == KEEP_STATES_SYNCED_OUTDIFF) { + for (i = 0; i < 2; i++) { + FDKmemcpy(&ppQmfBufferOutReal_F[2 * j - offset + i] + [hQmfTransposer->xOverQmf[0]], + &hQmfTransposer + ->qmfHBEBufReal_F[i][hQmfTransposer->xOverQmf[0]], + (QMF_SYNTH_CHANNELS - hQmfTransposer->xOverQmf[0]) * + sizeof(FIXP_DBL)); + FDKmemcpy(&ppQmfBufferOutImag_F[2 * j - offset + i] + [hQmfTransposer->xOverQmf[0]], + &hQmfTransposer + ->qmfHBEBufImag_F[i][hQmfTransposer->xOverQmf[0]], + (QMF_SYNTH_CHANNELS - hQmfTransposer->xOverQmf[0]) * + sizeof(FIXP_DBL)); + } + } else { + for (i = 0; i < 2; i++) { + FDKmemcpy(&ppQmfBufferOutReal_F[2 * j + i + ov_len] + [hQmfTransposer->xOverQmf[0]], + &hQmfTransposer + ->qmfHBEBufReal_F[i][hQmfTransposer->xOverQmf[0]], + (QMF_SYNTH_CHANNELS - hQmfTransposer->xOverQmf[0]) * + sizeof(FIXP_DBL)); + FDKmemcpy(&ppQmfBufferOutImag_F[2 * j + i + ov_len] + [hQmfTransposer->xOverQmf[0]], + &hQmfTransposer + ->qmfHBEBufImag_F[i][hQmfTransposer->xOverQmf[0]], + (QMF_SYNTH_CHANNELS - hQmfTransposer->xOverQmf[0]) * + sizeof(FIXP_DBL)); + } + } + } + } + + /* move slots up */ + for (i = 0; i < HBE_MAX_OUT_SLOTS - 2; i++) { + FDKmemcpy( + &hQmfTransposer->qmfHBEBufReal_F[i][hQmfTransposer->xOverQmf[0]], + &hQmfTransposer->qmfHBEBufReal_F[i + 2][hQmfTransposer->xOverQmf[0]], + (QMF_SYNTH_CHANNELS - hQmfTransposer->xOverQmf[0]) * + sizeof(FIXP_DBL)); + FDKmemcpy( + &hQmfTransposer->qmfHBEBufImag_F[i][hQmfTransposer->xOverQmf[0]], + &hQmfTransposer->qmfHBEBufImag_F[i + 2][hQmfTransposer->xOverQmf[0]], + (QMF_SYNTH_CHANNELS - hQmfTransposer->xOverQmf[0]) * + sizeof(FIXP_DBL)); + } + + /* finally set last two slot to zero */ + for (i = 0; i < 2; i++) { + FDKmemset(&hQmfTransposer->qmfHBEBufReal_F[HBE_MAX_OUT_SLOTS - 1 - i] + [hQmfTransposer->xOverQmf[0]], + 0, + (QMF_SYNTH_CHANNELS - hQmfTransposer->xOverQmf[0]) * + sizeof(FIXP_DBL)); + FDKmemset(&hQmfTransposer->qmfHBEBufImag_F[HBE_MAX_OUT_SLOTS - 1 - i] + [hQmfTransposer->xOverQmf[0]], + 0, + (QMF_SYNTH_CHANNELS - hQmfTransposer->xOverQmf[0]) * + sizeof(FIXP_DBL)); + } + } /* qmfVocoderColsIn */ + + if (keepStatesSyncedMode != KEEP_STATES_SYNCED_NOOUT) { + if (keepStatesSyncedMode == KEEP_STATES_SYNCED_OUTDIFF) { + for (i = 0; i < ov_len + LPC_ORDER; i++) { + for (band = hQmfTransposer->startBand; band < hQmfTransposer->stopBand; + band++) { + FIXP_DBL tmpR = ppQmfBufferOutReal_F[i][band]; + FIXP_DBL tmpI = ppQmfBufferOutImag_F[i][band]; + + ppQmfBufferOutReal_F[i][band] = + fMult(tmpR, cos_F[band]) - + fMult(tmpI, (-cos_F[64 - band - 1])); /* sum should be <= 1 + because of sin/cos + multiplication */ + ppQmfBufferOutImag_F[i][band] = + fMult(tmpR, (-cos_F[64 - band - 1])) + + fMult(tmpI, cos_F[band]); /* sum should by <= 1 because of sin/cos + multiplication */ + } + } + } else { + for (i = offset; i < hQmfTransposer->noCols; i++) { + for (band = hQmfTransposer->startBand; band < hQmfTransposer->stopBand; + band++) { + FIXP_DBL tmpR = ppQmfBufferOutReal_F[i + ov_len][band]; + FIXP_DBL tmpI = ppQmfBufferOutImag_F[i + ov_len][band]; + + ppQmfBufferOutReal_F[i + ov_len][band] = + fMult(tmpR, cos_F[band]) - + fMult(tmpI, (-cos_F[64 - band - 1])); /* sum should be <= 1 + because of sin/cos + multiplication */ + ppQmfBufferOutImag_F[i + ov_len][band] = + fMult(tmpR, (-cos_F[64 - band - 1])) + + fMult(tmpI, cos_F[band]); /* sum should by <= 1 because of sin/cos + multiplication */ + } + } + } + } + + *scale_hb = EXP2SCALE(scale_factor_hbe); +} + +int* GetxOverBandQmfTransposer(HANDLE_HBE_TRANSPOSER hQmfTransposer) { + if (hQmfTransposer) + return hQmfTransposer->xOverQmf; + else + return NULL; +} + +int Get41SbrQmfTransposer(HANDLE_HBE_TRANSPOSER hQmfTransposer) { + if (hQmfTransposer != NULL) + return hQmfTransposer->bSbr41; + else + return 0; +} diff --git a/fdk-aac/libSBRdec/src/hbe.h b/fdk-aac/libSBRdec/src/hbe.h new file mode 100644 index 0000000..fdffe1e --- /dev/null +++ b/fdk-aac/libSBRdec/src/hbe.h @@ -0,0 +1,200 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +#ifndef HBE_H +#define HBE_H + +#include "sbrdecoder.h" + +#ifndef QMF_SYNTH_CHANNELS +#define QMF_SYNTH_CHANNELS (64) +#endif + +#define HBE_QMF_FILTER_STATE_ANA_SIZE (400) +#define HBE_QMF_FILTER_STATE_SYN_SIZE (200) + +#ifndef MAX_NUM_PATCHES_HBE +#define MAX_NUM_PATCHES_HBE (6) +#endif +#define MAX_STRETCH_HBE (4) + +typedef enum { + KEEP_STATES_SYNCED_OFF = 0, /*!< normal QMF transposer behaviour */ + KEEP_STATES_SYNCED_NORMAL = 1, /*!< QMF transposer called for syncing of + states the last 8/14 slots are calculated in + case next frame is HBE */ + KEEP_STATES_SYNCED_OUTDIFF = + 2, /*!< QMF transposer behaviour as in normal case, but the calculated + slots are directly written to overlap area of buffer; only used in + resetSbrDec function */ + KEEP_STATES_SYNCED_NOOUT = + 3 /*!< QMF transposer is called for syncing of states only, not output + is generated at all; only used in resetSbrDec function */ +} KEEP_STATES_SYNCED_MODE; + +struct hbeTransposer { + int xOverQmf[MAX_NUM_PATCHES_HBE]; + + int maxStretch; + int timeDomainWinLen; + int qmfInBufSize; + int qmfOutBufSize; + int noCols; + int noChannels; + int startBand; + int stopBand; + int bSbr41; + + INT_PCM *inBuf_F; + FIXP_DBL **qmfInBufReal_F; + FIXP_DBL **qmfInBufImag_F; + + FIXP_DBL *qmfBufferCodecTempSlot_F; + + QMF_FILTER_BANK HBEAnalysiscQMF; + QMF_FILTER_BANK HBESynthesisQMF; + + FIXP_DBL const *synthesisQmfPreModCos_F; + FIXP_DBL const *synthesisQmfPreModSin_F; + + FIXP_QAS anaQmfStates[HBE_QMF_FILTER_STATE_ANA_SIZE]; + FIXP_QSS synQmfStates[HBE_QMF_FILTER_STATE_SYN_SIZE]; + + FIXP_DBL **qmfHBEBufReal_F; + FIXP_DBL **qmfHBEBufImag_F; + + int bXProducts[MAX_STRETCH_HBE]; + + int kstart; + int synthSize; + + int highband_exp[2]; + int target_exp[2]; +}; + +typedef struct hbeTransposer *HANDLE_HBE_TRANSPOSER; + +SBR_ERROR QmfTransposerCreate(HANDLE_HBE_TRANSPOSER *hQmfTransposer, + const int frameSize, int bDisableCrossProducts, + int bSbr41); + +SBR_ERROR QmfTransposerReInit(HANDLE_HBE_TRANSPOSER hQmfTransposer, + UCHAR *FreqBandTable[2], UCHAR NSfb[2]); + +void QmfTransposerClose(HANDLE_HBE_TRANSPOSER hQmfTransposer); + +void QmfTransposerApply(HANDLE_HBE_TRANSPOSER hQmfTransposer, + FIXP_DBL **qmfBufferCodecReal, + FIXP_DBL **qmfBufferCodecImag, int nColsIn, + FIXP_DBL **ppQmfBufferOutReal_F, + FIXP_DBL **ppQmfBufferOutImag_F, + FIXP_DBL lpcFilterStatesReal[2 + (3 * (4))][(64)], + FIXP_DBL lpcFilterStatesImag[2 + (3 * (4))][(64)], + int pitchInBins, int scale_lb, int scale_hbe, + int *scale_hb, int timeStep, int firstSlotOffsset, + int ov_len, + KEEP_STATES_SYNCED_MODE keepStatesSyncedMode); + +int *GetxOverBandQmfTransposer(HANDLE_HBE_TRANSPOSER hQmfTransposer); + +int Get41SbrQmfTransposer(HANDLE_HBE_TRANSPOSER hQmfTransposer); +#endif /* HBE_H */ diff --git a/fdk-aac/libSBRdec/src/huff_dec.cpp b/fdk-aac/libSBRdec/src/huff_dec.cpp new file mode 100644 index 0000000..90c9541 --- /dev/null +++ b/fdk-aac/libSBRdec/src/huff_dec.cpp @@ -0,0 +1,137 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief Huffman Decoder +*/ + +#include "huff_dec.h" + +/***************************************************************************/ +/*! + \brief Decodes one huffman code word + + Reads bits from the bitstream until a valid codeword is found. + The table entries are interpreted either as index to the next entry + or - if negative - as the codeword. + + \return decoded value + + \author + +****************************************************************************/ +int DecodeHuffmanCW(Huffman h, /*!< pointer to huffman codebook table */ + HANDLE_FDK_BITSTREAM hBs) /*!< Handle to Bitbuffer */ +{ + SCHAR index = 0; + int value, bit; + + while (index >= 0) { + bit = FDKreadBits(hBs, 1); + index = h[index][bit]; + } + + value = index + 64; /* Add offset */ + + return value; +} diff --git a/fdk-aac/libSBRdec/src/huff_dec.h b/fdk-aac/libSBRdec/src/huff_dec.h new file mode 100644 index 0000000..9aa62b4 --- /dev/null +++ b/fdk-aac/libSBRdec/src/huff_dec.h @@ -0,0 +1,117 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief Huffman Decoder +*/ +#ifndef HUFF_DEC_H +#define HUFF_DEC_H + +#include "sbrdecoder.h" +#include "FDK_bitstream.h" + +typedef const SCHAR (*Huffman)[2]; + +int DecodeHuffmanCW(Huffman h, HANDLE_FDK_BITSTREAM hBitBuf); + +#endif diff --git a/fdk-aac/libSBRdec/src/lpp_tran.cpp b/fdk-aac/libSBRdec/src/lpp_tran.cpp new file mode 100644 index 0000000..2ef07eb --- /dev/null +++ b/fdk-aac/libSBRdec/src/lpp_tran.cpp @@ -0,0 +1,1471 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief Low Power Profile Transposer + This module provides the transposer. The main entry point is lppTransposer(). + The function generates high frequency content by copying data from the low + band (provided by core codec) into the high band. This process is also + referred to as "patching". The function also implements spectral whitening by + means of inverse filtering based on LPC coefficients. + + Together with the QMF filterbank the transposer can be tested using a supplied + test program. See main_audio.cpp for details. This module does use fractional + arithmetic and the accuracy of the computations has an impact on the overall + sound quality. The module also needs to take into account the different + scaling of spectral data. + + \sa lppTransposer(), main_audio.cpp, sbr_scale.h, \ref documentationOverview +*/ + +#ifdef __ANDROID__ +#include "log/log.h" +#endif + +#include "lpp_tran.h" + +#include "sbr_ram.h" +#include "sbr_rom.h" + +#include "genericStds.h" +#include "autocorr2nd.h" + +#include "HFgen_preFlat.h" + +#if defined(__arm__) +#include "arm/lpp_tran_arm.cpp" +#endif + +#define LPC_SCALE_FACTOR 2 + +/*! + * + * \brief Get bandwidth expansion factor from filtering level + * + * Returns a filter parameter (bandwidth expansion factor) depending on + * the desired filtering level signalled in the bitstream. + * When switching the filtering level from LOW to OFF, an additional + * level is being inserted to achieve a smooth transition. + */ + +static FIXP_DBL mapInvfMode(INVF_MODE mode, INVF_MODE prevMode, + WHITENING_FACTORS whFactors) { + switch (mode) { + case INVF_LOW_LEVEL: + if (prevMode == INVF_OFF) + return whFactors.transitionLevel; + else + return whFactors.lowLevel; + + case INVF_MID_LEVEL: + return whFactors.midLevel; + + case INVF_HIGH_LEVEL: + return whFactors.highLevel; + + default: + if (prevMode == INVF_LOW_LEVEL) + return whFactors.transitionLevel; + else + return whFactors.off; + } +} + +/*! + * + * \brief Perform inverse filtering level emphasis + * + * Retrieve bandwidth expansion factor and apply smoothing for each filter band + * + */ + +static void inverseFilteringLevelEmphasis( + HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transposer */ + UCHAR nInvfBands, /*!< Number of bands for inverse filtering */ + INVF_MODE *sbr_invf_mode, /*!< Current inverse filtering modes */ + INVF_MODE *sbr_invf_mode_prev, /*!< Previous inverse filtering modes */ + FIXP_DBL *bwVector /*!< Resulting filtering levels */ +) { + for (int i = 0; i < nInvfBands; i++) { + FIXP_DBL accu; + FIXP_DBL bwTmp = mapInvfMode(sbr_invf_mode[i], sbr_invf_mode_prev[i], + hLppTrans->pSettings->whFactors); + + if (bwTmp < hLppTrans->bwVectorOld[i]) { + accu = fMultDiv2(FL2FXCONST_DBL(0.75f), bwTmp) + + fMultDiv2(FL2FXCONST_DBL(0.25f), hLppTrans->bwVectorOld[i]); + } else { + accu = fMultDiv2(FL2FXCONST_DBL(0.90625f), bwTmp) + + fMultDiv2(FL2FXCONST_DBL(0.09375f), hLppTrans->bwVectorOld[i]); + } + + if (accu<FL2FXCONST_DBL(0.015625f)>> 1) { + bwVector[i] = FL2FXCONST_DBL(0.0f); + } else { + bwVector[i] = fixMin(accu << 1, FL2FXCONST_DBL(0.99609375f)); + } + } +} + +/* Resulting autocorrelation determinant exponent */ +#define ACDET_EXP \ + (2 * (DFRACT_BITS + sbrScaleFactor->lb_scale + 10 - ac.det_scale)) +#define AC_EXP (-sbrScaleFactor->lb_scale + LPC_SCALE_FACTOR) +#define ALPHA_EXP (-sbrScaleFactor->lb_scale + LPC_SCALE_FACTOR + 1) +/* Resulting transposed QMF values exponent 16 bit normalized samplebits + * assumed. */ +#define QMFOUT_EXP ((SAMPLE_BITS - 15) - sbrScaleFactor->lb_scale) + +static inline void calc_qmfBufferReal(FIXP_DBL **qmfBufferReal, + const FIXP_DBL *const lowBandReal, + const int startSample, + const int stopSample, const UCHAR hiBand, + const int dynamicScale, const int descale, + const FIXP_SGL a0r, const FIXP_SGL a1r) { + FIXP_DBL accu1, accu2; + int i; + + for (i = 0; i < stopSample - startSample; i++) { + accu1 = fMultDiv2(a1r, lowBandReal[i]); + accu1 = (fMultDiv2(a0r, lowBandReal[i + 1]) + accu1); + accu1 = accu1 >> dynamicScale; + + accu1 <<= 1; + accu2 = (lowBandReal[i + 2] >> descale); + qmfBufferReal[i + startSample][hiBand] = accu1 + accu2; + } +} + +/*! + * + * \brief Perform transposition by patching of subband samples. + * This function serves as the main entry point into the module. The function + * determines the areas for the patching process (these are the source range as + * well as the target range) and implements spectral whitening by means of + * inverse filtering. The function autoCorrelation2nd() is an auxiliary function + * for calculating the LPC coefficients for the filtering. The actual + * calculation of the LPC coefficients and the implementation of the filtering + * are done as part of lppTransposer(). + * + * Note that the filtering is done on all available QMF subsamples, whereas the + * patching is only done on those QMF subsamples that will be used in the next + * QMF synthesis. The filtering is also implemented before the patching includes + * further dependencies on parameters from the SBR data. + * + */ + +void lppTransposer( + HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transposer */ + QMF_SCALE_FACTOR *sbrScaleFactor, /*!< Scaling factors */ + FIXP_DBL **qmfBufferReal, /*!< Pointer to pointer to real part of subband + samples (source) */ + + FIXP_DBL *degreeAlias, /*!< Vector for results of aliasing estimation */ + FIXP_DBL **qmfBufferImag, /*!< Pointer to pointer to imaginary part of + subband samples (source) */ + const int useLP, const int fPreWhitening, const int v_k_master0, + const int timeStep, /*!< Time step of envelope */ + const int firstSlotOffs, /*!< Start position in time */ + const int lastSlotOffs, /*!< Number of overlap-slots into next frame */ + const int nInvfBands, /*!< Number of bands for inverse filtering */ + INVF_MODE *sbr_invf_mode, /*!< Current inverse filtering modes */ + INVF_MODE *sbr_invf_mode_prev /*!< Previous inverse filtering modes */ +) { + INT bwIndex[MAX_NUM_PATCHES]; + FIXP_DBL bwVector[MAX_NUM_PATCHES]; /*!< pole moving factors */ + FIXP_DBL preWhiteningGains[(64) / 2]; + int preWhiteningGains_exp[(64) / 2]; + + int i; + int loBand, start, stop; + TRANSPOSER_SETTINGS *pSettings = hLppTrans->pSettings; + PATCH_PARAM *patchParam = pSettings->patchParam; + int patch; + + FIXP_SGL alphar[LPC_ORDER], a0r, a1r; + FIXP_SGL alphai[LPC_ORDER], a0i = 0, a1i = 0; + FIXP_SGL bw = FL2FXCONST_SGL(0.0f); + + int autoCorrLength; + + FIXP_DBL k1, k1_below = 0, k1_below2 = 0; + + ACORR_COEFS ac; + int startSample; + int stopSample; + int stopSampleClear; + + int comLowBandScale; + int ovLowBandShift; + int lowBandShift; + /* int ovHighBandShift;*/ + + alphai[0] = FL2FXCONST_SGL(0.0f); + alphai[1] = FL2FXCONST_SGL(0.0f); + + startSample = firstSlotOffs * timeStep; + stopSample = pSettings->nCols + lastSlotOffs * timeStep; + FDK_ASSERT((lastSlotOffs * timeStep) <= pSettings->overlap); + + inverseFilteringLevelEmphasis(hLppTrans, nInvfBands, sbr_invf_mode, + sbr_invf_mode_prev, bwVector); + + stopSampleClear = stopSample; + + autoCorrLength = pSettings->nCols + pSettings->overlap; + + if (pSettings->noOfPatches > 0) { + /* Set upper subbands to zero: + This is required in case that the patches do not cover the complete + highband (because the last patch would be too short). Possible + optimization: Clearing bands up to usb would be sufficient here. */ + int targetStopBand = + patchParam[pSettings->noOfPatches - 1].targetStartBand + + patchParam[pSettings->noOfPatches - 1].numBandsInPatch; + + int memSize = ((64) - targetStopBand) * sizeof(FIXP_DBL); + + if (!useLP) { + for (i = startSample; i < stopSampleClear; i++) { + FDKmemclear(&qmfBufferReal[i][targetStopBand], memSize); + FDKmemclear(&qmfBufferImag[i][targetStopBand], memSize); + } + } else { + for (i = startSample; i < stopSampleClear; i++) { + FDKmemclear(&qmfBufferReal[i][targetStopBand], memSize); + } + } + } +#ifdef __ANDROID__ + else { + // Safetynet logging + android_errorWriteLog(0x534e4554, "112160868"); + } +#endif + + /* init bwIndex for each patch */ + FDKmemclear(bwIndex, sizeof(bwIndex)); + + /* + Calc common low band scale factor + */ + comLowBandScale = + fixMin(sbrScaleFactor->ov_lb_scale, sbrScaleFactor->lb_scale); + + ovLowBandShift = sbrScaleFactor->ov_lb_scale - comLowBandScale; + lowBandShift = sbrScaleFactor->lb_scale - comLowBandScale; + /* ovHighBandShift = firstSlotOffs == 0 ? ovLowBandShift:0;*/ + + if (fPreWhitening) { + sbrDecoder_calculateGainVec( + qmfBufferReal, qmfBufferImag, + DFRACT_BITS - 1 - 16 - + sbrScaleFactor->ov_lb_scale, /* convert scale to exponent */ + DFRACT_BITS - 1 - 16 - + sbrScaleFactor->lb_scale, /* convert scale to exponent */ + pSettings->overlap, preWhiteningGains, preWhiteningGains_exp, + v_k_master0, startSample, stopSample); + } + + /* outer loop over bands to do analysis only once for each band */ + + if (!useLP) { + start = pSettings->lbStartPatching; + stop = pSettings->lbStopPatching; + } else { + start = fixMax(1, pSettings->lbStartPatching - 2); + stop = patchParam[0].targetStartBand; + } + + for (loBand = start; loBand < stop; loBand++) { + FIXP_DBL lowBandReal[(((1024) / (32) * (4) / 2) + (3 * (4))) + LPC_ORDER]; + FIXP_DBL *plowBandReal = lowBandReal; + FIXP_DBL **pqmfBufferReal = + qmfBufferReal + firstSlotOffs * timeStep /* + pSettings->overlap */; + FIXP_DBL lowBandImag[(((1024) / (32) * (4) / 2) + (3 * (4))) + LPC_ORDER]; + FIXP_DBL *plowBandImag = lowBandImag; + FIXP_DBL **pqmfBufferImag = + qmfBufferImag + firstSlotOffs * timeStep /* + pSettings->overlap */; + int resetLPCCoeffs = 0; + int dynamicScale = DFRACT_BITS - 1 - LPC_SCALE_FACTOR; + int acDetScale = 0; /* scaling of autocorrelation determinant */ + + for (i = 0; + i < LPC_ORDER + firstSlotOffs * timeStep /*+pSettings->overlap*/; + i++) { + *plowBandReal++ = hLppTrans->lpcFilterStatesRealLegSBR[i][loBand]; + if (!useLP) + *plowBandImag++ = hLppTrans->lpcFilterStatesImagLegSBR[i][loBand]; + } + + /* + Take old slope length qmf slot source values out of (overlap)qmf buffer + */ + if (!useLP) { + for (i = 0; + i < pSettings->nCols + pSettings->overlap - firstSlotOffs * timeStep; + i++) { + *plowBandReal++ = (*pqmfBufferReal++)[loBand]; + *plowBandImag++ = (*pqmfBufferImag++)[loBand]; + } + } else { + /* pSettings->overlap is always even */ + FDK_ASSERT((pSettings->overlap & 1) == 0); + for (i = 0; i < ((pSettings->nCols + pSettings->overlap - + firstSlotOffs * timeStep) >> + 1); + i++) { + *plowBandReal++ = (*pqmfBufferReal++)[loBand]; + *plowBandReal++ = (*pqmfBufferReal++)[loBand]; + } + if (pSettings->nCols & 1) { + *plowBandReal++ = (*pqmfBufferReal++)[loBand]; + } + } + + /* + Determine dynamic scaling value. + */ + dynamicScale = + fixMin(dynamicScale, + getScalefactor(lowBandReal, LPC_ORDER + pSettings->overlap) + + ovLowBandShift); + dynamicScale = + fixMin(dynamicScale, + getScalefactor(&lowBandReal[LPC_ORDER + pSettings->overlap], + pSettings->nCols) + + lowBandShift); + if (!useLP) { + dynamicScale = + fixMin(dynamicScale, + getScalefactor(lowBandImag, LPC_ORDER + pSettings->overlap) + + ovLowBandShift); + dynamicScale = + fixMin(dynamicScale, + getScalefactor(&lowBandImag[LPC_ORDER + pSettings->overlap], + pSettings->nCols) + + lowBandShift); + } + dynamicScale = fixMax( + 0, dynamicScale - 1); /* one additional bit headroom to prevent -1.0 */ + + /* + Scale temporal QMF buffer. + */ + scaleValues(&lowBandReal[0], LPC_ORDER + pSettings->overlap, + dynamicScale - ovLowBandShift); + scaleValues(&lowBandReal[LPC_ORDER + pSettings->overlap], pSettings->nCols, + dynamicScale - lowBandShift); + + if (!useLP) { + scaleValues(&lowBandImag[0], LPC_ORDER + pSettings->overlap, + dynamicScale - ovLowBandShift); + scaleValues(&lowBandImag[LPC_ORDER + pSettings->overlap], + pSettings->nCols, dynamicScale - lowBandShift); + } + + if (!useLP) { + acDetScale += autoCorr2nd_cplx(&ac, lowBandReal + LPC_ORDER, + lowBandImag + LPC_ORDER, autoCorrLength); + } else { + acDetScale += + autoCorr2nd_real(&ac, lowBandReal + LPC_ORDER, autoCorrLength); + } + + /* Examine dynamic of determinant in autocorrelation. */ + acDetScale += 2 * (comLowBandScale + dynamicScale); + acDetScale *= 2; /* two times reflection coefficent scaling */ + acDetScale += ac.det_scale; /* ac scaling of determinant */ + + /* In case of determinant < 10^-38, resetLPCCoeffs=1 has to be enforced. */ + if (acDetScale > 126) { + resetLPCCoeffs = 1; + } + + alphar[1] = FL2FXCONST_SGL(0.0f); + if (!useLP) alphai[1] = FL2FXCONST_SGL(0.0f); + + if (ac.det != FL2FXCONST_DBL(0.0f)) { + FIXP_DBL tmp, absTmp, absDet; + + absDet = fixp_abs(ac.det); + + if (!useLP) { + tmp = (fMultDiv2(ac.r01r, ac.r12r) >> (LPC_SCALE_FACTOR - 1)) - + ((fMultDiv2(ac.r01i, ac.r12i) + fMultDiv2(ac.r02r, ac.r11r)) >> + (LPC_SCALE_FACTOR - 1)); + } else { + tmp = (fMultDiv2(ac.r01r, ac.r12r) >> (LPC_SCALE_FACTOR - 1)) - + (fMultDiv2(ac.r02r, ac.r11r) >> (LPC_SCALE_FACTOR - 1)); + } + absTmp = fixp_abs(tmp); + + /* + Quick check: is first filter coeff >= 1(4) + */ + { + INT scale; + FIXP_DBL result = fDivNorm(absTmp, absDet, &scale); + scale = scale + ac.det_scale; + + if ((scale > 0) && (result >= (FIXP_DBL)MAXVAL_DBL >> scale)) { + resetLPCCoeffs = 1; + } else { + alphar[1] = FX_DBL2FX_SGL(scaleValue(result, scale)); + if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) { + alphar[1] = -alphar[1]; + } + } + } + + if (!useLP) { + tmp = (fMultDiv2(ac.r01i, ac.r12r) >> (LPC_SCALE_FACTOR - 1)) + + ((fMultDiv2(ac.r01r, ac.r12i) - + (FIXP_DBL)fMultDiv2(ac.r02i, ac.r11r)) >> + (LPC_SCALE_FACTOR - 1)); + + absTmp = fixp_abs(tmp); + + /* + Quick check: is second filter coeff >= 1(4) + */ + { + INT scale; + FIXP_DBL result = fDivNorm(absTmp, absDet, &scale); + scale = scale + ac.det_scale; + + if ((scale > 0) && + (result >= /*FL2FXCONST_DBL(1.f)*/ (FIXP_DBL)MAXVAL_DBL >> + scale)) { + resetLPCCoeffs = 1; + } else { + alphai[1] = FX_DBL2FX_SGL(scaleValue(result, scale)); + if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) { + alphai[1] = -alphai[1]; + } + } + } + } + } + + alphar[0] = FL2FXCONST_SGL(0.0f); + if (!useLP) alphai[0] = FL2FXCONST_SGL(0.0f); + + if (ac.r11r != FL2FXCONST_DBL(0.0f)) { + /* ac.r11r is always >=0 */ + FIXP_DBL tmp, absTmp; + + if (!useLP) { + tmp = (ac.r01r >> (LPC_SCALE_FACTOR + 1)) + + (fMultDiv2(alphar[1], ac.r12r) + fMultDiv2(alphai[1], ac.r12i)); + } else { + if (ac.r01r >= FL2FXCONST_DBL(0.0f)) + tmp = (ac.r01r >> (LPC_SCALE_FACTOR + 1)) + + fMultDiv2(alphar[1], ac.r12r); + else + tmp = -((-ac.r01r) >> (LPC_SCALE_FACTOR + 1)) + + fMultDiv2(alphar[1], ac.r12r); + } + + absTmp = fixp_abs(tmp); + + /* + Quick check: is first filter coeff >= 1(4) + */ + + if (absTmp >= (ac.r11r >> 1)) { + resetLPCCoeffs = 1; + } else { + INT scale; + FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale); + alphar[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1)); + + if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f))) + alphar[0] = -alphar[0]; + } + + if (!useLP) { + tmp = (ac.r01i >> (LPC_SCALE_FACTOR + 1)) + + (fMultDiv2(alphai[1], ac.r12r) - fMultDiv2(alphar[1], ac.r12i)); + + absTmp = fixp_abs(tmp); + + /* + Quick check: is second filter coeff >= 1(4) + */ + if (absTmp >= (ac.r11r >> 1)) { + resetLPCCoeffs = 1; + } else { + INT scale; + FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale); + alphai[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1)); + if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f))) + alphai[0] = -alphai[0]; + } + } + } + + if (!useLP) { + /* Now check the quadratic criteria */ + if ((fMultDiv2(alphar[0], alphar[0]) + fMultDiv2(alphai[0], alphai[0])) >= + FL2FXCONST_DBL(0.5f)) + resetLPCCoeffs = 1; + if ((fMultDiv2(alphar[1], alphar[1]) + fMultDiv2(alphai[1], alphai[1])) >= + FL2FXCONST_DBL(0.5f)) + resetLPCCoeffs = 1; + } + + if (resetLPCCoeffs) { + alphar[0] = FL2FXCONST_SGL(0.0f); + alphar[1] = FL2FXCONST_SGL(0.0f); + if (!useLP) { + alphai[0] = FL2FXCONST_SGL(0.0f); + alphai[1] = FL2FXCONST_SGL(0.0f); + } + } + + if (useLP) { + /* Aliasing detection */ + if (ac.r11r == FL2FXCONST_DBL(0.0f)) { + k1 = FL2FXCONST_DBL(0.0f); + } else { + if (fixp_abs(ac.r01r) >= fixp_abs(ac.r11r)) { + if (fMultDiv2(ac.r01r, ac.r11r) < FL2FX_DBL(0.0f)) { + k1 = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_SGL(1.0f)*/; + } else { + /* Since this value is squared later, it must not ever become -1.0f. + */ + k1 = (FIXP_DBL)(MINVAL_DBL + 1) /*FL2FXCONST_SGL(-1.0f)*/; + } + } else { + INT scale; + FIXP_DBL result = + fDivNorm(fixp_abs(ac.r01r), fixp_abs(ac.r11r), &scale); + k1 = scaleValue(result, scale); + + if (!((ac.r01r < FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f)))) { + k1 = -k1; + } + } + } + if ((loBand > 1) && (loBand < v_k_master0)) { + /* Check if the gain should be locked */ + FIXP_DBL deg = + /*FL2FXCONST_DBL(1.0f)*/ (FIXP_DBL)MAXVAL_DBL - fPow2(k1_below); + degreeAlias[loBand] = FL2FXCONST_DBL(0.0f); + if (((loBand & 1) == 0) && (k1 < FL2FXCONST_DBL(0.0f))) { + if (k1_below < FL2FXCONST_DBL(0.0f)) { /* 2-Ch Aliasing Detection */ + degreeAlias[loBand] = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_DBL(1.0f)*/; + if (k1_below2 > + FL2FXCONST_DBL(0.0f)) { /* 3-Ch Aliasing Detection */ + degreeAlias[loBand - 1] = deg; + } + } else if (k1_below2 > + FL2FXCONST_DBL(0.0f)) { /* 3-Ch Aliasing Detection */ + degreeAlias[loBand] = deg; + } + } + if (((loBand & 1) == 1) && (k1 > FL2FXCONST_DBL(0.0f))) { + if (k1_below > FL2FXCONST_DBL(0.0f)) { /* 2-CH Aliasing Detection */ + degreeAlias[loBand] = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_DBL(1.0f)*/; + if (k1_below2 < + FL2FXCONST_DBL(0.0f)) { /* 3-CH Aliasing Detection */ + degreeAlias[loBand - 1] = deg; + } + } else if (k1_below2 < + FL2FXCONST_DBL(0.0f)) { /* 3-CH Aliasing Detection */ + degreeAlias[loBand] = deg; + } + } + } + /* remember k1 values of the 2 QMF channels below the current channel */ + k1_below2 = k1_below; + k1_below = k1; + } + + patch = 0; + + while (patch < pSettings->noOfPatches) { /* inner loop over every patch */ + + int hiBand = loBand + patchParam[patch].targetBandOffs; + + if (loBand < patchParam[patch].sourceStartBand || + loBand >= patchParam[patch].sourceStopBand + //|| hiBand >= hLppTrans->pSettings->noChannels + ) { + /* Lowband not in current patch - proceed */ + patch++; + continue; + } + + FDK_ASSERT(hiBand < (64)); + + /* bwIndex[patch] is already initialized with value from previous band + * inside this patch */ + while (hiBand >= pSettings->bwBorders[bwIndex[patch]] && + bwIndex[patch] < MAX_NUM_PATCHES - 1) { + bwIndex[patch]++; + } + + /* + Filter Step 2: add the left slope with the current filter to the buffer + pure source values are already in there + */ + bw = FX_DBL2FX_SGL(bwVector[bwIndex[patch]]); + + a0r = FX_DBL2FX_SGL( + fMult(bw, alphar[0])); /* Apply current bandwidth expansion factor */ + + if (!useLP) a0i = FX_DBL2FX_SGL(fMult(bw, alphai[0])); + bw = FX_DBL2FX_SGL(fPow2(bw)); + a1r = FX_DBL2FX_SGL(fMult(bw, alphar[1])); + if (!useLP) a1i = FX_DBL2FX_SGL(fMult(bw, alphai[1])); + + /* + Filter Step 3: insert the middle part which won't be windowed + */ + if (bw <= FL2FXCONST_SGL(0.0f)) { + if (!useLP) { + int descale = + fixMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale)); + for (i = startSample; i < stopSample; i++) { + FIXP_DBL accu1, accu2; + accu1 = lowBandReal[LPC_ORDER + i] >> descale; + accu2 = lowBandImag[LPC_ORDER + i] >> descale; + if (fPreWhitening) { + accu1 = scaleValueSaturate( + fMultDiv2(accu1, preWhiteningGains[loBand]), + preWhiteningGains_exp[loBand] + 1); + accu2 = scaleValueSaturate( + fMultDiv2(accu2, preWhiteningGains[loBand]), + preWhiteningGains_exp[loBand] + 1); + } + qmfBufferReal[i][hiBand] = accu1; + qmfBufferImag[i][hiBand] = accu2; + } + } else { + int descale = + fixMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale)); + for (i = startSample; i < stopSample; i++) { + qmfBufferReal[i][hiBand] = lowBandReal[LPC_ORDER + i] >> descale; + } + } + } else { /* bw <= 0 */ + + if (!useLP) { + int descale = + fixMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale)); +#ifdef FUNCTION_LPPTRANSPOSER_func1 + lppTransposer_func1( + lowBandReal + LPC_ORDER + startSample, + lowBandImag + LPC_ORDER + startSample, + qmfBufferReal + startSample, qmfBufferImag + startSample, + stopSample - startSample, (int)hiBand, dynamicScale, descale, a0r, + a0i, a1r, a1i, fPreWhitening, preWhiteningGains[loBand], + preWhiteningGains_exp[loBand] + 1); +#else + for (i = startSample; i < stopSample; i++) { + FIXP_DBL accu1, accu2; + + accu1 = (fMultDiv2(a0r, lowBandReal[LPC_ORDER + i - 1]) - + fMultDiv2(a0i, lowBandImag[LPC_ORDER + i - 1]) + + fMultDiv2(a1r, lowBandReal[LPC_ORDER + i - 2]) - + fMultDiv2(a1i, lowBandImag[LPC_ORDER + i - 2])) >> + dynamicScale; + accu2 = (fMultDiv2(a0i, lowBandReal[LPC_ORDER + i - 1]) + + fMultDiv2(a0r, lowBandImag[LPC_ORDER + i - 1]) + + fMultDiv2(a1i, lowBandReal[LPC_ORDER + i - 2]) + + fMultDiv2(a1r, lowBandImag[LPC_ORDER + i - 2])) >> + dynamicScale; + + accu1 = (lowBandReal[LPC_ORDER + i] >> descale) + (accu1 << 1); + accu2 = (lowBandImag[LPC_ORDER + i] >> descale) + (accu2 << 1); + if (fPreWhitening) { + accu1 = scaleValueSaturate( + fMultDiv2(accu1, preWhiteningGains[loBand]), + preWhiteningGains_exp[loBand] + 1); + accu2 = scaleValueSaturate( + fMultDiv2(accu2, preWhiteningGains[loBand]), + preWhiteningGains_exp[loBand] + 1); + } + qmfBufferReal[i][hiBand] = accu1; + qmfBufferImag[i][hiBand] = accu2; + } +#endif + } else { + FDK_ASSERT(dynamicScale >= 0); + calc_qmfBufferReal( + qmfBufferReal, &(lowBandReal[LPC_ORDER + startSample - 2]), + startSample, stopSample, hiBand, dynamicScale, + fMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale)), a0r, + a1r); + } + } /* bw <= 0 */ + + patch++; + + } /* inner loop over patches */ + + /* + * store the unmodified filter coefficients if there is + * an overlapping envelope + *****************************************************************/ + + } /* outer loop over bands (loBand) */ + + if (useLP) { + for (loBand = pSettings->lbStartPatching; + loBand < pSettings->lbStopPatching; loBand++) { + patch = 0; + while (patch < pSettings->noOfPatches) { + UCHAR hiBand = loBand + patchParam[patch].targetBandOffs; + + if (loBand < patchParam[patch].sourceStartBand || + loBand >= patchParam[patch].sourceStopBand || + hiBand >= (64) /* Highband out of range (biterror) */ + ) { + /* Lowband not in current patch or highband out of range (might be + * caused by biterrors)- proceed */ + patch++; + continue; + } + + if (hiBand != patchParam[patch].targetStartBand) + degreeAlias[hiBand] = degreeAlias[loBand]; + + patch++; + } + } /* end for loop */ + } + + for (i = 0; i < nInvfBands; i++) { + hLppTrans->bwVectorOld[i] = bwVector[i]; + } + + /* + set high band scale factor + */ + sbrScaleFactor->hb_scale = comLowBandScale - (LPC_SCALE_FACTOR); +} + +void lppTransposerHBE( + HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transposer */ + HANDLE_HBE_TRANSPOSER hQmfTransposer, + QMF_SCALE_FACTOR *sbrScaleFactor, /*!< Scaling factors */ + FIXP_DBL **qmfBufferReal, /*!< Pointer to pointer to real part of subband + samples (source) */ + FIXP_DBL **qmfBufferImag, /*!< Pointer to pointer to imaginary part of + subband samples (source) */ + const int timeStep, /*!< Time step of envelope */ + const int firstSlotOffs, /*!< Start position in time */ + const int lastSlotOffs, /*!< Number of overlap-slots into next frame */ + const int nInvfBands, /*!< Number of bands for inverse filtering */ + INVF_MODE *sbr_invf_mode, /*!< Current inverse filtering modes */ + INVF_MODE *sbr_invf_mode_prev /*!< Previous inverse filtering modes */ +) { + INT bwIndex; + FIXP_DBL bwVector[MAX_NUM_PATCHES_HBE]; /*!< pole moving factors */ + + int i; + int loBand, start, stop; + TRANSPOSER_SETTINGS *pSettings = hLppTrans->pSettings; + PATCH_PARAM *patchParam = pSettings->patchParam; + + FIXP_SGL alphar[LPC_ORDER], a0r, a1r; + FIXP_SGL alphai[LPC_ORDER], a0i = 0, a1i = 0; + FIXP_SGL bw = FL2FXCONST_SGL(0.0f); + + int autoCorrLength; + + ACORR_COEFS ac; + int startSample; + int stopSample; + int stopSampleClear; + + int comBandScale; + int ovLowBandShift; + int lowBandShift; + /* int ovHighBandShift;*/ + + alphai[0] = FL2FXCONST_SGL(0.0f); + alphai[1] = FL2FXCONST_SGL(0.0f); + + startSample = firstSlotOffs * timeStep; + stopSample = pSettings->nCols + lastSlotOffs * timeStep; + + inverseFilteringLevelEmphasis(hLppTrans, nInvfBands, sbr_invf_mode, + sbr_invf_mode_prev, bwVector); + + stopSampleClear = stopSample; + + autoCorrLength = pSettings->nCols + pSettings->overlap; + + if (pSettings->noOfPatches > 0) { + /* Set upper subbands to zero: + This is required in case that the patches do not cover the complete + highband (because the last patch would be too short). Possible + optimization: Clearing bands up to usb would be sufficient here. */ + int targetStopBand = + patchParam[pSettings->noOfPatches - 1].targetStartBand + + patchParam[pSettings->noOfPatches - 1].numBandsInPatch; + + int memSize = ((64) - targetStopBand) * sizeof(FIXP_DBL); + + for (i = startSample; i < stopSampleClear; i++) { + FDKmemclear(&qmfBufferReal[i][targetStopBand], memSize); + FDKmemclear(&qmfBufferImag[i][targetStopBand], memSize); + } + } +#ifdef __ANDROID__ + else { + // Safetynet logging + android_errorWriteLog(0x534e4554, "112160868"); + } +#endif + + /* + Calc common low band scale factor + */ + comBandScale = sbrScaleFactor->hb_scale; + + ovLowBandShift = sbrScaleFactor->hb_scale - comBandScale; + lowBandShift = sbrScaleFactor->hb_scale - comBandScale; + /* ovHighBandShift = firstSlotOffs == 0 ? ovLowBandShift:0;*/ + + /* outer loop over bands to do analysis only once for each band */ + + start = hQmfTransposer->startBand; + stop = hQmfTransposer->stopBand; + + for (loBand = start; loBand < stop; loBand++) { + bwIndex = 0; + + FIXP_DBL lowBandReal[(((1024) / (32) * (4) / 2) + (3 * (4))) + LPC_ORDER]; + FIXP_DBL lowBandImag[(((1024) / (32) * (4) / 2) + (3 * (4))) + LPC_ORDER]; + + int resetLPCCoeffs = 0; + int dynamicScale = DFRACT_BITS - 1 - LPC_SCALE_FACTOR; + int acDetScale = 0; /* scaling of autocorrelation determinant */ + + for (i = 0; i < LPC_ORDER; i++) { + lowBandReal[i] = hLppTrans->lpcFilterStatesRealHBE[i][loBand]; + lowBandImag[i] = hLppTrans->lpcFilterStatesImagHBE[i][loBand]; + } + + for (; i < LPC_ORDER + firstSlotOffs * timeStep; i++) { + lowBandReal[i] = hLppTrans->lpcFilterStatesRealHBE[i][loBand]; + lowBandImag[i] = hLppTrans->lpcFilterStatesImagHBE[i][loBand]; + } + + /* + Take old slope length qmf slot source values out of (overlap)qmf buffer + */ + for (i = firstSlotOffs * timeStep; + i < pSettings->nCols + pSettings->overlap; i++) { + lowBandReal[i + LPC_ORDER] = qmfBufferReal[i][loBand]; + lowBandImag[i + LPC_ORDER] = qmfBufferImag[i][loBand]; + } + + /* store unmodified values to buffer */ + for (i = 0; i < LPC_ORDER + pSettings->overlap; i++) { + hLppTrans->lpcFilterStatesRealHBE[i][loBand] = + qmfBufferReal[pSettings->nCols - LPC_ORDER + i][loBand]; + hLppTrans->lpcFilterStatesImagHBE[i][loBand] = + qmfBufferImag[pSettings->nCols - LPC_ORDER + i][loBand]; + } + + /* + Determine dynamic scaling value. + */ + dynamicScale = + fixMin(dynamicScale, + getScalefactor(lowBandReal, LPC_ORDER + pSettings->overlap) + + ovLowBandShift); + dynamicScale = + fixMin(dynamicScale, + getScalefactor(&lowBandReal[LPC_ORDER + pSettings->overlap], + pSettings->nCols) + + lowBandShift); + dynamicScale = + fixMin(dynamicScale, + getScalefactor(lowBandImag, LPC_ORDER + pSettings->overlap) + + ovLowBandShift); + dynamicScale = + fixMin(dynamicScale, + getScalefactor(&lowBandImag[LPC_ORDER + pSettings->overlap], + pSettings->nCols) + + lowBandShift); + + dynamicScale = fixMax( + 0, dynamicScale - 1); /* one additional bit headroom to prevent -1.0 */ + + /* + Scale temporal QMF buffer. + */ + scaleValues(&lowBandReal[0], LPC_ORDER + pSettings->overlap, + dynamicScale - ovLowBandShift); + scaleValues(&lowBandReal[LPC_ORDER + pSettings->overlap], pSettings->nCols, + dynamicScale - lowBandShift); + scaleValues(&lowBandImag[0], LPC_ORDER + pSettings->overlap, + dynamicScale - ovLowBandShift); + scaleValues(&lowBandImag[LPC_ORDER + pSettings->overlap], pSettings->nCols, + dynamicScale - lowBandShift); + + acDetScale += autoCorr2nd_cplx(&ac, lowBandReal + LPC_ORDER, + lowBandImag + LPC_ORDER, autoCorrLength); + + /* Examine dynamic of determinant in autocorrelation. */ + acDetScale += 2 * (comBandScale + dynamicScale); + acDetScale *= 2; /* two times reflection coefficent scaling */ + acDetScale += ac.det_scale; /* ac scaling of determinant */ + + /* In case of determinant < 10^-38, resetLPCCoeffs=1 has to be enforced. */ + if (acDetScale > 126) { + resetLPCCoeffs = 1; + } + + alphar[1] = FL2FXCONST_SGL(0.0f); + alphai[1] = FL2FXCONST_SGL(0.0f); + + if (ac.det != FL2FXCONST_DBL(0.0f)) { + FIXP_DBL tmp, absTmp, absDet; + + absDet = fixp_abs(ac.det); + + tmp = (fMultDiv2(ac.r01r, ac.r12r) >> (LPC_SCALE_FACTOR - 1)) - + ((fMultDiv2(ac.r01i, ac.r12i) + fMultDiv2(ac.r02r, ac.r11r)) >> + (LPC_SCALE_FACTOR - 1)); + absTmp = fixp_abs(tmp); + + /* + Quick check: is first filter coeff >= 1(4) + */ + { + INT scale; + FIXP_DBL result = fDivNorm(absTmp, absDet, &scale); + scale = scale + ac.det_scale; + + if ((scale > 0) && (result >= (FIXP_DBL)MAXVAL_DBL >> scale)) { + resetLPCCoeffs = 1; + } else { + alphar[1] = FX_DBL2FX_SGL(scaleValue(result, scale)); + if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) { + alphar[1] = -alphar[1]; + } + } + } + + tmp = (fMultDiv2(ac.r01i, ac.r12r) >> (LPC_SCALE_FACTOR - 1)) + + ((fMultDiv2(ac.r01r, ac.r12i) - + (FIXP_DBL)fMultDiv2(ac.r02i, ac.r11r)) >> + (LPC_SCALE_FACTOR - 1)); + + absTmp = fixp_abs(tmp); + + /* + Quick check: is second filter coeff >= 1(4) + */ + { + INT scale; + FIXP_DBL result = fDivNorm(absTmp, absDet, &scale); + scale = scale + ac.det_scale; + + if ((scale > 0) && + (result >= /*FL2FXCONST_DBL(1.f)*/ (FIXP_DBL)MAXVAL_DBL >> scale)) { + resetLPCCoeffs = 1; + } else { + alphai[1] = FX_DBL2FX_SGL(scaleValue(result, scale)); + if ((tmp < FL2FX_DBL(0.0f)) ^ (ac.det < FL2FX_DBL(0.0f))) { + alphai[1] = -alphai[1]; + } + } + } + } + + alphar[0] = FL2FXCONST_SGL(0.0f); + alphai[0] = FL2FXCONST_SGL(0.0f); + + if (ac.r11r != FL2FXCONST_DBL(0.0f)) { + /* ac.r11r is always >=0 */ + FIXP_DBL tmp, absTmp; + + tmp = (ac.r01r >> (LPC_SCALE_FACTOR + 1)) + + (fMultDiv2(alphar[1], ac.r12r) + fMultDiv2(alphai[1], ac.r12i)); + + absTmp = fixp_abs(tmp); + + /* + Quick check: is first filter coeff >= 1(4) + */ + + if (absTmp >= (ac.r11r >> 1)) { + resetLPCCoeffs = 1; + } else { + INT scale; + FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale); + alphar[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1)); + + if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f))) + alphar[0] = -alphar[0]; + } + + tmp = (ac.r01i >> (LPC_SCALE_FACTOR + 1)) + + (fMultDiv2(alphai[1], ac.r12r) - fMultDiv2(alphar[1], ac.r12i)); + + absTmp = fixp_abs(tmp); + + /* + Quick check: is second filter coeff >= 1(4) + */ + if (absTmp >= (ac.r11r >> 1)) { + resetLPCCoeffs = 1; + } else { + INT scale; + FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale); + alphai[0] = FX_DBL2FX_SGL(scaleValue(result, scale + 1)); + if ((tmp > FL2FX_DBL(0.0f)) ^ (ac.r11r < FL2FX_DBL(0.0f))) { + alphai[0] = -alphai[0]; + } + } + } + + /* Now check the quadratic criteria */ + if ((fMultDiv2(alphar[0], alphar[0]) + fMultDiv2(alphai[0], alphai[0])) >= + FL2FXCONST_DBL(0.5f)) { + resetLPCCoeffs = 1; + } + if ((fMultDiv2(alphar[1], alphar[1]) + fMultDiv2(alphai[1], alphai[1])) >= + FL2FXCONST_DBL(0.5f)) { + resetLPCCoeffs = 1; + } + + if (resetLPCCoeffs) { + alphar[0] = FL2FXCONST_SGL(0.0f); + alphar[1] = FL2FXCONST_SGL(0.0f); + alphai[0] = FL2FXCONST_SGL(0.0f); + alphai[1] = FL2FXCONST_SGL(0.0f); + } + + while (bwIndex < MAX_NUM_PATCHES - 1 && + loBand >= pSettings->bwBorders[bwIndex]) { + bwIndex++; + } + + /* + Filter Step 2: add the left slope with the current filter to the buffer + pure source values are already in there + */ + bw = FX_DBL2FX_SGL(bwVector[bwIndex]); + + a0r = FX_DBL2FX_SGL( + fMult(bw, alphar[0])); /* Apply current bandwidth expansion factor */ + a0i = FX_DBL2FX_SGL(fMult(bw, alphai[0])); + bw = FX_DBL2FX_SGL(fPow2(bw)); + a1r = FX_DBL2FX_SGL(fMult(bw, alphar[1])); + a1i = FX_DBL2FX_SGL(fMult(bw, alphai[1])); + + /* + Filter Step 3: insert the middle part which won't be windowed + */ + if (bw <= FL2FXCONST_SGL(0.0f)) { + int descale = fixMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale)); + for (i = startSample; i < stopSample; i++) { + qmfBufferReal[i][loBand] = lowBandReal[LPC_ORDER + i] >> descale; + qmfBufferImag[i][loBand] = lowBandImag[LPC_ORDER + i] >> descale; + } + } else { /* bw <= 0 */ + + int descale = fixMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale)); + + for (i = startSample; i < stopSample; i++) { + FIXP_DBL accu1, accu2; + + accu1 = (fMultDiv2(a0r, lowBandReal[LPC_ORDER + i - 1]) - + fMultDiv2(a0i, lowBandImag[LPC_ORDER + i - 1]) + + fMultDiv2(a1r, lowBandReal[LPC_ORDER + i - 2]) - + fMultDiv2(a1i, lowBandImag[LPC_ORDER + i - 2])) >> + dynamicScale; + accu2 = (fMultDiv2(a0i, lowBandReal[LPC_ORDER + i - 1]) + + fMultDiv2(a0r, lowBandImag[LPC_ORDER + i - 1]) + + fMultDiv2(a1i, lowBandReal[LPC_ORDER + i - 2]) + + fMultDiv2(a1r, lowBandImag[LPC_ORDER + i - 2])) >> + dynamicScale; + + qmfBufferReal[i][loBand] = + (lowBandReal[LPC_ORDER + i] >> descale) + (accu1 << 1); + qmfBufferImag[i][loBand] = + (lowBandImag[LPC_ORDER + i] >> descale) + (accu2 << 1); + } + } /* bw <= 0 */ + + /* + * store the unmodified filter coefficients if there is + * an overlapping envelope + *****************************************************************/ + + } /* outer loop over bands (loBand) */ + + for (i = 0; i < nInvfBands; i++) { + hLppTrans->bwVectorOld[i] = bwVector[i]; + } + + /* + set high band scale factor + */ + sbrScaleFactor->hb_scale = comBandScale - (LPC_SCALE_FACTOR); +} + +/*! + * + * \brief Initialize one low power transposer instance + * + * + */ +SBR_ERROR +createLppTransposer( + HANDLE_SBR_LPP_TRANS hs, /*!< Handle of low power transposer */ + TRANSPOSER_SETTINGS *pSettings, /*!< Pointer to settings */ + const int highBandStartSb, /*!< ? */ + UCHAR *v_k_master, /*!< Master table */ + const int numMaster, /*!< Valid entries in master table */ + const int usb, /*!< Highband area stop subband */ + const int timeSlots, /*!< Number of time slots */ + const int nCols, /*!< Number of colums (codec qmf bank) */ + UCHAR *noiseBandTable, /*!< Mapping of SBR noise bands to QMF bands */ + const int noNoiseBands, /*!< Number of noise bands */ + UINT fs, /*!< Sample Frequency */ + const int chan, /*!< Channel number */ + const int overlap) { + /* FB inverse filtering settings */ + hs->pSettings = pSettings; + + pSettings->nCols = nCols; + pSettings->overlap = overlap; + + switch (timeSlots) { + case 15: + case 16: + break; + + default: + return SBRDEC_UNSUPPORTED_CONFIG; /* Unimplemented */ + } + + if (chan == 0) { + /* Init common data only once */ + hs->pSettings->nCols = nCols; + + return resetLppTransposer(hs, highBandStartSb, v_k_master, numMaster, + noiseBandTable, noNoiseBands, usb, fs); + } + return SBRDEC_OK; +} + +static int findClosestEntry(UCHAR goalSb, UCHAR *v_k_master, UCHAR numMaster, + UCHAR direction) { + int index; + + if (goalSb <= v_k_master[0]) return v_k_master[0]; + + if (goalSb >= v_k_master[numMaster]) return v_k_master[numMaster]; + + if (direction) { + index = 0; + while (v_k_master[index] < goalSb) { + index++; + } + } else { + index = numMaster; + while (v_k_master[index] > goalSb) { + index--; + } + } + + return v_k_master[index]; +} + +/*! + * + * \brief Reset memory for one lpp transposer instance + * + * \return SBRDEC_OK on success, SBRDEC_UNSUPPORTED_CONFIG on error + */ +SBR_ERROR +resetLppTransposer( + HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transposer */ + UCHAR highBandStartSb, /*!< High band area: start subband */ + UCHAR *v_k_master, /*!< Master table */ + UCHAR numMaster, /*!< Valid entries in master table */ + UCHAR *noiseBandTable, /*!< Mapping of SBR noise bands to QMF bands */ + UCHAR noNoiseBands, /*!< Number of noise bands */ + UCHAR usb, /*!< High band area: stop subband */ + UINT fs /*!