diff options
| author | Fraunhofer IIS FDK <audio-fdk@iis.fraunhofer.de> | 2018-02-26 20:17:00 +0100 |
|---|---|---|
| committer | Jean-Michel Trivi <jmtrivi@google.com> | 2018-04-19 11:21:15 -0700 |
| commit | 6cfabd35363c3ef5e3b209b867169a500b3ccc3c (patch) | |
| tree | 01c0a19f2735e8b5d2407555fe992d4230d089eb /libSBRenc/src | |
| parent | 6288a1e34c4dede4c2806beb1736ece6580558c7 (diff) | |
| download | fdk-aac-6cfabd35363c3ef5e3b209b867169a500b3ccc3c.tar.gz fdk-aac-6cfabd35363c3ef5e3b209b867169a500b3ccc3c.tar.bz2 fdk-aac-6cfabd35363c3ef5e3b209b867169a500b3ccc3c.zip | |
Upgrade to FDKv2
Bug: 71430241
Test: CTS DecoderTest and DecoderTestAacDrc
original-Change-Id: Iaa20f749b8a04d553b20247cfe1a8930ebbabe30
Apply clang-format also on header files.
original-Change-Id: I14de1ef16bbc79ec0283e745f98356a10efeb2e4
Fixes for MPEG-D DRC
original-Change-Id: If1de2d74bbbac84b3f67de3b88b83f6a23b8a15c
Catch unsupported tw_mdct at an early stage
original-Change-Id: Ied9dd00d754162a0e3ca1ae3e6b854315d818afe
Fixing PVC transition frames
original-Change-Id: Ib75725abe39252806c32d71176308f2c03547a4e
Move qmf bands sanity check
original-Change-Id: Iab540c3013c174d9490d2ae100a4576f51d8dbc4
Initialize scaling variable
original-Change-Id: I3c4087101b70e998c71c1689b122b0d7762e0f9e
Add 16 qmf band configuration to getSlotNrgHQ()
original-Change-Id: I49a5d30f703a1b126ff163df9656db2540df21f1
Always apply byte alignment at the end of the AudioMuxElement
original-Change-Id: I42d560287506d65d4c3de8bfe3eb9a4ebeb4efc7
Setup SBR element only if no parse error exists
original-Change-Id: I1915b73704bc80ab882b9173d6bec59cbd073676
Additional array index check in HCR
original-Change-Id: I18cc6e501ea683b5009f1bbee26de8ddd04d8267
Fix fade-in index selection in concealment module
original-Change-Id: Ibf802ed6ed8c05e9257e1f3b6d0ac1162e9b81c1
Enable explicit backward compatible parser for AAC_LD
original-Change-Id: I27e9c678dcb5d40ed760a6d1e06609563d02482d
Skip spatial specific config in explicit backward compatible ASC
original-Change-Id: Iff7cc365561319e886090cedf30533f562ea4d6e
Update flags description in decoder API
original-Change-Id: I9a5b4f8da76bb652f5580cbd3ba9760425c43830
Add QMF domain reset function
original-Change-Id: I4f89a8a2c0277d18103380134e4ed86996e9d8d6
DRC upgrade v2.1.0
original-Change-Id: I5731c0540139dab220094cd978ef42099fc45b74
Fix integer overflow in sqrtFixp_lookup()
original-Change-Id: I429a6f0d19aa2cc957e0f181066f0ca73968c914
Fix integer overflow in invSqrtNorm2()
original-Change-Id: I84de5cbf9fb3adeb611db203fe492fabf4eb6155
Fix integer overflow in GenerateRandomVector()
original-Change-Id: I3118a641008bd9484d479e5b0b1ee2b5d7d44d74
Fix integer overflow in adjustTimeSlot_EldGrid()
original-Change-Id: I29d503c247c5c8282349b79df940416a512fb9d5
Fix integer overflow in FDKsbrEnc_codeEnvelope()
original-Change-Id: I6b34b61ebb9d525b0c651ed08de2befc1f801449
Follow-up on: Fix integer overflow in adjustTimeSlot_EldGrid()
original-Change-Id: I6f8f578cc7089e5eb7c7b93e580b72ca35ad689a
Fix integer overflow in get_pk_v2()
original-Change-Id: I63375bed40d45867f6eeaa72b20b1f33e815938c
Fix integer overflow in Syn_filt_zero()
original-Change-Id: Ie0c02fdfbe03988f9d3b20d10cd9fe4c002d1279
Fix integer overflow in CFac_CalcFacSignal()
original-Change-Id: Id2d767c40066c591b51768e978eb8af3b803f0c5
Fix integer overflow in FDKaacEnc_FDKaacEnc_calcPeNoAH()
original-Change-Id: Idcbd0f4a51ae2550ed106aa6f3d678d1f9724841
Fix integer overflow in sbrDecoder_calculateGainVec()
original-Change-Id: I7081bcbe29c5cede9821b38d93de07c7add2d507
Fix integer overflow in CLpc_SynthesisLattice()
original-Change-Id: I4a95ddc18de150102352d4a1845f06094764c881
Fix integer overflow in Pred_Lt4()
original-Change-Id: I4dbd012b2de7d07c3e70a47b92e3bfae8dbc750a
Fix integer overflow in FDKsbrEnc_InitSbrFastTransientDetector()
original-Change-Id: I788cbec1a4a00f44c2f3a72ad7a4afa219807d04
Fix unsigned integer overflow in FDKaacEnc_WriteBitstream()
original-Change-Id: I68fc75166e7d2cd5cd45b18dbe3d8c2a92f1822a
Fix unsigned integer overflow in FDK_MetadataEnc_Init()
original-Change-Id: Ie8d025f9bcdb2442c704bd196e61065c03c10af4
Fix overflow in pseudo random number generators
original-Change-Id: I3e2551ee01356297ca14e3788436ede80bd5513c
Fix unsigned integer overflow in sbrDecoder_Parse()
original-Change-Id: I3f231b2f437e9c37db4d5b964164686710eee971
Fix unsigned integer overflow in longsub()
original-Change-Id: I73c2bc50415cac26f1f5a29e125bbe75f9180a6e
Fix unsigned integer overflow in CAacDecoder_DecodeFrame()
original-Change-Id: Ifce2db4b1454b46fa5f887e9d383f1cc43b291e4
Fix overflow at CLpdChannelStream_Read()
original-Change-Id: Idb9d822ce3a4272e4794b643644f5434e2d4bf3f
Fix unsigned integer overflow in Hcr_State_BODY_SIGN_ESC__ESC_WORD()
original-Change-Id: I1ccf77c0015684b85534c5eb97162740a870b71c
Fix unsigned integer overflow in UsacConfig_Parse()
original-Change-Id: Ie6d27f84b6ae7eef092ecbff4447941c77864d9f
Fix unsigned integer overflow in aacDecoder_drcParse()
original-Change-Id: I713f28e883eea3d70b6fa56a7b8f8c22bcf66ca0
Fix unsigned integer overflow in aacDecoder_drcReadCompression()
original-Change-Id: Ia34dfeb88c4705c558bce34314f584965cafcf7a
Fix unsigned integer overflow in CDataStreamElement_Read()
original-Change-Id: Iae896cc1d11f0a893d21be6aa90bd3e60a2c25f0
Fix unsigned integer overflow in transportDec_AdjustEndOfAccessUnit()
original-Change-Id: I64cf29a153ee784bb4a16fdc088baabebc0007dc
Fix unsigned integer overflow in transportDec_GetAuBitsRemaining()
original-Change-Id: I975b3420faa9c16a041874ba0db82e92035962e4
Fix unsigned integer overflow in extractExtendedData()
original-Change-Id: I2a59eb09e2053cfb58dfb75fcecfad6b85a80a8f
Fix signed integer overflow in CAacDecoder_ExtPayloadParse()
original-Change-Id: I4ad5ca4e3b83b5d964f1c2f8c5e7b17c477c7929
Fix unsigned integer overflow in CAacDecoder_DecodeFrame()
original-Change-Id: I29a39df77d45c52a0c9c5c83c1ba81f8d0f25090
Follow-up on: Fix integer overflow in CLpc_SynthesisLattice()
original-Change-Id: I8fb194ffc073a3432a380845be71036a272d388f
Fix signed integer overflow in _interpolateDrcGain()
original-Change-Id: I879ec9ab14005069a7c47faf80e8bc6e03d22e60
Fix unsigned integer overflow in FDKreadBits()
original-Change-Id: I1f47a6a8037ff70375aa8844947d5681bb4287ad
Fix unsigned integer overflow in FDKbyteAlign()
original-Change-Id: Id5f3a11a0c9e50fc6f76ed6c572dbd4e9f2af766
Fix unsigned integer overflow in FDK_get32()
original-Change-Id: I9d33b8e97e3d38cbb80629cb859266ca0acdce96
Fix unsigned integer overflow in FDK_pushBack()
original-Change-Id: Ic87f899bc8c6acf7a377a8ca7f3ba74c3a1e1c19
Fix unsigned integer overflow in FDK_pushForward()
original-Change-Id: I3b754382f6776a34be1602e66694ede8e0b8effc
Fix unsigned integer overflow in ReadPsData()
original-Change-Id: I25361664ba8139e32bbbef2ca8c106a606ce9c37
Fix signed integer overflow in E_UTIL_residu()
original-Change-Id: I8c3abd1f437ee869caa8fb5903ce7d3d641b6aad
REVERT: Follow-up on: Integer overflow in CLpc_SynthesisLattice().
original-Change-Id: I3d340099acb0414795c8dfbe6362bc0a8f045f9b
Follow-up on: Fix integer overflow in CLpc_SynthesisLattice()
original-Change-Id: I4aedb8b3a187064e9f4d985175aa55bb99cc7590
Follow-up on: Fix unsigned integer overflow in aacDecoder_drcParse()
original-Change-Id: I2aa2e13916213bf52a67e8b0518e7bf7e57fb37d
Fix integer overflow in acelp
original-Change-Id: Ie6390c136d84055f8b728aefbe4ebef6e029dc77
Fix unsigned integer overflow in aacDecoder_UpdateBitStreamCounters()
original-Change-Id: I391ffd97ddb0b2c184cba76139bfb356a3b4d2e2
Adjust concealment default settings
original-Change-Id: I6a95db935a327c47df348030bcceafcb29f54b21
Saturate estimatedStartPos
original-Change-Id: I27be2085e0ae83ec9501409f65e003f6bcba1ab6
Negative shift exponent in _interpolateDrcGain()
original-Change-Id: I18edb26b26d002aafd5e633d4914960f7a359c29
Negative shift exponent in calculateICC()
original-Change-Id: I3dcd2ae98d2eb70ee0d59750863cbb2a6f4f8aba
Too large shift exponent in FDK_put()
original-Change-Id: Ib7d9aaa434d2d8de4a13b720ca0464b31ca9b671
Too large shift exponent in CalcInvLdData()
original-Change-Id: I43e6e78d4cd12daeb1dcd5d82d1798bdc2550262
Member access within null pointer of type SBR_CHANNEL
original-Change-Id: Idc5e4ea8997810376d2f36bbdf628923b135b097
Member access within null pointer of type CpePersistentData
original-Change-Id: Ib6c91cb0d37882768e5baf63324e429589de0d9d
Member access within null pointer FDKaacEnc_psyMain()
original-Change-Id: I7729b7f4479970531d9dc823abff63ca52e01997
Member access within null pointer FDKaacEnc_GetPnsParam()
original-Change-Id: I9aa3b9f3456ae2e0f7483dbd5b3dde95fc62da39
Member access within null pointer FDKsbrEnc_EnvEncodeFrame()
original-Change-Id: I67936f90ea714e90b3e81bc0dd1472cc713eb23a
Add HCR sanity check
original-Change-Id: I6c1d9732ebcf6af12f50b7641400752f74be39f7
Fix memory issue for HBE edge case with 8:3 SBR
original-Change-Id: I11ea58a61e69fbe8bf75034b640baee3011e63e9
Additional SBR parametrization sanity check for ELD
original-Change-Id: Ie26026fbfe174c2c7b3691f6218b5ce63e322140
Add MPEG-D DRC channel layout check
original-Change-Id: Iea70a74f171b227cce636a9eac4ba662777a2f72
Additional out-of-bounds checks in MPEG-D DRC
original-Change-Id: Ife4a8c3452c6fde8a0a09e941154a39a769777d4
Change-Id: Ic63cb2f628720f54fe9b572b0cb528e2599c624e
Diffstat (limited to 'libSBRenc/src')
45 files changed, 12617 insertions, 12426 deletions
diff --git a/libSBRenc/src/bit_sbr.cpp b/libSBRenc/src/bit_sbr.cpp index 9200e01..5a65e98 100644 --- a/libSBRenc/src/bit_sbr.cpp +++ b/libSBRenc/src/bit_sbr.cpp @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,14 +90,21 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ /*! \file - \brief SBR bit writing routines + \brief SBR bit writing routines $Revision: 93300 $ */ - #include "bit_sbr.h" #include "code_env.h" @@ -95,71 +113,54 @@ amm-info@iis.fraunhofer.de #include "ps_main.h" -typedef enum { - SBR_ID_SCE = 1, - SBR_ID_CPE -} SBR_ELEMENT_TYPE; - - -static INT encodeSbrData (HANDLE_SBR_ENV_DATA sbrEnvDataLeft, - HANDLE_SBR_ENV_DATA sbrEnvDataRight, - HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_COMMON_DATA cmonData, - SBR_ELEMENT_TYPE sbrElem, - INT coupling, - UINT sbrSyntaxFlags); - -static INT encodeSbrHeader (HANDLE_SBR_HEADER_DATA sbrHeaderData, - HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, - HANDLE_COMMON_DATA cmonData); - - -static INT encodeSbrHeaderData (HANDLE_SBR_HEADER_DATA sbrHeaderData, - HANDLE_FDK_BITSTREAM hBitStream); - -static INT encodeSbrSingleChannelElement (HANDLE_SBR_ENV_DATA sbrEnvData, - HANDLE_FDK_BITSTREAM hBitStream - ,HANDLE_PARAMETRIC_STEREO hParametricStereo - ,UINT sbrSyntaxFlags - ); - - +typedef enum { SBR_ID_SCE = 1, SBR_ID_CPE } SBR_ELEMENT_TYPE; -static INT encodeSbrChannelPairElement (HANDLE_SBR_ENV_DATA sbrEnvDataLeft, - HANDLE_SBR_ENV_DATA sbrEnvDataRight, - HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_FDK_BITSTREAM hBitStream, - INT coupling); +static INT encodeSbrData(HANDLE_SBR_ENV_DATA sbrEnvDataLeft, + HANDLE_SBR_ENV_DATA sbrEnvDataRight, + HANDLE_PARAMETRIC_STEREO hParametricStereo, + HANDLE_COMMON_DATA cmonData, SBR_ELEMENT_TYPE sbrElem, + INT coupling, UINT sbrSyntaxFlags); +static INT encodeSbrHeader(HANDLE_SBR_HEADER_DATA sbrHeaderData, + HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, + HANDLE_COMMON_DATA cmonData); -static INT encodeSbrGrid (HANDLE_SBR_ENV_DATA sbrEnvData, - HANDLE_FDK_BITSTREAM hBitStream); +static INT encodeSbrHeaderData(HANDLE_SBR_HEADER_DATA sbrHeaderData, + HANDLE_FDK_BITSTREAM hBitStream); -static int encodeLowDelaySbrGrid ( HANDLE_SBR_ENV_DATA sbrEnvData, - HANDLE_FDK_BITSTREAM hBitStream, - int transmitFreqs); +static INT encodeSbrSingleChannelElement( + HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream, + HANDLE_PARAMETRIC_STEREO hParametricStereo, const UINT sbrSyntaxFlags); -static INT encodeSbrDtdf (HANDLE_SBR_ENV_DATA sbrEnvData, - HANDLE_FDK_BITSTREAM hBitStream); +static INT encodeSbrChannelPairElement( + HANDLE_SBR_ENV_DATA sbrEnvDataLeft, HANDLE_SBR_ENV_DATA sbrEnvDataRight, + HANDLE_PARAMETRIC_STEREO hParametricStereo, HANDLE_FDK_BITSTREAM hBitStream, + const INT coupling, const UINT sbrSyntaxFlags); -static INT writeNoiseLevelData (HANDLE_SBR_ENV_DATA sbrEnvData, - HANDLE_FDK_BITSTREAM hBitStream, - INT coupling); +static INT encodeSbrGrid(HANDLE_SBR_ENV_DATA sbrEnvData, + HANDLE_FDK_BITSTREAM hBitStream); -static INT writeEnvelopeData (HANDLE_SBR_ENV_DATA sbrEnvData, - HANDLE_FDK_BITSTREAM hBitStream, - INT coupling); +static int encodeLowDelaySbrGrid(HANDLE_SBR_ENV_DATA sbrEnvData, + HANDLE_FDK_BITSTREAM hBitStream, + const int transmitFreqs, + const UINT sbrSyntaxFlags); -static INT writeSyntheticCodingData (HANDLE_SBR_ENV_DATA sbrEnvData, - HANDLE_FDK_BITSTREAM hBitStream); +static INT encodeSbrDtdf(HANDLE_SBR_ENV_DATA sbrEnvData, + HANDLE_FDK_BITSTREAM hBitStream); +static INT writeNoiseLevelData(HANDLE_SBR_ENV_DATA sbrEnvData, + HANDLE_FDK_BITSTREAM hBitStream, INT coupling); -static INT encodeExtendedData (HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_FDK_BITSTREAM hBitStream); +static INT writeEnvelopeData(HANDLE_SBR_ENV_DATA sbrEnvData, + HANDLE_FDK_BITSTREAM hBitStream, INT coupling); +static INT writeSyntheticCodingData(HANDLE_SBR_ENV_DATA sbrEnvData, + HANDLE_FDK_BITSTREAM hBitStream); +static INT encodeExtendedData(HANDLE_PARAMETRIC_STEREO hParametricStereo, + HANDLE_FDK_BITSTREAM hBitStream); -static INT getSbrExtendedDataSize (HANDLE_PARAMETRIC_STEREO hParametricStereo); +static INT getSbrExtendedDataSize(HANDLE_PARAMETRIC_STEREO hParametricStereo); /***************************************************************************** @@ -170,40 +171,26 @@ static INT getSbrExtendedDataSize (HANDLE_PARAMETRIC_STEREO hParametricStereo); output: *****************************************************************************/ -INT -FDKsbrEnc_WriteEnvSingleChannelElement( - HANDLE_SBR_HEADER_DATA sbrHeaderData, - HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, - HANDLE_SBR_ENV_DATA sbrEnvData, - HANDLE_COMMON_DATA cmonData, - UINT sbrSyntaxFlags - ) +INT FDKsbrEnc_WriteEnvSingleChannelElement( + HANDLE_SBR_HEADER_DATA sbrHeaderData, + HANDLE_PARAMETRIC_STEREO hParametricStereo, + HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, HANDLE_SBR_ENV_DATA sbrEnvData, + HANDLE_COMMON_DATA cmonData, UINT sbrSyntaxFlags) { INT payloadBits = 0; - cmonData->sbrHdrBits = 0; + cmonData->sbrHdrBits = 0; cmonData->sbrDataBits = 0; /* write pure sbr data */ if (sbrEnvData != NULL) { - /* write header */ - payloadBits += encodeSbrHeader (sbrHeaderData, - sbrBitstreamData, - cmonData); - + payloadBits += encodeSbrHeader(sbrHeaderData, sbrBitstreamData, cmonData); /* write data */ - payloadBits += encodeSbrData (sbrEnvData, - NULL, - hParametricStereo, - cmonData, - SBR_ID_SCE, - 0, - sbrSyntaxFlags); - + payloadBits += encodeSbrData(sbrEnvData, NULL, hParametricStereo, cmonData, + SBR_ID_SCE, 0, sbrSyntaxFlags); } return payloadBits; } @@ -217,83 +204,65 @@ FDKsbrEnc_WriteEnvSingleChannelElement( output: *****************************************************************************/ -INT -FDKsbrEnc_WriteEnvChannelPairElement (HANDLE_SBR_HEADER_DATA sbrHeaderData, - HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, - HANDLE_SBR_ENV_DATA sbrEnvDataLeft, - HANDLE_SBR_ENV_DATA sbrEnvDataRight, - HANDLE_COMMON_DATA cmonData, - UINT sbrSyntaxFlags) +INT FDKsbrEnc_WriteEnvChannelPairElement( + HANDLE_SBR_HEADER_DATA sbrHeaderData, + HANDLE_PARAMETRIC_STEREO hParametricStereo, + HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, + HANDLE_SBR_ENV_DATA sbrEnvDataLeft, HANDLE_SBR_ENV_DATA sbrEnvDataRight, + HANDLE_COMMON_DATA cmonData, UINT sbrSyntaxFlags) { INT payloadBits = 0; - cmonData->sbrHdrBits = 0; + cmonData->sbrHdrBits = 0; cmonData->sbrDataBits = 0; /* write pure sbr data */ if ((sbrEnvDataLeft != NULL) && (sbrEnvDataRight != NULL)) { - /* write header */ - payloadBits += encodeSbrHeader (sbrHeaderData, - sbrBitstreamData, - cmonData); + payloadBits += encodeSbrHeader(sbrHeaderData, sbrBitstreamData, cmonData); /* write data */ - payloadBits += encodeSbrData (sbrEnvDataLeft, - sbrEnvDataRight, - hParametricStereo, - cmonData, - SBR_ID_CPE, - sbrHeaderData->coupling, - sbrSyntaxFlags); - + payloadBits += encodeSbrData(sbrEnvDataLeft, sbrEnvDataRight, + hParametricStereo, cmonData, SBR_ID_CPE, + sbrHeaderData->coupling, sbrSyntaxFlags); } return payloadBits; } -INT -FDKsbrEnc_CountSbrChannelPairElement (HANDLE_SBR_HEADER_DATA sbrHeaderData, - HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, - HANDLE_SBR_ENV_DATA sbrEnvDataLeft, - HANDLE_SBR_ENV_DATA sbrEnvDataRight, - HANDLE_COMMON_DATA cmonData, - UINT sbrSyntaxFlags) -{ +INT FDKsbrEnc_CountSbrChannelPairElement( + HANDLE_SBR_HEADER_DATA sbrHeaderData, + HANDLE_PARAMETRIC_STEREO hParametricStereo, + HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, + HANDLE_SBR_ENV_DATA sbrEnvDataLeft, HANDLE_SBR_ENV_DATA sbrEnvDataRight, + HANDLE_COMMON_DATA cmonData, UINT sbrSyntaxFlags) { INT payloadBits; INT bitPos = FDKgetValidBits(&cmonData->sbrBitbuf); - payloadBits = FDKsbrEnc_WriteEnvChannelPairElement(sbrHeaderData, - hParametricStereo, - sbrBitstreamData, - sbrEnvDataLeft, - sbrEnvDataRight, - cmonData, - sbrSyntaxFlags); + payloadBits = FDKsbrEnc_WriteEnvChannelPairElement( + sbrHeaderData, hParametricStereo, sbrBitstreamData, sbrEnvDataLeft, + sbrEnvDataRight, cmonData, sbrSyntaxFlags); - FDKpushBack(&cmonData->sbrBitbuf, (FDKgetValidBits(&cmonData->sbrBitbuf) - bitPos) ); + FDKpushBack(&cmonData->sbrBitbuf, + (FDKgetValidBits(&cmonData->sbrBitbuf) - bitPos)); return payloadBits; } +void sbrEncoder_GetHeader(SBR_ENCODER *sbrEncoder, HANDLE_FDK_BITSTREAM hBs, + INT element_index, int fSendHeaders) { + encodeSbrHeaderData(&sbrEncoder->sbrElement[element_index]->sbrHeaderData, + hBs); -void sbrEncoder_GetHeader(SBR_ENCODER *sbrEncoder, - HANDLE_FDK_BITSTREAM hBs, - INT element_index, - int fSendHeaders) -{ - encodeSbrHeaderData (&sbrEncoder->sbrElement[element_index]->sbrHeaderData, hBs); - if (fSendHeaders == 0) { /* Prevent header being embedded into the SBR payload. */ - sbrEncoder->sbrElement[element_index]->sbrBitstreamData.NrSendHeaderData = -1; + sbrEncoder->sbrElement[element_index]->sbrBitstreamData.NrSendHeaderData = + -1; sbrEncoder->sbrElement[element_index]->sbrBitstreamData.HeaderActive = 0; - sbrEncoder->sbrElement[element_index]->sbrBitstreamData.CountSendHeaderData = -1; + sbrEncoder->sbrElement[element_index] + ->sbrBitstreamData.CountSendHeaderData = -1; } } - /***************************************************************************** functionname: encodeSbrHeader @@ -303,20 +272,16 @@ void sbrEncoder_GetHeader(SBR_ENCODER *sbrEncoder, output: *****************************************************************************/ -static INT -encodeSbrHeader (HANDLE_SBR_HEADER_DATA sbrHeaderData, - HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, - HANDLE_COMMON_DATA cmonData) -{ +static INT encodeSbrHeader(HANDLE_SBR_HEADER_DATA sbrHeaderData, + HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, + HANDLE_COMMON_DATA cmonData) { INT payloadBits = 0; if (sbrBitstreamData->HeaderActive) { - payloadBits += FDKwriteBits (&cmonData->sbrBitbuf, 1, 1); - payloadBits += encodeSbrHeaderData (sbrHeaderData, - &cmonData->sbrBitbuf); - } - else { - payloadBits += FDKwriteBits (&cmonData->sbrBitbuf, 0, 1); + payloadBits += FDKwriteBits(&cmonData->sbrBitbuf, 1, 1); + payloadBits += encodeSbrHeaderData(sbrHeaderData, &cmonData->sbrBitbuf); + } else { + payloadBits += FDKwriteBits(&cmonData->sbrBitbuf, 0, 1); } cmonData->sbrHdrBits = payloadBits; @@ -324,8 +289,6 @@ encodeSbrHeader (HANDLE_SBR_HEADER_DATA sbrHeaderData, return payloadBits; } - - /***************************************************************************** functionname: encodeSbrHeaderData @@ -336,57 +299,54 @@ encodeSbrHeader (HANDLE_SBR_HEADER_DATA sbrHeaderData, output: *****************************************************************************/ -static INT -encodeSbrHeaderData (HANDLE_SBR_HEADER_DATA sbrHeaderData, - HANDLE_FDK_BITSTREAM hBitStream) +static INT encodeSbrHeaderData(HANDLE_SBR_HEADER_DATA sbrHeaderData, + HANDLE_FDK_BITSTREAM hBitStream) { INT payloadBits = 0; if (sbrHeaderData != NULL) { - payloadBits += FDKwriteBits (hBitStream, sbrHeaderData->sbr_amp_res, - SI_SBR_AMP_RES_BITS); - payloadBits += FDKwriteBits (hBitStream, sbrHeaderData->sbr_start_frequency, - SI_SBR_START_FREQ_BITS); - payloadBits += FDKwriteBits (hBitStream, sbrHeaderData->sbr_stop_frequency, - SI_SBR_STOP_FREQ_BITS); - payloadBits += FDKwriteBits (hBitStream, sbrHeaderData->sbr_xover_band, - SI_SBR_XOVER_BAND_BITS); - - payloadBits += FDKwriteBits (hBitStream, 0, - SI_SBR_RESERVED_BITS); - - payloadBits += FDKwriteBits (hBitStream, sbrHeaderData->header_extra_1, - SI_SBR_HEADER_EXTRA_1_BITS); - payloadBits += FDKwriteBits (hBitStream, sbrHeaderData->header_extra_2, - SI_SBR_HEADER_EXTRA_2_BITS); - + payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->sbr_amp_res, + SI_SBR_AMP_RES_BITS); + payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->sbr_start_frequency, + SI_SBR_START_FREQ_BITS); + payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->sbr_stop_frequency, + SI_SBR_STOP_FREQ_BITS); + payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->sbr_xover_band, + SI_SBR_XOVER_BAND_BITS); + + payloadBits += FDKwriteBits(hBitStream, 0, SI_SBR_RESERVED_BITS); + + payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->header_extra_1, + SI_SBR_HEADER_EXTRA_1_BITS); + payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->header_extra_2, + SI_SBR_HEADER_EXTRA_2_BITS); if (sbrHeaderData->header_extra_1) { - payloadBits += FDKwriteBits (hBitStream, sbrHeaderData->freqScale, - SI_SBR_FREQ_SCALE_BITS); - payloadBits += FDKwriteBits (hBitStream, sbrHeaderData->alterScale, - SI_SBR_ALTER_SCALE_BITS); - payloadBits += FDKwriteBits (hBitStream, sbrHeaderData->sbr_noise_bands, - SI_SBR_NOISE_BANDS_BITS); + payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->freqScale, + SI_SBR_FREQ_SCALE_BITS); + payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->alterScale, + SI_SBR_ALTER_SCALE_BITS); + payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->sbr_noise_bands, + SI_SBR_NOISE_BANDS_BITS); } /* sbrHeaderData->header_extra_1 */ if (sbrHeaderData->header_extra_2) { - payloadBits += FDKwriteBits (hBitStream, sbrHeaderData->sbr_limiter_bands, - SI_SBR_LIMITER_BANDS_BITS); - payloadBits += FDKwriteBits (hBitStream, sbrHeaderData->sbr_limiter_gains, - SI_SBR_LIMITER_GAINS_BITS); - payloadBits += FDKwriteBits (hBitStream, sbrHeaderData->sbr_interpol_freq, - SI_SBR_INTERPOL_FREQ_BITS); - payloadBits += FDKwriteBits (hBitStream, sbrHeaderData->sbr_smoothing_length, - SI_SBR_SMOOTHING_LENGTH_BITS); + payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->sbr_limiter_bands, + SI_SBR_LIMITER_BANDS_BITS); + payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->sbr_limiter_gains, + SI_SBR_LIMITER_GAINS_BITS); + payloadBits += FDKwriteBits(hBitStream, sbrHeaderData->sbr_interpol_freq, + SI_SBR_INTERPOL_FREQ_BITS); + payloadBits += + FDKwriteBits(hBitStream, sbrHeaderData->sbr_smoothing_length, + SI_SBR_SMOOTHING_LENGTH_BITS); } /* sbrHeaderData->header_extra_2 */ - } /* sbrHeaderData != NULL */ + } /* sbrHeaderData != NULL */ return payloadBits; } - /***************************************************************************** functionname: encodeSbrData @@ -396,27 +356,27 @@ encodeSbrHeaderData (HANDLE_SBR_HEADER_DATA sbrHeaderData, output: *****************************************************************************/ -static INT -encodeSbrData (HANDLE_SBR_ENV_DATA sbrEnvDataLeft, - HANDLE_SBR_ENV_DATA sbrEnvDataRight, - HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_COMMON_DATA cmonData, - SBR_ELEMENT_TYPE sbrElem, - INT coupling, - UINT sbrSyntaxFlags) -{ +static INT encodeSbrData(HANDLE_SBR_ENV_DATA sbrEnvDataLeft, + HANDLE_SBR_ENV_DATA sbrEnvDataRight, + HANDLE_PARAMETRIC_STEREO hParametricStereo, + HANDLE_COMMON_DATA cmonData, SBR_ELEMENT_TYPE sbrElem, + INT coupling, UINT sbrSyntaxFlags) { INT payloadBits = 0; switch (sbrElem) { - case SBR_ID_SCE: - payloadBits += encodeSbrSingleChannelElement (sbrEnvDataLeft, &cmonData->sbrBitbuf, hParametricStereo, sbrSyntaxFlags); - break; - case SBR_ID_CPE: - payloadBits += encodeSbrChannelPairElement (sbrEnvDataLeft, sbrEnvDataRight, hParametricStereo, &cmonData->sbrBitbuf, coupling); - break; - default: - /* we never should apply SBR to any other element type */ - FDK_ASSERT (0); + case SBR_ID_SCE: + payloadBits += + encodeSbrSingleChannelElement(sbrEnvDataLeft, &cmonData->sbrBitbuf, + hParametricStereo, sbrSyntaxFlags); + break; + case SBR_ID_CPE: + payloadBits += encodeSbrChannelPairElement( + sbrEnvDataLeft, sbrEnvDataRight, hParametricStereo, + &cmonData->sbrBitbuf, coupling, sbrSyntaxFlags); + break; + default: + /* we never should apply SBR to any other element type */ + FDK_ASSERT(0); } cmonData->sbrDataBits = payloadBits; @@ -424,13 +384,10 @@ encodeSbrData (HANDLE_SBR_ENV_DATA sbrEnvDataLeft, return payloadBits; } -#define MODE_FREQ_TANS 1 -#define MODE_NO_FREQ_TRAN 0 -#define LD_TRANSMISSION MODE_FREQ_TANS -static int encodeFreqs (int mode) { - return ((mode & MODE_FREQ_TANS) ? 1 : 0); -} - +#define MODE_FREQ_TANS 1 +#define MODE_NO_FREQ_TRAN 0 +#define LD_TRANSMISSION MODE_FREQ_TANS +static int encodeFreqs(int mode) { return ((mode & MODE_FREQ_TANS) ? 1 : 0); } /***************************************************************************** @@ -441,51 +398,47 @@ static int encodeFreqs (int mode) { output: *****************************************************************************/ -static INT -encodeSbrSingleChannelElement (HANDLE_SBR_ENV_DATA sbrEnvData, - HANDLE_FDK_BITSTREAM hBitStream - ,HANDLE_PARAMETRIC_STEREO hParametricStereo - ,UINT sbrSyntaxFlags - ) -{ +static INT encodeSbrSingleChannelElement( + HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream, + HANDLE_PARAMETRIC_STEREO hParametricStereo, const UINT sbrSyntaxFlags) { INT i, payloadBits = 0; - payloadBits += FDKwriteBits (hBitStream, 0, SI_SBR_DATA_EXTRA_BITS); /* no reserved bits */ + payloadBits += FDKwriteBits(hBitStream, 0, + SI_SBR_DATA_EXTRA_BITS); /* no reserved bits */ if (sbrEnvData->ldGrid) { - if ( sbrEnvData->hSbrBSGrid->frameClass != FIXFIXonly ) { - /* encode normal SbrGrid */ - payloadBits += encodeSbrGrid (sbrEnvData, hBitStream); - } else { - /* use FIXFIXonly frame Grid */ - payloadBits += encodeLowDelaySbrGrid ( sbrEnvData, hBitStream, encodeFreqs(LD_TRANSMISSION)); - } - } - else - { + if (sbrEnvData->hSbrBSGrid->frameClass != FIXFIXonly) { + /* encode normal SbrGrid */ + payloadBits += encodeSbrGrid(sbrEnvData, hBitStream); + } else { + /* use FIXFIXonly frame Grid */ + payloadBits += encodeLowDelaySbrGrid( + sbrEnvData, hBitStream, encodeFreqs(LD_TRANSMISSION), sbrSyntaxFlags); + } + } else { if (sbrSyntaxFlags & SBR_SYNTAX_SCALABLE) { - payloadBits += FDKwriteBits (hBitStream, 1, SI_SBR_COUPLING_BITS); + payloadBits += FDKwriteBits(hBitStream, 1, SI_SBR_COUPLING_BITS); } - payloadBits += encodeSbrGrid (sbrEnvData, hBitStream); + payloadBits += encodeSbrGrid(sbrEnvData, hBitStream); } - payloadBits += encodeSbrDtdf (sbrEnvData, hBitStream); + payloadBits += encodeSbrDtdf(sbrEnvData, hBitStream); for (i = 0; i < sbrEnvData->noOfnoisebands; i++) { - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->sbr_invf_mode_vec[i], SI_SBR_INVF_MODE_BITS); + payloadBits += FDKwriteBits(hBitStream, sbrEnvData->sbr_invf_mode_vec[i], + SI_SBR_INVF_MODE_BITS); } - payloadBits += writeEnvelopeData (sbrEnvData, hBitStream, 0); - payloadBits += writeNoiseLevelData (sbrEnvData, hBitStream, 0); + payloadBits += writeEnvelopeData(sbrEnvData, hBitStream, 0); + payloadBits += writeNoiseLevelData(sbrEnvData, hBitStream, 0); - payloadBits += writeSyntheticCodingData (sbrEnvData,hBitStream); + payloadBits += writeSyntheticCodingData(sbrEnvData, hBitStream); payloadBits += encodeExtendedData(hParametricStereo, hBitStream); return payloadBits; } - /***************************************************************************** functionname: encodeSbrChannelPairElement @@ -495,97 +448,104 @@ encodeSbrSingleChannelElement (HANDLE_SBR_ENV_DATA sbrEnvData, output: *****************************************************************************/ -static INT -encodeSbrChannelPairElement (HANDLE_SBR_ENV_DATA sbrEnvDataLeft, - HANDLE_SBR_ENV_DATA sbrEnvDataRight, - HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_FDK_BITSTREAM hBitStream, - INT coupling) -{ +static INT encodeSbrChannelPairElement( + HANDLE_SBR_ENV_DATA sbrEnvDataLeft, HANDLE_SBR_ENV_DATA sbrEnvDataRight, + HANDLE_PARAMETRIC_STEREO hParametricStereo, HANDLE_FDK_BITSTREAM hBitStream, + const INT coupling, const UINT sbrSyntaxFlags) { INT payloadBits = 0; INT i = 0; - payloadBits += FDKwriteBits (hBitStream, 0, SI_SBR_DATA_EXTRA_BITS); /* no reserved bits */ + payloadBits += FDKwriteBits(hBitStream, 0, + SI_SBR_DATA_EXTRA_BITS); /* no reserved bits */ - payloadBits += FDKwriteBits (hBitStream, coupling, SI_SBR_COUPLING_BITS); + payloadBits += FDKwriteBits(hBitStream, coupling, SI_SBR_COUPLING_BITS); if (coupling) { if (sbrEnvDataLeft->ldGrid) { - if ( sbrEnvDataLeft->hSbrBSGrid->frameClass != FIXFIXonly ) { - /* normal SbrGrid */ - payloadBits += encodeSbrGrid (sbrEnvDataLeft, hBitStream); + if (sbrEnvDataLeft->hSbrBSGrid->frameClass != FIXFIXonly) { + /* normal SbrGrid */ + payloadBits += encodeSbrGrid(sbrEnvDataLeft, hBitStream); - } else { - /* FIXFIXonly frame Grid */ - payloadBits += encodeLowDelaySbrGrid ( sbrEnvDataLeft, hBitStream, encodeFreqs(LD_TRANSMISSION)); - } + } else { + /* FIXFIXonly frame Grid */ + payloadBits += + encodeLowDelaySbrGrid(sbrEnvDataLeft, hBitStream, + encodeFreqs(LD_TRANSMISSION), sbrSyntaxFlags); + } } else - payloadBits += encodeSbrGrid (sbrEnvDataLeft, hBitStream); + payloadBits += encodeSbrGrid(sbrEnvDataLeft, hBitStream); - payloadBits += encodeSbrDtdf (sbrEnvDataLeft, hBitStream); - payloadBits += encodeSbrDtdf (sbrEnvDataRight, hBitStream); + payloadBits += encodeSbrDtdf(sbrEnvDataLeft, hBitStream); + payloadBits += encodeSbrDtdf(sbrEnvDataRight, hBitStream); for (i = 0; i < sbrEnvDataLeft->noOfnoisebands; i++) { - payloadBits += FDKwriteBits (hBitStream, sbrEnvDataLeft->sbr_invf_mode_vec[i], SI_SBR_INVF_MODE_BITS); + payloadBits += + FDKwriteBits(hBitStream, sbrEnvDataLeft->sbr_invf_mode_vec[i], + SI_SBR_INVF_MODE_BITS); } - payloadBits += writeEnvelopeData (sbrEnvDataLeft, hBitStream,1); - payloadBits += writeNoiseLevelData (sbrEnvDataLeft, hBitStream,1); - payloadBits += writeEnvelopeData (sbrEnvDataRight, hBitStream,1); - payloadBits += writeNoiseLevelData (sbrEnvDataRight, hBitStream,1); + payloadBits += writeEnvelopeData(sbrEnvDataLeft, hBitStream, 1); + payloadBits += writeNoiseLevelData(sbrEnvDataLeft, hBitStream, 1); + payloadBits += writeEnvelopeData(sbrEnvDataRight, hBitStream, 1); + payloadBits += writeNoiseLevelData(sbrEnvDataRight, hBitStream, 1); - payloadBits += writeSyntheticCodingData (sbrEnvDataLeft,hBitStream); - payloadBits += writeSyntheticCodingData (sbrEnvDataRight,hBitStream); + payloadBits += writeSyntheticCodingData(sbrEnvDataLeft, hBitStream); + payloadBits += writeSyntheticCodingData(sbrEnvDataRight, hBitStream); } else { /* no coupling */ FDK_ASSERT(sbrEnvDataLeft->ldGrid == sbrEnvDataRight->ldGrid); if (sbrEnvDataLeft->ldGrid || sbrEnvDataRight->ldGrid) { - /* sbrEnvDataLeft (left channel) */ - if ( sbrEnvDataLeft->hSbrBSGrid->frameClass != FIXFIXonly) { + /* sbrEnvDataLeft (left channel) */ + if (sbrEnvDataLeft->hSbrBSGrid->frameClass != FIXFIXonly) { /* no FIXFIXonly Frame so we dont need encodeLowDelaySbrGrid */ /* normal SbrGrid */ - payloadBits += encodeSbrGrid (sbrEnvDataLeft, hBitStream); + payloadBits += encodeSbrGrid(sbrEnvDataLeft, hBitStream); } else { /* FIXFIXonly frame Grid */ - payloadBits += encodeLowDelaySbrGrid ( sbrEnvDataLeft, hBitStream, encodeFreqs(LD_TRANSMISSION)); + payloadBits += + encodeLowDelaySbrGrid(sbrEnvDataLeft, hBitStream, + encodeFreqs(LD_TRANSMISSION), sbrSyntaxFlags); } /* sbrEnvDataRight (right channel) */ - if ( sbrEnvDataRight->hSbrBSGrid->frameClass != FIXFIXonly) { + if (sbrEnvDataRight->hSbrBSGrid->frameClass != FIXFIXonly) { /* no FIXFIXonly Frame so we dont need encodeLowDelaySbrGrid */ /* normal SbrGrid */ - payloadBits += encodeSbrGrid (sbrEnvDataRight, hBitStream); + payloadBits += encodeSbrGrid(sbrEnvDataRight, hBitStream); } else { /* FIXFIXonly frame Grid */ - payloadBits += encodeLowDelaySbrGrid ( sbrEnvDataRight, hBitStream, encodeFreqs(LD_TRANSMISSION)); + payloadBits += + encodeLowDelaySbrGrid(sbrEnvDataRight, hBitStream, + encodeFreqs(LD_TRANSMISSION), sbrSyntaxFlags); } - } else - { - payloadBits += encodeSbrGrid (sbrEnvDataLeft, hBitStream); - payloadBits += encodeSbrGrid (sbrEnvDataRight, hBitStream); + } else { + payloadBits += encodeSbrGrid(sbrEnvDataLeft, hBitStream); + payloadBits += encodeSbrGrid(sbrEnvDataRight, hBitStream); } - payloadBits += encodeSbrDtdf (sbrEnvDataLeft, hBitStream); - payloadBits += encodeSbrDtdf (sbrEnvDataRight, hBitStream); + payloadBits += encodeSbrDtdf(sbrEnvDataLeft, hBitStream); + payloadBits += encodeSbrDtdf(sbrEnvDataRight, hBitStream); for (i = 0; i < sbrEnvDataLeft->noOfnoisebands; i++) { - payloadBits += FDKwriteBits (hBitStream, sbrEnvDataLeft->sbr_invf_mode_vec[i], - SI_SBR_INVF_MODE_BITS); + payloadBits += + FDKwriteBits(hBitStream, sbrEnvDataLeft->sbr_invf_mode_vec[i], + SI_SBR_INVF_MODE_BITS); } for (i = 0; i < sbrEnvDataRight->noOfnoisebands; i++) { - payloadBits += FDKwriteBits (hBitStream, sbrEnvDataRight->sbr_invf_mode_vec[i], - SI_SBR_INVF_MODE_BITS); + payloadBits += + FDKwriteBits(hBitStream, sbrEnvDataRight->sbr_invf_mode_vec[i], + SI_SBR_INVF_MODE_BITS); } - payloadBits += writeEnvelopeData (sbrEnvDataLeft, hBitStream,0); - payloadBits += writeEnvelopeData (sbrEnvDataRight, hBitStream,0); - payloadBits += writeNoiseLevelData (sbrEnvDataLeft, hBitStream,0); - payloadBits += writeNoiseLevelData (sbrEnvDataRight, hBitStream,0); + payloadBits += writeEnvelopeData(sbrEnvDataLeft, hBitStream, 0); + payloadBits += writeEnvelopeData(sbrEnvDataRight, hBitStream, 0); + payloadBits += writeNoiseLevelData(sbrEnvDataLeft, hBitStream, 0); + payloadBits += writeNoiseLevelData(sbrEnvDataRight, hBitStream, 0); - payloadBits += writeSyntheticCodingData (sbrEnvDataLeft,hBitStream); - payloadBits += writeSyntheticCodingData (sbrEnvDataRight,hBitStream); + payloadBits += writeSyntheticCodingData(sbrEnvDataLeft, hBitStream); + payloadBits += writeSyntheticCodingData(sbrEnvDataRight, hBitStream); } /* coupling */ @@ -594,14 +554,13 @@ encodeSbrChannelPairElement (HANDLE_SBR_ENV_DATA sbrEnvDataLeft, return payloadBits; } -static INT ceil_ln2(INT x) -{ - INT tmp=-1; - while((1<<++tmp) < x); - return(tmp); +static INT ceil_ln2(INT x) { + INT tmp = -1; + while ((1 << ++tmp) < x) + ; + return (tmp); } - /***************************************************************************** functionname: encodeSbrGrid @@ -612,91 +571,95 @@ static INT ceil_ln2(INT x) output: *****************************************************************************/ -static INT -encodeSbrGrid (HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream) -{ +static INT encodeSbrGrid(HANDLE_SBR_ENV_DATA sbrEnvData, + HANDLE_FDK_BITSTREAM hBitStream) { INT payloadBits = 0; INT i, temp; INT bufferFrameStart = sbrEnvData->hSbrBSGrid->bufferFrameStart; - INT numberTimeSlots = sbrEnvData->hSbrBSGrid->numberTimeSlots; + INT numberTimeSlots = sbrEnvData->hSbrBSGrid->numberTimeSlots; if (sbrEnvData->ldGrid) - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hSbrBSGrid->frameClass, - SBR_CLA_BITS_LD); + payloadBits += FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->frameClass, + SBR_CLA_BITS_LD); else - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hSbrBSGrid->frameClass, - SBR_CLA_BITS); + payloadBits += FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->frameClass, + SBR_CLA_BITS); switch (sbrEnvData->hSbrBSGrid->frameClass) { - case FIXFIXonly: - FDK_ASSERT(0 /* Fatal error in encodeSbrGrid! */); - break; - case FIXFIX: - temp = ceil_ln2(sbrEnvData->hSbrBSGrid->bs_num_env); - payloadBits += FDKwriteBits (hBitStream, temp, SBR_ENV_BITS); - if ((sbrEnvData->ldGrid) && (sbrEnvData->hSbrBSGrid->bs_num_env==1)) - payloadBits += FDKwriteBits(hBitStream, sbrEnvData->currentAmpResFF, SI_SBR_AMP_RES_BITS); - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->hSbrBSGrid->v_f[0], SBR_RES_BITS); - - break; - - case FIXVAR: - case VARFIX: - if (sbrEnvData->hSbrBSGrid->frameClass == FIXVAR) - temp = sbrEnvData->hSbrBSGrid->bs_abs_bord - (bufferFrameStart + numberTimeSlots); - else - temp = sbrEnvData->hSbrBSGrid->bs_abs_bord - bufferFrameStart; - - payloadBits += FDKwriteBits (hBitStream, temp, SBR_ABS_BITS); - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->hSbrBSGrid->n, SBR_NUM_BITS); - - for (i = 0; i < sbrEnvData->hSbrBSGrid->n; i++) { - temp = (sbrEnvData->hSbrBSGrid->bs_rel_bord[i] - 2) >> 1; - payloadBits += FDKwriteBits (hBitStream, temp, SBR_REL_BITS); - } + case FIXFIXonly: + FDK_ASSERT(0 /* Fatal error in encodeSbrGrid! */); + break; + case FIXFIX: + temp = ceil_ln2(sbrEnvData->hSbrBSGrid->bs_num_env); + payloadBits += FDKwriteBits(hBitStream, temp, SBR_ENV_BITS); + if ((sbrEnvData->ldGrid) && (sbrEnvData->hSbrBSGrid->bs_num_env == 1)) + payloadBits += FDKwriteBits(hBitStream, sbrEnvData->currentAmpResFF, + SI_SBR_AMP_RES_BITS); + payloadBits += FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->v_f[0], + SBR_RES_BITS); - temp = ceil_ln2(sbrEnvData->hSbrBSGrid->n + 2); - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->hSbrBSGrid->p, temp); + break; - for (i = 0; i < sbrEnvData->hSbrBSGrid->n + 1; i++) { - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->hSbrBSGrid->v_f[i], - SBR_RES_BITS); - } - break; + case FIXVAR: + case VARFIX: + if (sbrEnvData->hSbrBSGrid->frameClass == FIXVAR) + temp = sbrEnvData->hSbrBSGrid->bs_abs_bord - + (bufferFrameStart + numberTimeSlots); + else + temp = sbrEnvData->hSbrBSGrid->bs_abs_bord - bufferFrameStart; + + payloadBits += FDKwriteBits(hBitStream, temp, SBR_ABS_BITS); + payloadBits += + FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->n, SBR_NUM_BITS); + + for (i = 0; i < sbrEnvData->hSbrBSGrid->n; i++) { + temp = (sbrEnvData->hSbrBSGrid->bs_rel_bord[i] - 2) >> 1; + payloadBits += FDKwriteBits(hBitStream, temp, SBR_REL_BITS); + } - case VARVAR: - temp = sbrEnvData->hSbrBSGrid->bs_abs_bord_0 - bufferFrameStart; - payloadBits += FDKwriteBits (hBitStream, temp, SBR_ABS_BITS); - temp = sbrEnvData->hSbrBSGrid->bs_abs_bord_1 - (bufferFrameStart + numberTimeSlots); - payloadBits += FDKwriteBits (hBitStream, temp, SBR_ABS_BITS); + temp = ceil_ln2(sbrEnvData->hSbrBSGrid->n + 2); + payloadBits += FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->p, temp); - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->hSbrBSGrid->bs_num_rel_0, SBR_NUM_BITS); - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->hSbrBSGrid->bs_num_rel_1, SBR_NUM_BITS); + for (i = 0; i < sbrEnvData->hSbrBSGrid->n + 1; i++) { + payloadBits += FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->v_f[i], + SBR_RES_BITS); + } + break; - for (i = 0; i < sbrEnvData->hSbrBSGrid->bs_num_rel_0; i++) { - temp = (sbrEnvData->hSbrBSGrid->bs_rel_bord_0[i] - 2) >> 1; - payloadBits += FDKwriteBits (hBitStream, temp, SBR_REL_BITS); - } + case VARVAR: + temp = sbrEnvData->hSbrBSGrid->bs_abs_bord_0 - bufferFrameStart; + payloadBits += FDKwriteBits(hBitStream, temp, SBR_ABS_BITS); + temp = sbrEnvData->hSbrBSGrid->bs_abs_bord_1 - + (bufferFrameStart + numberTimeSlots); + payloadBits += FDKwriteBits(hBitStream, temp, SBR_ABS_BITS); + + payloadBits += FDKwriteBits( + hBitStream, sbrEnvData->hSbrBSGrid->bs_num_rel_0, SBR_NUM_BITS); + payloadBits += FDKwriteBits( + hBitStream, sbrEnvData->hSbrBSGrid->bs_num_rel_1, SBR_NUM_BITS); + + for (i = 0; i < sbrEnvData->hSbrBSGrid->bs_num_rel_0; i++) { + temp = (sbrEnvData->hSbrBSGrid->bs_rel_bord_0[i] - 2) >> 1; + payloadBits += FDKwriteBits(hBitStream, temp, SBR_REL_BITS); + } - for (i = 0; i < sbrEnvData->hSbrBSGrid->bs_num_rel_1; i++) { - temp = (sbrEnvData->hSbrBSGrid->bs_rel_bord_1[i] - 2) >> 1; - payloadBits += FDKwriteBits (hBitStream, temp, SBR_REL_BITS); - } + for (i = 0; i < sbrEnvData->hSbrBSGrid->bs_num_rel_1; i++) { + temp = (sbrEnvData->hSbrBSGrid->bs_rel_bord_1[i] - 2) >> 1; + payloadBits += FDKwriteBits(hBitStream, temp, SBR_REL_BITS); + } - temp = ceil_ln2(sbrEnvData->hSbrBSGrid->bs_num_rel_0 + - sbrEnvData->hSbrBSGrid->bs_num_rel_1 + 2); - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->hSbrBSGrid->p, temp); + temp = ceil_ln2(sbrEnvData->hSbrBSGrid->bs_num_rel_0 + + sbrEnvData->hSbrBSGrid->bs_num_rel_1 + 2); + payloadBits += FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->p, temp); - temp = sbrEnvData->hSbrBSGrid->bs_num_rel_0 + - sbrEnvData->hSbrBSGrid->bs_num_rel_1 + 1; + temp = sbrEnvData->hSbrBSGrid->bs_num_rel_0 + + sbrEnvData->hSbrBSGrid->bs_num_rel_1 + 1; - for (i = 0; i < temp; i++) { - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->hSbrBSGrid->v_fLR[i], - SBR_RES_BITS); - } - break; + for (i = 0; i < temp; i++) { + payloadBits += FDKwriteBits( + hBitStream, sbrEnvData->hSbrBSGrid->v_fLR[i], SBR_RES_BITS); + } + break; } return payloadBits; @@ -715,12 +678,10 @@ encodeSbrGrid (HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream) output: *****************************************************************************/ -static int -encodeLowDelaySbrGrid ( HANDLE_SBR_ENV_DATA sbrEnvData, - HANDLE_FDK_BITSTREAM hBitStream, - int transmitFreqs - ) -{ +static int encodeLowDelaySbrGrid(HANDLE_SBR_ENV_DATA sbrEnvData, + HANDLE_FDK_BITSTREAM hBitStream, + const int transmitFreqs, + const UINT sbrSyntaxFlags) { int payloadBits = 0; int i; @@ -728,21 +689,25 @@ encodeLowDelaySbrGrid ( HANDLE_SBR_ENV_DATA sbrEnvData, /* write frameClass [1 bit] for FIXFIXonly Grid */ payloadBits += FDKwriteBits(hBitStream, 1, SBR_CLA_BITS_LD); - /* absolute Borders are fix: 0,X,X,X,nTimeSlots; so we dont have to transmit them */ + /* absolute Borders are fix: 0,X,X,X,nTimeSlots; so we dont have to transmit + * them */ /* only transmit the transient position! */ /* with this info (b1) we can reconstruct the Frame on Decoder side : */ /* border[0] = 0; border[1] = b1; border[2]=b1+2; border[3] = nrTimeSlots */ /* use 3 or 4bits for transient border (border) */ if (sbrEnvData->hSbrBSGrid->numberTimeSlots == 8) - payloadBits += FDKwriteBits ( hBitStream, sbrEnvData->hSbrBSGrid->bs_abs_bord, 3); + payloadBits += + FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->bs_abs_bord, 3); else - payloadBits += FDKwriteBits ( hBitStream, sbrEnvData->hSbrBSGrid->bs_abs_bord, 4); + payloadBits += + FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->bs_abs_bord, 4); if (transmitFreqs) { /* write FreqRes grid */ for (i = 0; i < sbrEnvData->hSbrBSGrid->bs_num_env; i++) { - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->hSbrBSGrid->v_f[i], SBR_RES_BITS); + payloadBits += FDKwriteBits(hBitStream, sbrEnvData->hSbrBSGrid->v_f[i], + SBR_RES_BITS); } } @@ -752,30 +717,28 @@ encodeLowDelaySbrGrid ( HANDLE_SBR_ENV_DATA sbrEnvData, /***************************************************************************** functionname: encodeSbrDtdf - description: writes bits that describes the direction of the envelopes of a frame - returns: number of bits written - input: - output: + description: writes bits that describes the direction of the envelopes of a +frame returns: number of bits written input: output: *****************************************************************************/ -static INT -encodeSbrDtdf (HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream) -{ +static INT encodeSbrDtdf(HANDLE_SBR_ENV_DATA sbrEnvData, + HANDLE_FDK_BITSTREAM hBitStream) { INT i, payloadBits = 0, noOfNoiseEnvelopes; noOfNoiseEnvelopes = sbrEnvData->noOfEnvelopes > 1 ? 2 : 1; for (i = 0; i < sbrEnvData->noOfEnvelopes; ++i) { - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->domain_vec[i], SBR_DIR_BITS); + payloadBits += + FDKwriteBits(hBitStream, sbrEnvData->domain_vec[i], SBR_DIR_BITS); } for (i = 0; i < noOfNoiseEnvelopes; ++i) { - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->domain_vec_noise[i], SBR_DIR_BITS); + payloadBits += + FDKwriteBits(hBitStream, sbrEnvData->domain_vec_noise[i], SBR_DIR_BITS); } return payloadBits; } - /***************************************************************************** functionname: writeNoiseLevelData @@ -785,87 +748,101 @@ encodeSbrDtdf (HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream) output: *****************************************************************************/ -static INT -writeNoiseLevelData (HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream, INT coupling) -{ +static INT writeNoiseLevelData(HANDLE_SBR_ENV_DATA sbrEnvData, + HANDLE_FDK_BITSTREAM hBitStream, INT coupling) { INT j, i, payloadBits = 0; INT nNoiseEnvelopes = sbrEnvData->noOfEnvelopes > 1 ? 2 : 1; for (i = 0; i < nNoiseEnvelopes; i++) { switch (sbrEnvData->domain_vec_noise[i]) { - case FREQ: - if (coupling && sbrEnvData->balance) { - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->sbr_noise_levels[i * sbrEnvData->noOfnoisebands], - sbrEnvData->si_sbr_start_noise_bits_balance); - } else { - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->sbr_noise_levels[i * sbrEnvData->noOfnoisebands], - sbrEnvData->si_sbr_start_noise_bits); - } + case FREQ: + if (coupling && sbrEnvData->balance) { + payloadBits += FDKwriteBits( + hBitStream, + sbrEnvData->sbr_noise_levels[i * sbrEnvData->noOfnoisebands], + sbrEnvData->si_sbr_start_noise_bits_balance); + } else { + payloadBits += FDKwriteBits( + hBitStream, + sbrEnvData->sbr_noise_levels[i * sbrEnvData->noOfnoisebands], + sbrEnvData->si_sbr_start_noise_bits); + } - for (j = 1 + i * sbrEnvData->noOfnoisebands; j < (sbrEnvData->noOfnoisebands * (1 + i)); j++) { - if (coupling) { - if (sbrEnvData->balance) { - /* coupling && balance */ - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hufftableNoiseBalanceFreqC[sbrEnvData->sbr_noise_levels[j] + - CODE_BOOK_SCF_LAV_BALANCE11], - sbrEnvData->hufftableNoiseBalanceFreqL[sbrEnvData->sbr_noise_levels[j] + - CODE_BOOK_SCF_LAV_BALANCE11]); + for (j = 1 + i * sbrEnvData->noOfnoisebands; + j < (sbrEnvData->noOfnoisebands * (1 + i)); j++) { + if (coupling) { + if (sbrEnvData->balance) { + /* coupling && balance */ + payloadBits += FDKwriteBits(hBitStream, + sbrEnvData->hufftableNoiseBalanceFreqC + [sbrEnvData->sbr_noise_levels[j] + + CODE_BOOK_SCF_LAV_BALANCE11], + sbrEnvData->hufftableNoiseBalanceFreqL + [sbrEnvData->sbr_noise_levels[j] + + CODE_BOOK_SCF_LAV_BALANCE11]); + } else { + /* coupling && !balance */ + payloadBits += FDKwriteBits( + hBitStream, + sbrEnvData->hufftableNoiseLevelFreqC + [sbrEnvData->sbr_noise_levels[j] + CODE_BOOK_SCF_LAV11], + sbrEnvData->hufftableNoiseLevelFreqL + [sbrEnvData->sbr_noise_levels[j] + CODE_BOOK_SCF_LAV11]); + } } else { - /* coupling && !balance */ - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hufftableNoiseLevelFreqC[sbrEnvData->sbr_noise_levels[j] + - CODE_BOOK_SCF_LAV11], - sbrEnvData->hufftableNoiseLevelFreqL[sbrEnvData->sbr_noise_levels[j] + - CODE_BOOK_SCF_LAV11]); + /* !coupling */ + payloadBits += FDKwriteBits( + hBitStream, + sbrEnvData + ->hufftableNoiseFreqC[sbrEnvData->sbr_noise_levels[j] + + CODE_BOOK_SCF_LAV11], + sbrEnvData + ->hufftableNoiseFreqL[sbrEnvData->sbr_noise_levels[j] + + CODE_BOOK_SCF_LAV11]); } - } else { - /* !coupling */ - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hufftableNoiseFreqC[sbrEnvData->sbr_noise_levels[j] + - CODE_BOOK_SCF_LAV11], - sbrEnvData->hufftableNoiseFreqL[sbrEnvData->sbr_noise_levels[j] + - CODE_BOOK_SCF_LAV11]); } - } - break; - - case TIME: - for (j = i * sbrEnvData->noOfnoisebands; j < (sbrEnvData->noOfnoisebands * (1 + i)); j++) { - if (coupling) { - if (sbrEnvData->balance) { - /* coupling && balance */ - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hufftableNoiseBalanceTimeC[sbrEnvData->sbr_noise_levels[j] + - CODE_BOOK_SCF_LAV_BALANCE11], - sbrEnvData->hufftableNoiseBalanceTimeL[sbrEnvData->sbr_noise_levels[j] + - CODE_BOOK_SCF_LAV_BALANCE11]); + break; + + case TIME: + for (j = i * sbrEnvData->noOfnoisebands; + j < (sbrEnvData->noOfnoisebands * (1 + i)); j++) { + if (coupling) { + if (sbrEnvData->balance) { + /* coupling && balance */ + payloadBits += FDKwriteBits(hBitStream, + sbrEnvData->hufftableNoiseBalanceTimeC + [sbrEnvData->sbr_noise_levels[j] + + CODE_BOOK_SCF_LAV_BALANCE11], + sbrEnvData->hufftableNoiseBalanceTimeL + [sbrEnvData->sbr_noise_levels[j] + + CODE_BOOK_SCF_LAV_BALANCE11]); + } else { + /* coupling && !balance */ + payloadBits += FDKwriteBits( + hBitStream, + sbrEnvData->hufftableNoiseLevelTimeC + [sbrEnvData->sbr_noise_levels[j] + CODE_BOOK_SCF_LAV11], + sbrEnvData->hufftableNoiseLevelTimeL + [sbrEnvData->sbr_noise_levels[j] + CODE_BOOK_SCF_LAV11]); + } } else { - /* coupling && !balance */ - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hufftableNoiseLevelTimeC[sbrEnvData->sbr_noise_levels[j] + - CODE_BOOK_SCF_LAV11], - sbrEnvData->hufftableNoiseLevelTimeL[sbrEnvData->sbr_noise_levels[j] + - CODE_BOOK_SCF_LAV11]); + /* !coupling */ + payloadBits += FDKwriteBits( + hBitStream, + sbrEnvData + ->hufftableNoiseLevelTimeC[sbrEnvData->sbr_noise_levels[j] + + CODE_BOOK_SCF_LAV11], + sbrEnvData + ->hufftableNoiseLevelTimeL[sbrEnvData->sbr_noise_levels[j] + + CODE_BOOK_SCF_LAV11]); } - } else { - /* !coupling */ - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hufftableNoiseLevelTimeC[sbrEnvData->sbr_noise_levels[j] + - CODE_BOOK_SCF_LAV11], - sbrEnvData->hufftableNoiseLevelTimeL[sbrEnvData->sbr_noise_levels[j] + - CODE_BOOK_SCF_LAV11]); } - } - break; + break; } } return payloadBits; } - /***************************************************************************** functionname: writeEnvelopeData @@ -875,64 +852,85 @@ writeNoiseLevelData (HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitSt output: *****************************************************************************/ -static INT -writeEnvelopeData (HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStream, INT coupling) -{ +static INT writeEnvelopeData(HANDLE_SBR_ENV_DATA sbrEnvData, + HANDLE_FDK_BITSTREAM hBitStream, INT coupling) { INT payloadBits = 0, j, i, delta; - for (j = 0; j < sbrEnvData->noOfEnvelopes; j++) { /* loop over all envelopes */ + for (j = 0; j < sbrEnvData->noOfEnvelopes; + j++) { /* loop over all envelopes */ if (sbrEnvData->domain_vec[j] == FREQ) { if (coupling && sbrEnvData->balance) { - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->ienvelope[j][0], sbrEnvData->si_sbr_start_env_bits_balance); + payloadBits += FDKwriteBits(hBitStream, sbrEnvData->ienvelope[j][0], + sbrEnvData->si_sbr_start_env_bits_balance); } else { - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->ienvelope[j][0], sbrEnvData->si_sbr_start_env_bits); + payloadBits += FDKwriteBits(hBitStream, sbrEnvData->ienvelope[j][0], + sbrEnvData->si_sbr_start_env_bits); } } - for (i = 1 - sbrEnvData->domain_vec[j]; i < sbrEnvData->noScfBands[j]; i++) { + for (i = 1 - sbrEnvData->domain_vec[j]; i < sbrEnvData->noScfBands[j]; + i++) { delta = sbrEnvData->ienvelope[j][i]; if (coupling && sbrEnvData->balance) { - FDK_ASSERT (fixp_abs (delta) <= sbrEnvData->codeBookScfLavBalance); + FDK_ASSERT(fixp_abs(delta) <= sbrEnvData->codeBookScfLavBalance); } else { - FDK_ASSERT (fixp_abs (delta) <= sbrEnvData->codeBookScfLav); + FDK_ASSERT(fixp_abs(delta) <= sbrEnvData->codeBookScfLav); } if (coupling) { if (sbrEnvData->balance) { if (sbrEnvData->domain_vec[j]) { /* coupling && balance && TIME */ - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hufftableBalanceTimeC[delta + sbrEnvData->codeBookScfLavBalance], - sbrEnvData->hufftableBalanceTimeL[delta + sbrEnvData->codeBookScfLavBalance]); + payloadBits += FDKwriteBits( + hBitStream, + sbrEnvData + ->hufftableBalanceTimeC[delta + + sbrEnvData->codeBookScfLavBalance], + sbrEnvData + ->hufftableBalanceTimeL[delta + + sbrEnvData->codeBookScfLavBalance]); } else { /* coupling && balance && FREQ */ - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hufftableBalanceFreqC[delta + sbrEnvData->codeBookScfLavBalance], - sbrEnvData->hufftableBalanceFreqL[delta + sbrEnvData->codeBookScfLavBalance]); + payloadBits += FDKwriteBits( + hBitStream, + sbrEnvData + ->hufftableBalanceFreqC[delta + + sbrEnvData->codeBookScfLavBalance], + sbrEnvData + ->hufftableBalanceFreqL[delta + + sbrEnvData->codeBookScfLavBalance]); } } else { if (sbrEnvData->domain_vec[j]) { /* coupling && !balance && TIME */ - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hufftableLevelTimeC[delta + sbrEnvData->codeBookScfLav], - sbrEnvData->hufftableLevelTimeL[delta + sbrEnvData->codeBookScfLav]); + payloadBits += FDKwriteBits( + hBitStream, + sbrEnvData + ->hufftableLevelTimeC[delta + sbrEnvData->codeBookScfLav], + sbrEnvData + ->hufftableLevelTimeL[delta + sbrEnvData->codeBookScfLav]); } else { /* coupling && !balance && FREQ */ - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hufftableLevelFreqC[delta + sbrEnvData->codeBookScfLav], - sbrEnvData->hufftableLevelFreqL[delta + sbrEnvData->codeBookScfLav]); + payloadBits += FDKwriteBits( + hBitStream, + sbrEnvData + ->hufftableLevelFreqC[delta + sbrEnvData->codeBookScfLav], + sbrEnvData + ->hufftableLevelFreqL[delta + sbrEnvData->codeBookScfLav]); } } } else { if (sbrEnvData->domain_vec[j]) { /* !coupling && TIME */ - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hufftableTimeC[delta + sbrEnvData->codeBookScfLav], - sbrEnvData->hufftableTimeL[delta + sbrEnvData->codeBookScfLav]); + payloadBits += FDKwriteBits( + hBitStream, + sbrEnvData->hufftableTimeC[delta + sbrEnvData->codeBookScfLav], + sbrEnvData->hufftableTimeL[delta + sbrEnvData->codeBookScfLav]); } else { /* !coupling && FREQ */ - payloadBits += FDKwriteBits (hBitStream, - sbrEnvData->hufftableFreqC[delta + sbrEnvData->codeBookScfLav], - sbrEnvData->hufftableFreqL[delta + sbrEnvData->codeBookScfLav]); + payloadBits += FDKwriteBits( + hBitStream, + sbrEnvData->hufftableFreqC[delta + sbrEnvData->codeBookScfLav], + sbrEnvData->hufftableFreqL[delta + sbrEnvData->codeBookScfLav]); } } } @@ -940,7 +938,6 @@ writeEnvelopeData (HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStre return payloadBits; } - /***************************************************************************** functionname: encodeExtendedData @@ -950,49 +947,51 @@ writeEnvelopeData (HANDLE_SBR_ENV_DATA sbrEnvData, HANDLE_FDK_BITSTREAM hBitStre output: *****************************************************************************/ -static INT encodeExtendedData (HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_FDK_BITSTREAM hBitStream) -{ +static INT encodeExtendedData(HANDLE_PARAMETRIC_STEREO hParametricStereo, + HANDLE_FDK_BITSTREAM hBitStream) { INT extDataSize; INT payloadBits = 0; extDataSize = getSbrExtendedDataSize(hParametricStereo); - if (extDataSize != 0) { - INT maxExtSize = (1<<SI_SBR_EXTENSION_SIZE_BITS) - 1; + INT maxExtSize = (1 << SI_SBR_EXTENSION_SIZE_BITS) - 1; INT writtenNoBits = 0; /* needed to byte align the extended data */ - payloadBits += FDKwriteBits (hBitStream, 1, SI_SBR_EXTENDED_DATA_BITS); + payloadBits += FDKwriteBits(hBitStream, 1, SI_SBR_EXTENDED_DATA_BITS); FDK_ASSERT(extDataSize <= SBR_EXTENDED_DATA_MAX_CNT); if (extDataSize < maxExtSize) { - payloadBits += FDKwriteBits (hBitStream, extDataSize, SI_SBR_EXTENSION_SIZE_BITS); + payloadBits += + FDKwriteBits(hBitStream, extDataSize, SI_SBR_EXTENSION_SIZE_BITS); } else { - payloadBits += FDKwriteBits (hBitStream, maxExtSize, SI_SBR_EXTENSION_SIZE_BITS); - payloadBits += FDKwriteBits (hBitStream, extDataSize - maxExtSize, SI_SBR_EXTENSION_ESC_COUNT_BITS); + payloadBits += + FDKwriteBits(hBitStream, maxExtSize, SI_SBR_EXTENSION_SIZE_BITS); + payloadBits += FDKwriteBits(hBitStream, extDataSize - maxExtSize, + SI_SBR_EXTENSION_ESC_COUNT_BITS); } /* parametric coding signalled here? */ - if(hParametricStereo){ - writtenNoBits += FDKwriteBits (hBitStream, EXTENSION_ID_PS_CODING, SI_SBR_EXTENSION_ID_BITS); - writtenNoBits += FDKsbrEnc_PSEnc_WritePSData(hParametricStereo, hBitStream); + if (hParametricStereo) { + writtenNoBits += FDKwriteBits(hBitStream, EXTENSION_ID_PS_CODING, + SI_SBR_EXTENSION_ID_BITS); + writtenNoBits += + FDKsbrEnc_PSEnc_WritePSData(hParametricStereo, hBitStream); } payloadBits += writtenNoBits; /* byte alignment */ - writtenNoBits = writtenNoBits%8; - if(writtenNoBits) + writtenNoBits = writtenNoBits % 8; + if (writtenNoBits) payloadBits += FDKwriteBits(hBitStream, 0, (8 - writtenNoBits)); } else { - payloadBits += FDKwriteBits (hBitStream, 0, SI_SBR_EXTENDED_DATA_BITS); + payloadBits += FDKwriteBits(hBitStream, 0, SI_SBR_EXTENDED_DATA_BITS); } return payloadBits; } - /***************************************************************************** functionname: writeSyntheticCodingData @@ -1002,18 +1001,18 @@ static INT encodeExtendedData (HANDLE_PARAMETRIC_STEREO hParametricStereo, output: *****************************************************************************/ -static INT writeSyntheticCodingData (HANDLE_SBR_ENV_DATA sbrEnvData, - HANDLE_FDK_BITSTREAM hBitStream) +static INT writeSyntheticCodingData(HANDLE_SBR_ENV_DATA sbrEnvData, + HANDLE_FDK_BITSTREAM hBitStream) { INT i; INT payloadBits = 0; - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->addHarmonicFlag, 1); + payloadBits += FDKwriteBits(hBitStream, sbrEnvData->addHarmonicFlag, 1); if (sbrEnvData->addHarmonicFlag) { for (i = 0; i < sbrEnvData->noHarmonics; i++) { - payloadBits += FDKwriteBits (hBitStream, sbrEnvData->addHarmonic[i], 1); + payloadBits += FDKwriteBits(hBitStream, sbrEnvData->addHarmonic[i], 1); } } @@ -1031,9 +1030,7 @@ static INT writeSyntheticCodingData (HANDLE_SBR_ENV_DATA sbrEnvData, output: *****************************************************************************/ -static INT -getSbrExtendedDataSize (HANDLE_PARAMETRIC_STEREO hParametricStereo) -{ +static INT getSbrExtendedDataSize(HANDLE_PARAMETRIC_STEREO hParametricStereo) { INT extDataBits = 0; /* add your new extended data counting methods here */ @@ -1042,16 +1039,11 @@ getSbrExtendedDataSize (HANDLE_PARAMETRIC_STEREO hParametricStereo) no extended data */ - if(hParametricStereo){ + if (hParametricStereo) { /* PS extended data */ extDataBits += SI_SBR_EXTENSION_ID_BITS; extDataBits += FDKsbrEnc_PSEnc_WritePSData(hParametricStereo, NULL); } - return (extDataBits+7) >> 3; + return (extDataBits + 7) >> 3; } - - - - - diff --git a/libSBRenc/src/bit_sbr.h b/libSBRenc/src/bit_sbr.h index de4ac89..e90f52c 100644 --- a/libSBRenc/src/bit_sbr.h +++ b/libSBRenc/src/bit_sbr.h @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,14 +90,22 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ /*! \file - \brief SBR bit writing + \brief SBR bit writing $Revision: 92790 $ */ -#ifndef __BIT_SBR_H -#define __BIT_SBR_H +#ifndef BIT_SBR_H +#define BIT_SBR_H #include "sbr_def.h" #include "cmondata.h" @@ -94,20 +113,22 @@ amm-info@iis.fraunhofer.de struct SBR_ENV_DATA; -struct SBR_BITSTREAM_DATA -{ +struct SBR_BITSTREAM_DATA { INT TotalBits; INT PayloadBits; INT FillBits; INT HeaderActive; - INT NrSendHeaderData; /**< input from commandline */ - INT CountSendHeaderData; /**< modulo count. If < 0 then no counting is done (no SBR headers) */ + INT HeaderActiveDelay; /**< sbr payload and its header is delayed depending on + encoder configuration*/ + INT NrSendHeaderData; /**< input from commandline */ + INT CountSendHeaderData; /**< modulo count. If < 0 then no counting is done + (no SBR headers) */ + INT rightBorderFIX; /**< force VARFIX or FIXFIX frames */ }; -typedef struct SBR_BITSTREAM_DATA *HANDLE_SBR_BITSTREAM_DATA; +typedef struct SBR_BITSTREAM_DATA *HANDLE_SBR_BITSTREAM_DATA; -struct SBR_HEADER_DATA -{ +struct SBR_HEADER_DATA { AMP_RES sbr_amp_res; INT sbr_start_frequency; INT sbr_stop_frequency; @@ -133,13 +154,10 @@ struct SBR_HEADER_DATA /* element of singlechannelelement */ - }; typedef struct SBR_HEADER_DATA *HANDLE_SBR_HEADER_DATA; -struct SBR_ENV_DATA -{ - +struct SBR_ENV_DATA { INT sbr_xpos_ctrl; FREQ_RES freq_res_fixfix[2]; UCHAR fResTransIsLow; @@ -167,7 +185,6 @@ struct SBR_ENV_DATA const UCHAR *hufftableLevelFreqL; const UCHAR *hufftableBalanceFreqL; - const UCHAR *hufftableNoiseTimeL; const INT *hufftableNoiseTimeC; const UCHAR *hufftableNoiseFreqL; @@ -188,7 +205,6 @@ struct SBR_ENV_DATA INT addHarmonicFlag; UCHAR addHarmonic[MAX_FREQ_COEFFS]; - /* calculated helper vars */ INT si_sbr_start_env_bits_balance; INT si_sbr_start_env_bits; @@ -205,7 +221,10 @@ struct SBR_ENV_DATA INT balance; AMP_RES init_sbr_amp_res; AMP_RES currentAmpResFF; - FIXP_DBL ton_HF[SBR_GLOBAL_TONALITY_VALUES]; /* tonality is scaled by 2^19/0.524288f (fract part of RELAXATION) */ + FIXP_DBL + ton_HF[SBR_GLOBAL_TONALITY_VALUES]; /* tonality is scaled by + 2^19/0.524288f (fract part of + RELAXATION) */ FIXP_DBL global_tonality; /* extended data */ @@ -218,41 +237,31 @@ struct SBR_ENV_DATA }; typedef struct SBR_ENV_DATA *HANDLE_SBR_ENV_DATA; - - -INT FDKsbrEnc_WriteEnvSingleChannelElement(struct SBR_HEADER_DATA *sbrHeaderData, - struct T_PARAMETRIC_STEREO *hParametricStereo, - struct SBR_BITSTREAM_DATA *sbrBitstreamData, - struct SBR_ENV_DATA *sbrEnvData, - struct COMMON_DATA *cmonData, - UINT sbrSyntaxFlags); - - -INT FDKsbrEnc_WriteEnvChannelPairElement(struct SBR_HEADER_DATA *sbrHeaderData, - struct T_PARAMETRIC_STEREO *hParametricStereo, - struct SBR_BITSTREAM_DATA *sbrBitstreamData, - struct SBR_ENV_DATA *sbrEnvDataLeft, - struct SBR_ENV_DATA *sbrEnvDataRight, - struct COMMON_DATA *cmonData, - UINT sbrSyntaxFlags); - - - -INT FDKsbrEnc_CountSbrChannelPairElement (struct SBR_HEADER_DATA *sbrHeaderData, - struct T_PARAMETRIC_STEREO *hParametricStereo, - struct SBR_BITSTREAM_DATA *sbrBitstreamData, - struct SBR_ENV_DATA *sbrEnvDataLeft, - struct SBR_ENV_DATA *sbrEnvDataRight, - struct COMMON_DATA *cmonData, - UINT sbrSyntaxFlags); - - +INT FDKsbrEnc_WriteEnvSingleChannelElement( + struct SBR_HEADER_DATA *sbrHeaderData, + struct T_PARAMETRIC_STEREO *hParametricStereo, + struct SBR_BITSTREAM_DATA *sbrBitstreamData, + struct SBR_ENV_DATA *sbrEnvData, struct COMMON_DATA *cmonData, + UINT sbrSyntaxFlags); + +INT FDKsbrEnc_WriteEnvChannelPairElement( + struct SBR_HEADER_DATA *sbrHeaderData, + struct T_PARAMETRIC_STEREO *hParametricStereo, + struct SBR_BITSTREAM_DATA *sbrBitstreamData, + struct SBR_ENV_DATA *sbrEnvDataLeft, struct SBR_ENV_DATA *sbrEnvDataRight, + struct COMMON_DATA *cmonData, UINT sbrSyntaxFlags); + +INT FDKsbrEnc_CountSbrChannelPairElement( + struct SBR_HEADER_DATA *sbrHeaderData, + struct T_PARAMETRIC_STEREO *hParametricStereo, + struct SBR_BITSTREAM_DATA *sbrBitstreamData, + struct SBR_ENV_DATA *sbrEnvDataLeft, struct SBR_ENV_DATA *sbrEnvDataRight, + struct COMMON_DATA *cmonData, UINT sbrSyntaxFlags); /* debugging and tuning functions */ /*#define SBR_ENV_STATISTICS */ - /*#define SBR_PAYLOAD_MONITOR*/ #endif diff --git a/libSBRenc/src/cmondata.h b/libSBRenc/src/cmondata.h index 32e6993..0779b4d 100644 --- a/libSBRenc/src/cmondata.h +++ b/libSBRenc/src/cmondata.h @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,32 +90,38 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ /*! \file - \brief Core Coder's and SBR's shared data structure definition + \brief Core Coder's and SBR's shared data structure definition $Revision: + 92790 $ */ -#ifndef __SBR_CMONDATA_H -#define __SBR_CMONDATA_H +#ifndef CMONDATA_H +#define CMONDATA_H #include "FDK_bitstream.h" - struct COMMON_DATA { - INT sbrHdrBits; /**< number of SBR header bits */ - INT sbrDataBits; /**< number of SBR data bits */ - INT sbrFillBits; /**< number of SBR fill bits */ - FDK_BITSTREAM sbrBitbuf; /**< the SBR data bitbuffer */ - FDK_BITSTREAM tmpWriteBitbuf; /**< helper var for writing header*/ - INT xOverFreq; /**< the SBR crossover frequency */ - INT dynBwEnabled; /**< indicates if dynamic bandwidth is enabled */ - INT sbrNumChannels; /**< number of channels (meaning mono or stereo) */ - INT dynXOverFreqEnc; /**< encoder dynamic crossover frequency */ + INT sbrHdrBits; /**< number of SBR header bits */ + INT sbrDataBits; /**< number of SBR data bits */ + INT sbrFillBits; /**< number of SBR fill bits */ + FDK_BITSTREAM sbrBitbuf; /**< the SBR data bitbuffer */ + FDK_BITSTREAM tmpWriteBitbuf; /**< helper var for writing header*/ + INT xOverFreq; /**< the SBR crossover frequency */ + INT dynBwEnabled; /**< indicates if dynamic bandwidth is enabled */ + INT sbrNumChannels; /**< number of channels (meaning mono or stereo) */ + INT dynXOverFreqEnc; /**< encoder dynamic crossover frequency */ }; typedef struct COMMON_DATA *HANDLE_COMMON_DATA; - - #endif diff --git a/libSBRenc/src/code_env.cpp b/libSBRenc/src/code_env.cpp index e1a28d5..fb0f6a4 100644 --- a/libSBRenc/src/code_env.cpp +++ b/libSBRenc/src/code_env.cpp @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,10 +90,18 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ #include "code_env.h" -#include "sbr_rom.h" +#include "sbrenc_rom.h" /***************************************************************************** @@ -93,100 +112,98 @@ amm-info@iis.fraunhofer.de output: *****************************************************************************/ -INT -FDKsbrEnc_InitSbrHuffmanTables (HANDLE_SBR_ENV_DATA sbrEnvData, - HANDLE_SBR_CODE_ENVELOPE henv, - HANDLE_SBR_CODE_ENVELOPE hnoise, - AMP_RES amp_res) -{ - if ( (!henv) || (!hnoise) || (!sbrEnvData) ) - return (1); /* not init. */ +INT FDKsbrEnc_InitSbrHuffmanTables(HANDLE_SBR_ENV_DATA sbrEnvData, + HANDLE_SBR_CODE_ENVELOPE henv, + HANDLE_SBR_CODE_ENVELOPE hnoise, + AMP_RES amp_res) { + if ((!henv) || (!hnoise) || (!sbrEnvData)) return (1); /* not init. */ sbrEnvData->init_sbr_amp_res = amp_res; switch (amp_res) { - case SBR_AMP_RES_3_0: - /*envelope data*/ - - /*Level/Pan - coding */ - sbrEnvData->hufftableLevelTimeC = v_Huff_envelopeLevelC11T; - sbrEnvData->hufftableLevelTimeL = v_Huff_envelopeLevelL11T; - sbrEnvData->hufftableBalanceTimeC = bookSbrEnvBalanceC11T; - sbrEnvData->hufftableBalanceTimeL = bookSbrEnvBalanceL11T; - - sbrEnvData->hufftableLevelFreqC = v_Huff_envelopeLevelC11F; - sbrEnvData->hufftableLevelFreqL = v_Huff_envelopeLevelL11F; - sbrEnvData->hufftableBalanceFreqC = bookSbrEnvBalanceC11F; - sbrEnvData->hufftableBalanceFreqL = bookSbrEnvBalanceL11F; - - /*Right/Left - coding */ - sbrEnvData->hufftableTimeC = v_Huff_envelopeLevelC11T; - sbrEnvData->hufftableTimeL = v_Huff_envelopeLevelL11T; - sbrEnvData->hufftableFreqC = v_Huff_envelopeLevelC11F; - sbrEnvData->hufftableFreqL = v_Huff_envelopeLevelL11F; - - sbrEnvData->codeBookScfLavBalance = CODE_BOOK_SCF_LAV_BALANCE11; - sbrEnvData->codeBookScfLav = CODE_BOOK_SCF_LAV11; - - sbrEnvData->si_sbr_start_env_bits = SI_SBR_START_ENV_BITS_AMP_RES_3_0; - sbrEnvData->si_sbr_start_env_bits_balance = SI_SBR_START_ENV_BITS_BALANCE_AMP_RES_3_0; - break; - - case SBR_AMP_RES_1_5: - /*envelope data*/ - - /*Level/Pan - coding */ - sbrEnvData->hufftableLevelTimeC = v_Huff_envelopeLevelC10T; - sbrEnvData->hufftableLevelTimeL = v_Huff_envelopeLevelL10T; - sbrEnvData->hufftableBalanceTimeC = bookSbrEnvBalanceC10T; - sbrEnvData->hufftableBalanceTimeL = bookSbrEnvBalanceL10T; - - sbrEnvData->hufftableLevelFreqC = v_Huff_envelopeLevelC10F; - sbrEnvData->hufftableLevelFreqL = v_Huff_envelopeLevelL10F; - sbrEnvData->hufftableBalanceFreqC = bookSbrEnvBalanceC10F; - sbrEnvData->hufftableBalanceFreqL = bookSbrEnvBalanceL10F; - - /*Right/Left - coding */ - sbrEnvData->hufftableTimeC = v_Huff_envelopeLevelC10T; - sbrEnvData->hufftableTimeL = v_Huff_envelopeLevelL10T; - sbrEnvData->hufftableFreqC = v_Huff_envelopeLevelC10F; - sbrEnvData->hufftableFreqL = v_Huff_envelopeLevelL10F; - - sbrEnvData->codeBookScfLavBalance = CODE_BOOK_SCF_LAV_BALANCE10; - sbrEnvData->codeBookScfLav = CODE_BOOK_SCF_LAV10; - - sbrEnvData->si_sbr_start_env_bits = SI_SBR_START_ENV_BITS_AMP_RES_1_5; - sbrEnvData->si_sbr_start_env_bits_balance = SI_SBR_START_ENV_BITS_BALANCE_AMP_RES_1_5; - break; - - default: - return (1); /* undefined amp_res mode */ + case SBR_AMP_RES_3_0: + /*envelope data*/ + + /*Level/Pan - coding */ + sbrEnvData->hufftableLevelTimeC = v_Huff_envelopeLevelC11T; + sbrEnvData->hufftableLevelTimeL = v_Huff_envelopeLevelL11T; + sbrEnvData->hufftableBalanceTimeC = bookSbrEnvBalanceC11T; + sbrEnvData->hufftableBalanceTimeL = bookSbrEnvBalanceL11T; + + sbrEnvData->hufftableLevelFreqC = v_Huff_envelopeLevelC11F; + sbrEnvData->hufftableLevelFreqL = v_Huff_envelopeLevelL11F; + sbrEnvData->hufftableBalanceFreqC = bookSbrEnvBalanceC11F; + sbrEnvData->hufftableBalanceFreqL = bookSbrEnvBalanceL11F; + + /*Right/Left - coding */ + sbrEnvData->hufftableTimeC = v_Huff_envelopeLevelC11T; + sbrEnvData->hufftableTimeL = v_Huff_envelopeLevelL11T; + sbrEnvData->hufftableFreqC = v_Huff_envelopeLevelC11F; + sbrEnvData->hufftableFreqL = v_Huff_envelopeLevelL11F; + + sbrEnvData->codeBookScfLavBalance = CODE_BOOK_SCF_LAV_BALANCE11; + sbrEnvData->codeBookScfLav = CODE_BOOK_SCF_LAV11; + + sbrEnvData->si_sbr_start_env_bits = SI_SBR_START_ENV_BITS_AMP_RES_3_0; + sbrEnvData->si_sbr_start_env_bits_balance = + SI_SBR_START_ENV_BITS_BALANCE_AMP_RES_3_0; + break; + + case SBR_AMP_RES_1_5: + /*envelope data*/ + + /*Level/Pan - coding */ + sbrEnvData->hufftableLevelTimeC = v_Huff_envelopeLevelC10T; + sbrEnvData->hufftableLevelTimeL = v_Huff_envelopeLevelL10T; + sbrEnvData->hufftableBalanceTimeC = bookSbrEnvBalanceC10T; + sbrEnvData->hufftableBalanceTimeL = bookSbrEnvBalanceL10T; + + sbrEnvData->hufftableLevelFreqC = v_Huff_envelopeLevelC10F; + sbrEnvData->hufftableLevelFreqL = v_Huff_envelopeLevelL10F; + sbrEnvData->hufftableBalanceFreqC = bookSbrEnvBalanceC10F; + sbrEnvData->hufftableBalanceFreqL = bookSbrEnvBalanceL10F; + + /*Right/Left - coding */ + sbrEnvData->hufftableTimeC = v_Huff_envelopeLevelC10T; + sbrEnvData->hufftableTimeL = v_Huff_envelopeLevelL10T; + sbrEnvData->hufftableFreqC = v_Huff_envelopeLevelC10F; + sbrEnvData->hufftableFreqL = v_Huff_envelopeLevelL10F; + + sbrEnvData->codeBookScfLavBalance = CODE_BOOK_SCF_LAV_BALANCE10; + sbrEnvData->codeBookScfLav = CODE_BOOK_SCF_LAV10; + + sbrEnvData->si_sbr_start_env_bits = SI_SBR_START_ENV_BITS_AMP_RES_1_5; + sbrEnvData->si_sbr_start_env_bits_balance = + SI_SBR_START_ENV_BITS_BALANCE_AMP_RES_1_5; + break; + + default: + return (1); /* undefined amp_res mode */ } /* these are common to both amp_res values */ /*Noise data*/ /*Level/Pan - coding */ - sbrEnvData->hufftableNoiseLevelTimeC = v_Huff_NoiseLevelC11T; - sbrEnvData->hufftableNoiseLevelTimeL = v_Huff_NoiseLevelL11T; + sbrEnvData->hufftableNoiseLevelTimeC = v_Huff_NoiseLevelC11T; + sbrEnvData->hufftableNoiseLevelTimeL = v_Huff_NoiseLevelL11T; sbrEnvData->hufftableNoiseBalanceTimeC = bookSbrNoiseBalanceC11T; sbrEnvData->hufftableNoiseBalanceTimeL = bookSbrNoiseBalanceL11T; - sbrEnvData->hufftableNoiseLevelFreqC = v_Huff_envelopeLevelC11F; - sbrEnvData->hufftableNoiseLevelFreqL = v_Huff_envelopeLevelL11F; + sbrEnvData->hufftableNoiseLevelFreqC = v_Huff_envelopeLevelC11F; + sbrEnvData->hufftableNoiseLevelFreqL = v_Huff_envelopeLevelL11F; sbrEnvData->hufftableNoiseBalanceFreqC = bookSbrEnvBalanceC11F; sbrEnvData->hufftableNoiseBalanceFreqL = bookSbrEnvBalanceL11F; - /*Right/Left - coding */ - sbrEnvData->hufftableNoiseTimeC = v_Huff_NoiseLevelC11T; - sbrEnvData->hufftableNoiseTimeL = v_Huff_NoiseLevelL11T; - sbrEnvData->hufftableNoiseFreqC = v_Huff_envelopeLevelC11F; - sbrEnvData->hufftableNoiseFreqL = v_Huff_envelopeLevelL11F; - - sbrEnvData->si_sbr_start_noise_bits = SI_SBR_START_NOISE_BITS_AMP_RES_3_0; - sbrEnvData->si_sbr_start_noise_bits_balance = SI_SBR_START_NOISE_BITS_BALANCE_AMP_RES_3_0; + sbrEnvData->hufftableNoiseTimeC = v_Huff_NoiseLevelC11T; + sbrEnvData->hufftableNoiseTimeL = v_Huff_NoiseLevelL11T; + sbrEnvData->hufftableNoiseFreqC = v_Huff_envelopeLevelC11F; + sbrEnvData->hufftableNoiseFreqL = v_Huff_envelopeLevelL11F; + sbrEnvData->si_sbr_start_noise_bits = SI_SBR_START_NOISE_BITS_AMP_RES_3_0; + sbrEnvData->si_sbr_start_noise_bits_balance = + SI_SBR_START_NOISE_BITS_BALANCE_AMP_RES_3_0; /* init envelope tables and codebooks */ henv->codeBookScfLavBalanceTime = sbrEnvData->codeBookScfLavBalance; @@ -209,7 +226,6 @@ FDKsbrEnc_InitSbrHuffmanTables (HANDLE_SBR_ENV_DATA sbrEnvData, henv->start_bits = sbrEnvData->si_sbr_start_env_bits; henv->start_bits_balance = sbrEnvData->si_sbr_start_env_bits_balance; - /* init noise tables and codebooks */ hnoise->codeBookScfLavBalanceTime = CODE_BOOK_SCF_LAV_BALANCE11; @@ -226,14 +242,14 @@ FDKsbrEnc_InitSbrHuffmanTables (HANDLE_SBR_ENV_DATA sbrEnvData, hnoise->hufftableBalanceFreqL = sbrEnvData->hufftableNoiseBalanceFreqL; hnoise->hufftableFreqL = sbrEnvData->hufftableNoiseFreqL; - hnoise->start_bits = sbrEnvData->si_sbr_start_noise_bits; hnoise->start_bits_balance = sbrEnvData->si_sbr_start_noise_bits_balance; - /* No delta coding in time from the previous frame due to 1.5dB FIx-FIX rule */ + /* No delta coding in time from the previous frame due to 1.5dB FIx-FIX rule + */ henv->upDate = 0; hnoise->upDate = 0; - return (0); + return (0); } /******************************************************************************* @@ -248,33 +264,24 @@ FDKsbrEnc_InitSbrHuffmanTables (HANDLE_SBR_ENV_DATA sbrEnvData, Return: INT *******************************************************************************/ -static INT indexLow2High(INT offset, INT index, FREQ_RES res) -{ - - if(res == FREQ_RES_LOW) - { - 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); +static INT indexLow2High(INT offset, INT index, FREQ_RES res) { + if (res == FREQ_RES_LOW) { + 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); + } else + return (index); } - - /******************************************************************************* Functionname: mapLowResEnergyVal ******************************************************************************* @@ -286,43 +293,31 @@ static INT indexLow2High(INT offset, INT index, FREQ_RES res) Return: none *******************************************************************************/ -static void mapLowResEnergyVal(SCHAR currVal, SCHAR* prevData, INT offset, INT index, FREQ_RES res) -{ - - if(res == FREQ_RES_LOW) - { - 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; - } +static void mapLowResEnergyVal(SCHAR currVal, SCHAR *prevData, INT offset, + INT index, FREQ_RES res) { + if (res == FREQ_RES_LOW) { + 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 + } else prevData[index] = currVal; } - - /******************************************************************************* Functionname: computeBits ******************************************************************************* @@ -338,36 +333,31 @@ static void mapLowResEnergyVal(SCHAR currVal, SCHAR* prevData, INT offset, INT i Return: INT *******************************************************************************/ -static INT -computeBits (SCHAR *delta, - INT codeBookScfLavLevel, - INT codeBookScfLavBalance, - const UCHAR * hufftableLevel, - const UCHAR * hufftableBalance, INT coupling, INT channel) -{ +static INT computeBits(SCHAR *delta, INT codeBookScfLavLevel, + INT codeBookScfLavBalance, const UCHAR *hufftableLevel, + const UCHAR *hufftableBalance, INT coupling, + INT channel) { INT index; INT delta_bits = 0; if (coupling) { - if (channel == 1) - { - if (*delta < 0) - index = fixMax(*delta, -codeBookScfLavBalance); - else - index = fixMin(*delta, codeBookScfLavBalance); - - if (index != *delta) { - *delta = index; - return (10000); - } + if (channel == 1) { + if (*delta < 0) + index = fixMax(*delta, -codeBookScfLavBalance); + else + index = fixMin(*delta, codeBookScfLavBalance); - delta_bits = hufftableBalance[index + codeBookScfLavBalance]; + if (index != *delta) { + *delta = index; + return (10000); } - else { + + delta_bits = hufftableBalance[index + codeBookScfLavBalance]; + } else { if (*delta < 0) index = fixMax(*delta, -codeBookScfLavLevel); else - index = fixMin(*delta, codeBookScfLavLevel); + index = fixMin(*delta, codeBookScfLavLevel); if (index != *delta) { *delta = index; @@ -375,12 +365,11 @@ computeBits (SCHAR *delta, } delta_bits = hufftableLevel[index + codeBookScfLavLevel]; } - } - else { + } else { if (*delta < 0) index = fixMax(*delta, -codeBookScfLavLevel); else - index = fixMin(*delta, codeBookScfLavLevel); + index = fixMin(*delta, codeBookScfLavLevel); if (index != *delta) { *delta = index; @@ -392,9 +381,6 @@ computeBits (SCHAR *delta, return (delta_bits); } - - - /******************************************************************************* Functionname: FDKsbrEnc_codeEnvelope ******************************************************************************* @@ -414,18 +400,12 @@ computeBits (SCHAR *delta, *directionVec is modified *******************************************************************************/ -void -FDKsbrEnc_codeEnvelope(SCHAR *sfb_nrg, - const FREQ_RES *freq_res, - SBR_CODE_ENVELOPE *h_sbrCodeEnvelope, - INT *directionVec, - INT coupling, - INT nEnvelopes, - INT channel, - INT headerActive) -{ +void FDKsbrEnc_codeEnvelope(SCHAR *sfb_nrg, const FREQ_RES *freq_res, + SBR_CODE_ENVELOPE *h_sbrCodeEnvelope, + INT *directionVec, INT coupling, INT nEnvelopes, + INT channel, INT headerActive) { INT i, no_of_bands, band; - FIXP_DBL tmp1,tmp2,tmp3,dF_edge_1stEnv; + FIXP_DBL tmp1, tmp2, tmp3, dF_edge_1stEnv; SCHAR *ptr_nrg; INT codeBookScfLavLevelTime; @@ -447,9 +427,10 @@ FDKsbrEnc_codeEnvelope(SCHAR *sfb_nrg, SCHAR delta_T[MAX_FREQ_COEFFS]; SCHAR last_nrg, curr_nrg; - tmp1 = FL2FXCONST_DBL(0.5f) >> (DFRACT_BITS-16-1); - tmp2 = h_sbrCodeEnvelope->dF_edge_1stEnv >> (DFRACT_BITS-16); - tmp3 = (FIXP_DBL)(((INT)(LONG)h_sbrCodeEnvelope->dF_edge_incr*h_sbrCodeEnvelope->dF_edge_incr_fac) >> (DFRACT_BITS-16)); + tmp1 = FL2FXCONST_DBL(0.5f) >> (DFRACT_BITS - 16 - 1); + tmp2 = h_sbrCodeEnvelope->dF_edge_1stEnv >> (DFRACT_BITS - 16); + tmp3 = (FIXP_DBL)fMult(h_sbrCodeEnvelope->dF_edge_incr, + ((FIXP_DBL)h_sbrCodeEnvelope->dF_edge_incr_fac) << 15); dF_edge_1stEnv = tmp1 + tmp2 + tmp3; @@ -462,8 +443,7 @@ FDKsbrEnc_codeEnvelope(SCHAR *sfb_nrg, hufftableBalanceTimeL = h_sbrCodeEnvelope->hufftableBalanceTimeL; hufftableLevelFreqL = h_sbrCodeEnvelope->hufftableLevelFreqL; hufftableBalanceFreqL = h_sbrCodeEnvelope->hufftableBalanceFreqL; - } - else { + } else { codeBookScfLavLevelTime = h_sbrCodeEnvelope->codeBookScfLavTime; codeBookScfLavLevelFreq = h_sbrCodeEnvelope->codeBookScfLavFreq; codeBookScfLavBalanceTime = h_sbrCodeEnvelope->codeBookScfLavTime; @@ -474,28 +454,23 @@ FDKsbrEnc_codeEnvelope(SCHAR *sfb_nrg, hufftableBalanceFreqL = h_sbrCodeEnvelope->hufftableFreqL; } - if(coupling == 1 && channel == 1) - envDataTableCompFactor = 1; /*should be one when the new huffman-tables are ready*/ + if (coupling == 1 && channel == 1) + envDataTableCompFactor = + 1; /*should be one when the new huffman-tables are ready*/ else envDataTableCompFactor = 0; - - if (h_sbrCodeEnvelope->deltaTAcrossFrames == 0) - h_sbrCodeEnvelope->upDate = 0; + if (h_sbrCodeEnvelope->deltaTAcrossFrames == 0) h_sbrCodeEnvelope->upDate = 0; /* no delta coding in time in case of a header */ - if (headerActive) - h_sbrCodeEnvelope->upDate = 0; - + if (headerActive) h_sbrCodeEnvelope->upDate = 0; - for (i = 0; i < nEnvelopes; i++) - { + for (i = 0; i < nEnvelopes; i++) { if (freq_res[i] == FREQ_RES_HIGH) no_of_bands = h_sbrCodeEnvelope->nSfb[FREQ_RES_HIGH]; else no_of_bands = h_sbrCodeEnvelope->nSfb[FREQ_RES_LOW]; - ptr_nrg = sfb_nrg; curr_nrg = *ptr_nrg; @@ -506,107 +481,96 @@ FDKsbrEnc_codeEnvelope(SCHAR *sfb_nrg, else delta_F_bits = h_sbrCodeEnvelope->start_bits; + if (h_sbrCodeEnvelope->upDate != 0) { + delta_T[0] = (curr_nrg - h_sbrCodeEnvelope->sfb_nrg_prev[0]) >> + envDataTableCompFactor; - if(h_sbrCodeEnvelope->upDate != 0) - { - delta_T[0] = (curr_nrg - h_sbrCodeEnvelope->sfb_nrg_prev[0]) >> envDataTableCompFactor; - - delta_T_bits = computeBits (&delta_T[0], - codeBookScfLavLevelTime, - codeBookScfLavBalanceTime, - hufftableLevelTimeL, - hufftableBalanceTimeL, coupling, channel); + delta_T_bits = computeBits(&delta_T[0], codeBookScfLavLevelTime, + codeBookScfLavBalanceTime, hufftableLevelTimeL, + hufftableBalanceTimeL, coupling, channel); } - - mapLowResEnergyVal(curr_nrg, h_sbrCodeEnvelope->sfb_nrg_prev, offset, 0, freq_res[i]); + mapLowResEnergyVal(curr_nrg, h_sbrCodeEnvelope->sfb_nrg_prev, offset, 0, + freq_res[i]); /* ensure that nrg difference is not higher than codeBookScfLavXXXFreq */ - if ( coupling && channel == 1 ) { + if (coupling && channel == 1) { for (band = no_of_bands - 1; band > 0; band--) { - if ( ptr_nrg[band] - ptr_nrg[band-1] > codeBookScfLavBalanceFreq ) { - ptr_nrg[band-1] = ptr_nrg[band] - codeBookScfLavBalanceFreq; + if (ptr_nrg[band] - ptr_nrg[band - 1] > codeBookScfLavBalanceFreq) { + ptr_nrg[band - 1] = ptr_nrg[band] - codeBookScfLavBalanceFreq; } } for (band = 1; band < no_of_bands; band++) { - if ( ptr_nrg[band-1] - ptr_nrg[band] > codeBookScfLavBalanceFreq ) { - ptr_nrg[band] = ptr_nrg[band-1] - codeBookScfLavBalanceFreq; + if (ptr_nrg[band - 1] - ptr_nrg[band] > codeBookScfLavBalanceFreq) { + ptr_nrg[band] = ptr_nrg[band - 1] - codeBookScfLavBalanceFreq; } } - } - else { + } else { for (band = no_of_bands - 1; band > 0; band--) { - if ( ptr_nrg[band] - ptr_nrg[band-1] > codeBookScfLavLevelFreq ) { - ptr_nrg[band-1] = ptr_nrg[band] - codeBookScfLavLevelFreq; + if (ptr_nrg[band] - ptr_nrg[band - 1] > codeBookScfLavLevelFreq) { + ptr_nrg[band - 1] = ptr_nrg[band] - codeBookScfLavLevelFreq; } } for (band = 1; band < no_of_bands; band++) { - if ( ptr_nrg[band-1] - ptr_nrg[band] > codeBookScfLavLevelFreq ) { - ptr_nrg[band] = ptr_nrg[band-1] - codeBookScfLavLevelFreq; + if (ptr_nrg[band - 1] - ptr_nrg[band] > codeBookScfLavLevelFreq) { + ptr_nrg[band] = ptr_nrg[band - 1] - codeBookScfLavLevelFreq; } } } - /* Coding loop*/ - for (band = 1; band < no_of_bands; band++) - { + for (band = 1; band < no_of_bands; band++) { last_nrg = (*ptr_nrg); ptr_nrg++; curr_nrg = (*ptr_nrg); delta_F[band] = (curr_nrg - last_nrg) >> envDataTableCompFactor; - delta_F_bits += computeBits (&delta_F[band], - codeBookScfLavLevelFreq, - codeBookScfLavBalanceFreq, - hufftableLevelFreqL, - hufftableBalanceFreqL, coupling, channel); + delta_F_bits += computeBits( + &delta_F[band], codeBookScfLavLevelFreq, codeBookScfLavBalanceFreq, + hufftableLevelFreqL, hufftableBalanceFreqL, coupling, channel); - if(h_sbrCodeEnvelope->upDate != 0) - { - delta_T[band] = curr_nrg - h_sbrCodeEnvelope->sfb_nrg_prev[indexLow2High(offset, band, freq_res[i])]; + if (h_sbrCodeEnvelope->upDate != 0) { + delta_T[band] = + curr_nrg - + h_sbrCodeEnvelope + ->sfb_nrg_prev[indexLow2High(offset, band, freq_res[i])]; delta_T[band] = delta_T[band] >> envDataTableCompFactor; } - mapLowResEnergyVal(curr_nrg, h_sbrCodeEnvelope->sfb_nrg_prev, offset, band, freq_res[i]); + mapLowResEnergyVal(curr_nrg, h_sbrCodeEnvelope->sfb_nrg_prev, offset, + band, freq_res[i]); - if(h_sbrCodeEnvelope->upDate != 0) - { - delta_T_bits += computeBits (&delta_T[band], - codeBookScfLavLevelTime, - codeBookScfLavBalanceTime, - hufftableLevelTimeL, - hufftableBalanceTimeL, coupling, channel); + if (h_sbrCodeEnvelope->upDate != 0) { + delta_T_bits += computeBits( + &delta_T[band], codeBookScfLavLevelTime, codeBookScfLavBalanceTime, + hufftableLevelTimeL, hufftableBalanceTimeL, coupling, channel); } } /* Replace sfb_nrg with deltacoded samples and set flag */ if (i == 0) { INT tmp_bits; - tmp_bits = (((delta_T_bits * dF_edge_1stEnv) >> (DFRACT_BITS-18)) + (FIXP_DBL)1) >> 1; + tmp_bits = (((delta_T_bits * dF_edge_1stEnv) >> (DFRACT_BITS - 18)) + + (FIXP_DBL)1) >> + 1; use_dT = (h_sbrCodeEnvelope->upDate != 0 && (delta_F_bits > tmp_bits)); - } - else + } else use_dT = (delta_T_bits < delta_F_bits && h_sbrCodeEnvelope->upDate != 0); - if (use_dT) - { + if (use_dT) { directionVec[i] = TIME; - FDKmemcpy (sfb_nrg, delta_T, no_of_bands * sizeof (SCHAR)); - } - else { + FDKmemcpy(sfb_nrg, delta_T, no_of_bands * sizeof(SCHAR)); + } else { h_sbrCodeEnvelope->upDate = 0; directionVec[i] = FREQ; - FDKmemcpy (sfb_nrg, delta_F, no_of_bands * sizeof (SCHAR)); + FDKmemcpy(sfb_nrg, delta_F, no_of_bands * sizeof(SCHAR)); } sfb_nrg += no_of_bands; h_sbrCodeEnvelope->upDate = 1; } - } - /******************************************************************************* Functionname: FDKsbrEnc_InitSbrCodeEnvelope ******************************************************************************* @@ -618,15 +582,11 @@ FDKsbrEnc_codeEnvelope(SCHAR *sfb_nrg, Return: *******************************************************************************/ -INT -FDKsbrEnc_InitSbrCodeEnvelope (HANDLE_SBR_CODE_ENVELOPE h_sbrCodeEnvelope, - INT *nSfb, - INT deltaTAcrossFrames, - FIXP_DBL dF_edge_1stEnv, - FIXP_DBL dF_edge_incr) -{ - - FDKmemclear(h_sbrCodeEnvelope,sizeof(SBR_CODE_ENVELOPE)); +INT FDKsbrEnc_InitSbrCodeEnvelope(HANDLE_SBR_CODE_ENVELOPE h_sbrCodeEnvelope, + INT *nSfb, INT deltaTAcrossFrames, + FIXP_DBL dF_edge_1stEnv, + FIXP_DBL dF_edge_incr) { + FDKmemclear(h_sbrCodeEnvelope, sizeof(SBR_CODE_ENVELOPE)); h_sbrCodeEnvelope->deltaTAcrossFrames = deltaTAcrossFrames; h_sbrCodeEnvelope->dF_edge_1stEnv = dF_edge_1stEnv; @@ -635,7 +595,8 @@ FDKsbrEnc_InitSbrCodeEnvelope (HANDLE_SBR_CODE_ENVELOPE h_sbrCodeEnvelope, h_sbrCodeEnvelope->upDate = 0; h_sbrCodeEnvelope->nSfb[FREQ_RES_LOW] = nSfb[FREQ_RES_LOW]; h_sbrCodeEnvelope->nSfb[FREQ_RES_HIGH] = nSfb[FREQ_RES_HIGH]; - h_sbrCodeEnvelope->offset = 2*h_sbrCodeEnvelope->nSfb[FREQ_RES_LOW] - h_sbrCodeEnvelope->nSfb[FREQ_RES_HIGH]; + h_sbrCodeEnvelope->offset = 2 * h_sbrCodeEnvelope->nSfb[FREQ_RES_LOW] - + h_sbrCodeEnvelope->nSfb[FREQ_RES_HIGH]; return (0); } diff --git a/libSBRenc/src/code_env.h b/libSBRenc/src/code_env.h index 50a365e..673a783 100644 --- a/libSBRenc/src/code_env.h +++ b/libSBRenc/src/code_env.h @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,22 +90,29 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ /*! \file - \brief DPCM Envelope coding + \brief DPCM Envelope coding $Revision: 92790 $ */ -#ifndef __CODE_ENV_H -#define __CODE_ENV_H +#ifndef CODE_ENV_H +#define CODE_ENV_H #include "sbr_def.h" #include "bit_sbr.h" #include "fram_gen.h" -typedef struct -{ +typedef struct { INT offset; INT upDate; INT nSfb[2]; @@ -104,7 +122,6 @@ typedef struct FIXP_DBL dF_edge_incr; INT dF_edge_incr_fac; - INT codeBookScfLavTime; INT codeBookScfLavFreq; @@ -116,7 +133,6 @@ typedef struct INT start_bits; INT start_bits_balance; - const UCHAR *hufftableTimeL; const UCHAR *hufftableFreqL; @@ -124,30 +140,22 @@ typedef struct const UCHAR *hufftableBalanceTimeL; const UCHAR *hufftableLevelFreqL; const UCHAR *hufftableBalanceFreqL; -} -SBR_CODE_ENVELOPE; +} SBR_CODE_ENVELOPE; typedef SBR_CODE_ENVELOPE *HANDLE_SBR_CODE_ENVELOPE; +void FDKsbrEnc_codeEnvelope(SCHAR *sfb_nrg, const FREQ_RES *freq_res, + SBR_CODE_ENVELOPE *h_sbrCodeEnvelope, + INT *directionVec, INT coupling, INT nEnvelopes, + INT channel, INT headerActive); +INT FDKsbrEnc_InitSbrCodeEnvelope(HANDLE_SBR_CODE_ENVELOPE h_sbrCodeEnvelope, + INT *nSfb, INT deltaTAcrossFrames, + FIXP_DBL dF_edge_1stEnv, + FIXP_DBL dF_edge_incr); -void -FDKsbrEnc_codeEnvelope (SCHAR *sfb_nrg, - const FREQ_RES *freq_res, - SBR_CODE_ENVELOPE * h_sbrCodeEnvelope, - INT *directionVec, INT coupling, INT nEnvelopes, INT channel, - INT headerActive); - -INT -FDKsbrEnc_InitSbrCodeEnvelope (HANDLE_SBR_CODE_ENVELOPE h_sbrCodeEnvelope, - INT *nSfb, - INT deltaTAcrossFrames, - FIXP_DBL dF_edge_1stEnv, - FIXP_DBL dF_edge_incr); - -INT -FDKsbrEnc_InitSbrHuffmanTables (struct SBR_ENV_DATA* sbrEnvData, - HANDLE_SBR_CODE_ENVELOPE henv, - HANDLE_SBR_CODE_ENVELOPE hnoise, - AMP_RES amp_res); +INT FDKsbrEnc_InitSbrHuffmanTables(struct SBR_ENV_DATA *sbrEnvData, + HANDLE_SBR_CODE_ENVELOPE henv, + HANDLE_SBR_CODE_ENVELOPE hnoise, + AMP_RES amp_res); #endif diff --git a/libSBRenc/src/env_bit.cpp b/libSBRenc/src/env_bit.cpp index ea31183..41812ac 100644 --- a/libSBRenc/src/env_bit.cpp +++ b/libSBRenc/src/env_bit.cpp @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,7 +90,15 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ /*! \file @@ -89,13 +108,12 @@ amm-info@iis.fraunhofer.de #include "env_bit.h" #include "cmondata.h" - #ifndef min -#define min(a,b) ( a < b ? a:b) +#define min(a, b) (a < b ? a : b) #endif #ifndef max -#define max(a,b) ( a > b ? a:b) +#define max(a, b) (a > b ? a : b) #endif /* ***************************** crcAdvance **********************************/ @@ -107,27 +125,22 @@ amm-info@iis.fraunhofer.de * This function updates the crc register * */ -static void crcAdvance(USHORT crcPoly, - USHORT crcMask, - USHORT *crc, - ULONG bValue, - INT bBits - ) -{ +static void crcAdvance(USHORT crcPoly, USHORT crcMask, USHORT *crc, + ULONG bValue, INT bBits) { INT i; USHORT flag; - for (i=bBits-1; i>=0; i--) { - flag = ((*crc) & crcMask) ? (1) : (0) ; - flag ^= (bValue & (1<<i)) ? (1) : (0) ; + for (i = bBits - 1; i >= 0; i--) { + flag = ((*crc) & crcMask) ? (1) : (0); + flag ^= (bValue & (1 << i)) ? (1) : (0); - (*crc)<<=1; - if(flag) (*crc) ^= crcPoly; + (*crc) <<= 1; + if (flag) (*crc) ^= crcPoly; } } - -/* ***************************** FDKsbrEnc_InitSbrBitstream **********************************/ +/* ***************************** FDKsbrEnc_InitSbrBitstream + * **********************************/ /** * @fn * @brief Inittialisation of sbr bitstream, write of dummy header and CRC @@ -137,36 +150,35 @@ static void crcAdvance(USHORT crcPoly, * */ -INT FDKsbrEnc_InitSbrBitstream(HANDLE_COMMON_DATA hCmonData, - UCHAR *memoryBase, /*!