< SBR output sampling frequency */ +) { + TRANSPOSER_SETTINGS *pSettings = hLppTrans->pSettings; + PATCH_PARAM *patchParam = pSettings->patchParam; + + int i, patch; + int targetStopBand; + int sourceStartBand; + int patchDistance; + int numBandsInPatch; + + int lsb = v_k_master[0]; /* Start subband expressed in "non-critical" sampling + terms*/ + int xoverOffset = highBandStartSb - + lsb; /* Calculate distance in QMF bands between k0 and kx */ + int startFreqHz; + + int desiredBorder; + + usb = fixMin(usb, v_k_master[numMaster]); /* Avoid endless loops (compare with + float code). */ + + /* + * Plausibility check + */ + + if (pSettings->nCols == 64) { + if (lsb < 4) { + /* 4:1 SBR Requirement k0 >= 4 missed! */ + return SBRDEC_UNSUPPORTED_CONFIG; + } + } else if (lsb - SHIFT_START_SB < 4) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + + /* + * Initialize the patching parameter + */ + /* ISO/IEC 14496-3 (Figure 4.48): goalSb = round( 2.048e6 / fs ) */ + desiredBorder = (((2048000 * 2) / fs) + 1) >> 1; + + desiredBorder = findClosestEntry(desiredBorder, v_k_master, numMaster, + 1); /* Adapt region to master-table */ + + /* First patch */ + sourceStartBand = SHIFT_START_SB + xoverOffset; + targetStopBand = lsb + xoverOffset; /* upperBand */ + + /* Even (odd) numbered channel must be patched to even (odd) numbered channel + */ + patch = 0; + while (targetStopBand < usb) { + /* Too many patches? + Allow MAX_NUM_PATCHES+1 patches here. + we need to check later again, since patch might be the highest patch + AND contain less than 3 bands => actual number of patches will be reduced + by 1. + */ + if (patch > MAX_NUM_PATCHES) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + + patchParam[patch].guardStartBand = targetStopBand; + patchParam[patch].targetStartBand = targetStopBand; + + numBandsInPatch = + desiredBorder - targetStopBand; /* Get the desired range of the patch */ + + if (numBandsInPatch >= lsb - sourceStartBand) { + /* Desired number bands are not available -> patch whole source range */ + patchDistance = + targetStopBand - sourceStartBand; /* Get the targetOffset */ + patchDistance = + patchDistance & ~1; /* Rounding off odd numbers and make all even */ + numBandsInPatch = + lsb - (targetStopBand - + patchDistance); /* Update number of bands to be patched */ + numBandsInPatch = findClosestEntry(targetStopBand + numBandsInPatch, + v_k_master, numMaster, 0) - + targetStopBand; /* Adapt region to master-table */ + } + + if (pSettings->nCols == 64) { + if (numBandsInPatch == 0 && sourceStartBand == SHIFT_START_SB) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + } + + /* Desired number bands are available -> get the minimal even patching + * distance */ + patchDistance = + numBandsInPatch + targetStopBand - lsb; /* Get minimal distance */ + patchDistance = (patchDistance + 1) & + ~1; /* Rounding up odd numbers and make all even */ + + if (numBandsInPatch > 0) { + patchParam[patch].sourceStartBand = targetStopBand - patchDistance; + patchParam[patch].targetBandOffs = patchDistance; + patchParam[patch].numBandsInPatch = numBandsInPatch; + patchParam[patch].sourceStopBand = + patchParam[patch].sourceStartBand + numBandsInPatch; + + targetStopBand += patchParam[patch].numBandsInPatch; + patch++; + } + + /* All patches but first */ + sourceStartBand = SHIFT_START_SB; + + /* Check if we are close to desiredBorder */ + if (desiredBorder - targetStopBand < 3) /* MPEG doc */ + { + desiredBorder = usb; + } + } + + patch--; + + /* If highest patch contains less than three subband: skip it */ + if ((patch > 0) && (patchParam[patch].numBandsInPatch < 3)) { + patch--; + targetStopBand = + patchParam[patch].targetStartBand + patchParam[patch].numBandsInPatch; + } + + /* now check if we don't have one too many */ + if (patch >= MAX_NUM_PATCHES) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + + pSettings->noOfPatches = patch + 1; + + /* Check lowest and highest source subband */ + pSettings->lbStartPatching = targetStopBand; + pSettings->lbStopPatching = 0; + for (patch = 0; patch < pSettings->noOfPatches; patch++) { + pSettings->lbStartPatching = + fixMin(pSettings->lbStartPatching, patchParam[patch].sourceStartBand); + pSettings->lbStopPatching = + fixMax(pSettings->lbStopPatching, patchParam[patch].sourceStopBand); + } + + for (i = 0; i < noNoiseBands; i++) { + pSettings->bwBorders[i] = noiseBandTable[i + 1]; + } + for (; i < MAX_NUM_NOISE_VALUES; i++) { + pSettings->bwBorders[i] = 255; + } + + /* + * Choose whitening factors + */ + + startFreqHz = + ((lsb + xoverOffset) * fs) >> 7; /* Shift does a division by 2*(64) */ + + for (i = 1; i < NUM_WHFACTOR_TABLE_ENTRIES; i++) { + if (startFreqHz < FDK_sbrDecoder_sbr_whFactorsIndex[i]) break; + } + i--; + + pSettings->whFactors.off = FDK_sbrDecoder_sbr_whFactorsTable[i][0]; + pSettings->whFactors.transitionLevel = + FDK_sbrDecoder_sbr_whFactorsTable[i][1]; + pSettings->whFactors.lowLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][2]; + pSettings->whFactors.midLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][3]; + pSettings->whFactors.highLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][4]; + + return SBRDEC_OK; +} diff --git a/fdk-aac/libSBRdec/src/lpp_tran.h b/fdk-aac/libSBRdec/src/lpp_tran.h new file mode 100644 index 0000000..51b4395 --- /dev/null +++ b/fdk-aac/libSBRdec/src/lpp_tran.h @@ -0,0 +1,275 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief Low Power Profile Transposer +*/ + +#ifndef LPP_TRAN_H +#define LPP_TRAN_H + +#include "sbrdecoder.h" +#include "hbe.h" +#include "qmf.h" + +/* + Common +*/ +#define QMF_OUT_SCALE 8 + +/* + Frequency scales +*/ + +/* + Env-Adjust +*/ +#define MAX_NOISE_ENVELOPES 2 +#define MAX_NOISE_COEFFS 5 +#define MAX_NUM_NOISE_VALUES (MAX_NOISE_ENVELOPES * MAX_NOISE_COEFFS) +#define MAX_NUM_LIMITERS 12 + +/* Set MAX_ENVELOPES to the largest value of all supported BSFORMATs + by overriding MAX_ENVELOPES in the correct order: */ +#define MAX_ENVELOPES_LEGACY 5 +#define MAX_ENVELOPES_USAC 8 +#define MAX_ENVELOPES MAX_ENVELOPES_USAC + +#define MAX_FREQ_COEFFS_DUAL_RATE 48 +#define MAX_FREQ_COEFFS_QUAD_RATE 56 +#define MAX_FREQ_COEFFS MAX_FREQ_COEFFS_QUAD_RATE + +#define MAX_FREQ_COEFFS_FS44100 35 +#define MAX_FREQ_COEFFS_FS48000 32 + +#define MAX_NUM_ENVELOPE_VALUES (MAX_ENVELOPES * MAX_FREQ_COEFFS) + +#define MAX_GAIN_EXP 34 +/* Maximum gain will be sqrt(0.5 * 2^MAX_GAIN_EXP) + example: 34=99dB */ +#define MAX_GAIN_CONCEAL_EXP 1 +/* Maximum gain will be sqrt(0.5 * 2^MAX_GAIN_CONCEAL_EXP) in concealment case + * (0dB) */ + +/* + LPP Transposer +*/ +#define LPC_ORDER 2 + +#define MAX_INVF_BANDS MAX_NOISE_COEFFS + +#define MAX_NUM_PATCHES 6 +#define SHIFT_START_SB 1 /*!< lowest subband of source range */ + +typedef enum { + INVF_OFF = 0, + INVF_LOW_LEVEL, + INVF_MID_LEVEL, + INVF_HIGH_LEVEL, + INVF_SWITCHED /* not a real choice but used here to control behaviour */ +} INVF_MODE; + +/** parameter set for one single patch */ +typedef struct { + UCHAR sourceStartBand; /*!< first band in lowbands where to take the samples + from */ + UCHAR + sourceStopBand; /*!< first band in lowbands which is not included in the + patch anymore */ + UCHAR guardStartBand; /*!< first band in highbands to be filled with zeros in + order to reduce interferences between patches */ + UCHAR + targetStartBand; /*!< first band in highbands to be filled with whitened + lowband signal */ + UCHAR targetBandOffs; /*!< difference between 'startTargetBand' and + 'startSourceBand' */ + UCHAR numBandsInPatch; /*!< number of consecutive bands in this one patch */ +} PATCH_PARAM; + +/** whitening factors for different levels of whitening + need to be initialized corresponding to crossover frequency */ +typedef struct { + FIXP_DBL off; /*!< bw factor for signal OFF */ + FIXP_DBL transitionLevel; + FIXP_DBL lowLevel; /*!< bw factor for signal LOW_LEVEL */ + FIXP_DBL midLevel; /*!< bw factor for signal MID_LEVEL */ + FIXP_DBL highLevel; /*!< bw factor for signal HIGH_LEVEL */ +} WHITENING_FACTORS; + +/*! The transposer settings are calculated on a header reset and are shared by + * both channels. */ +typedef struct { + UCHAR nCols; /*!< number subsamples of a codec frame */ + UCHAR noOfPatches; /*!< number of patches */ + UCHAR lbStartPatching; /*!< first band of lowbands that will be patched */ + UCHAR lbStopPatching; /*!< first band that won't be patched anymore*/ + UCHAR bwBorders[MAX_NUM_NOISE_VALUES]; /*!< spectral bands with different + inverse filtering levels */ + + PATCH_PARAM + patchParam[MAX_NUM_PATCHES]; /*!< new parameter set for patching */ + WHITENING_FACTORS + whFactors; /*!< the pole moving factors for certain + whitening levels as indicated in the bitstream + depending on the crossover frequency */ + UCHAR overlap; /*!< Overlap size */ +} TRANSPOSER_SETTINGS; + +typedef struct { + TRANSPOSER_SETTINGS *pSettings; /*!< Common settings for both channels */ + FIXP_DBL + bwVectorOld[MAX_NUM_PATCHES]; /*!< pole moving factors of past frame */ + FIXP_DBL lpcFilterStatesRealLegSBR[LPC_ORDER + (3 * (4))][( + 32)]; /*!< pointer array to save filter states */ + + FIXP_DBL lpcFilterStatesImagLegSBR[LPC_ORDER + (3 * (4))][( + 32)]; /*!< pointer array to save filter states */ + + FIXP_DBL lpcFilterStatesRealHBE[LPC_ORDER + (3 * (4))][( + 64)]; /*!< pointer array to save filter states */ + FIXP_DBL lpcFilterStatesImagHBE[LPC_ORDER + (3 * (4))][( + 64)]; /*!< pointer array to save filter states */ +} SBR_LPP_TRANS; + +typedef SBR_LPP_TRANS *HANDLE_SBR_LPP_TRANS; + +void lppTransposer(HANDLE_SBR_LPP_TRANS hLppTrans, + QMF_SCALE_FACTOR *sbrScaleFactor, FIXP_DBL **qmfBufferReal, + + FIXP_DBL *degreeAlias, FIXP_DBL **qmfBufferImag, + const int useLP, const int fPreWhitening, + const int v_k_master0, const int timeStep, + const int firstSlotOffset, const int lastSlotOffset, + const int nInvfBands, INVF_MODE *sbr_invf_mode, + INVF_MODE *sbr_invf_mode_prev); + +void lppTransposerHBE( + HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transposer */ + HANDLE_HBE_TRANSPOSER hQmfTransposer, + QMF_SCALE_FACTOR *sbrScaleFactor, /*!< Scaling factors */ + FIXP_DBL **qmfBufferReal, /*!< Pointer to pointer to real part of subband + samples (source) */ + FIXP_DBL **qmfBufferImag, /*!< Pointer to pointer to imaginary part of + subband samples (source) */ + const int timeStep, /*!< Time step of envelope */ + const int firstSlotOffs, /*!< Start position in time */ + const int lastSlotOffs, /*!< Number of overlap-slots into next frame */ + const int nInvfBands, /*!< Number of bands for inverse filtering */ + INVF_MODE *sbr_invf_mode, /*!< Current inverse filtering modes */ + INVF_MODE *sbr_invf_mode_prev /*!< Previous inverse filtering modes */ +); + +SBR_ERROR +createLppTransposer(HANDLE_SBR_LPP_TRANS hLppTrans, + TRANSPOSER_SETTINGS *pSettings, const int highBandStartSb, + UCHAR *v_k_master, const int numMaster, const int usb, + const int timeSlots, const int nCols, UCHAR *noiseBandTable, + const int noNoiseBands, UINT fs, const int chan, + const int overlap); + +SBR_ERROR +resetLppTransposer(HANDLE_SBR_LPP_TRANS hLppTrans, UCHAR highBandStartSb, + UCHAR *v_k_master, UCHAR numMaster, UCHAR *noiseBandTable, + UCHAR noNoiseBands, UCHAR usb, UINT fs); + +#endif /* LPP_TRAN_H */ diff --git a/fdk-aac/libSBRdec/src/psbitdec.cpp b/fdk-aac/libSBRdec/src/psbitdec.cpp new file mode 100644 index 0000000..82bb65b --- /dev/null +++ b/fdk-aac/libSBRdec/src/psbitdec.cpp @@ -0,0 +1,594 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +#include "psbitdec.h" + +#include "sbr_rom.h" +#include "huff_dec.h" + +/* PS dec privat functions */ +SBR_ERROR ResetPsDec(HANDLE_PS_DEC h_ps_d); + +/***************************************************************************/ +/*! + \brief huffman decoding by codebook table + + \return index of huffman codebook table + +****************************************************************************/ +static SCHAR decode_huff_cw( + Huffman h, /*!< pointer to huffman codebook table */ + HANDLE_FDK_BITSTREAM hBitBuf, /*!< Handle to Bitbuffer */ + int *length) /*!< length of huffman codeword (or NULL) */ +{ + UCHAR bit = 0; + SCHAR index = 0; + UCHAR bitCount = 0; + + while (index >= 0) { + bit = FDKreadBits(hBitBuf, 1); + bitCount++; + index = h[index][bit]; + } + if (length) { + *length = bitCount; + } + return (index + 64); /* Add offset */ +} + +/***************************************************************************/ +/*! + \brief helper function - limiting of value to min/max values + + \return limited value + +****************************************************************************/ + +static SCHAR limitMinMax(SCHAR i, SCHAR min, SCHAR max) { + if (i < min) + return min; + else if (i > max) + return max; + else + return i; +} + +/***************************************************************************/ +/*! + \brief Decodes delta values in-place and updates + data buffers according to quantization classes. + + When delta coded in frequency the first element is deltacode from zero. + aIndex buffer is decoded from delta values to actual values. + + \return none + +****************************************************************************/ +static void deltaDecodeArray( + SCHAR enable, SCHAR *aIndex, /*!< ICC/IID parameters */ + SCHAR *aPrevFrameIndex, /*!< ICC/IID parameters of previous frame */ + SCHAR DtDf, UCHAR nrElements, /*!< as conveyed in bitstream */ + /*!< output array size: nrElements*stride */ + UCHAR stride, /*!< 1=dflt, 2=half freq. resolution */ + SCHAR minIdx, SCHAR maxIdx) { + int i; + + /* Delta decode */ + if (enable == 1) { + if (DtDf == 0) { /* Delta coded in freq */ + aIndex[0] = 0 + aIndex[0]; + aIndex[0] = limitMinMax(aIndex[0], minIdx, maxIdx); + for (i = 1; i < nrElements; i++) { + aIndex[i] = aIndex[i - 1] + aIndex[i]; + aIndex[i] = limitMinMax(aIndex[i], minIdx, maxIdx); + } + } else { /* Delta time */ + for (i = 0; i < nrElements; i++) { + aIndex[i] = aPrevFrameIndex[i * stride] + aIndex[i]; + aIndex[i] = limitMinMax(aIndex[i], minIdx, maxIdx); + } + } + } else { /* No data is sent, set index to zero */ + for (i = 0; i < nrElements; i++) { + aIndex[i] = 0; + } + } + if (stride == 2) { + for (i = nrElements * stride - 1; i > 0; i--) { + aIndex[i] = aIndex[i >> 1]; + } + } +} + +/***************************************************************************/ +/*! + \brief Mapping of ICC/IID parameters to 20 stereo bands + + \return none + +****************************************************************************/ +static void map34IndexTo20(SCHAR *aIndex, /*!< decoded ICC/IID parameters */ + UCHAR noBins) /*!< number of stereo bands */ +{ + aIndex[0] = (2 * aIndex[0] + aIndex[1]) / 3; + aIndex[1] = (aIndex[1] + 2 * aIndex[2]) / 3; + aIndex[2] = (2 * aIndex[3] + aIndex[4]) / 3; + aIndex[3] = (aIndex[4] + 2 * aIndex[5]) / 3; + aIndex[4] = (aIndex[6] + aIndex[7]) / 2; + aIndex[5] = (aIndex[8] + aIndex[9]) / 2; + aIndex[6] = aIndex[10]; + aIndex[7] = aIndex[11]; + aIndex[8] = (aIndex[12] + aIndex[13]) / 2; + aIndex[9] = (aIndex[14] + aIndex[15]) / 2; + aIndex[10] = aIndex[16]; + /* For IPD/OPD it stops here */ + + if (noBins == NO_HI_RES_BINS) { + aIndex[11] = aIndex[17]; + aIndex[12] = aIndex[18]; + aIndex[13] = aIndex[19]; + aIndex[14] = (aIndex[20] + aIndex[21]) / 2; + aIndex[15] = (aIndex[22] + aIndex[23]) / 2; + aIndex[16] = (aIndex[24] + aIndex[25]) / 2; + aIndex[17] = (aIndex[26] + aIndex[27]) / 2; + aIndex[18] = (aIndex[28] + aIndex[29] + aIndex[30] + aIndex[31]) / 4; + aIndex[19] = (aIndex[32] + aIndex[33]) / 2; + } +} + +/***************************************************************************/ +/*! + \brief Decodes delta coded IID, ICC, IPD and OPD indices + + \return PS processing flag. If set to 1 + +****************************************************************************/ +int DecodePs(struct PS_DEC *h_ps_d, /*!< PS handle */ + const UCHAR frameError, /*!< Flag telling that frame had errors */ + PS_DEC_COEFFICIENTS *pScratch) { + MPEG_PS_BS_DATA *pBsData; + UCHAR gr, env; + int bPsHeaderValid, bPsDataAvail; + + /* Assign Scratch */ + h_ps_d->specificTo.mpeg.pCoef = pScratch; + + /* Shortcuts to avoid deferencing and keep the code readable */ + pBsData = &h_ps_d->bsData[h_ps_d->processSlot].mpeg; + bPsHeaderValid = pBsData->bPsHeaderValid; + bPsDataAvail = + (h_ps_d->bPsDataAvail[h_ps_d->processSlot] == ppt_mpeg) ? 1 : 0; + + /*************************************************************************************** + * Decide whether to process or to conceal PS data or not. */ + + if ((h_ps_d->psDecodedPrv && !frameError && !bPsDataAvail) || + (!h_ps_d->psDecodedPrv && + (frameError || !bPsDataAvail || !bPsHeaderValid))) { + /* Don't apply PS processing. + * Declare current PS header and bitstream data invalid. */ + pBsData->bPsHeaderValid = 0; + h_ps_d->bPsDataAvail[h_ps_d->processSlot] = ppt_none; + return (0); + } + + if (frameError || + !bPsHeaderValid) { /* no new PS data available (e.g. frame loss) */ + /* => keep latest data constant (i.e. FIX with noEnv=0) */ + pBsData->noEnv = 0; + } + + /*************************************************************************************** + * Decode bitstream payload or prepare parameter for concealment: + */ + for (env = 0; env < pBsData->noEnv; env++) { + SCHAR *aPrevIidIndex; + SCHAR *aPrevIccIndex; + + UCHAR noIidSteps = pBsData->bFineIidQ ? NO_IID_STEPS_FINE : NO_IID_STEPS; + + if (env == 0) { + aPrevIidIndex = h_ps_d->specificTo.mpeg.aIidPrevFrameIndex; + aPrevIccIndex = h_ps_d->specificTo.mpeg.aIccPrevFrameIndex; + } else { + aPrevIidIndex = pBsData->aaIidIndex[env - 1]; + aPrevIccIndex = pBsData->aaIccIndex[env - 1]; + } + + deltaDecodeArray(pBsData->bEnableIid, pBsData->aaIidIndex[env], + aPrevIidIndex, pBsData->abIidDtFlag[env], + FDK_sbrDecoder_aNoIidBins[pBsData->freqResIid], + (pBsData->freqResIid) ? 1 : 2, -noIidSteps, noIidSteps); + + deltaDecodeArray(pBsData->bEnableIcc, pBsData->aaIccIndex[env], + aPrevIccIndex, pBsData->abIccDtFlag[env], + FDK_sbrDecoder_aNoIccBins[pBsData->freqResIcc], + (pBsData->freqResIcc) ? 1 : 2, 0, NO_ICC_STEPS - 1); + } /* for (env=0; env<pBsData->noEnv; env++) */ + + /* handling of FIX noEnv=0 */ + if (pBsData->noEnv == 0) { + /* set noEnv=1, keep last parameters or force 0 if not enabled */ + pBsData->noEnv = 1; + + if (pBsData->bEnableIid) { + pBsData->bFineIidQ = h_ps_d->specificTo.mpeg.bPrevFrameFineIidQ; + pBsData->freqResIid = h_ps_d->specificTo.mpeg.prevFreqResIid; + for (gr = 0; gr < NO_HI_RES_IID_BINS; gr++) { + pBsData->aaIidIndex[pBsData->noEnv - 1][gr] = + h_ps_d->specificTo.mpeg.aIidPrevFrameIndex[gr]; + } + } else { + for (gr = 0; gr < NO_HI_RES_IID_BINS; gr++) { + pBsData->aaIidIndex[pBsData->noEnv - 1][gr] = 0; + } + } + + if (pBsData->bEnableIcc) { + pBsData->freqResIcc = h_ps_d->specificTo.mpeg.prevFreqResIcc; + for (gr = 0; gr < NO_HI_RES_ICC_BINS; gr++) { + pBsData->aaIccIndex[pBsData->noEnv - 1][gr] = + h_ps_d->specificTo.mpeg.aIccPrevFrameIndex[gr]; + } + } else { + for (gr = 0; gr < NO_HI_RES_ICC_BINS; gr++) { + pBsData->aaIccIndex[pBsData->noEnv - 1][gr] = 0; + } + } + } + + /* Update previous frame Iid quantization */ + h_ps_d->specificTo.mpeg.bPrevFrameFineIidQ = pBsData->bFineIidQ; + + /* Update previous frequency resolution for IID */ + h_ps_d->specificTo.mpeg.prevFreqResIid = pBsData->freqResIid; + + /* Update previous frequency resolution for ICC */ + h_ps_d->specificTo.mpeg.prevFreqResIcc = pBsData->freqResIcc; + + /* Update previous frame index buffers */ + for (gr = 0; gr < NO_HI_RES_IID_BINS; gr++) { + h_ps_d->specificTo.mpeg.aIidPrevFrameIndex[gr] = + pBsData->aaIidIndex[pBsData->noEnv - 1][gr]; + } + for (gr = 0; gr < NO_HI_RES_ICC_BINS; gr++) { + h_ps_d->specificTo.mpeg.aIccPrevFrameIndex[gr] = + pBsData->aaIccIndex[pBsData->noEnv - 1][gr]; + } + + /* PS data from bitstream (if avail) was decoded now */ + h_ps_d->bPsDataAvail[h_ps_d->processSlot] = ppt_none; + + /* handling of env borders for FIX & VAR */ + if (pBsData->bFrameClass == 0) { + /* FIX_BORDERS NoEnv=0,1,2,4 */ + pBsData->aEnvStartStop[0] = 0; + for (env = 1; env < pBsData->noEnv; env++) { + pBsData->aEnvStartStop[env] = + (env * h_ps_d->noSubSamples) / pBsData->noEnv; + } + pBsData->aEnvStartStop[pBsData->noEnv] = h_ps_d->noSubSamples; + /* 1024 (32 slots) env borders: 0, 8, 16, 24, 32 */ + /* 960 (30 slots) env borders: 0, 7, 15, 22, 30 */ + } else { /* if (h_ps_d->bFrameClass == 0) */ + /* VAR_BORDERS NoEnv=1,2,3,4 */ + pBsData->aEnvStartStop[0] = 0; + + /* handle case aEnvStartStop[noEnv]<noSubSample for VAR_BORDERS by + duplicating last PS parameters and incrementing noEnv */ + if (pBsData->aEnvStartStop[pBsData->noEnv] < h_ps_d->noSubSamples) { + for (gr = 0; gr < NO_HI_RES_IID_BINS; gr++) { + pBsData->aaIidIndex[pBsData->noEnv][gr] = + pBsData->aaIidIndex[pBsData->noEnv - 1][gr]; + } + for (gr = 0; gr < NO_HI_RES_ICC_BINS; gr++) { + pBsData->aaIccIndex[pBsData->noEnv][gr] = + pBsData->aaIccIndex[pBsData->noEnv - 1][gr]; + } + pBsData->noEnv++; + pBsData->aEnvStartStop[pBsData->noEnv] = h_ps_d->noSubSamples; + } + + /* enforce strictly monotonic increasing borders */ + for (env = 1; env < pBsData->noEnv; env++) { + UCHAR thr; + thr = (UCHAR)h_ps_d->noSubSamples - (pBsData->noEnv - env); + if (pBsData->aEnvStartStop[env] > thr) { + pBsData->aEnvStartStop[env] = thr; + } else { + thr = pBsData->aEnvStartStop[env - 1] + 1; + if (pBsData->aEnvStartStop[env] < thr) { + pBsData->aEnvStartStop[env] = thr; + } + } + } + } /* if (h_ps_d->bFrameClass == 0) ... else */ + + /* copy data prior to possible 20<->34 in-place mapping */ + for (env = 0; env < pBsData->noEnv; env++) { + UCHAR i; + for (i = 0; i < NO_HI_RES_IID_BINS; i++) { + h_ps_d->specificTo.mpeg.pCoef->aaIidIndexMapped[env][i] = + pBsData->aaIidIndex[env][i]; + } + for (i = 0; i < NO_HI_RES_ICC_BINS; i++) { + h_ps_d->specificTo.mpeg.pCoef->aaIccIndexMapped[env][i] = + pBsData->aaIccIndex[env][i]; + } + } + + /* MPEG baseline PS */ + /* Baseline version of PS always uses the hybrid filter structure with 20 + * stereo bands. */ + /* If ICC/IID parameters for 34 stereo bands are decoded they have to be + * mapped to 20 */ + /* stereo bands. */ + /* Additionaly the IPD/OPD parameters won't be used. */ + + for (env = 0; env < pBsData->noEnv; env++) { + if (pBsData->freqResIid == 2) + map34IndexTo20(h_ps_d->specificTo.mpeg.pCoef->aaIidIndexMapped[env], + NO_HI_RES_IID_BINS); + if (pBsData->freqResIcc == 2) + map34IndexTo20(h_ps_d->specificTo.mpeg.pCoef->aaIccIndexMapped[env], + NO_HI_RES_ICC_BINS); + + /* IPD/OPD is disabled in baseline version and thus was removed here */ + } + + return (1); +} + +/***************************************************************************/ +/*! + + \brief Reads parametric stereo data from bitstream + + \return + +****************************************************************************/ +unsigned int ReadPsData( + HANDLE_PS_DEC h_ps_d, /*!< handle to struct PS_DEC */ + HANDLE_FDK_BITSTREAM hBitBuf, /*!< handle to struct BIT_BUF */ + int nBitsLeft /*!< max number of bits available */ +) { + MPEG_PS_BS_DATA *pBsData; + + UCHAR gr, env; + SCHAR dtFlag; + INT startbits; + Huffman CurrentTable; + SCHAR bEnableHeader; + + if (!h_ps_d) return 0; + + pBsData = &h_ps_d->bsData[h_ps_d->bsReadSlot].mpeg; + + if (h_ps_d->bsReadSlot != h_ps_d->bsLastSlot) { + /* Copy last header data */ + FDKmemcpy(pBsData, &h_ps_d->bsData[h_ps_d->bsLastSlot].mpeg, + sizeof(MPEG_PS_BS_DATA)); + } + + startbits = (INT)FDKgetValidBits(hBitBuf); + + bEnableHeader = (SCHAR)FDKreadBits(hBitBuf, 1); + + /* Read header */ + if (bEnableHeader) { + pBsData->bPsHeaderValid = 1; + pBsData->bEnableIid = (UCHAR)FDKreadBits(hBitBuf, 1); + if (pBsData->bEnableIid) { + pBsData->modeIid = (UCHAR)FDKreadBits(hBitBuf, 3); + } + + pBsData->bEnableIcc = (UCHAR)FDKreadBits(hBitBuf, 1); + if (pBsData->bEnableIcc) { + pBsData->modeIcc = (UCHAR)FDKreadBits(hBitBuf, 3); + } + + pBsData->bEnableExt = (UCHAR)FDKreadBits(hBitBuf, 1); + } + + pBsData->bFrameClass = (UCHAR)FDKreadBits(hBitBuf, 1); + if (pBsData->bFrameClass == 0) { + /* FIX_BORDERS NoEnv=0,1,2,4 */ + pBsData->noEnv = + FDK_sbrDecoder_aFixNoEnvDecode[(UCHAR)FDKreadBits(hBitBuf, 2)]; + /* all additional handling of env borders is now in DecodePs() */ + } else { + /* VAR_BORDERS NoEnv=1,2,3,4 */ + pBsData->noEnv = 1 + (UCHAR)FDKreadBits(hBitBuf, 2); + for (env = 1; env < pBsData->noEnv + 1; env++) + pBsData->aEnvStartStop[env] = ((UCHAR)FDKreadBits(hBitBuf, 5)) + 1; + /* all additional handling of env borders is now in DecodePs() */ + } + + /* verify that IID & ICC modes (quant grid, freq res) are supported */ + if ((pBsData->modeIid > 5) || (pBsData->modeIcc > 5)) { + /* no useful PS data could be read from bitstream */ + h_ps_d->bPsDataAvail[h_ps_d->bsReadSlot] = ppt_none; + /* discard all remaining bits */ + nBitsLeft -= startbits - (INT)FDKgetValidBits(hBitBuf); + while (nBitsLeft > 0) { + int i = nBitsLeft; + if (i > 8) { + i = 8; + } + FDKreadBits(hBitBuf, i); + nBitsLeft -= i; + } + return (UINT)(startbits - (INT)FDKgetValidBits(hBitBuf)); + } + + if (pBsData->modeIid > 2) { + pBsData->freqResIid = pBsData->modeIid - 3; + pBsData->bFineIidQ = 1; + } else { + pBsData->freqResIid = pBsData->modeIid; + pBsData->bFineIidQ = 0; + } + + if (pBsData->modeIcc > 2) { + pBsData->freqResIcc = pBsData->modeIcc - 3; + } else { + pBsData->freqResIcc = pBsData->modeIcc; + } + + /* Extract IID data */ + if (pBsData->bEnableIid) { + for (env = 0; env < pBsData->noEnv; env++) { + dtFlag = (SCHAR)FDKreadBits(hBitBuf, 1); + if (!dtFlag) { + if (pBsData->bFineIidQ) + CurrentTable = (Huffman)&aBookPsIidFineFreqDecode; + else + CurrentTable = (Huffman)&aBookPsIidFreqDecode; + } else { + if (pBsData->bFineIidQ) + CurrentTable = (Huffman)&aBookPsIidFineTimeDecode; + else + CurrentTable = (Huffman)&aBookPsIidTimeDecode; + } + + for (gr = 0; gr < FDK_sbrDecoder_aNoIidBins[pBsData->freqResIid]; gr++) + pBsData->aaIidIndex[env][gr] = + decode_huff_cw(CurrentTable, hBitBuf, NULL); + pBsData->abIidDtFlag[env] = dtFlag; + } + } + + /* Extract ICC data */ + if (pBsData->bEnableIcc) { + for (env = 0; env < pBsData->noEnv; env++) { + dtFlag = (SCHAR)FDKreadBits(hBitBuf, 1); + if (!dtFlag) + CurrentTable = (Huffman)&aBookPsIccFreqDecode; + else + CurrentTable = (Huffman)&aBookPsIccTimeDecode; + + for (gr = 0; gr < FDK_sbrDecoder_aNoIccBins[pBsData->freqResIcc]; gr++) + pBsData->aaIccIndex[env][gr] = + decode_huff_cw(CurrentTable, hBitBuf, NULL); + pBsData->abIccDtFlag[env] = dtFlag; + } + } + + if (pBsData->bEnableExt) { + /*! + Decoders that support only the baseline version of the PS tool are allowed + to ignore the IPD/OPD data, but according header data has to be parsed. + ISO/IEC 14496-3 Subpart 8 Annex 4 + */ + + int cnt = FDKreadBits(hBitBuf, PS_EXTENSION_SIZE_BITS); + if (cnt == (1 << PS_EXTENSION_SIZE_BITS) - 1) { + cnt += FDKreadBits(hBitBuf, PS_EXTENSION_ESC_COUNT_BITS); + } + while (cnt--) FDKreadBits(hBitBuf, 8); + } + + /* new PS data was read from bitstream */ + h_ps_d->bPsDataAvail[h_ps_d->bsReadSlot] = ppt_mpeg; + + return (startbits - (INT)FDKgetValidBits(hBitBuf)); +} diff --git a/fdk-aac/libSBRdec/src/psbitdec.h b/fdk-aac/libSBRdec/src/psbitdec.h new file mode 100644 index 0000000..f0fc43a --- /dev/null +++ b/fdk-aac/libSBRdec/src/psbitdec.h @@ -0,0 +1,116 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +#ifndef PSBITDEC_H +#define PSBITDEC_H + +#include "sbrdecoder.h" + +#include "psdec.h" + +unsigned int ReadPsData(struct PS_DEC *h_ps_d, HANDLE_FDK_BITSTREAM hBs, + int nBitsLeft); + +int DecodePs(struct PS_DEC *h_ps_d, const UCHAR frameError, + PS_DEC_COEFFICIENTS *pCoef); + +#endif /* PSBITDEC_H */ diff --git a/fdk-aac/libSBRdec/src/psdec.cpp b/fdk-aac/libSBRdec/src/psdec.cpp new file mode 100644 index 0000000..b31b310 --- /dev/null +++ b/fdk-aac/libSBRdec/src/psdec.cpp @@ -0,0 +1,722 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief parametric stereo decoder +*/ + +#include "psdec.h" + +#include "FDK_bitbuffer.h" + +#include "sbr_rom.h" +#include "sbr_ram.h" + +#include "FDK_tools_rom.h" + +#include "genericStds.h" + +#include "FDK_trigFcts.h" + +/********************************************************************/ +/* MLQUAL DEFINES */ +/********************************************************************/ + +#define FRACT_ZERO FRACT_BITS - 1 +/********************************************************************/ + +SBR_ERROR ResetPsDec(HANDLE_PS_DEC h_ps_d); + +/***** HELPERS *****/ + +/***************************************************************************/ +/*! + \brief Creates one instance of the PS_DEC struct + + \return Error info + +****************************************************************************/ +int CreatePsDec(HANDLE_PS_DEC *h_PS_DEC, /*!< pointer to the module state */ + int aacSamplesPerFrame) { + SBR_ERROR errorInfo = SBRDEC_OK; + HANDLE_PS_DEC h_ps_d; + int i; + + if (*h_PS_DEC == NULL) { + /* Get ps dec ram */ + h_ps_d = GetRam_ps_dec(); + if (h_ps_d == NULL) { + goto bail; + } + } else { + /* Reset an open instance */ + h_ps_d = *h_PS_DEC; + } + + /* + * Create Analysis Hybrid filterbank. + */ + FDKhybridAnalysisOpen(&h_ps_d->specificTo.mpeg.hybridAnalysis, + h_ps_d->specificTo.mpeg.pHybridAnaStatesLFdmx, + sizeof(h_ps_d->specificTo.mpeg.pHybridAnaStatesLFdmx), + NULL, 0); + + /* initialisation */ + switch (aacSamplesPerFrame) { + case 960: + h_ps_d->noSubSamples = 30; /* col */ + break; + case 1024: + h_ps_d->noSubSamples = 32; /* col */ + break; + default: + h_ps_d->noSubSamples = -1; + break; + } + + if (h_ps_d->noSubSamples > MAX_NUM_COL || h_ps_d->noSubSamples <= 0) { + goto bail; + } + h_ps_d->noChannels = NO_QMF_CHANNELS; /* row */ + + h_ps_d->psDecodedPrv = 0; + h_ps_d->procFrameBased = -1; + for (i = 0; i < (1) + 1; i++) { + h_ps_d->bPsDataAvail[i] = ppt_none; + } + { + int error; + error = FDKdecorrelateOpen(&(h_ps_d->specificTo.mpeg.apDecor), + h_ps_d->specificTo.mpeg.decorrBufferCplx, + (2 * ((825) + (373)))); + if (error) goto bail; + } + + for (i = 0; i < (1) + 1; i++) { + FDKmemclear(&h_ps_d->bsData[i].mpeg, sizeof(MPEG_PS_BS_DATA)); + } + + errorInfo = ResetPsDec(h_ps_d); + + if (errorInfo != SBRDEC_OK) goto bail; + + *h_PS_DEC = h_ps_d; + + return 0; + +bail: + if (h_ps_d != NULL) { + DeletePsDec(&h_ps_d); + } + + return -1; +} /*END CreatePsDec */ + +/***************************************************************************/ +/*! + \brief Delete one instance of the PS_DEC struct + + \return Error info + +****************************************************************************/ +int DeletePsDec(HANDLE_PS_DEC *h_PS_DEC) /*!< pointer to the module state */ +{ + if (*h_PS_DEC == NULL) { + return -1; + } + + { + HANDLE_PS_DEC h_ps_d = *h_PS_DEC; + FDKdecorrelateClose(&(h_ps_d->specificTo.mpeg.apDecor)); + } + + FreeRam_ps_dec(h_PS_DEC); + + return 0; +} /*END DeletePsDec */ + +/***************************************************************************/ +/*! + \brief resets some values of the PS handle to default states + + \return + +****************************************************************************/ +SBR_ERROR ResetPsDec(HANDLE_PS_DEC h_ps_d) /*!< pointer to the module state */ +{ + SBR_ERROR errorInfo = SBRDEC_OK; + INT i; + + /* explicitly init state variables to safe values (until first ps header + * arrives) */ + + h_ps_d->specificTo.mpeg.lastUsb = 0; + + /* + * Initialize Analysis Hybrid filterbank. + */ + FDKhybridAnalysisInit(&h_ps_d->specificTo.mpeg.hybridAnalysis, THREE_TO_TEN, + NO_QMF_BANDS_HYBRID20, NO_QMF_BANDS_HYBRID20, 1); + + /* + * Initialize Synthesis Hybrid filterbank. + */ + for (i = 0; i < 2; i++) { + FDKhybridSynthesisInit(&h_ps_d->specificTo.mpeg.hybridSynthesis[i], + THREE_TO_TEN, NO_QMF_CHANNELS, NO_QMF_CHANNELS); + } + { + INT error; + error = FDKdecorrelateInit(&h_ps_d->specificTo.mpeg.apDecor, 71, DECORR_PS, + DUCKER_AUTOMATIC, 0, 0, 0, 0, 1, /* isLegacyPS */ + 1); + if (error) return SBRDEC_NOT_INITIALIZED; + } + + for (i = 0; i < NO_IID_GROUPS; i++) { + h_ps_d->specificTo.mpeg.h11rPrev[i] = FL2FXCONST_DBL(0.5f); + h_ps_d->specificTo.mpeg.h12rPrev[i] = FL2FXCONST_DBL(0.5f); + } + + FDKmemclear(h_ps_d->specificTo.mpeg.h21rPrev, + sizeof(h_ps_d->specificTo.mpeg.h21rPrev)); + FDKmemclear(h_ps_d->specificTo.mpeg.h22rPrev, + sizeof(h_ps_d->specificTo.mpeg.h22rPrev)); + + return errorInfo; +} + +/***************************************************************************/ +/*! + \brief Feed delaylines when parametric stereo is switched on. + \return +****************************************************************************/ +void PreparePsProcessing(HANDLE_PS_DEC h_ps_d, + const FIXP_DBL *const *const rIntBufferLeft, + const FIXP_DBL *const *const iIntBufferLeft, + const int scaleFactorLowBand) { + if (h_ps_d->procFrameBased == + 1) /* If we have switched from frame to slot based processing */ + { /* fill hybrid delay buffer. */ + int i, j; + + for (i = 0; i < HYBRID_FILTER_DELAY; i++) { + FIXP_DBL qmfInputData[2][NO_QMF_BANDS_HYBRID20]; + FIXP_DBL hybridOutputData[2][NO_SUB_QMF_CHANNELS]; + + for (j = 0; j < NO_QMF_BANDS_HYBRID20; j++) { + qmfInputData[0][j] = + scaleValue(rIntBufferLeft[i][j], scaleFactorLowBand); + qmfInputData[1][j] = + scaleValue(iIntBufferLeft[i][j], scaleFactorLowBand); + } + + FDKhybridAnalysisApply(&h_ps_d->specificTo.mpeg.hybridAnalysis, + qmfInputData[0], qmfInputData[1], + hybridOutputData[0], hybridOutputData[1]); + } + h_ps_d->procFrameBased = 0; /* switch to slot based processing. */ + + } /* procFrameBased==1 */ +} + +void initSlotBasedRotation( + HANDLE_PS_DEC h_ps_d, /*!< pointer to the module state */ + int env, int usb) { + INT group = 0; + INT bin = 0; + INT noIidSteps, noFactors; + + FIXP_SGL invL; + FIXP_DBL ScaleL, ScaleR; + FIXP_DBL Alpha, Beta, AlphasValue; + FIXP_DBL h11r, h12r, h21r, h22r; + + const FIXP_DBL *PScaleFactors; + + if (h_ps_d->bsData[h_ps_d->processSlot].mpeg.bFineIidQ) { + PScaleFactors = ScaleFactorsFine; /* values are shiftet right by one */ + noIidSteps = NO_IID_STEPS_FINE; + noFactors = NO_IID_LEVELS_FINE; + } else { + PScaleFactors = ScaleFactors; /* values are shiftet right by one */ + noIidSteps = NO_IID_STEPS; + noFactors = NO_IID_LEVELS; + } + + /* dequantize and decode */ + for (group = 0; group < NO_IID_GROUPS; group++) { + bin = bins2groupMap20[group]; + + /*! + <h3> type 'A' rotation </h3> + mixing procedure R_a, used in baseline version<br> + + Scale-factor vectors c1 and c2 are precalculated in initPsTables () and + stored in scaleFactors[] and scaleFactorsFine[] = pScaleFactors []. From the + linearized IID parameters (intensity differences), two scale factors are + calculated. They are used to obtain the coefficients h11... h22. + */ + + /* ScaleR and ScaleL are scaled by 1 shift right */ + + ScaleL = ScaleR = 0; + if (noIidSteps + h_ps_d->specificTo.mpeg.pCoef->aaIidIndexMapped[env][bin] >= 0 && noIidSteps + h_ps_d->specificTo.mpeg.pCoef->aaIidIndexMapped[env][bin] < noFactors) + ScaleR = PScaleFactors[noIidSteps + h_ps_d->specificTo.mpeg.pCoef + ->aaIidIndexMapped[env][bin]]; + if (noIidSteps - h_ps_d->specificTo.mpeg.pCoef->aaIidIndexMapped[env][bin] >= 0 && noIidSteps - h_ps_d->specificTo.mpeg.pCoef->aaIidIndexMapped[env][bin] < noFactors) + ScaleL = PScaleFactors[noIidSteps - h_ps_d->specificTo.mpeg.pCoef + ->aaIidIndexMapped[env][bin]]; + + AlphasValue = 0; + if (h_ps_d->specificTo.mpeg.pCoef->aaIccIndexMapped[env][bin] >= 0) + AlphasValue = Alphas[h_ps_d->specificTo.mpeg.pCoef->aaIccIndexMapped[env][bin]]; + Beta = fMult( + fMult(AlphasValue, + (ScaleR - ScaleL)), + FIXP_SQRT05); + Alpha = + AlphasValue >> 1; + + /* Alpha and Beta are now both scaled by 2 shifts right */ + + /* calculate the coefficients h11... h22 from scale-factors and ICC + * parameters */ + + /* h values are scaled by 1 shift right */ + { + FIXP_DBL trigData[4]; + + inline_fixp_cos_sin(Beta + Alpha, Beta - Alpha, 2, trigData); + h11r = fMult(ScaleL, trigData[0]); + h12r = fMult(ScaleR, trigData[2]); + h21r = fMult(ScaleL, trigData[1]); + h22r = fMult(ScaleR, trigData[3]); + } + /*****************************************************************************************/ + /* Interpolation of the matrices H11... H22: */ + /* */ + /* H11(k,n) = H11(k,n[e]) + (n-n[e]) * (H11(k,n[e+1] - H11(k,n[e])) / + * (n[e+1] - n[e]) */ + /* ... */ + /*****************************************************************************************/ + + /* invL = 1/(length of envelope) */ + invL = FX_DBL2FX_SGL(GetInvInt( + h_ps_d->bsData[h_ps_d->processSlot].mpeg.aEnvStartStop[env + 1] - + h_ps_d->bsData[h_ps_d->processSlot].mpeg.aEnvStartStop[env])); + + h_ps_d->specificTo.mpeg.pCoef->H11r[group] = + h_ps_d->specificTo.mpeg.h11rPrev[group]; + h_ps_d->specificTo.mpeg.pCoef->H12r[group] = + h_ps_d->specificTo.mpeg.h12rPrev[group]; + h_ps_d->specificTo.mpeg.pCoef->H21r[group] = + h_ps_d->specificTo.mpeg.h21rPrev[group]; + h_ps_d->specificTo.mpeg.pCoef->H22r[group] = + h_ps_d->specificTo.mpeg.h22rPrev[group]; + + h_ps_d->specificTo.mpeg.pCoef->DeltaH11r[group] = + fMult(h11r - h_ps_d->specificTo.mpeg.pCoef->H11r[group], invL); + h_ps_d->specificTo.mpeg.pCoef->DeltaH12r[group] = + fMult(h12r - h_ps_d->specificTo.mpeg.pCoef->H12r[group], invL); + h_ps_d->specificTo.mpeg.pCoef->DeltaH21r[group] = + fMult(h21r - h_ps_d->specificTo.mpeg.pCoef->H21r[group], invL); + h_ps_d->specificTo.mpeg.pCoef->DeltaH22r[group] = + fMult(h22r - h_ps_d->specificTo.mpeg.pCoef->H22r[group], invL); + + /* update prev coefficients for interpolation in next envelope */ + + h_ps_d->specificTo.mpeg.h11rPrev[group] = h11r; + h_ps_d->specificTo.mpeg.h12rPrev[group] = h12r; + h_ps_d->specificTo.mpeg.h21rPrev[group] = h21r; + h_ps_d->specificTo.mpeg.h22rPrev[group] = h22r; + + } /* group loop */ +} + +static const UCHAR groupTable[NO_IID_GROUPS + 1] = { + 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, + 12, 13, 14, 15, 16, 18, 21, 25, 30, 42, 71}; + +static void applySlotBasedRotation( + HANDLE_PS_DEC h_ps_d, /*!< pointer to the module state */ + + FIXP_DBL *mHybridRealLeft, /*!< hybrid values real left */ + FIXP_DBL *mHybridImagLeft, /*!< hybrid values imag left */ + + FIXP_DBL *mHybridRealRight, /*!< hybrid values real right */ + FIXP_DBL *mHybridImagRight /*!< hybrid values imag right */ +) { + INT group; + INT subband; + + /**********************************************************************************************/ + /*! + <h2> Mapping </h2> + + The number of stereo bands that is actually used depends on the number of + availble parameters for IID and ICC: <pre> nr. of IID para.| nr. of ICC para. + | nr. of Stereo bands + ----------------|------------------|------------------- + 10,20 | 10,20 | 20 + 10,20 | 34 | 34 + 34 | 10,20 | 34 + 34 | 34 | 34 + </pre> + In the case the number of parameters for IIS and ICC differs from the number + of stereo bands, a mapping from the lower number to the higher number of + parameters is applied. Index mapping of IID and ICC parameters is already done + in psbitdec.cpp. Further mapping is not needed here in baseline version. + **********************************************************************************************/ + + /************************************************************************************************/ + /*! + <h2> Mixing </h2> + + To generate the QMF subband signals for the subband samples n = n[e]+1 ,,, + n_[e+1] the parameters at position n[e] and n[e+1] are required as well as the + subband domain signals s_k(n) and d_k(n) for n = n[e]+1... n_[e+1]. n[e] + represents the start position for envelope e. The border positions n[e] are + handled in DecodePS(). + + The stereo sub subband signals are constructed as: + <pre> + l_k(n) = H11(k,n) s_k(n) + H21(k,n) d_k(n) + r_k(n) = H21(k,n) s_k(n) + H22(k,n) d_k(n) + </pre> + In order to obtain the matrices H11(k,n)... H22 (k,n), the vectors h11(b)... + h22(b) need to be calculated first (b: parameter index). Depending on ICC mode + either mixing procedure R_a or R_b is used for that. For both procedures, the + parameters for parameter position n[e+1] is used. + ************************************************************************************************/ + + /************************************************************************************************/ + /*! + <h2>Phase parameters </h2> + With disabled phase parameters (which is the case in baseline version), the + H-matrices are just calculated by: + + <pre> + H11(k,n[e+1] = h11(b(k)) + (...) + b(k): parameter index according to mapping table + </pre> + + <h2>Processing of the samples in the sub subbands </h2> + this loop includes the interpolation of the coefficients Hxx + ************************************************************************************************/ + + /******************************************************/ + /* construct stereo sub subband signals according to: */ + /* */ + /* l_k(n) = H11(k,n) s_k(n) + H21(k,n) d_k(n) */ + /* r_k(n) = H12(k,n) s_k(n) + H22(k,n) d_k(n) */ + /******************************************************/ + PS_DEC_COEFFICIENTS *pCoef = h_ps_d->specificTo.mpeg.pCoef; + + for (group = 0; group < NO_IID_GROUPS; group++) { + pCoef->H11r[group] += pCoef->DeltaH11r[group]; + pCoef->H12r[group] += pCoef->DeltaH12r[group]; + pCoef->H21r[group] += pCoef->DeltaH21r[group]; + pCoef->H22r[group] += pCoef->DeltaH22r[group]; + + const int start = groupTable[group]; + const int stop = groupTable[group + 1]; + for (subband = start; subband < stop; subband++) { + FIXP_DBL tmpLeft = + fMultAdd(fMultDiv2(pCoef->H11r[group], mHybridRealLeft[subband]), + pCoef->H21r[group], mHybridRealRight[subband]); + FIXP_DBL tmpRight = + fMultAdd(fMultDiv2(pCoef->H12r[group], mHybridRealLeft[subband]), + pCoef->H22r[group], mHybridRealRight[subband]); + mHybridRealLeft[subband] = tmpLeft; + mHybridRealRight[subband] = tmpRight; + + tmpLeft = + fMultAdd(fMultDiv2(pCoef->H11r[group], mHybridImagLeft[subband]), + pCoef->H21r[group], mHybridImagRight[subband]); + tmpRight = + fMultAdd(fMultDiv2(pCoef->H12r[group], mHybridImagLeft[subband]), + pCoef->H22r[group], mHybridImagRight[subband]); + mHybridImagLeft[subband] = tmpLeft; + mHybridImagRight[subband] = tmpRight; + } /* subband */ + } +} + +/***************************************************************************/ +/*! + \brief Applies IID, ICC, IPD and OPD parameters to the current frame. + + \return none + +****************************************************************************/ +void ApplyPsSlot( + HANDLE_PS_DEC h_ps_d, /*!< handle PS_DEC*/ + FIXP_DBL **rIntBufferLeft, /*!< real bands left qmf channel (38x64) */ + FIXP_DBL **iIntBufferLeft, /*!< imag bands left qmf channel (38x64) */ + FIXP_DBL *rIntBufferRight, /*!< real bands right qmf channel (38x64) */ + FIXP_DBL *iIntBufferRight, /*!< imag bands right qmf channel (38x64) */ + const int scaleFactorLowBand_no_ov, const int scaleFactorLowBand, + const int scaleFactorHighBand, const int lsb, const int usb) { +/*! +The 64-band QMF representation of the monaural signal generated by the SBR tool +is used as input of the PS tool. After the PS processing, the outputs of the +left and right hybrid synthesis filterbanks are used to generate the stereo +output signal. + +<pre> + + ------------- ---------- ------------- + | Hybrid | M_n[k,m] | | L_n[k,m] | Hybrid | l[n] + m[n] --->| analysis |--------->| |--------->| synthesis |-----> + ------------- | Stereo | ------------- + | | recon- | + | | stuction | + \|/ | | + ------------- | | + | De- | D_n[k,m] | | + | correlation |--------->| | + ------------- | | ------------- + | | R_n[k,m] | Hybrid | r[n] + | |--------->| synthesis |-----> + IID, ICC ------------------------>| | | filter bank | +(IPD, OPD) ---------- ------------- + +m[n]: QMF represantation of the mono input +M_n[k,m]: (sub-)sub-band domain signals of the mono input +D_n[k,m]: decorrelated (sub-)sub-band domain signals +L_n[k,m]: (sub-)sub-band domain signals of the left output +R_n[k,m]: (sub-)sub-band domain signals of the right output +l[n],r[n]: left/right output signals + +</pre> +*/ +#define NO_HYBRID_DATA_BANDS (71) + + int i; + FIXP_DBL qmfInputData[2][NO_QMF_BANDS_HYBRID20]; + FIXP_DBL *hybridData[2][2]; + C_ALLOC_SCRATCH_START(pHybridData, FIXP_DBL, 4 * NO_HYBRID_DATA_BANDS); + + hybridData[0][0] = + pHybridData + 0 * NO_HYBRID_DATA_BANDS; /* left real hybrid data */ + hybridData[0][1] = + pHybridData + 1 * NO_HYBRID_DATA_BANDS; /* left imag hybrid data */ + hybridData[1][0] = + pHybridData + 2 * NO_HYBRID_DATA_BANDS; /* right real hybrid data */ + hybridData[1][1] = + pHybridData + 3 * NO_HYBRID_DATA_BANDS; /* right imag hybrid data */ + + /*! + Hybrid analysis filterbank: + The lower 3 (5) of the 64 QMF subbands are further split to provide better + frequency resolution. for PS processing. For the 10 and 20 stereo bands + configuration, the QMF band H_0(w) is split up into 8 (sub-) sub-bands and the + QMF bands H_1(w) and H_2(w) are spit into 2 (sub-) 4th. (See figures 8.20 + and 8.22 of ISO/IEC 14496-3:2001/FDAM 2:2004(E) ) + */ + + /* + * Hybrid analysis. + */ + + /* Get qmf input data and apply descaling */ + for (i = 0; i < NO_QMF_BANDS_HYBRID20; i++) { + qmfInputData[0][i] = scaleValue(rIntBufferLeft[HYBRID_FILTER_DELAY][i], + scaleFactorLowBand_no_ov); + qmfInputData[1][i] = scaleValue(iIntBufferLeft[HYBRID_FILTER_DELAY][i], + scaleFactorLowBand_no_ov); + } + + /* LF - part */ + FDKhybridAnalysisApply(&h_ps_d->specificTo.mpeg.hybridAnalysis, + qmfInputData[0], qmfInputData[1], hybridData[0][0], + hybridData[0][1]); + + /* HF - part */ + /* bands up to lsb */ + scaleValues(&hybridData[0][0][NO_SUB_QMF_CHANNELS - 2], + &rIntBufferLeft[0][NO_QMF_BANDS_HYBRID20], + lsb - NO_QMF_BANDS_HYBRID20, scaleFactorLowBand); + scaleValues(&hybridData[0][1][NO_SUB_QMF_CHANNELS - 2], + &iIntBufferLeft[0][NO_QMF_BANDS_HYBRID20], + lsb - NO_QMF_BANDS_HYBRID20, scaleFactorLowBand); + + /* bands from lsb to usb */ + scaleValues(&hybridData[0][0][lsb + (NO_SUB_QMF_CHANNELS - 2 - + NO_QMF_BANDS_HYBRID20)], + &rIntBufferLeft[0][lsb], usb - lsb, scaleFactorHighBand); + scaleValues(&hybridData[0][1][lsb + (NO_SUB_QMF_CHANNELS - 2 - + NO_QMF_BANDS_HYBRID20)], + &iIntBufferLeft[0][lsb], usb - lsb, scaleFactorHighBand); + + /* bands from usb to NO_SUB_QMF_CHANNELS which should be zero for non-overlap + slots but can be non-zero for overlap slots */ + FDKmemcpy( + &hybridData[0][0] + [usb + (NO_SUB_QMF_CHANNELS - 2 - NO_QMF_BANDS_HYBRID20)], + &rIntBufferLeft[0][usb], sizeof(FIXP_DBL) * (NO_QMF_CHANNELS - usb)); + FDKmemcpy( + &hybridData[0][1] + [usb + (NO_SUB_QMF_CHANNELS - 2 - NO_QMF_BANDS_HYBRID20)], + &iIntBufferLeft[0][usb], sizeof(FIXP_DBL) * (NO_QMF_CHANNELS - usb)); + + /*! + Decorrelation: + By means of all-pass filtering and delaying, the (sub-)sub-band samples s_k(n) + are converted into de-correlated (sub-)sub-band samples d_k(n). + - k: frequency in hybrid spectrum + - n: time index + */ + + FDKdecorrelateApply(&h_ps_d->specificTo.mpeg.apDecor, + &hybridData[0][0][0], /* left real hybrid data */ + &hybridData[0][1][0], /* left imag hybrid data */ + &hybridData[1][0][0], /* right real hybrid data */ + &hybridData[1][1][0], /* right imag hybrid data */ + 0 /* startHybBand */ + ); + + /*! + Stereo Processing: + The sets of (sub-)sub-band samples s_k(n) and d_k(n) are processed according + to the stereo cues which are defined per stereo band. + */ + + applySlotBasedRotation(h_ps_d, + &hybridData[0][0][0], /* left real hybrid data */ + &hybridData[0][1][0], /* left imag hybrid data */ + &hybridData[1][0][0], /* right real hybrid data */ + &hybridData[1][1][0] /* right imag hybrid data */ + ); + + /*! + Hybrid synthesis filterbank: + The stereo processed hybrid subband signals l_k(n) and r_k(n) are fed into the + hybrid synthesis filterbanks which are identical to the 64 complex synthesis + filterbank of the SBR tool. The input to the filterbank are slots of 64 QMF + samples. For each slot the filterbank outputs one block of 64 samples of one + reconstructed stereo channel. The hybrid synthesis filterbank is computed + seperatly for the left and right channel. + */ + + /* + * Hybrid synthesis. + */ + for (i = 0; i < 2; i++) { + FDKhybridSynthesisApply( + &h_ps_d->specificTo.mpeg.hybridSynthesis[i], + hybridData[i][0], /* real hybrid data */ + hybridData[i][1], /* imag hybrid data */ + (i == 0) ? rIntBufferLeft[0] + : rIntBufferRight, /* output real qmf buffer */ + (i == 0) ? iIntBufferLeft[0] + : iIntBufferRight /* output imag qmf buffer */ + ); + } + + /* free temporary hybrid qmf values of one timeslot */ + C_ALLOC_SCRATCH_END(pHybridData, FIXP_DBL, 4 * NO_HYBRID_DATA_BANDS); + +} /* END ApplyPsSlot */ diff --git a/fdk-aac/libSBRdec/src/psdec.h b/fdk-aac/libSBRdec/src/psdec.h new file mode 100644 index 0000000..029eac4 --- /dev/null +++ b/fdk-aac/libSBRdec/src/psdec.h @@ -0,0 +1,333 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief Sbr decoder +*/ +#ifndef PSDEC_H +#define PSDEC_H + +#include "sbrdecoder.h" +#include "FDK_hybrid.h" + +#include "FDK_decorrelate.h" + +/* This PS decoder implements the baseline version. So it always uses the */ +/* hybrid filter structure for 20 stereo bands and does not implemet IPD/OPD */ +/* synthesis. The baseline version has to support the complete PS bitstream */ +/* syntax. But IPD/OPD data is ignored and set to 0. If 34 stereo band config */ +/* is used in the bitstream for IIS/ICC the decoded parameters are mapped to */ +/* 20 stereo bands. */ + +#include "FDK_bitstream.h" + +#define SCAL_HEADROOM (2) + +#define PS_EXTENSION_SIZE_BITS (4) +#define PS_EXTENSION_ESC_COUNT_BITS (8) + +#define NO_QMF_CHANNELS (64) +#define MAX_NUM_COL (32) + +#define NO_QMF_BANDS_HYBRID20 (3) +#define NO_SUB_QMF_CHANNELS (12) +#define HYBRID_FILTER_DELAY (6) + +#define MAX_NO_PS_ENV (4 + 1) /* +1 needed for VAR_BORDER */ + +#define NO_HI_RES_BINS (34) +#define NO_MID_RES_BINS (20) +#define NO_LOW_RES_BINS (10) + +#define NO_HI_RES_IID_BINS (NO_HI_RES_BINS) +#define NO_HI_RES_ICC_BINS (NO_HI_RES_BINS) + +#define NO_MID_RES_IID_BINS (NO_MID_RES_BINS) +#define NO_MID_RES_ICC_BINS (NO_MID_RES_BINS) + +#define NO_LOW_RES_IID_BINS (NO_LOW_RES_BINS) +#define NO_LOW_RES_ICC_BINS (NO_LOW_RES_BINS) + +#define SUBQMF_GROUPS (10) +#define QMF_GROUPS (12) + +//#define SUBQMF_GROUPS_HI_RES ( 32 ) +//#define QMF_GROUPS_HI_RES ( 18 ) + +#define NO_IID_GROUPS (SUBQMF_GROUPS + QMF_GROUPS) +//#define NO_IID_GROUPS_HI_RES ( SUBQMF_GROUPS_HI_RES + +// QMF_GROUPS_HI_RES ) + +#define NO_IID_STEPS (7) /* 1 .. + 7 */ +#define NO_IID_STEPS_FINE (15) /* 1 .. +15 */ +#define NO_ICC_STEPS (8) /* 0 .. + 7 */ + +#define NO_IID_LEVELS (2 * NO_IID_STEPS + 1) /* - 7 .. + 7 */ +#define NO_IID_LEVELS_FINE (2 * NO_IID_STEPS_FINE + 1) /* -15 .. +15 */ +#define NO_ICC_LEVELS (NO_ICC_STEPS) /* 0 .. + 7 */ + +#define FIXP_SQRT05 ((FIXP_DBL)0x5a827980) /* 1/SQRT2 */ + +struct PS_DEC_COEFFICIENTS { + FIXP_DBL H11r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ + FIXP_DBL H12r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ + FIXP_DBL H21r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ + FIXP_DBL H22r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ + + FIXP_DBL + DeltaH11r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ + FIXP_DBL + DeltaH12r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ + FIXP_DBL + DeltaH21r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ + FIXP_DBL + DeltaH22r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ + + SCHAR + aaIidIndexMapped[MAX_NO_PS_ENV] + [NO_HI_RES_IID_BINS]; /*!< The mapped IID index for all + envelopes and all IID bins */ + SCHAR + aaIccIndexMapped[MAX_NO_PS_ENV] + [NO_HI_RES_ICC_BINS]; /*!< The mapped ICC index for all + envelopes and all ICC bins */ +}; + +typedef enum { ppt_none = 0, ppt_mpeg = 1, ppt_drm = 2 } PS_PAYLOAD_TYPE; + +typedef struct { + UCHAR bPsHeaderValid; /*!< set if new header is available from bitstream */ + + UCHAR bEnableIid; /*!< One bit denoting the presence of IID parameters */ + UCHAR bEnableIcc; /*!< One bit denoting the presence of ICC parameters */ + UCHAR bEnableExt; /*!< The PS extension layer is enabled using the enable_ext + bit. If it is set to %1 the IPD and OPD parameters are + sent. If it is disabled, i.e. %0, the extension layer is + skipped. */ + + UCHAR + modeIid; /*!< The configuration of IID parameters (number of bands and + quantisation grid, iid_quant) is determined by iid_mode. */ + UCHAR modeIcc; /*!< The configuration of Inter-channel Coherence parameters + (number of bands and quantisation grid) is determined by + icc_mode. */ + + UCHAR freqResIid; /*!< 0=low, 1=mid or 2=high frequency resolution for iid */ + UCHAR freqResIcc; /*!< 0=low, 1=mid or 2=high frequency resolution for icc */ + + UCHAR bFineIidQ; /*!< Use fine Iid quantisation. */ + + UCHAR bFrameClass; /*!< The frame_class bit determines whether the parameter + positions of the current frame are uniformly spaced + accross the frame or they are defined using the + positions described by border_position. + */ + + UCHAR noEnv; /*!< The number of envelopes per frame */ + UCHAR aEnvStartStop[MAX_NO_PS_ENV + 1]; /*!< In case of variable parameter + spacing the parameter positions are + determined by border_position */ + + SCHAR abIidDtFlag[MAX_NO_PS_ENV]; /*!< Deltacoding time/freq flag for IID, 0 + => freq */ + SCHAR abIccDtFlag[MAX_NO_PS_ENV]; /*!< Deltacoding time/freq flag for ICC, 0 + => freq */ + + SCHAR + aaIidIndex[MAX_NO_PS_ENV] + [NO_HI_RES_IID_BINS]; /*!< The IID index for all envelopes and + all IID bins */ + SCHAR + aaIccIndex[MAX_NO_PS_ENV] + [NO_HI_RES_ICC_BINS]; /*!< The ICC index for all envelopes and + all ICC bins */ + +} MPEG_PS_BS_DATA; + +struct PS_DEC { + SCHAR noSubSamples; + SCHAR noChannels; + + SCHAR procFrameBased; /*!< Helper to detected switching from frame based to + slot based processing + */ + + PS_PAYLOAD_TYPE + bPsDataAvail[(1) + 1]; /*!< set if new data available from bitstream */ + UCHAR psDecodedPrv; /*!< set if PS has been processed in the last frame */ + + /* helpers for frame delay line */ + UCHAR bsLastSlot; /*!< Index of last read slot. */ + UCHAR bsReadSlot; /*!< Index of current read slot for additional delay. */ + UCHAR processSlot; /*!< Index of current slot for processing (need for add. + delay). */ + + union { /* Bitstream data */ + MPEG_PS_BS_DATA + mpeg; /*!< Struct containing all MPEG specific PS data from bitstream. + */ + } bsData[(1) + 1]; + + shouldBeUnion { /* Static data */ + struct { + SCHAR aIidPrevFrameIndex[NO_HI_RES_IID_BINS]; /*!< The IID index for + previous frame */ + SCHAR aIccPrevFrameIndex[NO_HI_RES_ICC_BINS]; /*!< The ICC index for + previous frame */ + UCHAR + bPrevFrameFineIidQ; /*!< The IID quantization of the previous frame */ + UCHAR prevFreqResIid; /*!< Frequency resolution for IID of the previous + frame */ + UCHAR prevFreqResIcc; /*!< Frequency resolution for ICC of the previous + frame */ + UCHAR lastUsb; /*!< uppermost WMF delay band of last frame */ + + FIXP_DBL pHybridAnaStatesLFdmx + [2 * 13 * NO_QMF_BANDS_HYBRID20]; /*!< Memory used in hybrid analysis + for filter states. */ + FDK_ANA_HYB_FILTER hybridAnalysis; + FDK_SYN_HYB_FILTER hybridSynthesis[2]; + + DECORR_DEC apDecor; /*!< Decorrelator instance. */ + FIXP_DBL decorrBufferCplx[(2 * ((825) + (373)))]; + + FIXP_DBL h11rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy) + coefficients */ + FIXP_DBL h12rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy) + coefficients */ + FIXP_DBL h21rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy) + coefficients */ + FIXP_DBL h22rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy) + coefficients */ + + PS_DEC_COEFFICIENTS + *pCoef; /*!< temporal coefficients are on reusable scratch memory */ + + } mpeg; + } + specificTo; +}; + +typedef struct PS_DEC *HANDLE_PS_DEC; + +int CreatePsDec(HANDLE_PS_DEC *h_PS_DEC, int aacSamplesPerFrame); + +int DeletePsDec(HANDLE_PS_DEC *h_PS_DEC); + +void PreparePsProcessing(HANDLE_PS_DEC h_ps_d, + const FIXP_DBL *const *const rIntBufferLeft, + const FIXP_DBL *const *const iIntBufferLeft, + const int scaleFactorLowBand); + +void initSlotBasedRotation(HANDLE_PS_DEC h_ps_d, int env, int usb); + +void ApplyPsSlot( + HANDLE_PS_DEC h_ps_d, /* parametric stereo decoder handle */ + FIXP_DBL **rIntBufferLeft, /* real values of left qmf timeslot */ + FIXP_DBL **iIntBufferLeft, /* imag values of left qmf timeslot */ + FIXP_DBL *rIntBufferRight, /* real values of right qmf timeslot */ + FIXP_DBL *iIntBufferRight, /* imag values of right qmf timeslot */ + const int scaleFactorLowBand_no_ov, const int scaleFactorLowBand, + const int scaleFactorHighBand, const int lsb, const int usb); + +#endif /* PSDEC_H */ diff --git a/fdk-aac/libSBRdec/src/psdec_drm.cpp b/fdk-aac/libSBRdec/src/psdec_drm.cpp new file mode 100644 index 0000000..6971f53 --- /dev/null +++ b/fdk-aac/libSBRdec/src/psdec_drm.cpp @@ -0,0 +1,108 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief parametric stereo decoder for Digital radio mondial +*/ + +#include "psdec_drm.h" diff --git a/fdk-aac/libSBRdec/src/psdec_drm.h b/fdk-aac/libSBRdec/src/psdec_drm.h new file mode 100644 index 0000000..5e2575d --- /dev/null +++ b/fdk-aac/libSBRdec/src/psdec_drm.h @@ -0,0 +1,113 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief parametric stereo decoder for digital radio mondial +*/ + +#ifndef PSDEC_DRM_H +#define PSDEC_DRM_H + +#include "sbrdecoder.h" + +#endif /* PSDEC_DRM_H */ diff --git a/fdk-aac/libSBRdec/src/psdecrom_drm.cpp b/fdk-aac/libSBRdec/src/psdecrom_drm.cpp new file mode 100644 index 0000000..2033a83 --- /dev/null +++ b/fdk-aac/libSBRdec/src/psdecrom_drm.cpp @@ -0,0 +1,108 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief rom tables for Drm parametric stereo decoder +*/ + +#include "psdec_drm.h" diff --git a/fdk-aac/libSBRdec/src/pvc_dec.cpp b/fdk-aac/libSBRdec/src/pvc_dec.cpp new file mode 100644 index 0000000..b477122 --- /dev/null +++ b/fdk-aac/libSBRdec/src/pvc_dec.cpp @@ -0,0 +1,683 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): Matthias Hildenbrand + + Description: Decode Predictive Vector Coding Data + +*******************************************************************************/ + +#include "pvc_dec.h" + +/* PVC interal definitions */ +#define PVC_DIVMODE_BITS 3 +#define PVC_NSMODE_BITS 1 +#define PVC_REUSEPVCID_BITS 1 +#define PVC_PVCID_BITS 7 +#define PVC_GRIDINFO_BITS 1 +#define PVC_NQMFBAND 64 +#define PVC_NBLOW 3 /* max. number of grouped QMF subbands below SBR range */ + +#define PVC_NTAB1 3 +#define PVC_NTAB2 128 +#define PVC_ID_NBIT 7 + +/* Exponent of pPvcStaticData->Esg and predictedEsg in dB domain. + max(Esg) = 10*log10(2^15*2^15) = 90.30; + min(Esg) = 10*log10(0.1) = -10 + max of predicted Esg seems to be higher than 90dB but 7 Bit should be enough. +*/ +#define PVC_ESG_EXP 7 + +#define LOG10FAC 0.752574989159953f /* == 10/log2(10) * 2^-2 */ +#define LOG10FAC_INV 0.664385618977472f /* == log2(10)/10 * 2^1 */ + +RAM_ALIGN +LNK_SECTION_CONSTDATA +static const FIXP_SGL pvc_SC_16[] = { + FX_DBL2FXCONST_SGL(0x14413695), FX_DBL2FXCONST_SGL(0x1434b6cb), + FX_DBL2FXCONST_SGL(0x140f27c7), FX_DBL2FXCONST_SGL(0x13d0591d), + FX_DBL2FXCONST_SGL(0x1377f502), FX_DBL2FXCONST_SGL(0x130577d6), + FX_DBL2FXCONST_SGL(0x12782266), FX_DBL2FXCONST_SGL(0x11cee459), + FX_DBL2FXCONST_SGL(0x11083a2a), FX_DBL2FXCONST_SGL(0x1021f5e9), + FX_DBL2FXCONST_SGL(0x0f18e17c), FX_DBL2FXCONST_SGL(0x0de814ca), + FX_DBL2FXCONST_SGL(0x0c87a568), FX_DBL2FXCONST_SGL(0x0ae9b167), + FX_DBL2FXCONST_SGL(0x08f24226), FX_DBL2FXCONST_SGL(0x06575ed5), +}; + +RAM_ALIGN +LNK_SECTION_CONSTDATA +static const FIXP_SGL pvc_SC_12[] = { + FX_DBL2FXCONST_SGL(0x1aba6b3e), FX_DBL2FXCONST_SGL(0x1a9d164e), + FX_DBL2FXCONST_SGL(0x1a44d56d), FX_DBL2FXCONST_SGL(0x19b0d742), + FX_DBL2FXCONST_SGL(0x18df969a), FX_DBL2FXCONST_SGL(0x17ce91a0), + FX_DBL2FXCONST_SGL(0x1679c3fa), FX_DBL2FXCONST_SGL(0x14daabfc), + FX_DBL2FXCONST_SGL(0x12e65221), FX_DBL2FXCONST_SGL(0x1088d125), + FX_DBL2FXCONST_SGL(0x0d9907b3), FX_DBL2FXCONST_SGL(0x09a80e9d), +}; + +RAM_ALIGN +LNK_SECTION_CONSTDATA +static const FIXP_SGL pvc_SC_4[] = { + FX_DBL2FXCONST_SGL(0x4ad6ab0f), + FX_DBL2FXCONST_SGL(0x47ef0dbe), + FX_DBL2FXCONST_SGL(0x3eee7496), + FX_DBL2FXCONST_SGL(0x2e4bd29d), +}; + +RAM_ALIGN +LNK_SECTION_CONSTDATA +static const FIXP_SGL pvc_SC_3[] = { + FX_DBL2FXCONST_SGL(0x610dc761), + FX_DBL2FXCONST_SGL(0x5a519a3d), + FX_DBL2FXCONST_SGL(0x44a09e62), +}; + +static const UCHAR g_3a_pvcTab1_mode1[PVC_NTAB1][PVC_NBLOW][PVC_NBHIGH_MODE1] = + {{{0x4F, 0x5B, 0x57, 0x52, 0x4D, 0x65, 0x45, 0x57}, + {0xF3, 0x0F, 0x18, 0x20, 0x19, 0x4F, 0x3D, 0x23}, + {0x78, 0x57, 0x55, 0x50, 0x50, 0x20, 0x36, 0x37}}, + {{0x4C, 0x5F, 0x53, 0x37, 0x1E, 0xFD, 0x15, 0x0A}, + {0x05, 0x0E, 0x28, 0x41, 0x48, 0x6E, 0x54, 0x5B}, + {0x59, 0x47, 0x40, 0x40, 0x3D, 0x33, 0x3F, 0x39}}, + {{0x47, 0x5F, 0x57, 0x34, 0x3C, 0x2E, 0x2E, 0x31}, + {0xFA, 0x13, 0x23, 0x4E, 0x44, 0x7C, 0x34, 0x38}, + {0x63, 0x43, 0x41, 0x3D, 0x35, 0x19, 0x3D, 0x33}}}; + +static const UCHAR g_2a_pvcTab2_mode1[PVC_NTAB2][PVC_NBHIGH_MODE1] = { + {0xCB, 0xD1, 0xCC, 0xD2, 0xE2, 0xEB, 0xE7, 0xE8}, + {0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80}, + {0x84, 0x8C, 0x88, 0x83, 0x90, 0x93, 0x86, 0x80}, + {0xD7, 0xD8, 0xC0, 0xC7, 0xCF, 0xE5, 0xF1, 0xF6}, + {0xA5, 0xA6, 0xAA, 0xA8, 0xB0, 0xB1, 0xB8, 0xB8}, + {0xD7, 0xCB, 0xC1, 0xC3, 0xC5, 0xC9, 0xC9, 0xCE}, + {0xCA, 0xB5, 0xB8, 0xB3, 0xAC, 0xB6, 0xBB, 0xB8}, + {0xC1, 0xC4, 0xC3, 0xC5, 0xC6, 0xCA, 0xCA, 0xCB}, + {0xE0, 0xE1, 0xD8, 0xCD, 0xCB, 0xCB, 0xCE, 0xCC}, + {0xDB, 0xE1, 0xDF, 0xDB, 0xDC, 0xD9, 0xD9, 0xD6}, + {0xE0, 0xDE, 0xDD, 0xDD, 0xE0, 0xE3, 0xE5, 0xE6}, + {0xCA, 0xD2, 0xCD, 0xCE, 0xD5, 0xDB, 0xD9, 0xDB}, + {0xD2, 0xE0, 0xDB, 0xD5, 0xDB, 0xDE, 0xE3, 0xE1}, + {0xE5, 0xDB, 0xD0, 0xD2, 0xD8, 0xDD, 0xDB, 0xDD}, + {0xC0, 0xB5, 0xBF, 0xDD, 0xE3, 0xDC, 0xDC, 0xE4}, + {0xDB, 0xCE, 0xC6, 0xCF, 0xCF, 0xD1, 0xD3, 0xD4}, + {0xC9, 0xD7, 0xDA, 0xE2, 0xE9, 0xE7, 0xDF, 0xDC}, + {0x0A, 0x07, 0x0A, 0x08, 0x19, 0x24, 0x1F, 0x22}, + {0x1E, 0x1F, 0x11, 0x0E, 0x22, 0x2D, 0x33, 0x32}, + {0xF0, 0xDA, 0xDC, 0x18, 0x1F, 0x19, 0x0A, 0x1E}, + {0x09, 0xF8, 0xE6, 0x05, 0x19, 0x11, 0x0E, 0x0B}, + {0x09, 0x10, 0x0E, 0xE6, 0xF4, 0x20, 0x22, 0xFA}, + {0xF2, 0xE5, 0xF8, 0x0E, 0x18, 0x15, 0x0D, 0x10}, + {0x15, 0x13, 0x16, 0x0A, 0x0D, 0x1F, 0x1D, 0x1B}, + {0xFA, 0xFF, 0xFE, 0xFF, 0x09, 0x11, 0x03, 0x0B}, + {0xFE, 0xFA, 0xF2, 0xF8, 0x0C, 0x1E, 0x11, 0x12}, + {0xFA, 0xF8, 0x0B, 0x17, 0x1D, 0x17, 0x0E, 0x16}, + {0x00, 0xF3, 0xFD, 0x0A, 0x1C, 0x17, 0xFD, 0x08}, + {0xEA, 0xEA, 0x03, 0x12, 0x1E, 0x14, 0x09, 0x04}, + {0x02, 0xFE, 0x04, 0xFB, 0x0C, 0x0E, 0x07, 0x02}, + {0xF6, 0x02, 0x07, 0x0B, 0x17, 0x17, 0x01, 0xFF}, + {0xF5, 0xFB, 0xFE, 0x04, 0x12, 0x14, 0x0C, 0x0D}, + {0x10, 0x10, 0x0E, 0x04, 0x07, 0x11, 0x0F, 0x13}, + {0x0C, 0x0F, 0xFB, 0xF2, 0x0A, 0x12, 0x09, 0x0D}, + {0x0D, 0x1D, 0xF1, 0xF4, 0x2A, 0x06, 0x3B, 0x32}, + {0xFC, 0x08, 0x06, 0x02, 0x0E, 0x17, 0x08, 0x0E}, + {0x07, 0x02, 0xEE, 0xEE, 0x2B, 0xF6, 0x23, 0x13}, + {0x04, 0x02, 0x05, 0x08, 0x0B, 0x0E, 0xFB, 0xFB}, + {0x00, 0x04, 0x10, 0x18, 0x22, 0x25, 0x1D, 0x1F}, + {0xFB, 0x0D, 0x07, 0x00, 0x0C, 0x0F, 0xFC, 0x02}, + {0x00, 0x00, 0x00, 0x01, 0x05, 0x07, 0x03, 0x05}, + {0x04, 0x05, 0x08, 0x13, 0xFF, 0xEB, 0x0C, 0x06}, + {0x05, 0x13, 0x0E, 0x0B, 0x12, 0x15, 0x09, 0x0A}, + {0x09, 0x03, 0x09, 0x05, 0x12, 0x16, 0x11, 0x12}, + {0x14, 0x1A, 0x06, 0x01, 0x10, 0x11, 0xFE, 0x02}, + {0x01, 0x0B, 0x0B, 0x0C, 0x18, 0x21, 0x10, 0x13}, + {0x12, 0x0D, 0x0A, 0x10, 0x1C, 0x1D, 0x0D, 0x10}, + {0x03, 0x09, 0x14, 0x15, 0x1B, 0x1A, 0x01, 0xFF}, + {0x08, 0x12, 0x13, 0x0E, 0x16, 0x1D, 0x14, 0x1B}, + {0x07, 0x15, 0x1C, 0x1B, 0x20, 0x21, 0x11, 0x0E}, + {0x12, 0x18, 0x19, 0x17, 0x20, 0x25, 0x1A, 0x1E}, + {0x0C, 0x1A, 0x1D, 0x22, 0x2F, 0x33, 0x27, 0x28}, + {0x0E, 0x1A, 0x17, 0x10, 0x0A, 0x0E, 0xFF, 0x06}, + {0x1A, 0x1C, 0x18, 0x14, 0x1A, 0x16, 0x0A, 0x0E}, + {0x1E, 0x27, 0x25, 0x26, 0x27, 0x2A, 0x21, 0x21}, + {0xF1, 0x0A, 0x16, 0x1C, 0x28, 0x25, 0x15, 0x19}, + {0x08, 0x12, 0x09, 0x08, 0x16, 0x17, 0xEF, 0xF6}, + {0x0C, 0x0B, 0x00, 0xFC, 0x04, 0x09, 0xFC, 0x03}, + {0xFB, 0xF1, 0xF8, 0x26, 0x24, 0x18, 0x1D, 0x20}, + {0xF9, 0x01, 0x0C, 0x0F, 0x07, 0x08, 0x06, 0x07}, + {0x07, 0x06, 0x08, 0x04, 0x07, 0x0D, 0x07, 0x09}, + {0xFE, 0x01, 0x06, 0x05, 0x13, 0x1B, 0x14, 0x19}, + {0x09, 0x0C, 0x0E, 0x01, 0x08, 0x05, 0xFB, 0xFD}, + {0x07, 0x06, 0x03, 0x0A, 0x16, 0x12, 0x04, 0x07}, + {0x04, 0x01, 0x00, 0x04, 0x1F, 0x20, 0x0E, 0x0A}, + {0x03, 0xFF, 0xF6, 0xFB, 0x15, 0x1A, 0x00, 0x03}, + {0xFC, 0x18, 0x0B, 0x2D, 0x35, 0x23, 0x12, 0x09}, + {0x02, 0xFE, 0x01, 0xFF, 0x0C, 0x11, 0x0D, 0x0F}, + {0xFA, 0xE9, 0xD9, 0xFF, 0x0D, 0x05, 0x0D, 0x10}, + {0xF1, 0xE0, 0xF0, 0x01, 0x06, 0x06, 0x06, 0x10}, + {0xE9, 0xD4, 0xD7, 0x0F, 0x14, 0x0B, 0x0D, 0x16}, + {0x00, 0xFF, 0xEE, 0xE5, 0xFF, 0x08, 0x02, 0xF9}, + {0xE0, 0xDA, 0xE5, 0xFE, 0x09, 0x02, 0xF9, 0x04}, + {0xE0, 0xE2, 0xF4, 0x09, 0x13, 0x0C, 0x0D, 0x09}, + {0xFC, 0x02, 0x04, 0xFF, 0x00, 0xFF, 0xF8, 0xF7}, + {0xFE, 0xFB, 0xED, 0xF2, 0xFE, 0xFE, 0x08, 0x0C}, + {0xF3, 0xEF, 0xD0, 0xE3, 0x05, 0x11, 0xFD, 0xFF}, + {0xFA, 0xEF, 0xEA, 0xFE, 0x0D, 0x0E, 0xFE, 0x02}, + {0xF7, 0xFB, 0xDB, 0xDF, 0x14, 0xDD, 0x07, 0xFE}, + {0xFE, 0x08, 0x00, 0xDB, 0xE5, 0x1A, 0x13, 0xED}, + {0xF9, 0xFE, 0xFF, 0xF4, 0xF3, 0x00, 0x05, 0x02}, + {0xEF, 0xDE, 0xD8, 0xEB, 0xEA, 0xF5, 0x0E, 0x19}, + {0xFB, 0xFC, 0xFA, 0xEC, 0xEB, 0xED, 0xEE, 0xE8}, + {0xEE, 0xFC, 0xFD, 0x00, 0x04, 0xFC, 0xF0, 0xF5}, + {0x00, 0xFA, 0xF4, 0xF1, 0xF5, 0xFA, 0xFB, 0xF9}, + {0xEB, 0xF0, 0xDF, 0xE3, 0xEF, 0x07, 0x02, 0x05}, + {0xF7, 0xF0, 0xE6, 0xE7, 0x06, 0x15, 0x06, 0x0C}, + {0xF1, 0xE4, 0xD8, 0xEA, 0x06, 0xF2, 0x07, 0x09}, + {0xFF, 0xFE, 0xFE, 0xF9, 0xFF, 0xFF, 0x02, 0xF9}, + {0xDD, 0xF4, 0xF0, 0xF1, 0xFF, 0xFF, 0xEA, 0xF1}, + {0xF0, 0xF1, 0xFD, 0x03, 0x03, 0xFE, 0x00, 0x05}, + {0xF1, 0xF6, 0xE0, 0xDF, 0xF5, 0x01, 0xF4, 0xF8}, + {0x02, 0x03, 0xE5, 0xDC, 0xE7, 0xFD, 0x02, 0x08}, + {0xEC, 0xF1, 0xF5, 0xEC, 0xF2, 0xF8, 0xF6, 0xEE}, + {0xF3, 0xF4, 0xF6, 0xF4, 0xF5, 0xF1, 0xE7, 0xEA}, + {0xF7, 0xF3, 0xEC, 0xEA, 0xEF, 0xF0, 0xEE, 0xF1}, + {0xEB, 0xF6, 0xFB, 0xFA, 0xEF, 0xF3, 0xF3, 0xF7}, + {0x01, 0x03, 0xF1, 0xF6, 0x05, 0xF8, 0xE1, 0xEB}, + {0xF5, 0xF6, 0xF6, 0xF4, 0xFB, 0xFB, 0xFF, 0x00}, + {0xF8, 0x01, 0xFB, 0xFA, 0xFF, 0x03, 0xFE, 0x04}, + {0x04, 0xFB, 0x03, 0xFD, 0xF5, 0xF7, 0xF6, 0xFB}, + {0x06, 0x09, 0xFB, 0xF4, 0xF9, 0xFA, 0xFC, 0xFF}, + {0xF5, 0xF6, 0xF1, 0xEE, 0xF5, 0xF8, 0xF5, 0xF9}, + {0xF5, 0xF9, 0xFA, 0xFC, 0x07, 0x09, 0x01, 0xFB}, + {0xD7, 0xE9, 0xE8, 0xEC, 0x00, 0x0C, 0xFE, 0xF1}, + {0xEC, 0x04, 0xE9, 0xDF, 0x03, 0xE8, 0x00, 0xFA}, + {0xE6, 0xE2, 0xFF, 0x0A, 0x13, 0x01, 0x00, 0xF7}, + {0xF1, 0xFA, 0xF7, 0xF5, 0x01, 0x06, 0x05, 0x0A}, + {0xF6, 0xF6, 0xFC, 0xF6, 0xE8, 0x11, 0xF2, 0xFE}, + {0xFE, 0x08, 0x05, 0x12, 0xFD, 0xD0, 0x0E, 0x07}, + {0xF1, 0xFE, 0xF7, 0xF2, 0xFB, 0x02, 0xFA, 0xF8}, + {0xF4, 0xEA, 0xEC, 0xF3, 0xFE, 0x01, 0xF7, 0xF6}, + {0xFF, 0xFA, 0xFB, 0xF9, 0xFF, 0x01, 0x04, 0x03}, + {0x00, 0xF9, 0xF4, 0xFC, 0x05, 0xFC, 0xF7, 0xFB}, + {0xF8, 0xFF, 0xEF, 0xEC, 0xFB, 0x04, 0xF8, 0x03}, + {0xEB, 0xF1, 0xED, 0xF4, 0x02, 0x0E, 0x0B, 0x04}, + {0xF7, 0x01, 0xF8, 0xF4, 0xF8, 0xEF, 0xF8, 0x04}, + {0xEB, 0xF0, 0xF7, 0xFC, 0x10, 0x0D, 0xF8, 0xF8}, + {0xE8, 0xFE, 0xEE, 0xE8, 0xED, 0xF7, 0xF5, 0xF8}, + {0xED, 0xEB, 0xE9, 0xEA, 0xF2, 0xF5, 0xF4, 0xF9}, + {0xEA, 0xF2, 0xEF, 0xEE, 0xF9, 0xFE, 0xFD, 0x02}, + {0xFA, 0xFD, 0x02, 0x0D, 0xFA, 0xE4, 0x0F, 0x01}, + {0xFF, 0x08, 0x05, 0xF6, 0xF7, 0xFB, 0xF1, 0xF1}, + {0xF4, 0xEC, 0xEE, 0xF6, 0xEE, 0xEE, 0xF8, 0x06}, + {0xE8, 0xFA, 0xF8, 0xE8, 0xF8, 0xE9, 0xEE, 0xF9}, + {0xE5, 0xE9, 0xF0, 0x00, 0x00, 0xEF, 0xF3, 0xF8}, + {0xF7, 0xFB, 0xFB, 0xF7, 0xF9, 0xF9, 0xF5, 0xF0}, + {0xFD, 0xFF, 0xF2, 0xEE, 0xF2, 0xF5, 0xF1, 0xF3}}; + +static const UCHAR g_3a_pvcTab1_mode2[PVC_NTAB1][PVC_NBLOW][PVC_NBHIGH_MODE2] = + {{{0x11, 0x27, 0x0F, 0xFD, 0x04, 0xFC}, + {0x00, 0xBE, 0xE3, 0xF4, 0xDB, 0xF0}, + {0x09, 0x1E, 0x18, 0x1A, 0x21, 0x1B}}, + {{0x16, 0x28, 0x2B, 0x29, 0x25, 0x32}, + {0xF2, 0xE9, 0xE4, 0xE5, 0xE2, 0xD4}, + {0x0E, 0x0B, 0x0C, 0x0D, 0x0D, 0x0E}}, + {{0x2E, 0x3C, 0x20, 0x16, 0x1B, 0x1A}, + {0xE4, 0xC6, 0xE5, 0xF4, 0xDC, 0xDC}, + {0x0F, 0x1B, 0x18, 0x14, 0x1E, 0x1A}}}; + +static const UCHAR g_2a_pvcTab2_mode2[PVC_NTAB2][PVC_NBHIGH_MODE2] = { + {0x26, 0x25, 0x11, 0x0C, 0xFA, 0x15}, {0x1B, 0x18, 0x11, 0x0E, 0x0E, 0x0E}, + {0x12, 0x10, 0x10, 0x10, 0x11, 0x10}, {0x1E, 0x24, 0x19, 0x15, 0x14, 0x12}, + {0x24, 0x16, 0x12, 0x13, 0x15, 0x1C}, {0xEA, 0xED, 0xEB, 0xEA, 0xEC, 0xEB}, + {0xFC, 0xFD, 0xFD, 0xFC, 0xFE, 0xFE}, {0x0F, 0x0C, 0x0B, 0x0A, 0x0B, 0x0B}, + {0x22, 0x0B, 0x16, 0x18, 0x13, 0x19}, {0x1C, 0x14, 0x1D, 0x20, 0x19, 0x1A}, + {0x10, 0x08, 0x00, 0xFF, 0x02, 0x05}, {0x06, 0x07, 0x05, 0x03, 0x05, 0x04}, + {0x2A, 0x1F, 0x12, 0x12, 0x11, 0x18}, {0x19, 0x19, 0x02, 0x04, 0x00, 0x04}, + {0x18, 0x17, 0x17, 0x15, 0x16, 0x15}, {0x21, 0x1E, 0x1B, 0x19, 0x1C, 0x1B}, + {0x3C, 0x35, 0x20, 0x1D, 0x30, 0x34}, {0x3A, 0x1F, 0x37, 0x38, 0x33, 0x31}, + {0x37, 0x34, 0x25, 0x27, 0x35, 0x34}, {0x34, 0x2E, 0x32, 0x31, 0x34, 0x31}, + {0x36, 0x33, 0x2F, 0x2F, 0x32, 0x2F}, {0x35, 0x20, 0x2F, 0x32, 0x2F, 0x2C}, + {0x2E, 0x2B, 0x2F, 0x34, 0x36, 0x30}, {0x3F, 0x39, 0x30, 0x28, 0x29, 0x29}, + {0x3C, 0x30, 0x32, 0x37, 0x39, 0x36}, {0x37, 0x36, 0x30, 0x2B, 0x26, 0x24}, + {0x44, 0x38, 0x2F, 0x2D, 0x2D, 0x2D}, {0x38, 0x2B, 0x2C, 0x2C, 0x30, 0x2D}, + {0x37, 0x36, 0x2F, 0x23, 0x2D, 0x32}, {0x3C, 0x39, 0x29, 0x2E, 0x38, 0x37}, + {0x3B, 0x3A, 0x35, 0x32, 0x31, 0x2D}, {0x32, 0x31, 0x2F, 0x2C, 0x2D, 0x28}, + {0x2C, 0x31, 0x32, 0x30, 0x32, 0x2D}, {0x35, 0x34, 0x34, 0x34, 0x35, 0x33}, + {0x34, 0x38, 0x3B, 0x3C, 0x3E, 0x3A}, {0x3E, 0x3C, 0x3B, 0x3A, 0x3C, 0x39}, + {0x3D, 0x41, 0x46, 0x41, 0x3D, 0x38}, {0x44, 0x41, 0x40, 0x3E, 0x3F, 0x3A}, + {0x47, 0x47, 0x47, 0x42, 0x44, 0x40}, {0x4C, 0x4A, 0x4A, 0x46, 0x49, 0x45}, + {0x53, 0x52, 0x52, 0x4C, 0x4E, 0x49}, {0x41, 0x3D, 0x39, 0x2C, 0x2E, 0x2E}, + {0x2D, 0x37, 0x36, 0x30, 0x28, 0x36}, {0x3B, 0x32, 0x2E, 0x2D, 0x2D, 0x29}, + {0x40, 0x39, 0x36, 0x35, 0x36, 0x32}, {0x30, 0x2D, 0x2D, 0x2E, 0x31, 0x30}, + {0x38, 0x3D, 0x3B, 0x37, 0x35, 0x34}, {0x44, 0x3D, 0x3C, 0x38, 0x37, 0x33}, + {0x3A, 0x36, 0x37, 0x37, 0x39, 0x36}, {0x32, 0x36, 0x37, 0x30, 0x2E, 0x2A}, + {0x3C, 0x33, 0x33, 0x31, 0x33, 0x30}, {0x30, 0x31, 0x36, 0x37, 0x38, 0x34}, + {0x26, 0x27, 0x2E, 0x29, 0x1C, 0x16}, {0x14, 0x15, 0x1F, 0x17, 0x15, 0x1C}, + {0x38, 0x2D, 0x18, 0x13, 0x1E, 0x2B}, {0x30, 0x22, 0x17, 0x1A, 0x26, 0x2B}, + {0x24, 0x20, 0x1F, 0x10, 0x0C, 0x11}, {0x27, 0x1F, 0x13, 0x17, 0x24, 0x2A}, + {0x2F, 0x13, 0x18, 0x13, 0x2A, 0x32}, {0x31, 0x1E, 0x1E, 0x1E, 0x21, 0x28}, + {0x2A, 0x12, 0x19, 0x17, 0x16, 0x24}, {0x27, 0x0F, 0x16, 0x1D, 0x17, 0x1C}, + {0x2F, 0x26, 0x25, 0x22, 0x20, 0x22}, {0x1E, 0x1B, 0x1E, 0x18, 0x1E, 0x24}, + {0x31, 0x26, 0x0E, 0x15, 0x15, 0x25}, {0x2D, 0x22, 0x1E, 0x14, 0x10, 0x22}, + {0x25, 0x1B, 0x18, 0x11, 0x13, 0x1F}, {0x2F, 0x1B, 0x13, 0x1B, 0x18, 0x22}, + {0x21, 0x24, 0x1D, 0x1C, 0x1D, 0x1B}, {0x23, 0x1E, 0x28, 0x29, 0x27, 0x25}, + {0x2E, 0x2A, 0x1D, 0x17, 0x26, 0x2D}, {0x31, 0x2C, 0x1A, 0x0E, 0x1A, 0x24}, + {0x26, 0x16, 0x20, 0x1D, 0x14, 0x1E}, {0x29, 0x20, 0x1B, 0x1B, 0x17, 0x17}, + {0x1D, 0x06, 0x1A, 0x1E, 0x1B, 0x1D}, {0x2B, 0x23, 0x1F, 0x1F, 0x1D, 0x1C}, + {0x27, 0x1A, 0x0C, 0x0E, 0x0F, 0x1A}, {0x29, 0x1D, 0x1E, 0x22, 0x22, 0x24}, + {0x20, 0x21, 0x1B, 0x18, 0x13, 0x21}, {0x27, 0x0E, 0x10, 0x14, 0x10, 0x1A}, + {0x26, 0x24, 0x25, 0x25, 0x26, 0x28}, {0x1A, 0x24, 0x25, 0x29, 0x26, 0x24}, + {0x1D, 0x1D, 0x15, 0x12, 0x0F, 0x18}, {0x1E, 0x14, 0x13, 0x12, 0x14, 0x18}, + {0x16, 0x13, 0x13, 0x1A, 0x1B, 0x1D}, {0x20, 0x27, 0x22, 0x24, 0x1A, 0x19}, + {0x1F, 0x17, 0x19, 0x18, 0x17, 0x18}, {0x20, 0x1B, 0x1C, 0x1C, 0x1B, 0x1A}, + {0x23, 0x19, 0x1D, 0x1F, 0x1E, 0x21}, {0x26, 0x1F, 0x1D, 0x1B, 0x19, 0x1A}, + {0x23, 0x1E, 0x1F, 0x20, 0x1F, 0x1E}, {0x29, 0x20, 0x22, 0x20, 0x20, 0x1F}, + {0x26, 0x23, 0x21, 0x22, 0x23, 0x23}, {0x29, 0x1F, 0x24, 0x25, 0x26, 0x29}, + {0x2B, 0x22, 0x25, 0x27, 0x23, 0x21}, {0x29, 0x21, 0x19, 0x0E, 0x22, 0x2D}, + {0x32, 0x29, 0x1F, 0x1C, 0x1B, 0x21}, {0x1E, 0x1A, 0x1E, 0x24, 0x25, 0x25}, + {0x24, 0x1D, 0x21, 0x22, 0x22, 0x25}, {0x2C, 0x25, 0x21, 0x22, 0x23, 0x25}, + {0x24, 0x1E, 0x21, 0x26, 0x2B, 0x2C}, {0x28, 0x24, 0x1B, 0x1F, 0x28, 0x2D}, + {0x23, 0x13, 0x16, 0x22, 0x22, 0x29}, {0x1B, 0x23, 0x1C, 0x20, 0x14, 0x0D}, + {0x1E, 0x16, 0x1A, 0x1E, 0x1C, 0x1D}, {0x2B, 0x1C, 0x1D, 0x20, 0x1B, 0x1C}, + {0x1C, 0x1B, 0x23, 0x1F, 0x19, 0x1E}, {0x21, 0x23, 0x26, 0x20, 0x20, 0x22}, + {0x1D, 0x0B, 0x19, 0x1E, 0x11, 0x19}, {0x18, 0x17, 0x16, 0x17, 0x14, 0x16}, + {0x16, 0x19, 0x1C, 0x20, 0x21, 0x22}, {0x30, 0x1E, 0x22, 0x24, 0x25, 0x26}, + {0x1B, 0x1F, 0x17, 0x1D, 0x1E, 0x21}, {0x32, 0x2B, 0x27, 0x1F, 0x1B, 0x1A}, + {0x28, 0x20, 0x1A, 0x1B, 0x1F, 0x23}, {0x32, 0x21, 0x20, 0x21, 0x1D, 0x1F}, + {0x22, 0x18, 0x12, 0x15, 0x1B, 0x20}, {0x27, 0x27, 0x2A, 0x24, 0x21, 0x21}, + {0x1E, 0x0F, 0x0D, 0x1A, 0x1D, 0x23}, {0x28, 0x25, 0x27, 0x21, 0x17, 0x25}, + {0x2B, 0x27, 0x23, 0x19, 0x13, 0x14}, {0x25, 0x2B, 0x22, 0x22, 0x20, 0x21}, + {0x27, 0x1B, 0x16, 0x17, 0x0F, 0x15}, {0x29, 0x26, 0x23, 0x15, 0x1E, 0x28}, + {0x24, 0x1C, 0x19, 0x1A, 0x18, 0x19}, {0x2D, 0x15, 0x27, 0x2B, 0x24, 0x23}, + {0x2C, 0x12, 0x1F, 0x23, 0x1F, 0x20}, {0x25, 0x0F, 0x22, 0x27, 0x1F, 0x21}}; + +static const UCHAR g_a_pvcTab1_dp_mode1[PVC_NTAB1 - 1] = {17, 68}; +static const UCHAR g_a_pvcTab1_dp_mode2[PVC_NTAB1 - 1] = {16, 52}; +/* fractional exponent which corresponds to Q representation value */ +static const SCHAR g_a_scalingCoef_mode1[PVC_NBLOW + 1] = { + -1, -1, 0, 6}; /* { 8, 8, 7, 1 }; Q scaling */ +static const SCHAR g_a_scalingCoef_mode2[PVC_NBLOW + 1] = { + 0, 0, 1, 7}; /* { 7, 7, 6, 0 }; Q scaling */ + +int pvcInitFrame(PVC_STATIC_DATA *pPvcStaticData, + PVC_DYNAMIC_DATA *pPvcDynamicData, const UCHAR pvcMode, + const UCHAR ns, const int RATE, const int kx, + const int pvcBorder0, const UCHAR *pPvcID) { + int lbw, hbw, i, temp; + pPvcDynamicData->pvc_mode = pvcMode; + pPvcDynamicData->kx = kx; + pPvcDynamicData->RATE = RATE; + + switch (pvcMode) { + case 0: + /* legacy SBR, nothing to do */ + return 0; + case 1: + pPvcDynamicData->nbHigh = 8; + pPvcDynamicData->pPVCTab1 = (const UCHAR *)g_3a_pvcTab1_mode1; + pPvcDynamicData->pPVCTab2 = (const UCHAR *)g_2a_pvcTab2_mode1; + pPvcDynamicData->pPVCTab1_dp = g_a_pvcTab1_dp_mode1; + pPvcDynamicData->pScalingCoef = g_a_scalingCoef_mode1; + hbw = 8 / RATE; + break; + case 2: + pPvcDynamicData->nbHigh = 6; + pPvcDynamicData->pPVCTab1 = (const UCHAR *)g_3a_pvcTab1_mode2; + pPvcDynamicData->pPVCTab2 = (const UCHAR *)g_2a_pvcTab2_mode2; + pPvcDynamicData->pPVCTab1_dp = g_a_pvcTab1_dp_mode2; + pPvcDynamicData->pScalingCoef = g_a_scalingCoef_mode2; + hbw = 12 / RATE; + break; + default: + /* invalid pvcMode */ + return 1; + } + + pPvcDynamicData->pvcBorder0 = pvcBorder0; + UCHAR pvcBorder0_last = pPvcStaticData->pvcBorder0; + pPvcStaticData->pvcBorder0 = pvcBorder0; + pPvcDynamicData->pPvcID = pPvcID; + + pPvcDynamicData->ns = ns; + switch (ns) { + case 16: + pPvcDynamicData->pSCcoeffs = pvc_SC_16; + break; + case 12: + pPvcDynamicData->pSCcoeffs = pvc_SC_12; + break; + case 4: + pPvcDynamicData->pSCcoeffs = pvc_SC_4; + break; + case 3: + pPvcDynamicData->pSCcoeffs = pvc_SC_3; + break; + default: + return 1; + } + + /* in the lower part of Esg-array there are previous values of Esg (from last + call to this function In case of an previous legay-SBR frame, or if there + was a change in cross-over FQ the value of first PVC SBR timeslot is + propagated to prev-values in order to have reasonable values for + smooth-filtering + */ + if ((pPvcStaticData->pvc_mode_last == 0) || (pPvcStaticData->kx_last != kx)) { + pPvcDynamicData->pastEsgSlotsAvail = 0; + } else { + pPvcDynamicData->pastEsgSlotsAvail = PVC_NS_MAX - pvcBorder0_last; + } + + lbw = 8 / RATE; + + temp = kx; + for (i = PVC_NBLOW; i >= 0; i--) { + pPvcDynamicData->sg_offset_low[i] = temp; + temp -= lbw; + } + + temp = 0; + for (i = 0; i <= pPvcDynamicData->nbHigh; i++) { + pPvcDynamicData->sg_offset_high_kx[i] = temp; + temp += hbw; + } + + return 0; +} + +/* call if pvcMode = 1,2 */ +void pvcDecodeFrame(PVC_STATIC_DATA *pPvcStaticData, + PVC_DYNAMIC_DATA *pPvcDynamicData, FIXP_DBL **qmfBufferReal, + FIXP_DBL **qmfBufferImag, const int overlap, + const int qmfExponentOverlap, + const int qmfExponentCurrent) { + int t; + FIXP_DBL *predictedEsgSlot; + int RATE = pPvcDynamicData->RATE; + int pvcBorder0 = pPvcDynamicData->pvcBorder0; + + for (t = pvcBorder0; t < PVC_NTIMESLOT; t++) { + int *pPredEsg_exp = &pPvcDynamicData->predEsg_exp[t]; + predictedEsgSlot = pPvcDynamicData->predEsg[t]; + + pvcDecodeTimeSlot( + pPvcStaticData, pPvcDynamicData, &qmfBufferReal[t * RATE], + &qmfBufferImag[t * RATE], + (t * RATE < overlap) ? qmfExponentOverlap : qmfExponentCurrent, + pvcBorder0, t, predictedEsgSlot, pPredEsg_exp); + } + + return; +} + +void pvcDecodeTimeSlot(PVC_STATIC_DATA *pPvcStaticData, + PVC_DYNAMIC_DATA *pPvcDynamicData, + FIXP_DBL **qmfSlotReal, FIXP_DBL **qmfSlotImag, + const int qmfExponent, const int pvcBorder0, + const int timeSlotNumber, FIXP_DBL predictedEsgSlot[], + int *predictedEsg_exp) { + int i, band, ksg, ksg_start = 0; + int RATE = pPvcDynamicData->RATE; + int Esg_index = pPvcStaticData->Esg_slot_index; + const SCHAR *sg_borders = pPvcDynamicData->sg_offset_low; + FIXP_DBL *pEsg = pPvcStaticData->Esg[Esg_index]; + FIXP_DBL E[PVC_NBLOW] = {0}; + + /* Subband grouping in QMF subbands below SBR range */ + /* Within one timeslot ( i = [0...(RATE-1)] QMF subsamples) calculate energy + E(ib,t) and group them to Esg(ksg,t). Then transfer values to logarithmical + domain and store them for time domain smoothing. (7.5.6.3 Subband grouping + in QMF subbands below SBR range) + */ + for (ksg = 0; sg_borders[ksg] < 0; ksg++) { + pEsg[ksg] = FL2FXCONST_DBL(-10.0 / (1 << PVC_ESG_EXP)); /* 10*log10(0.1) */ + ksg_start++; + } + + for (i = 0; i < RATE; i++) { + FIXP_DBL *qmfR, *qmfI; + qmfR = qmfSlotReal[i]; + qmfI = qmfSlotImag[i]; + for (ksg = ksg_start; ksg < PVC_NBLOW; ksg++) { + for (band = sg_borders[ksg]; band < sg_borders[ksg + 1]; band++) { + /* The division by 8 == (RATE*lbw) is required algorithmically */ + E[ksg] += (fPow2Div2(qmfR[band]) + fPow2Div2(qmfI[band])) >> 2; + } + } + } + for (ksg = ksg_start; ksg < PVC_NBLOW; ksg++) { + if (E[ksg] > (FIXP_DBL)0) { + /* 10/log2(10) = 0.752574989159953 * 2^2 */ + int exp_log; + FIXP_DBL nrg = CalcLog2(E[ksg], 2 * qmfExponent, &exp_log); + nrg = fMult(nrg, FL2FXCONST_SGL(LOG10FAC)); + nrg = scaleValue(nrg, exp_log - PVC_ESG_EXP + 2); + pEsg[ksg] = fMax(nrg, FL2FXCONST_DBL(-10.0 / (1 << PVC_ESG_EXP))); + } else { + pEsg[ksg] = + FL2FXCONST_DBL(-10.0 / (1 << PVC_ESG_EXP)); /* 10*log10(0.1) */ + } + } + + /* Time domain smoothing of subband-grouped energy */ + { + int idx = pPvcStaticData->Esg_slot_index; + FIXP_DBL *pEsg_filt; + FIXP_SGL SCcoeff; + + E[0] = E[1] = E[2] = (FIXP_DBL)0; + for (i = 0; i < pPvcDynamicData->ns; i++) { + SCcoeff = pPvcDynamicData->pSCcoeffs[i]; + pEsg_filt = pPvcStaticData->Esg[idx]; + /* Div2 is compensated by scaling of coeff table */ + E[0] = fMultAddDiv2(E[0], pEsg_filt[0], SCcoeff); + E[1] = fMultAddDiv2(E[1], pEsg_filt[1], SCcoeff); + E[2] = fMultAddDiv2(E[2], pEsg_filt[2], SCcoeff); + if (i >= pPvcDynamicData->pastEsgSlotsAvail) { + /* if past Esg values are not available use the ones from the last valid + * slot */ + continue; + } + if (idx > 0) { + idx--; + } else { + idx += PVC_NS_MAX - 1; + } + } + } + + /* SBR envelope scalefactor prediction */ + { + int E_high_exp[PVC_NBHIGH_MAX]; + int E_high_exp_max = 0; + int pvcTab1ID; + int pvcTab2ID = (int)pPvcDynamicData->pPvcID[timeSlotNumber]; + const UCHAR *pTab1, *pTab2; + if (pvcTab2ID < pPvcDynamicData->pPVCTab1_dp[0]) { + pvcTab1ID = 0; + } else if (pvcTab2ID < pPvcDynamicData->pPVCTab1_dp[1]) { + pvcTab1ID = 1; + } else { + pvcTab1ID = 2; + } + pTab1 = &(pPvcDynamicData + ->pPVCTab1[pvcTab1ID * PVC_NBLOW * pPvcDynamicData->nbHigh]); + pTab2 = &(pPvcDynamicData->pPVCTab2[pvcTab2ID * pPvcDynamicData->nbHigh]); + for (ksg = 0; ksg < pPvcDynamicData->nbHigh; ksg++) { + FIXP_SGL predCoeff; + FIXP_DBL accu; + int predCoeff_exp, kb; + E_high_exp[ksg] = 0; + + /* residual part */ + accu = ((LONG)(SCHAR)*pTab2++) << (DFRACT_BITS - 8 - PVC_ESG_EXP + + pPvcDynamicData->pScalingCoef[3]); + + /* linear combination of lower grouped energies part */ + for (kb = 0; kb < PVC_NBLOW; kb++) { + predCoeff = (FIXP_SGL)( + (SHORT)(SCHAR)pTab1[kb * pPvcDynamicData->nbHigh + ksg] << 8); + predCoeff_exp = pPvcDynamicData->pScalingCoef[kb] + + 1; /* +1 to compensate for Div2 */ + accu += fMultDiv2(E[kb], predCoeff) << predCoeff_exp; + } + /* convert back to linear domain */ + accu = fMult(accu, FL2FXCONST_SGL(LOG10FAC_INV)); + accu = f2Pow( + accu, PVC_ESG_EXP - 1, + &predCoeff_exp); /* -1 compensates for exponent of LOG10FAC_INV */ + predictedEsgSlot[ksg] = accu; + E_high_exp[ksg] = predCoeff_exp; + if (predCoeff_exp > E_high_exp_max) { + E_high_exp_max = predCoeff_exp; + } + } + + /* rescale output vector according to largest exponent */ + for (ksg = 0; ksg < pPvcDynamicData->nbHigh; ksg++) { + int scale = E_high_exp[ksg] - E_high_exp_max; + predictedEsgSlot[ksg] = scaleValue(predictedEsgSlot[ksg], scale); + } + *predictedEsg_exp = E_high_exp_max; + } + + pPvcStaticData->Esg_slot_index = + (pPvcStaticData->Esg_slot_index + 1) & (PVC_NS_MAX - 1); + pPvcDynamicData->pastEsgSlotsAvail = + fMin(pPvcDynamicData->pastEsgSlotsAvail + 1, PVC_NS_MAX - 1); + return; +} + +/* call if pvcMode = 0,1,2 */ +void pvcEndFrame(PVC_STATIC_DATA *pPvcStaticData, + PVC_DYNAMIC_DATA *pPvcDynamicData) { + pPvcStaticData->pvc_mode_last = pPvcDynamicData->pvc_mode; + pPvcStaticData->kx_last = pPvcDynamicData->kx; + + if (pPvcDynamicData->pvc_mode == 0) return; + + { + int t, max = -100; + for (t = pPvcDynamicData->pvcBorder0; t < PVC_NTIMESLOT; t++) { + if (pPvcDynamicData->predEsg_exp[t] > max) { + max = pPvcDynamicData->predEsg_exp[t]; + } + } + pPvcDynamicData->predEsg_expMax = max; + } + return; +} + +void expandPredEsg(const PVC_DYNAMIC_DATA *pPvcDynamicData, const int timeSlot, + const int lengthOutputVector, FIXP_DBL *pOutput, + SCHAR *pOutput_exp) { + int k = 0, ksg; + const FIXP_DBL *predEsg = pPvcDynamicData->predEsg[timeSlot]; + + for (ksg = 0; ksg < pPvcDynamicData->nbHigh; ksg++) { + for (; k < pPvcDynamicData->sg_offset_high_kx[ksg + 1]; k++) { + pOutput[k] = predEsg[ksg]; + pOutput_exp[k] = (SCHAR)pPvcDynamicData->predEsg_exp[timeSlot]; + } + } + ksg--; + for (; k < lengthOutputVector; k++) { + pOutput[k] = predEsg[ksg]; + pOutput_exp[k] = (SCHAR)pPvcDynamicData->predEsg_exp[timeSlot]; + } + + return; +} diff --git a/fdk-aac/libSBRdec/src/pvc_dec.h b/fdk-aac/libSBRdec/src/pvc_dec.h new file mode 100644 index 0000000..f5a467f --- /dev/null +++ b/fdk-aac/libSBRdec/src/pvc_dec.h @@ -0,0 +1,238 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): Matthias Hildenbrand + + Description: Decode Predictive Vector Coding Data + +*******************************************************************************/ + +#ifndef PVC_DEC_H +#define PVC_DEC_H + +#include "common_fix.h" + +#define PVC_DIVMODE_BITS 3 +#define PVC_REUSEPVCID_BITS 1 +#define PVC_PVCID_BITS 7 +#define PVC_GRIDINFO_BITS 1 + +#define MAX_PVC_ENVELOPES 2 +#define PVC_NTIMESLOT 16 +#define PVC_NBLOW 3 /* max. number of grouped QMF subbands below SBR range */ + +#define PVC_NBHIGH_MODE1 8 +#define PVC_NBHIGH_MODE2 6 +#define PVC_NBHIGH_MAX (PVC_NBHIGH_MODE1) +#define PVC_NS_MAX 16 + +/** Data for each PVC instance which needs to be persistent accross SBR frames + */ +typedef struct { + UCHAR kx_last; /**< Xover frequency of last frame */ + UCHAR pvc_mode_last; /**< PVC mode of last frame */ + UCHAR Esg_slot_index; /**< Ring buffer index to current Esg time slot */ + UCHAR pvcBorder0; /**< Start SBR time slot of PVC frame */ + FIXP_DBL Esg[PVC_NS_MAX][PVC_NBLOW]; /**< Esg(ksg,t) of current and 15 + previous time slots (ring buffer) in + logarithmical domain */ +} PVC_STATIC_DATA; + +/** Data for each PVC instance which is valid during one SBR frame */ +typedef struct { + UCHAR pvc_mode; /**< PVC mode 1 or 2, 0 means legacy SBR */ + UCHAR pvcBorder0; /**< Start SBR time slot of PVC frame */ + UCHAR kx; /**< Index of the first QMF subband in the SBR range */ + UCHAR RATE; /**< Number of QMF subband samples per time slot (2 or 4) */ + UCHAR ns; /**< Number of time slots for time-domain smoothing of Esg(ksg,t) */ + const UCHAR + *pPvcID; /**< Pointer to prediction coefficient matrix index table */ + UCHAR pastEsgSlotsAvail; /**< Number of past Esg(ksg,t) which are available + for smoothing filter */ + const FIXP_SGL *pSCcoeffs; /**< Pointer to smoothing window table */ + SCHAR + sg_offset_low[PVC_NBLOW + 1]; /**< Offset table for PVC grouping of SBR + subbands below SBR range */ + SCHAR sg_offset_high_kx[PVC_NBHIGH_MAX + 1]; /**< Offset table for PVC + grouping of SBR subbands in + SBR range (relativ to kx) */ + UCHAR nbHigh; /**< Number of grouped QMF subbands in the SBR range */ + const SCHAR *pScalingCoef; /**< Pointer to scaling coeff table */ + const UCHAR *pPVCTab1; /**< PVC mode 1 table */ + const UCHAR *pPVCTab2; /**< PVC mode 2 table */ + const UCHAR *pPVCTab1_dp; /**< Mapping of pvcID to PVC mode 1 table */ + FIXP_DBL predEsg[PVC_NTIMESLOT] + [PVC_NBHIGH_MAX]; /**< Predicted Energy in linear domain */ + int predEsg_exp[PVC_NTIMESLOT]; /**< Exponent of predicted Energy in linear + domain */ + int predEsg_expMax; /**< Maximum of predEsg_exp[] */ +} PVC_DYNAMIC_DATA; + +/** + * \brief Initialize PVC data structures for current frame (call if pvcMode = + * 0,1,2) + * \param[in] pPvcStaticData Pointer to PVC persistent data + * \param[out] pPvcDynamicData Pointer to PVC dynamic data + * \param[in] pvcMode PVC mode 1 or 2, 0 means legacy SBR + * \param[in] ns Number of time slots for time-domain smoothing of Esg(ksg,t) + * \param[in] RATE Number of QMF subband samples per time slot (2 or 4) + * \param[in] kx Index of the first QMF subband in the SBR range + * \param[in] pvcBorder0 Start SBR time slot of PVC frame + * \param[in] pPvcID Pointer to array of PvcIDs read from bitstream + */ +int pvcInitFrame(PVC_STATIC_DATA *pPvcStaticData, + PVC_DYNAMIC_DATA *pPvcDynamicData, const UCHAR pvcMode, + const UCHAR ns, const int RATE, const int kx, + const int pvcBorder0, const UCHAR *pPvcID); + +/** + * \brief Wrapper function for pvcDecodeTimeSlot() to decode PVC data of one + * frame (call if pvcMode = 1,2) + * \param[in,out] pPvcStaticData Pointer to PVC persistent data + * \param[in,out] pPvcDynamicData Pointer to PVC dynamic data + * \param[in] qmfBufferReal Pointer to array with real QMF subbands + * \param[in] qmfBufferImag Pointer to array with imag QMF subbands + * \param[in] overlap Number of QMF overlap slots + * \param[in] qmfExponentOverlap Exponent of qmfBuffer (low part) of overlap + * slots + * \param[in] qmfExponentCurrent Exponent of qmfBuffer (low part) + */ +void pvcDecodeFrame(PVC_STATIC_DATA *pPvcStaticData, + PVC_DYNAMIC_DATA *pPvcDynamicData, FIXP_DBL **qmfBufferReal, + FIXP_DBL **qmfBufferImag, const int overlap, + const int qmfExponentOverlap, const int qmfExponentCurrent); + +/** + * \brief Decode PVC data for one SBR time slot (call if pvcMode = 1,2) + * \param[in,out] pPvcStaticData Pointer to PVC persistent data + * \param[in,out] pPvcDynamicData Pointer to PVC dynamic data + * \param[in] qmfBufferReal Pointer to array with real QMF subbands + * \param[in] qmfBufferImag Pointer to array with imag QMF subbands + * \param[in] qmfExponent Exponent of qmfBuffer of current time slot + * \param[in] pvcBorder0 Start SBR time slot of PVC frame + * \param[in] timeSlotNumber Number of current SBR time slot (0..15) + * \param[out] predictedEsgSlot Predicted Energy of current time slot + * \param[out] predictedEsg_exp Exponent of predicted Energy of current time + * slot + */ +void pvcDecodeTimeSlot(PVC_STATIC_DATA *pPvcStaticData, + PVC_DYNAMIC_DATA *pPvcDynamicData, + FIXP_DBL **qmfSlotReal, FIXP_DBL **qmfSlotImag, + const int qmfExponent, const int pvcBorder0, + const int timeSlotNumber, FIXP_DBL predictedEsgSlot[], + int *predictedEsg_exp); + +/** + * \brief Finish the current PVC frame (call if pvcMode = 0,1,2) + * \param[in,out] pPvcStaticData Pointer to PVC persistent data + * \param[in,out] pPvcDynamicData Pointer to PVC dynamic data + */ +void pvcEndFrame(PVC_STATIC_DATA *pPvcStaticData, + PVC_DYNAMIC_DATA *pPvcDynamicData); + +/** + * \brief Expand predicted PVC grouped energies to full QMF subband resolution + * \param[in] pPvcDynamicData Pointer to PVC dynamic data + * \param[in] timeSlot Number of current SBR time slot (0..15) + * \param[in] lengthOutputVector Lenght of output vector + * \param[out] pOutput Output array for predicted energies + * \param[out] pOutput_exp Exponent of predicted energies + */ +void expandPredEsg(const PVC_DYNAMIC_DATA *pPvcDynamicData, const int timeSlot, + const int lengthOutputVector, FIXP_DBL *pOutput, + SCHAR *pOutput_exp); + +#endif /* PVC_DEC_H*/ diff --git a/fdk-aac/libSBRdec/src/sbr_crc.cpp b/fdk-aac/libSBRdec/src/sbr_crc.cpp new file mode 100644 index 0000000..ba0fd05 --- /dev/null +++ b/fdk-aac/libSBRdec/src/sbr_crc.cpp @@ -0,0 +1,192 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief CRC check coutines +*/ + +#include "sbr_crc.h" + +#include "FDK_bitstream.h" +#include "transcendent.h" + +#define MAXCRCSTEP 16 +#define MAXCRCSTEP_LD 4 + +/*! + \brief crc calculation +*/ +static ULONG calcCRC(HANDLE_CRC hCrcBuf, ULONG bValue, int nBits) { + int i; + ULONG bMask = (1UL << (nBits - 1)); + + for (i = 0; i < nBits; i++, bMask >>= 1) { + USHORT flag = (hCrcBuf->crcState & hCrcBuf->crcMask) ? 1 : 0; + USHORT flag1 = (bMask & bValue) ? 1 : 0; + + flag ^= flag1; + hCrcBuf->crcState <<= 1; + if (flag) hCrcBuf->crcState ^= hCrcBuf->crcPoly; + } + + return (hCrcBuf->crcState); +} + +/*! + \brief crc +*/ +static int getCrc(HANDLE_FDK_BITSTREAM hBs, ULONG NrBits) { + int i; + CRC_BUFFER CrcBuf; + + CrcBuf.crcState = SBR_CRC_START; + CrcBuf.crcPoly = SBR_CRC_POLY; + CrcBuf.crcMask = SBR_CRC_MASK; + + int CrcStep = NrBits >> MAXCRCSTEP_LD; + + int CrcNrBitsRest = (NrBits - CrcStep * MAXCRCSTEP); + ULONG bValue; + + for (i = 0; i < CrcStep; i++) { + bValue = FDKreadBits(hBs, MAXCRCSTEP); + calcCRC(&CrcBuf, bValue, MAXCRCSTEP); + } + + bValue = FDKreadBits(hBs, CrcNrBitsRest); + calcCRC(&CrcBuf, bValue, CrcNrBitsRest); + + return (CrcBuf.crcState & SBR_CRC_RANGE); +} + +/*! + \brief crc interface + \return 1: CRC OK, 0: CRC check failure +*/ +int SbrCrcCheck(HANDLE_FDK_BITSTREAM hBs, /*!< handle to bit-buffer */ + LONG NrBits) /*!