< Pointer to bitstream buffer */ - INT memorySize, /*!< Length of bitstream buffer in bytes */ - HANDLE_FDK_CRCINFO hCrcInfo, - UINT sbrSyntaxFlags) /*!< SBR syntax flags */ +INT FDKsbrEnc_InitSbrBitstream( + HANDLE_COMMON_DATA hCmonData, + UCHAR *memoryBase, /*!< Pointer to bitstream buffer */ + INT memorySize, /*!< Length of bitstream buffer in bytes */ + HANDLE_FDK_CRCINFO hCrcInfo, UINT sbrSyntaxFlags) /*!< SBR syntax flags */ { INT crcRegion = 0; /* reset bit buffer */ FDKresetBitbuffer(&hCmonData->sbrBitbuf, BS_WRITER); - FDKinitBitStream(&hCmonData->tmpWriteBitbuf, memoryBase, - memorySize, 0, BS_WRITER); + FDKinitBitStream(&hCmonData->tmpWriteBitbuf, memoryBase, memorySize, 0, + BS_WRITER); if (sbrSyntaxFlags & SBR_SYNTAX_CRC) { - if (sbrSyntaxFlags & SBR_SYNTAX_DRM_CRC) - { /* Init and start CRC region */ - FDKwriteBits (&hCmonData->sbrBitbuf, 0x0, SI_SBR_DRM_CRC_BITS); - FDKcrcInit( hCrcInfo, 0x001d, 0xFFFF, SI_SBR_DRM_CRC_BITS ); - crcRegion = FDKcrcStartReg( hCrcInfo, &hCmonData->sbrBitbuf, 0 ); + if (sbrSyntaxFlags & SBR_SYNTAX_DRM_CRC) { /* Init and start CRC region */ + FDKwriteBits(&hCmonData->sbrBitbuf, 0x0, SI_SBR_DRM_CRC_BITS); + FDKcrcInit(hCrcInfo, 0x001d, 0xFFFF, SI_SBR_DRM_CRC_BITS); + crcRegion = FDKcrcStartReg(hCrcInfo, &hCmonData->sbrBitbuf, 0); } else { - FDKwriteBits (&hCmonData->sbrBitbuf, 0x0, SI_SBR_CRC_BITS); + FDKwriteBits(&hCmonData->sbrBitbuf, 0x0, SI_SBR_CRC_BITS); } } return (crcRegion); } - -/* ************************** FDKsbrEnc_AssembleSbrBitstream *******************************/ +/* ************************** FDKsbrEnc_AssembleSbrBitstream + * *******************************/ /** * @fn * @brief Formats the SBR payload @@ -176,48 +188,43 @@ INT FDKsbrEnc_InitSbrBitstream(HANDLE_COMMON_DATA hCmonData, * */ -void -FDKsbrEnc_AssembleSbrBitstream( HANDLE_COMMON_DATA hCmonData, - HANDLE_FDK_CRCINFO hCrcInfo, - INT crcRegion, - UINT sbrSyntaxFlags) -{ - USHORT crcReg = SBR_CRCINIT; - INT numCrcBits,i; +void FDKsbrEnc_AssembleSbrBitstream(HANDLE_COMMON_DATA hCmonData, + HANDLE_FDK_CRCINFO hCrcInfo, INT crcRegion, + UINT sbrSyntaxFlags) { + USHORT crcReg = SBR_CRCINIT; + INT numCrcBits, i; /* check if SBR is present */ - if ( hCmonData==NULL ) - return; + if (hCmonData == NULL) return; hCmonData->sbrFillBits = 0; /* Fill bits are written only for GA streams */ - if ( sbrSyntaxFlags & SBR_SYNTAX_DRM_CRC ) - { + if (sbrSyntaxFlags & SBR_SYNTAX_DRM_CRC) { /* * Calculate and write DRM CRC */ - FDKcrcEndReg( hCrcInfo, &hCmonData->sbrBitbuf, crcRegion ); - FDKwriteBits( &hCmonData->tmpWriteBitbuf, FDKcrcGetCRC(hCrcInfo)^0xFF, SI_SBR_DRM_CRC_BITS ); - } - else - { - if ( !(sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) ) - { - /* Do alignment here, because its defined as part of the sbr_extension_data */ + FDKcrcEndReg(hCrcInfo, &hCmonData->sbrBitbuf, crcRegion); + FDKwriteBits(&hCmonData->tmpWriteBitbuf, FDKcrcGetCRC(hCrcInfo) ^ 0xFF, + SI_SBR_DRM_CRC_BITS); + } else { + if (!(sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY)) { + /* Do alignment here, because its defined as part of the + * sbr_extension_data */ int sbrLoad = hCmonData->sbrHdrBits + hCmonData->sbrDataBits; - if ( sbrSyntaxFlags & SBR_SYNTAX_CRC ) { + if (sbrSyntaxFlags & SBR_SYNTAX_CRC) { sbrLoad += SI_SBR_CRC_BITS; } - sbrLoad += 4; /* Do byte Align with 4 bit offset. ISO/IEC 14496-3:2005(E) page 39. */ + sbrLoad += 4; /* Do byte Align with 4 bit offset. ISO/IEC 14496-3:2005(E) + page 39. */ hCmonData->sbrFillBits = (8 - (sbrLoad % 8)) % 8; /* append fill bits */ - FDKwriteBits(&hCmonData->sbrBitbuf, 0, hCmonData->sbrFillBits ); + FDKwriteBits(&hCmonData->sbrBitbuf, 0, hCmonData->sbrFillBits); FDK_ASSERT(FDKgetValidBits(&hCmonData->sbrBitbuf) % 8 == 4); } @@ -225,26 +232,26 @@ FDKsbrEnc_AssembleSbrBitstream( HANDLE_COMMON_DATA hCmonData, /* calculate crc */ - if ( sbrSyntaxFlags & SBR_SYNTAX_CRC ) { - FDK_BITSTREAM tmpCRCBuf = hCmonData->sbrBitbuf; - FDKresetBitbuffer( &tmpCRCBuf, BS_READER ); + if (sbrSyntaxFlags & SBR_SYNTAX_CRC) { + FDK_BITSTREAM tmpCRCBuf = hCmonData->sbrBitbuf; + FDKresetBitbuffer(&tmpCRCBuf, BS_READER); - numCrcBits = hCmonData->sbrHdrBits + hCmonData->sbrDataBits + hCmonData->sbrFillBits; + numCrcBits = hCmonData->sbrHdrBits + hCmonData->sbrDataBits + + hCmonData->sbrFillBits; - for(i=0;i<numCrcBits;i++){ + for (i = 0; i < numCrcBits; i++) { INT bit; - bit = FDKreadBits(&tmpCRCBuf,1); - crcAdvance(SBR_CRC_POLY,SBR_CRC_MASK,&crcReg,bit,1); + bit = FDKreadBits(&tmpCRCBuf, 1); + crcAdvance(SBR_CRC_POLY, SBR_CRC_MASK, &crcReg, bit, 1); } crcReg &= (SBR_CRC_RANGE); /* * Write CRC data. */ - FDKwriteBits (&hCmonData->tmpWriteBitbuf, crcReg, SI_SBR_CRC_BITS); + FDKwriteBits(&hCmonData->tmpWriteBitbuf, crcReg, SI_SBR_CRC_BITS); } } FDKsyncCache(&hCmonData->tmpWriteBitbuf); } - diff --git a/libSBRenc/src/env_bit.h b/libSBRenc/src/env_bit.h index 038a32a..b91802c 100644 --- a/libSBRenc/src/env_bit.h +++ b/libSBRenc/src/env_bit.h @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,48 +90,46 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ /*! \file \brief Remaining SBR Bit Writing Routines */ -#ifndef BIT_ENV_H -#define BIT_ENV_H +#ifndef ENV_BIT_H +#define ENV_BIT_H #include "sbr_encoder.h" #include "FDK_crc.h" /* 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_RANGE (0x03FF) -#define SBR_CRC_MAXREGS 1 -#define SBR_CRCINIT (0x0) - - -#define SI_SBR_CRC_ENABLE_BITS 0 -#define SI_SBR_CRC_BITS 10 -#define SI_SBR_DRM_CRC_BITS 8 +#define SBR_CRC_POLY (0x0233) +#define SBR_CRC_MASK (0x0200) +#define SBR_CRC_RANGE (0x03FF) +#define SBR_CRC_MAXREGS 1 +#define SBR_CRCINIT (0x0) +#define SI_SBR_CRC_ENABLE_BITS 0 +#define SI_SBR_CRC_BITS 10 +#define SI_SBR_DRM_CRC_BITS 8 struct COMMON_DATA; -INT FDKsbrEnc_InitSbrBitstream(struct COMMON_DATA *hCmonData, - UCHAR *memoryBase, - INT memorySize, - HANDLE_FDK_CRCINFO hCrcInfo, - UINT sbrSyntaxFlags); - -void -FDKsbrEnc_AssembleSbrBitstream (struct COMMON_DATA *hCmonData, - HANDLE_FDK_CRCINFO hCrcInfo, - INT crcReg, - UINT sbrSyntaxFlags); - - - +INT FDKsbrEnc_InitSbrBitstream(struct COMMON_DATA *hCmonData, UCHAR *memoryBase, + INT memorySize, HANDLE_FDK_CRCINFO hCrcInfo, + UINT sbrSyntaxFlags); +void FDKsbrEnc_AssembleSbrBitstream(struct COMMON_DATA *hCmonData, + HANDLE_FDK_CRCINFO hCrcInfo, INT crcRegion, + UINT sbrSyntaxFlags); -#endif /* #ifndef BIT_ENV_H */ +#endif /* #ifndef ENV_BIT_H */ diff --git a/libSBRenc/src/env_est.cpp b/libSBRenc/src/env_est.cpp index 4fcda51..0eb8425 100644 --- a/libSBRenc/src/env_est.cpp +++ b/libSBRenc/src/env_est.cpp @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,7 +90,15 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ #include "env_est.h" #include "tran_det.h" @@ -89,20 +108,18 @@ amm-info@iis.fraunhofer.de #include "fram_gen.h" #include "bit_sbr.h" #include "cmondata.h" -#include "sbr_ram.h" - +#include "sbrenc_ram.h" #include "genericStds.h" #define QUANT_ERROR_THRES 200 #define Y_NRG_SCALE 5 /* noCols = 32 -> shift(5) */ +#define MAX_NRG_SLOTS_LD 16 - -static const UCHAR panTable[2][10] = { { 0, 2, 4, 6, 8,12,16,20,24}, - { 0, 2, 4, 8,12, 0, 0, 0, 0 } }; +static const UCHAR panTable[2][10] = {{0, 2, 4, 6, 8, 12, 16, 20, 24}, + {0, 2, 4, 8, 12, 0, 0, 0, 0}}; static const UCHAR maxIndex[2] = {9, 5}; - /****************************************************************************** Functionname: FDKsbrEnc_GetTonality ******************************************************************************/ @@ -124,64 +141,64 @@ static const UCHAR maxIndex[2] = {9, 5}; scaled by 2^(RELAXATION_SHIFT+2)*RELAXATION_FRACT ****************************************************************************/ -static FIXP_DBL FDKsbrEnc_GetTonality( - const FIXP_DBL *const *quotaMatrix, - const INT noEstPerFrame, - const INT startIndex, - const FIXP_DBL *const *Energies, - const UCHAR startBand, - const INT stopBand, - const INT numberCols - ) -{ +static FIXP_DBL FDKsbrEnc_GetTonality(const FIXP_DBL *const *quotaMatrix, + const INT noEstPerFrame, + const INT startIndex, + const FIXP_DBL *const *Energies, + const UCHAR startBand, const INT stopBand, + const INT numberCols) { UCHAR b, e, k; - INT no_enMaxBand[SBR_MAX_ENERGY_VALUES] = { -1, -1, -1, -1, -1 }; - FIXP_DBL energyMax[SBR_MAX_ENERGY_VALUES] = { FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f) }; + INT no_enMaxBand[SBR_MAX_ENERGY_VALUES] = {-1, -1, -1, -1, -1}; + FIXP_DBL energyMax[SBR_MAX_ENERGY_VALUES] = { + FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), + FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f)}; FIXP_DBL energyMaxMin = MAXVAL_DBL; /* min. energy in energyMax array */ - UCHAR posEnergyMaxMin = 0; /* min. energy in energyMax array position */ - FIXP_DBL tonalityBand[SBR_MAX_ENERGY_VALUES] = { FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f) }; + UCHAR posEnergyMaxMin = 0; /* min. energy in energyMax array position */ + FIXP_DBL tonalityBand[SBR_MAX_ENERGY_VALUES] = { + FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), + FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f)}; FIXP_DBL globalTonality = FL2FXCONST_DBL(0.0f); - FIXP_DBL energyBand[QMF_CHANNELS]; - INT maxNEnergyValues; /* max. number of max. energy values */ + FIXP_DBL energyBand[64]; + INT maxNEnergyValues; /* max. number of max. energy values */ /*** Sum up energies for each band ***/ - FDK_ASSERT(numberCols==15||numberCols==16); + FDK_ASSERT(numberCols == 15 || numberCols == 16); /* numberCols is always 15 or 16 for ELD. In case of 16 bands, the energyBands are initialized with the [15]th column. The rest of the column energies are added in the next step. */ - if (numberCols==15) { - for (b=startBand; b<stopBand; b++) { - energyBand[b]=FL2FXCONST_DBL(0.0f); + if (numberCols == 15) { + for (b = startBand; b < stopBand; b++) { + energyBand[b] = FL2FXCONST_DBL(0.0f); } } else { - for (b=startBand; b<stopBand; b++) { - energyBand[b]=Energies[15][b]>>4; + for (b = startBand; b < stopBand; b++) { + energyBand[b] = Energies[15][b] >> 4; } } - for (k=0; k<15; k++) { - for (b=startBand; b<stopBand; b++) { - energyBand[b] += Energies[k][b]>>4; + for (k = 0; k < 15; k++) { + for (b = startBand; b < stopBand; b++) { + energyBand[b] += Energies[k][b] >> 4; } } /*** Determine 5 highest band-energies ***/ - maxNEnergyValues = fMin(SBR_MAX_ENERGY_VALUES, stopBand-startBand); + maxNEnergyValues = fMin(SBR_MAX_ENERGY_VALUES, stopBand - startBand); /* Get min. value in energyMax array */ energyMaxMin = energyMax[0] = energyBand[startBand]; no_enMaxBand[0] = startBand; posEnergyMaxMin = 0; - for (k=1; k<maxNEnergyValues; k++) { - energyMax[k] = energyBand[startBand+k]; - no_enMaxBand[k] = startBand+k; + for (k = 1; k < maxNEnergyValues; k++) { + energyMax[k] = energyBand[startBand + k]; + no_enMaxBand[k] = startBand + k; if (energyMaxMin > energyMax[k]) { energyMaxMin = energyMax[k]; posEnergyMaxMin = k; } } - for (b=startBand+maxNEnergyValues; b<stopBand; b++) { + for (b = startBand + maxNEnergyValues; b < stopBand; b++) { if (energyBand[b] > energyMaxMin) { energyMax[posEnergyMaxMin] = energyBand[b]; no_enMaxBand[posEnergyMaxMin] = b; @@ -189,7 +206,7 @@ static FIXP_DBL FDKsbrEnc_GetTonality( /* Again, get min. value in energyMax array */ energyMaxMin = energyMax[0]; posEnergyMaxMin = 0; - for (k=1; k<maxNEnergyValues; k++) { + for (k = 1; k < maxNEnergyValues; k++) { if (energyMaxMin > energyMax[k]) { energyMaxMin = energyMax[k]; posEnergyMaxMin = k; @@ -200,12 +217,13 @@ static FIXP_DBL FDKsbrEnc_GetTonality( /*** End determine 5 highest band-energies ***/ /* Get tonality values for 5 highest energies */ - for (e=0; e<maxNEnergyValues; e++) { - tonalityBand[e]=FL2FXCONST_DBL(0.0f); - for (k=0; k<noEstPerFrame; k++) { + for (e = 0; e < maxNEnergyValues; e++) { + tonalityBand[e] = FL2FXCONST_DBL(0.0f); + for (k = 0; k < noEstPerFrame; k++) { tonalityBand[e] += quotaMatrix[startIndex + k][no_enMaxBand[e]] >> 1; } - globalTonality += tonalityBand[e] >> 2; /* headroom of 2+1 (max. 5 additions) */ + globalTonality += + tonalityBand[e] >> 2; /* headroom of 2+1 (max. 5 additions) */ } return globalTonality; @@ -221,34 +239,36 @@ static FIXP_DBL FDKsbrEnc_GetTonality( ****************************************************************************/ LNK_SECTION_CODE_L1 -static void -FDKsbrEnc_getEnergyFromCplxQmfData(FIXP_DBL **RESTRICT energyValues,/*!< the result of the operation */ - FIXP_DBL **RESTRICT realValues, /*!< the real part of the QMF subsamples */ - FIXP_DBL **RESTRICT imagValues, /*!< the imaginary part of the QMF subsamples */ - INT numberBands, /*!< number of QMF bands */ - INT numberCols, /*!< number of QMF subsamples */ - INT *qmfScale, /*!< sclefactor of QMF subsamples */ - INT *energyScale) /*!< scalefactor of energies */ +static void FDKsbrEnc_getEnergyFromCplxQmfData( + FIXP_DBL **RESTRICT energyValues, /*!< the result of the operation */ + FIXP_DBL **RESTRICT realValues, /*!< the real part of the QMF subsamples */ + FIXP_DBL **RESTRICT + imagValues, /*!< the imaginary part of the QMF subsamples */ + INT numberBands, /*!< number of QMF bands */ + INT numberCols, /*!< number of QMF subsamples */ + INT *qmfScale, /*!< sclefactor of QMF subsamples */ + INT *energyScale) /*!< scalefactor of energies */ { int j, k; int scale; FIXP_DBL max_val = FL2FXCONST_DBL(0.0f); /* Get Scratch buffer */ - C_ALLOC_SCRATCH_START(tmpNrg, FIXP_DBL, QMF_CHANNELS*QMF_MAX_TIME_SLOTS/2); + C_ALLOC_SCRATCH_START(tmpNrg, FIXP_DBL, 32 * 64 / 2) /* Get max possible scaling of QMF data */ scale = DFRACT_BITS; - for (k=0; k<numberCols; k++) { - scale = fixMin(scale, fixMin(getScalefactor(realValues[k], numberBands), getScalefactor(imagValues[k], numberBands))); + for (k = 0; k < numberCols; k++) { + scale = fixMin(scale, fixMin(getScalefactor(realValues[k], numberBands), + getScalefactor(imagValues[k], numberBands))); } /* Tweak scaling stability for zero signal to non-zero signal transitions */ - if (scale >= DFRACT_BITS-1) { - scale = (FRACT_BITS-1-*qmfScale); + if (scale >= DFRACT_BITS - 1) { + scale = (FRACT_BITS - 1 - *qmfScale); } - /* prevent scaling of QFM values to -1.f */ - scale = fixMax(0,scale-1); + /* prevent scaling of QMF values to -1.f */ + scale = fixMax(0, scale - 1); /* Update QMF scale */ *qmfScale += scale; @@ -259,22 +279,23 @@ FDKsbrEnc_getEnergyFromCplxQmfData(FIXP_DBL **RESTRICT energyValues,/*!< the res */ { FIXP_DBL *nrgValues = tmpNrg; - for (k=0; k<numberCols; k+=2) - { + for (k = 0; k < numberCols; k += 2) { /* Load band vector addresses of 2 consecutive timeslots */ FIXP_DBL *RESTRICT r0 = realValues[k]; FIXP_DBL *RESTRICT i0 = imagValues[k]; - FIXP_DBL *RESTRICT r1 = realValues[k+1]; - FIXP_DBL *RESTRICT i1 = imagValues[k+1]; - for (j=0; j<numberBands; j++) - { - FIXP_DBL energy; - FIXP_DBL tr0,tr1,ti0,ti1; + FIXP_DBL *RESTRICT r1 = realValues[k + 1]; + FIXP_DBL *RESTRICT i1 = imagValues[k + 1]; + for (j = 0; j < numberBands; j++) { + FIXP_DBL energy; + FIXP_DBL tr0, tr1, ti0, ti1; /* Read QMF values of 2 timeslots */ - tr0 = r0[j]; tr1 = r1[j]; ti0 = i0[j]; ti1 = i1[j]; + tr0 = r0[j]; + tr1 = r1[j]; + ti0 = i0[j]; + ti1 = i1[j]; - /* Scale QMF Values and Calc Energy of both timeslots */ + /* Scale QMF Values and Calc Energy average of both timeslots */ tr0 <<= scale; ti0 <<= scale; energy = fPow2AddDiv2(fPow2Div2(tr0), ti0) >> 1; @@ -288,18 +309,23 @@ FDKsbrEnc_getEnergyFromCplxQmfData(FIXP_DBL **RESTRICT energyValues,/*!< the res max_val = fixMax(max_val, energy); /* Write back scaled QMF values */ - r0[j] = tr0; r1[j] = tr1; i0[j] = ti0; i1[j] = ti1; + r0[j] = tr0; + r1[j] = tr1; + i0[j] = ti0; + i1[j] = ti1; } } } /* energyScale: scalefactor energies of current frame */ - *energyScale = 2*(*qmfScale)-1; /* if qmfScale > 0: nr of right shifts otherwise nr of left shifts */ + *energyScale = + 2 * (*qmfScale) - + 1; /* if qmfScale > 0: nr of right shifts otherwise nr of left shifts */ /* Scale timeslot pair energies and write to output buffer */ scale = CountLeadingBits(max_val); { - FIXP_DBL *nrgValues = tmpNrg; - for (k=0; k<numberCols>>1; k++) { + FIXP_DBL *nrgValues = tmpNrg; + for (k = 0; k<numberCols>> 1; k++) { scaleValues(energyValues[k], nrgValues, numberBands, scale); nrgValues += numberBands; } @@ -307,41 +333,43 @@ FDKsbrEnc_getEnergyFromCplxQmfData(FIXP_DBL **RESTRICT energyValues,/*!< the res } /* Free Scratch buffer */ - C_ALLOC_SCRATCH_END(tmpNrg, FIXP_DBL, QMF_CHANNELS*QMF_MAX_TIME_SLOTS/2); + C_ALLOC_SCRATCH_END(tmpNrg, FIXP_DBL, 32 * 64 / 2) } LNK_SECTION_CODE_L1 -static void -FDKsbrEnc_getEnergyFromCplxQmfDataFull(FIXP_DBL **RESTRICT energyValues,/*!< the result of the operation */ - FIXP_DBL **RESTRICT realValues, /*!< the real part of the QMF subsamples */ - FIXP_DBL **RESTRICT imagValues, /*!< the imaginary part of the QMF subsamples */ - int numberBands, /*!< number of QMF bands */ - int numberCols, /*!< number of QMF subsamples */ - int *qmfScale, /*!< sclefactor of QMF subsamples */ - int *energyScale) /*!< scalefactor of energies */ +static void FDKsbrEnc_getEnergyFromCplxQmfDataFull( + FIXP_DBL **RESTRICT energyValues, /*!< the result of the operation */ + FIXP_DBL **RESTRICT realValues, /*!< the real part of the QMF subsamples */ + FIXP_DBL **RESTRICT + imagValues, /*!< the imaginary part of the QMF subsamples */ + int numberBands, /*!< number of QMF bands */ + int numberCols, /*!< number of QMF subsamples */ + int *qmfScale, /*!< scalefactor of QMF subsamples */ + int *energyScale) /*!< scalefactor of energies */ { int j, k; int scale; FIXP_DBL max_val = FL2FXCONST_DBL(0.0f); /* Get Scratch buffer */ - C_ALLOC_SCRATCH_START(tmpNrg, FIXP_DBL, QMF_MAX_TIME_SLOTS*QMF_CHANNELS/2); + C_ALLOC_SCRATCH_START(tmpNrg, FIXP_DBL, MAX_NRG_SLOTS_LD * 64) - FDK_ASSERT(numberBands <= QMF_CHANNELS); - FDK_ASSERT(numberCols <= QMF_MAX_TIME_SLOTS/2); + FDK_ASSERT(numberCols <= MAX_NRG_SLOTS_LD); + FDK_ASSERT(numberBands <= 64); /* Get max possible scaling of QMF data */ scale = DFRACT_BITS; - for (k=0; k<numberCols; k++) { - scale = fixMin(scale, fixMin(getScalefactor(realValues[k], numberBands), getScalefactor(imagValues[k], numberBands))); + for (k = 0; k < numberCols; k++) { + scale = fixMin(scale, fixMin(getScalefactor(realValues[k], numberBands), + getScalefactor(imagValues[k], numberBands))); } /* Tweak scaling stability for zero signal to non-zero signal transitions */ - if (scale >= DFRACT_BITS-1) { - scale = (FRACT_BITS-1-*qmfScale); + if (scale >= DFRACT_BITS - 1) { + scale = (FRACT_BITS - 1 - *qmfScale); } /* prevent scaling of QFM values to -1.f */ - scale = fixMax(0,scale-1); + scale = fixMax(0, scale - 1); /* Update QMF scale */ *qmfScale += scale; @@ -352,20 +380,19 @@ FDKsbrEnc_getEnergyFromCplxQmfDataFull(FIXP_DBL **RESTRICT energyValues,/*!< the */ { FIXP_DBL *nrgValues = tmpNrg; - for (k=0; k<numberCols; k++) - { - /* Load band vector addresses of 2 consecutive timeslots */ + for (k = 0; k < numberCols; k++) { + /* Load band vector addresses of 1 timeslot */ FIXP_DBL *RESTRICT r0 = realValues[k]; FIXP_DBL *RESTRICT i0 = imagValues[k]; - for (j=0; j<numberBands; j++) - { - FIXP_DBL energy; - FIXP_DBL tr0,ti0; + for (j = 0; j < numberBands; j++) { + FIXP_DBL energy; + FIXP_DBL tr0, ti0; - /* Read QMF values of 2 timeslots */ - tr0 = r0[j]; ti0 = i0[j]; + /* Read QMF values of 1 timeslot */ + tr0 = r0[j]; + ti0 = i0[j]; - /* Scale QMF Values and Calc Energy of both timeslots */ + /* Scale QMF Values and Calc Energy */ tr0 <<= scale; ti0 <<= scale; energy = fPow2AddDiv2(fPow2Div2(tr0), ti0); @@ -374,18 +401,21 @@ FDKsbrEnc_getEnergyFromCplxQmfDataFull(FIXP_DBL **RESTRICT energyValues,/*!< the max_val = fixMax(max_val, energy); /* Write back scaled QMF values */ - r0[j] = tr0; i0[j] = ti0; + r0[j] = tr0; + i0[j] = ti0; } } } /* energyScale: scalefactor energies of current frame */ - *energyScale = 2*(*qmfScale)-1; /* if qmfScale > 0: nr of right shifts otherwise nr of left shifts */ + *energyScale = + 2 * (*qmfScale) - + 1; /* if qmfScale > 0: nr of right shifts otherwise nr of left shifts */ /* Scale timeslot pair energies and write to output buffer */ scale = CountLeadingBits(max_val); { - FIXP_DBL *nrgValues = tmpNrg; - for (k=0; k<numberCols; k++) { + FIXP_DBL *nrgValues = tmpNrg; + for (k = 0; k < numberCols; k++) { scaleValues(energyValues[k], nrgValues, numberBands, scale); nrgValues += numberBands; } @@ -393,7 +423,7 @@ FDKsbrEnc_getEnergyFromCplxQmfDataFull(FIXP_DBL **RESTRICT energyValues,/*!< the } /* Free Scratch buffer */ - C_ALLOC_SCRATCH_END(tmpNrg, FIXP_DBL, QMF_MAX_TIME_SLOTS*QMF_CHANNELS/2); + C_ALLOC_SCRATCH_END(tmpNrg, FIXP_DBL, MAX_NRG_SLOTS_LD * 64) } /***************************************************************************/ @@ -404,12 +434,10 @@ FDKsbrEnc_getEnergyFromCplxQmfDataFull(FIXP_DBL **RESTRICT energyValues,/*!< the \return the quantized pan value ****************************************************************************/ -static INT -mapPanorama(INT nrgVal, /*! integer value of the energy */ - INT ampRes, /*! amplitude resolution [1.5/3dB] */ - INT *quantError /*! quantization error of energy val*/ - ) -{ +static INT mapPanorama(INT nrgVal, /*! integer value of the energy */ + INT ampRes, /*! amplitude resolution [1.5/3dB] */ + INT *quantError /*! quantization error of energy val*/ +) { int i; INT min_val, val; UCHAR panIndex; @@ -422,7 +450,7 @@ mapPanorama(INT nrgVal, /*! integer value of the energy */ min_val = FDK_INT_MAX; panIndex = 0; for (i = 0; i < maxIndex[ampRes]; i++) { - val = fixp_abs ((nrgVal - (INT)panTable[ampRes][i])); + val = fixp_abs((nrgVal - (INT)panTable[ampRes][i])); if (val < min_val) { min_val = val; @@ -430,12 +458,12 @@ mapPanorama(INT nrgVal, /*! integer value of the energy */ } } - *quantError=min_val; + *quantError = min_val; - return panTable[ampRes][maxIndex[ampRes]-1] + sign * panTable[ampRes][panIndex]; + return panTable[ampRes][maxIndex[ampRes] - 1] + + sign * panTable[ampRes][panIndex]; } - /***************************************************************************/ /*! @@ -444,34 +472,37 @@ mapPanorama(INT nrgVal, /*! integer value of the energy */ \return void ****************************************************************************/ -static void -sbrNoiseFloorLevelsQuantisation(SCHAR *RESTRICT iNoiseLevels, /*! quantized noise levels */ - FIXP_DBL *RESTRICT NoiseLevels, /*! the noise levels */ - INT coupling /*! the coupling flag */ - ) -{ +static void sbrNoiseFloorLevelsQuantisation( + SCHAR *RESTRICT iNoiseLevels, /*! quantized noise levels */ + FIXP_DBL *RESTRICT + NoiseLevels, /*! the noise levels. Exponent = LD_DATA_SHIFT */ + INT coupling /*! the coupling flag */ +) { INT i; INT tmp, dummy; /* Quantisation, similar to sfb quant... */ for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) { - /* tmp = NoiseLevels[i] > (PFLOAT)30.0f ? 30: (INT) (NoiseLevels[i] + (PFLOAT)0.5); */ - /* 30>>6 = 0.46875 */ + /* tmp = NoiseLevels[i] > (PFLOAT)30.0f ? 30: (INT) (NoiseLevels[i] + + * (PFLOAT)0.5); */ + /* 30>>LD_DATA_SHIFT = 0.46875 */ if ((FIXP_DBL)NoiseLevels[i] > FL2FXCONST_DBL(0.46875f)) { tmp = 30; - } - else { - /* tmp = (INT)((FIXP_DBL)NoiseLevels[i] + (FL2FXCONST_DBL(0.5f)>>(*/ /* FRACT_BITS+ */ /* 6-1)));*/ - /* tmp = tmp >> (DFRACT_BITS-1-6); */ /* conversion to integer happens here */ - /* rounding is done by shifting one bit less than necessary to the right, adding '1' and then shifting the final bit */ - tmp = ((((INT)NoiseLevels[i])>>(DFRACT_BITS-1-LD_DATA_SHIFT)) ); /* conversion to integer */ - if (tmp != 0) - tmp += 1; + } else { + /* tmp = (INT)((FIXP_DBL)NoiseLevels[i] + (FL2FXCONST_DBL(0.5f)>>(*/ + /* FRACT_BITS+ */ /* 6-1)));*/ + /* tmp = tmp >> (DFRACT_BITS-1-LD_DATA_SHIFT); */ /* conversion to integer + happens here */ + /* rounding is done by shifting one bit less than necessary to the right, + * adding '1' and then shifting the final bit */ + tmp = ((((INT)NoiseLevels[i]) >> + (DFRACT_BITS - 1 - LD_DATA_SHIFT))); /* conversion to integer */ + if (tmp != 0) tmp += 1; } if (coupling) { tmp = tmp < -30 ? -30 : tmp; - tmp = mapPanorama (tmp,1,&dummy); + tmp = mapPanorama(tmp, 1, &dummy); } iNoiseLevels[i] = tmp; } @@ -485,60 +516,76 @@ sbrNoiseFloorLevelsQuantisation(SCHAR *RESTRICT iNoiseLevels, /*! quantized n \return void ****************************************************************************/ -static void -coupleNoiseFloor(FIXP_DBL *RESTRICT noise_level_left, /*! noise level left (modified)*/ - FIXP_DBL *RESTRICT noise_level_right /*! noise level right (modified)*/ - ) -{ - FIXP_DBL cmpValLeft,cmpValRight; +static void coupleNoiseFloor( + FIXP_DBL *RESTRICT noise_level_left, /*! noise level left (modified)*/ + FIXP_DBL *RESTRICT noise_level_right /*! noise level right (modified)*/ +) { + FIXP_DBL cmpValLeft, cmpValRight; INT i; - FIXP_DBL temp1,temp2; + FIXP_DBL temp1, temp2; for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) { - /* Calculation of the power function using ld64: z = x^y; z' = CalcLd64(z) = y*CalcLd64(x)/64; z = CalcInvLd64(z'); */ - cmpValLeft = NOISE_FLOOR_OFFSET_64 - noise_level_left[i]; + cmpValLeft = NOISE_FLOOR_OFFSET_64 - noise_level_left[i]; cmpValRight = NOISE_FLOOR_OFFSET_64 - noise_level_right[i]; if (cmpValRight < FL2FXCONST_DBL(0.0f)) { temp1 = CalcInvLdData(NOISE_FLOOR_OFFSET_64 - noise_level_right[i]); - } - else { + } else { temp1 = CalcInvLdData(NOISE_FLOOR_OFFSET_64 - noise_level_right[i]); - temp1 = temp1 << (DFRACT_BITS-1-LD_DATA_SHIFT-1); /* INT to fract conversion of result, if input of CalcInvLdData is positiv */ + temp1 = temp1 << (DFRACT_BITS - 1 - LD_DATA_SHIFT - + 1); /* INT to fract conversion of result, if input of + CalcInvLdData is positiv */ } if (cmpValLeft < FL2FXCONST_DBL(0.0f)) { temp2 = CalcInvLdData(NOISE_FLOOR_OFFSET_64 - noise_level_left[i]); - } - else { + } else { temp2 = CalcInvLdData(NOISE_FLOOR_OFFSET_64 - noise_level_left[i]); - temp2 = temp2 << (DFRACT_BITS-1-LD_DATA_SHIFT-1); /* INT to fract conversion of result, if input of CalcInvLdData is positiv */ + temp2 = temp2 << (DFRACT_BITS - 1 - LD_DATA_SHIFT - + 1); /* INT to fract conversion of result, if input of + CalcInvLdData is positiv */ } - - if ((cmpValLeft < FL2FXCONST_DBL(0.0f)) && (cmpValRight < FL2FXCONST_DBL(0.0f))) { - noise_level_left[i] = NOISE_FLOOR_OFFSET_64 - (CalcLdData(((temp1>>1) + (temp2>>1)))); /* no scaling needed! both values are dfract */ + if ((cmpValLeft < FL2FXCONST_DBL(0.0f)) && + (cmpValRight < FL2FXCONST_DBL(0.0f))) { + noise_level_left[i] = + NOISE_FLOOR_OFFSET_64 - + (CalcLdData( + ((temp1 >> 1) + + (temp2 >> 1)))); /* no scaling needed! both values are dfract */ noise_level_right[i] = CalcLdData(temp2) - CalcLdData(temp1); } - if ((cmpValLeft >= FL2FXCONST_DBL(0.0f)) && (cmpValRight >= FL2FXCONST_DBL(0.0f))) { - noise_level_left[i] = NOISE_FLOOR_OFFSET_64 - (CalcLdData(((temp1>>1) + (temp2>>1))) + FL2FXCONST_DBL(0.109375f)); /* scaled with 7/64 */ + if ((cmpValLeft >= FL2FXCONST_DBL(0.0f)) && + (cmpValRight >= FL2FXCONST_DBL(0.0f))) { + noise_level_left[i] = NOISE_FLOOR_OFFSET_64 - + (CalcLdData(((temp1 >> 1) + (temp2 >> 1))) + + FL2FXCONST_DBL(0.109375f)); /* scaled with 7/64 */ noise_level_right[i] = CalcLdData(temp2) - CalcLdData(temp1); } - if ((cmpValLeft >= FL2FXCONST_DBL(0.0f)) && (cmpValRight < FL2FXCONST_DBL(0.0f))) { - noise_level_left[i] = NOISE_FLOOR_OFFSET_64 - (CalcLdData(((temp1>>(7+1)) + (temp2>>1))) + FL2FXCONST_DBL(0.109375f)); /* scaled with 7/64 */ - noise_level_right[i] = (CalcLdData(temp2) + FL2FXCONST_DBL(0.109375f)) - CalcLdData(temp1); + if ((cmpValLeft >= FL2FXCONST_DBL(0.0f)) && + (cmpValRight < FL2FXCONST_DBL(0.0f))) { + noise_level_left[i] = NOISE_FLOOR_OFFSET_64 - + (CalcLdData(((temp1 >> (7 + 1)) + (temp2 >> 1))) + + FL2FXCONST_DBL(0.109375f)); /* scaled with 7/64 */ + noise_level_right[i] = + (CalcLdData(temp2) + FL2FXCONST_DBL(0.109375f)) - CalcLdData(temp1); } - if ((cmpValLeft < FL2FXCONST_DBL(0.0f)) && (cmpValRight >= FL2FXCONST_DBL(0.0f))) { - noise_level_left[i] = NOISE_FLOOR_OFFSET_64 - (CalcLdData(((temp1>>1) + (temp2>>(7+1)))) + FL2FXCONST_DBL(0.109375f)); /* scaled with 7/64 */ - noise_level_right[i] = CalcLdData(temp2) - (CalcLdData(temp1) + FL2FXCONST_DBL(0.109375f)); /* scaled with 7/64 */ + if ((cmpValLeft < FL2FXCONST_DBL(0.0f)) && + (cmpValRight >= FL2FXCONST_DBL(0.0f))) { + noise_level_left[i] = NOISE_FLOOR_OFFSET_64 - + (CalcLdData(((temp1 >> 1) + (temp2 >> (7 + 1)))) + + FL2FXCONST_DBL(0.109375f)); /* scaled with 7/64 */ + noise_level_right[i] = CalcLdData(temp2) - + (CalcLdData(temp1) + + FL2FXCONST_DBL(0.109375f)); /* scaled with 7/64 */ } } } @@ -546,22 +593,23 @@ coupleNoiseFloor(FIXP_DBL *RESTRICT noise_level_left, /*! noise level left (mod /***************************************************************************/ /*! - \brief Calculation of energy starting in lower band (li) up to upper band (ui) - over slots (start_pos) to (stop_pos) + \brief Calculation of energy starting in lower band (li) up to upper band +(ui) over slots (start_pos) to (stop_pos) \return void ****************************************************************************/ -static FIXP_DBL -getEnvSfbEnergy(INT li, /*! lower band */ - INT ui, /*! upper band */ - INT start_pos, /*! start slot */ - INT stop_pos, /*! stop slot */ - INT border_pos, /*! slots scaling border */ - FIXP_DBL **YBuffer, /*! sfb energy buffer */ - INT YBufferSzShift, /*! Energy buffer index scale */ - INT scaleNrg0, /*! scaling of lower slots */ - INT scaleNrg1) /*! scaling of upper slots */ + +static FIXP_DBL getEnvSfbEnergy( + INT li, /*! lower band */ + INT ui, /*! upper band */ + INT start_pos, /*! start slot */ + INT stop_pos, /*! stop slot */ + INT border_pos, /*! slots scaling border */ + FIXP_DBL **YBuffer, /*! sfb energy buffer */ + INT YBufferSzShift, /*! Energy buffer index scale */ + INT scaleNrg0, /*! scaling of lower slots */ + INT scaleNrg1) /*! scaling of upper slots */ { /* use dynamic scaling for outer energy loop; energies are critical and every bit is important */ @@ -569,30 +617,33 @@ getEnvSfbEnergy(INT li, /*! lower band */ FIXP_DBL nrgSum, nrg1, nrg2, accu1, accu2; INT dynScale, dynScale1, dynScale2; - if(ui-li==0) dynScale = DFRACT_BITS-1; + if (ui - li == 0) + dynScale = DFRACT_BITS - 1; else - dynScale = CalcLdInt(ui-li)>>(DFRACT_BITS-1-LD_DATA_SHIFT); + dynScale = CalcLdInt(ui - li) >> (DFRACT_BITS - 1 - LD_DATA_SHIFT); - sc0 = fixMin(scaleNrg0,Y_NRG_SCALE); sc1 = fixMin(scaleNrg1,Y_NRG_SCALE); + sc0 = fixMin(scaleNrg0, Y_NRG_SCALE); + sc1 = fixMin(scaleNrg1, Y_NRG_SCALE); /* dynScale{1,2} is set such that the right shift below is positive */ - dynScale1 = fixMin((scaleNrg0-sc0),dynScale); - dynScale2 = fixMin((scaleNrg1-sc1),dynScale); + dynScale1 = fixMin((scaleNrg0 - sc0), dynScale); + dynScale2 = fixMin((scaleNrg1 - sc1), dynScale); nrgSum = accu1 = accu2 = (FIXP_DBL)0; for (k = li; k < ui; k++) { nrg1 = nrg2 = (FIXP_DBL)0; for (l = start_pos; l < border_pos; l++) { - nrg1 += YBuffer[l>>YBufferSzShift][k] >> sc0; + nrg1 += YBuffer[l >> YBufferSzShift][k] >> sc0; } for (; l < stop_pos; l++) { - nrg2 += YBuffer[l>>YBufferSzShift][k] >> sc1; + nrg2 += YBuffer[l >> YBufferSzShift][k] >> sc1; } - accu1 += (nrg1>>dynScale1); - accu2 += (nrg2>>dynScale2); + accu1 += (nrg1 >> dynScale1); + accu2 += (nrg2 >> dynScale2); } /* This shift factor is always positive. See comment above. */ - nrgSum += ( accu1 >> fixMin((scaleNrg0-sc0-dynScale1),(DFRACT_BITS-1)) ) - + ( accu2 >> fixMin((scaleNrg1-sc1-dynScale2),(DFRACT_BITS-1)) ); + nrgSum += + (accu1 >> fixMin((scaleNrg0 - sc0 - dynScale1), (DFRACT_BITS - 1))) + + (accu2 >> fixMin((scaleNrg1 - sc1 - dynScale2), (DFRACT_BITS - 1))); return nrgSum; } @@ -605,27 +656,30 @@ getEnvSfbEnergy(INT li, /*! lower band */ \return void ****************************************************************************/ -static FIXP_DBL -mhLoweringEnergy(FIXP_DBL nrg, INT M) -{ +static FIXP_DBL mhLoweringEnergy(FIXP_DBL nrg, INT M) { /* - Compensating for the fact that we in the decoder map the "average energy to every QMF - band, and use this when we calculate the boost-factor. Since the mapped energy isn't - the average energy but the maximum energy in case of missing harmonic creation, we will - in the boost function calculate that too much limiting has been applied and hence we will - boost the signal although it isn't called for. Hence we need to compensate for this by - lowering the transmitted energy values for the sines so they will get the correct level + Compensating for the fact that we in the decoder map the "average energy to + every QMF band, and use this when we calculate the boost-factor. Since the + mapped energy isn't the average energy but the maximum energy in case of + missing harmonic creation, we will in the boost function calculate that too + much limiting has been applied and hence we will boost the signal although + it isn't called for. Hence we need to compensate for this by lowering the + transmitted energy values for the sines so they will get the correct level after the boost is applied. */ - if(M > 2){ + if (M > 2) { INT tmpScale; tmpScale = CountLeadingBits(nrg); nrg <<= tmpScale; - nrg = fMult(nrg, FL2FXCONST_DBL(0.398107267f)); /* The maximum boost is 1.584893, so the maximum attenuation should be square(1/1.584893) = 0.398107267 */ + nrg = fMult(nrg, FL2FXCONST_DBL(0.398107267f)); /* The maximum boost + is 1.584893, so the + maximum attenuation + should be + square(1/1.584893) = + 0.398107267 */ nrg >>= tmpScale; - } - else{ - if(M > 1){ + } else { + if (M > 1) { nrg >>= 1; } } @@ -641,22 +695,17 @@ mhLoweringEnergy(FIXP_DBL nrg, INT M) \return void ****************************************************************************/ -static FIXP_DBL nmhLoweringEnergy( - FIXP_DBL nrg, - const FIXP_DBL nrgSum, - const INT nrgSum_scale, - const INT M - ) -{ - if (nrg>FL2FXCONST_DBL(0)) { - int sc=0; +static FIXP_DBL nmhLoweringEnergy(FIXP_DBL nrg, const FIXP_DBL nrgSum, + const INT nrgSum_scale, const INT M) { + if (nrg > FL2FXCONST_DBL(0)) { + int sc = 0; /* gain = nrgSum / (nrg*(M+1)) */ - FIXP_DBL gain = fMult(fDivNorm(nrgSum, nrg, &sc), GetInvInt(M+1)); + FIXP_DBL gain = fMult(fDivNorm(nrgSum, nrg, &sc), GetInvInt(M + 1)); sc += nrgSum_scale; /* reduce nrg if gain smaller 1.f */ - if ( !((sc>=0) && ( gain > ((FIXP_DBL)MAXVAL_DBL>>sc) )) ) { - nrg = fMult(scaleValue(gain,sc), nrg); + if (!