< max. CRC length */ +{ + int crcResult = 1; + ULONG NrCrcBits; + ULONG crcCheckResult; + LONG NrBitsAvailable; + ULONG crcCheckSum; + + crcCheckSum = FDKreadBits(hBs, 10); + + NrBitsAvailable = FDKgetValidBits(hBs); + if (NrBitsAvailable <= 0) { + return 0; + } + + NrCrcBits = fixMin((INT)NrBits, (INT)NrBitsAvailable); + + crcCheckResult = getCrc(hBs, NrCrcBits); + FDKpushBack(hBs, (NrBitsAvailable - FDKgetValidBits(hBs))); + + if (crcCheckResult != crcCheckSum) { + crcResult = 0; + } + + return (crcResult); +} diff --git a/fdk-aac/libSBRdec/src/sbr_crc.h b/fdk-aac/libSBRdec/src/sbr_crc.h new file mode 100644 index 0000000..9633717 --- /dev/null +++ b/fdk-aac/libSBRdec/src/sbr_crc.h @@ -0,0 +1,138 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief CRC checking routines +*/ +#ifndef SBR_CRC_H +#define SBR_CRC_H + +#include "sbrdecoder.h" + +#include "FDK_bitstream.h" + +/* some useful crc polynoms: + +crc5: x^5+x^4+x^2+x^1+1 +crc6: x^6+x^5+x^3+x^2+x+1 +crc7: x^7+x^6+x^2+1 +crc8: x^8+x^2+x+x+1 +*/ + +/* default SBR CRC */ /* G(x) = x^10 + x^9 + x^5 + x^4 + x + 1 */ +#define SBR_CRC_POLY 0x0233 +#define SBR_CRC_MASK 0x0200 +#define SBR_CRC_START 0x0000 +#define SBR_CRC_RANGE 0x03FF + +typedef struct { + USHORT crcState; + USHORT crcMask; + USHORT crcPoly; +} CRC_BUFFER; + +typedef CRC_BUFFER *HANDLE_CRC; + +int SbrCrcCheck(HANDLE_FDK_BITSTREAM hBitBuf, LONG NrCrcBits); + +#endif diff --git a/fdk-aac/libSBRdec/src/sbr_deb.cpp b/fdk-aac/libSBRdec/src/sbr_deb.cpp new file mode 100644 index 0000000..13cd211 --- /dev/null +++ b/fdk-aac/libSBRdec/src/sbr_deb.cpp @@ -0,0 +1,108 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief Print selected debug messages +*/ + +#include "sbr_deb.h" diff --git a/fdk-aac/libSBRdec/src/sbr_deb.h b/fdk-aac/libSBRdec/src/sbr_deb.h new file mode 100644 index 0000000..97d572a --- /dev/null +++ b/fdk-aac/libSBRdec/src/sbr_deb.h @@ -0,0 +1,113 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief Debugging aids +*/ + +#ifndef SBR_DEB_H +#define SBR_DEB_H + +#include "sbrdecoder.h" + +#endif diff --git a/fdk-aac/libSBRdec/src/sbr_dec.cpp b/fdk-aac/libSBRdec/src/sbr_dec.cpp new file mode 100644 index 0000000..30611e7 --- /dev/null +++ b/fdk-aac/libSBRdec/src/sbr_dec.cpp @@ -0,0 +1,1480 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief Sbr decoder + This module provides the actual decoder implementation. The SBR data (side + information) is already decoded. Only three functions are provided: + + \li 1.) createSbrDec(): One time initialization + \li 2.) resetSbrDec(): Called by sbr_Apply() when the information contained in + an SBR_HEADER_ELEMENT requires a reset and recalculation of important SBR + structures. \li 3.) sbr_dec(): The actual decoder. Calls the different tools + such as filterbanks, lppTransposer(), and calculateSbrEnvelope() [the envelope + adjuster]. + + \sa sbr_dec(), \ref documentationOverview +*/ + +#include "sbr_dec.h" + +#include "sbr_ram.h" +#include "env_extr.h" +#include "env_calc.h" +#include "scale.h" +#include "FDK_matrixCalloc.h" +#include "hbe.h" + +#include "genericStds.h" + +#include "sbrdec_drc.h" + +static void copyHarmonicSpectrum(int *xOverQmf, FIXP_DBL **qmfReal, + FIXP_DBL **qmfImag, int noCols, int overlap, + KEEP_STATES_SYNCED_MODE keepStatesSynced) { + int patchBands; + int patch, band, col, target, sourceBands, i; + int numPatches = 0; + int slotOffset = 0; + + FIXP_DBL **ppqmfReal = qmfReal + overlap; + FIXP_DBL **ppqmfImag = qmfImag + overlap; + + if (keepStatesSynced == KEEP_STATES_SYNCED_NORMAL) { + slotOffset = noCols - overlap - LPC_ORDER; + } + + if (keepStatesSynced == KEEP_STATES_SYNCED_OUTDIFF) { + ppqmfReal = qmfReal; + ppqmfImag = qmfImag; + } + + for (i = 1; i < MAX_NUM_PATCHES; i++) { + if (xOverQmf[i] != 0) { + numPatches++; + } + } + + for (patch = (MAX_STRETCH_HBE - 1); patch < numPatches; patch++) { + patchBands = xOverQmf[patch + 1] - xOverQmf[patch]; + target = xOverQmf[patch]; + sourceBands = xOverQmf[MAX_STRETCH_HBE - 1] - xOverQmf[MAX_STRETCH_HBE - 2]; + + while (patchBands > 0) { + int numBands = sourceBands; + int startBand = xOverQmf[MAX_STRETCH_HBE - 1] - 1; + if (target + numBands >= xOverQmf[patch + 1]) { + numBands = xOverQmf[patch + 1] - target; + } + if ((((target + numBands - 1) % 2) + + ((xOverQmf[MAX_STRETCH_HBE - 1] - 1) % 2)) % + 2) { + if (numBands == sourceBands) { + numBands--; + } else { + startBand--; + } + } + if (keepStatesSynced == KEEP_STATES_SYNCED_OUTDIFF) { + for (col = slotOffset; col < overlap + LPC_ORDER; col++) { + i = 0; + for (band = numBands; band > 0; band--) { + if ((target + band - 1 < 64) && + (target + band - 1 < xOverQmf[patch + 1])) { + ppqmfReal[col][target + band - 1] = ppqmfReal[col][startBand - i]; + ppqmfImag[col][target + band - 1] = ppqmfImag[col][startBand - i]; + i++; + } + } + } + } else { + for (col = slotOffset; col < noCols; col++) { + i = 0; + for (band = numBands; band > 0; band--) { + if ((target + band - 1 < 64) && + (target + band - 1 < xOverQmf[patch + 1])) { + ppqmfReal[col][target + band - 1] = ppqmfReal[col][startBand - i]; + ppqmfImag[col][target + band - 1] = ppqmfImag[col][startBand - i]; + i++; + } + } + } + } + target += numBands; + patchBands -= numBands; + } + } +} + +/*! + \brief SBR decoder core function for one channel + + \image html BufferMgmtDetailed-1632.png + + Besides the filter states of the QMF filter bank and the LPC-states of + the LPP-Transposer, processing is mainly based on four buffers: + #timeIn, #timeOut, #WorkBuffer2 and #OverlapBuffer. The #WorkBuffer2 + is reused for all channels and might be used by the core decoder, a + static overlap buffer is required for each channel. Due to in-place + processing, #timeIn and #timeOut point to identical locations. + + The spectral data is organized in so-called slots. Each slot + contains 64 bands of complex data. The number of slots per frame + depends on the frame size. For mp3PRO, there are 18 slots per frame + and 6 slots per #OverlapBuffer. It is not necessary to have the slots + in located consecutive address ranges. + + To optimize memory usage and to minimize the number of memory + accesses, the memory management is organized as follows (slot numbers + based on mp3PRO): + + 1.) Input time domain signal is located in #timeIn. The last slots + (0..5) of the spectral data of the previous frame are located in the + #OverlapBuffer. In addition, #frameData of the current frame resides + in the upper part of #timeIn. + + 2.) During the cplxAnalysisQmfFiltering(), 32 samples from #timeIn are + transformed into a slot of up to 32 complex spectral low band values at a + time. The first spectral slot -- nr. 6 -- is written at slot number + zero of #WorkBuffer2. #WorkBuffer2 will be completely filled with + spectral data. + + 3.) LPP-Transposition in lppTransposer() is processed on 24 slots. During the + transposition, the high band part of the spectral data is replicated + based on the low band data. + + Envelope Adjustment is processed on the high band part of the spectral + data only by calculateSbrEnvelope(). + + 4.) The cplxSynthesisQmfFiltering() creates 64 time domain samples out + of a slot of 64 complex spectral values at a time. The first 6 slots + in #timeOut are filled from the results of spectral slots 0..5 in the + #OverlapBuffer. The consecutive slots in timeOut are now filled with + the results of spectral slots 6..17. + + 5.) The preprocessed slots 18..23 have to be stored in the + #OverlapBuffer. + +*/ + +void sbr_dec( + HANDLE_SBR_DEC hSbrDec, /*!< handle to Decoder channel */ + INT_PCM *timeIn, /*!< pointer to input time signal */ + INT_PCM *timeOut, /*!< pointer to output time signal */ + HANDLE_SBR_DEC hSbrDecRight, /*!< handle to Decoder channel right */ + INT_PCM *timeOutRight, /*!< pointer to output time signal */ + const int strideOut, /*!< Time data traversal strideOut */ + HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ + HANDLE_SBR_FRAME_DATA hFrameData, /*!< Control data of current frame */ + HANDLE_SBR_PREV_FRAME_DATA + hPrevFrameData, /*!< Some control data of last frame */ + const int applyProcessing, /*!< Flag for SBR operation */ + HANDLE_PS_DEC h_ps_d, const UINT flags, const int codecFrameSize) { + int i, slot, reserve; + int saveLbScale; + int lastSlotOffs; + FIXP_DBL maxVal; + + /* temporary pointer / variable for QMF; + required as we want to use temporary buffer + creating one frame delay for HBE in LP mode */ + INT_PCM *pTimeInQmf = timeIn; + + /* Number of QMF timeslots in the overlap buffer: */ + int ov_len = hSbrDec->LppTrans.pSettings->overlap; + + /* Number of QMF slots per frame */ + int noCols = hHeaderData->numberTimeSlots * hHeaderData->timeStep; + + /* create pointer array for data to use for HBE and legacy sbr */ + FIXP_DBL *pLowBandReal[(3 * 4) + 2 * ((1024) / (32) * (4) / 2)]; + FIXP_DBL *pLowBandImag[(3 * 4) + 2 * ((1024) / (32) * (4) / 2)]; + + /* set pReal to where QMF analysis writes in case of legacy SBR */ + FIXP_DBL **pReal = pLowBandReal + ov_len; + FIXP_DBL **pImag = pLowBandImag + ov_len; + + /* map QMF buffer to pointer array (Overlap + Frame)*/ + for (i = 0; i < noCols + ov_len; i++) { + pLowBandReal[i] = hSbrDec->qmfDomainInCh->hQmfSlotsReal[i]; + pLowBandImag[i] = hSbrDec->qmfDomainInCh->hQmfSlotsImag[i]; + } + + if ((flags & SBRDEC_USAC_HARMONICSBR)) { + /* in case of harmonic SBR and no HBE_LP map additional buffer for + one more frame to pointer arry */ + for (i = 0; i < noCols; i++) { + pLowBandReal[i + noCols + ov_len] = hSbrDec->hQmfHBESlotsReal[i]; + pLowBandImag[i + noCols + ov_len] = hSbrDec->hQmfHBESlotsImag[i]; + } + + /* shift scale values according to buffer */ + hSbrDec->scale_ov = hSbrDec->scale_lb; + hSbrDec->scale_lb = hSbrDec->scale_hbe; + + /* set pReal to where QMF analysis writes in case of HBE */ + pReal += noCols; + pImag += noCols; + if (flags & SBRDEC_SKIP_QMF_ANA) { + /* stereoCfgIndex3 with HBE */ + FDK_QmfDomain_QmfData2HBE(hSbrDec->qmfDomainInCh, + hSbrDec->hQmfHBESlotsReal, + hSbrDec->hQmfHBESlotsImag); + } else { + /* We have to move old hbe frame data to lb area of buffer */ + for (i = 0; i < noCols; i++) { + FDKmemcpy(pLowBandReal[ov_len + i], hSbrDec->hQmfHBESlotsReal[i], + hHeaderData->numberOfAnalysisBands * sizeof(FIXP_DBL)); + FDKmemcpy(pLowBandImag[ov_len + i], hSbrDec->hQmfHBESlotsImag[i], + hHeaderData->numberOfAnalysisBands * sizeof(FIXP_DBL)); + } + } + } + + /* + low band codec signal subband filtering + */ + + if (flags & SBRDEC_SKIP_QMF_ANA) { + if (!(flags & SBRDEC_USAC_HARMONICSBR)) /* stereoCfgIndex3 w/o HBE */ + FDK_QmfDomain_WorkBuffer2ProcChannel(hSbrDec->qmfDomainInCh); + } else { + C_AALLOC_SCRATCH_START(qmfTemp, FIXP_DBL, 2 * (64)); + qmfAnalysisFiltering(&hSbrDec->qmfDomainInCh->fb, pReal, pImag, + &hSbrDec->qmfDomainInCh->scaling, pTimeInQmf, 0, 1, + qmfTemp); + + C_AALLOC_SCRATCH_END(qmfTemp, FIXP_DBL, 2 * (64)); + } + + /* + Clear upper half of spectrum + */ + if (!((flags & SBRDEC_USAC_HARMONICSBR) && + (hFrameData->sbrPatchingMode == 0))) { + int nAnalysisBands = hHeaderData->numberOfAnalysisBands; + + if (!(flags & SBRDEC_LOW_POWER)) { + for (slot = ov_len; slot < noCols + ov_len; slot++) { + FDKmemclear(&pLowBandReal[slot][nAnalysisBands], + ((64) - nAnalysisBands) * sizeof(FIXP_DBL)); + FDKmemclear(&pLowBandImag[slot][nAnalysisBands], + ((64) - nAnalysisBands) * sizeof(FIXP_DBL)); + } + } else { + for (slot = ov_len; slot < noCols + ov_len; slot++) { + FDKmemclear(&pLowBandReal[slot][nAnalysisBands], + ((64) - nAnalysisBands) * sizeof(FIXP_DBL)); + } + } + } + + /* + Shift spectral data left to gain accuracy in transposer and adjustor + */ + /* Range was increased from lsb to no_channels because in some cases (e.g. + USAC conf eSbr_4_Pvc.mp4 and some HBE cases) it could be observed that the + signal between lsb and no_channels is used for the patching process. + */ + maxVal = maxSubbandSample(pReal, (flags & SBRDEC_LOW_POWER) ? NULL : pImag, 0, + hSbrDec->qmfDomainInCh->fb.no_channels, 0, noCols); + + reserve = fixMax(0, CntLeadingZeros(maxVal) - 1); + reserve = fixMin(reserve, + DFRACT_BITS - 1 - hSbrDec->qmfDomainInCh->scaling.lb_scale); + + /* If all data is zero, lb_scale could become too large */ + rescaleSubbandSamples(pReal, (flags & SBRDEC_LOW_POWER) ? NULL : pImag, 0, + hSbrDec->qmfDomainInCh->fb.no_channels, 0, noCols, + reserve); + + hSbrDec->qmfDomainInCh->scaling.lb_scale += reserve; + + if ((flags & SBRDEC_USAC_HARMONICSBR)) { + /* actually this is our hbe_scale */ + hSbrDec->scale_hbe = hSbrDec->qmfDomainInCh->scaling.lb_scale; + /* the real lb_scale is stored in scale_lb from sbr */ + hSbrDec->qmfDomainInCh->scaling.lb_scale = hSbrDec->scale_lb; + } + /* + save low band scale, wavecoding or parametric stereo may modify it + */ + saveLbScale = hSbrDec->qmfDomainInCh->scaling.lb_scale; + + if (applyProcessing) { + UCHAR *borders = hFrameData->frameInfo.borders; + lastSlotOffs = borders[hFrameData->frameInfo.nEnvelopes] - + hHeaderData->numberTimeSlots; + + FIXP_DBL degreeAlias[(64)]; + PVC_DYNAMIC_DATA pvcDynamicData; + pvcInitFrame( + &hSbrDec->PvcStaticData, &pvcDynamicData, + (hHeaderData->frameErrorFlag ? 0 : hHeaderData->bs_info.pvc_mode), + hFrameData->ns, hHeaderData->timeStep, + hHeaderData->freqBandData.lowSubband, + hFrameData->frameInfo.pvcBorders[0], hFrameData->pvcID); + + if (!hHeaderData->frameErrorFlag && (hHeaderData->bs_info.pvc_mode > 0)) { + pvcDecodeFrame(&hSbrDec->PvcStaticData, &pvcDynamicData, pLowBandReal, + pLowBandImag, ov_len, + SCALE2EXP(hSbrDec->qmfDomainInCh->scaling.ov_lb_scale), + SCALE2EXP(hSbrDec->qmfDomainInCh->scaling.lb_scale)); + } + pvcEndFrame(&hSbrDec->PvcStaticData, &pvcDynamicData); + + /* The transposer will override most values in degreeAlias[]. + The array needs to be cleared at least from lowSubband to highSubband + before. */ + if (flags & SBRDEC_LOW_POWER) + FDKmemclear(°reeAlias[hHeaderData->freqBandData.lowSubband], + (hHeaderData->freqBandData.highSubband - + hHeaderData->freqBandData.lowSubband) * + sizeof(FIXP_DBL)); + + /* + Inverse filtering of lowband and transposition into the SBR-frequency + range + */ + + { + KEEP_STATES_SYNCED_MODE keepStatesSyncedMode = + ((flags & SBRDEC_USAC_HARMONICSBR) && + (hFrameData->sbrPatchingMode != 0)) + ? KEEP_STATES_SYNCED_NORMAL + : KEEP_STATES_SYNCED_OFF; + + if (flags & SBRDEC_USAC_HARMONICSBR) { + if (flags & SBRDEC_QUAD_RATE) { + pReal -= 32; + pImag -= 32; + } + + if ((hSbrDec->savedStates == 0) && (hFrameData->sbrPatchingMode == 1)) { + /* copy saved states from previous frame to legacy SBR lpc filterstate + * buffer */ + for (i = 0; i < LPC_ORDER + ov_len; i++) { + FDKmemcpy( + hSbrDec->LppTrans.lpcFilterStatesRealLegSBR[i], + hSbrDec->codecQMFBufferReal[noCols - LPC_ORDER - ov_len + i], + hSbrDec->hHBE->noChannels * sizeof(FIXP_DBL)); + FDKmemcpy( + hSbrDec->LppTrans.lpcFilterStatesImagLegSBR[i], + hSbrDec->codecQMFBufferImag[noCols - LPC_ORDER - ov_len + i], + hSbrDec->hHBE->noChannels * sizeof(FIXP_DBL)); + } + } + + /* saving unmodified QMF states in case we are switching from legacy SBR + * to HBE */ + for (i = 0; i < hSbrDec->hHBE->noCols; i++) { + FDKmemcpy(hSbrDec->codecQMFBufferReal[i], pLowBandReal[ov_len + i], + hSbrDec->hHBE->noChannels * sizeof(FIXP_DBL)); + FDKmemcpy(hSbrDec->codecQMFBufferImag[i], pLowBandImag[ov_len + i], + hSbrDec->hHBE->noChannels * sizeof(FIXP_DBL)); + } + + QmfTransposerApply( + hSbrDec->hHBE, pReal, pImag, noCols, pLowBandReal, pLowBandImag, + hSbrDec->LppTrans.lpcFilterStatesRealHBE, + hSbrDec->LppTrans.lpcFilterStatesImagHBE, + hFrameData->sbrPitchInBins, hSbrDec->scale_lb, hSbrDec->scale_hbe, + &hSbrDec->qmfDomainInCh->scaling.hb_scale, hHeaderData->timeStep, + borders[0], ov_len, keepStatesSyncedMode); + + if (flags & SBRDEC_QUAD_RATE) { + int *xOverQmf = GetxOverBandQmfTransposer(hSbrDec->hHBE); + + copyHarmonicSpectrum(xOverQmf, pLowBandReal, pLowBandImag, noCols, + ov_len, keepStatesSyncedMode); + } + } + } + + if ((flags & SBRDEC_USAC_HARMONICSBR) && + (hFrameData->sbrPatchingMode == 0)) { + hSbrDec->prev_frame_lSbr = 0; + hSbrDec->prev_frame_hbeSbr = 1; + + lppTransposerHBE( + &hSbrDec->LppTrans, hSbrDec->hHBE, &hSbrDec->qmfDomainInCh->scaling, + pLowBandReal, pLowBandImag, hHeaderData->timeStep, borders[0], + lastSlotOffs, hHeaderData->freqBandData.nInvfBands, + hFrameData->sbr_invf_mode, hPrevFrameData->sbr_invf_mode); + + } else { + if (flags & SBRDEC_USAC_HARMONICSBR) { + for (i = 0; i < LPC_ORDER + hSbrDec->LppTrans.pSettings->overlap; i++) { + /* + Store the unmodified qmf Slots values for upper part of spectrum + (required for LPC filtering) required if next frame is a HBE frame + */ + FDKmemcpy(hSbrDec->LppTrans.lpcFilterStatesRealHBE[i], + hSbrDec->qmfDomainInCh + ->hQmfSlotsReal[hSbrDec->hHBE->noCols - LPC_ORDER + i], + (64) * sizeof(FIXP_DBL)); + FDKmemcpy(hSbrDec->LppTrans.lpcFilterStatesImagHBE[i], + hSbrDec->qmfDomainInCh + ->hQmfSlotsImag[hSbrDec->hHBE->noCols - LPC_ORDER + i], + (64) * sizeof(FIXP_DBL)); + } + } + { + hSbrDec->prev_frame_lSbr = 1; + hSbrDec->prev_frame_hbeSbr = 0; + } + + lppTransposer( + &hSbrDec->LppTrans, &hSbrDec->qmfDomainInCh->scaling, pLowBandReal, + degreeAlias, // only used if useLP = 1 + pLowBandImag, flags & SBRDEC_LOW_POWER, + hHeaderData->bs_info.sbr_preprocessing, + hHeaderData->freqBandData.v_k_master[0], hHeaderData->timeStep, + borders[0], lastSlotOffs, hHeaderData->freqBandData.nInvfBands, + hFrameData->sbr_invf_mode, hPrevFrameData->sbr_invf_mode); + } + + /* + Adjust envelope of current frame. + */ + + if ((hFrameData->sbrPatchingMode != + hSbrDec->SbrCalculateEnvelope.sbrPatchingMode)) { + ResetLimiterBands(hHeaderData->freqBandData.limiterBandTable, + &hHeaderData->freqBandData.noLimiterBands, + hHeaderData->freqBandData.freqBandTable[0], + hHeaderData->freqBandData.nSfb[0], + hSbrDec->LppTrans.pSettings->patchParam, + hSbrDec->LppTrans.pSettings->noOfPatches, + hHeaderData->bs_data.limiterBands, + hFrameData->sbrPatchingMode, + (flags & SBRDEC_USAC_HARMONICSBR) && + (hFrameData->sbrPatchingMode == 0) + ? GetxOverBandQmfTransposer(hSbrDec->hHBE) + : NULL, + Get41SbrQmfTransposer(hSbrDec->hHBE)); + + hSbrDec->SbrCalculateEnvelope.sbrPatchingMode = + hFrameData->sbrPatchingMode; + } + + calculateSbrEnvelope( + &hSbrDec->qmfDomainInCh->scaling, &hSbrDec->SbrCalculateEnvelope, + hHeaderData, hFrameData, &pvcDynamicData, pLowBandReal, pLowBandImag, + flags & SBRDEC_LOW_POWER, + + degreeAlias, flags, + (hHeaderData->frameErrorFlag || hPrevFrameData->frameErrorFlag)); + +#if (SBRDEC_MAX_HB_FADE_FRAMES > 0) + /* Avoid hard onsets of high band */ + if (hHeaderData->frameErrorFlag) { + if (hSbrDec->highBandFadeCnt < SBRDEC_MAX_HB_FADE_FRAMES) { + hSbrDec->highBandFadeCnt += 1; + } + } else { + if (hSbrDec->highBandFadeCnt > + 0) { /* Manipulate high band scale factor to get a smooth fade-in */ + hSbrDec->qmfDomainInCh->scaling.hb_scale += hSbrDec->highBandFadeCnt; + hSbrDec->qmfDomainInCh->scaling.hb_scale = + fMin(hSbrDec->qmfDomainInCh->scaling.hb_scale, DFRACT_BITS - 1); + hSbrDec->highBandFadeCnt -= 1; + } + } + +#endif + /* + Update hPrevFrameData (to be used in the next frame) + */ + for (i = 0; i < hHeaderData->freqBandData.nInvfBands; i++) { + hPrevFrameData->sbr_invf_mode[i] = hFrameData->sbr_invf_mode[i]; + } + hPrevFrameData->coupling = hFrameData->coupling; + hPrevFrameData->stopPos = borders[hFrameData->frameInfo.nEnvelopes]; + hPrevFrameData->ampRes = hFrameData->ampResolutionCurrentFrame; + hPrevFrameData->prevSbrPitchInBins = hFrameData->sbrPitchInBins; + /* could be done in extractFrameInfo_pvc() but hPrevFrameData is not + * available there */ + FDKmemcpy(&hPrevFrameData->prevFrameInfo, &hFrameData->frameInfo, + sizeof(FRAME_INFO)); + } else { + /* rescale from lsb to nAnalysisBands in order to compensate scaling with + * hb_scale in this area, done by synthesisFiltering*/ + int rescale; + int lsb; + int length; + + /* Reset hb_scale if no highband is present, because hb_scale is considered + * in the QMF-synthesis */ + hSbrDec->qmfDomainInCh->scaling.hb_scale = saveLbScale; + + rescale = hSbrDec->qmfDomainInCh->scaling.hb_scale - + hSbrDec->qmfDomainInCh->scaling.ov_lb_scale; + lsb = hSbrDec->qmfDomainOutCh->fb.lsb; + length = (hSbrDec->qmfDomainInCh->fb.no_channels - lsb); + + if ((rescale < 0) && (length > 0)) { + if (!(flags & SBRDEC_LOW_POWER)) { + for (i = 0; i < ov_len; i++) { + scaleValues(&pLowBandReal[i][lsb], length, rescale); + scaleValues(&pLowBandImag[i][lsb], length, rescale); + } + } else { + for (i = 0; i < ov_len; i++) { + scaleValues(&pLowBandReal[i][lsb], length, rescale); + } + } + } + } + + if (!(flags & SBRDEC_USAC_HARMONICSBR)) { + int length = hSbrDec->qmfDomainInCh->fb.lsb; + if (flags & SBRDEC_SYNTAX_USAC) { + length = hSbrDec->qmfDomainInCh->fb.no_channels; + } + + /* in case of legacy sbr saving of filter states here */ + for (i = 0; i < LPC_ORDER + ov_len; i++) { + /* + Store the unmodified qmf Slots values (required for LPC filtering) + */ + if (!(flags & SBRDEC_LOW_POWER)) { + FDKmemcpy(hSbrDec->LppTrans.lpcFilterStatesRealLegSBR[i], + pLowBandReal[noCols - LPC_ORDER + i], + length * sizeof(FIXP_DBL)); + FDKmemcpy(hSbrDec->LppTrans.lpcFilterStatesImagLegSBR[i], + pLowBandImag[noCols - LPC_ORDER + i], + length * sizeof(FIXP_DBL)); + } else + FDKmemcpy(hSbrDec->LppTrans.lpcFilterStatesRealLegSBR[i], + pLowBandReal[noCols - LPC_ORDER + i], + length * sizeof(FIXP_DBL)); + } + } + + /* + Synthesis subband filtering. + */ + + if (!(flags & SBRDEC_PS_DECODED)) { + if (!(flags & SBRDEC_SKIP_QMF_SYN)) { + int outScalefactor = 0; + + if (h_ps_d != NULL) { + h_ps_d->procFrameBased = 1; /* we here do frame based processing */ + } + + sbrDecoder_drcApply(&hSbrDec->sbrDrcChannel, pLowBandReal, + (flags & SBRDEC_LOW_POWER) ? NULL : pLowBandImag, + hSbrDec->qmfDomainOutCh->fb.no_col, &outScalefactor); + + qmfChangeOutScalefactor(&hSbrDec->qmfDomainOutCh->fb, outScalefactor); + + { + HANDLE_FREQ_BAND_DATA hFreq = &hHeaderData->freqBandData; + int save_usb = hSbrDec->qmfDomainOutCh->fb.usb; + +#if (QMF_MAX_SYNTHESIS_BANDS <= 64) + C_AALLOC_SCRATCH_START(qmfTemp, FIXP_DBL, 2 * QMF_MAX_SYNTHESIS_BANDS); +#else + C_AALLOC_STACK_START(qmfTemp, FIXP_DBL, 2 * QMF_MAX_SYNTHESIS_BANDS); +#endif + if (hSbrDec->qmfDomainOutCh->fb.usb < hFreq->ov_highSubband) { + /* we need to patch usb for this frame as overlap may contain higher + frequency range if headerchange occured; fb. usb is always limited + to maximum fb.no_channels; In case of wrongly decoded headers it + might be that ov_highSubband is higher than the number of synthesis + channels (fb.no_channels), which is forbidden, therefore we need to + limit ov_highSubband with fMin function to avoid not allowed usb in + synthesis filterbank. */ + hSbrDec->qmfDomainOutCh->fb.usb = + fMin((UINT)hFreq->ov_highSubband, + (UINT)hSbrDec->qmfDomainOutCh->fb.no_channels); + } + { + qmfSynthesisFiltering( + &hSbrDec->qmfDomainOutCh->fb, pLowBandReal, + (flags & SBRDEC_LOW_POWER) ? NULL : pLowBandImag, + &hSbrDec->qmfDomainInCh->scaling, + hSbrDec->LppTrans.pSettings->overlap, timeOut, strideOut, + qmfTemp); + } + /* restore saved value */ + hSbrDec->qmfDomainOutCh->fb.usb = save_usb; + hFreq->ov_highSubband = save_usb; +#if (QMF_MAX_SYNTHESIS_BANDS <= 64) + C_AALLOC_SCRATCH_END(qmfTemp, FIXP_DBL, 2 * QMF_MAX_SYNTHESIS_BANDS); +#else + C_AALLOC_STACK_END(qmfTemp, FIXP_DBL, 2 * QMF_MAX_SYNTHESIS_BANDS); +#endif + } + } + + } else { /* (flags & SBRDEC_PS_DECODED) */ + INT sdiff; + INT scaleFactorHighBand, scaleFactorLowBand_ov, scaleFactorLowBand_no_ov; + + HANDLE_QMF_FILTER_BANK synQmf = &hSbrDec->qmfDomainOutCh->fb; + HANDLE_QMF_FILTER_BANK synQmfRight = &hSbrDecRight->qmfDomainOutCh->fb; + + /* adapt scaling */ + sdiff = hSbrDec->qmfDomainInCh->scaling.lb_scale - + reserve; /* Scaling difference */ + scaleFactorHighBand = sdiff - hSbrDec->qmfDomainInCh->scaling.hb_scale; + scaleFactorLowBand_ov = sdiff - hSbrDec->qmfDomainInCh->scaling.ov_lb_scale; + scaleFactorLowBand_no_ov = sdiff - hSbrDec->qmfDomainInCh->scaling.lb_scale; + + /* Scale of low band overlapping QMF data */ + scaleFactorLowBand_ov = + fMin(DFRACT_BITS - 1, fMax(-(DFRACT_BITS - 1), scaleFactorLowBand_ov)); + /* Scale of low band current QMF data */ + scaleFactorLowBand_no_ov = fMin( + DFRACT_BITS - 1, fMax(-(DFRACT_BITS - 1), scaleFactorLowBand_no_ov)); + /* Scale of current high band */ + scaleFactorHighBand = + fMin(DFRACT_BITS - 1, fMax(-(DFRACT_BITS - 1), scaleFactorHighBand)); + + if (h_ps_d->procFrameBased == 1) /* If we have switched from frame to slot + based processing copy filter states */ + { /* procFrameBased will be unset later */ + /* copy filter states from left to right */ + /* was ((640)-(64))*sizeof(FIXP_QSS) + flexible amount of synthesis bands needed for QMF based resampling + */ + FDK_ASSERT(hSbrDec->qmfDomainInCh->pGlobalConf->nBandsSynthesis <= + QMF_MAX_SYNTHESIS_BANDS); + FDKmemcpy(synQmfRight->FilterStates, synQmf->FilterStates, + 9 * hSbrDec->qmfDomainInCh->pGlobalConf->nBandsSynthesis * + sizeof(FIXP_QSS)); + } + + /* Feed delaylines when parametric stereo is switched on. */ + PreparePsProcessing(h_ps_d, pLowBandReal, pLowBandImag, + scaleFactorLowBand_ov); + + /* use the same synthese qmf values for left and right channel */ + synQmfRight->no_col = synQmf->no_col; + synQmfRight->lsb = synQmf->lsb; + synQmfRight->usb = synQmf->usb; + + int env = 0; + + { +#if (QMF_MAX_SYNTHESIS_BANDS <= 64) + C_AALLOC_SCRATCH_START(pWorkBuffer, FIXP_DBL, + 2 * QMF_MAX_SYNTHESIS_BANDS); +#else + C_AALLOC_STACK_START(pWorkBuffer, FIXP_DBL, 2 * QMF_MAX_SYNTHESIS_BANDS); +#endif + + int maxShift = 0; + + if (hSbrDec->sbrDrcChannel.enable != 0) { + if (hSbrDec->sbrDrcChannel.prevFact_exp > maxShift) { + maxShift = hSbrDec->sbrDrcChannel.prevFact_exp; + } + if (hSbrDec->sbrDrcChannel.currFact_exp > maxShift) { + maxShift = hSbrDec->sbrDrcChannel.currFact_exp; + } + if (hSbrDec->sbrDrcChannel.nextFact_exp > maxShift) { + maxShift = hSbrDec->sbrDrcChannel.nextFact_exp; + } + } + + /* copy DRC data to right channel (with PS both channels use the same DRC + * gains) */ + FDKmemcpy(&hSbrDecRight->sbrDrcChannel, &hSbrDec->sbrDrcChannel, + sizeof(SBRDEC_DRC_CHANNEL)); + + for (i = 0; i < synQmf->no_col; i++) { /* ----- no_col loop ----- */ + + INT outScalefactorR, outScalefactorL; + + /* qmf timeslot of right channel */ + FIXP_DBL *rQmfReal = pWorkBuffer; + FIXP_DBL *rQmfImag = pWorkBuffer + synQmf->no_channels; + + { + if (i == + h_ps_d->bsData[h_ps_d->processSlot].mpeg.aEnvStartStop[env]) { + initSlotBasedRotation(h_ps_d, env, + hHeaderData->freqBandData.highSubband); + env++; + } + + ApplyPsSlot( + h_ps_d, /* parametric stereo decoder handle */ + (pLowBandReal + i), /* one timeslot of left/mono channel */ + (pLowBandImag + i), /* one timeslot of left/mono channel */ + rQmfReal, /* one timeslot or right channel */ + rQmfImag, /* one timeslot or right channel */ + scaleFactorLowBand_no_ov, + (i < hSbrDec->LppTrans.pSettings->overlap) + ? scaleFactorLowBand_ov + : scaleFactorLowBand_no_ov, + scaleFactorHighBand, synQmf->lsb, synQmf->usb); + + outScalefactorL = outScalefactorR = 1; /* psDiffScale! (MPEG-PS) */ + } + + sbrDecoder_drcApplySlot(/* right channel */ + &hSbrDecRight->sbrDrcChannel, rQmfReal, + rQmfImag, i, synQmfRight->no_col, maxShift); + + outScalefactorR += maxShift; + + sbrDecoder_drcApplySlot(/* left channel */ + &hSbrDec->sbrDrcChannel, *(pLowBandReal + i), + *(pLowBandImag + i), i, synQmf->no_col, + maxShift); + + outScalefactorL += maxShift; + + if (!(flags & SBRDEC_SKIP_QMF_SYN)) { + qmfSynthesisFilteringSlot( + synQmfRight, rQmfReal, /* QMF real buffer */ + rQmfImag, /* QMF imag buffer */ + outScalefactorL, outScalefactorL, + timeOutRight + (i * synQmf->no_channels * strideOut), strideOut, + pWorkBuffer); + + qmfSynthesisFilteringSlot( + synQmf, *(pLowBandReal + i), /* QMF real buffer */ + *(pLowBandImag + i), /* QMF imag buffer */ + outScalefactorR, outScalefactorR, + timeOut + (i * synQmf->no_channels * strideOut), strideOut, + pWorkBuffer); + } + } /* no_col loop i */ +#if (QMF_MAX_SYNTHESIS_BANDS <= 64) + C_AALLOC_SCRATCH_END(pWorkBuffer, FIXP_DBL, 2 * QMF_MAX_SYNTHESIS_BANDS); +#else + C_AALLOC_STACK_END(pWorkBuffer, FIXP_DBL, 2 * QMF_MAX_SYNTHESIS_BANDS); +#endif + } + } + + sbrDecoder_drcUpdateChannel(&hSbrDec->sbrDrcChannel); + + /* + Update overlap buffer + Even bands above usb are copied to avoid outdated spectral data in case + the stop frequency raises. + */ + + if (!(flags & SBRDEC_SKIP_QMF_SYN)) { + { + FDK_QmfDomain_SaveOverlap(hSbrDec->qmfDomainInCh, 0); + FDK_ASSERT(hSbrDec->qmfDomainInCh->scaling.ov_lb_scale == saveLbScale); + } + } + + hSbrDec->savedStates = 0; + + /* Save current frame status */ + hPrevFrameData->frameErrorFlag = hHeaderData->frameErrorFlag; + hSbrDec->applySbrProc_old = applyProcessing; + +} /* sbr_dec() */ + +/*! + \brief Creates sbr decoder structure + \return errorCode, 0 if successful +*/ +SBR_ERROR +createSbrDec(SBR_CHANNEL *hSbrChannel, + HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ + TRANSPOSER_SETTINGS *pSettings, + const int downsampleFac, /*!< Downsampling factor */ + const UINT qmfFlags, /*!< flags -> 1: HQ/LP selector, 2: CLDFB */ + const UINT flags, const int overlap, + int chan, /*!< Channel for which to assign buffers etc. */ + int codecFrameSize) + +{ + SBR_ERROR err = SBRDEC_OK; + int timeSlots = + hHeaderData->numberTimeSlots; /* Number of SBR slots per frame */ + int noCols = + timeSlots * hHeaderData->timeStep; /* Number of QMF slots per frame */ + HANDLE_SBR_DEC hs = &(hSbrChannel->SbrDec); + +#if (SBRDEC_MAX_HB_FADE_FRAMES > 0) + hs->highBandFadeCnt = SBRDEC_MAX_HB_FADE_FRAMES; + +#endif + hs->scale_hbe = 15; + hs->scale_lb = 15; + hs->scale_ov = 15; + + hs->prev_frame_lSbr = 0; + hs->prev_frame_hbeSbr = 0; + + hs->codecFrameSize = codecFrameSize; + + /* + create envelope calculator + */ + err = createSbrEnvelopeCalc(&hs->SbrCalculateEnvelope, hHeaderData, chan, + flags); + if (err != SBRDEC_OK) { + return err; + } + + initSbrPrevFrameData(&hSbrChannel->prevFrameData, timeSlots); + + /* + create transposer + */ + err = createLppTransposer( + &hs->LppTrans, pSettings, hHeaderData->freqBandData.lowSubband, + hHeaderData->freqBandData.v_k_master, hHeaderData->freqBandData.numMaster, + hHeaderData->freqBandData.highSubband, timeSlots, noCols, + hHeaderData->freqBandData.freqBandTableNoise, + hHeaderData->freqBandData.nNfb, hHeaderData->sbrProcSmplRate, chan, + overlap); + if (err != SBRDEC_OK) { + return err; + } + + if (flags & SBRDEC_USAC_HARMONICSBR) { + int noChannels, bSbr41 = flags & SBRDEC_QUAD_RATE ? 1 : 0; + + noChannels = + QMF_SYNTH_CHANNELS / + ((bSbr41 + 1) * 2); /* 32 for (32:64 and 24:64) and 16 for 16:64 */ + + /* shared memory between hbeLightTimeDelayBuffer and hQmfHBESlotsReal if + * SBRDEC_HBE_ENABLE */ + hSbrChannel->SbrDec.tmp_memory = (FIXP_DBL **)fdkCallocMatrix2D_aligned( + noCols, noChannels, sizeof(FIXP_DBL)); + if (hSbrChannel->SbrDec.tmp_memory == NULL) { + return SBRDEC_MEM_ALLOC_FAILED; + } + + hSbrChannel->SbrDec.hQmfHBESlotsReal = hSbrChannel->SbrDec.tmp_memory; + hSbrChannel->SbrDec.hQmfHBESlotsImag = + (FIXP_DBL **)fdkCallocMatrix2D_aligned(noCols, noChannels, + sizeof(FIXP_DBL)); + if (hSbrChannel->SbrDec.hQmfHBESlotsImag == NULL) { + return SBRDEC_MEM_ALLOC_FAILED; + } + + /* buffers containing unmodified qmf data; required when switching from + * legacy SBR to HBE */ + /* buffer can be used as LPCFilterstates buffer because legacy SBR needs + * exactly these values for LPC filtering */ + hSbrChannel->SbrDec.codecQMFBufferReal = + (FIXP_DBL **)fdkCallocMatrix2D_aligned(noCols, noChannels, + sizeof(FIXP_DBL)); + if (hSbrChannel->SbrDec.codecQMFBufferReal == NULL) { + return SBRDEC_MEM_ALLOC_FAILED; + } + + hSbrChannel->SbrDec.codecQMFBufferImag = + (FIXP_DBL **)fdkCallocMatrix2D_aligned(noCols, noChannels, + sizeof(FIXP_DBL)); + if (hSbrChannel->SbrDec.codecQMFBufferImag == NULL) { + return SBRDEC_MEM_ALLOC_FAILED; + } + + err = QmfTransposerCreate(&hs->hHBE, codecFrameSize, 0, bSbr41); + if (err != SBRDEC_OK) { + return err; + } + } + + return err; +} + +/*! + \brief Delete sbr decoder structure + \return errorCode, 0 if successful +*/ +int deleteSbrDec(SBR_CHANNEL *hSbrChannel) { + HANDLE_SBR_DEC hs = &hSbrChannel->SbrDec; + + deleteSbrEnvelopeCalc(&hs->SbrCalculateEnvelope); + + if (hs->tmp_memory != NULL) { + FDK_FREE_MEMORY_2D_ALIGNED(hs->tmp_memory); + } + + /* modify here */ + FDK_FREE_MEMORY_2D_ALIGNED(hs->hQmfHBESlotsImag); + + if (hs->hHBE != NULL) QmfTransposerClose(hs->hHBE); + + if (hs->codecQMFBufferReal != NULL) { + FDK_FREE_MEMORY_2D_ALIGNED(hs->codecQMFBufferReal); + } + + if (hs->codecQMFBufferImag != NULL) { + FDK_FREE_MEMORY_2D_ALIGNED(hs->codecQMFBufferImag); + } + + return 0; +} + +/*! + \brief resets sbr decoder structure + \return errorCode, 0 if successful +*/ +SBR_ERROR +resetSbrDec(HANDLE_SBR_DEC hSbrDec, HANDLE_SBR_HEADER_DATA hHeaderData, + HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData, const int downsampleFac, + const UINT flags, HANDLE_SBR_FRAME_DATA hFrameData) { + SBR_ERROR sbrError = SBRDEC_OK; + int i; + FIXP_DBL *pLowBandReal[128]; + FIXP_DBL *pLowBandImag[128]; + int useLP = flags & SBRDEC_LOW_POWER; + + int old_lsb = hSbrDec->qmfDomainInCh->fb.lsb; + int old_usb = hSbrDec->qmfDomainInCh->fb.usb; + int new_lsb = hHeaderData->freqBandData.lowSubband; + /* int new_usb = hHeaderData->freqBandData.highSubband; */ + int l, startBand, stopBand, startSlot, size; + + FIXP_DBL **OverlapBufferReal = hSbrDec->qmfDomainInCh->hQmfSlotsReal; + FIXP_DBL **OverlapBufferImag = hSbrDec->qmfDomainInCh->hQmfSlotsImag; + + /* in case the previous frame was not active in terms of SBR processing, the + full band from 0 to no_channels was rescaled and not overwritten. Thats why + the scaling factor lb_scale can be seen as assigned to all bands from 0 to + no_channels in the previous frame. The same states for the current frame if + the current frame is not active in terms of SBR processing + */ + int applySbrProc = (hHeaderData->syncState == SBR_ACTIVE || + (hHeaderData->frameErrorFlag == 0 && + hHeaderData->syncState == SBR_HEADER)); + int applySbrProc_old = hSbrDec->applySbrProc_old; + + if (!applySbrProc) { + new_lsb = (hSbrDec->qmfDomainInCh->fb).no_channels; + } + if (!applySbrProc_old) { + old_lsb = (hSbrDec->qmfDomainInCh->fb).no_channels; + old_usb = old_lsb; + } + + resetSbrEnvelopeCalc(&hSbrDec->SbrCalculateEnvelope); + + /* Change lsb and usb */ + /* Synthesis */ + FDK_ASSERT(hSbrDec->qmfDomainOutCh != NULL); + hSbrDec->qmfDomainOutCh->fb.lsb = + fixMin((INT)hSbrDec->qmfDomainOutCh->fb.no_channels, + (INT)hHeaderData->freqBandData.lowSubband); + hSbrDec->qmfDomainOutCh->fb.usb = + fixMin((INT)hSbrDec->qmfDomainOutCh->fb.no_channels, + (INT)hHeaderData->freqBandData.highSubband); + /* Analysis */ + FDK_ASSERT(hSbrDec->qmfDomainInCh != NULL); + hSbrDec->qmfDomainInCh->fb.lsb = hSbrDec->qmfDomainOutCh->fb.lsb; + hSbrDec->qmfDomainInCh->fb.usb = hSbrDec->qmfDomainOutCh->fb.usb; + + /* + The following initialization of spectral data in the overlap buffer + is required for dynamic x-over or a change of the start-freq for 2 reasons: + + 1. If the lowband gets _wider_, unadjusted data would remain + + 2. If the lowband becomes _smaller_, the highest bands of the old lowband + must be cleared because the whitening would be affected + */ + startBand = old_lsb; + stopBand = new_lsb; + startSlot = fMax(0, hHeaderData->timeStep * (hPrevFrameData->stopPos - + hHeaderData->numberTimeSlots)); + size = fMax(0, stopBand - startBand); + + /* in case of USAC we don't want to zero out the memory, as this can lead to + holes in the spectrum; fix shall only be applied for USAC not for MPEG-4 + SBR, in this case setting zero remains */ + if (!(flags & SBRDEC_SYNTAX_USAC)) { + /* keep already adjusted data in the x-over-area */ + if (!useLP) { + for (l = startSlot; l < hSbrDec->LppTrans.pSettings->overlap; l++) { + FDKmemclear(&OverlapBufferReal[l][startBand], size * sizeof(FIXP_DBL)); + FDKmemclear(&OverlapBufferImag[l][startBand], size * sizeof(FIXP_DBL)); + } + } else { + for (l = startSlot; l < hSbrDec->LppTrans.pSettings->overlap; l++) { + FDKmemclear(&OverlapBufferReal[l][startBand], size * sizeof(FIXP_DBL)); + } + } + + /* + reset LPC filter states + */ + startBand = fixMin(old_lsb, new_lsb); + stopBand = fixMax(old_lsb, new_lsb); + size = fixMax(0, stopBand - startBand); + + FDKmemclear(&hSbrDec->LppTrans.lpcFilterStatesRealLegSBR[0][startBand], + size * sizeof(FIXP_DBL)); + FDKmemclear(&hSbrDec->LppTrans.lpcFilterStatesRealLegSBR[1][startBand], + size * sizeof(FIXP_DBL)); + if (!useLP) { + FDKmemclear(&hSbrDec->LppTrans.lpcFilterStatesImagLegSBR[0][startBand], + size * sizeof(FIXP_DBL)); + FDKmemclear(&hSbrDec->LppTrans.lpcFilterStatesImagLegSBR[1][startBand], + size * sizeof(FIXP_DBL)); + } + } + + if (startSlot != 0) { + int source_exp, target_exp, delta_exp, target_lsb, target_usb, reserve; + FIXP_DBL maxVal; + + /* + Rescale already processed spectral data between old and new x-over + frequency. This must be done because of the separate scalefactors for + lowband and highband. + */ + + /* We have four relevant transitions to cover: + 1. old_usb is lower than new_lsb; old SBR area is completely below new SBR + area. + -> entire old area was highband and belongs to lowband now + and has to be rescaled. + 2. old_lsb is higher than new_usb; new SBR area is completely below old SBR + area. + -> old area between new_lsb and old_lsb was lowband and belongs to + highband now and has to be rescaled to match new highband scale. + 3. old_lsb is lower and old_usb is higher than new_lsb; old and new SBR + areas overlap. + -> old area between old_lsb and new_lsb was highband and belongs to + lowband now and has to be rescaled to match new lowband scale. + 4. new_lsb is lower and new_usb_is higher than old_lsb; old and new SBR + areas overlap. + -> old area between new_lsb and old_usb was lowband and belongs to + highband now and has to be rescaled to match new highband scale. + */ + + if (new_lsb > old_lsb) { + /* case 1 and 3 */ + source_exp = SCALE2EXP(hSbrDec->qmfDomainInCh->scaling.ov_hb_scale); + target_exp = SCALE2EXP(hSbrDec->qmfDomainInCh->scaling.ov_lb_scale); + + startBand = old_lsb; + + if (new_lsb >= old_usb) { + /* case 1 */ + stopBand = old_usb; + } else { + /* case 3 */ + stopBand = new_lsb; + } + + target_lsb = 0; + target_usb = old_lsb; + } else { + /* case 2 and 4 */ + source_exp = SCALE2EXP(hSbrDec->qmfDomainInCh->scaling.ov_lb_scale); + target_exp = SCALE2EXP(hSbrDec->qmfDomainInCh->scaling.ov_hb_scale); + + startBand = new_lsb; + stopBand = old_lsb; + + target_lsb = old_lsb; + target_usb = old_usb; + } + + maxVal = + maxSubbandSample(OverlapBufferReal, (useLP) ? NULL : OverlapBufferImag, + startBand, stopBand, 0, startSlot); + + reserve = ((LONG)maxVal != 0 ? CntLeadingZeros(maxVal) - 1 : 0); + reserve = fixMin( + reserve, + DFRACT_BITS - 1 - + EXP2SCALE( + source_exp)); /* what is this line for, why do we need it? */ + + /* process only if x-over-area is not dominant after rescale; + otherwise I'm not sure if all buffers are scaled correctly; + */ + if (target_exp - (source_exp - reserve) >= 0) { + rescaleSubbandSamples(OverlapBufferReal, + (useLP) ? NULL : OverlapBufferImag, startBand, + stopBand, 0, startSlot, reserve); + source_exp -= reserve; + } + + delta_exp = target_exp - source_exp; + + if (delta_exp < 0) { /* x-over-area is dominant */ + startBand = target_lsb; + stopBand = target_usb; + delta_exp = -delta_exp; + + if (new_lsb > old_lsb) { + /* The lowband has to be rescaled */ + hSbrDec->qmfDomainInCh->scaling.ov_lb_scale = EXP2SCALE(source_exp); + } else { + /* The highband has to be rescaled */ + hSbrDec->qmfDomainInCh->scaling.ov_hb_scale = EXP2SCALE(source_exp); + } + } + + FDK_ASSERT(startBand <= stopBand); + + if (!useLP) { + for (l = 0; l < startSlot; l++) { + scaleValues(OverlapBufferReal[l] + startBand, stopBand - startBand, + -delta_exp); + scaleValues(OverlapBufferImag[l] + startBand, stopBand - startBand, + -delta_exp); + } + } else + for (l = 0; l < startSlot; l++) { + scaleValues(OverlapBufferReal[l] + startBand, stopBand - startBand, + -delta_exp); + } + } /* startSlot != 0 */ + + /* + Initialize transposer and limiter + */ + sbrError = resetLppTransposer( + &hSbrDec->LppTrans, hHeaderData->freqBandData.lowSubband, + hHeaderData->freqBandData.v_k_master, hHeaderData->freqBandData.numMaster, + hHeaderData->freqBandData.freqBandTableNoise, + hHeaderData->freqBandData.nNfb, hHeaderData->freqBandData.highSubband, + hHeaderData->sbrProcSmplRate); + if (sbrError != SBRDEC_OK) return sbrError; + + hSbrDec->savedStates = 0; + + if ((flags & SBRDEC_USAC_HARMONICSBR) && applySbrProc) { + sbrError = QmfTransposerReInit(hSbrDec->hHBE, + hHeaderData->freqBandData.freqBandTable, + hHeaderData->freqBandData.nSfb); + if (sbrError != SBRDEC_OK) return sbrError; + + /* copy saved states from previous frame to legacy SBR lpc filterstate + * buffer */ + for (i = 0; i < LPC_ORDER + hSbrDec->LppTrans.pSettings->overlap; i++) { + FDKmemcpy( + hSbrDec->LppTrans.lpcFilterStatesRealLegSBR[i], + hSbrDec->codecQMFBufferReal[hSbrDec->hHBE->noCols - LPC_ORDER - + hSbrDec->LppTrans.pSettings->overlap + i], + hSbrDec->hHBE->noChannels * sizeof(FIXP_DBL)); + FDKmemcpy( + hSbrDec->LppTrans.lpcFilterStatesImagLegSBR[i], + hSbrDec->codecQMFBufferImag[hSbrDec->hHBE->noCols - LPC_ORDER - + hSbrDec->LppTrans.pSettings->overlap + i], + hSbrDec->hHBE->noChannels * sizeof(FIXP_DBL)); + } + hSbrDec->savedStates = 1; + + { + /* map QMF buffer to pointer array (Overlap + Frame)*/ + for (i = 0; i < hSbrDec->LppTrans.pSettings->overlap + LPC_ORDER; i++) { + pLowBandReal[i] = hSbrDec->LppTrans.lpcFilterStatesRealHBE[i]; + pLowBandImag[i] = hSbrDec->LppTrans.lpcFilterStatesImagHBE[i]; + } + + /* map QMF buffer to pointer array (Overlap + Frame)*/ + for (i = 0; i < hSbrDec->hHBE->noCols; i++) { + pLowBandReal[i + hSbrDec->LppTrans.pSettings->overlap + LPC_ORDER] = + hSbrDec->codecQMFBufferReal[i]; + pLowBandImag[i + hSbrDec->LppTrans.pSettings->overlap + LPC_ORDER] = + hSbrDec->codecQMFBufferImag[i]; + } + + if (flags & SBRDEC_QUAD_RATE) { + if (hFrameData->sbrPatchingMode == 0) { + int *xOverQmf = GetxOverBandQmfTransposer(hSbrDec->hHBE); + + /* in case of harmonic SBR and no HBE_LP map additional buffer for + one more frame to pointer arry */ + for (i = 0; i < hSbrDec->hHBE->noCols / 2; i++) { + pLowBandReal[i + hSbrDec->hHBE->noCols + + hSbrDec->LppTrans.pSettings->overlap + LPC_ORDER] = + hSbrDec->hQmfHBESlotsReal[i]; + pLowBandImag[i + hSbrDec->hHBE->noCols + + hSbrDec->LppTrans.pSettings->overlap + LPC_ORDER] = + hSbrDec->hQmfHBESlotsImag[i]; + } + + QmfTransposerApply( + hSbrDec->hHBE, + pLowBandReal + hSbrDec->LppTrans.pSettings->overlap + + hSbrDec->hHBE->noCols / 2 + LPC_ORDER, + pLowBandImag + hSbrDec->LppTrans.pSettings->overlap + + hSbrDec->hHBE->noCols / 2 + LPC_ORDER, + hSbrDec->hHBE->noCols, pLowBandReal, pLowBandImag, + hSbrDec->LppTrans.lpcFilterStatesRealHBE, + hSbrDec->LppTrans.lpcFilterStatesImagHBE, + hPrevFrameData->prevSbrPitchInBins, hSbrDec->scale_lb, + hSbrDec->scale_hbe, &hSbrDec->qmfDomainInCh->scaling.hb_scale, + hHeaderData->timeStep, hFrameData->frameInfo.borders[0], + hSbrDec->LppTrans.pSettings->overlap, KEEP_STATES_SYNCED_OUTDIFF); + + copyHarmonicSpectrum( + xOverQmf, pLowBandReal, pLowBandImag, hSbrDec->hHBE->noCols, + hSbrDec->LppTrans.pSettings->overlap, KEEP_STATES_SYNCED_OUTDIFF); + } + } else { + /* in case of harmonic SBR and no HBE_LP map additional buffer for + one more frame to pointer arry */ + for (i = 0; i < hSbrDec->hHBE->noCols; i++) { + pLowBandReal[i + hSbrDec->hHBE->noCols + + hSbrDec->LppTrans.pSettings->overlap + LPC_ORDER] = + hSbrDec->hQmfHBESlotsReal[i]; + pLowBandImag[i + hSbrDec->hHBE->noCols + + hSbrDec->LppTrans.pSettings->overlap + LPC_ORDER] = + hSbrDec->hQmfHBESlotsImag[i]; + } + + if (hFrameData->sbrPatchingMode == 0) { + QmfTransposerApply( + hSbrDec->hHBE, + pLowBandReal + hSbrDec->LppTrans.pSettings->overlap + LPC_ORDER, + pLowBandImag + hSbrDec->LppTrans.pSettings->overlap + LPC_ORDER, + hSbrDec->hHBE->noCols, pLowBandReal, pLowBandImag, + hSbrDec->LppTrans.lpcFilterStatesRealHBE, + hSbrDec->LppTrans.lpcFilterStatesImagHBE, + 0 /* not required for keeping states updated in this frame*/, + hSbrDec->scale_lb, hSbrDec->scale_lb, + &hSbrDec->qmfDomainInCh->scaling.hb_scale, hHeaderData->timeStep, + hFrameData->frameInfo.borders[0], + hSbrDec->LppTrans.pSettings->overlap, KEEP_STATES_SYNCED_NOOUT); + } + + QmfTransposerApply( + hSbrDec->hHBE, + pLowBandReal + hSbrDec->LppTrans.pSettings->overlap + + hSbrDec->hHBE->noCols + LPC_ORDER, + pLowBandImag + hSbrDec->LppTrans.pSettings->overlap + + hSbrDec->hHBE->noCols + LPC_ORDER, + hSbrDec->hHBE->noCols, pLowBandReal, pLowBandImag, + hSbrDec->LppTrans.lpcFilterStatesRealHBE, + hSbrDec->LppTrans.lpcFilterStatesImagHBE, + hPrevFrameData->prevSbrPitchInBins, hSbrDec->scale_lb, + hSbrDec->scale_hbe, &hSbrDec->qmfDomainInCh->scaling.hb_scale, + hHeaderData->timeStep, hFrameData->frameInfo.borders[0], + hSbrDec->LppTrans.pSettings->overlap, KEEP_STATES_SYNCED_OUTDIFF); + } + + if (hFrameData->sbrPatchingMode == 0) { + for (i = startSlot; i < hSbrDec->LppTrans.pSettings->overlap; i++) { + /* + Store the unmodified qmf Slots values for upper part of spectrum + (required for LPC filtering) required if next frame is a HBE frame + */ + FDKmemcpy(hSbrDec->qmfDomainInCh->hQmfSlotsReal[i], + hSbrDec->LppTrans.lpcFilterStatesRealHBE[i + LPC_ORDER], + (64) * sizeof(FIXP_DBL)); + FDKmemcpy(hSbrDec->qmfDomainInCh->hQmfSlotsImag[i], + hSbrDec->LppTrans.lpcFilterStatesImagHBE[i + LPC_ORDER], + (64) * sizeof(FIXP_DBL)); + } + + for (i = startSlot; i < hSbrDec->LppTrans.pSettings->overlap; i++) { + /* + Store the unmodified qmf Slots values for upper part of spectrum + (required for LPC filtering) required if next frame is a HBE frame + */ + FDKmemcpy( + hSbrDec->qmfDomainInCh->hQmfSlotsReal[i], + hSbrDec->codecQMFBufferReal[hSbrDec->hHBE->noCols - + hSbrDec->LppTrans.pSettings->overlap + + i], + new_lsb * sizeof(FIXP_DBL)); + FDKmemcpy( + hSbrDec->qmfDomainInCh->hQmfSlotsImag[i], + hSbrDec->codecQMFBufferImag[hSbrDec->hHBE->noCols - + hSbrDec->LppTrans.pSettings->overlap + + i], + new_lsb * sizeof(FIXP_DBL)); + } + } + } + } + + { + int adapt_lb = 0, diff = 0, + new_scale = hSbrDec->qmfDomainInCh->scaling.ov_lb_scale; + + if ((hSbrDec->qmfDomainInCh->scaling.ov_lb_scale != + hSbrDec->qmfDomainInCh->scaling.lb_scale) && + startSlot != 0) { + /* we need to adapt spectrum to have equal scale factor, always larger + * than zero */ + diff = SCALE2EXP(hSbrDec->qmfDomainInCh->scaling.ov_lb_scale) - + SCALE2EXP(hSbrDec->qmfDomainInCh->scaling.lb_scale); + + if (diff > 0) { + adapt_lb = 1; + diff = -diff; + new_scale = hSbrDec->qmfDomainInCh->scaling.ov_lb_scale; + } + + stopBand = new_lsb; + } + + if (hFrameData->sbrPatchingMode == 1) { + /* scale states from LegSBR filterstates buffer */ + for (i = 0; i < hSbrDec->LppTrans.pSettings->overlap + LPC_ORDER; i++) { + scaleValues(hSbrDec->LppTrans.lpcFilterStatesRealLegSBR[i], new_lsb, + diff); + if (!useLP) { + scaleValues(hSbrDec->LppTrans.lpcFilterStatesImagLegSBR[i], new_lsb, + diff); + } + } + + if (flags & SBRDEC_SYNTAX_USAC) { + /* get missing states between old and new x_over from LegSBR + * filterstates buffer */ + /* in case of legacy SBR we leave these values zeroed out */ + for (i = startSlot; i < hSbrDec->LppTrans.pSettings->overlap; i++) { + FDKmemcpy(&OverlapBufferReal[i][old_lsb], + &hSbrDec->LppTrans + .lpcFilterStatesRealLegSBR[LPC_ORDER + i][old_lsb], + fMax(new_lsb - old_lsb, 0) * sizeof(FIXP_DBL)); + if (!useLP) { + FDKmemcpy(&OverlapBufferImag[i][old_lsb], + &hSbrDec->LppTrans + .lpcFilterStatesImagLegSBR[LPC_ORDER + i][old_lsb], + fMax(new_lsb - old_lsb, 0) * sizeof(FIXP_DBL)); + } + } + } + + if (new_lsb > old_lsb) { + stopBand = old_lsb; + } + } + if ((adapt_lb == 1) && (stopBand > startBand)) { + for (l = startSlot; l < hSbrDec->LppTrans.pSettings->overlap; l++) { + scaleValues(OverlapBufferReal[l] + startBand, stopBand - startBand, + diff); + if (!useLP) { + scaleValues(OverlapBufferImag[l] + startBand, stopBand - startBand, + diff); + } + } + } + hSbrDec->qmfDomainInCh->scaling.ov_lb_scale = new_scale; + } + + sbrError = ResetLimiterBands(hHeaderData->freqBandData.limiterBandTable, + &hHeaderData->freqBandData.noLimiterBands, + hHeaderData->freqBandData.freqBandTable[0], + hHeaderData->freqBandData.nSfb[0], + hSbrDec->LppTrans.pSettings->patchParam, + hSbrDec->LppTrans.pSettings->noOfPatches, + hHeaderData->bs_data.limiterBands, + hFrameData->sbrPatchingMode, + GetxOverBandQmfTransposer(hSbrDec->hHBE), + Get41SbrQmfTransposer(hSbrDec->hHBE)); + + hSbrDec->SbrCalculateEnvelope.sbrPatchingMode = hFrameData->sbrPatchingMode; + + return sbrError; +} diff --git a/fdk-aac/libSBRdec/src/sbr_dec.h b/fdk-aac/libSBRdec/src/sbr_dec.h new file mode 100644 index 0000000..156da03 --- /dev/null +++ b/fdk-aac/libSBRdec/src/sbr_dec.h @@ -0,0 +1,204 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief Sbr decoder +*/ +#ifndef SBR_DEC_H +#define SBR_DEC_H + +#include "sbrdecoder.h" + +#include "lpp_tran.h" +#include "qmf.h" +#include "env_calc.h" +#include "FDK_audio.h" + +#include "sbrdec_drc.h" + +#include "pvc_dec.h" + +#include "hbe.h" + +enum SBRDEC_QMF_SKIP { + qmfSkipNothing = 0, + qmfSkipAnalysis = 1 << 0, + qmfSkipSynthesis = 1 << 1 +}; + +typedef struct { + SBR_CALCULATE_ENVELOPE SbrCalculateEnvelope; + SBR_LPP_TRANS LppTrans; + PVC_STATIC_DATA PvcStaticData; + + /* do scale handling in sbr an not in qmf */ + SHORT scale_ov; + SHORT scale_lb; + SHORT scale_hbe; + + SHORT prev_frame_lSbr; + SHORT prev_frame_hbeSbr; + + int codecFrameSize; + + HANDLE_HBE_TRANSPOSER hHBE; + + HANDLE_FDK_QMF_DOMAIN_IN qmfDomainInCh; + HANDLE_FDK_QMF_DOMAIN_OUT qmfDomainOutCh; + + SBRDEC_DRC_CHANNEL sbrDrcChannel; + +#if (SBRDEC_MAX_HB_FADE_FRAMES > 0) + INT highBandFadeCnt; /* counter for fading in high-band signal smoothly */ + +#endif + FIXP_DBL **tmp_memory; /* shared memory between hbeLightTimeDelayBuffer and + hQmfHBESlotsReal */ + + FIXP_DBL **hQmfHBESlotsReal; + FIXP_DBL **hQmfHBESlotsImag; + + FIXP_DBL **codecQMFBufferReal; + FIXP_DBL **codecQMFBufferImag; + UCHAR savedStates; + int applySbrProc_old; +} SBR_DEC; + +typedef SBR_DEC *HANDLE_SBR_DEC; + +typedef struct { + SBR_FRAME_DATA frameData[(1) + 1]; + SBR_PREV_FRAME_DATA prevFrameData; + SBR_DEC SbrDec; +} SBR_CHANNEL; + +typedef SBR_CHANNEL *HANDLE_SBR_CHANNEL; + +void sbr_dec( + HANDLE_SBR_DEC hSbrDec, /*!< handle to Decoder channel */ + INT_PCM *timeIn, /*!< pointer to input time signal */ + INT_PCM *timeOut, /*!< pointer to output time signal */ + HANDLE_SBR_DEC hSbrDecRight, /*!< handle to Decoder channel right */ + INT_PCM *timeOutRight, /*!< pointer to output time signal */ + INT strideOut, /*!< Time data traversal strideOut */ + HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ + HANDLE_SBR_FRAME_DATA hFrameData, /*!< Control data of current frame */ + HANDLE_SBR_PREV_FRAME_DATA + hPrevFrameData, /*!< Some control data of last frame */ + const int applyProcessing, /*!< Flag for SBR operation */ + HANDLE_PS_DEC h_ps_d, const UINT flags, const int codecFrameSize); + +SBR_ERROR +createSbrDec(SBR_CHANNEL *hSbrChannel, HANDLE_SBR_HEADER_DATA hHeaderData, + TRANSPOSER_SETTINGS *pSettings, const int downsampleFac, + const UINT qmfFlags, const UINT flags, const int overlap, int chan, + int codecFrameSize); + +int deleteSbrDec(SBR_CHANNEL *hSbrChannel); + +SBR_ERROR +resetSbrDec(HANDLE_SBR_DEC hSbrDec, HANDLE_SBR_HEADER_DATA hHeaderData, + HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData, const int downsampleFac, + const UINT flags, HANDLE_SBR_FRAME_DATA hFrameData); + +#endif diff --git a/fdk-aac/libSBRdec/src/sbr_ram.cpp b/fdk-aac/libSBRdec/src/sbr_ram.cpp new file mode 100644 index 0000000..8b35fd2 --- /dev/null +++ b/fdk-aac/libSBRdec/src/sbr_ram.cpp @@ -0,0 +1,191 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief Memory layout + + This module declares all static and dynamic memory spaces +*/ + +#include "sbr_ram.h" + +#define WORKBUFFER1_TAG 2 +#define WORKBUFFER2_TAG 3 + +/*! + \name StaticSbrData + + Static memory areas, must not be overwritten in other sections of the decoder +*/ +/* @{ */ + +/*! SBR Decoder main structure */ +C_ALLOC_MEM(Ram_SbrDecoder, struct SBR_DECODER_INSTANCE, 1) +/*! SBR Decoder element data <br> + Dimension: (8) */ +C_ALLOC_MEM2(Ram_SbrDecElement, SBR_DECODER_ELEMENT, 1, (8)) +/*! SBR Decoder individual channel data <br> + Dimension: (8) */ +C_ALLOC_MEM2(Ram_SbrDecChannel, SBR_CHANNEL, 1, (8) + 1) + +/*! Static Data of PS */ + +C_ALLOC_MEM(Ram_ps_dec, struct PS_DEC, 1) + +/* @} */ + +/*! + \name DynamicSbrData + + Dynamic memory areas, might be reused in other algorithm sections, + e.g. the core decoder + <br> + Depending on the mode set by DONT_USE_CORE_WORKBUFFER, workbuffers are + defined additionally to the CoreWorkbuffer. + <br> + The size of WorkBuffers is ((1024) / (32) * (4) / 2)*(64) = 2048. + <br> + WorkBuffer2 is a pointer to the CoreWorkBuffer wich is reused here in the SBR + part. In case of DONT_USE_CORE_WORKBUFFER, the CoreWorkbuffer is not used and + the according Workbuffer2 is defined locally in this file. <br> WorkBuffer1 is + reused in the AAC core (-> aacdecoder.cpp, aac_ram.cpp) <br> + + Use of WorkBuffers: + <pre> + + ------------------------------------------------------------- + AAC core: + + CoreWorkbuffer: spectral coefficients + WorkBuffer1: CAacDecoderChannelInfo, CAacDecoderDynamicData + + ------------------------------------------------------------- + SBR part: + ---------------------------------------------- + Low Power Mode (useLP=1 or LOW_POWER_SBR_ONLY), see assignLcTimeSlots() + + SLOT_BASED_PROTOTYPE_SYN_FILTER + + WorkBuffer1 WorkBuffer2(=CoreWorkbuffer) + ________________ ________________ + | RealLeft | | RealRight | + |________________| |________________| + + ---------------------------------------------- + High Quality Mode (!LOW_POWER_SBR_ONLY and useLP=0), see + assignHqTimeSlots() + + SLOTBASED_PS + + WorkBuffer1 WorkBuffer2(=CoreWorkbuffer) + ________________ ________________ + | Real/Imag | interleaved | Real/Imag | + interleaved + |________________| first half actual ch |________________| second + half actual ch + + ------------------------------------------------------------- + + </pre> + +*/ diff --git a/fdk-aac/libSBRdec/src/sbr_ram.h b/fdk-aac/libSBRdec/src/sbr_ram.h new file mode 100644 index 0000000..e00f8b5 --- /dev/null +++ b/fdk-aac/libSBRdec/src/sbr_ram.h @@ -0,0 +1,186 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! +\file +\brief Memory layout +*/ +#ifndef SBR_RAM_H +#define SBR_RAM_H + +#include "sbrdecoder.h" + +#include "env_extr.h" +#include "sbr_dec.h" + +#define SBRDEC_MAX_CH_PER_ELEMENT (2) + +#define FRAME_OK (0) +#define FRAME_ERROR (1) +#define FRAME_ERROR_ALLSLOTS (2) + +typedef struct { + SBR_CHANNEL *pSbrChannel[SBRDEC_MAX_CH_PER_ELEMENT]; + TRANSPOSER_SETTINGS + transposerSettings; /* Common transport settings for each individual + channel of an element */ + HANDLE_FDK_BITSTREAM hBs; + + MP4_ELEMENT_ID + elementID; /* Element ID set during initialization. Can be used for + concealment */ + int nChannels; /* Number of elements output channels (=2 in case of PS) */ + + UCHAR frameErrorFlag[(1) + 1]; /* Frame error status (for every slot in the + delay line). Will be copied into header at + the very beginning of decodeElement() + routine. */ + + UCHAR useFrameSlot; /* Index which defines which slot will be decoded/filled + next (used with additional delay) */ + UCHAR useHeaderSlot[(1) + 1]; /* Index array that provides the link between + header and frame data (important when + processing with additional delay). */ +} SBR_DECODER_ELEMENT; + +struct SBR_DECODER_INSTANCE { + SBR_DECODER_ELEMENT *pSbrElement[(8)]; + SBR_HEADER_DATA sbrHeader[( + 8)][(1) + 1]; /* Sbr header for each individual channel of an element */ + + HANDLE_FDK_QMF_DOMAIN pQmfDomain; + + HANDLE_PS_DEC hParametricStereoDec; + + /* Global parameters */ + AUDIO_OBJECT_TYPE coreCodec; /* AOT of core codec */ + int numSbrElements; + int numSbrChannels; + INT sampleRateIn; /* SBR decoder input sampling rate; might be different than + the transposer input sampling rate. */ + INT sampleRateOut; /* Sampling rate of the SBR decoder output audio samples. + */ + USHORT codecFrameSize; + UCHAR synDownsampleFac; + INT downscaleFactor; + UCHAR numDelayFrames; /* The current number of additional delay frames used + for processing. */ + UCHAR harmonicSBR; + UCHAR + numFlushedFrames; /* The variable counts the number of frames which are + flushed consecutively. */ + + UINT flags; +}; + +H_ALLOC_MEM(Ram_SbrDecElement, SBR_DECODER_ELEMENT) +H_ALLOC_MEM(Ram_SbrDecChannel, SBR_CHANNEL) +H_ALLOC_MEM(Ram_SbrDecoder, struct SBR_DECODER_INSTANCE) + +H_ALLOC_MEM(Ram_sbr_QmfStatesSynthesis, FIXP_QSS) +H_ALLOC_MEM(Ram_sbr_OverlapBuffer, FIXP_DBL) + +H_ALLOC_MEM(Ram_sbr_HBEOverlapBuffer, FIXP_DBL) + +H_ALLOC_MEM(Ram_ps_dec, PS_DEC) + +#endif /* SBR_RAM_H */ diff --git a/fdk-aac/libSBRdec/src/sbr_rom.cpp b/fdk-aac/libSBRdec/src/sbr_rom.cpp new file mode 100644 index 0000000..8a6688a --- /dev/null +++ b/fdk-aac/libSBRdec/src/sbr_rom.cpp @@ -0,0 +1,1705 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief Definition of constant tables + + This module contains most of the constant data that can be stored in ROM. +*/ + +#include "sbr_rom.h" + +/*! + \name StartStopBands + \brief Start and stop subbands of the highband. + + k_o = startMin + offset[bs_start_freq]; + startMin = {3000,4000,5000} * (128/FS_sbr) / FS_sbr < 32Khz, 32Khz <= FS_sbr < + 64KHz, 64KHz <= FS_sbr The stop subband can also be calculated to save memory + by defining #CALC_STOP_BAND. +*/ +//@{ +/* tables were created with ../addon/octave/sbr_start_freq_table.m */ +const UCHAR FDK_sbrDecoder_sbr_start_freq_16[][16] = { + {16, 17, 18, 19, 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31}, + {4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19}}; +const UCHAR FDK_sbrDecoder_sbr_start_freq_22[][16] = { + {12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 24, 26, 28, 30}, + {4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 18, 20, 22}}; +const UCHAR FDK_sbrDecoder_sbr_start_freq_24[][16] = { + {11, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 25, 27, 29, 32}, + {3, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 17, 19, 21, 24}}; +const UCHAR FDK_sbrDecoder_sbr_start_freq_32[][16] = { + {10, 12, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 25, 27, 29, 32}, + {2, 4, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 17, 19, 21, 24}}; +const UCHAR FDK_sbrDecoder_sbr_start_freq_40[][16] = { + {12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 24, 26, 28, 30, 32}, + {5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 17, 19, 21, 23, 25}}; +const UCHAR FDK_sbrDecoder_sbr_start_freq_44[][16] = { + {8, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 21, 23, 25, 28, 32}, + {2, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 15, 17, 19, 22, 26}}; +const UCHAR FDK_sbrDecoder_sbr_start_freq_48[][16] = { + {7, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 20, 22, 24, 27, 31}, + {1, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 14, 16, 18, 21, 25}}; +const UCHAR FDK_sbrDecoder_sbr_start_freq_64[][16] = { + {6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 19, 21, 23, 26, 30}, + {1, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 14, 16, 18, 21, 25}}; +const UCHAR FDK_sbrDecoder_sbr_start_freq_88[][16] = { + {5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 16, 18, 20, 23, 27, 31}, + {2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 13, 15, 17, 20, 24, 28}}; +const UCHAR FDK_sbrDecoder_sbr_start_freq_192[16] = { + 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 12, 14, 16, 19, 23, 27}; +const UCHAR FDK_sbrDecoder_sbr_start_freq_176[16] = { + 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 13, 15, 17, 20, 24, 28}; +const UCHAR FDK_sbrDecoder_sbr_start_freq_128[16] = { + 1, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 14, 16, 18, 21, 25}; + +//@} + +/*! + \name Whitening + \brief Coefficients for spectral whitening in the transposer +*/ +//@{ +/*! Assignment of whitening tuning depending on the crossover frequency */ +const USHORT FDK_sbrDecoder_sbr_whFactorsIndex[NUM_WHFACTOR_TABLE_ENTRIES] = { + 0, 5000, 6000, 6500, 7000, 7500, 8000, 9000, 10000}; + +/*! + \brief Whithening levels tuning table + + With the current tuning, there are some redundant entries: + + \li NUM_WHFACTOR_TABLE_ENTRIES can be reduced by 3, + \li the first coloumn can be eliminated. + +*/ +const FIXP_DBL + FDK_sbrDecoder_sbr_whFactorsTable[NUM_WHFACTOR_TABLE_ENTRIES][6] = { + /* OFF_LEVEL, TRANSITION_LEVEL, LOW_LEVEL, MID_LEVEL, HIGH_LEVEL */ + {FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), + FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* < 5000 */ + {FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), + FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 5000 < 6000 */ + {FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), + FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 6000 < 6500 */ + {FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), + FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 6500 < 7000 */ + {FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), + FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 7000 < 7500 */ + {FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), + FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 7500 < 8000 */ + {FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), + FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 8000 < 9000 */ + {FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), + FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 9000 < 10000 */ + {FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), + FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* > 10000 */ +}; + +//@} + +/*! + \name EnvAdj + \brief Constants and tables used for envelope adjustment +*/ +//@{ + +/*! Mantissas of gain limits */ +const FIXP_SGL FDK_sbrDecoder_sbr_limGains_m[4] = { + FL2FXCONST_SGL(0.5011932025f), /*!< -3 dB. Gain limit when limiterGains in + frameData is 0 */ + FL2FXCONST_SGL( + 0.5f), /*!< 0 dB. Gain limit when limiterGains in frameData is 1 */ + FL2FXCONST_SGL(0.9976346258f), /*!< +3 dB. Gain limit when limiterGains in + frameData is 2 */ + FL2FXCONST_SGL(0.6776263578f) /*!< Inf. Gain limit when limiterGains in + frameData is 3 */ +}; + +/*! Exponents of gain limits */ +const UCHAR FDK_sbrDecoder_sbr_limGains_e[4] = {0, 1, 1, 67}; + +/*! Constants for calculating the number of limiter bands */ +const FIXP_SGL FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4[4] = { + FL2FXCONST_SGL(1.0f / 4.0f), FL2FXCONST_SGL(1.2f / 4.0f), + FL2FXCONST_SGL(2.0f / 4.0f), FL2FXCONST_SGL(3.0f / 4.0f)}; + +/*! Constants for calculating the number of limiter bands */ +const FIXP_DBL FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4_DBL[4] = { + FL2FXCONST_DBL(1.0f / 4.0f), FL2FXCONST_DBL(1.2f / 4.0f), + FL2FXCONST_DBL(2.0f / 4.0f), FL2FXCONST_DBL(3.0f / 4.0f)}; + +/*! Ratio of old gains and noise levels for the first 4 timeslots of an envelope + */ +const FIXP_SGL FDK_sbrDecoder_sbr_smoothFilter[4] = { + FL2FXCONST_SGL(0.66666666666666f), FL2FXCONST_SGL(0.36516383427084f), + FL2FXCONST_SGL(0.14699433520835f), FL2FXCONST_SGL(0.03183050093751f)}; + +/*! Real and imaginary part of random noise which will be modulated + to the desired level. An accuracy of 13 bits is sufficient for these + random numbers. +*/ +const FIXP_SGL FDK_sbrDecoder_sbr_randomPhase[SBR_NF_NO_RANDOM_VAL][2] = { + {FL2FXCONST_SGL(-0.99948153278296f / 8.0), + FL2FXCONST_SGL(-0.59483417516607f / 8.0)}, + {FL2FXCONST_SGL(0.97113454393991f / 8.0), + FL2FXCONST_SGL(-0.67528515225647f / 8.0)}, + {FL2FXCONST_SGL(0.14130051758487f / 8.0), + FL2FXCONST_SGL(-0.95090983575689f / 8.0)}, + {FL2FXCONST_SGL(-0.47005496701697f / 8.0), + FL2FXCONST_SGL(-0.37340549728647f / 8.0)}, + {FL2FXCONST_SGL(0.80705063769351f / 8.0), + FL2FXCONST_SGL(0.29653668284408f / 8.0)}, + {FL2FXCONST_SGL(-0.38981478896926f / 8.0), + FL2FXCONST_SGL(0.89572605717087f / 8.0)}, + {FL2FXCONST_SGL(-0.01053049862020f / 8.0), + FL2FXCONST_SGL(-0.66959058036166f / 8.0)}, + {FL2FXCONST_SGL(-0.91266367957293f / 8.0), + FL2FXCONST_SGL(-0.11522938140034f / 8.0)}, + {FL2FXCONST_SGL(0.54840422910309f / 8.0), + FL2FXCONST_SGL(0.75221367176302f / 8.0)}, + {FL2FXCONST_SGL(0.40009252867955f / 8.0), + FL2FXCONST_SGL(-0.98929400334421f / 8.0)}, + {FL2FXCONST_SGL(-0.99867974711855f / 8.0), + FL2FXCONST_SGL(-0.88147068645358f / 8.0)}, + {FL2FXCONST_SGL(-0.95531076805040f / 8.0), + FL2FXCONST_SGL(0.90908757154593f / 8.0)}, + {FL2FXCONST_SGL(-0.45725933317144f / 8.0), + FL2FXCONST_SGL(-0.56716323646760f / 8.0)}, + {FL2FXCONST_SGL(-0.72929675029275f / 8.0), + FL2FXCONST_SGL(-0.98008272727324f / 8.0)}, + {FL2FXCONST_SGL(0.75622801399036f / 8.0), + FL2FXCONST_SGL(0.20950329995549f / 8.0)}, + {FL2FXCONST_SGL(0.07069442601050f / 8.0), + FL2FXCONST_SGL(-0.78247898470706f / 8.0)}, + {FL2FXCONST_SGL(0.74496252926055f / 8.0), + FL2FXCONST_SGL(-0.91169004445807f / 8.0)}, + {FL2FXCONST_SGL(-0.96440182703856f / 8.0), + FL2FXCONST_SGL(-0.94739918296622f / 8.0)}, + {FL2FXCONST_SGL(0.30424629369539f / 8.0), + FL2FXCONST_SGL(-0.49438267012479f / 8.0)}, + {FL2FXCONST_SGL(0.66565033746925f / 8.0), + FL2FXCONST_SGL(0.64652935542491f / 8.0)}, + {FL2FXCONST_SGL(0.91697008020594f / 8.0), + FL2FXCONST_SGL(0.17514097332009f / 8.0)}, + {FL2FXCONST_SGL(-0.70774918760427f / 8.0), + FL2FXCONST_SGL(0.52548653416543f / 8.0)}, + {FL2FXCONST_SGL(-0.70051415345560f / 8.0), + FL2FXCONST_SGL(-0.45340028808763f / 8.0)}, + {FL2FXCONST_SGL(-0.99496513054797f / 8.0), + FL2FXCONST_SGL(-0.90071908066973f / 8.0)}, + {FL2FXCONST_SGL(0.98164490790123f / 8.0), + FL2FXCONST_SGL(-0.77463155528697f / 8.0)}, + {FL2FXCONST_SGL(-0.54671580548181f / 8.0), + FL2FXCONST_SGL(-0.02570928536004f / 8.0)}, + {FL2FXCONST_SGL(-0.01689629065389f / 8.0), + FL2FXCONST_SGL(0.00287506445732f / 8.0)}, + {FL2FXCONST_SGL(-0.86110349531986f / 8.0), + FL2FXCONST_SGL(0.42548583726477f / 8.0)}, + {FL2FXCONST_SGL(-0.98892980586032f / 8.0), + FL2FXCONST_SGL(-0.87881132267556f / 8.0)}, + {FL2FXCONST_SGL(0.51756627678691f / 8.0), + FL2FXCONST_SGL(0.66926784710139f / 8.0)}, + {FL2FXCONST_SGL(-0.99635026409640f / 8.0), + FL2FXCONST_SGL(-0.58107730574765f / 8.0)}, + {FL2FXCONST_SGL(-0.99969370862163f / 8.0), + FL2FXCONST_SGL(0.98369989360250f / 8.0)}, + {FL2FXCONST_SGL(0.55266258627194f / 8.0), + FL2FXCONST_SGL(0.59449057465591f / 8.0)}, + {FL2FXCONST_SGL(0.34581177741673f / 8.0), + FL2FXCONST_SGL(0.94879421061866f / 8.0)}, + {FL2FXCONST_SGL(0.62664209577999f / 8.0), + FL2FXCONST_SGL(-0.74402970906471f / 8.0)}, + {FL2FXCONST_SGL(-0.77149701404973f / 8.0), + FL2FXCONST_SGL(-0.33883658042801f / 8.0)}, + {FL2FXCONST_SGL(-0.91592244254432f / 8.0), + FL2FXCONST_SGL(0.03687901376713f / 8.0)}, + {FL2FXCONST_SGL(-0.76285492357887f / 8.0), + FL2FXCONST_SGL(-0.91371867919124f / 8.0)}, + {FL2FXCONST_SGL(0.79788337195331f / 8.0), + FL2FXCONST_SGL(-0.93180971199849f / 8.0)}, + {FL2FXCONST_SGL(0.54473080610200f / 8.0), + FL2FXCONST_SGL(-0.11919206037186f / 8.0)}, + {FL2FXCONST_SGL(-0.85639281671058f / 8.0), + FL2FXCONST_SGL(0.42429854760451f / 8.0)}, + {FL2FXCONST_SGL(-0.92882402971423f / 8.0), + FL2FXCONST_SGL(0.27871809078609f / 8.0)}, + {FL2FXCONST_SGL(-0.11708371046774f / 8.0), + FL2FXCONST_SGL(-0.99800843444966f / 8.0)}, + {FL2FXCONST_SGL(0.21356749817493f / 8.0), + FL2FXCONST_SGL(-0.90716295627033f / 8.0)}, + {FL2FXCONST_SGL(-0.76191692573909f / 8.0), + FL2FXCONST_SGL(0.99768118356265f / 8.0)}, + {FL2FXCONST_SGL(0.98111043100884f / 8.0), + FL2FXCONST_SGL(-0.95854459734407f / 8.0)}, + {FL2FXCONST_SGL(-0.85913269895572f / 8.0), + FL2FXCONST_SGL(0.95766566168880f / 8.0)}, + {FL2FXCONST_SGL(-0.93307242253692f / 8.0), + FL2FXCONST_SGL(0.49431757696466f / 8.0)}, + {FL2FXCONST_SGL(0.30485754879632f / 8.0), + FL2FXCONST_SGL(-0.70540034357529f / 8.0)}, + {FL2FXCONST_SGL(0.85289650925190f / 8.0), + FL2FXCONST_SGL(0.46766131791044f / 8.0)}, + {FL2FXCONST_SGL(0.91328082618125f / 8.0), + FL2FXCONST_SGL(-0.99839597361769f / 8.0)}, + {FL2FXCONST_SGL(-0.05890199924154f / 8.0), + FL2FXCONST_SGL(0.70741827819497f / 8.0)}, + {FL2FXCONST_SGL(0.28398686150148f / 8.0), + FL2FXCONST_SGL(0.34633555702188f / 8.0)}, + {FL2FXCONST_SGL(0.95258164539612f / 8.0), + FL2FXCONST_SGL(-0.54893416026939f / 8.0)}, + {FL2FXCONST_SGL(-0.78566324168507f / 8.0), + FL2FXCONST_SGL(-0.75568541079691f / 8.0)}, + {FL2FXCONST_SGL(-0.95789495447877f / 8.0), + FL2FXCONST_SGL(-0.20423194696966f / 8.0)}, + {FL2FXCONST_SGL(0.82411158711197f / 8.0), + FL2FXCONST_SGL(0.96654618432562f / 8.0)}, + {FL2FXCONST_SGL(-0.65185446735885f / 8.0), + FL2FXCONST_SGL(-0.88734990773289f / 8.0)}, + {FL2FXCONST_SGL(-0.93643603134666f / 8.0), + FL2FXCONST_SGL(0.99870790442385f / 8.0)}, + {FL2FXCONST_SGL(0.91427159529618f / 8.0), + FL2FXCONST_SGL(-0.98290505544444f / 8.0)}, + {FL2FXCONST_SGL(-0.70395684036886f / 8.0), + FL2FXCONST_SGL(0.58796798221039f / 8.0)}, + {FL2FXCONST_SGL(0.00563771969365f / 8.0), + FL2FXCONST_SGL(0.61768196727244f / 8.0)}, + {FL2FXCONST_SGL(0.89065051931895f / 8.0), + FL2FXCONST_SGL(0.52783352697585f / 8.0)}, + {FL2FXCONST_SGL(-0.68683707712762f / 8.0), + FL2FXCONST_SGL(0.80806944710339f / 8.0)}, + {FL2FXCONST_SGL(0.72165342518718f / 8.0), + FL2FXCONST_SGL(-0.69259857349564f / 8.0)}, + {FL2FXCONST_SGL(-0.62928247730667f / 8.0), + FL2FXCONST_SGL(0.13627037407335f / 8.0)}, + {FL2FXCONST_SGL(0.29938434065514f / 8.0), + FL2FXCONST_SGL(-0.46051329682246f / 8.0)}, + {FL2FXCONST_SGL(-0.91781958879280f / 8.0), + FL2FXCONST_SGL(-0.74012716684186f / 8.0)}, + {FL2FXCONST_SGL(0.99298717043688f / 8.0), + FL2FXCONST_SGL(0.40816610075661f / 8.0)}, + {FL2FXCONST_SGL(0.82368298622748f / 8.0), + FL2FXCONST_SGL(-0.74036047190173f / 8.0)}, + {FL2FXCONST_SGL(-0.98512833386833f / 8.0), + FL2FXCONST_SGL(-0.99972330709594f / 8.0)}, + {FL2FXCONST_SGL(-0.95915368242257f / 8.0), + FL2FXCONST_SGL(-0.99237800466040f / 8.0)}, + {FL2FXCONST_SGL(-0.21411126572790f / 8.0), + FL2FXCONST_SGL(-0.93424819052545f / 8.0)}, + {FL2FXCONST_SGL(-0.68821476106884f / 8.0), + FL2FXCONST_SGL(-0.26892306315457f / 8.0)}, + {FL2FXCONST_SGL(0.91851997982317f / 8.0), + FL2FXCONST_SGL(0.09358228901785f / 8.0)}, + {FL2FXCONST_SGL(-0.96062769559127f / 8.0), + FL2FXCONST_SGL(0.36099095133739f / 8.0)}, + {FL2FXCONST_SGL(0.51646184922287f / 8.0), + FL2FXCONST_SGL(-0.71373332873917f / 8.0)}, + {FL2FXCONST_SGL(0.61130721139669f / 8.0), + FL2FXCONST_SGL(0.46950141175917f / 8.0)}, + {FL2FXCONST_SGL(0.47336129371299f / 8.0), + FL2FXCONST_SGL(-0.27333178296162f / 8.0)}, + {FL2FXCONST_SGL(0.90998308703519f / 8.0), + FL2FXCONST_SGL(0.96715662938132f / 8.0)}, + {FL2FXCONST_SGL(0.44844799194357f / 8.0), + FL2FXCONST_SGL(0.99211574628306f / 8.0)}, + {FL2FXCONST_SGL(0.66614891079092f / 8.0), + FL2FXCONST_SGL(0.96590176169121f / 8.0)}, + {FL2FXCONST_SGL(0.74922239129237f / 8.0), + FL2FXCONST_SGL(-0.89879858826087f / 8.0)}, + {FL2FXCONST_SGL(-0.99571588506485f / 8.0), + FL2FXCONST_SGL(0.52785521494349f / 8.0)}, + {FL2FXCONST_SGL(0.97401082477563f / 8.0), + FL2FXCONST_SGL(-0.16855870075190f / 8.0)}, + {FL2FXCONST_SGL(0.72683747733879f / 8.0), + FL2FXCONST_SGL(-0.48060774432251f / 8.0)}, + {FL2FXCONST_SGL(0.95432193457128f / 8.0), + FL2FXCONST_SGL(0.68849603408441f / 8.0)}, + {FL2FXCONST_SGL(-0.72962208425191f / 8.0), + FL2FXCONST_SGL(-0.76608443420917f / 8.0)}, + {FL2FXCONST_SGL(-0.85359479233537f / 8.0), + FL2FXCONST_SGL(0.88738125901579f / 8.0)}, + {FL2FXCONST_SGL(-0.81412430338535f / 8.0), + FL2FXCONST_SGL(-0.97480768049637f / 8.0)}, + {FL2FXCONST_SGL(-0.87930772356786f / 8.0), + FL2FXCONST_SGL(0.74748307690436f / 8.0)}, + {FL2FXCONST_SGL(-0.71573331064977f / 8.0), + FL2FXCONST_SGL(-0.98570608178923f / 8.0)}, + {FL2FXCONST_SGL(0.83524300028228f / 8.0), + FL2FXCONST_SGL(0.83702537075163f / 8.0)}, + {FL2FXCONST_SGL(-0.48086065601423f / 8.0), + FL2FXCONST_SGL(-0.98848504923531f / 8.0)}, + {FL2FXCONST_SGL(0.97139128574778f / 8.0), + FL2FXCONST_SGL(0.80093621198236f / 8.0)}, + {FL2FXCONST_SGL(0.51992825347895f / 8.0), + FL2FXCONST_SGL(0.80247631400510f / 8.0)}, + {FL2FXCONST_SGL(-0.00848591195325f / 8.0), + FL2FXCONST_SGL(-0.76670128000486f / 8.0)}, + {FL2FXCONST_SGL(-0.70294374303036f / 8.0), + FL2FXCONST_SGL(0.55359910445577f / 8.0)}, + {FL2FXCONST_SGL(-0.95894428168140f / 8.0), + FL2FXCONST_SGL(-0.43265504344783f / 8.0)}, + {FL2FXCONST_SGL(0.97079252950321f / 8.0), + FL2FXCONST_SGL(0.09325857238682f / 8.0)}, + {FL2FXCONST_SGL(-0.92404293670797f / 8.0), + FL2FXCONST_SGL(0.85507704027855f / 8.0)}, + {FL2FXCONST_SGL(-0.69506469500450f / 8.0), + FL2FXCONST_SGL(0.98633412625459f / 8.0)}, + {FL2FXCONST_SGL(0.26559203620024f / 8.0), + FL2FXCONST_SGL(0.73314307966524f / 8.0)}, + {FL2FXCONST_SGL(0.28038443336943f / 8.0), + FL2FXCONST_SGL(0.14537913654427f / 8.0)}, + {FL2FXCONST_SGL(-0.74138124825523f / 8.0), + FL2FXCONST_SGL(0.99310339807762f / 8.0)}, + {FL2FXCONST_SGL(-0.01752795995444f / 8.0), + FL2FXCONST_SGL(-0.82616635284178f / 8.0)}, + {FL2FXCONST_SGL(-0.55126773094930f / 8.0), + FL2FXCONST_SGL(-0.98898543862153f / 8.0)}, + {FL2FXCONST_SGL(0.97960898850996f / 8.0), + FL2FXCONST_SGL(-0.94021446752851f / 8.0)}, + {FL2FXCONST_SGL(-0.99196309146936f / 8.0), + FL2FXCONST_SGL(0.67019017358456f / 8.0)}, + {FL2FXCONST_SGL(-0.67684928085260f / 8.0), + FL2FXCONST_SGL(0.12631491649378f / 8.0)}, + {FL2FXCONST_SGL(0.09140039465500f / 8.0), + FL2FXCONST_SGL(-0.20537731453108f / 8.0)}, + {FL2FXCONST_SGL(-0.71658965751996f / 8.0), + FL2FXCONST_SGL(-0.97788200391224f / 8.0)}, + {FL2FXCONST_SGL(0.81014640078925f / 8.0), + FL2FXCONST_SGL(0.53722648362443f / 8.0)}, + {FL2FXCONST_SGL(0.40616991671205f / 8.0), + FL2FXCONST_SGL(-0.26469008598449f / 8.0)}, + {FL2FXCONST_SGL(-0.67680188682972f / 8.0), + FL2FXCONST_SGL(0.94502052337695f / 8.0)}, + {FL2FXCONST_SGL(0.86849774348749f / 8.0), + FL2FXCONST_SGL(-0.18333598647899f / 8.0)}, + {FL2FXCONST_SGL(-0.99500381284851f / 8.0), + FL2FXCONST_SGL(-0.02634122068550f / 8.0)}, + {FL2FXCONST_SGL(0.84329189340667f / 8.0), + FL2FXCONST_SGL(0.10406957462213f / 8.0)}, + {FL2FXCONST_SGL(-0.09215968531446f / 8.0), + FL2FXCONST_SGL(0.69540012101253f / 8.0)}, + {FL2FXCONST_SGL(0.99956173327206f / 8.0), + FL2FXCONST_SGL(-0.12358542001404f / 8.0)}, + {FL2FXCONST_SGL(-0.79732779473535f / 8.0), + FL2FXCONST_SGL(-0.91582524736159f / 8.0)}, + {FL2FXCONST_SGL(0.96349973642406f / 8.0), + FL2FXCONST_SGL(0.96640458041000f / 8.0)}, + {FL2FXCONST_SGL(-0.79942778496547f / 8.0), + FL2FXCONST_SGL(0.64323902822857f / 8.0)}, + {FL2FXCONST_SGL(-0.11566039853896f / 8.0), + FL2FXCONST_SGL(0.28587846253726f / 8.0)}, + {FL2FXCONST_SGL(-0.39922954514662f / 8.0), + FL2FXCONST_SGL(0.94129601616966f / 8.0)}, + {FL2FXCONST_SGL(0.99089197565987f / 8.0), + FL2FXCONST_SGL(-0.92062625581587f / 8.0)}, + {FL2FXCONST_SGL(0.28631285179909f / 8.0), + FL2FXCONST_SGL(-0.91035047143603f / 8.0)}, + {FL2FXCONST_SGL(-0.83302725605608f / 8.0), + FL2FXCONST_SGL(-0.67330410892084f / 8.0)}, + {FL2FXCONST_SGL(0.95404443402072f / 8.0), + FL2FXCONST_SGL(0.49162765398743f / 8.0)}, + {FL2FXCONST_SGL(-0.06449863579434f / 8.0), + FL2FXCONST_SGL(0.03250560813135f / 8.0)}, + {FL2FXCONST_SGL(-0.99575054486311f / 8.0), + FL2FXCONST_SGL(0.42389784469507f / 8.0)}, + {FL2FXCONST_SGL(-0.65501142790847f / 8.0), + FL2FXCONST_SGL(0.82546114655624f / 8.0)}, + {FL2FXCONST_SGL(-0.81254441908887f / 8.0), + FL2FXCONST_SGL(-0.51627234660629f / 8.0)}, + {FL2FXCONST_SGL(-0.99646369485481f / 8.0), + FL2FXCONST_SGL(0.84490533520752f / 8.0)}, + {FL2FXCONST_SGL(0.00287840603348f / 8.0), + FL2FXCONST_SGL(0.64768261158166f / 8.0)}, + {FL2FXCONST_SGL(0.70176989408455f / 8.0), + FL2FXCONST_SGL(-0.20453028573322f / 8.0)}, + {FL2FXCONST_SGL(0.96361882270190f / 8.0), + FL2FXCONST_SGL(0.40706967140989f / 8.0)}, + {FL2FXCONST_SGL(-0.68883758192426f / 8.0), + FL2FXCONST_SGL(0.91338958840772f / 8.0)}, + {FL2FXCONST_SGL(-0.34875585502238f / 8.0), + FL2FXCONST_SGL(0.71472290693300f / 8.0)}, + {FL2FXCONST_SGL(0.91980081243087f / 8.0), + FL2FXCONST_SGL(0.66507455644919f / 8.0)}, + {FL2FXCONST_SGL(-0.99009048343881f / 8.0), + FL2FXCONST_SGL(0.85868021604848f / 8.0)}, + {FL2FXCONST_SGL(0.68865791458395f / 8.0), + FL2FXCONST_SGL(0.55660316809678f / 8.0)}, + {FL2FXCONST_SGL(-0.99484402129368f / 8.0), + FL2FXCONST_SGL(-0.20052559254934f / 8.0)}, + {FL2FXCONST_SGL(0.94214511408023f / 8.0), + FL2FXCONST_SGL(-0.99696425367461f / 8.0)}, + {FL2FXCONST_SGL(-0.67414626793544f / 8.0), + FL2FXCONST_SGL(0.49548221180078f / 8.0)}, + {FL2FXCONST_SGL(-0.47339353684664f / 8.0), + FL2FXCONST_SGL(-0.85904328834047f / 8.0)}, + {FL2FXCONST_SGL(0.14323651387360f / 8.0), + FL2FXCONST_SGL(-0.94145598222488f / 8.0)}, + {FL2FXCONST_SGL(-0.29268293575672f / 8.0), + FL2FXCONST_SGL(0.05759224927952f / 8.0)}, + {FL2FXCONST_SGL(0.43793861458754f / 8.0), + FL2FXCONST_SGL(-0.78904969892724f / 8.0)}, + {FL2FXCONST_SGL(-0.36345126374441f / 8.0), + FL2FXCONST_SGL(0.64874435357162f / 8.0)}, + {FL2FXCONST_SGL(-0.08750604656825f / 8.0), + FL2FXCONST_SGL(0.97686944362527f / 8.0)}, + {FL2FXCONST_SGL(-0.96495267812511f / 8.0), + FL2FXCONST_SGL(-0.53960305946511f / 8.0)}, + {FL2FXCONST_SGL(0.55526940659947f / 8.0), + FL2FXCONST_SGL(0.78891523734774f / 8.0)}, + {FL2FXCONST_SGL(0.73538215752630f / 8.0), + FL2FXCONST_SGL(0.96452072373404f / 8.0)}, + {FL2FXCONST_SGL(-0.30889773919437f / 8.0), + FL2FXCONST_SGL(-0.80664389776860f / 8.0)}, + {FL2FXCONST_SGL(0.03574995626194f / 8.0), + FL2FXCONST_SGL(-0.97325616900959f / 8.0)}, + {FL2FXCONST_SGL(0.98720684660488f / 8.0), + FL2FXCONST_SGL(0.48409133691962f / 8.0)}, + {FL2FXCONST_SGL(-0.81689296271203f / 8.0), + FL2FXCONST_SGL(-0.90827703628298f / 8.0)}, + {FL2FXCONST_SGL(0.67866860118215f / 8.0), + FL2FXCONST_SGL(0.81284503870856f / 8.0)}, + {FL2FXCONST_SGL(-0.15808569732583f / 8.0), + FL2FXCONST_SGL(0.85279555024382f / 8.0)}, + {FL2FXCONST_SGL(0.80723395114371f / 8.0), + FL2FXCONST_SGL(-0.24717418514605f / 8.0)}, + {FL2FXCONST_SGL(0.47788757329038f / 8.0), + FL2FXCONST_SGL(-0.46333147839295f / 8.0)}, + {FL2FXCONST_SGL(0.96367554763201f / 8.0), + FL2FXCONST_SGL(0.38486749303242f / 8.0)}, + {FL2FXCONST_SGL(-0.99143875716818f / 8.0), + FL2FXCONST_SGL(-0.24945277239809f / 8.0)}, + {FL2FXCONST_SGL(0.83081876925833f / 8.0), + FL2FXCONST_SGL(-0.94780851414763f / 8.0)}, + {FL2FXCONST_SGL(-0.58753191905341f / 8.0), + FL2FXCONST_SGL(0.01290772389163f / 8.0)}, + {FL2FXCONST_SGL(0.95538108220960f / 8.0), + FL2FXCONST_SGL(-0.85557052096538f / 8.0)}, + {FL2FXCONST_SGL(-0.96490920476211f / 8.0), + FL2FXCONST_SGL(-0.64020970923102f / 8.0)}, + {FL2FXCONST_SGL(-0.97327101028521f / 8.0), + FL2FXCONST_SGL(0.12378128133110f / 8.0)}, + {FL2FXCONST_SGL(0.91400366022124f / 8.0), + FL2FXCONST_SGL(0.57972471346930f / 8.0)}, + {FL2FXCONST_SGL(-0.99925837363824f / 8.0), + FL2FXCONST_SGL(0.71084847864067f / 8.0)}, + {FL2FXCONST_SGL(-0.86875903507313f / 8.0), + FL2FXCONST_SGL(-0.20291699203564f / 8.0)}, + {FL2FXCONST_SGL(-0.26240034795124f / 8.0), + FL2FXCONST_SGL(-0.68264554369108f / 8.0)}, + {FL2FXCONST_SGL(-0.24664412953388f / 8.0), + FL2FXCONST_SGL(-0.87642273115183f / 8.0)}, + {FL2FXCONST_SGL(0.02416275806869f / 8.0), + FL2FXCONST_SGL(0.27192914288905f / 8.0)}, + {FL2FXCONST_SGL(0.82068619590515f / 8.0), + FL2FXCONST_SGL(-0.85087787994476f / 8.0)}, + {FL2FXCONST_SGL(0.88547373760759f / 8.0), + FL2FXCONST_SGL(-0.89636802901469f / 8.0)}, + {FL2FXCONST_SGL(-0.18173078152226f / 8.0), + FL2FXCONST_SGL(-0.26152145156800f / 8.0)}, + {FL2FXCONST_SGL(0.09355476558534f / 8.0), + FL2FXCONST_SGL(0.54845123045604f / 8.0)}, + {FL2FXCONST_SGL(-0.54668414224090f / 8.0), + FL2FXCONST_SGL(0.95980774020221f / 8.0)}, + {FL2FXCONST_SGL(0.37050990604091f / 8.0), + FL2FXCONST_SGL(-0.59910140383171f / 8.0)}, + {FL2FXCONST_SGL(-0.70373594262891f / 8.0), + FL2FXCONST_SGL(0.91227665827081f / 8.0)}, + {FL2FXCONST_SGL(-0.34600785879594f / 8.0), + FL2FXCONST_SGL(-0.99441426144200f / 8.0)}, + {FL2FXCONST_SGL(-0.68774481731008f / 8.0), + FL2FXCONST_SGL(-0.30238837956299f / 8.0)}, + {FL2FXCONST_SGL(-0.26843291251234f / 8.0), + FL2FXCONST_SGL(0.83115668004362f / 8.0)}, + {FL2FXCONST_SGL(0.49072334613242f / 8.0), + FL2FXCONST_SGL(-0.45359708737775f / 8.0)}, + {FL2FXCONST_SGL(0.38975993093975f / 8.0), + FL2FXCONST_SGL(0.95515358099121f / 8.0)}, + {FL2FXCONST_SGL(-0.97757125224150f / 8.0), + FL2FXCONST_SGL(0.05305894580606f / 8.0)}, + {FL2FXCONST_SGL(-0.17325552859616f / 8.0), + FL2FXCONST_SGL(-0.92770672250494f / 8.0)}, + {FL2FXCONST_SGL(0.99948035025744f / 8.0), + FL2FXCONST_SGL(0.58285545563426f / 8.0)}, + {FL2FXCONST_SGL(-0.64946246527458f / 8.0), + FL2FXCONST_SGL(0.68645507104960f / 8.0)}, + {FL2FXCONST_SGL(-0.12016920576437f / 8.0), + FL2FXCONST_SGL(-0.57147322153312f / 8.0)}, + {FL2FXCONST_SGL(-0.58947456517751f / 8.0), + FL2FXCONST_SGL(-0.34847132454388f / 8.0)}, + {FL2FXCONST_SGL(-0.41815140454465f / 8.0), + FL2FXCONST_SGL(0.16276422358861f / 8.0)}, + {FL2FXCONST_SGL(0.99885650204884f / 8.0), + FL2FXCONST_SGL(0.11136095490444f / 8.0)}, + {FL2FXCONST_SGL(-0.56649614128386f / 8.0), + FL2FXCONST_SGL(-0.90494866361587f / 8.0)}, + {FL2FXCONST_SGL(0.94138021032330f / 8.0), + FL2FXCONST_SGL(0.35281916733018f / 8.0)}, + {FL2FXCONST_SGL(-0.75725076534641f / 8.0), + FL2FXCONST_SGL(0.53650549640587f / 8.0)}, + {FL2FXCONST_SGL(0.20541973692630f / 8.0), + FL2FXCONST_SGL(-0.94435144369918f / 8.0)}, + {FL2FXCONST_SGL(0.99980371023351f / 8.0), + FL2FXCONST_SGL(0.79835913565599f / 8.0)}, + {FL2FXCONST_SGL(0.29078277605775f / 8.0), + FL2FXCONST_SGL(0.35393777921520f / 8.0)}, + {FL2FXCONST_SGL(-0.62858772103030f / 8.0), + FL2FXCONST_SGL(0.38765693387102f / 8.0)}, + {FL2FXCONST_SGL(0.43440904467688f / 8.0), + FL2FXCONST_SGL(-0.98546330463232f / 8.0)}, + {FL2FXCONST_SGL(-0.98298583762390f / 8.0), + FL2FXCONST_SGL(0.21021524625209f / 8.0)}, + {FL2FXCONST_SGL(0.19513029146934f / 8.0), + FL2FXCONST_SGL(-0.94239832251867f / 8.0)}, + {FL2FXCONST_SGL(-0.95476662400101f / 8.0), + FL2FXCONST_SGL(0.98364554179143f / 8.0)}, + {FL2FXCONST_SGL(0.93379635304810f / 8.0), + FL2FXCONST_SGL(-0.70881994583682f / 8.0)}, + {FL2FXCONST_SGL(-0.85235410573336f / 8.0), + FL2FXCONST_SGL(-0.08342347966410f / 8.0)}, + {FL2FXCONST_SGL(-0.86425093011245f / 8.0), + FL2FXCONST_SGL(-0.45795025029466f / 8.0)}, + {FL2FXCONST_SGL(0.38879779059045f / 8.0), + FL2FXCONST_SGL(0.97274429344593f / 8.0)}, + {FL2FXCONST_SGL(0.92045124735495f / 8.0), + FL2FXCONST_SGL(-0.62433652524220f / 8.0)}, + {FL2FXCONST_SGL(0.89162532251878f / 8.0), + FL2FXCONST_SGL(0.54950955570563f / 8.0)}, + {FL2FXCONST_SGL(-0.36834336949252f / 8.0), + FL2FXCONST_SGL(0.96458298020975f / 8.0)}, + {FL2FXCONST_SGL(0.93891760988045f / 8.0), + FL2FXCONST_SGL(-0.89968353740388f / 8.0)}, + {FL2FXCONST_SGL(0.99267657565094f / 8.0), + FL2FXCONST_SGL(-0.03757034316958f / 8.0)}, + {FL2FXCONST_SGL(-0.94063471614176f / 8.0), + FL2FXCONST_SGL(0.41332338538963f / 8.0)}, + {FL2FXCONST_SGL(0.99740224117019f / 8.0), + FL2FXCONST_SGL(-0.16830494996370f / 8.0)}, + {FL2FXCONST_SGL(-0.35899413170555f / 8.0), + FL2FXCONST_SGL(-0.46633226649613f / 8.0)}, + {FL2FXCONST_SGL(0.05237237274947f / 8.0), + FL2FXCONST_SGL(-0.25640361602661f / 8.0)}, + {FL2FXCONST_SGL(0.36703583957424f / 8.0), + FL2FXCONST_SGL(-0.38653265641875f / 8.0)}, + {FL2FXCONST_SGL(0.91653180367913f / 8.0), + FL2FXCONST_SGL(-0.30587628726597f / 8.0)}, + {FL2FXCONST_SGL(0.69000803499316f / 8.0), + FL2FXCONST_SGL(0.90952171386132f / 8.0)}, + {FL2FXCONST_SGL(-0.38658751133527f / 8.0), + FL2FXCONST_SGL(0.99501571208985f / 8.0)}, + {FL2FXCONST_SGL(-0.29250814029851f / 8.0), + FL2FXCONST_SGL(0.37444994344615f / 8.0)}, + {FL2FXCONST_SGL(-0.60182204677608f / 8.0), + FL2FXCONST_SGL(0.86779651036123f / 8.0)}, + {FL2FXCONST_SGL(-0.97418588163217f / 8.0), + FL2FXCONST_SGL(0.96468523666475f / 8.0)}, + {FL2FXCONST_SGL(0.88461574003963f / 8.0), + FL2FXCONST_SGL(0.57508405276414f / 8.0)}, + {FL2FXCONST_SGL(0.05198933055162f / 8.0), + FL2FXCONST_SGL(0.21269661669964f / 8.0)}, + {FL2FXCONST_SGL(-0.53499621979720f / 8.0), + FL2FXCONST_SGL(0.97241553731237f / 8.0)}, + {FL2FXCONST_SGL(-0.49429560226497f / 8.0), + FL2FXCONST_SGL(0.98183865291903f / 8.0)}, + {FL2FXCONST_SGL(-0.98935142339139f / 8.0), + FL2FXCONST_SGL(-0.40249159006933f / 8.0)}, + {FL2FXCONST_SGL(-0.98081380091130f / 8.0), + FL2FXCONST_SGL(-0.72856895534041f / 8.0)}, + {FL2FXCONST_SGL(-0.27338148835532f / 8.0), + FL2FXCONST_SGL(0.99950922447209f / 8.0)}, + {FL2FXCONST_SGL(0.06310802338302f / 8.0), + FL2FXCONST_SGL(-0.54539587529618f / 8.0)}, + {FL2FXCONST_SGL(-0.20461677199539f / 8.0), + FL2FXCONST_SGL(-0.14209977628489f / 8.0)}, + {FL2FXCONST_SGL(0.66223843141647f / 8.0), + FL2FXCONST_SGL(0.72528579940326f / 8.0)}, + {FL2FXCONST_SGL(-0.84764345483665f / 8.0), + FL2FXCONST_SGL(0.02372316801261f / 8.0)}, + {FL2FXCONST_SGL(-0.89039863483811f / 8.0), + FL2FXCONST_SGL(0.88866581484602f / 8.0)}, + {FL2FXCONST_SGL(0.95903308477986f / 8.0), + FL2FXCONST_SGL(0.76744927173873f / 8.0)}, + {FL2FXCONST_SGL(0.73504123909879f / 8.0), + FL2FXCONST_SGL(-0.03747203173192f / 8.0)}, + {FL2FXCONST_SGL(-0.31744434966056f / 8.0), + FL2FXCONST_SGL(-0.36834111883652f / 8.0)}, + {FL2FXCONST_SGL(-0.34110827591623f / 8.0), + FL2FXCONST_SGL(0.40211222807691f / 8.0)}, + {FL2FXCONST_SGL(0.47803883714199f / 8.0), + FL2FXCONST_SGL(-0.39423219786288f / 8.0)}, + {FL2FXCONST_SGL(0.98299195879514f / 8.0), + FL2FXCONST_SGL(0.01989791390047f / 8.0)}, + {FL2FXCONST_SGL(-0.30963073129751f / 8.0), + FL2FXCONST_SGL(-0.18076720599336f / 8.0)}, + {FL2FXCONST_SGL(0.99992588229018f / 8.0), + FL2FXCONST_SGL(-0.26281872094289f / 8.0)}, + {FL2FXCONST_SGL(-0.93149731080767f / 8.0), + FL2FXCONST_SGL(-0.98313162570490f / 8.0)}, + {FL2FXCONST_SGL(0.99923472302773f / 8.0), + FL2FXCONST_SGL(-0.80142993767554f / 8.0)}, + {FL2FXCONST_SGL(-0.26024169633417f / 8.0), + FL2FXCONST_SGL(-0.75999759855752f / 8.0)}, + {FL2FXCONST_SGL(-0.35712514743563f / 8.0), + FL2FXCONST_SGL(0.19298963768574f / 8.0)}, + {FL2FXCONST_SGL(-0.99899084509530f / 8.0), + FL2FXCONST_SGL(0.74645156992493f / 8.0)}, + {FL2FXCONST_SGL(0.86557171579452f / 8.0), + FL2FXCONST_SGL(0.55593866696299f / 8.0)}, + {FL2FXCONST_SGL(0.33408042438752f / 8.0), + FL2FXCONST_SGL(0.86185953874709f / 8.0)}, + {FL2FXCONST_SGL(0.99010736374716f / 8.0), + FL2FXCONST_SGL(0.04602397576623f / 8.0)}, + {FL2FXCONST_SGL(-0.66694269691195f / 8.0), + FL2FXCONST_SGL(-0.91643611810148f / 8.0)}, + {FL2FXCONST_SGL(0.64016792079480f / 8.0), + FL2FXCONST_SGL(0.15649530836856f / 8.0)}, + {FL2FXCONST_SGL(0.99570534804836f / 8.0), + FL2FXCONST_SGL(0.45844586038111f / 8.0)}, + {FL2FXCONST_SGL(-0.63431466947340f / 8.0), + FL2FXCONST_SGL(0.21079116459234f / 8.0)}, + {FL2FXCONST_SGL(-0.07706847005931f / 8.0), + FL2FXCONST_SGL(-0.89581437101329f / 8.0)}, + {FL2FXCONST_SGL(0.98590090577724f / 8.0), + FL2FXCONST_SGL(0.88241721133981f / 8.0)}, + {FL2FXCONST_SGL(0.80099335254678f / 8.0), + FL2FXCONST_SGL(-0.36851896710853f / 8.0)}, + {FL2FXCONST_SGL(0.78368131392666f / 8.0), + FL2FXCONST_SGL(0.45506999802597f / 8.0)}, + {FL2FXCONST_SGL(0.08707806671691f / 8.0), + FL2FXCONST_SGL(0.80938994918745f / 8.0)}, + {FL2FXCONST_SGL(-0.86811883080712f / 8.0), + FL2FXCONST_SGL(0.39347308654705f / 8.0)}, + {FL2FXCONST_SGL(-0.39466529740375f / 8.0), + FL2FXCONST_SGL(-0.66809432114456f / 8.0)}, + {FL2FXCONST_SGL(0.97875325649683f / 8.0), + FL2FXCONST_SGL(-0.72467840967746f / 8.0)}, + {FL2FXCONST_SGL(-0.95038560288864f / 8.0), + FL2FXCONST_SGL(0.89563219587625f / 8.0)}, + {FL2FXCONST_SGL(0.17005239424212f / 8.0), + FL2FXCONST_SGL(0.54683053962658f / 8.0)}, + {FL2FXCONST_SGL(-0.76910792026848f / 8.0), + FL2FXCONST_SGL(-0.96226617549298f / 8.0)}, + {FL2FXCONST_SGL(0.99743281016846f / 8.0), + FL2FXCONST_SGL(0.42697157037567f / 8.0)}, + {FL2FXCONST_SGL(0.95437383549973f / 8.0), + FL2FXCONST_SGL(0.97002324109952f / 8.0)}, + {FL2FXCONST_SGL(0.99578905365569f / 8.0), + FL2FXCONST_SGL(-0.54106826257356f / 8.0)}, + {FL2FXCONST_SGL(0.28058259829990f / 8.0), + FL2FXCONST_SGL(-0.85361420634036f / 8.0)}, + {FL2FXCONST_SGL(0.85256524470573f / 8.0), + FL2FXCONST_SGL(-0.64567607735589f / 8.0)}, + {FL2FXCONST_SGL(-0.50608540105128f / 8.0), + FL2FXCONST_SGL(-0.65846015480300f / 8.0)}, + {FL2FXCONST_SGL(-0.97210735183243f / 8.0), + FL2FXCONST_SGL(-0.23095213067791f / 8.0)}, + {FL2FXCONST_SGL(0.95424048234441f / 8.0), + FL2FXCONST_SGL(-0.99240147091219f / 8.0)}, + {FL2FXCONST_SGL(-0.96926570524023f / 8.0), + FL2FXCONST_SGL(0.73775654896574f / 8.0)}, + {FL2FXCONST_SGL(0.30872163214726f / 8.0), + FL2FXCONST_SGL(0.41514960556126f / 8.0)}, + {FL2FXCONST_SGL(-0.24523839572639f / 8.0), + FL2FXCONST_SGL(0.63206633394807f / 8.0)}, + {FL2FXCONST_SGL(-0.33813265086024f / 8.0), + FL2FXCONST_SGL(-0.38661779441897f / 8.0)}, + {FL2FXCONST_SGL(-0.05826828420146f / 8.0), + FL2FXCONST_SGL(-0.06940774188029f / 8.0)}, + {FL2FXCONST_SGL(-0.22898461455054f / 8.0), + FL2FXCONST_SGL(0.97054853316316f / 8.0)}, + {FL2FXCONST_SGL(-0.18509915019881f / 8.0), + FL2FXCONST_SGL(0.47565762892084f / 8.0)}, + {FL2FXCONST_SGL(-0.10488238045009f / 8.0), + FL2FXCONST_SGL(-0.87769947402394f / 8.0)}, + {FL2FXCONST_SGL(-0.71886586182037f / 8.0), + FL2FXCONST_SGL(0.78030982480538f / 8.0)}, + {FL2FXCONST_SGL(0.99793873738654f / 8.0), + FL2FXCONST_SGL(0.90041310491497f / 8.0)}, + {FL2FXCONST_SGL(0.57563307626120f / 8.0), + FL2FXCONST_SGL(-0.91034337352097f / 8.0)}, + {FL2FXCONST_SGL(0.28909646383717f / 8.0), + FL2FXCONST_SGL(0.96307783970534f / 8.0)}, + {FL2FXCONST_SGL(0.42188998312520f / 8.0), + FL2FXCONST_SGL(0.48148651230437f / 8.0)}, + {FL2FXCONST_SGL(0.93335049681047f / 8.0), + FL2FXCONST_SGL(-0.43537023883588f / 8.0)}, + {FL2FXCONST_SGL(-0.97087374418267f / 8.0), + FL2FXCONST_SGL(0.86636445711364f / 8.0)}, + {FL2FXCONST_SGL(0.36722871286923f / 8.0), + FL2FXCONST_SGL(0.65291654172961f / 8.0)}, + {FL2FXCONST_SGL(-0.81093025665696f / 8.0), + FL2FXCONST_SGL(0.08778370229363f / 8.0)}, + {FL2FXCONST_SGL(-0.26240603062237f / 8.0), + FL2FXCONST_SGL(-0.92774095379098f / 8.0)}, + {FL2FXCONST_SGL(0.83996497984604f / 8.0), + FL2FXCONST_SGL(0.55839849139647f / 8.0)}, + {FL2FXCONST_SGL(-0.99909615720225f / 8.0), + FL2FXCONST_SGL(-0.96024605713970f / 8.0)}, + {FL2FXCONST_SGL(0.74649464155061f / 8.0), + FL2FXCONST_SGL(0.12144893606462f / 8.0)}, + {FL2FXCONST_SGL(-0.74774595569805f / 8.0), + FL2FXCONST_SGL(-0.26898062008959f / 8.0)}, + {FL2FXCONST_SGL(0.95781667469567f / 8.0), + FL2FXCONST_SGL(-0.79047927052628f / 8.0)}, + {FL2FXCONST_SGL(0.95472308713099f / 8.0), + FL2FXCONST_SGL(-0.08588776019550f / 8.0)}, + {FL2FXCONST_SGL(0.48708332746299f / 8.0), + FL2FXCONST_SGL(0.99999041579432f / 8.0)}, + {FL2FXCONST_SGL(0.46332038247497f / 8.0), + FL2FXCONST_SGL(0.10964126185063f / 8.0)}, + {FL2FXCONST_SGL(-0.76497004940162f / 8.0), + FL2FXCONST_SGL(0.89210929242238f / 8.0)}, + {FL2FXCONST_SGL(0.57397389364339f / 8.0), + FL2FXCONST_SGL(0.35289703373760f / 8.0)}, + {FL2FXCONST_SGL(0.75374316974495f / 8.0), + FL2FXCONST_SGL(0.96705214651335f / 8.0)}, + {FL2FXCONST_SGL(-0.59174397685714f / 8.0), + FL2FXCONST_SGL(-0.89405370422752f / 8.0)}, + {FL2FXCONST_SGL(0.75087906691890f / 8.0), + FL2FXCONST_SGL(-0.29612672982396f / 8.0)}, + {FL2FXCONST_SGL(-0.98607857336230f / 8.0), + FL2FXCONST_SGL(0.25034911730023f / 8.0)}, + {FL2FXCONST_SGL(-0.40761056640505f / 8.0), + FL2FXCONST_SGL(-0.90045573444695f / 8.0)}, + {FL2FXCONST_SGL(0.66929266740477f / 8.0), + FL2FXCONST_SGL(0.98629493401748f / 8.0)}, + {FL2FXCONST_SGL(-0.97463695257310f / 8.0), + FL2FXCONST_SGL(-0.00190223301301f / 8.0)}, + {FL2FXCONST_SGL(0.90145509409859f / 8.0), + FL2FXCONST_SGL(0.99781390365446f / 8.0)}, + {FL2FXCONST_SGL(-0.87259289048043f / 8.0), + FL2FXCONST_SGL(0.99233587353666f / 8.0)}, + {FL2FXCONST_SGL(-0.91529461447692f / 8.0), + FL2FXCONST_SGL(-0.15698707534206f / 8.0)}, + {FL2FXCONST_SGL(-0.03305738840705f / 8.0), + FL2FXCONST_SGL(-0.37205262859764f / 8.0)}, + {FL2FXCONST_SGL(0.07223051368337f / 8.0), + FL2FXCONST_SGL(-0.88805001733626f / 8.0)}, + {FL2FXCONST_SGL(0.99498012188353f / 8.0), + FL2FXCONST_SGL(0.97094358113387f / 8.0)}, + {FL2FXCONST_SGL(-0.74904939500519f / 8.0), + FL2FXCONST_SGL(0.99985483641521f / 8.0)}, + {FL2FXCONST_SGL(0.04585228574211f / 8.0), + FL2FXCONST_SGL(0.99812337444082f / 8.0)}, + {FL2FXCONST_SGL(-0.89054954257993f / 8.0), + FL2FXCONST_SGL(-0.31791913188064f / 8.0)}, + {FL2FXCONST_SGL(-0.83782144651251f / 8.0), + FL2FXCONST_SGL(0.97637632547466f / 8.0)}, + {FL2FXCONST_SGL(0.33454804933804f / 8.0), + FL2FXCONST_SGL(-0.86231516800408f / 8.0)}, + {FL2FXCONST_SGL(-0.99707579362824f / 8.0), + FL2FXCONST_SGL(0.93237990079441f / 8.0)}, + {FL2FXCONST_SGL(-0.22827527843994f / 8.0), + FL2FXCONST_SGL(0.18874759397997f / 8.0)}, + {FL2FXCONST_SGL(0.67248046289143f / 8.0), + FL2FXCONST_SGL(-0.03646211390569f / 8.0)}, + {FL2FXCONST_SGL(-0.05146538187944f / 8.0), + FL2FXCONST_SGL(-0.92599700120679f / 8.0)}, + {FL2FXCONST_SGL(0.99947295749905f / 8.0), + FL2FXCONST_SGL(0.93625229707912f / 8.0)}, + {FL2FXCONST_SGL(0.66951124390363f / 8.0), + FL2FXCONST_SGL(0.98905825623893f / 8.0)}, + {FL2FXCONST_SGL(-0.99602956559179f / 8.0), + FL2FXCONST_SGL(-0.44654715757688f / 8.0)}, + {FL2FXCONST_SGL(0.82104905483590f / 8.0), + FL2FXCONST_SGL(0.99540741724928f / 8.0)}, + {FL2FXCONST_SGL(0.99186510988782f / 8.0), + FL2FXCONST_SGL(0.72023001312947f / 8.0)}, + {FL2FXCONST_SGL(-0.65284592392918f / 8.0), + FL2FXCONST_SGL(0.52186723253637f / 8.0)}, + {FL2FXCONST_SGL(0.93885443798188f / 8.0), + FL2FXCONST_SGL(-0.74895312615259f / 8.0)}, + {FL2FXCONST_SGL(0.96735248738388f / 8.0), + FL2FXCONST_SGL(0.90891816978629f / 8.0)}, + {FL2FXCONST_SGL(-0.22225968841114f / 8.0), + FL2FXCONST_SGL(0.57124029781228f / 8.0)}, + {FL2FXCONST_SGL(-0.44132783753414f / 8.0), + FL2FXCONST_SGL(-0.92688840659280f / 8.0)}, + {FL2FXCONST_SGL(-0.85694974219574f / 8.0), + FL2FXCONST_SGL(0.88844532719844f / 8.0)}, + {FL2FXCONST_SGL(0.91783042091762f / 8.0), + FL2FXCONST_SGL(-0.46356892383970f / 8.0)}, + {FL2FXCONST_SGL(0.72556974415690f / 8.0), + FL2FXCONST_SGL(-0.99899555770747f / 8.0)}, + {FL2FXCONST_SGL(-0.99711581834508f / 8.0), + FL2FXCONST_SGL(0.58211560180426f / 8.0)}, + {FL2FXCONST_SGL(0.77638976371966f / 8.0), + FL2FXCONST_SGL(0.94321834873819f / 8.0)}, + {FL2FXCONST_SGL(0.07717324253925f / 8.0), + FL2FXCONST_SGL(0.58638399856595f / 8.0)}, + {FL2FXCONST_SGL(-0.56049829194163f / 8.0), + FL2FXCONST_SGL(0.82522301569036f / 8.0)}, + {FL2FXCONST_SGL(0.98398893639988f / 8.0), + FL2FXCONST_SGL(0.39467440420569f / 8.0)}, + {FL2FXCONST_SGL(0.47546946844938f / 8.0), + FL2FXCONST_SGL(0.68613044836811f / 8.0)}, + {FL2FXCONST_SGL(0.65675089314631f / 8.0), + FL2FXCONST_SGL(0.18331637134880f / 8.0)}, + {FL2FXCONST_SGL(0.03273375457980f / 8.0), + FL2FXCONST_SGL(-0.74933109564108f / 8.0)}, + {FL2FXCONST_SGL(-0.38684144784738f / 8.0), + FL2FXCONST_SGL(0.51337349030406f / 8.0)}, + {FL2FXCONST_SGL(-0.97346267944545f / 8.0), + FL2FXCONST_SGL(-0.96549364384098f / 8.0)}, + {FL2FXCONST_SGL(-0.53282156061942f / 8.0), + FL2FXCONST_SGL(-0.91423265091354f / 8.0)}, + {FL2FXCONST_SGL(0.99817310731176f / 8.0), + FL2FXCONST_SGL(0.61133572482148f / 8.0)}, + {FL2FXCONST_SGL(-0.50254500772635f / 8.0), + FL2FXCONST_SGL(-0.88829338134294f / 8.0)}, + {FL2FXCONST_SGL(0.01995873238855f / 8.0), + FL2FXCONST_SGL(0.85223515096765f / 8.0)}, + {FL2FXCONST_SGL(0.99930381973804f / 8.0), + FL2FXCONST_SGL(0.94578896296649f / 8.0)}, + {FL2FXCONST_SGL(0.82907767600783f / 8.0), + FL2FXCONST_SGL(-0.06323442598128f / 8.0)}, + {FL2FXCONST_SGL(-0.58660709669728f / 8.0), + FL2FXCONST_SGL(0.96840773806582f / 8.0)}, + {FL2FXCONST_SGL(-0.17573736667267f / 8.0), + FL2FXCONST_SGL(-0.48166920859485f / 8.0)}, + {FL2FXCONST_SGL(0.83434292401346f / 8.0), + FL2FXCONST_SGL(-0.13023450646997f / 8.0)}, + {FL2FXCONST_SGL(0.05946491307025f / 8.0), + FL2FXCONST_SGL(0.20511047074866f / 8.0)}, + {FL2FXCONST_SGL(0.81505484574602f / 8.0), + FL2FXCONST_SGL(-0.94685947861369f / 8.0)}, + {FL2FXCONST_SGL(-0.44976380954860f / 8.0), + FL2FXCONST_SGL(0.40894572671545f / 8.0)}, + {FL2FXCONST_SGL(-0.89746474625671f / 8.0), + FL2FXCONST_SGL(0.99846578838537f / 8.0)}, + {FL2FXCONST_SGL(0.39677256130792f / 8.0), + FL2FXCONST_SGL(-0.74854668609359f / 8.0)}, + {FL2FXCONST_SGL(-0.07588948563079f / 8.0), + FL2FXCONST_SGL(0.74096214084170f / 8.0)}, + {FL2FXCONST_SGL(0.76343198951445f / 8.0), + FL2FXCONST_SGL(0.41746629422634f / 8.0)}, + {FL2FXCONST_SGL(-0.74490104699626f / 8.0), + FL2FXCONST_SGL(0.94725911744610f / 8.0)}, + {FL2FXCONST_SGL(0.64880119792759f / 8.0), + FL2FXCONST_SGL(0.41336660830571f / 8.0)}, + {FL2FXCONST_SGL(0.62319537462542f / 8.0), + FL2FXCONST_SGL(-0.93098313552599f / 8.0)}, + {FL2FXCONST_SGL(0.42215817594807f / 8.0), + FL2FXCONST_SGL(-0.07712787385208f / 8.0)}, + {FL2FXCONST_SGL(0.02704554141885f / 8.0), + FL2FXCONST_SGL(-0.05417518053666f / 8.0)}, + {FL2FXCONST_SGL(0.80001773566818f / 8.0), + FL2FXCONST_SGL(0.91542195141039f / 8.0)}, + {FL2FXCONST_SGL(-0.79351832348816f / 8.0), + FL2FXCONST_SGL(-0.36208897989136f / 8.0)}, + {FL2FXCONST_SGL(0.63872359151636f / 8.0), + FL2FXCONST_SGL(0.08128252493444f / 8.0)}, + {FL2FXCONST_SGL(0.52890520960295f / 8.0), + FL2FXCONST_SGL(0.60048872455592f / 8.0)}, + {FL2FXCONST_SGL(0.74238552914587f / 8.0), + FL2FXCONST_SGL(0.04491915291044f / 8.0)}, + {FL2FXCONST_SGL(0.99096131449250f / 8.0), + FL2FXCONST_SGL(-0.19451182854402f / 8.0)}, + {FL2FXCONST_SGL(-0.80412329643109f / 8.0), + FL2FXCONST_SGL(-0.88513818199457f / 8.0)}, + {FL2FXCONST_SGL(-0.64612616129736f / 8.0), + FL2FXCONST_SGL(0.72198674804544f / 8.0)}, + {FL2FXCONST_SGL(0.11657770663191f / 8.0), + FL2FXCONST_SGL(-0.83662833815041f / 8.0)}, + {FL2FXCONST_SGL(-0.95053182488101f / 8.0), + FL2FXCONST_SGL(-0.96939905138082f / 8.0)}, + {FL2FXCONST_SGL(-0.62228872928622f / 8.0), + FL2FXCONST_SGL(0.82767262846661f / 8.0)}, + {FL2FXCONST_SGL(0.03004475787316f / 8.0), + FL2FXCONST_SGL(-0.99738896333384f / 8.0)}, + {FL2FXCONST_SGL(-0.97987214341034f / 8.0), + FL2FXCONST_SGL(0.36526129686425f / 8.0)}, + {FL2FXCONST_SGL(-0.99986980746200f / 8.0), + FL2FXCONST_SGL(-0.36021610299715f / 8.0)}, + {FL2FXCONST_SGL(0.89110648599879f / 8.0), + FL2FXCONST_SGL(-0.97894250343044f / 8.0)}, + {FL2FXCONST_SGL(0.10407960510582f / 8.0), + FL2FXCONST_SGL(0.77357793811619f / 8.0)}, + {FL2FXCONST_SGL(0.95964737821728f / 8.0), + FL2FXCONST_SGL(-0.35435818285502f / 8.0)}, + {FL2FXCONST_SGL(0.50843233159162f / 8.0), + FL2FXCONST_SGL(0.96107691266205f / 8.0)}, + {FL2FXCONST_SGL(0.17006334670615f / 8.0), + FL2FXCONST_SGL(-0.76854025314829f / 8.0)}, + {FL2FXCONST_SGL(0.25872675063360f / 8.0), + FL2FXCONST_SGL(0.99893303933816f / 8.0)}, + {FL2FXCONST_SGL(-0.01115998681937f / 8.0), + FL2FXCONST_SGL(0.98496019742444f / 8.0)}, + {FL2FXCONST_SGL(-0.79598702973261f / 8.0), + FL2FXCONST_SGL(0.97138411318894f / 8.0)}, + {FL2FXCONST_SGL(-0.99264708948101f / 8.0), + FL2FXCONST_SGL(-0.99542822402536f / 8.0)}, + {FL2FXCONST_SGL(-0.99829663752818f / 8.0), + FL2FXCONST_SGL(0.01877138824311f / 8.0)}, + {FL2FXCONST_SGL(-0.70801016548184f / 8.0), + FL2FXCONST_SGL(0.33680685948117f / 8.0)}, + {FL2FXCONST_SGL(-0.70467057786826f / 8.0), + FL2FXCONST_SGL(0.93272777501857f / 8.0)}, + {FL2FXCONST_SGL(0.99846021905254f / 8.0), + FL2FXCONST_SGL(-0.98725746254433f / 8.0)}, + {FL2FXCONST_SGL(-0.63364968534650f / 8.0), + FL2FXCONST_SGL(-0.16473594423746f / 8.0)}, + {FL2FXCONST_SGL(-0.16258217500792f / 8.0), + FL2FXCONST_SGL(-0.95939125400802f / 8.0)}, + {FL2FXCONST_SGL(-0.43645594360633f / 8.0), + FL2FXCONST_SGL(-0.94805030113284f / 8.0)}, + {FL2FXCONST_SGL(-0.99848471702976f / 8.0), + FL2FXCONST_SGL(0.96245166923809f / 8.0)}, + {FL2FXCONST_SGL(-0.16796458968998f / 8.0), + FL2FXCONST_SGL(-0.98987511890470f / 8.0)}, + {FL2FXCONST_SGL(-0.87979225745213f / 8.0), + FL2FXCONST_SGL(-0.71725725041680f / 8.0)}, + {FL2FXCONST_SGL(0.44183099021786f / 8.0), + FL2FXCONST_SGL(-0.93568974498761f / 8.0)}, + {FL2FXCONST_SGL(0.93310180125532f / 8.0), + FL2FXCONST_SGL(-0.99913308068246f / 8.0)}, + {FL2FXCONST_SGL(-0.93941931782002f / 8.0), + FL2FXCONST_SGL(-0.56409379640356f / 8.0)}, + {FL2FXCONST_SGL(-0.88590003188677f / 8.0), + FL2FXCONST_SGL(0.47624600491382f / 8.0)}, + {FL2FXCONST_SGL(0.99971463703691f / 8.0), + FL2FXCONST_SGL(-0.83889954253462f / 8.0)}, + {FL2FXCONST_SGL(-0.75376385639978f / 8.0), + FL2FXCONST_SGL(0.00814643438625f / 8.0)}, + {FL2FXCONST_SGL(0.93887685615875f / 8.0), + FL2FXCONST_SGL(-0.11284528204636f / 8.0)}, + {FL2FXCONST_SGL(0.85126435782309f / 8.0), + FL2FXCONST_SGL(0.52349251543547f / 8.0)}, + {FL2FXCONST_SGL(0.39701421446381f / 8.0), + FL2FXCONST_SGL(0.81779634174316f / 8.0)}, + {FL2FXCONST_SGL(-0.37024464187437f / 8.0), + FL2FXCONST_SGL(-0.87071656222959f / 8.0)}, + {FL2FXCONST_SGL(-0.36024828242896f / 8.0), + FL2FXCONST_SGL(0.34655735648287f / 8.0)}, + {FL2FXCONST_SGL(-0.93388812549209f / 8.0), + FL2FXCONST_SGL(-0.84476541096429f / 8.0)}, + {FL2FXCONST_SGL(-0.65298804552119f / 8.0), + FL2FXCONST_SGL(-0.18439575450921f / 8.0)}, + {FL2FXCONST_SGL(0.11960319006843f / 8.0), + FL2FXCONST_SGL(0.99899346780168f / 8.0)}, + {FL2FXCONST_SGL(0.94292565553160f / 8.0), + FL2FXCONST_SGL(0.83163906518293f / 8.0)}, + {FL2FXCONST_SGL(0.75081145286948f / 8.0), + FL2FXCONST_SGL(-0.35533223142265f / 8.0)}, + {FL2FXCONST_SGL(0.56721979748394f / 8.0), + FL2FXCONST_SGL(-0.24076836414499f / 8.0)}, + {FL2FXCONST_SGL(0.46857766746029f / 8.0), + FL2FXCONST_SGL(-0.30140233457198f / 8.0)}, + {FL2FXCONST_SGL(0.97312313923635f / 8.0), + FL2FXCONST_SGL(-0.99548191630031f / 8.0)}, + {FL2FXCONST_SGL(-0.38299976567017f / 8.0), + FL2FXCONST_SGL(0.98516909715427f / 8.0)}, + {FL2FXCONST_SGL(0.41025800019463f / 8.0), + FL2FXCONST_SGL(0.02116736935734f / 8.0)}, + {FL2FXCONST_SGL(0.09638062008048f / 8.0), + FL2FXCONST_SGL(0.04411984381457f / 8.0)}, + {FL2FXCONST_SGL(-0.85283249275397f / 8.0), + FL2FXCONST_SGL(0.91475563922421f / 8.0)}, + {FL2FXCONST_SGL(0.88866808958124f / 8.0), + FL2FXCONST_SGL(-0.99735267083226f / 8.0)}, + {FL2FXCONST_SGL(-0.48202429536989f / 8.0), + FL2FXCONST_SGL(-0.96805608884164f / 8.0)}, + {FL2FXCONST_SGL(0.27572582416567f / 8.0), + FL2FXCONST_SGL(0.58634753335832f / 8.0)}, + {FL2FXCONST_SGL(-0.65889129659168f / 8.0), + FL2FXCONST_SGL(0.58835634138583f / 8.0)}, + {FL2FXCONST_SGL(0.98838086953732f / 8.0), + FL2FXCONST_SGL(0.99994349600236f / 8.0)}, + {FL2FXCONST_SGL(-0.20651349620689f / 8.0), + FL2FXCONST_SGL(0.54593044066355f / 8.0)}, + {FL2FXCONST_SGL(-0.62126416356920f / 8.0), + FL2FXCONST_SGL(-0.59893681700392f / 8.0)}, + {FL2FXCONST_SGL(0.20320105410437f / 8.0), + FL2FXCONST_SGL(-0.86879180355289f / 8.0)}, + {FL2FXCONST_SGL(-0.97790548600584f / 8.0), + FL2FXCONST_SGL(0.96290806999242f / 8.0)}, + {FL2FXCONST_SGL(0.11112534735126f / 8.0), + FL2FXCONST_SGL(0.21484763313301f / 8.0)}, + {FL2FXCONST_SGL(-0.41368337314182f / 8.0), + FL2FXCONST_SGL(0.28216837680365f / 8.0)}, + {FL2FXCONST_SGL(0.24133038992960f / 8.0), + FL2FXCONST_SGL(0.51294362630238f / 8.0)}, + {FL2FXCONST_SGL(-0.66393410674885f / 8.0), + FL2FXCONST_SGL(-0.08249679629081f / 8.0)}, + {FL2FXCONST_SGL(-0.53697829178752f / 8.0), + FL2FXCONST_SGL(-0.97649903936228f / 8.0)}, + {FL2FXCONST_SGL(-0.97224737889348f / 8.0), + FL2FXCONST_SGL(0.22081333579837f / 8.0)}, + {FL2FXCONST_SGL(0.87392477144549f / 8.0), + FL2FXCONST_SGL(-0.12796173740361f / 8.0)}, + {FL2FXCONST_SGL(0.19050361015753f / 8.0), + FL2FXCONST_SGL(0.01602615387195f / 8.0)}, + {FL2FXCONST_SGL(-0.46353441212724f / 8.0), + FL2FXCONST_SGL(-0.95249041539006f / 8.0)}, + {FL2FXCONST_SGL(-0.07064096339021f / 8.0), + FL2FXCONST_SGL(-0.94479803205886f / 8.0)}, + {FL2FXCONST_SGL(-0.92444085484466f / 8.0), + FL2FXCONST_SGL(-0.10457590187436f / 8.0)}, + {FL2FXCONST_SGL(-0.83822593578728f / 8.0), + FL2FXCONST_SGL(-0.01695043208885f / 8.0)}, + {FL2FXCONST_SGL(0.75214681811150f / 8.0), + FL2FXCONST_SGL(-0.99955681042665f / 8.0)}, + {FL2FXCONST_SGL(-0.42102998829339f / 8.0), + FL2FXCONST_SGL(0.99720941999394f / 8.0)}, + {FL2FXCONST_SGL(-0.72094786237696f / 8.0), + FL2FXCONST_SGL(-0.35008961934255f / 8.0)}, + {FL2FXCONST_SGL(0.78843311019251f / 8.0), + FL2FXCONST_SGL(0.52851398958271f / 8.0)}, + {FL2FXCONST_SGL(0.97394027897442f / 8.0), + FL2FXCONST_SGL(-0.26695944086561f / 8.0)}, + {FL2FXCONST_SGL(0.99206463477946f / 8.0), + FL2FXCONST_SGL(-0.57010120849429f / 8.0)}, + {FL2FXCONST_SGL(0.76789609461795f / 8.0), + FL2FXCONST_SGL(-0.76519356730966f / 8.0)}, + {FL2FXCONST_SGL(-0.82002421836409f / 8.0), + FL2FXCONST_SGL(-0.73530179553767f / 8.0)}, + {FL2FXCONST_SGL(0.81924990025724f / 8.0), + FL2FXCONST_SGL(0.99698425250579f / 8.0)}, + {FL2FXCONST_SGL(-0.26719850873357f / 8.0), + FL2FXCONST_SGL(0.68903369776193f / 8.0)}, + {FL2FXCONST_SGL(-0.43311260380975f / 8.0), + FL2FXCONST_SGL(0.85321815947490f / 8.0)}, + {FL2FXCONST_SGL(0.99194979673836f / 8.0), + FL2FXCONST_SGL(0.91876249766422f / 8.0)}, + {FL2FXCONST_SGL(-0.80692001248487f / 8.0), + FL2FXCONST_SGL(-0.32627540663214f / 8.0)}, + {FL2FXCONST_SGL(0.43080003649976f / 8.0), + FL2FXCONST_SGL(-0.21919095636638f / 8.0)}, + {FL2FXCONST_SGL(0.67709491937357f / 8.0), + FL2FXCONST_SGL(-0.95478075822906f / 8.0)}, + {FL2FXCONST_SGL(0.56151770568316f / 8.0), + FL2FXCONST_SGL(-0.70693811747778f / 8.0)}, + {FL2FXCONST_SGL(0.10831862810749f / 8.0), + FL2FXCONST_SGL(-0.08628837174592f / 8.0)}, + {FL2FXCONST_SGL(0.91229417540436f / 8.0), + FL2FXCONST_SGL(-0.65987351408410f / 8.0)}, + {FL2FXCONST_SGL(-0.48972893932274f / 8.0), + FL2FXCONST_SGL(0.56289246362686f / 8.0)}, + {FL2FXCONST_SGL(-0.89033658689697f / 8.0), + FL2FXCONST_SGL(-0.71656563987082f / 8.0)}, + {FL2FXCONST_SGL(0.65269447475094f / 8.0), + FL2FXCONST_SGL(0.65916004833932f / 8.0)}, + {FL2FXCONST_SGL(0.67439478141121f / 8.0), + FL2FXCONST_SGL(-0.81684380846796f / 8.0)}, + {FL2FXCONST_SGL(-0.47770832416973f / 8.0), + FL2FXCONST_SGL(-0.16789556203025f / 8.0)}, + {FL2FXCONST_SGL(-0.99715979260878f / 8.0), + FL2FXCONST_SGL(-0.93565784007648f / 8.0)}, + {FL2FXCONST_SGL(-0.90889593602546f / 8.0), + FL2FXCONST_SGL(0.62034397054380f / 8.0)}, + {FL2FXCONST_SGL(-0.06618622548177f / 8.0), + FL2FXCONST_SGL(-0.23812217221359f / 8.0)}, + {FL2FXCONST_SGL(0.99430266919728f / 8.0), + FL2FXCONST_SGL(0.18812555317553f / 8.0)}, + {FL2FXCONST_SGL(0.97686402381843f / 8.0), + FL2FXCONST_SGL(-0.28664534366620f / 8.0)}, + {FL2FXCONST_SGL(0.94813650221268f / 8.0), + FL2FXCONST_SGL(-0.97506640027128f / 8.0)}, + {FL2FXCONST_SGL(-0.95434497492853f / 8.0), + FL2FXCONST_SGL(-0.79607978501983f / 8.0)}, + {FL2FXCONST_SGL(-0.49104783137150f / 8.0), + FL2FXCONST_SGL(0.32895214359663f / 8.0)}, + {FL2FXCONST_SGL(0.99881175120751f / 8.0), + FL2FXCONST_SGL(0.88993983831354f / 8.0)}, + {FL2FXCONST_SGL(0.50449166760303f / 8.0), + FL2FXCONST_SGL(-0.85995072408434f / 8.0)}, + {FL2FXCONST_SGL(0.47162891065108f / 8.0), + FL2FXCONST_SGL(-0.18680204049569f / 8.0)}, + {FL2FXCONST_SGL(-0.62081581361840f / 8.0), + FL2FXCONST_SGL(0.75000676218956f / 8.0)}, + {FL2FXCONST_SGL(-0.43867015250812f / 8.0), + FL2FXCONST_SGL(0.99998069244322f / 8.0)}, + {FL2FXCONST_SGL(0.98630563232075f / 8.0), + FL2FXCONST_SGL(-0.53578899600662f / 8.0)}, + {FL2FXCONST_SGL(-0.61510362277374f / 8.0), + FL2FXCONST_SGL(-0.89515019899997f / 8.0)}, + {FL2FXCONST_SGL(-0.03841517601843f / 8.0), + FL2FXCONST_SGL(-0.69888815681179f / 8.0)}, + {FL2FXCONST_SGL(-0.30102157304644f / 8.0), + FL2FXCONST_SGL(-0.07667808922205f / 8.0)}, + {FL2FXCONST_SGL(0.41881284182683f / 8.0), + FL2FXCONST_SGL(0.02188098922282f / 8.0)}, + {FL2FXCONST_SGL(-0.86135454941237f / 8.0), + FL2FXCONST_SGL(0.98947480909359f / 8.0)}, + {FL2FXCONST_SGL(0.67226861393788f / 8.0), + FL2FXCONST_SGL(-0.13494389011014f / 8.0)}, + {FL2FXCONST_SGL(-0.70737398842068f / 8.0), + FL2FXCONST_SGL(-0.76547349325992f / 8.0)}, + {FL2FXCONST_SGL(0.94044946687963f / 8.0), + FL2FXCONST_SGL(0.09026201157416f / 8.0)}, + {FL2FXCONST_SGL(-0.82386352534327f / 8.0), + FL2FXCONST_SGL(0.08924768823676f / 8.0)}, + {FL2FXCONST_SGL(-0.32070666698656f / 8.0), + FL2FXCONST_SGL(0.50143421908753f / 8.0)}, + {FL2FXCONST_SGL(0.57593163224487f / 8.0), + FL2FXCONST_SGL(-0.98966422921509f / 8.0)}, + {FL2FXCONST_SGL(-0.36326018419965f / 8.0), + FL2FXCONST_SGL(0.07440243123228f / 8.0)}, + {FL2FXCONST_SGL(0.99979044674350f / 8.0), + FL2FXCONST_SGL(-0.14130287347405f / 8.0)}, + {FL2FXCONST_SGL(-0.92366023326932f / 8.0), + FL2FXCONST_SGL(-0.97979298068180f / 8.0)}, + {FL2FXCONST_SGL(-0.44607178518598f / 8.0), + FL2FXCONST_SGL(-0.54233252016394f / 8.0)}, + {FL2FXCONST_SGL(0.44226800932956f / 8.0), + FL2FXCONST_SGL(0.71326756742752f / 8.0)}, + {FL2FXCONST_SGL(0.03671907158312f / 8.0), + FL2FXCONST_SGL(0.63606389366675f / 8.0)}, + {FL2FXCONST_SGL(0.52175424682195f / 8.0), + FL2FXCONST_SGL(-0.85396826735705f / 8.0)}, + {FL2FXCONST_SGL(-0.94701139690956f / 8.0), + FL2FXCONST_SGL(-0.01826348194255f / 8.0)}, + {FL2FXCONST_SGL(-0.98759606946049f / 8.0), + FL2FXCONST_SGL(0.82288714303073f / 8.0)}, + {FL2FXCONST_SGL(0.87434794743625f / 8.0), + FL2FXCONST_SGL(0.89399495655433f / 8.0)}, + {FL2FXCONST_SGL(-0.93412041758744f / 8.0), + FL2FXCONST_SGL(0.41374052024363f / 8.0)}, + {FL2FXCONST_SGL(0.96063943315511f / 8.0), + FL2FXCONST_SGL(0.93116709541280f / 8.0)}, + {FL2FXCONST_SGL(0.97534253457837f / 8.0), + FL2FXCONST_SGL(0.86150930812689f / 8.0)}, + {FL2FXCONST_SGL(0.99642466504163f / 8.0), + FL2FXCONST_SGL(0.70190043427512f / 8.0)}, + {FL2FXCONST_SGL(-0.94705089665984f / 8.0), + FL2FXCONST_SGL(-0.29580042814306f / 8.0)}, + {FL2FXCONST_SGL(0.91599807087376f / 8.0), + FL2FXCONST_SGL(-0.98147830385781f / 8.0)}}; +//@} + +/* +static const FIXP_SGL harmonicPhase [2][4] = { + { 1.0, 0.0, -1.0, 0.0}, + { 0.0, 1.0, 0.0, -1.0} +}; +*/ + +/* tables for SBR and AAC LD */ +/* table for 8 time slot index */ +const int FDK_sbrDecoder_envelopeTable_8[8][5] = { + /* transientIndex nEnv, tranIdx, shortEnv, border1, border2, ... */ + /* borders from left to right side; -1 = not in use */ + /*[|T-|------]*/ {2, 0, 0, 1, -1}, + /*[|-T-|-----]*/ {2, 0, 0, 2, -1}, + /*[--|T-|----]*/ {3, 1, 1, 2, 4}, + /*[---|T-|---]*/ {3, 1, 1, 3, 5}, + /*[----|T-|--]*/ {3, 1, 1, 4, 6}, + /*[-----|T--|]*/ {2, 1, 1, 5, -1}, + /*[------|T-|]*/ {2, 1, 1, 6, -1}, + /*[-------|T|]*/ {2, 1, 1, 7, -1}, +}; + +/* table for 15 time slot index */ +const int FDK_sbrDecoder_envelopeTable_15[15][6] = { + /* transientIndex nEnv, tranIdx, shortEnv, border1, border2, ... */ + /* length from left to right side; -1 = not in use */ + /*[|T---|------------]*/ {2, 0, 0, 4, -1, -1}, + /*[|-T---|-----------]*/ {2, 0, 0, 5, -1, -1}, + /*[|--|T---|---------]*/ {3, 1, 1, 2, 6, -1}, + /*[|---|T---|--------]*/ {3, 1, 1, 3, 7, -1}, + /*[|----|T---|-------]*/ {3, 1, 1, 4, 8, -1}, + /*[|-----|T---|------]*/ {3, 1, 1, 5, 9, -1}, + /*[|------|T---|-----]*/ {3, 1, 1, 6, 10, -1}, + /*[|-------|T---|----]*/ {3, 1, 1, 7, 11, -1}, + /*[|--------|T---|---]*/ {3, 1, 1, 8, 12, -1}, + /*[|---------|T---|--]*/ {3, 1, 1, 9, 13, -1}, + /*[|----------|T----|]*/ {2, 1, 1, 10, -1, -1}, + /*[|-----------|T---|]*/ {2, 1, 1, 11, -1, -1}, + /*[|------------|T--|]*/ {2, 1, 1, 12, -1, -1}, + /*[|-------------|T-|]*/ {2, 1, 1, 13, -1, -1}, + /*[|--------------|T|]*/ {2, 1, 1, 14, -1, -1}, +}; + +/* table for 16 time slot index */ +const int FDK_sbrDecoder_envelopeTable_16[16][6] = { + /* transientIndex nEnv, tranIdx, shortEnv, border1, border2, ... */ + /* length from left to right side; -1 = not in use */ + /*[|T---|------------|]*/ {2, 0, 0, 4, -1, -1}, + /*[|-T---|-----------|]*/ {2, 0, 0, 5, -1, -1}, + /*[|--|T---|----------]*/ {3, 1, 1, 2, 6, -1}, + /*[|---|T---|---------]*/ {3, 1, 1, 3, 7, -1}, + /*[|----|T---|--------]*/ {3, 1, 1, 4, 8, -1}, + /*[|-----|T---|-------]*/ {3, 1, 1, 5, 9, -1}, + /*[|------|T---|------]*/ {3, 1, 1, 6, 10, -1}, + /*[|-------|T---|-----]*/ {3, 1, 1, 7, 11, -1}, + /*[|--------|T---|----]*/ {3, 1, 1, 8, 12, -1}, + /*[|---------|T---|---]*/ {3, 1, 1, 9, 13, -1}, + /*[|----------|T---|--]*/ {3, 1, 1, 10, 14, -1}, + /*[|-----------|T----|]*/ {2, 1, 1, 11, -1, -1}, + /*[|------------|T---|]*/ {2, 1, 1, 12, -1, -1}, + /*[|-------------|T--|]*/ {2, 1, 1, 13, -1, -1}, + /*[|--------------|T-|]*/ {2, 1, 1, 14, -1, -1}, + /*[|---------------|T|]*/ {2, 1, 1, 15, -1, -1}, +}; + +/*! + \name FrameInfoDefaults + + Predefined envelope positions for the FIX-FIX case (static framing) +*/ +//@{ +const FRAME_INFO FDK_sbrDecoder_sbr_frame_info1_15 = { + 0, 1, {0, 15, 0, 0, 0, 0}, {1, 0, 0, 0, 0}, -1, 1, {0, 15, 0}, {0, 0, 0}, + 0, 0}; +const FRAME_INFO FDK_sbrDecoder_sbr_frame_info2_15 = { + 0, 2, {0, 8, 15, 0, 0, 0}, {1, 1, 0, 0, 0}, -1, 2, {0, 8, 15}, {0, 0, 0}, + 0, 0}; +const FRAME_INFO FDK_sbrDecoder_sbr_frame_info4_15 = { + 0, 4, {0, 4, 8, 12, 15, 0}, {1, 1, 1, 1, 0}, -1, 2, {0, 8, 15}, {0, 0, 0}, + 0, 0}; +#if (MAX_ENVELOPES >= 8) +const FRAME_INFO FDK_sbrDecoder_sbr_frame_info8_15 = { + 0, + 8, + {0, 2, 4, 6, 8, 10, 12, 14, 15}, + {1, 1, 1, 1, 1, 1, 1, 1}, + -1, + 2, + {0, 8, 15}, + {0, 0, 0}, + 0, + 0}; +#endif + +const FRAME_INFO FDK_sbrDecoder_sbr_frame_info1_16 = { + 0, 1, {0, 16, 0, 0, 0, 0}, {1, 0, 0, 0, 0}, -1, 1, {0, 16, 0}, {0, 0, 0}, + 0, 0}; +const FRAME_INFO FDK_sbrDecoder_sbr_frame_info2_16 = { + 0, 2, {0, 8, 16, 0, 0, 0}, {1, 1, 0, 0, 0}, -1, 2, {0, 8, 16}, {0, 0, 0}, + 0, 0}; +const FRAME_INFO FDK_sbrDecoder_sbr_frame_info4_16 = { + 0, 4, {0, 4, 8, 12, 16, 0}, {1, 1, 1, 1, 0}, -1, 2, {0, 8, 16}, {0, 0, 0}, + 0, 0}; + +#if (MAX_ENVELOPES >= 8) +const FRAME_INFO FDK_sbrDecoder_sbr_frame_info8_16 = { + 0, + 8, + {0, 2, 4, 6, 8, 10, 12, 14, 16}, + {1, 1, 1, 1, 1, 1, 1, 1}, + -1, + 2, + {0, 8, 16}, + {0, 0, 0}, + 0, + 0}; +#endif + +//@} + +/*! + \name SBR_HuffmanTables + + SBR Huffman Table Overview: \n + \n + o envelope level, 1.5 dB: \n + 1) sbr_huffBook_EnvLevel10T[120][2] \n + 2) sbr_huffBook_EnvLevel10F[120][2] \n + \n + o envelope balance, 1.5 dB: \n + 3) sbr_huffBook_EnvBalance10T[48][2] \n + 4) sbr_huffBook_EnvBalance10F[48][2] \n + \n + o envelope level, 3.0 dB: \n + 5) sbr_huffBook_EnvLevel11T[62][2] \n + 6) sbr_huffBook_EnvLevel11F[62][2] \n + \n + o envelope balance, 3.0 dB: \n + 7) sbr_huffBook_EnvBalance11T[24][2] \n + 8) sbr_huffBook_EnvBalance11F[24][2] \n + \n + o noise level, 3.0 dB: \n + 9) sbr_huffBook_NoiseLevel11T[62][2] \n + -) (sbr_huffBook_EnvLevel11F[62][2] is used for freq dir)\n + \n + o noise balance, 3.0 dB: \n + 10) sbr_huffBook_NoiseBalance11T[24][2]\n + -) (sbr_huffBook_EnvBalance11F[24][2] is used for freq dir)\n + \n + (1.5 dB is never used for noise) + +*/ +//@{ +const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel10T[120][2] = { + {1, 2}, {-64, -65}, {3, 4}, {-63, -66}, {5, 6}, + {-62, -67}, {7, 8}, {-61, -68}, {9, 10}, {-60, -69}, + {11, 12}, {-59, -70}, {13, 14}, {-58, -71}, {15, 16}, + {-57, -72}, {17, 18}, {-73, -56}, {19, 21}, {-74, 20}, + {-55, -75}, {22, 26}, {23, 24}, {-54, -76}, {-77, 25}, + {-53, -78}, {27, 34}, {28, 29}, {-52, -79}, {30, 31}, + {-80, -51}, {32, 33}, {-83, -82}, {-81, -50}, {35, 57}, + {36, 40}, {37, 38}, {-88, -84}, {-48, 39}, {-90, -85}, + {41, 46}, {42, 43}, {-49, -87}, {44, 45}, {-89, -86}, + {-124, -123}, {47, 50}, {48, 49}, {-122, -121}, {-120, -119}, + {51, 54}, {52, 53}, {-118, -117}, {-116, -115}, {55, 56}, + {-114, -113}, {-112, -111}, {58, 89}, {59, 74}, {60, 67}, + {61, 64}, {62, 63}, {-110, -109}, {-108, -107}, {65, 66}, + {-106, -105}, {-104, -103}, {68, 71}, {69, 70}, {-102, -101}, + {-100, -99}, {72, 73}, {-98, -97}, {-96, -95}, {75, 82}, + {76, 79}, {77, 78}, {-94, -93}, {-92, -91}, {80, 81}, + {-47, -46}, {-45, -44}, {83, 86}, {84, 85}, {-43, -42}, + {-41, -40}, {87, 88}, {-39, -38}, {-37, -36}, {90, 105}, + {91, 98}, {92, 95}, {93, 94}, {-35, -34}, {-33, -32}, + {96, 97}, {-31, -30}, {-29, -28}, {99, 102}, {100, 101}, + {-27, -26}, {-25, -24}, {103, 104}, {-23, -22}, {-21, -20}, + {106, 113}, {107, 110}, {108, 109}, {-19, -18}, {-17, -16}, + {111, 112}, {-15, -14}, {-13, -12}, {114, 117}, {115, 116}, + {-11, -10}, {-9, -8}, {118, 119}, {-7, -6}, {-5, -4}}; + +const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel10F[120][2] = { + {1, 2}, {-64, -65}, {3, 4}, {-63, -66}, {5, 6}, + {-67, -62}, {7, 8}, {-68, -61}, {9, 10}, {-69, -60}, + {11, 13}, {-70, 12}, {-59, -71}, {14, 16}, {-58, 15}, + {-72, -57}, {17, 19}, {-73, 18}, {-56, -74}, {20, 23}, + {21, 22}, {-55, -75}, {-54, -53}, {24, 27}, {25, 26}, + {-76, -52}, {-77, -51}, {28, 31}, {29, 30}, {-50, -78}, + {-79, -49}, {32, 36}, {33, 34}, {-48, -47}, {-80, 35}, + {-81, -82}, {37, 47}, {38, 41}, {39, 40}, {-83, -46}, + {-45, -84}, {42, 44}, {-85, 43}, {-44, -43}, {45, 46}, + {-88, -87}, {-86, -90}, {48, 66}, {49, 56}, {50, 53}, + {51, 52}, {-92, -42}, {-41, -39}, {54, 55}, {-105, -89}, + {-38, -37}, {57, 60}, {58, 59}, {-94, -91}, {-40, -36}, + {61, 63}, {-20, 62}, {-115, -110}, {64, 65}, {-108, -107}, + {-101, -97}, {67, 89}, {68, 75}, {69, 72}, {70, 71}, + {-95, -93}, {-34, -27}, {73, 74}, {-22, -17}, {-16, -124}, + {76, 82}, {77, 79}, {-123, 78}, {-122, -121}, {80, 81}, + {-120, -119}, {-118, -117}, {83, 86}, {84, 85}, {-116, -114}, + {-113, -112}, {87, 88}, {-111, -109}, {-106, -104}, {90, 105}, + {91, 98}, {92, 95}, {93, 94}, {-103, -102}, {-100, -99}, + {96, 97}, {-98, -96}, {-35, -33}, {99, 102}, {100, 101}, + {-32, -31}, {-30, -29}, {103, 104}, {-28, -26}, {-25, -24}, + {106, 113}, {107, 110}, {108, 109}, {-23, -21}, {-19, -18}, + {111, 112}, {-15, -14}, {-13, -12}, {114, 117}, {115, 116}, + {-11, -10}, {-9, -8}, {118, 119}, {-7, -6}, {-5, -4}}; + +const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance10T[48][2] = { + {-64, 1}, {-63, 2}, {-65, 3}, {-62, 4}, {-66, 5}, {-61, 6}, + {-67, 7}, {-60, 8}, {-68, 9}, {10, 11}, {-69, -59}, {12, 13}, + {-70, -58}, {14, 28}, {15, 21}, {16, 18}, {-57, 17}, {-71, -56}, + {19, 20}, {-88, -87}, {-86, -85}, {22, 25}, {23, 24}, {-84, -83}, + {-82, -81}, {26, 27}, {-80, -79}, {-78, -77}, {29, 36}, {30, 33}, + {31, 32}, {-76, -75}, {-74, -73}, {34, 35}, {-72, -55}, {-54, -53}, + {37, 41}, {38, 39}, {-52, -51}, {-50, 40}, {-49, -48}, {42, 45}, + {43, 44}, {-47, -46}, {-45, -44}, {46, 47}, {-43, -42}, {-41, -40}}; + +const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance10F[48][2] = { + {-64, 1}, {-65, 2}, {-63, 3}, {-66, 4}, {-62, 5}, {-61, 6}, + {-67, 7}, {-68, 8}, {-60, 9}, {10, 11}, {-69, -59}, {-70, 12}, + {-58, 13}, {14, 17}, {-71, 15}, {-57, 16}, {-56, -73}, {18, 32}, + {19, 25}, {20, 22}, {-72, 21}, {-88, -87}, {23, 24}, {-86, -85}, + {-84, -83}, {26, 29}, {27, 28}, {-82, -81}, {-80, -79}, {30, 31}, + {-78, -77}, {-76, -75}, {33, 40}, {34, 37}, {35, 36}, {-74, -55}, + {-54, -53}, {38, 39}, {-52, -51}, {-50, -49}, {41, 44}, {42, 43}, + {-48, -47}, {-46, -45}, {45, 46}, {-44, -43}, {-42, 47}, {-41, -40}}; + +const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel11T[62][2] = { + {-64, 1}, {-65, 2}, {-63, 3}, {-66, 4}, {-62, 5}, {-67, 6}, + {-61, 7}, {-68, 8}, {-60, 9}, {10, 11}, {-69, -59}, {12, 14}, + {-70, 13}, {-71, -58}, {15, 18}, {16, 17}, {-72, -57}, {-73, -74}, + {19, 22}, {-56, 20}, {-55, 21}, {-54, -77}, {23, 31}, {24, 25}, + {-75, -76}, {26, 27}, {-78, -53}, {28, 29}, {-52, -95}, {-94, 30}, + {-93, -92}, {32, 47}, {33, 40}, {34, 37}, {35, 36}, {-91, -90}, + {-89, -88}, {38, 39}, {-87, -86}, {-85, -84}, {41, 44}, {42, 43}, + {-83, -82}, {-81, -80}, {45, 46}, {-79, -51}, {-50, -49}, {48, 55}, + {49, 52}, {50, 51}, {-48, -47}, {-46, -45}, {53, 54}, {-44, -43}, + {-42, -41}, {56, 59}, {57, 58}, {-40, -39}, {-38, -37}, {60, 61}, + {-36, -35}, {-34, -33}}; + +const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel11F[62][2] = { + {-64, 1}, {-65, 2}, {-63, 3}, {-66, 4}, {-62, 5}, {-67, 6}, + {7, 8}, {-61, -68}, {9, 10}, {-60, -69}, {11, 12}, {-59, -70}, + {13, 14}, {-58, -71}, {15, 16}, {-57, -72}, {17, 19}, {-56, 18}, + {-55, -73}, {20, 24}, {21, 22}, {-74, -54}, {-53, 23}, {-75, -76}, + {25, 30}, {26, 27}, {-52, -51}, {28, 29}, {-77, -79}, {-50, -49}, + {31, 39}, {32, 35}, {33, 34}, {-78, -46}, {-82, -88}, {36, 37}, + {-83, -48}, {-47, 38}, {-86, -85}, {40, 47}, {41, 44}, {42, 43}, + {-80, -44}, {-43, -42}, {45, 46}, {-39, -87}, {-84, -40}, {48, 55}, + {49, 52}, {50, 51}, {-95, -94}, {-93, -92}, {53, 54}, {-91, -90}, + {-89, -81}, {56, 59}, {57, 58}, {-45, -41}, {-38, -37}, {60, 61}, + {-36, -35}, {-34, -33}}; + +const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance11T[24][2] = { + {-64, 1}, {-63, 2}, {-65, 3}, {-66, 4}, {-62, 5}, {-61, 6}, + {-67, 7}, {-68, 8}, {-60, 9}, {10, 16}, {11, 13}, {-69, 12}, + {-76, -75}, {14, 15}, {-74, -73}, {-72, -71}, {17, 20}, {18, 19}, + {-70, -59}, {-58, -57}, {21, 22}, {-56, -55}, {-54, 23}, {-53, -52}}; + +const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance11F[24][2] = { + {-64, 1}, {-65, 2}, {-63, 3}, {-66, 4}, {-62, 5}, {-61, 6}, + {-67, 7}, {-68, 8}, {-60, 9}, {10, 13}, {-69, 11}, {-59, 12}, + {-58, -76}, {14, 17}, {15, 16}, {-75, -74}, {-73, -72}, {18, 21}, + {19, 20}, {-71, -70}, {-57, -56}, {22, 23}, {-55, -54}, {-53, -52}}; + +const SCHAR FDK_sbrDecoder_sbr_huffBook_NoiseLevel11T[62][2] = { + {-64, 1}, {-63, 2}, {-65, 3}, {-66, 4}, {-62, 5}, {-67, 6}, + {7, 8}, {-61, -68}, {9, 30}, {10, 15}, {-60, 11}, {-69, 12}, + {13, 14}, {-59, -53}, {-95, -94}, {16, 23}, {17, 20}, {18, 19}, + {-93, -92}, {-91, -90}, {21, 22}, {-89, -88}, {-87, -86}, {24, 27}, + {25, 26}, {-85, -84}, {-83, -82}, {28, 29}, {-81, -80}, {-79, -78}, + {31, 46}, {32, 39}, {33, 36}, {34, 35}, {-77, -76}, {-75, -74}, + {37, 38}, {-73, -72}, {-71, -70}, {40, 43}, {41, 42}, {-58, -57}, + {-56, -55}, {44, 45}, {-54, -52}, {-51, -50}, {47, 54}, {48, 51}, + {49, 50}, {-49, -48}, {-47, -46}, {52, 53}, {-45, -44}, {-43, -42}, + {55, 58}, {56, 57}, {-41, -40}, {-39, -38}, {59, 60}, {-37, -36}, + {-35, 61}, {-34, -33}}; + +const SCHAR FDK_sbrDecoder_sbr_huffBook_NoiseBalance11T[24][2] = { + {-64, 1}, {-65, 2}, {-63, 3}, {4, 9}, {-66, 5}, {-62, 6}, + {7, 8}, {-76, -75}, {-74, -73}, {10, 17}, {11, 14}, {12, 13}, + {-72, -71}, {-70, -69}, {15, 16}, {-68, -67}, {-61, -60}, {18, 21}, + {19, 20}, {-59, -58}, {-57, -56}, {22, 23}, {-55, -54}, {-53, -52}}; +//@} + +/*! + \name parametric stereo + \brief constants used by the parametric stereo part of the decoder + +*/ + +/* constants used in psbitdec.cpp */ + +/* FIX_BORDER can have 0, 1, 2, 4 envelopes */ +const UCHAR FDK_sbrDecoder_aFixNoEnvDecode[4] = {0, 1, 2, 4}; + +/* IID & ICC Huffman codebooks */ +const SCHAR aBookPsIidTimeDecode[28][2] = { + {-64, 1}, {-65, 2}, {-63, 3}, {-66, 4}, {-62, 5}, {-67, 6}, + {-61, 7}, {-68, 8}, {-60, 9}, {-69, 10}, {-59, 11}, {-70, 12}, + {-58, 13}, {-57, 14}, {-71, 15}, {16, 17}, {-56, -72}, {18, 21}, + {19, 20}, {-55, -78}, {-77, -76}, {22, 25}, {23, 24}, {-75, -74}, + {-73, -54}, {26, 27}, {-53, -52}, {-51, -50}}; + +const SCHAR aBookPsIidFreqDecode[28][2] = { + {-64, 1}, {2, 3}, {-63, -65}, {4, 5}, {-62, -66}, {6, 7}, + {-61, -67}, {8, 9}, {-68, -60}, {-59, 10}, {-69, 11}, {-58, 12}, + {-70, 13}, {-71, 14}, {-57, 15}, {16, 17}, {-56, -72}, {18, 19}, + {-55, -54}, {20, 21}, {-73, -53}, {22, 24}, {-74, 23}, {-75, -78}, + {25, 26}, {-77, -76}, {-52, 27}, {-51, -50}}; + +const SCHAR aBookPsIccTimeDecode[14][2] = { + {-64, 1}, {-63, 2}, {-65, 3}, {-62, 4}, {-66, 5}, {-61, 6}, {-67, 7}, + {-60, 8}, {-68, 9}, {-59, 10}, {-69, 11}, {-58, 12}, {-70, 13}, {-71, -57}}; + +const SCHAR aBookPsIccFreqDecode[14][2] = { + {-64, 1}, {-63, 2}, {-65, 3}, {-62, 4}, {-66, 5}, {-61, 6}, {-67, 7}, + {-60, 8}, {-59, 9}, {-68, 10}, {-58, 11}, {-69, 12}, {-57, 13}, {-70, -71}}; + +/* IID-fine Huffman codebooks */ + +const SCHAR aBookPsIidFineTimeDecode[60][2] = { + {1, -64}, {-63, 2}, {3, -65}, {4, 59}, {5, 7}, {6, -67}, + {-68, -60}, {-61, 8}, {9, 11}, {-59, 10}, {-70, -58}, {12, 41}, + {13, 20}, {14, -71}, {-55, 15}, {-53, 16}, {17, -77}, {18, 19}, + {-85, -84}, {-46, -45}, {-57, 21}, {22, 40}, {23, 29}, {-51, 24}, + {25, 26}, {-83, -82}, {27, 28}, {-90, -38}, {-92, -91}, {30, 37}, + {31, 34}, {32, 33}, {-35, -34}, {-37, -36}, {35, 36}, {-94, -93}, + {-89, -39}, {38, -79}, {39, -81}, {-88, -40}, {-74, -54}, {42, -69}, + {43, 44}, {-72, -56}, {45, 52}, {46, 50}, {47, -76}, {-49, 48}, + {-47, 49}, {-87, -41}, {-52, 51}, {-78, -50}, {53, -73}, {54, -75}, + {55, 57}, {56, -80}, {-86, -42}, {-48, 58}, {-44, -43}, {-66, -62}}; + +const SCHAR aBookPsIidFineFreqDecode[60][2] = { + {1, -64}, {2, 4}, {3, -65}, {-66, -62}, {-63, 5}, {6, 7}, + {-67, -61}, {8, 9}, {-68, -60}, {10, 11}, {-69, -59}, {12, 13}, + {-70, -58}, {14, 18}, {-57, 15}, {16, -72}, {-54, 17}, {-75, -53}, + {19, 37}, {-56, 20}, {21, -73}, {22, 29}, {23, -76}, {24, -78}, + {25, 28}, {26, 27}, {-85, -43}, {-83, -45}, {-81, -47}, {-52, 30}, + {-50, 31}, {32, -79}, {33, 34}, {-82, -46}, {35, 36}, {-90, -89}, + {-92, -91}, {38, -71}, {-55, 39}, {40, -74}, {41, 50}, {42, -77}, + {-49, 43}, {44, 47}, {45, 46}, {-86, -42}, {-88, -87}, {48, 49}, + {-39, -38}, {-41, -40}, {-51, 51}, {52, 59}, {53, 56}, {54, 55}, + {-35, -34}, {-37, -36}, {57, 58}, {-94, -93}, {-84, -44}, {-80, -48}}; + +/* constants used in psdec.cpp */ + +/* the values of the following 3 tables are shiftet right by 1 ! */ +const FIXP_DBL ScaleFactors[NO_IID_LEVELS] = { + + 0x5a5ded00, 0x59cd0400, 0x58c29680, 0x564c2e80, 0x52a3d480, + 0x4c8be080, 0x46df3080, 0x40000000, 0x384ba5c0, 0x304c2980, + 0x24e9f640, 0x1b4a2940, 0x11b5c0a0, 0x0b4e2540, 0x0514ea90}; + +const FIXP_DBL ScaleFactorsFine[NO_IID_LEVELS_FINE] = { + + 0x5a825c00, 0x5a821c00, 0x5a815100, 0x5a7ed000, 0x5a76e600, 0x5a5ded00, + 0x5a39b880, 0x59f1fd00, 0x5964d680, 0x5852ca00, 0x564c2e80, 0x54174480, + 0x50ea7500, 0x4c8be080, 0x46df3080, 0x40000000, 0x384ba5c0, 0x304c2980, + 0x288dd240, 0x217a2900, 0x1b4a2940, 0x13c5ece0, 0x0e2b0090, 0x0a178ef0, + 0x072ab798, 0x0514ea90, 0x02dc5944, 0x019bf87c, 0x00e7b173, 0x00824b8b, + 0x00494568}; +const FIXP_DBL Alphas[NO_ICC_LEVELS] = { + + 0x00000000, 0x0b6b5be0, 0x12485f80, 0x1da2fa40, + 0x2637ebc0, 0x3243f6c0, 0x466b7480, 0x6487ed80}; + +const UCHAR bins2groupMap20[NO_IID_GROUPS] = { + 0, 0, 1, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19}; + +const UCHAR FDK_sbrDecoder_aNoIidBins[3] = { + NO_LOW_RES_IID_BINS, NO_MID_RES_IID_BINS, NO_HI_RES_IID_BINS}; + +const UCHAR FDK_sbrDecoder_aNoIccBins[3] = { + NO_LOW_RES_ICC_BINS, NO_MID_RES_ICC_BINS, NO_HI_RES_ICC_BINS}; + +/************************************************************************/ +/*! + \brief Create lookup tables for some arithmetic functions + + The tables would normally be defined as const arrays, + but initialization at run time allows to specify their accuracy. +*/ +/************************************************************************/ + +/* 1/x-table: (example for INV_TABLE_BITS 8) + + The table covers an input range from 0.5 to 1.0 with a step size of 1/512, + starting at 0.5 + 1/512. + Each table entry corresponds to an input interval starting 1/1024 below the + exact value and ending 1/1024 above it. + + The table is actually a 0.5/x-table, so that the output range is again + 0.5...1.0 and the exponent of the result must be increased by 1. + + Input range Index in table result + ------------------------------------------------------------------- + 0.500000...0.500976 - 0.5 / 0.500000 = 1.000000 + 0.500976...0.502930 0 0.5 / 0.501953 = 0.996109 + 0.502930...0.500488 1 0.5 / 0.503906 = 0.992248 + ... + 0.999023...1.000000 255 0.5 / 1.000000 = 0.500000 + + for (i=0; i<INV_TABLE_SIZE; i++) { + d = 0.5f / ( 0.5f+(double)(i+1)/(INV_TABLE_SIZE*2) ) ; + invTable[i] = FL2FX_SGL(d); + } +*/ +const FIXP_SGL FDK_sbrDecoder_invTable[INV_TABLE_SIZE] = { + 0x7f80, 0x7f01, 0x7e83, 0x7e07, 0x7d8b, 0x7d11, 0x7c97, 0x7c1e, 0x7ba6, + 0x7b2f, 0x7ab9, 0x7a44, 0x79cf, 0x795c, 0x78e9, 0x7878, 0x7807, 0x7796, + 0x7727, 0x76b9, 0x764b, 0x75de, 0x7572, 0x7506, 0x749c, 0x7432, 0x73c9, + 0x7360, 0x72f9, 0x7292, 0x722c, 0x71c6, 0x7161, 0x70fd, 0x709a, 0x7037, + 0x6fd5, 0x6f74, 0x6f13, 0x6eb3, 0x6e54, 0x6df5, 0x6d97, 0x6d39, 0x6cdc, + 0x6c80, 0x6c24, 0x6bc9, 0x6b6f, 0x6b15, 0x6abc, 0x6a63, 0x6a0b, 0x69b3, + 0x695c, 0x6906, 0x68b0, 0x685a, 0x6806, 0x67b1, 0x675e, 0x670a, 0x66b8, + 0x6666, 0x6614, 0x65c3, 0x6572, 0x6522, 0x64d2, 0x6483, 0x6434, 0x63e6, + 0x6399, 0x634b, 0x62fe, 0x62b2, 0x6266, 0x621b, 0x61d0, 0x6185, 0x613b, + 0x60f2, 0x60a8, 0x6060, 0x6017, 0x5fcf, 0x5f88, 0x5f41, 0x5efa, 0x5eb4, + 0x5e6e, 0x5e28, 0x5de3, 0x5d9f, 0x5d5a, 0x5d17, 0x5cd3, 0x5c90, 0x5c4d, + 0x5c0b, 0x5bc9, 0x5b87, 0x5b46, 0x5b05, 0x5ac4, 0x5a84, 0x5a44, 0x5a05, + 0x59c6, 0x5987, 0x5949, 0x590a, 0x58cd, 0x588f, 0x5852, 0x5815, 0x57d9, + 0x579d, 0x5761, 0x5725, 0x56ea, 0x56af, 0x5675, 0x563b, 0x5601, 0x55c7, + 0x558e, 0x5555, 0x551c, 0x54e3, 0x54ab, 0x5473, 0x543c, 0x5405, 0x53ce, + 0x5397, 0x5360, 0x532a, 0x52f4, 0x52bf, 0x5289, 0x5254, 0x521f, 0x51eb, + 0x51b7, 0x5183, 0x514f, 0x511b, 0x50e8, 0x50b5, 0x5082, 0x5050, 0x501d, + 0x4feb, 0x4fba, 0x4f88, 0x4f57, 0x4f26, 0x4ef5, 0x4ec4, 0x4e94, 0x4e64, + 0x4e34, 0x4e04, 0x4dd5, 0x4da6, 0x4d77, 0x4d48, 0x4d19, 0x4ceb, 0x4cbd, + 0x4c8f, 0x4c61, 0x4c34, 0x4c07, 0x4bd9, 0x4bad, 0x4b80, 0x4b54, 0x4b27, + 0x4afb, 0x4acf, 0x4aa4, 0x4a78, 0x4a4d, 0x4a22, 0x49f7, 0x49cd, 0x49a2, + 0x4978, 0x494e, 0x4924, 0x48fa, 0x48d1, 0x48a7, 0x487e, 0x4855, 0x482d, + 0x4804, 0x47dc, 0x47b3, 0x478b, 0x4763, 0x473c, 0x4714, 0x46ed, 0x46c5, + 0x469e, 0x4677, 0x4651, 0x462a, 0x4604, 0x45de, 0x45b8, 0x4592, 0x456c, + 0x4546, 0x4521, 0x44fc, 0x44d7, 0x44b2, 0x448d, 0x4468, 0x4444, 0x441f, + 0x43fb, 0x43d7, 0x43b3, 0x4390, 0x436c, 0x4349, 0x4325, 0x4302, 0x42df, + 0x42bc, 0x4299, 0x4277, 0x4254, 0x4232, 0x4210, 0x41ee, 0x41cc, 0x41aa, + 0x4189, 0x4167, 0x4146, 0x4125, 0x4104, 0x40e3, 0x40c2, 0x40a1, 0x4081, + 0x4060, 0x4040, 0x4020, 0x4000}; diff --git a/fdk-aac/libSBRdec/src/sbr_rom.h b/fdk-aac/libSBRdec/src/sbr_rom.h new file mode 100644 index 0000000..039743c --- /dev/null +++ b/fdk-aac/libSBRdec/src/sbr_rom.h @@ -0,0 +1,216 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! +\file +\brief Declaration of constant tables +*/ +#ifndef SBR_ROM_H +#define SBR_ROM_H + +#include "sbrdecoder.h" +#include "env_extr.h" +#include "qmf.h" + +#define INV_INT_TABLE_SIZE 49 +#define SBR_NF_NO_RANDOM_VAL \ + 512 /*!< Size of random number array for noise floor */ + +/* + Frequency scales +*/ + +/* if defined(SBRDEC_RATIO_16_64_ENABLE) ((4) = 4) else ((4) = 2) */ +extern const UCHAR FDK_sbrDecoder_sbr_start_freq_16[(4) / 2][16]; +extern const UCHAR FDK_sbrDecoder_sbr_start_freq_22[(4) / 2][16]; +extern const UCHAR FDK_sbrDecoder_sbr_start_freq_24[(4) / 2][16]; +extern const UCHAR FDK_sbrDecoder_sbr_start_freq_32[(4) / 2][16]; +extern const UCHAR FDK_sbrDecoder_sbr_start_freq_40[(4) / 2][16]; +extern const UCHAR FDK_sbrDecoder_sbr_start_freq_44[(4) / 2][16]; +extern const UCHAR FDK_sbrDecoder_sbr_start_freq_48[(4) / 2][16]; +extern const UCHAR FDK_sbrDecoder_sbr_start_freq_64[(4) / 2][16]; +extern const UCHAR FDK_sbrDecoder_sbr_start_freq_88[(4) / 2][16]; +extern const UCHAR FDK_sbrDecoder_sbr_start_freq_192[16]; +extern const UCHAR FDK_sbrDecoder_sbr_start_freq_176[16]; +extern const UCHAR FDK_sbrDecoder_sbr_start_freq_128[16]; + +/* + Low-Power-Profile Transposer +*/ +#define NUM_WHFACTOR_TABLE_ENTRIES 9 +extern const USHORT + FDK_sbrDecoder_sbr_whFactorsIndex[NUM_WHFACTOR_TABLE_ENTRIES]; +extern const FIXP_DBL + FDK_sbrDecoder_sbr_whFactorsTable[NUM_WHFACTOR_TABLE_ENTRIES][6]; + +/* + Envelope Adjustor +*/ +extern const FIXP_SGL FDK_sbrDecoder_sbr_limGains_m[4]; +extern const UCHAR FDK_sbrDecoder_sbr_limGains_e[4]; +extern const FIXP_SGL FDK_sbrDecoder_sbr_limGainsPvc_m[4]; +extern const UCHAR FDK_sbrDecoder_sbr_limGainsPvc_e[4]; +extern const FIXP_SGL FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4[4]; +extern const FIXP_DBL FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4_DBL[4]; +extern const FIXP_SGL FDK_sbrDecoder_sbr_smoothFilter[4]; +extern const FIXP_SGL FDK_sbrDecoder_sbr_randomPhase[SBR_NF_NO_RANDOM_VAL][2]; + +/* + Envelope Extractor +*/ +extern const int FDK_sbrDecoder_envelopeTable_8[8][5]; +extern const int FDK_sbrDecoder_envelopeTable_15[15][6]; +extern const int FDK_sbrDecoder_envelopeTable_16[16][6]; +extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info1_15; +extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info2_15; +extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info4_15; +extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info8_15; +extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info1_16; +extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info2_16; +extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info4_16; +extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info8_16; + +extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel10T[120][2]; +extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel10F[120][2]; +extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance10T[48][2]; +extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance10F[48][2]; +extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel11T[62][2]; +extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel11F[62][2]; +extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance11T[24][2]; +extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance11F[24][2]; +extern const SCHAR FDK_sbrDecoder_sbr_huffBook_NoiseLevel11T[62][2]; +extern const SCHAR FDK_sbrDecoder_sbr_huffBook_NoiseBalance11T[24][2]; + +/* + Parametric stereo +*/ + +/* FIX_BORDER can have 0, 1, 2, 4 envelops */ +extern const UCHAR FDK_sbrDecoder_aFixNoEnvDecode[4]; + +/* IID & ICC Huffman codebooks */ +extern const SCHAR aBookPsIidTimeDecode[28][2]; +extern const SCHAR aBookPsIidFreqDecode[28][2]; +extern const SCHAR aBookPsIccTimeDecode[14][2]; +extern const SCHAR aBookPsIccFreqDecode[14][2]; + +/* IID-fine Huffman codebooks */ + +extern const SCHAR aBookPsIidFineTimeDecode[60][2]; +extern const SCHAR aBookPsIidFineFreqDecode[60][2]; + +/* the values of the following 3 tables are shiftet right by 1 ! */ +extern const FIXP_DBL ScaleFactors[NO_IID_LEVELS]; +extern const FIXP_DBL ScaleFactorsFine[NO_IID_LEVELS_FINE]; +extern const FIXP_DBL Alphas[NO_ICC_LEVELS]; + +extern const UCHAR bins2groupMap20[NO_IID_GROUPS]; +extern const UCHAR FDK_sbrDecoder_aNoIidBins[3]; +extern const UCHAR FDK_sbrDecoder_aNoIccBins[3]; + +/* Lookup tables for some arithmetic functions */ + +#define INV_TABLE_BITS 8 +#define INV_TABLE_SIZE (1 << INV_TABLE_BITS) +extern const FIXP_SGL FDK_sbrDecoder_invTable[INV_TABLE_SIZE]; + +#endif // SBR_ROM_H diff --git a/fdk-aac/libSBRdec/src/sbrdec_drc.cpp b/fdk-aac/libSBRdec/src/sbrdec_drc.cpp new file mode 100644 index 0000000..2d73f32 --- /dev/null +++ b/fdk-aac/libSBRdec/src/sbrdec_drc.cpp @@ -0,0 +1,528 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): Christian Griebel + + Description: Dynamic range control (DRC) decoder tool for SBR + +*******************************************************************************/ + +#include "sbrdec_drc.h" + +/* DRC - Offset table for QMF interpolation. Shifted by one index position. + The table defines the (short) window borders rounded to the nearest QMF + timeslot. It has the size 16 because it is accessed with the + drcInterpolationScheme that is read from the bitstream with 4 bit. */ +static const UCHAR winBorderToColMappingTab[2][16] = { + /*-1, 0, 1, 2, 3, 4, 5, 6, 7, 8 */ + {0, 0, 4, 8, 12, 16, 20, 24, 28, 32, 32, 32, 32, 32, 32, + 32}, /* 1024 framing */ + {0, 0, 4, 8, 11, 15, 19, 23, 26, 30, 30, 30, 30, 30, 30, + 30} /* 960 framing */ +}; + +/*! + \brief Initialize DRC QMF factors + + \hDrcData Handle to DRC channel data. + + \return none +*/ +void sbrDecoder_drcInitChannel(HANDLE_SBR_DRC_CHANNEL hDrcData) { + int band; + + if (hDrcData == NULL) { + return; + } + + for (band = 0; band < (64); band++) { + hDrcData->prevFact_mag[band] = FL2FXCONST_DBL(0.5f); + } + + for (band = 0; band < SBRDEC_MAX_DRC_BANDS; band++) { + hDrcData->currFact_mag[band] = FL2FXCONST_DBL(0.5f); + hDrcData->nextFact_mag[band] = FL2FXCONST_DBL(0.5f); + } + + hDrcData->prevFact_exp = 1; + hDrcData->currFact_exp = 1; + hDrcData->nextFact_exp = 1; + + hDrcData->numBandsCurr = 1; + hDrcData->numBandsNext = 1; + + hDrcData->winSequenceCurr = 0; + hDrcData->winSequenceNext = 0; + + hDrcData->drcInterpolationSchemeCurr = 0; + hDrcData->drcInterpolationSchemeNext = 0; + + hDrcData->enable = 0; +} + +/*! + \brief Swap DRC QMF scaling factors after they have been applied. + + \hDrcData Handle to DRC channel data. + + \return none +*/ +void sbrDecoder_drcUpdateChannel(HANDLE_SBR_DRC_CHANNEL hDrcData) { + if (hDrcData == NULL) { + return; + } + if (hDrcData->enable != 1) { + return; + } + + /* swap previous data */ + FDKmemcpy(hDrcData->currFact_mag, hDrcData->nextFact_mag, + SBRDEC_MAX_DRC_BANDS * sizeof(FIXP_DBL)); + + hDrcData->currFact_exp = hDrcData->nextFact_exp; + + hDrcData->numBandsCurr = hDrcData->numBandsNext; + + FDKmemcpy(hDrcData->bandTopCurr, hDrcData->bandTopNext, + SBRDEC_MAX_DRC_BANDS * sizeof(USHORT)); + + hDrcData->drcInterpolationSchemeCurr = hDrcData->drcInterpolationSchemeNext; + + hDrcData->winSequenceCurr = hDrcData->winSequenceNext; +} + +/*! + \brief Apply DRC factors slot based. + + \hDrcData Handle to DRC channel data. + \qmfRealSlot Pointer to real valued QMF data of one time slot. + \qmfImagSlot Pointer to the imaginary QMF data of one time slot. + \col Number of the time slot. + \numQmfSubSamples Total number of time slots for one frame. + \scaleFactor Pointer to the out scale factor of the time slot. + + \return None. +*/ +void sbrDecoder_drcApplySlot(HANDLE_SBR_DRC_CHANNEL hDrcData, + FIXP_DBL *qmfRealSlot, FIXP_DBL *qmfImagSlot, + int col, int numQmfSubSamples, int maxShift) { + const UCHAR *winBorderToColMap; + + int band, bottomMdct, topMdct, bin, useLP; + int indx = numQmfSubSamples - (numQmfSubSamples >> 1) - 10; /* l_border */ + int frameLenFlag = (numQmfSubSamples == 30) ? 1 : 0; + int frameSize = (frameLenFlag == 1) ? 960 : 1024; + + const FIXP_DBL *fact_mag = NULL; + INT fact_exp = 0; + UINT numBands = 0; + USHORT *bandTop = NULL; + int shortDrc = 0; + + FIXP_DBL alphaValue = FL2FXCONST_DBL(0.0f); + + if (hDrcData == NULL) { + return; + } + if (hDrcData->enable != 1) { + return; + } + + winBorderToColMap = winBorderToColMappingTab[frameLenFlag]; + + useLP = (qmfImagSlot == NULL) ? 1 : 0; + + col += indx; + bottomMdct = 0; + + /* get respective data and calc interpolation factor */ + if (col < (numQmfSubSamples >> 1)) { /* first half of current frame */ + if (hDrcData->winSequenceCurr != 2) { /* long window */ + int j = col + (numQmfSubSamples >> 1); + + if (hDrcData->drcInterpolationSchemeCurr == 0) { + INT k = (frameLenFlag) ? 0x4444445 : 0x4000000; + + alphaValue = (FIXP_DBL)(j * k); + } else { + if (j >= (int)winBorderToColMap[hDrcData->drcInterpolationSchemeCurr]) { + alphaValue = (FIXP_DBL)MAXVAL_DBL; + } + } + } else { /* short windows */ + shortDrc = 1; + } + + fact_mag = hDrcData->currFact_mag; + fact_exp = hDrcData->currFact_exp; + numBands = hDrcData->numBandsCurr; + bandTop = hDrcData->bandTopCurr; + } else if (col < numQmfSubSamples) { /* second half of current frame */ + if (hDrcData->winSequenceNext != 2) { /* next: long window */ + int j = col - (numQmfSubSamples >> 1); + + if (hDrcData->drcInterpolationSchemeNext == 0) { + INT k = (frameLenFlag) ? 0x4444445 : 0x4000000; + + alphaValue = (FIXP_DBL)(j * k); + } else { + if (j >= (int)winBorderToColMap[hDrcData->drcInterpolationSchemeNext]) { + alphaValue = (FIXP_DBL)MAXVAL_DBL; + } + } + + fact_mag = hDrcData->nextFact_mag; + fact_exp = hDrcData->nextFact_exp; + numBands = hDrcData->numBandsNext; + bandTop = hDrcData->bandTopNext; + } else { /* next: short windows */ + if (hDrcData->winSequenceCurr != 2) { /* current: long window */ + alphaValue = (FIXP_DBL)0; + + fact_mag = hDrcData->nextFact_mag; + fact_exp = hDrcData->nextFact_exp; + numBands = hDrcData->numBandsNext; + bandTop = hDrcData->bandTopNext; + } else { /* current: short windows */ + shortDrc = 1; + + fact_mag = hDrcData->currFact_mag; + fact_exp = hDrcData->currFact_exp; + numBands = hDrcData->numBandsCurr; + bandTop = hDrcData->bandTopCurr; + } + } + } else { /* first half of next frame */ + if (hDrcData->winSequenceNext != 2) { /* long window */ + int j = col - (numQmfSubSamples >> 1); + + if (hDrcData->drcInterpolationSchemeNext == 0) { + INT k = (frameLenFlag) ? 0x4444445 : 0x4000000; + + alphaValue = (FIXP_DBL)(j * k); + } else { + if (j >= (int)winBorderToColMap[hDrcData->drcInterpolationSchemeNext]) { + alphaValue = (FIXP_DBL)MAXVAL_DBL; + } + } + } else { /* short windows */ + shortDrc = 1; + } + + fact_mag = hDrcData->nextFact_mag; + fact_exp = hDrcData->nextFact_exp; + numBands = hDrcData->numBandsNext; + bandTop = hDrcData->bandTopNext; + + col -= numQmfSubSamples; + } + + /* process bands */ + for (band = 0; band < (int)numBands; band++) { + int bottomQmf, topQmf; + + FIXP_DBL drcFact_mag = (FIXP_DBL)MAXVAL_DBL; + + topMdct = (bandTop[band] + 1) << 2; + + if (!shortDrc) { /* long window */ + if (frameLenFlag) { + /* 960 framing */ + bottomQmf = fMultIfloor((FIXP_DBL)0x4444445, bottomMdct); + topQmf = fMultIfloor((FIXP_DBL)0x4444445, topMdct); + + topMdct = 30 * topQmf; + } else { + /* 1024 framing */ + topMdct &= ~0x1f; + + bottomQmf = bottomMdct >> 5; + topQmf = topMdct >> 5; + } + + if (band == ((int)numBands - 1)) { + topQmf = (64); + } + + for (bin = bottomQmf; bin < topQmf; bin++) { + FIXP_DBL drcFact1_mag = hDrcData->prevFact_mag[bin]; + FIXP_DBL drcFact2_mag = fact_mag[band]; + + /* normalize scale factors */ + if (hDrcData->prevFact_exp < maxShift) { + drcFact1_mag >>= maxShift - hDrcData->prevFact_exp; + } + if (fact_exp < maxShift) { + drcFact2_mag >>= maxShift - fact_exp; + } + + /* interpolate */ + if (alphaValue == (FIXP_DBL)0) { + drcFact_mag = drcFact1_mag; + } else if (alphaValue == (FIXP_DBL)MAXVAL_DBL) { + drcFact_mag = drcFact2_mag; + } else { + drcFact_mag = + fMult(alphaValue, drcFact2_mag) + + fMult(((FIXP_DBL)MAXVAL_DBL - alphaValue), drcFact1_mag); + } + + /* apply scaling */ + qmfRealSlot[bin] = fMult(qmfRealSlot[bin], drcFact_mag); + if (!useLP) { + qmfImagSlot[bin] = fMult(qmfImagSlot[bin], drcFact_mag); + } + + /* save previous factors */ + if (col == (numQmfSubSamples >> 1) - 1) { + hDrcData->prevFact_mag[bin] = fact_mag[band]; + } + } + } else { /* short windows */ + unsigned startWinIdx, stopWinIdx; + int startCol, stopCol; + FIXP_DBL invFrameSizeDiv8 = + (frameLenFlag) ? (FIXP_DBL)0x1111112 : (FIXP_DBL)0x1000000; + + /* limit top at the frame borders */ + if (topMdct < 0) { + topMdct = 0; + } + if (topMdct >= frameSize) { + topMdct = frameSize - 1; + } + + if (frameLenFlag) { + /* 960 framing */ + topMdct = fMultIfloor((FIXP_DBL)0x78000000, + fMultIfloor((FIXP_DBL)0x22222223, topMdct) << 2); + + startWinIdx = fMultIfloor(invFrameSizeDiv8, bottomMdct) + + 1; /* winBorderToColMap table has offset of 1 */ + stopWinIdx = fMultIceil(invFrameSizeDiv8 - (FIXP_DBL)1, topMdct) + 1; + } else { + /* 1024 framing */ + topMdct &= ~0x03; + + startWinIdx = fMultIfloor(invFrameSizeDiv8, bottomMdct) + 1; + stopWinIdx = fMultIceil(invFrameSizeDiv8, topMdct) + 1; + } + + /* startCol is truncated to the nearest corresponding start subsample in + the QMF of the short window bottom is present in:*/ + startCol = (int)winBorderToColMap[startWinIdx]; + + /* stopCol is rounded upwards to the nearest corresponding stop subsample + in the QMF of the short window top is present in. */ + stopCol = (int)winBorderToColMap[stopWinIdx]; + + bottomQmf = fMultIfloor(invFrameSizeDiv8, + ((bottomMdct % (numQmfSubSamples << 2)) << 5)); + topQmf = fMultIfloor(invFrameSizeDiv8, + ((topMdct % (numQmfSubSamples << 2)) << 5)); + + /* extend last band */ + if (band == ((int)numBands - 1)) { + topQmf = (64); + stopCol = numQmfSubSamples; + stopWinIdx = 10; + } + + if (topQmf == 0) { + if (frameLenFlag) { + FIXP_DBL rem = fMult(invFrameSizeDiv8, + (FIXP_DBL)(topMdct << (DFRACT_BITS - 12))); + if ((LONG)rem & (LONG)0x1F) { + stopWinIdx -= 1; + stopCol = (int)winBorderToColMap[stopWinIdx]; + } + } + topQmf = (64); + } + + /* save previous factors */ + if (stopCol == numQmfSubSamples) { + int tmpBottom = bottomQmf; + + if ((int)winBorderToColMap[8] > startCol) { + tmpBottom = 0; /* band starts in previous short window */ + } + + for (bin = tmpBottom; bin < topQmf; bin++) { + hDrcData->prevFact_mag[bin] = fact_mag[band]; + } + } + + /* apply */ + if ((col >= startCol) && (col < stopCol)) { + if (col >= (int)winBorderToColMap[startWinIdx + 1]) { + bottomQmf = 0; /* band starts in previous short window */ + } + if (col < (int)winBorderToColMap[stopWinIdx - 1]) { + topQmf = (64); /* band ends in next short window */ + } + + drcFact_mag = fact_mag[band]; + + /* normalize scale factor */ + if (fact_exp < maxShift) { + drcFact_mag >>= maxShift - fact_exp; + } + + /* apply scaling */ + for (bin = bottomQmf; bin < topQmf; bin++) { + qmfRealSlot[bin] = fMult(qmfRealSlot[bin], drcFact_mag); + if (!useLP) { + qmfImagSlot[bin] = fMult(qmfImagSlot[bin], drcFact_mag); + } + } + } + } + + bottomMdct = topMdct; + } /* end of bands loop */ + + if (col == (numQmfSubSamples >> 1) - 1) { + hDrcData->prevFact_exp = fact_exp; + } +} + +/*! + \brief Apply DRC factors frame based. + + \hDrcData Handle to DRC channel data. + \qmfRealSlot Pointer to real valued QMF data of the whole frame. + \qmfImagSlot Pointer to the imaginary QMF data of the whole frame. + \numQmfSubSamples Total number of time slots for one frame. + \scaleFactor Pointer to the out scale factor of the frame. + + \return None. +*/ +void sbrDecoder_drcApply(HANDLE_SBR_DRC_CHANNEL hDrcData, + FIXP_DBL **QmfBufferReal, FIXP_DBL **QmfBufferImag, + int numQmfSubSamples, int *scaleFactor) { + int col; + int maxShift = 0; + + if (hDrcData == NULL) { + return; + } + if (hDrcData->enable == 0) { + return; /* Avoid changing the scaleFactor even though the processing is + disabled. */ + } + + /* get max scale factor */ + if (hDrcData->prevFact_exp > maxShift) { + maxShift = hDrcData->prevFact_exp; + } + if (hDrcData->currFact_exp > maxShift) { + maxShift = hDrcData->currFact_exp; + } + if (hDrcData->nextFact_exp > maxShift) { + maxShift = hDrcData->nextFact_exp; + } + + for (col = 0; col < numQmfSubSamples; col++) { + FIXP_DBL *qmfSlotReal = QmfBufferReal[col]; + FIXP_DBL *qmfSlotImag = (QmfBufferImag == NULL) ? NULL : QmfBufferImag[col]; + + sbrDecoder_drcApplySlot(hDrcData, qmfSlotReal, qmfSlotImag, col, + numQmfSubSamples, maxShift); + } + + *scaleFactor += maxShift; +} diff --git a/fdk-aac/libSBRdec/src/sbrdec_drc.h b/fdk-aac/libSBRdec/src/sbrdec_drc.h new file mode 100644 index 0000000..2eb0e20 --- /dev/null +++ b/fdk-aac/libSBRdec/src/sbrdec_drc.h @@ -0,0 +1,149 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): Christian Griebel + + Description: Dynamic range control (DRC) decoder tool for SBR + +*******************************************************************************/ + +#ifndef SBRDEC_DRC_H +#define SBRDEC_DRC_H + +#include "sbrdecoder.h" + +#define SBRDEC_MAX_DRC_CHANNELS (8) +#define SBRDEC_MAX_DRC_BANDS (16) + +typedef struct { + FIXP_DBL prevFact_mag[(64)]; + INT prevFact_exp; + + FIXP_DBL currFact_mag[SBRDEC_MAX_DRC_BANDS]; + FIXP_DBL nextFact_mag[SBRDEC_MAX_DRC_BANDS]; + INT currFact_exp; + INT nextFact_exp; + + UINT numBandsCurr; + UINT numBandsNext; + USHORT bandTopCurr[SBRDEC_MAX_DRC_BANDS]; + USHORT bandTopNext[SBRDEC_MAX_DRC_BANDS]; + + SHORT drcInterpolationSchemeCurr; + SHORT drcInterpolationSchemeNext; + + SHORT enable; + + UCHAR winSequenceCurr; + UCHAR winSequenceNext; + +} SBRDEC_DRC_CHANNEL; + +typedef SBRDEC_DRC_CHANNEL *HANDLE_SBR_DRC_CHANNEL; + +void sbrDecoder_drcInitChannel(HANDLE_SBR_DRC_CHANNEL hDrcData); + +void sbrDecoder_drcUpdateChannel(HANDLE_SBR_DRC_CHANNEL hDrcData); + +void sbrDecoder_drcApplySlot(HANDLE_SBR_DRC_CHANNEL hDrcData, + FIXP_DBL *qmfRealSlot, FIXP_DBL *qmfImagSlot, + int col, int numQmfSubSamples, int maxShift); + +void sbrDecoder_drcApply(HANDLE_SBR_DRC_CHANNEL hDrcData, + FIXP_DBL **QmfBufferReal, FIXP_DBL **QmfBufferImag, + int numQmfSubSamples, int *scaleFactor); + +#endif /* SBRDEC_DRC_H */ diff --git a/fdk-aac/libSBRdec/src/sbrdec_freq_sca.cpp b/fdk-aac/libSBRdec/src/sbrdec_freq_sca.cpp new file mode 100644 index 0000000..165f94b --- /dev/null +++ b/fdk-aac/libSBRdec/src/sbrdec_freq_sca.cpp @@ -0,0 +1,835 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief Frequency scale calculation +*/ + +#include "sbrdec_freq_sca.h" + +#include "transcendent.h" +#include "sbr_rom.h" +#include "env_extr.h" + +#include "genericStds.h" /* need log() for debug-code only */ + +#define MAX_OCTAVE 29 +#define MAX_SECOND_REGION 50 + +static int numberOfBands(FIXP_SGL bpo_div16, int start, int stop, int warpFlag); +static void CalcBands(UCHAR *diff, UCHAR start, UCHAR stop, UCHAR num_bands); +static SBR_ERROR modifyBands(UCHAR max_band, UCHAR *diff, UCHAR length); +static void cumSum(UCHAR start_value, UCHAR *diff, UCHAR length, + UCHAR *start_adress); + +/*! + \brief Retrieve QMF-band where the SBR range starts + + Convert startFreq which was read from the bitstream into a + QMF-channel number. + + \return Number of start band +*/ +static UCHAR getStartBand( + UINT fs, /*!< Output sampling frequency */ + UCHAR startFreq, /*!< Index to table of possible start bands */ + UINT headerDataFlags) /*!< Info to SBR mode */ +{ + INT band; + UINT fsMapped = fs; + SBR_RATE rate = DUAL; + + if (headerDataFlags & (SBRDEC_SYNTAX_USAC | SBRDEC_SYNTAX_RSVD50)) { + if (headerDataFlags & SBRDEC_QUAD_RATE) { + rate = QUAD; + } + fsMapped = sbrdec_mapToStdSampleRate(fs, 1); + } + + FDK_ASSERT(2 * (rate + 1) <= (4)); + + switch (fsMapped) { + case 192000: + band = FDK_sbrDecoder_sbr_start_freq_192[startFreq]; + break; + case 176400: + band = FDK_sbrDecoder_sbr_start_freq_176[startFreq]; + break; + case 128000: + band = FDK_sbrDecoder_sbr_start_freq_128[startFreq]; + break; + case 96000: + case 88200: + band = FDK_sbrDecoder_sbr_start_freq_88[rate][startFreq]; + break; + case 64000: + band = FDK_sbrDecoder_sbr_start_freq_64[rate][startFreq]; + break; + case 48000: + band = FDK_sbrDecoder_sbr_start_freq_48[rate][startFreq]; + break; + case 44100: + band = FDK_sbrDecoder_sbr_start_freq_44[rate][startFreq]; + break; + case 40000: + band = FDK_sbrDecoder_sbr_start_freq_40[rate][startFreq]; + break; + case 32000: + band = FDK_sbrDecoder_sbr_start_freq_32[rate][startFreq]; + break; + case 24000: + band = FDK_sbrDecoder_sbr_start_freq_24[rate][startFreq]; + break; + case 22050: + band = FDK_sbrDecoder_sbr_start_freq_22[rate][startFreq]; + break; + case 16000: + band = FDK_sbrDecoder_sbr_start_freq_16[rate][startFreq]; + break; + default: + band = 255; + } + + return band; +} + +/*! + \brief Retrieve QMF-band where the SBR range starts + + Convert startFreq which was read from the bitstream into a + QMF-channel number. + + \return Number of start band +*/ +static UCHAR getStopBand( + UINT fs, /*!< Output sampling frequency */ + UCHAR stopFreq, /*!< Index to table of possible start bands */ + UINT headerDataFlags, /*!< Info to SBR mode */ + UCHAR k0) /*!< Start freq index */ +{ + UCHAR k2; + + if (stopFreq < 14) { + INT stopMin; + INT num = 2 * (64); + UCHAR diff_tot[MAX_OCTAVE + MAX_SECOND_REGION]; + UCHAR *diff0 = diff_tot; + UCHAR *diff1 = diff_tot + MAX_OCTAVE; + + if (headerDataFlags & SBRDEC_QUAD_RATE) { + num >>= 1; + } + + if (fs < 32000) { + stopMin = (((2 * 6000 * num) / fs) + 1) >> 1; + } else { + if (fs < 64000) { + stopMin = (((2 * 8000 * num) / fs) + 1) >> 1; + } else { + stopMin = (((2 * 10000 * num) / fs) + 1) >> 1; + } + } + + /* + Choose a stop band between k1 and 64 depending on stopFreq (0..13), + based on a logarithmic scale. + The vectors diff0 and diff1 are used temporarily here. + */ + CalcBands(diff0, stopMin, 64, 13); + shellsort(diff0, 13); + cumSum(stopMin, diff0, 13, diff1); + k2 = diff1[stopFreq]; + } else if (stopFreq == 14) + k2 = 2 * k0; + else + k2 = 3 * k0; + + /* Limit to Nyquist */ + if (k2 > (64)) k2 = (64); + + /* Range checks */ + /* 1 <= difference <= 48; 1 <= fs <= 96000 */ + { + UCHAR max_freq_coeffs = (headerDataFlags & SBRDEC_QUAD_RATE) + ? MAX_FREQ_COEFFS_QUAD_RATE + : MAX_FREQ_COEFFS; + if (((k2 - k0) > max_freq_coeffs) || (k2 <= k0)) { + return 255; + } + } + + if (headerDataFlags & SBRDEC_QUAD_RATE) { + return k2; /* skip other checks: (k2 - k0) must be <= + MAX_FREQ_COEFFS_QUAD_RATE for all fs */ + } + if (headerDataFlags & (SBRDEC_SYNTAX_USAC | SBRDEC_SYNTAX_RSVD50)) { + /* 1 <= difference <= 35; 42000 <= fs <= 96000 */ + if ((fs >= 42000) && ((k2 - k0) > MAX_FREQ_COEFFS_FS44100)) { + return 255; + } + /* 1 <= difference <= 32; 46009 <= fs <= 96000 */ + if ((fs >= 46009) && ((k2 - k0) > MAX_FREQ_COEFFS_FS48000)) { + return 255; + } + } else { + /* 1 <= difference <= 35; fs == 44100 */ + if ((fs == 44100) && ((k2 - k0) > MAX_FREQ_COEFFS_FS44100)) { + return 255; + } + /* 1 <= difference <= 32; 48000 <= fs <= 96000 */ + if ((fs >= 48000) && ((k2 - k0) > MAX_FREQ_COEFFS_FS48000)) { + return 255; + } + } + + return k2; +} + +/*! + \brief Generates master frequency tables + + Frequency tables are calculated according to the selected domain + (linear/logarithmic) and granularity. + IEC 14496-3 4.6.18.3.2.1 + + \return errorCode, 0 if successful +*/ +SBR_ERROR +sbrdecUpdateFreqScale( + UCHAR *v_k_master, /*!< Master table to be created */ + UCHAR *numMaster, /*!< Number of entries in master table */ + UINT fs, /*!< SBR working sampling rate */ + HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Control data from bitstream */ + UINT flags) { + FIXP_SGL bpo_div16; /* bands_per_octave divided by 16 */ + INT dk = 0; + + /* Internal variables */ + UCHAR k0, k2, i; + UCHAR num_bands0 = 0; + UCHAR num_bands1 = 0; + UCHAR diff_tot[MAX_OCTAVE + MAX_SECOND_REGION]; + UCHAR *diff0 = diff_tot; + UCHAR *diff1 = diff_tot + MAX_OCTAVE; + INT k2_achived; + INT k2_diff; + INT incr = 0; + + /* + Determine start band + */ + if (flags & SBRDEC_QUAD_RATE) { + fs >>= 1; + } + + k0 = getStartBand(fs, hHeaderData->bs_data.startFreq, flags); + if (k0 == 255) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + + /* + Determine stop band + */ + k2 = getStopBand(fs, hHeaderData->bs_data.stopFreq, flags, k0); + if (k2 == 255) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + + if (hHeaderData->bs_data.freqScale > 0) { /* Bark */ + INT k1; + + if (hHeaderData->bs_data.freqScale == 1) { + bpo_div16 = FL2FXCONST_SGL(12.0f / 16.0f); + } else if (hHeaderData->bs_data.freqScale == 2) { + bpo_div16 = FL2FXCONST_SGL(10.0f / 16.0f); + } else { + bpo_div16 = FL2FXCONST_SGL(8.0f / 16.0f); + } + + /* Ref: ISO/IEC 23003-3, Figure 12 - Flowchart calculation of fMaster for + * 4:1 system when bs_freq_scale > 0 */ + if (flags & SBRDEC_QUAD_RATE) { + if ((SHORT)k0 < (SHORT)(bpo_div16 >> ((FRACT_BITS - 1) - 4))) { + bpo_div16 = (FIXP_SGL)(k0 & (UCHAR)0xfe) + << ((FRACT_BITS - 1) - 4); /* bpo_div16 = floor(k0/2)*2 */ + } + } + + if (1000 * k2 > 2245 * k0) { /* Two or more regions */ + k1 = 2 * k0; + + num_bands0 = numberOfBands(bpo_div16, k0, k1, 0); + num_bands1 = + numberOfBands(bpo_div16, k1, k2, hHeaderData->bs_data.alterScale); + if (num_bands0 < 1) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + if (num_bands1 < 1) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + + CalcBands(diff0, k0, k1, num_bands0); + shellsort(diff0, num_bands0); + if (diff0[0] == 0) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + + cumSum(k0, diff0, num_bands0, v_k_master); + + CalcBands(diff1, k1, k2, num_bands1); + shellsort(diff1, num_bands1); + if (diff0[num_bands0 - 1] > diff1[0]) { + SBR_ERROR err; + + err = modifyBands(diff0[num_bands0 - 1], diff1, num_bands1); + if (err) return SBRDEC_UNSUPPORTED_CONFIG; + } + + /* Add 2nd region */ + cumSum(k1, diff1, num_bands1, &v_k_master[num_bands0]); + *numMaster = num_bands0 + num_bands1; /* Output nr of bands */ + + } else { /* Only one region */ + k1 = k2; + + num_bands0 = numberOfBands(bpo_div16, k0, k1, 0); + if (num_bands0 < 1) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + CalcBands(diff0, k0, k1, num_bands0); + shellsort(diff0, num_bands0); + if (diff0[0] == 0) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + + cumSum(k0, diff0, num_bands0, v_k_master); + *numMaster = num_bands0; /* Output nr of bands */ + } + } else { /* Linear mode */ + if (hHeaderData->bs_data.alterScale == 0) { + dk = 1; + /* FLOOR to get to few number of bands (next lower even number) */ + num_bands0 = (k2 - k0) & 254; + } else { + dk = 2; + num_bands0 = (((k2 - k0) >> 1) + 1) & 254; /* ROUND to the closest fit */ + } + + if (num_bands0 < 1) { + return SBRDEC_UNSUPPORTED_CONFIG; + /* We must return already here because 'i' can become negative below. */ + } + + k2_achived = k0 + num_bands0 * dk; + k2_diff = k2 - k2_achived; + + for (i = 0; i < num_bands0; i++) diff_tot[i] = dk; + + /* If linear scale wasn't achieved */ + /* and we got too wide SBR area */ + if (k2_diff < 0) { + incr = 1; + i = 0; + } + + /* If linear scale wasn't achieved */ + /* and we got too small SBR area */ + if (k2_diff > 0) { + incr = -1; + i = num_bands0 - 1; + } + + /* Adjust diff vector to get sepc. SBR range */ + while (k2_diff != 0) { + diff_tot[i] = diff_tot[i] - incr; + i = i + incr; + k2_diff = k2_diff + incr; + } + + cumSum(k0, diff_tot, num_bands0, v_k_master); /* cumsum */ + *numMaster = num_bands0; /* Output nr of bands */ + } + + if (*numMaster < 1) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + + /* Ref: ISO/IEC 23003-3 Cor.3, "In 7.5.5.2, add to the requirements:"*/ + if (flags & SBRDEC_QUAD_RATE) { + int k; + for (k = 1; k < *numMaster; k++) { + if (!(v_k_master[k] - v_k_master[k - 1] <= k0 - 2)) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + } + } + + /* + Print out the calculated table + */ + + return SBRDEC_OK; +} + +/*! + \brief Calculate frequency ratio of one SBR band + + All SBR bands should span a constant frequency range in the logarithmic + domain. This function calculates the ratio of any SBR band's upper and lower + frequency. + + \return num_band-th root of k_start/k_stop +*/ +static FIXP_SGL calcFactorPerBand(int k_start, int k_stop, int num_bands) { + /* Scaled bandfactor and step 1 bit right to avoid overflow + * use double data type */ + FIXP_DBL bandfactor = FL2FXCONST_DBL(0.25f); /* Start value */ + FIXP_DBL step = FL2FXCONST_DBL(0.125f); /* Initial increment for factor */ + + int direction = 1; + + /* Because saturation can't be done in INT IIS, + * changed start and stop data type from FIXP_SGL to FIXP_DBL */ + FIXP_DBL start = k_start << (DFRACT_BITS - 8); + FIXP_DBL stop = k_stop << (DFRACT_BITS - 8); + + FIXP_DBL temp; + + int j, i = 0; + + while (step > FL2FXCONST_DBL(0.0f)) { + i++; + temp = stop; + + /* Calculate temp^num_bands: */ + for (j = 0; j < num_bands; j++) + // temp = fMult(temp,bandfactor); + temp = fMultDiv2(temp, bandfactor) << 2; + + if (temp < start) { /* Factor too strong, make it weaker */ + if (direction == 0) + /* Halfen step. Right shift is not done as fract because otherwise the + lowest bit cannot be cleared due to rounding */ + step = (FIXP_DBL)((LONG)step >> 1); + direction = 1; + bandfactor = bandfactor + step; + } else { /* Factor is too weak: make it stronger */ + if (direction == 1) step = (FIXP_DBL)((LONG)step >> 1); + direction = 0; + bandfactor = bandfactor - step; + } + + if (i > 100) { + step = FL2FXCONST_DBL(0.0f); + } + } + return FX_DBL2FX_SGL(bandfactor << 1); +} + +/*! + \brief Calculate number of SBR bands between start and stop band + + Given the number of bands per octave, this function calculates how many + bands fit in the given frequency range. + When the warpFlag is set, the 'band density' is decreased by a factor + of 1/1.3 + + \return number of bands +*/ +static int numberOfBands( + FIXP_SGL bpo_div16, /*!< Input: number of bands per octave divided by 16 */ + int start, /*!< First QMF band of SBR frequency range */ + int stop, /*!< Last QMF band of SBR frequency range + 1 */ + int warpFlag) /*!< Stretching flag */ +{ + FIXP_SGL num_bands_div128; + int num_bands; + + num_bands_div128 = + FX_DBL2FX_SGL(fMult(FDK_getNumOctavesDiv8(start, stop), bpo_div16)); + + if (warpFlag) { + /* Apply the warp factor of 1.3 to get wider bands. We use a value + of 32768/25200 instead of the exact value to avoid critical cases + of rounding. + */ + num_bands_div128 = FX_DBL2FX_SGL( + fMult(num_bands_div128, FL2FXCONST_SGL(25200.0 / 32768.0))); + } + + /* add scaled 1 for rounding to even numbers: */ + num_bands_div128 = num_bands_div128 + FL2FXCONST_SGL(1.0f / 128.0f); + /* scale back to right aligned integer and double the value: */ + num_bands = 2 * ((LONG)num_bands_div128 >> (FRACT_BITS - 7)); + + return (num_bands); +} + +/*! + \brief Calculate width of SBR bands + + Given the desired number of bands within the SBR frequency range, + this function calculates the width of each SBR band in QMF channels. + The bands get wider from start to stop (bark scale). +*/ +static void CalcBands(UCHAR *diff, /*!< Vector of widths to be calculated */ + UCHAR start, /*!< Lower end of subband range */ + UCHAR stop, /*!< Upper end of subband range */ + UCHAR num_bands) /*!< Desired number of bands */ +{ + int i; + int previous; + int current; + FIXP_SGL exact, temp; + FIXP_SGL bandfactor = calcFactorPerBand(start, stop, num_bands); + + previous = stop; /* Start with highest QMF channel */ + exact = (FIXP_SGL)( + stop << (FRACT_BITS - 8)); /* Shift left to gain some accuracy */ + + for (i = num_bands - 1; i >= 0; i--) { + /* Calculate border of next lower sbr band */ + exact = FX_DBL2FX_SGL(fMult(exact, bandfactor)); + + /* Add scaled 0.5 for rounding: + We use a value 128/256 instead of 0.5 to avoid some critical cases of + rounding. */ + temp = exact + FL2FXCONST_SGL(128.0 / 32768.0); + + /* scale back to right alinged integer: */ + current = (LONG)temp >> (FRACT_BITS - 8); + + /* Save width of band i */ + diff[i] = previous - current; + previous = current; + } +} + +/*! + \brief Calculate cumulated sum vector from delta vector +*/ +static void cumSum(UCHAR start_value, UCHAR *diff, UCHAR length, + UCHAR *start_adress) { + int i; + start_adress[0] = start_value; + for (i = 1; i <= length; i++) + start_adress[i] = start_adress[i - 1] + diff[i - 1]; +} + +/*! + \brief Adapt width of frequency bands in the second region + + If SBR spans more than 2 octaves, the upper part of a bark-frequency-scale + is calculated separately. This function tries to avoid that the second region + starts with a band smaller than the highest band of the first region. +*/ +static SBR_ERROR modifyBands(UCHAR max_band_previous, UCHAR *diff, + UCHAR length) { + int change = max_band_previous - diff[0]; + + /* Limit the change so that the last band cannot get narrower than the first + * one */ + if (change > (diff[length - 1] - diff[0]) >> 1) + change = (diff[length - 1] - diff[0]) >> 1; + + diff[0] += change; + diff[length - 1] -= change; + shellsort(diff, length); + + return SBRDEC_OK; +} + +/*! + \brief Update high resolution frequency band table +*/ +static void sbrdecUpdateHiRes(UCHAR *h_hires, UCHAR *num_hires, + UCHAR *v_k_master, UCHAR num_bands, + UCHAR xover_band) { + UCHAR i; + + *num_hires = num_bands - xover_band; + + for (i = xover_band; i <= num_bands; i++) { + h_hires[i - xover_band] = v_k_master[i]; + } +} + +/*! + \brief Build low resolution table out of high resolution table +*/ +static void sbrdecUpdateLoRes(UCHAR *h_lores, UCHAR *num_lores, UCHAR *h_hires, + UCHAR num_hires) { + UCHAR i; + + if ((num_hires & 1) == 0) { + /* If even number of hires bands */ + *num_lores = num_hires >> 1; + /* Use every second lores=hires[0,2,4...] */ + for (i = 0; i <= *num_lores; i++) h_lores[i] = h_hires[i * 2]; + } else { + /* Odd number of hires, which means xover is odd */ + *num_lores = (num_hires + 1) >> 1; + /* Use lores=hires[0,1,3,5 ...] */ + h_lores[0] = h_hires[0]; + for (i = 1; i <= *num_lores; i++) { + h_lores[i] = h_hires[i * 2 - 1]; + } + } +} + +/*! + \brief Derive a low-resolution frequency-table from the master frequency + table +*/ +void sbrdecDownSampleLoRes(UCHAR *v_result, UCHAR num_result, + UCHAR *freqBandTableRef, UCHAR num_Ref) { + int step; + int i, j; + int org_length, result_length; + int v_index[MAX_FREQ_COEFFS >> 1]; + + /* init */ + org_length = num_Ref; + result_length = num_result; + + v_index[0] = 0; /* Always use left border */ + i = 0; + while (org_length > 0) { + /* Create downsample vector */ + i++; + step = org_length / result_length; + org_length = org_length - step; + result_length--; + v_index[i] = v_index[i - 1] + step; + } + + for (j = 0; j <= i; j++) { + /* Use downsample vector to index LoResolution vector */ + v_result[j] = freqBandTableRef[v_index[j]]; + } +} + +/*! + \brief Sorting routine +*/ +void shellsort(UCHAR *in, UCHAR n) { + int i, j, v, w; + int inc = 1; + + do + inc = 3 * inc + 1; + while (inc <= n); + + do { + inc = inc / 3; + for (i = inc; i < n; i++) { + v = in[i]; + j = i; + while ((w = in[j - inc]) > v) { + in[j] = w; + j -= inc; + if (j < inc) break; + } + in[j] = v; + } + } while (inc > 1); +} + +/*! + \brief Reset frequency band tables + \return errorCode, 0 if successful +*/ +SBR_ERROR +resetFreqBandTables(HANDLE_SBR_HEADER_DATA hHeaderData, const UINT flags) { + SBR_ERROR err = SBRDEC_OK; + int k2, kx, lsb, usb; + int intTemp; + UCHAR nBandsLo, nBandsHi; + HANDLE_FREQ_BAND_DATA hFreq = &hHeaderData->freqBandData; + + /* Calculate master frequency function */ + err = sbrdecUpdateFreqScale(hFreq->v_k_master, &hFreq->numMaster, + hHeaderData->sbrProcSmplRate, hHeaderData, flags); + + if (err || (hHeaderData->bs_info.xover_band > hFreq->numMaster)) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + + /* Derive Hiresolution from master frequency function */ + sbrdecUpdateHiRes(hFreq->freqBandTable[1], &nBandsHi, hFreq->v_k_master, + hFreq->numMaster, hHeaderData->bs_info.xover_band); + /* Derive Loresolution from Hiresolution */ + sbrdecUpdateLoRes(hFreq->freqBandTable[0], &nBandsLo, hFreq->freqBandTable[1], + nBandsHi); + + hFreq->nSfb[0] = nBandsLo; + hFreq->nSfb[1] = nBandsHi; + + /* Check index to freqBandTable[0] */ + if (!(nBandsLo > 0) || + (nBandsLo > (((hHeaderData->numberOfAnalysisBands == 16) + ? MAX_FREQ_COEFFS_QUAD_RATE + : MAX_FREQ_COEFFS_DUAL_RATE) >> + 1))) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + + lsb = hFreq->freqBandTable[0][0]; + usb = hFreq->freqBandTable[0][nBandsLo]; + + /* Check for start frequency border k_x: + - ISO/IEC 14496-3 4.6.18.3.6 Requirements + - ISO/IEC 23003-3 7.5.5.2 Modifications and additions to the MPEG-4 SBR + tool + */ + /* Note that lsb > as hHeaderData->numberOfAnalysisBands is a valid SBR config + * for 24 band QMF analysis. */ + if ((lsb > ((flags & SBRDEC_QUAD_RATE) ? 16 : (32))) || (lsb >= usb)) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + + /* Calculate number of noise bands */ + + k2 = hFreq->freqBandTable[1][nBandsHi]; + kx = hFreq->freqBandTable[1][0]; + + if (hHeaderData->bs_data.noise_bands == 0) { + hFreq->nNfb = 1; + } else /* Calculate no of noise bands 1,2 or 3 bands/octave */ + { + /* Fetch number of octaves divided by 32 */ + intTemp = (LONG)FDK_getNumOctavesDiv8(kx, k2) >> 2; + + /* Integer-Multiplication with number of bands: */ + intTemp = intTemp * hHeaderData->bs_data.noise_bands; + + /* Add scaled 0.5 for rounding: */ + intTemp = intTemp + (LONG)FL2FXCONST_SGL(0.5f / 32.0f); + + /* Convert to right-aligned integer: */ + intTemp = intTemp >> (FRACT_BITS - 1 /*sign*/ - 5 /* rescale */); + + if (intTemp == 0) intTemp = 1; + + hFreq->nNfb = intTemp; + } + + hFreq->nInvfBands = hFreq->nNfb; + + if (hFreq->nNfb > MAX_NOISE_COEFFS) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + + /* Get noise bands */ + sbrdecDownSampleLoRes(hFreq->freqBandTableNoise, hFreq->nNfb, + hFreq->freqBandTable[0], nBandsLo); + + /* save old highband; required for overlap in usac + when headerchange occurs at XVAR and VARX frame; */ + hFreq->ov_highSubband = hFreq->highSubband; + + hFreq->lowSubband = lsb; + hFreq->highSubband = usb; + + return SBRDEC_OK; +} diff --git a/fdk-aac/libSBRdec/src/sbrdec_freq_sca.h b/fdk-aac/libSBRdec/src/sbrdec_freq_sca.h new file mode 100644 index 0000000..7e6b8e8 --- /dev/null +++ b/fdk-aac/libSBRdec/src/sbrdec_freq_sca.h @@ -0,0 +1,127 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief Frequency scale prototypes +*/ +#ifndef SBRDEC_FREQ_SCA_H +#define SBRDEC_FREQ_SCA_H + +#include "sbrdecoder.h" +#include "env_extr.h" + +typedef enum { DUAL, QUAD } SBR_RATE; + +SBR_ERROR +sbrdecUpdateFreqScale(UCHAR *v_k_master, UCHAR *numMaster, UINT fs, + HANDLE_SBR_HEADER_DATA headerData, UINT flags); + +void sbrdecDownSampleLoRes(UCHAR *v_result, UCHAR num_result, + UCHAR *freqBandTableRef, UCHAR num_Ref); + +void shellsort(UCHAR *in, UCHAR n); + +SBR_ERROR +resetFreqBandTables(HANDLE_SBR_HEADER_DATA hHeaderData, const UINT flags); + +#endif diff --git a/fdk-aac/libSBRdec/src/sbrdecoder.cpp b/fdk-aac/libSBRdec/src/sbrdecoder.cpp new file mode 100644 index 0000000..4bc6f69 --- /dev/null +++ b/fdk-aac/libSBRdec/src/sbrdecoder.cpp @@ -0,0 +1,2023 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief SBR decoder frontend + This module provides a frontend to the SBR decoder. The function openSBR() is + called for initialization. The function sbrDecoder_Apply() is called for each + frame. sbr_Apply() will call the required functions to decode the raw SBR data + (provided by env_extr.cpp), to decode the envelope data and noise floor levels + [decodeSbrData()], and to finally apply SBR to the current frame [sbr_dec()]. + + \sa sbrDecoder_Apply(), \ref documentationOverview +*/ + +/*! + \page documentationOverview Overview of important information resources and + source code documentation + + As part of this documentation you can find more extensive descriptions about + key concepts and algorithms at the following locations: + + <h2>Programming</h2> + + \li Buffer management: sbrDecoder_Apply() and sbr_dec() + \li Internal scale factors to maximize SNR on fixed point processors: + #QMF_SCALE_FACTOR \li Special mantissa-exponent format: Created in + requantizeEnvelopeData() and used in calculateSbrEnvelope() + + <h2>Algorithmic details</h2> + \li About the SBR data format: \ref SBR_HEADER_ELEMENT and \ref + SBR_STANDARD_ELEMENT \li Details about the bitstream decoder: env_extr.cpp \li + Details about the QMF filterbank and the provided polyphase implementation: + qmf_dec.cpp \li Details about the transposer: lpp_tran.cpp \li Details about + the envelope adjuster: env_calc.cpp + +*/ + +#include "sbrdecoder.h" + +#include "FDK_bitstream.h" + +#include "sbrdec_freq_sca.h" +#include "env_extr.h" +#include "sbr_dec.h" +#include "env_dec.h" +#include "sbr_crc.h" +#include "sbr_ram.h" +#include "sbr_rom.h" +#include "lpp_tran.h" +#include "transcendent.h" + +#include "FDK_crc.h" + +#include "sbrdec_drc.h" + +#include "psbitdec.h" + +/* Decoder library info */ +#define SBRDECODER_LIB_VL0 3 +#define SBRDECODER_LIB_VL1 0 +#define SBRDECODER_LIB_VL2 0 +#define SBRDECODER_LIB_TITLE "SBR Decoder" +#ifdef __ANDROID__ +#define SBRDECODER_LIB_BUILD_DATE "" +#define SBRDECODER_LIB_BUILD_TIME "" +#else +#define SBRDECODER_LIB_BUILD_DATE __DATE__ +#define SBRDECODER_LIB_BUILD_TIME __TIME__ +#endif + +static void setFrameErrorFlag(SBR_DECODER_ELEMENT *pSbrElement, UCHAR value) { + if (pSbrElement != NULL) { + switch (value) { + case FRAME_ERROR_ALLSLOTS: + FDKmemset(pSbrElement->frameErrorFlag, FRAME_ERROR, + sizeof(pSbrElement->frameErrorFlag)); + break; + default: + pSbrElement->frameErrorFlag[pSbrElement->useFrameSlot] = value; + } + } +} + +static UCHAR getHeaderSlot(UCHAR currentSlot, UCHAR hdrSlotUsage[(1) + 1]) { + UINT occupied = 0; + int s; + UCHAR slot = hdrSlotUsage[currentSlot]; + + FDK_ASSERT((1) + 1 < 32); + + for (s = 0; s < (1) + 1; s++) { + if ((hdrSlotUsage[s] == slot) && (s != slot)) { + occupied = 1; + break; + } + } + + if (occupied) { + occupied = 0; + + for (s = 0; s < (1) + 1; s++) { + occupied |= 1 << hdrSlotUsage[s]; + } + for (s = 0; s < (1) + 1; s++) { + if (!(occupied & 0x1)) { + slot = s; + break; + } + occupied >>= 1; + } + } + + return slot; +} + +static void copySbrHeader(HANDLE_SBR_HEADER_DATA hDst, + const HANDLE_SBR_HEADER_DATA hSrc) { + /* copy the whole header memory (including pointers) */ + FDKmemcpy(hDst, hSrc, sizeof(SBR_HEADER_DATA)); + + /* update pointers */ + hDst->freqBandData.freqBandTable[0] = hDst->freqBandData.freqBandTableLo; + hDst->freqBandData.freqBandTable[1] = hDst->freqBandData.freqBandTableHi; +} + +static int compareSbrHeader(const HANDLE_SBR_HEADER_DATA hHdr1, + const HANDLE_SBR_HEADER_DATA hHdr2) { + int result = 0; + + /* compare basic data */ + result |= (hHdr1->syncState != hHdr2->syncState) ? 1 : 0; + result |= (hHdr1->status != hHdr2->status) ? 1 : 0; + result |= (hHdr1->frameErrorFlag != hHdr2->frameErrorFlag) ? 1 : 0; + result |= (hHdr1->numberTimeSlots != hHdr2->numberTimeSlots) ? 1 : 0; + result |= + (hHdr1->numberOfAnalysisBands != hHdr2->numberOfAnalysisBands) ? 1 : 0; + result |= (hHdr1->timeStep != hHdr2->timeStep) ? 1 : 0; + result |= (hHdr1->sbrProcSmplRate != hHdr2->sbrProcSmplRate) ? 1 : 0; + + /* compare bitstream data */ + result |= + FDKmemcmp(&hHdr1->bs_data, &hHdr2->bs_data, sizeof(SBR_HEADER_DATA_BS)); + result |= + FDKmemcmp(&hHdr1->bs_dflt, &hHdr2->bs_dflt, sizeof(SBR_HEADER_DATA_BS)); + result |= FDKmemcmp(&hHdr1->bs_info, &hHdr2->bs_info, + sizeof(SBR_HEADER_DATA_BS_INFO)); + + /* compare frequency band data */ + result |= FDKmemcmp(&hHdr1->freqBandData, &hHdr2->freqBandData, + (8 + MAX_NUM_LIMITERS + 1) * sizeof(UCHAR)); + result |= FDKmemcmp(hHdr1->freqBandData.freqBandTableLo, + hHdr2->freqBandData.freqBandTableLo, + (MAX_FREQ_COEFFS / 2 + 1) * sizeof(UCHAR)); + result |= FDKmemcmp(hHdr1->freqBandData.freqBandTableHi, + hHdr2->freqBandData.freqBandTableHi, + (MAX_FREQ_COEFFS + 1) * sizeof(UCHAR)); + result |= FDKmemcmp(hHdr1->freqBandData.freqBandTableNoise, + hHdr2->freqBandData.freqBandTableNoise, + (MAX_NOISE_COEFFS + 1) * sizeof(UCHAR)); + result |= + FDKmemcmp(hHdr1->freqBandData.v_k_master, hHdr2->freqBandData.v_k_master, + (MAX_FREQ_COEFFS + 1) * sizeof(UCHAR)); + + return result; +} + +/*! + \brief Reset SBR decoder. + + Reset should only be called if SBR has been sucessfully detected by + an appropriate checkForPayload() function. + + \return Error code. +*/ +static SBR_ERROR sbrDecoder_ResetElement(HANDLE_SBRDECODER self, + int sampleRateIn, int sampleRateOut, + int samplesPerFrame, + const MP4_ELEMENT_ID elementID, + const int elementIndex, + const int overlap) { + SBR_ERROR sbrError = SBRDEC_OK; + HANDLE_SBR_HEADER_DATA hSbrHeader; + UINT qmfFlags = 0; + + int i, synDownsampleFac; + + /* USAC: assuming theoretical case 8 kHz output sample rate with 4:1 SBR */ + const int sbr_min_sample_rate_in = IS_USAC(self->coreCodec) ? 2000 : 6400; + + /* Check in/out samplerates */ + if (sampleRateIn < sbr_min_sample_rate_in || sampleRateIn > (96000)) { + sbrError = SBRDEC_UNSUPPORTED_CONFIG; + goto bail; + } + + if (sampleRateOut > (96000)) { + sbrError = SBRDEC_UNSUPPORTED_CONFIG; + goto bail; + } + + /* Set QMF mode flags */ + if (self->flags & SBRDEC_LOW_POWER) qmfFlags |= QMF_FLAG_LP; + + if (self->coreCodec == AOT_ER_AAC_ELD) { + if (self->flags & SBRDEC_LD_MPS_QMF) { + qmfFlags |= QMF_FLAG_MPSLDFB; + } else { + qmfFlags |= QMF_FLAG_CLDFB; + } + } + + /* Set downsampling factor for synthesis filter bank */ + if (sampleRateOut == 0) { + /* no single rate mode */ + sampleRateOut = + sampleRateIn + << 1; /* In case of implicit signalling, assume dual rate SBR */ + } + + if (sampleRateIn == sampleRateOut) { + synDownsampleFac = 2; + self->flags |= SBRDEC_DOWNSAMPLE; + } else { + synDownsampleFac = 1; + self->flags &= ~SBRDEC_DOWNSAMPLE; + } + + self->synDownsampleFac = synDownsampleFac; + self->sampleRateOut = sampleRateOut; + + { + for (i = 0; i < (1) + 1; i++) { + int setDflt; + hSbrHeader = &(self->sbrHeader[elementIndex][i]); + setDflt = ((hSbrHeader->syncState == SBR_NOT_INITIALIZED) || + (self->flags & SBRDEC_FORCE_RESET)) + ? 1 + : 0; + + /* init a default header such that we can at least do upsampling later */ + sbrError = initHeaderData(hSbrHeader, sampleRateIn, sampleRateOut, + self->downscaleFactor, samplesPerFrame, + self->flags, setDflt); + + /* Set synchState to UPSAMPLING in case it already is initialized */ + hSbrHeader->syncState = hSbrHeader->syncState > UPSAMPLING + ? UPSAMPLING + : hSbrHeader->syncState; + } + } + + if (sbrError != SBRDEC_OK) { + goto bail; + } + + if (!self->pQmfDomain->globalConf.qmfDomainExplicitConfig) { + self->pQmfDomain->globalConf.flags_requested |= qmfFlags; + self->pQmfDomain->globalConf.nBandsAnalysis_requested = + self->sbrHeader[elementIndex][0].numberOfAnalysisBands; + self->pQmfDomain->globalConf.nBandsSynthesis_requested = + (synDownsampleFac == 1) ? 64 : 32; /* may be overwritten by MPS */ + self->pQmfDomain->globalConf.nBandsSynthesis_requested /= + self->downscaleFactor; + self->pQmfDomain->globalConf.nQmfTimeSlots_requested = + self->sbrHeader[elementIndex][0].numberTimeSlots * + self->sbrHeader[elementIndex][0].timeStep; + self->pQmfDomain->globalConf.nQmfOvTimeSlots_requested = overlap; + self->pQmfDomain->globalConf.nQmfProcBands_requested = 64; /* always 64 */ + self->pQmfDomain->globalConf.nQmfProcChannels_requested = + 1; /* may be overwritten by MPS */ + } + + /* Init SBR channels going to be assigned to a SBR element */ + { + int ch; + for (ch = 0; ch < self->pSbrElement[elementIndex]->nChannels; ch++) { + int headerIndex = + getHeaderSlot(self->pSbrElement[elementIndex]->useFrameSlot, + self->pSbrElement[elementIndex]->useHeaderSlot); + + /* and create sbrDec */ + sbrError = + createSbrDec(self->pSbrElement[elementIndex]->pSbrChannel[ch], + &self->sbrHeader[elementIndex][headerIndex], + &self->pSbrElement[elementIndex]->transposerSettings, + synDownsampleFac, qmfFlags, self->flags, overlap, ch, + self->codecFrameSize); + + if (sbrError != SBRDEC_OK) { + goto bail; + } + } + } + + // FDKmemclear(sbr_OverlapBuffer, sizeof(sbr_OverlapBuffer)); + + if (self->numSbrElements == 1) { + switch (self->coreCodec) { + case AOT_AAC_LC: + case AOT_SBR: + case AOT_PS: + case AOT_ER_AAC_SCAL: + case AOT_DRM_AAC: + case AOT_DRM_SURROUND: + if (CreatePsDec(&self->hParametricStereoDec, samplesPerFrame)) { + sbrError = SBRDEC_CREATE_ERROR; + goto bail; + } + break; + default: + break; + } + } + + /* Init frame delay slot handling */ + self->pSbrElement[elementIndex]->useFrameSlot = 0; + for (i = 0; i < ((1) + 1); i++) { + self->pSbrElement[elementIndex]->useHeaderSlot[i] = i; + } + +bail: + + return sbrError; +} + +/*! + \brief Assign QMF domain provided QMF channels to SBR channels. + + \return void +*/ +static void sbrDecoder_AssignQmfChannels2SbrChannels(HANDLE_SBRDECODER self) { + int ch, el, absCh_offset = 0; + for (el = 0; el < self->numSbrElements; el++) { + if (self->pSbrElement[el] != NULL) { + for (ch = 0; ch < self->pSbrElement[el]->nChannels; ch++) { + FDK_ASSERT(((absCh_offset + ch) < ((8) + (1))) && + ((absCh_offset + ch) < ((8) + (1)))); + self->pSbrElement[el]->pSbrChannel[ch]->SbrDec.qmfDomainInCh = + &self->pQmfDomain->QmfDomainIn[absCh_offset + ch]; + self->pSbrElement[el]->pSbrChannel[ch]->SbrDec.qmfDomainOutCh = + &self->pQmfDomain->QmfDomainOut[absCh_offset + ch]; + } + absCh_offset += self->pSbrElement[el]->nChannels; + } + } +} + +SBR_ERROR sbrDecoder_Open(HANDLE_SBRDECODER *pSelf, + HANDLE_FDK_QMF_DOMAIN pQmfDomain) { + HANDLE_SBRDECODER self = NULL; + SBR_ERROR sbrError = SBRDEC_OK; + int elIdx; + + if ((pSelf == NULL) || (pQmfDomain == NULL)) { + return SBRDEC_INVALID_ARGUMENT; + } + + /* Get memory for this instance */ + self = GetRam_SbrDecoder(); + if (self == NULL) { + sbrError = SBRDEC_MEM_ALLOC_FAILED; + goto bail; + } + + self->pQmfDomain = pQmfDomain; + + /* + Already zero because of calloc + self->numSbrElements = 0; + self->numSbrChannels = 0; + self->codecFrameSize = 0; + */ + + self->numDelayFrames = (1); /* set to the max value by default */ + + /* Initialize header sync state */ + for (elIdx = 0; elIdx < (8); elIdx += 1) { + int i; + for (i = 0; i < (1) + 1; i += 1) { + self->sbrHeader[elIdx][i].syncState = SBR_NOT_INITIALIZED; + } + } + + *pSelf = self; + +bail: + return sbrError; +} + +/** + * \brief determine if the given core codec AOT can be processed or not. + * \param coreCodec core codec audio object type. + * \return 1 if SBR can be processed, 0 if SBR cannot be processed/applied. + */ +static int sbrDecoder_isCoreCodecValid(AUDIO_OBJECT_TYPE coreCodec) { + switch (coreCodec) { + case AOT_AAC_LC: + case AOT_SBR: + case AOT_PS: + case AOT_ER_AAC_SCAL: + case AOT_ER_AAC_ELD: + case AOT_DRM_AAC: + case AOT_DRM_SURROUND: + case AOT_USAC: + return 1; + default: + return 0; + } +} + +static void sbrDecoder_DestroyElement(HANDLE_SBRDECODER self, + const int elementIndex) { + if (self->pSbrElement[elementIndex] != NULL) { + int ch; + + for (ch = 0; ch < SBRDEC_MAX_CH_PER_ELEMENT; ch++) { + if (self->pSbrElement[elementIndex]->pSbrChannel[ch] != NULL) { + deleteSbrDec(self->pSbrElement[elementIndex]->pSbrChannel[ch]); + FreeRam_SbrDecChannel( + &self->pSbrElement[elementIndex]->pSbrChannel[ch]); + self->numSbrChannels -= 1; + } + } + FreeRam_SbrDecElement(&self->pSbrElement[elementIndex]); + self->numSbrElements -= 1; + } +} + +SBR_ERROR sbrDecoder_InitElement( + HANDLE_SBRDECODER self, const int sampleRateIn, const int sampleRateOut, + const int samplesPerFrame, const AUDIO_OBJECT_TYPE coreCodec, + const MP4_ELEMENT_ID elementID, const int elementIndex, + const UCHAR harmonicSBR, const UCHAR stereoConfigIndex, + const UCHAR configMode, UCHAR *configChanged, const INT downscaleFactor) { + SBR_ERROR sbrError = SBRDEC_OK; + int chCnt = 0; + int nSbrElementsStart; + int nSbrChannelsStart; + if (self == NULL) { + return SBRDEC_INVALID_ARGUMENT; + } + + nSbrElementsStart = self->numSbrElements; + nSbrChannelsStart = self->numSbrChannels; + + /* Check core codec AOT */ + if (!sbrDecoder_isCoreCodecValid(coreCodec) || elementIndex >= (8)) { + sbrError = SBRDEC_UNSUPPORTED_CONFIG; + goto bail; + } + + if (elementID != ID_SCE && elementID != ID_CPE && elementID != ID_LFE) { + sbrError = SBRDEC_UNSUPPORTED_CONFIG; + goto bail; + } + + if (self->sampleRateIn == sampleRateIn && + self->codecFrameSize == samplesPerFrame && self->coreCodec == coreCodec && + self->pSbrElement[elementIndex] != NULL && + self->pSbrElement[elementIndex]->elementID == elementID && + !(self->flags & SBRDEC_FORCE_RESET) && + ((sampleRateOut == 0) ? 1 : (self->sampleRateOut == sampleRateOut)) && + ((harmonicSBR == 2) ? 1 + : (self->harmonicSBR == + harmonicSBR)) /* The value 2 signalizes that + harmonicSBR shall be ignored in + the config change detection */ + ) { + /* Nothing to do */ + return SBRDEC_OK; + } else { + if (configMode & AC_CM_DET_CFG_CHANGE) { + *configChanged = 1; + } + } + + /* reaching this point the SBR-decoder gets (re-)configured */ + + /* The flags field is used for all elements! */ + self->flags &= + (SBRDEC_FORCE_RESET | SBRDEC_FLUSH); /* Keep the global flags. They will + be reset after decoding. */ + self->flags |= (downscaleFactor > 1) ? SBRDEC_ELD_DOWNSCALE : 0; + self->flags |= (coreCodec == AOT_ER_AAC_ELD) ? SBRDEC_ELD_GRID : 0; + self->flags |= (coreCodec == AOT_ER_AAC_SCAL) ? SBRDEC_SYNTAX_SCAL : 0; + self->flags |= + (coreCodec == AOT_DRM_AAC) ? SBRDEC_SYNTAX_SCAL | SBRDEC_SYNTAX_DRM : 0; + self->flags |= (coreCodec == AOT_DRM_SURROUND) + ? SBRDEC_SYNTAX_SCAL | SBRDEC_SYNTAX_DRM + : 0; + self->flags |= (coreCodec == AOT_USAC) ? SBRDEC_SYNTAX_USAC : 0; + /* Robustness: Take integer division rounding into consideration. E.g. 22050 + * Hz with 4:1 SBR => 5512 Hz core sampling rate. */ + self->flags |= (sampleRateIn == sampleRateOut / 4) ? SBRDEC_QUAD_RATE : 0; + self->flags |= (harmonicSBR == 1) ? SBRDEC_USAC_HARMONICSBR : 0; + + if (configMode & AC_CM_DET_CFG_CHANGE) { + return SBRDEC_OK; + } + + self->sampleRateIn = sampleRateIn; + self->codecFrameSize = samplesPerFrame; + self->coreCodec = coreCodec; + self->harmonicSBR = harmonicSBR; + self->downscaleFactor = downscaleFactor; + + /* Init SBR elements */ + { + int elChannels, ch; + + if (self->pSbrElement[elementIndex] == NULL) { + self->pSbrElement[elementIndex] = GetRam_SbrDecElement(elementIndex); + if (self->pSbrElement[elementIndex] == NULL) { + sbrError = SBRDEC_MEM_ALLOC_FAILED; + goto bail; + } + self->numSbrElements++; + } else { + self->numSbrChannels -= self->pSbrElement[elementIndex]->nChannels; + } + + /* Save element ID for sanity checks and to have a fallback for concealment. + */ + self->pSbrElement[elementIndex]->elementID = elementID; + + /* Determine amount of channels for this element */ + switch (elementID) { + case ID_NONE: + case ID_CPE: + elChannels = 2; + break; + case ID_LFE: + case ID_SCE: + elChannels = 1; + break; + default: + elChannels = 0; + break; + } + + /* Handle case of Parametric Stereo */ + if (elementIndex == 0 && elementID == ID_SCE) { + switch (coreCodec) { + case AOT_AAC_LC: + case AOT_SBR: + case AOT_PS: + case AOT_ER_AAC_SCAL: + case AOT_DRM_AAC: + case AOT_DRM_SURROUND: + elChannels = 2; + break; + default: + break; + } + } + + /* Sanity check to avoid memory leaks */ + if (elChannels < self->pSbrElement[elementIndex]->nChannels) { + self->numSbrChannels += self->pSbrElement[elementIndex]->nChannels; + sbrError = SBRDEC_PARSE_ERROR; + goto bail; + } + + self->pSbrElement[elementIndex]->nChannels = elChannels; + + for (ch = 0; ch < elChannels; ch++) { + if (self->pSbrElement[elementIndex]->pSbrChannel[ch] == NULL) { + self->pSbrElement[elementIndex]->pSbrChannel[ch] = + GetRam_SbrDecChannel(chCnt); + if (self->pSbrElement[elementIndex]->pSbrChannel[ch] == NULL) { + sbrError = SBRDEC_MEM_ALLOC_FAILED; + goto bail; + } + } + self->numSbrChannels++; + + sbrDecoder_drcInitChannel(&self->pSbrElement[elementIndex] + ->pSbrChannel[ch] + ->SbrDec.sbrDrcChannel); + + chCnt++; + } + } + + if (!self->pQmfDomain->globalConf.qmfDomainExplicitConfig) { + self->pQmfDomain->globalConf.nInputChannels_requested = + self->numSbrChannels; + self->pQmfDomain->globalConf.nOutputChannels_requested = + fMax((INT)self->numSbrChannels, + (INT)self->pQmfDomain->globalConf.nOutputChannels_requested); + } + + /* Make sure each SBR channel has one QMF channel assigned even if + * numSbrChannels or element set-up has changed. */ + sbrDecoder_AssignQmfChannels2SbrChannels(self); + + /* clear error flags for all delay slots */ + FDKmemclear(self->pSbrElement[elementIndex]->frameErrorFlag, + ((1) + 1) * sizeof(UCHAR)); + + { + int overlap; + + if (coreCodec == AOT_ER_AAC_ELD) { + overlap = 0; + } else if (self->flags & SBRDEC_QUAD_RATE) { + overlap = (3 * 4); + } else { + overlap = (3 * 2); + } + /* Initialize this instance */ + sbrError = sbrDecoder_ResetElement(self, sampleRateIn, sampleRateOut, + samplesPerFrame, elementID, elementIndex, + overlap); + } + +bail: + if (sbrError != SBRDEC_OK) { + if ((nSbrElementsStart < self->numSbrElements) || + (nSbrChannelsStart < self->numSbrChannels)) { + /* Free the memory allocated for this element */ + sbrDecoder_DestroyElement(self, elementIndex); + } else if ((elementIndex < (8)) && + (self->pSbrElement[elementIndex] != + NULL)) { /* Set error flag to trigger concealment */ + setFrameErrorFlag(self->pSbrElement[elementIndex], FRAME_ERROR); + } + } + + return sbrError; +} + +/** + * \brief Free config dependent SBR memory. + * \param self SBR decoder instance handle + */ +SBR_ERROR sbrDecoder_FreeMem(HANDLE_SBRDECODER *self) { + int i; + int elIdx; + + if (self != NULL && *self != NULL) { + for (i = 0; i < (8); i++) { + sbrDecoder_DestroyElement(*self, i); + } + + for (elIdx = 0; elIdx < (8); elIdx += 1) { + for (i = 0; i < (1) + 1; i += 1) { + (*self)->sbrHeader[elIdx][i].syncState = SBR_NOT_INITIALIZED; + } + } + } + + return SBRDEC_OK; +} + +/** + * \brief Apply decoded SBR header for one element. + * \param self SBR decoder instance handle + * \param hSbrHeader SBR header handle to be processed. + * \param hSbrChannel pointer array to the SBR element channels corresponding to + * the SBR header. + * \param headerStatus header status value returned from SBR header parser. + * \param numElementChannels amount of channels for the SBR element whos header + * is to be processed. + */ +static SBR_ERROR sbrDecoder_HeaderUpdate(HANDLE_SBRDECODER self, + HANDLE_SBR_HEADER_DATA hSbrHeader, + SBR_HEADER_STATUS headerStatus, + HANDLE_SBR_CHANNEL hSbrChannel[], + const int numElementChannels) { + SBR_ERROR errorStatus = SBRDEC_OK; + + /* + change of control data, reset decoder + */ + errorStatus = resetFreqBandTables(hSbrHeader, self->flags); + + if (errorStatus == SBRDEC_OK) { + if (hSbrHeader->syncState == UPSAMPLING && headerStatus != HEADER_RESET) { +#if (SBRDEC_MAX_HB_FADE_FRAMES > 0) + int ch; + for (ch = 0; ch < numElementChannels; ch += 1) { + hSbrChannel[ch]->SbrDec.highBandFadeCnt = SBRDEC_MAX_HB_FADE_FRAMES; + } + +#endif + /* As the default header would limit the frequency range, + lowSubband and highSubband must be patched. */ + hSbrHeader->freqBandData.lowSubband = hSbrHeader->numberOfAnalysisBands; + hSbrHeader->freqBandData.highSubband = hSbrHeader->numberOfAnalysisBands; + } + + /* Trigger a reset before processing this slot */ + hSbrHeader->status |= SBRDEC_HDR_STAT_RESET; + } + + return errorStatus; +} + +INT sbrDecoder_Header(HANDLE_SBRDECODER self, HANDLE_FDK_BITSTREAM hBs, + const INT sampleRateIn, const INT sampleRateOut, + const INT samplesPerFrame, + const AUDIO_OBJECT_TYPE coreCodec, + const MP4_ELEMENT_ID elementID, const INT elementIndex, + const UCHAR harmonicSBR, const UCHAR stereoConfigIndex, + const UCHAR configMode, UCHAR *configChanged, + const INT downscaleFactor) { + SBR_HEADER_STATUS headerStatus; + HANDLE_SBR_HEADER_DATA hSbrHeader; + SBR_ERROR sbrError = SBRDEC_OK; + int headerIndex; + UINT flagsSaved = + 0; /* flags should not be changed in AC_CM_DET_CFG_CHANGE - mode after + parsing */ + + if (self == NULL || elementIndex >= (8)) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + + if (!sbrDecoder_isCoreCodecValid(coreCodec)) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + + if (configMode & AC_CM_DET_CFG_CHANGE) { + flagsSaved = self->flags; /* store */ + } + + sbrError = sbrDecoder_InitElement( + self, sampleRateIn, sampleRateOut, samplesPerFrame, coreCodec, elementID, + elementIndex, harmonicSBR, stereoConfigIndex, configMode, configChanged, + downscaleFactor); + + if ((sbrError != SBRDEC_OK) || (elementID == ID_LFE)) { + goto bail; + } + + if (configMode & AC_CM_DET_CFG_CHANGE) { + hSbrHeader = NULL; + } else { + headerIndex = getHeaderSlot(self->pSbrElement[elementIndex]->useFrameSlot, + self->pSbrElement[elementIndex]->useHeaderSlot); + + hSbrHeader = &(self->sbrHeader[elementIndex][headerIndex]); + } + + headerStatus = sbrGetHeaderData(hSbrHeader, hBs, self->flags, 0, configMode); + + if (coreCodec == AOT_USAC) { + if (configMode & AC_CM_DET_CFG_CHANGE) { + self->flags = flagsSaved; /* restore */ + } + return sbrError; + } + + if (configMode & AC_CM_ALLOC_MEM) { + SBR_DECODER_ELEMENT *pSbrElement; + + pSbrElement = self->pSbrElement[elementIndex]; + + /* Sanity check */ + if (pSbrElement != NULL) { + if ((elementID == ID_CPE && pSbrElement->nChannels != 2) || + (elementID != ID_CPE && pSbrElement->nChannels != 1)) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + if (headerStatus == HEADER_RESET) { + sbrError = sbrDecoder_HeaderUpdate(self, hSbrHeader, headerStatus, + pSbrElement->pSbrChannel, + pSbrElement->nChannels); + + if (sbrError == SBRDEC_OK) { + hSbrHeader->syncState = SBR_HEADER; + hSbrHeader->status |= SBRDEC_HDR_STAT_UPDATE; + } + /* else { + Since we already have overwritten the old SBR header the only way out + is UPSAMPLING! This will be prepared in the next step. + } */ + } + } + } +bail: + if (configMode & AC_CM_DET_CFG_CHANGE) { + self->flags = flagsSaved; /* restore */ + } + return sbrError; +} + +SBR_ERROR sbrDecoder_SetParam(HANDLE_SBRDECODER self, const SBRDEC_PARAM param, + const INT value) { + SBR_ERROR errorStatus = SBRDEC_OK; + + /* configure the subsystems */ + switch (param) { + case SBR_SYSTEM_BITSTREAM_DELAY: + if (value < 0 || value > (1)) { + errorStatus = SBRDEC_SET_PARAM_FAIL; + break; + } + if (self == NULL) { + errorStatus = SBRDEC_NOT_INITIALIZED; + } else { + self->numDelayFrames = (UCHAR)value; + } + break; + case SBR_QMF_MODE: + if (self == NULL) { + errorStatus = SBRDEC_NOT_INITIALIZED; + } else { + if (value == 1) { + self->flags |= SBRDEC_LOW_POWER; + } else { + self->flags &= ~SBRDEC_LOW_POWER; + } + } + break; + case SBR_LD_QMF_TIME_ALIGN: + if (self == NULL) { + errorStatus = SBRDEC_NOT_INITIALIZED; + } else { + if (value == 1) { + self->flags |= SBRDEC_LD_MPS_QMF; + } else { + self->flags &= ~SBRDEC_LD_MPS_QMF; + } + } + break; + case SBR_FLUSH_DATA: + if (value != 0) { + if (self == NULL) { + errorStatus = SBRDEC_NOT_INITIALIZED; + } else { + self->flags |= SBRDEC_FLUSH; + } + } + break; + case SBR_CLEAR_HISTORY: + if (value != 0) { + if (self == NULL) { + errorStatus = SBRDEC_NOT_INITIALIZED; + } else { + self->flags |= SBRDEC_FORCE_RESET; + } + } + break; + case SBR_BS_INTERRUPTION: { + int elementIndex; + + if (self == NULL) { + errorStatus = SBRDEC_NOT_INITIALIZED; + break; + } + + /* Loop over SBR elements */ + for (elementIndex = 0; elementIndex < self->numSbrElements; + elementIndex++) { + if (self->pSbrElement[elementIndex] != NULL) { + HANDLE_SBR_HEADER_DATA hSbrHeader; + int headerIndex = + getHeaderSlot(self->pSbrElement[elementIndex]->useFrameSlot, + self->pSbrElement[elementIndex]->useHeaderSlot); + + hSbrHeader = &(self->sbrHeader[elementIndex][headerIndex]); + + /* Set sync state UPSAMPLING for the corresponding slot. + This switches off bitstream parsing until a new header arrives. */ + hSbrHeader->syncState = UPSAMPLING; + hSbrHeader->status |= SBRDEC_HDR_STAT_UPDATE; + } + } + } break; + + case SBR_SKIP_QMF: + if (self == NULL) { + errorStatus = SBRDEC_NOT_INITIALIZED; + } else { + if (value == 1) { + self->flags |= SBRDEC_SKIP_QMF_ANA; + } else { + self->flags &= ~SBRDEC_SKIP_QMF_ANA; + } + if (value == 2) { + self->flags |= SBRDEC_SKIP_QMF_SYN; + } else { + self->flags &= ~SBRDEC_SKIP_QMF_SYN; + } + } + break; + default: + errorStatus = SBRDEC_SET_PARAM_FAIL; + break; + } /* switch(param) */ + + return (errorStatus); +} + +static SBRDEC_DRC_CHANNEL *sbrDecoder_drcGetChannel( + const HANDLE_SBRDECODER self, const INT channel) { + SBRDEC_DRC_CHANNEL *pSbrDrcChannelData = NULL; + int elementIndex, elChanIdx = 0, numCh = 0; + + for (elementIndex = 0; (elementIndex < (8)) && (numCh <= channel); + elementIndex++) { + SBR_DECODER_ELEMENT *pSbrElement = self->pSbrElement[elementIndex]; + int c, elChannels; + + elChanIdx = 0; + if (pSbrElement == NULL) break; + + /* Determine amount of channels for this element */ + switch (pSbrElement->elementID) { + case ID_CPE: + elChannels = 2; + break; + case ID_LFE: + case ID_SCE: + elChannels = 1; + break; + case ID_NONE: + default: + elChannels = 0; + break; + } + + /* Limit with actual allocated element channels */ + elChannels = fMin(elChannels, pSbrElement->nChannels); + + for (c = 0; (c < elChannels) && (numCh <= channel); c++) { + if (pSbrElement->pSbrChannel[elChanIdx] != NULL) { + numCh++; + elChanIdx++; + } + } + } + elementIndex -= 1; + elChanIdx -= 1; + + if (elChanIdx < 0 || elementIndex < 0) { + return NULL; + } + + if (self->pSbrElement[elementIndex] != NULL) { + if (self->pSbrElement[elementIndex]->pSbrChannel[elChanIdx] != NULL) { + pSbrDrcChannelData = &self->pSbrElement[elementIndex] + ->pSbrChannel[elChanIdx] + ->SbrDec.sbrDrcChannel; + } + } + + return (pSbrDrcChannelData); +} + +SBR_ERROR sbrDecoder_drcFeedChannel(HANDLE_SBRDECODER self, INT ch, + UINT numBands, FIXP_DBL *pNextFact_mag, + INT nextFact_exp, + SHORT drcInterpolationScheme, + UCHAR winSequence, USHORT *pBandTop) { + SBRDEC_DRC_CHANNEL *pSbrDrcChannelData = NULL; + int band, isValidData = 0; + + if (self == NULL) { + return SBRDEC_NOT_INITIALIZED; + } + if (ch > (8) || pNextFact_mag == NULL) { + return SBRDEC_SET_PARAM_FAIL; + } + + /* Search for gain values different to 1.0f */ + for (band = 0; band < (int)numBands; band += 1) { + if (!((pNextFact_mag[band] == FL2FXCONST_DBL(0.5)) && + (nextFact_exp == 1)) && + !((pNextFact_mag[band] == (FIXP_DBL)MAXVAL_DBL) && + (nextFact_exp == 0))) { + isValidData = 1; + break; + } + } + + /* Find the right SBR channel */ + pSbrDrcChannelData = sbrDecoder_drcGetChannel(self, ch); + + if (pSbrDrcChannelData != NULL) { + if (pSbrDrcChannelData->enable || + isValidData) { /* Activate processing only with real and valid data */ + int i; + + pSbrDrcChannelData->enable = 1; + pSbrDrcChannelData->numBandsNext = numBands; + + pSbrDrcChannelData->winSequenceNext = winSequence; + pSbrDrcChannelData->drcInterpolationSchemeNext = drcInterpolationScheme; + pSbrDrcChannelData->nextFact_exp = nextFact_exp; + + for (i = 0; i < (int)numBands; i++) { + pSbrDrcChannelData->bandTopNext[i] = pBandTop[i]; + pSbrDrcChannelData->nextFact_mag[i] = pNextFact_mag[i]; + } + } + } + + return SBRDEC_OK; +} + +void sbrDecoder_drcDisable(HANDLE_SBRDECODER self, INT ch) { + SBRDEC_DRC_CHANNEL *pSbrDrcChannelData = NULL; + + if ((self == NULL) || (ch > (8)) || (self->numSbrElements == 0) || + (self->numSbrChannels == 0)) { + return; + } + + /* Find the right SBR channel */ + pSbrDrcChannelData = sbrDecoder_drcGetChannel(self, ch); + + if (pSbrDrcChannelData != NULL) { + sbrDecoder_drcInitChannel(pSbrDrcChannelData); + } +} + +SBR_ERROR sbrDecoder_Parse(HANDLE_SBRDECODER self, HANDLE_FDK_BITSTREAM hBs, + UCHAR *pDrmBsBuffer, USHORT drmBsBufferSize, + int *count, int bsPayLen, int crcFlag, + MP4_ELEMENT_ID prevElement, int elementIndex, + UINT acFlags, UINT acElFlags[]) { + SBR_DECODER_ELEMENT *hSbrElement = NULL; + HANDLE_SBR_HEADER_DATA hSbrHeader = NULL; + HANDLE_SBR_CHANNEL *pSbrChannel; + + SBR_FRAME_DATA *hFrameDataLeft = NULL; + SBR_FRAME_DATA *hFrameDataRight = NULL; + SBR_FRAME_DATA frameDataLeftCopy; + SBR_FRAME_DATA frameDataRightCopy; + + SBR_ERROR errorStatus = SBRDEC_OK; + SBR_HEADER_STATUS headerStatus = HEADER_NOT_PRESENT; + + INT startPos = FDKgetValidBits(hBs); + INT CRCLen = 0; + HANDLE_FDK_BITSTREAM hBsOriginal = hBs; + FDK_BITSTREAM bsBwd; + + FDK_CRCINFO crcInfo; + INT crcReg = 0; + USHORT drmSbrCrc = 0; + const int fGlobalIndependencyFlag = acFlags & AC_INDEP; + const int bs_pvc = acElFlags[elementIndex] & AC_EL_USAC_PVC; + const int bs_interTes = acElFlags[elementIndex] & AC_EL_USAC_ITES; + int stereo; + int fDoDecodeSbrData = 1; + + int lastSlot, lastHdrSlot = 0, thisHdrSlot = 0; + + if (*count <= 0) { + setFrameErrorFlag(self->pSbrElement[elementIndex], FRAME_ERROR); + return SBRDEC_OK; + } + + /* SBR sanity checks */ + if (self == NULL) { + errorStatus = SBRDEC_NOT_INITIALIZED; + goto bail; + } + + /* Reverse bits of DRM SBR payload */ + if ((self->flags & SBRDEC_SYNTAX_DRM) && *count > 0) { + int dataBytes, dataBits; + + FDK_ASSERT(drmBsBufferSize >= (512)); + dataBits = *count; + + if (dataBits > ((512) * 8)) { + /* do not flip more data than needed */ + dataBits = (512) * 8; + } + + dataBytes = (dataBits + 7) >> 3; + + int j; + + if ((j = (int)FDKgetValidBits(hBs)) != 8) { + FDKpushBiDirectional(hBs, (j - 8)); + } + + j = 0; + for (; dataBytes > 0; dataBytes--) { + int i; + UCHAR tmpByte; + UCHAR buffer = 0x00; + + tmpByte = (UCHAR)FDKreadBits(hBs, 8); + for (i = 0; i < 4; i++) { + int shift = 2 * i + 1; + buffer |= (tmpByte & (0x08 >> i)) << shift; + buffer |= (tmpByte & (0x10 << i)) >> shift; + } + pDrmBsBuffer[j++] = buffer; + FDKpushBack(hBs, 16); + } + + FDKinitBitStream(&bsBwd, pDrmBsBuffer, (512), dataBits, BS_READER); + + /* Use reversed data */ + hBs = &bsBwd; + bsPayLen = *count; + } + + /* Remember start position of SBR element */ + startPos = FDKgetValidBits(hBs); + + /* SBR sanity checks */ + if (self->pSbrElement[elementIndex] == NULL) { + errorStatus = SBRDEC_NOT_INITIALIZED; + goto bail; + } + hSbrElement = self->pSbrElement[elementIndex]; + + lastSlot = (hSbrElement->useFrameSlot > 0) ? hSbrElement->useFrameSlot - 1 + : self->numDelayFrames; + lastHdrSlot = hSbrElement->useHeaderSlot[lastSlot]; + thisHdrSlot = getHeaderSlot( + hSbrElement->useFrameSlot, + hSbrElement->useHeaderSlot); /* Get a free header slot not used by + frames not processed yet. */ + + /* Assign the free slot to store a new header if there is one. */ + hSbrHeader = &self->sbrHeader[elementIndex][thisHdrSlot]; + + pSbrChannel = hSbrElement->pSbrChannel; + stereo = (hSbrElement->elementID == ID_CPE) ? 1 : 0; + + hFrameDataLeft = &self->pSbrElement[elementIndex] + ->pSbrChannel[0] + ->frameData[hSbrElement->useFrameSlot]; + if (stereo) { + hFrameDataRight = &self->pSbrElement[elementIndex] + ->pSbrChannel[1] + ->frameData[hSbrElement->useFrameSlot]; + } + + /* store frameData; new parsed frameData possibly corrupted */ + FDKmemcpy(&frameDataLeftCopy, hFrameDataLeft, sizeof(SBR_FRAME_DATA)); + if (stereo) { + FDKmemcpy(&frameDataRightCopy, hFrameDataRight, sizeof(SBR_FRAME_DATA)); + } + + /* reset PS flag; will be set after PS was found */ + self->flags &= ~SBRDEC_PS_DECODED; + + if (hSbrHeader->status & SBRDEC_HDR_STAT_UPDATE) { + /* Got a new header from extern (e.g. from an ASC) */ + headerStatus = HEADER_OK; + hSbrHeader->status &= ~SBRDEC_HDR_STAT_UPDATE; + } else if (thisHdrSlot != lastHdrSlot) { + /* Copy the last header into this slot otherwise the + header compare will trigger more HEADER_RESETs than needed. */ + copySbrHeader(hSbrHeader, &self->sbrHeader[elementIndex][lastHdrSlot]); + } + + /* + Check if bit stream data is valid and matches the element context + */ + if (((prevElement != ID_SCE) && (prevElement != ID_CPE)) || + prevElement != hSbrElement->elementID) { + /* In case of LFE we also land here, since there is no LFE SBR element (do + * upsampling only) */ + fDoDecodeSbrData = 0; + } + + if (fDoDecodeSbrData) { + if ((INT)FDKgetValidBits(hBs) <= 0) { + fDoDecodeSbrData = 0; + } + } + + /* + SBR CRC-check + */ + if (fDoDecodeSbrData) { + if (crcFlag) { + switch (self->coreCodec) { + case AOT_ER_AAC_ELD: + FDKpushFor(hBs, 10); + /* check sbrcrc later: we don't know the payload length now */ + break; + case AOT_DRM_AAC: + case AOT_DRM_SURROUND: + drmSbrCrc = (USHORT)FDKreadBits(hBs, 8); + /* Setup CRC decoder */ + FDKcrcInit(&crcInfo, 0x001d, 0xFFFF, 8); + /* Start CRC region */ + crcReg = FDKcrcStartReg(&crcInfo, hBs, 0); + break; + default: + CRCLen = bsPayLen - 10; /* change: 0 => i */ + if (CRCLen < 0) { + fDoDecodeSbrData = 0; + } else { + fDoDecodeSbrData = SbrCrcCheck(hBs, CRCLen); + } + break; + } + } + } /* if (fDoDecodeSbrData) */ + + /* + Read in the header data and issue a reset if change occured + */ + if (fDoDecodeSbrData) { + int sbrHeaderPresent; + + if (self->flags & (SBRDEC_SYNTAX_RSVD50 | SBRDEC_SYNTAX_USAC)) { + SBR_HEADER_DATA_BS_INFO newSbrInfo; + int sbrInfoPresent; + + if (bs_interTes) { + self->flags |= SBRDEC_USAC_ITES; + } else { + self->flags &= ~SBRDEC_USAC_ITES; + } + + if (fGlobalIndependencyFlag) { + self->flags |= SBRDEC_USAC_INDEP; + sbrInfoPresent = 1; + sbrHeaderPresent = 1; + } else { + self->flags &= ~SBRDEC_USAC_INDEP; + sbrInfoPresent = FDKreadBit(hBs); + if (sbrInfoPresent) { + sbrHeaderPresent = FDKreadBit(hBs); + } else { + sbrHeaderPresent = 0; + } + } + + if (sbrInfoPresent) { + newSbrInfo.ampResolution = FDKreadBit(hBs); + newSbrInfo.xover_band = FDKreadBits(hBs, 4); + newSbrInfo.sbr_preprocessing = FDKreadBit(hBs); + if (bs_pvc) { + newSbrInfo.pvc_mode = FDKreadBits(hBs, 2); + /* bs_pvc_mode: 0 -> no PVC, 1 -> PVC mode 1, 2 -> PVC mode 2, 3 -> + * reserved */ + if (newSbrInfo.pvc_mode > 2) { + headerStatus = HEADER_ERROR; + } + if (stereo && newSbrInfo.pvc_mode > 0) { + /* bs_pvc is always transmitted but pvc_mode is set to zero in case + * of stereo SBR. The config might be wrong but we cannot tell for + * sure. */ + newSbrInfo.pvc_mode = 0; + } + } else { + newSbrInfo.pvc_mode = 0; + } + if (headerStatus != HEADER_ERROR) { + if (FDKmemcmp(&hSbrHeader->bs_info, &newSbrInfo, + sizeof(SBR_HEADER_DATA_BS_INFO))) { + /* in case of ampResolution and preprocessing change no full reset + * required */ + /* HEADER reset would trigger HBE transposer reset which breaks + * eSbr_3_Eaa.mp4 */ + if ((hSbrHeader->bs_info.pvc_mode != newSbrInfo.pvc_mode) || + (hSbrHeader->bs_info.xover_band != newSbrInfo.xover_band)) { + headerStatus = HEADER_RESET; + } else { + headerStatus = HEADER_OK; + } + + hSbrHeader->bs_info = newSbrInfo; + } else { + headerStatus = HEADER_OK; + } + } + } + if (headerStatus == HEADER_ERROR) { + /* Corrupt SBR info data, do not decode and switch to UPSAMPLING */ + hSbrHeader->syncState = UPSAMPLING; + fDoDecodeSbrData = 0; + sbrHeaderPresent = 0; + } + + if (sbrHeaderPresent && fDoDecodeSbrData) { + int useDfltHeader; + + useDfltHeader = FDKreadBit(hBs); + + if (useDfltHeader) { + sbrHeaderPresent = 0; + if (FDKmemcmp(&hSbrHeader->bs_data, &hSbrHeader->bs_dflt, + sizeof(SBR_HEADER_DATA_BS)) || + hSbrHeader->syncState != SBR_ACTIVE) { + hSbrHeader->bs_data = hSbrHeader->bs_dflt; + headerStatus = HEADER_RESET; + } + } + } + } else { + sbrHeaderPresent = FDKreadBit(hBs); + } + + if (sbrHeaderPresent) { + headerStatus = sbrGetHeaderData(hSbrHeader, hBs, self->flags, 1, 0); + } + + if (headerStatus == HEADER_RESET) { + errorStatus = sbrDecoder_HeaderUpdate( + self, hSbrHeader, headerStatus, pSbrChannel, hSbrElement->nChannels); + + if (errorStatus == SBRDEC_OK) { + hSbrHeader->syncState = SBR_HEADER; + } else { + hSbrHeader->syncState = SBR_NOT_INITIALIZED; + headerStatus = HEADER_ERROR; + } + } + + if (errorStatus != SBRDEC_OK) { + fDoDecodeSbrData = 0; + } + } /* if (fDoDecodeSbrData) */ + + /* + Print debugging output only if state has changed + */ + + /* read frame data */ + if ((hSbrHeader->syncState >= SBR_HEADER) && fDoDecodeSbrData) { + int sbrFrameOk; + /* read the SBR element data */ + if (!stereo && (self->hParametricStereoDec != NULL)) { + /* update slot index for PS bitstream parsing */ + self->hParametricStereoDec->bsLastSlot = + self->hParametricStereoDec->bsReadSlot; + self->hParametricStereoDec->bsReadSlot = hSbrElement->useFrameSlot; + } + sbrFrameOk = sbrGetChannelElement( + hSbrHeader, hFrameDataLeft, (stereo) ? hFrameDataRight : NULL, + &pSbrChannel[0]->prevFrameData, + pSbrChannel[0]->SbrDec.PvcStaticData.pvc_mode_last, hBs, + (stereo) ? NULL : self->hParametricStereoDec, self->flags, + self->pSbrElement[elementIndex]->transposerSettings.overlap); + + if (!sbrFrameOk) { + fDoDecodeSbrData = 0; + } else { + INT valBits; + + if (bsPayLen > 0) { + valBits = bsPayLen - ((INT)startPos - (INT)FDKgetValidBits(hBs)); + } else { + valBits = (INT)FDKgetValidBits(hBs); + } + + if (crcFlag) { + switch (self->coreCodec) { + case AOT_ER_AAC_ELD: { + /* late crc check for eld */ + INT payloadbits = + (INT)startPos - (INT)FDKgetValidBits(hBs) - startPos; + INT crcLen = payloadbits - 10; + FDKpushBack(hBs, payloadbits); + fDoDecodeSbrData = SbrCrcCheck(hBs, crcLen); + FDKpushFor(hBs, crcLen); + } break; + case AOT_DRM_AAC: + case AOT_DRM_SURROUND: + /* End CRC region */ + FDKcrcEndReg(&crcInfo, hBs, crcReg); + /* Check CRC */ + if ((FDKcrcGetCRC(&crcInfo) ^ 0xFF) != drmSbrCrc) { + fDoDecodeSbrData = 0; + if (headerStatus != HEADER_NOT_PRESENT) { + headerStatus = HEADER_ERROR; + hSbrHeader->syncState = SBR_NOT_INITIALIZED; + } + } + break; + default: + break; + } + } + + /* sanity check of remaining bits */ + if (valBits < 0) { + fDoDecodeSbrData = 0; + } else { + switch (self->coreCodec) { + case AOT_SBR: + case AOT_PS: + case AOT_AAC_LC: { + /* This sanity check is only meaningful with General Audio + * bitstreams */ + int alignBits = valBits & 0x7; + + if (valBits > alignBits) { + fDoDecodeSbrData = 0; + } + } break; + default: + /* No sanity check available */ + break; + } + } + } + } else { + /* The returned bit count will not be the actual payload size since we did + not parse the frame data. Return an error so that the caller can react + respectively. */ + errorStatus = SBRDEC_PARSE_ERROR; + } + + if (!fDoDecodeSbrData) { + /* Set error flag for this slot to trigger concealment */ + setFrameErrorFlag(self->pSbrElement[elementIndex], FRAME_ERROR); + /* restore old frameData for concealment */ + FDKmemcpy(hFrameDataLeft, &frameDataLeftCopy, sizeof(SBR_FRAME_DATA)); + if (stereo) { + FDKmemcpy(hFrameDataRight, &frameDataRightCopy, sizeof(SBR_FRAME_DATA)); + } + errorStatus = SBRDEC_PARSE_ERROR; + } else { + /* Everything seems to be ok so clear the error flag */ + setFrameErrorFlag(self->pSbrElement[elementIndex], FRAME_OK); + } + + if (!stereo) { + /* Turn coupling off explicitely to avoid access to absent right frame data + that might occur with corrupt bitstreams. */ + hFrameDataLeft->coupling = COUPLING_OFF; + } + +bail: + + if (self != NULL) { + if (self->flags & SBRDEC_SYNTAX_DRM) { + hBs = hBsOriginal; + } + + if (errorStatus != SBRDEC_NOT_INITIALIZED) { + int useOldHdr = + ((headerStatus == HEADER_NOT_PRESENT) || + (headerStatus == HEADER_ERROR) || + (headerStatus == HEADER_RESET && errorStatus == SBRDEC_PARSE_ERROR)) + ? 1 + : 0; + + if (!useOldHdr && (thisHdrSlot != lastHdrSlot) && (hSbrHeader != NULL)) { + useOldHdr |= + (compareSbrHeader(hSbrHeader, + &self->sbrHeader[elementIndex][lastHdrSlot]) == 0) + ? 1 + : 0; + } + + if (hSbrElement != NULL) { + if (useOldHdr != 0) { + /* Use the old header for this frame */ + hSbrElement->useHeaderSlot[hSbrElement->useFrameSlot] = lastHdrSlot; + } else { + /* Use the new header for this frame */ + hSbrElement->useHeaderSlot[hSbrElement->useFrameSlot] = thisHdrSlot; + } + + /* Move frame pointer to the next slot which is up to be decoded/applied + * next */ + hSbrElement->useFrameSlot = + (hSbrElement->useFrameSlot + 1) % (self->numDelayFrames + 1); + } + } + } + + *count -= startPos - (INT)FDKgetValidBits(hBs); + + return errorStatus; +} + +/** + * \brief Render one SBR element into time domain signal. + * \param self SBR decoder handle + * \param timeData pointer to output buffer + * \param channelMapping pointer to UCHAR array where next 2 channel offsets are + * stored. + * \param elementIndex enumerating index of the SBR element to render. + * \param numInChannels number of channels from core coder. + * \param numOutChannels pointer to a location to return number of output + * channels. + * \param psPossible flag indicating if PS is possible or not. + * \return SBRDEC_OK if successfull, else error code + */ +static SBR_ERROR sbrDecoder_DecodeElement( + HANDLE_SBRDECODER self, QDOM_PCM *input, INT_PCM *timeData, + const int timeDataSize, const FDK_channelMapDescr *const mapDescr, + const int mapIdx, int channelIndex, const int elementIndex, + const int numInChannels, int *numOutChannels, const int psPossible) { + SBR_DECODER_ELEMENT *hSbrElement = self->pSbrElement[elementIndex]; + HANDLE_SBR_CHANNEL *pSbrChannel = + self->pSbrElement[elementIndex]->pSbrChannel; + HANDLE_SBR_HEADER_DATA hSbrHeader = + &self->sbrHeader[elementIndex] + [hSbrElement->useHeaderSlot[hSbrElement->useFrameSlot]]; + HANDLE_PS_DEC h_ps_d = self->hParametricStereoDec; + + /* get memory for frame data from scratch */ + SBR_FRAME_DATA *hFrameDataLeft = NULL; + SBR_FRAME_DATA *hFrameDataRight = NULL; + + SBR_ERROR errorStatus = SBRDEC_OK; + + INT strideOut, offset0 = 255, offset0_block = 0, offset1 = 255, + offset1_block = 0; + INT codecFrameSize = self->codecFrameSize; + + int stereo = (hSbrElement->elementID == ID_CPE) ? 1 : 0; + int numElementChannels = + hSbrElement + ->nChannels; /* Number of channels of the current SBR element */ + + hFrameDataLeft = + &hSbrElement->pSbrChannel[0]->frameData[hSbrElement->useFrameSlot]; + if (stereo) { + hFrameDataRight = + &hSbrElement->pSbrChannel[1]->frameData[hSbrElement->useFrameSlot]; + } + + if (self->flags & SBRDEC_FLUSH) { + if (self->numFlushedFrames > self->numDelayFrames) { + int hdrIdx; + /* No valid SBR payload available, hence switch to upsampling (in all + * headers) */ + for (hdrIdx = 0; hdrIdx < ((1) + 1); hdrIdx += 1) { + self->sbrHeader[elementIndex][hdrIdx].syncState = UPSAMPLING; + } + } else { + /* Move frame pointer to the next slot which is up to be decoded/applied + * next */ + hSbrElement->useFrameSlot = + (hSbrElement->useFrameSlot + 1) % (self->numDelayFrames + 1); + /* Update header and frame data pointer because they have already been set + */ + hSbrHeader = + &self->sbrHeader[elementIndex] + [hSbrElement + ->useHeaderSlot[hSbrElement->useFrameSlot]]; + hFrameDataLeft = + &hSbrElement->pSbrChannel[0]->frameData[hSbrElement->useFrameSlot]; + if (stereo) { + hFrameDataRight = + &hSbrElement->pSbrChannel[1]->frameData[hSbrElement->useFrameSlot]; + } + } + } + + /* Update the header error flag */ + hSbrHeader->frameErrorFlag = + hSbrElement->frameErrorFlag[hSbrElement->useFrameSlot]; + + /* + Prepare filterbank for upsampling if no valid bit stream data is available. + */ + if (hSbrHeader->syncState == SBR_NOT_INITIALIZED) { + errorStatus = + initHeaderData(hSbrHeader, self->sampleRateIn, self->sampleRateOut, + self->downscaleFactor, codecFrameSize, self->flags, + 1 /* SET_DEFAULT_HDR */ + ); + + if (errorStatus != SBRDEC_OK) { + return errorStatus; + } + + hSbrHeader->syncState = UPSAMPLING; + + errorStatus = sbrDecoder_HeaderUpdate(self, hSbrHeader, HEADER_NOT_PRESENT, + pSbrChannel, hSbrElement->nChannels); + + if (errorStatus != SBRDEC_OK) { + hSbrHeader->syncState = SBR_NOT_INITIALIZED; + return errorStatus; + } + } + + /* reset */ + if (hSbrHeader->status & SBRDEC_HDR_STAT_RESET) { + int ch; + int applySbrProc = (hSbrHeader->syncState == SBR_ACTIVE || + (hSbrHeader->frameErrorFlag == 0 && + hSbrHeader->syncState == SBR_HEADER)); + for (ch = 0; ch < numElementChannels; ch++) { + SBR_ERROR errorStatusTmp = SBRDEC_OK; + + errorStatusTmp = resetSbrDec( + &pSbrChannel[ch]->SbrDec, hSbrHeader, &pSbrChannel[ch]->prevFrameData, + self->synDownsampleFac, self->flags, pSbrChannel[ch]->frameData); + + if (errorStatusTmp != SBRDEC_OK) { + hSbrHeader->syncState = UPSAMPLING; + } + } + if (applySbrProc) { + hSbrHeader->status &= ~SBRDEC_HDR_STAT_RESET; + } + } + + /* decoding */ + if ((hSbrHeader->syncState == SBR_ACTIVE) || + ((hSbrHeader->syncState == SBR_HEADER) && + (hSbrHeader->frameErrorFlag == 0))) { + errorStatus = SBRDEC_OK; + + decodeSbrData(hSbrHeader, hFrameDataLeft, &pSbrChannel[0]->prevFrameData, + (stereo) ? hFrameDataRight : NULL, + (stereo) ? &pSbrChannel[1]->prevFrameData : NULL); + + /* Now we have a full parameter set and can do parameter + based concealment instead of plain upsampling. */ + hSbrHeader->syncState = SBR_ACTIVE; + } + + if (timeDataSize < + hSbrHeader->numberTimeSlots * hSbrHeader->timeStep * + self->pQmfDomain->globalConf.nBandsSynthesis * + (psPossible ? fMax(2, numInChannels) : numInChannels)) { + return SBRDEC_OUTPUT_BUFFER_TOO_SMALL; + } + + { + self->flags &= ~SBRDEC_PS_DECODED; + C_ALLOC_SCRATCH_START(pPsScratch, struct PS_DEC_COEFFICIENTS, 1) + + /* decode PS data if available */ + if (h_ps_d != NULL && psPossible && (hSbrHeader->syncState == SBR_ACTIVE)) { + int applyPs = 1; + + /* define which frame delay line slot to process */ + h_ps_d->processSlot = hSbrElement->useFrameSlot; + + applyPs = DecodePs(h_ps_d, hSbrHeader->frameErrorFlag, pPsScratch); + self->flags |= (applyPs) ? SBRDEC_PS_DECODED : 0; + } + + offset0 = FDK_chMapDescr_getMapValue(mapDescr, channelIndex, mapIdx); + offset0_block = offset0 * codecFrameSize; + if (stereo || psPossible) { + /* the value of offset1 only matters if the condition is true, however if + it is not true channelIndex+1 may exceed the channel map resutling in an + error, though the value of offset1 is actually meaningless. This is + prevented here. */ + offset1 = FDK_chMapDescr_getMapValue(mapDescr, channelIndex + 1, mapIdx); + offset1_block = offset1 * codecFrameSize; + } + /* Set strides for reading and writing */ + if (psPossible) + strideOut = (numInChannels < 2) ? 2 : numInChannels; + else + strideOut = numInChannels; + + /* use same buffers for left and right channel and apply PS per timeslot */ + /* Process left channel */ + sbr_dec(&pSbrChannel[0]->SbrDec, input + offset0_block, timeData + offset0, + (self->flags & SBRDEC_PS_DECODED) ? &pSbrChannel[1]->SbrDec : NULL, + timeData + offset1, strideOut, hSbrHeader, hFrameDataLeft, + &pSbrChannel[0]->prevFrameData, + (hSbrHeader->syncState == SBR_ACTIVE), h_ps_d, self->flags, + codecFrameSize); + + if (stereo) { + /* Process right channel */ + sbr_dec(&pSbrChannel[1]->SbrDec, input + offset1_block, + timeData + offset1, NULL, NULL, strideOut, hSbrHeader, + hFrameDataRight, &pSbrChannel[1]->prevFrameData, + (hSbrHeader->syncState == SBR_ACTIVE), NULL, self->flags, + codecFrameSize); + } + + C_ALLOC_SCRATCH_END(pPsScratch, struct PS_DEC_COEFFICIENTS, 1) + } + + if (h_ps_d != NULL) { + /* save PS status for next run */ + h_ps_d->psDecodedPrv = (self->flags & SBRDEC_PS_DECODED) ? 1 : 0; + } + + if (psPossible && !(self->flags & SBRDEC_SKIP_QMF_SYN)) { + FDK_ASSERT(strideOut > 1); + if (!(self->flags & SBRDEC_PS_DECODED)) { + /* A decoder which is able to decode PS has to produce a stereo output + * even if no PS data is available. */ + /* So copy left channel to right channel. */ + int copyFrameSize = + codecFrameSize * self->pQmfDomain->QmfDomainOut->fb.no_channels; + copyFrameSize /= self->pQmfDomain->QmfDomainIn->fb.no_channels; + INT_PCM *ptr; + INT i; + FDK_ASSERT(strideOut == 2); + + ptr = timeData; + for (i = copyFrameSize >> 1; i--;) { + INT_PCM tmp; /* This temporal variable is required because some + compilers can't do *ptr++ = *ptr++ correctly. */ + tmp = *ptr++; + *ptr++ = tmp; + tmp = *ptr++; + *ptr++ = tmp; + } + } + *numOutChannels = 2; /* Output minimum two channels when PS is enabled. */ + } + + return errorStatus; +} + +SBR_ERROR sbrDecoder_Apply(HANDLE_SBRDECODER self, INT_PCM *input, + INT_PCM *timeData, const int timeDataSize, + int *numChannels, int *sampleRate, + const FDK_channelMapDescr *const mapDescr, + const int mapIdx, const int coreDecodedOk, + UCHAR *psDecoded) { + SBR_ERROR errorStatus = SBRDEC_OK; + + int psPossible; + int sbrElementNum; + int numCoreChannels; + int numSbrChannels = 0; + + if ((self == NULL) || (timeData == NULL) || (numChannels == NULL) || + (sampleRate == NULL) || (psDecoded == NULL) || + !FDK_chMapDescr_isValid(mapDescr)) { + return SBRDEC_INVALID_ARGUMENT; + } + + psPossible = *psDecoded; + numCoreChannels = *numChannels; + if (numCoreChannels <= 0) { + return SBRDEC_INVALID_ARGUMENT; + } + + if (self->numSbrElements < 1) { + /* exit immediately to avoid access violations */ + return SBRDEC_NOT_INITIALIZED; + } + + /* Sanity check of allocated SBR elements. */ + for (sbrElementNum = 0; sbrElementNum < self->numSbrElements; + sbrElementNum++) { + if (self->pSbrElement[sbrElementNum] == NULL) { + return SBRDEC_NOT_INITIALIZED; + } + } + + if (self->numSbrElements != 1 || self->pSbrElement[0]->elementID != ID_SCE) { + psPossible = 0; + } + + /* Make sure that even if no SBR data was found/parsed *psDecoded is returned + * 1 if psPossible was 0. */ + if (psPossible == 0) { + self->flags &= ~SBRDEC_PS_DECODED; + } + + /* replaces channel based reset inside sbr_dec() */ + if (((self->flags & SBRDEC_LOW_POWER) ? 1 : 0) != + ((self->pQmfDomain->globalConf.flags & QMF_FLAG_LP) ? 1 : 0)) { + if (self->flags & SBRDEC_LOW_POWER) { + self->pQmfDomain->globalConf.flags |= QMF_FLAG_LP; + self->pQmfDomain->globalConf.flags_requested |= QMF_FLAG_LP; + } else { + self->pQmfDomain->globalConf.flags &= ~QMF_FLAG_LP; + self->pQmfDomain->globalConf.flags_requested &= ~QMF_FLAG_LP; + } + if (FDK_QmfDomain_InitFilterBank(self->pQmfDomain, QMF_FLAG_KEEP_STATES)) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + } + if (self->numSbrChannels > self->pQmfDomain->globalConf.nInputChannels) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + + if (self->flags & SBRDEC_FLUSH) { + /* flushing is signalized, hence increment the flush frame counter */ + self->numFlushedFrames++; + } else { + /* no flushing is signalized, hence reset the flush frame counter */ + self->numFlushedFrames = 0; + } + + /* Loop over SBR elements */ + for (sbrElementNum = 0; sbrElementNum < self->numSbrElements; + sbrElementNum++) { + int numElementChan; + + if (psPossible && + self->pSbrElement[sbrElementNum]->pSbrChannel[1] == NULL) { + /* Disable PS and try decoding SBR mono. */ + psPossible = 0; + } + + numElementChan = + (self->pSbrElement[sbrElementNum]->elementID == ID_CPE) ? 2 : 1; + + /* If core signal is bad then force upsampling */ + if (!coreDecodedOk) { + setFrameErrorFlag(self->pSbrElement[sbrElementNum], FRAME_ERROR_ALLSLOTS); + } + + errorStatus = sbrDecoder_DecodeElement( + self, input, timeData, timeDataSize, mapDescr, mapIdx, numSbrChannels, + sbrElementNum, + numCoreChannels, /* is correct even for USC SCI==2 case */ + &numElementChan, psPossible); + + if (errorStatus != SBRDEC_OK) { + goto bail; + } + + numSbrChannels += numElementChan; + + if (numSbrChannels >= numCoreChannels) { + break; + } + } + + /* Update numChannels and samplerate */ + /* Do not mess with output channels in case of USAC. numSbrChannels != + * numChannels for stereoConfigIndex == 2 */ + if (!(self->flags & SBRDEC_SYNTAX_USAC)) { + *numChannels = numSbrChannels; + } + *sampleRate = self->sampleRateOut; + *psDecoded = (self->flags & SBRDEC_PS_DECODED) ? 1 : 0; + + /* Clear reset and flush flag because everything seems to be done + * successfully. */ + self->flags &= ~SBRDEC_FORCE_RESET; + self->flags &= ~SBRDEC_FLUSH; + +bail: + + return errorStatus; +} + +SBR_ERROR sbrDecoder_Close(HANDLE_SBRDECODER *pSelf) { + HANDLE_SBRDECODER self = *pSelf; + int i; + + if (self != NULL) { + if (self->hParametricStereoDec != NULL) { + DeletePsDec(&self->hParametricStereoDec); + } + + for (i = 0; i < (8); i++) { + sbrDecoder_DestroyElement(self, i); + } + + FreeRam_SbrDecoder(pSelf); + } + + return SBRDEC_OK; +} + +INT sbrDecoder_GetLibInfo(LIB_INFO *info) { + int i; + + if (info == NULL) { + return -1; + } + + /* search for next free tab */ + for (i = 0; i < FDK_MODULE_LAST; i++) { + if (info[i].module_id == FDK_NONE) break; + } + if (i == FDK_MODULE_LAST) return -1; + info += i; + + info->module_id = FDK_SBRDEC; + info->version = + LIB_VERSION(SBRDECODER_LIB_VL0, SBRDECODER_LIB_VL1, SBRDECODER_LIB_VL2); + LIB_VERSION_STRING(info); + info->build_date = SBRDECODER_LIB_BUILD_DATE; + info->build_time = SBRDECODER_LIB_BUILD_TIME; + info->title = SBRDECODER_LIB_TITLE; + + /* Set flags */ + info->flags = 0 | CAPF_SBR_HQ | CAPF_SBR_LP | CAPF_SBR_PS_MPEG | + CAPF_SBR_DRM_BS | CAPF_SBR_CONCEALMENT | CAPF_SBR_DRC | + CAPF_SBR_ELD_DOWNSCALE | CAPF_SBR_HBEHQ; + /* End of flags */ + + return 0; +} + +UINT sbrDecoder_GetDelay(const HANDLE_SBRDECODER self) { + UINT outputDelay = 0; + + if (self != NULL) { + UINT flags = self->flags; + + /* See chapter 1.6.7.2 of ISO/IEC 14496-3 for the GA-SBR figures below. */ + + /* Are we initialized? */ + if ((self->numSbrChannels > 0) && (self->numSbrElements > 0)) { + /* Add QMF synthesis delay */ + if ((flags & SBRDEC_ELD_GRID) && IS_LOWDELAY(self->coreCodec)) { + /* Low delay SBR: */ + if (!(flags & SBRDEC_SKIP_QMF_SYN)) { + outputDelay += + (flags & SBRDEC_DOWNSAMPLE) ? 32 : 64; /* QMF synthesis */ + if (flags & SBRDEC_LD_MPS_QMF) { + outputDelay += 32; + } + } + } else if (!IS_USAC(self->coreCodec)) { + /* By the method of elimination this is the GA (AAC-LC, HE-AAC, ...) + * branch: */ + outputDelay += (flags & SBRDEC_DOWNSAMPLE) ? 481 : 962; + if (flags & SBRDEC_SKIP_QMF_SYN) { + outputDelay -= 257; /* QMF synthesis */ + } + } + } + } + + return (outputDelay); +} diff --git a/fdk-aac/libSBRdec/src/transcendent.h b/fdk-aac/libSBRdec/src/transcendent.h new file mode 100644 index 0000000..0e815c2 --- /dev/null +++ b/fdk-aac/libSBRdec/src/transcendent.h @@ -0,0 +1,372 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief FDK Fixed Point Arithmetic Library Interface +*/ + +#ifndef TRANSCENDENT_H +#define TRANSCENDENT_H + +#include "sbrdecoder.h" +#include "sbr_rom.h" + +/************************************************************************/ +/*! + \brief Get number of octaves between frequencies a and b + + The Result is scaled with 1/8. + The valid range for a and b is 1 to LOG_DUALIS_TABLE_SIZE. + + \return ld(a/b) / 8 +*/ +/************************************************************************/ +static inline FIXP_SGL FDK_getNumOctavesDiv8(INT a, /*!< lower band */ + INT b) /*!< upper band */ +{ + return ((SHORT)((LONG)(CalcLdInt(b) - CalcLdInt(a)) >> (FRACT_BITS - 3))); +} + +/************************************************************************/ +/*! + \brief Add two values given by mantissa and exponent. + + Mantissas are in fract format with values between 0 and 1. <br> + The base for exponents is 2. Example: \f$ a = a\_m * 2^{a\_e} \f$<br> +*/ +/************************************************************************/ +inline void FDK_add_MantExp(FIXP_SGL a_m, /*!< Mantissa of 1st operand a */ + SCHAR a_e, /*!< Exponent of 1st operand a */ + FIXP_SGL b_m, /*!< Mantissa of 2nd operand b */ + SCHAR b_e, /*!< Exponent of 2nd operand b */ + FIXP_SGL *ptrSum_m, /*!< Mantissa of result */ + SCHAR *ptrSum_e) /*!< Exponent of result */ +{ + FIXP_DBL accu; + int shift; + int shiftAbs; + + FIXP_DBL shiftedMantissa; + FIXP_DBL otherMantissa; + + /* Equalize exponents of the summands. + For the smaller summand, the exponent is adapted and + for compensation, the mantissa is shifted right. */ + + shift = (int)(a_e - b_e); + + shiftAbs = (shift > 0) ? shift : -shift; + shiftAbs = (shiftAbs < DFRACT_BITS - 1) ? shiftAbs : DFRACT_BITS - 1; + shiftedMantissa = (shift > 0) ? (FX_SGL2FX_DBL(b_m) >> shiftAbs) + : (FX_SGL2FX_DBL(a_m) >> shiftAbs); + otherMantissa = (shift > 0) ? FX_SGL2FX_DBL(a_m) : FX_SGL2FX_DBL(b_m); + *ptrSum_e = (shift > 0) ? a_e : b_e; + + accu = (shiftedMantissa >> 1) + (otherMantissa >> 1); + /* shift by 1 bit to avoid overflow */ + + if ((accu >= (FL2FXCONST_DBL(0.5f) - (FIXP_DBL)1)) || + (accu <= FL2FXCONST_DBL(-0.5f))) + *ptrSum_e += 1; + else + accu = (shiftedMantissa + otherMantissa); + + *ptrSum_m = FX_DBL2FX_SGL(accu); +} + +inline void FDK_add_MantExp(FIXP_DBL a, /*!< Mantissa of 1st operand a */ + SCHAR a_e, /*!< Exponent of 1st operand a */ + FIXP_DBL b, /*!< Mantissa of 2nd operand b */ + SCHAR b_e, /*!< Exponent of 2nd operand b */ + FIXP_DBL *ptrSum, /*!< Mantissa of result */ + SCHAR *ptrSum_e) /*!< Exponent of result */ +{ + FIXP_DBL accu; + int shift; + int shiftAbs; + + FIXP_DBL shiftedMantissa; + FIXP_DBL otherMantissa; + + /* Equalize exponents of the summands. + For the smaller summand, the exponent is adapted and + for compensation, the mantissa is shifted right. */ + + shift = (int)(a_e - b_e); + + shiftAbs = (shift > 0) ? shift : -shift; + shiftAbs = (shiftAbs < DFRACT_BITS - 1) ? shiftAbs : DFRACT_BITS - 1; + shiftedMantissa = (shift > 0) ? (b >> shiftAbs) : (a >> shiftAbs); + otherMantissa = (shift > 0) ? a : b; + *ptrSum_e = (shift > 0) ? a_e : b_e; + + accu = (shiftedMantissa >> 1) + (otherMantissa >> 1); + /* shift by 1 bit to avoid overflow */ + + if ((accu >= (FL2FXCONST_DBL(0.5f) - (FIXP_DBL)1)) || + (accu <= FL2FXCONST_DBL(-0.5f))) + *ptrSum_e += 1; + else + accu = (shiftedMantissa + otherMantissa); + + *ptrSum = accu; +} + +/************************************************************************/ +/*! + \brief Divide two values given by mantissa and exponent. + + Mantissas are in fract format with values between 0 and 1. <br> + The base for exponents is 2. Example: \f$ a = a\_m * 2^{a\_e} \f$<br> + + For performance reasons, the division is based on a table lookup + which limits accuracy. +*/ +/************************************************************************/ +static inline void FDK_divide_MantExp( + FIXP_SGL a_m, /*!< Mantissa of dividend a */ + SCHAR a_e, /*!< Exponent of dividend a */ + FIXP_SGL b_m, /*!< Mantissa of divisor b */ + SCHAR b_e, /*!< Exponent of divisor b */ + FIXP_SGL *ptrResult_m, /*!< Mantissa of quotient a/b */ + SCHAR *ptrResult_e) /*!< Exponent of quotient a/b */ + +{ + int preShift, postShift, index, shift; + FIXP_DBL ratio_m; + FIXP_SGL bInv_m = FL2FXCONST_SGL(0.0f); + + preShift = CntLeadingZeros(FX_SGL2FX_DBL(b_m)); + + /* + Shift b into the range from 0..INV_TABLE_SIZE-1, + + E.g. 10 bits must be skipped for INV_TABLE_BITS 8: + - leave 8 bits as index for table + - skip sign bit, + - skip first bit of mantissa, because this is always the same (>0.5) + + We are dealing with energies, so we need not care + about negative numbers + */ + + /* + The first interval has half width so the lowest bit of the index is + needed for a doubled resolution. + */ + shift = (FRACT_BITS - 2 - INV_TABLE_BITS - preShift); + + index = (shift < 0) ? (LONG)b_m << (-shift) : (LONG)b_m >> shift; + + /* The index has INV_TABLE_BITS +1 valid bits here. Clear the other bits. */ + index &= (1 << (INV_TABLE_BITS + 1)) - 1; + + /* Remove offset of half an interval */ + index--; + + /* Now the lowest bit is shifted out */ + index = index >> 1; + + /* Fetch inversed mantissa from table: */ + bInv_m = (index < 0) ? bInv_m : FDK_sbrDecoder_invTable[index]; + + /* Multiply a with the inverse of b: */ + ratio_m = (index < 0) ? FX_SGL2FX_DBL(a_m >> 1) : fMultDiv2(bInv_m, a_m); + + postShift = CntLeadingZeros(ratio_m) - 1; + + *ptrResult_m = FX_DBL2FX_SGL(ratio_m << postShift); + *ptrResult_e = a_e - b_e + 1 + preShift - postShift; +} + +static inline void FDK_divide_MantExp( + FIXP_DBL a_m, /*!< Mantissa of dividend a */ + SCHAR a_e, /*!< Exponent of dividend a */ + FIXP_DBL b_m, /*!< Mantissa of divisor b */ + SCHAR b_e, /*!< Exponent of divisor b */ + FIXP_DBL *ptrResult_m, /*!< Mantissa of quotient a/b */ + SCHAR *ptrResult_e) /*!< Exponent of quotient a/b */ + +{ + int preShift, postShift, index, shift; + FIXP_DBL ratio_m; + FIXP_SGL bInv_m = FL2FXCONST_SGL(0.0f); + + preShift = CntLeadingZeros(b_m); + + /* + Shift b into the range from 0..INV_TABLE_SIZE-1, + + E.g. 10 bits must be skipped for INV_TABLE_BITS 8: + - leave 8 bits as index for table + - skip sign bit, + - skip first bit of mantissa, because this is always the same (>0.5) + + We are dealing with energies, so we need not care + about negative numbers + */ + + /* + The first interval has half width so the lowest bit of the index is + needed for a doubled resolution. + */ + shift = (DFRACT_BITS - 2 - INV_TABLE_BITS - preShift); + + index = (shift < 0) ? (LONG)b_m << (-shift) : (LONG)b_m >> shift; + + /* The index has INV_TABLE_BITS +1 valid bits here. Clear the other bits. */ + index &= (1 << (INV_TABLE_BITS + 1)) - 1; + + /* Remove offset of half an interval */ + index--; + + /* Now the lowest bit is shifted out */ + index = index >> 1; + + /* Fetch inversed mantissa from table: */ + bInv_m = (index < 0) ? bInv_m : FDK_sbrDecoder_invTable[index]; + + /* Multiply a with the inverse of b: */ + ratio_m = (index < 0) ? (a_m >> 1) : fMultDiv2(bInv_m, a_m); + + postShift = CntLeadingZeros(ratio_m) - 1; + + *ptrResult_m = ratio_m << postShift; + *ptrResult_e = a_e - b_e + 1 + preShift - postShift; +} + +/*! + \brief Calculate the squareroot of a number given by mantissa and exponent + + Mantissa is in fract format with values between 0 and 1. <br> + The base for the exponent is 2. Example: \f$ a = a\_m * 2^{a\_e} \f$<br> + The operand is addressed via pointers and will be overwritten with the result. + + For performance reasons, the square root is based on a table lookup + which limits accuracy. +*/ +static inline void FDK_sqrt_MantExp( + FIXP_DBL *mantissa, /*!< Pointer to mantissa */ + SCHAR *exponent, const SCHAR *destScale) { + FIXP_DBL input_m = *mantissa; + int input_e = (int)*exponent; + FIXP_DBL result = FL2FXCONST_DBL(0.0f); + int result_e = -FRACT_BITS; + + /* Call lookup square root, which does internally normalization. */ + result = sqrtFixp_lookup(input_m, &input_e); + result_e = input_e; + + /* Write result */ + if (exponent == destScale) { + *mantissa = result; + *exponent = result_e; + } else { + int shift = result_e - *destScale; + *mantissa = (shift >= 0) ? result << (INT)fixMin(DFRACT_BITS - 1, shift) + : result >> (INT)fixMin(DFRACT_BITS - 1, -shift); + *exponent = *destScale; + } +} + +#endif |