((sc >= 0) && (gain > ((FIXP_DBL)MAXVAL_DBL >> sc)))) { + nrg = fMult(scaleValue(gain, sc), nrg); } } return nrg; @@ -671,91 +720,92 @@ static FIXP_DBL nmhLoweringEnergy( \return void ****************************************************************************/ -static void -calculateSbrEnvelope (FIXP_DBL **RESTRICT YBufferLeft, /*! energy buffer left */ - FIXP_DBL **RESTRICT YBufferRight, /*! energy buffer right */ - int *RESTRICT YBufferScaleLeft, /*! scale energy buffer left */ - int *RESTRICT YBufferScaleRight, /*! scale energy buffer right */ - const SBR_FRAME_INFO *frame_info, /*! frame info vector */ - SCHAR *RESTRICT sfb_nrgLeft, /*! sfb energy buffer left */ - SCHAR *RESTRICT sfb_nrgRight, /*! sfb energy buffer right */ - HANDLE_SBR_CONFIG_DATA h_con, /*! handle to config data */ - HANDLE_ENV_CHANNEL h_sbr, /*! envelope channel handle */ - SBR_STEREO_MODE stereoMode, /*! stereo coding mode */ - INT* maxQuantError, /*! maximum quantization error, for panorama. */ - int YBufferSzShift) /*! Energy buffer index scale */ +static void calculateSbrEnvelope( + FIXP_DBL **RESTRICT YBufferLeft, /*! energy buffer left */ + FIXP_DBL **RESTRICT YBufferRight, /*! energy buffer right */ + int *RESTRICT YBufferScaleLeft, /*! scale energy buffer left */ + int *RESTRICT YBufferScaleRight, /*! scale energy buffer right */ + const SBR_FRAME_INFO *frame_info, /*! frame info vector */ + SCHAR *RESTRICT sfb_nrgLeft, /*! sfb energy buffer left */ + SCHAR *RESTRICT sfb_nrgRight, /*! sfb energy buffer right */ + HANDLE_SBR_CONFIG_DATA h_con, /*! handle to config data */ + HANDLE_ENV_CHANNEL h_sbr, /*! envelope channel handle */ + SBR_STEREO_MODE stereoMode, /*! stereo coding mode */ + INT *maxQuantError, /*! maximum quantization error, for panorama. */ + int YBufferSzShift) /*! Energy buffer index scale */ { - int i, j, m = 0; + int env, j, m = 0; INT no_of_bands, start_pos, stop_pos, li, ui; FREQ_RES freq_res; INT ca = 2 - h_sbr->encEnvData.init_sbr_amp_res; INT oneBitLess = 0; if (ca == 2) - oneBitLess = 1; /* LD_DATA_SHIFT => ld64 scaling; one bit less for rounding */ + oneBitLess = + 1; /* LD_DATA_SHIFT => ld64 scaling; one bit less for rounding */ INT quantError; INT nEnvelopes = frame_info->nEnvelopes; INT short_env = frame_info->shortEnv - 1; INT timeStep = h_sbr->sbrExtractEnvelope.time_step; - INT commonScale,scaleLeft0,scaleLeft1; - INT scaleRight0=0,scaleRight1=0; + INT commonScale, scaleLeft0, scaleLeft1; + INT scaleRight0 = 0, scaleRight1 = 0; - commonScale = fixMin(YBufferScaleLeft[0],YBufferScaleLeft[1]); + commonScale = fixMin(YBufferScaleLeft[0], YBufferScaleLeft[1]); if (stereoMode == SBR_COUPLING) { - commonScale = fixMin(commonScale,YBufferScaleRight[0]); - commonScale = fixMin(commonScale,YBufferScaleRight[1]); + commonScale = fixMin(commonScale, YBufferScaleRight[0]); + commonScale = fixMin(commonScale, YBufferScaleRight[1]); } commonScale = commonScale - 7; scaleLeft0 = YBufferScaleLeft[0] - commonScale; - scaleLeft1 = YBufferScaleLeft[1] - commonScale ; - FDK_ASSERT ((scaleLeft0 >= 0) && (scaleLeft1 >= 0)); + scaleLeft1 = YBufferScaleLeft[1] - commonScale; + FDK_ASSERT((scaleLeft0 >= 0) && (scaleLeft1 >= 0)); if (stereoMode == SBR_COUPLING) { scaleRight0 = YBufferScaleRight[0] - commonScale; scaleRight1 = YBufferScaleRight[1] - commonScale; - FDK_ASSERT ((scaleRight0 >= 0) && (scaleRight1 >= 0)); + FDK_ASSERT((scaleRight0 >= 0) && (scaleRight1 >= 0)); *maxQuantError = 0; } - for (i = 0; i < nEnvelopes; i++) { - - FIXP_DBL pNrgLeft[QMF_MAX_TIME_SLOTS]; - FIXP_DBL pNrgRight[QMF_MAX_TIME_SLOTS]; + for (env = 0; env < nEnvelopes; env++) { + FIXP_DBL pNrgLeft[32]; + FIXP_DBL pNrgRight[32]; int envNrg_scale; - FIXP_DBL envNrgLeft = FL2FXCONST_DBL(0.0f); + FIXP_DBL envNrgLeft = FL2FXCONST_DBL(0.0f); FIXP_DBL envNrgRight = FL2FXCONST_DBL(0.0f); - int missingHarmonic[QMF_MAX_TIME_SLOTS]; - int count[QMF_MAX_TIME_SLOTS]; + int missingHarmonic[32]; + int count[32]; - start_pos = timeStep * frame_info->borders[i]; - stop_pos = timeStep * frame_info->borders[i + 1]; - freq_res = frame_info->freqRes[i]; + start_pos = timeStep * frame_info->borders[env]; + stop_pos = timeStep * frame_info->borders[env + 1]; + freq_res = frame_info->freqRes[env]; no_of_bands = h_con->nSfb[freq_res]; - envNrg_scale = DFRACT_BITS-fNormz((FIXP_DBL)no_of_bands); - - if (i == short_env) { - stop_pos -= fixMax(2, timeStep); /* consider at least 2 QMF slots less for short envelopes (envelopes just before transients) */ + envNrg_scale = DFRACT_BITS - fNormz((FIXP_DBL)no_of_bands); + if (env == short_env) { + j = fMax(2, timeStep); /* consider at least 2 QMF slots less for short + envelopes (envelopes just before transients) */ + if ((stop_pos - start_pos - j) > 0) { + stop_pos = stop_pos - j; + } } - for (j = 0; j < no_of_bands; j++) { - FIXP_DBL nrgLeft = FL2FXCONST_DBL(0.0f); + FIXP_DBL nrgLeft = FL2FXCONST_DBL(0.0f); FIXP_DBL nrgRight = FL2FXCONST_DBL(0.0f); li = h_con->freqBandTable[freq_res][j]; ui = h_con->freqBandTable[freq_res][j + 1]; - if(freq_res == FREQ_RES_HIGH){ - if(j == 0 && ui-li > 1){ + if (freq_res == FREQ_RES_HIGH) { + if (j == 0 && ui - li > 1) { li++; } - } - else{ - if(j == 0 && ui-li > 2){ + } else { + if (j == 0 && ui - li > 2) { li++; } } @@ -766,25 +816,26 @@ calculateSbrEnvelope (FIXP_DBL **RESTRICT YBufferLeft, /*! energy buffer left * */ missingHarmonic[j] = 0; - if(h_sbr->encEnvData.addHarmonicFlag){ - - if(freq_res == FREQ_RES_HIGH){ - if(h_sbr->encEnvData.addHarmonic[j]){ /*A missing sine in the current band*/ + if (h_sbr->encEnvData.addHarmonicFlag) { + if (freq_res == FREQ_RES_HIGH) { + if (h_sbr->encEnvData + .addHarmonic[j]) { /*A missing sine in the current band*/ missingHarmonic[j] = 1; } - } - else{ + } else { INT i; INT startBandHigh = 0; INT stopBandHigh = 0; - while(h_con->freqBandTable[FREQ_RES_HIGH][startBandHigh] < h_con->freqBandTable[FREQ_RES_LOW][j]) + while (h_con->freqBandTable[FREQ_RES_HIGH][startBandHigh] < + h_con->freqBandTable[FREQ_RES_LOW][j]) startBandHigh++; - while(h_con->freqBandTable[FREQ_RES_HIGH][stopBandHigh] < h_con->freqBandTable[FREQ_RES_LOW][j + 1]) + while (h_con->freqBandTable[FREQ_RES_HIGH][stopBandHigh] < + h_con->freqBandTable[FREQ_RES_LOW][j + 1]) stopBandHigh++; - for(i = startBandHigh; i<stopBandHigh; i++){ - if(h_sbr->encEnvData.addHarmonic[i]){ + for (i = startBandHigh; i < stopBandHigh; i++) { + if (h_sbr->encEnvData.addHarmonic[i]) { missingHarmonic[j] = 1; } } @@ -792,105 +843,82 @@ calculateSbrEnvelope (FIXP_DBL **RESTRICT YBufferLeft, /*! energy buffer left * } /* - If a sine is missing in a scalefactorband, with more than one qmf channel - use the nrg from the channel with the largest nrg rather than the mean. - Compensate for the boost calculation in the decdoder. + If a sine is missing in a scalefactorband, with more than one qmf + channel use the nrg from the channel with the largest nrg rather than + the mean. Compensate for the boost calculation in the decdoder. */ - int border_pos = fixMin(stop_pos, h_sbr->sbrExtractEnvelope.YBufferWriteOffset<<YBufferSzShift); - - if(missingHarmonic[j]){ + int border_pos = + fixMin(stop_pos, h_sbr->sbrExtractEnvelope.YBufferWriteOffset + << YBufferSzShift); + if (missingHarmonic[j]) { int k; count[j] = stop_pos - start_pos; nrgLeft = FL2FXCONST_DBL(0.0f); for (k = li; k < ui; k++) { FIXP_DBL tmpNrg; - tmpNrg = getEnvSfbEnergy(k, - k+1, - start_pos, - stop_pos, - border_pos, - YBufferLeft, - YBufferSzShift, - scaleLeft0, + tmpNrg = getEnvSfbEnergy(k, k + 1, start_pos, stop_pos, border_pos, + YBufferLeft, YBufferSzShift, scaleLeft0, scaleLeft1); nrgLeft = fixMax(nrgLeft, tmpNrg); } /* Energy lowering compensation */ - nrgLeft = mhLoweringEnergy(nrgLeft, ui-li); + nrgLeft = mhLoweringEnergy(nrgLeft, ui - li); if (stereoMode == SBR_COUPLING) { - nrgRight = FL2FXCONST_DBL(0.0f); for (k = li; k < ui; k++) { FIXP_DBL tmpNrg; - tmpNrg = getEnvSfbEnergy(k, - k+1, - start_pos, - stop_pos, - border_pos, - YBufferRight, - YBufferSzShift, - scaleRight0, + tmpNrg = getEnvSfbEnergy(k, k + 1, start_pos, stop_pos, border_pos, + YBufferRight, YBufferSzShift, scaleRight0, scaleRight1); nrgRight = fixMax(nrgRight, tmpNrg); } /* Energy lowering compensation */ - nrgRight = mhLoweringEnergy(nrgRight, ui-li); + nrgRight = mhLoweringEnergy(nrgRight, ui - li); } } /* end missingHarmonic */ - else{ + else { count[j] = (stop_pos - start_pos) * (ui - li); - nrgLeft = getEnvSfbEnergy(li, - ui, - start_pos, - stop_pos, - border_pos, - YBufferLeft, - YBufferSzShift, - scaleLeft0, + nrgLeft = getEnvSfbEnergy(li, ui, start_pos, stop_pos, border_pos, + YBufferLeft, YBufferSzShift, scaleLeft0, scaleLeft1); if (stereoMode == SBR_COUPLING) { - nrgRight = getEnvSfbEnergy(li, - ui, - start_pos, - stop_pos, - border_pos, - YBufferRight, - YBufferSzShift, - scaleRight0, + nrgRight = getEnvSfbEnergy(li, ui, start_pos, stop_pos, border_pos, + YBufferRight, YBufferSzShift, scaleRight0, scaleRight1); } } /* !missingHarmonic */ /* save energies */ - pNrgLeft[j] = nrgLeft; + pNrgLeft[j] = nrgLeft; pNrgRight[j] = nrgRight; - envNrgLeft += (nrgLeft>>envNrg_scale); - envNrgRight += (nrgRight>>envNrg_scale); + envNrgLeft += (nrgLeft >> envNrg_scale); + envNrgRight += (nrgRight >> envNrg_scale); } /* j */ for (j = 0; j < no_of_bands; j++) { - FIXP_DBL nrgLeft2 = FL2FXCONST_DBL(0.0f); - FIXP_DBL nrgLeft = pNrgLeft[j]; + FIXP_DBL nrgLeft = pNrgLeft[j]; FIXP_DBL nrgRight = pNrgRight[j]; /* None missing harmonic Energy lowering compensation */ - if(!missingHarmonic[j] && h_sbr->fLevelProtect) { + if (!missingHarmonic[j] && h_sbr->fLevelProtect) { /* in case of missing energy in base band, reduce reference energy to prevent overflows in decoder output */ - nrgLeft = nmhLoweringEnergy(nrgLeft, envNrgLeft, envNrg_scale, no_of_bands); + nrgLeft = + nmhLoweringEnergy(nrgLeft, envNrgLeft, envNrg_scale, no_of_bands); if (stereoMode == SBR_COUPLING) { - nrgRight = nmhLoweringEnergy(nrgRight, envNrgRight, envNrg_scale, no_of_bands); + nrgRight = nmhLoweringEnergy(nrgRight, envNrgRight, envNrg_scale, + no_of_bands); } } @@ -900,31 +928,34 @@ calculateSbrEnvelope (FIXP_DBL **RESTRICT YBufferLeft, /*! energy buffer left * nrgLeft = (nrgRight + nrgLeft) >> 1; } - /* nrgLeft = f20_log2(nrgLeft / (PFLOAT)(count * h_sbr->sbrQmf.no_channels))+(PFLOAT)44; */ + /* nrgLeft = f20_log2(nrgLeft / (PFLOAT)(count * 64))+(PFLOAT)44; */ /* If nrgLeft == 0 then the Log calculations below do fail. */ - if (nrgLeft > FL2FXCONST_DBL(0.0f)) - { - FIXP_DBL tmp0,tmp1,tmp2,tmp3; + if (nrgLeft > FL2FXCONST_DBL(0.0f)) { + FIXP_DBL tmp0, tmp1, tmp2, tmp3; INT tmpScale; tmpScale = CountLeadingBits(nrgLeft); nrgLeft = nrgLeft << tmpScale; - tmp0 = CalcLdData(nrgLeft); /* scaled by 1/64 */ - tmp1 = ((FIXP_DBL) (commonScale+tmpScale)) << (DFRACT_BITS-1-LD_DATA_SHIFT-1); /* scaled by 1/64 */ - tmp2 = ((FIXP_DBL)(count[j]*h_con->noQmfBands)) << (DFRACT_BITS-1-14-1); - tmp2 = CalcLdData(tmp2); /* scaled by 1/64 */ - tmp3 = FL2FXCONST_DBL(0.6875f-0.21875f-0.015625f)>>1; /* scaled by 1/64 */ + tmp0 = CalcLdData(nrgLeft); /* scaled by 1/64 */ + tmp1 = ((FIXP_DBL)(commonScale + tmpScale)) + << (DFRACT_BITS - 1 - LD_DATA_SHIFT - 1); /* scaled by 1/64 */ + tmp2 = ((FIXP_DBL)(count[j] * 64)) << (DFRACT_BITS - 1 - 14 - 1); + tmp2 = CalcLdData(tmp2); /* scaled by 1/64 */ + tmp3 = FL2FXCONST_DBL(0.6875f - 0.21875f - 0.015625f) >> + 1; /* scaled by 1/64 */ - nrgLeft = ((tmp0-tmp2)>>1) + (tmp3 - tmp1); + nrgLeft = ((tmp0 - tmp2) >> 1) + (tmp3 - tmp1); } else { nrgLeft = FL2FXCONST_DBL(-1.0f); } /* ld64 to integer conversion */ - nrgLeft = fixMin(fixMax(nrgLeft,FL2FXCONST_DBL(0.0f)),(FL2FXCONST_DBL(0.5f)>>oneBitLess)); - nrgLeft = (FIXP_DBL)(LONG)nrgLeft >> (DFRACT_BITS-1-LD_DATA_SHIFT-1-oneBitLess-1); - sfb_nrgLeft[m] = ((INT)nrgLeft+1)>>1; /* rounding */ + nrgLeft = fixMin(fixMax(nrgLeft, FL2FXCONST_DBL(0.0f)), + (FL2FXCONST_DBL(0.5f) >> oneBitLess)); + nrgLeft = (FIXP_DBL)(LONG)nrgLeft >> + (DFRACT_BITS - 1 - LD_DATA_SHIFT - 1 - oneBitLess - 1); + sfb_nrgLeft[m] = ((INT)nrgLeft + 1) >> 1; /* rounding */ if (stereoMode == SBR_COUPLING) { FIXP_DBL scaleFract; @@ -936,14 +967,20 @@ calculateSbrEnvelope (FIXP_DBL **RESTRICT YBufferLeft, /*! energy buffer left * sc0 = CountLeadingBits(nrgLeft2); sc1 = CountLeadingBits(nrgRight); - scaleFract = ((FIXP_DBL)(sc0-sc1)) << (DFRACT_BITS-1-LD_DATA_SHIFT); /* scale value in ld64 representation */ - nrgRight = CalcLdData(nrgLeft2<<sc0) - CalcLdData(nrgRight<<sc1) - scaleFract; + scaleFract = + ((FIXP_DBL)(sc0 - sc1)) + << (DFRACT_BITS - 1 - + LD_DATA_SHIFT); /* scale value in ld64 representation */ + nrgRight = CalcLdData(nrgLeft2 << sc0) - CalcLdData(nrgRight << sc1) - + scaleFract; /* ld64 to integer conversion */ - nrgRight = (FIXP_DBL)(LONG)(nrgRight) >> (DFRACT_BITS-1-LD_DATA_SHIFT-1-oneBitLess); - nrgRight = (nrgRight+(FIXP_DBL)1)>>1; /* rounding */ + nrgRight = (FIXP_DBL)(LONG)(nrgRight) >> + (DFRACT_BITS - 1 - LD_DATA_SHIFT - 1 - oneBitLess); + nrgRight = (nrgRight + (FIXP_DBL)1) >> 1; /* rounding */ - sfb_nrgRight[m] = mapPanorama (nrgRight,h_sbr->encEnvData.init_sbr_amp_res,&quantError); + sfb_nrgRight[m] = mapPanorama( + nrgRight, h_sbr->encEnvData.init_sbr_amp_res, &quantError); *maxQuantError = fixMax(quantError, *maxQuantError); } @@ -951,21 +988,25 @@ calculateSbrEnvelope (FIXP_DBL **RESTRICT YBufferLeft, /*! energy buffer left * m++; } /* j */ - /* Do energy compensation for sines that are present in two - QMF-bands in the original, but will only occur in one band in - the decoder due to the synthetic sine coding.*/ + /* Do energy compensation for sines that are present in two + QMF-bands in the original, but will only occur in one band in + the decoder due to the synthetic sine coding.*/ if (h_con->useParametricCoding) { - m-=no_of_bands; + m -= no_of_bands; for (j = 0; j < no_of_bands; j++) { - if (freq_res==FREQ_RES_HIGH && h_sbr->sbrExtractEnvelope.envelopeCompensation[j]){ - sfb_nrgLeft[m] -= (ca * fixp_abs((INT)h_sbr->sbrExtractEnvelope.envelopeCompensation[j])); + if (freq_res == FREQ_RES_HIGH && + h_sbr->sbrExtractEnvelope.envelopeCompensation[j]) { + sfb_nrgLeft[m] -= + (ca * + fixp_abs( + (INT)h_sbr->sbrExtractEnvelope.envelopeCompensation[j])); } sfb_nrgLeft[m] = fixMax(0, sfb_nrgLeft[m]); m++; } } /* useParametricCoding */ - } /* i*/ + } /* env loop */ } /***************************************************************************/ @@ -984,96 +1025,73 @@ calculateSbrEnvelope (FIXP_DBL **RESTRICT YBufferLeft, /*! energy buffer left * ****************************************************************************/ LNK_SECTION_CODE_L1 -void -FDKsbrEnc_extractSbrEnvelope1 ( - HANDLE_SBR_CONFIG_DATA h_con, /*! handle to config data */ - HANDLE_SBR_HEADER_DATA sbrHeaderData, - HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, - HANDLE_ENV_CHANNEL hEnvChan, - HANDLE_COMMON_DATA hCmonData, - SBR_ENV_TEMP_DATA *eData, - SBR_FRAME_TEMP_DATA *fData - ) -{ - +void FDKsbrEnc_extractSbrEnvelope1( + HANDLE_SBR_CONFIG_DATA h_con, /*! handle to config data */ + HANDLE_SBR_HEADER_DATA sbrHeaderData, + HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, HANDLE_ENV_CHANNEL hEnvChan, + HANDLE_COMMON_DATA hCmonData, SBR_ENV_TEMP_DATA *eData, + SBR_FRAME_TEMP_DATA *fData) { HANDLE_SBR_EXTRACT_ENVELOPE sbrExtrEnv = &hEnvChan->sbrExtractEnvelope; if (sbrExtrEnv->YBufferSzShift == 0) - FDKsbrEnc_getEnergyFromCplxQmfDataFull(&sbrExtrEnv->YBuffer[sbrExtrEnv->YBufferWriteOffset], - sbrExtrEnv->rBuffer + sbrExtrEnv->rBufferReadOffset, - sbrExtrEnv->iBuffer + sbrExtrEnv->rBufferReadOffset, - h_con->noQmfBands, - sbrExtrEnv->no_cols, - &hEnvChan->qmfScale, - &sbrExtrEnv->YBufferScale[1]); + FDKsbrEnc_getEnergyFromCplxQmfDataFull( + &sbrExtrEnv->YBuffer[sbrExtrEnv->YBufferWriteOffset], + sbrExtrEnv->rBuffer + sbrExtrEnv->rBufferReadOffset, + sbrExtrEnv->iBuffer + sbrExtrEnv->rBufferReadOffset, h_con->noQmfBands, + sbrExtrEnv->no_cols, &hEnvChan->qmfScale, &sbrExtrEnv->YBufferScale[1]); else - FDKsbrEnc_getEnergyFromCplxQmfData(&sbrExtrEnv->YBuffer[sbrExtrEnv->YBufferWriteOffset], - sbrExtrEnv->rBuffer + sbrExtrEnv->rBufferReadOffset, - sbrExtrEnv->iBuffer + sbrExtrEnv->rBufferReadOffset, - h_con->noQmfBands, - sbrExtrEnv->no_cols, - &hEnvChan->qmfScale, - &sbrExtrEnv->YBufferScale[1]); - + FDKsbrEnc_getEnergyFromCplxQmfData( + &sbrExtrEnv->YBuffer[sbrExtrEnv->YBufferWriteOffset], + sbrExtrEnv->rBuffer + sbrExtrEnv->rBufferReadOffset, + sbrExtrEnv->iBuffer + sbrExtrEnv->rBufferReadOffset, h_con->noQmfBands, + sbrExtrEnv->no_cols, &hEnvChan->qmfScale, &sbrExtrEnv->YBufferScale[1]); + /* Energie values = + * sbrExtrEnv->YBuffer[sbrExtrEnv->YBufferWriteOffset][x].floatVal * + * (1<<2*7-sbrExtrEnv->YBufferScale[1]) */ /* Precalculation of Tonality Quotas COEFF Transform OK */ - FDKsbrEnc_CalculateTonalityQuotas(&hEnvChan->TonCorr, - sbrExtrEnv->rBuffer, - sbrExtrEnv->iBuffer, - h_con->freqBandTable[HI][h_con->nSfb[HI]], - hEnvChan->qmfScale); - - - if(h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { - FIXP_DBL tonality = FDKsbrEnc_GetTonality ( - hEnvChan->TonCorr.quotaMatrix, - hEnvChan->TonCorr.numberOfEstimatesPerFrame, - hEnvChan->TonCorr.startIndexMatrix, - sbrExtrEnv->YBuffer + sbrExtrEnv->YBufferWriteOffset, - h_con->freqBandTable[HI][0]+1, - h_con->noQmfBands, - sbrExtrEnv->no_cols - ); + FDKsbrEnc_CalculateTonalityQuotas( + &hEnvChan->TonCorr, sbrExtrEnv->rBuffer, sbrExtrEnv->iBuffer, + h_con->freqBandTable[HI][h_con->nSfb[HI]], hEnvChan->qmfScale); + + if (h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { + FIXP_DBL tonality = FDKsbrEnc_GetTonality( + hEnvChan->TonCorr.quotaMatrix, + hEnvChan->TonCorr.numberOfEstimatesPerFrame, + hEnvChan->TonCorr.startIndexMatrix, + sbrExtrEnv->YBuffer + sbrExtrEnv->YBufferWriteOffset, + h_con->freqBandTable[HI][0] + 1, h_con->noQmfBands, + sbrExtrEnv->no_cols); hEnvChan->encEnvData.ton_HF[1] = hEnvChan->encEnvData.ton_HF[0]; hEnvChan->encEnvData.ton_HF[0] = tonality; /* tonality is scaled by 2^19/0.524288f (fract part of RELAXATION) */ - hEnvChan->encEnvData.global_tonality = (hEnvChan->encEnvData.ton_HF[0]>>1) + (hEnvChan->encEnvData.ton_HF[1]>>1); + hEnvChan->encEnvData.global_tonality = + (hEnvChan->encEnvData.ton_HF[0] >> 1) + + (hEnvChan->encEnvData.ton_HF[1] >> 1); } - - /* Transient detection COEFF Transform OK */ - if(h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) - { - FDKsbrEnc_fastTransientDetect( - &hEnvChan->sbrFastTransientDetector, - sbrExtrEnv->YBuffer, - sbrExtrEnv->YBufferScale, - sbrExtrEnv->YBufferWriteOffset, - eData->transient_info - ); - - } - else - { - FDKsbrEnc_transientDetect(&hEnvChan->sbrTransientDetector, - sbrExtrEnv->YBuffer, - sbrExtrEnv->YBufferScale, - eData->transient_info, - sbrExtrEnv->YBufferWriteOffset, - sbrExtrEnv->YBufferSzShift, - sbrExtrEnv->time_step, - hEnvChan->SbrEnvFrame.frameMiddleSlot); - } + if (h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { + FDKsbrEnc_fastTransientDetect(&hEnvChan->sbrFastTransientDetector, + sbrExtrEnv->YBuffer, sbrExtrEnv->YBufferScale, + sbrExtrEnv->YBufferWriteOffset, + eData->transient_info); + } else { + FDKsbrEnc_transientDetect( + &hEnvChan->sbrTransientDetector, sbrExtrEnv->YBuffer, + sbrExtrEnv->YBufferScale, eData->transient_info, + sbrExtrEnv->YBufferWriteOffset, sbrExtrEnv->YBufferSzShift, + sbrExtrEnv->time_step, hEnvChan->SbrEnvFrame.frameMiddleSlot); + } /* Generate flags for 2 env in a FIXFIX-frame. @@ -1083,19 +1101,12 @@ FDKsbrEnc_extractSbrEnvelope1 ( /* frame Splitter COEFF Transform OK */ - FDKsbrEnc_frameSplitter(sbrExtrEnv->YBuffer, - sbrExtrEnv->YBufferScale, - &hEnvChan->sbrTransientDetector, - h_con->freqBandTable[1], - eData->transient_info, - sbrExtrEnv->YBufferWriteOffset, - sbrExtrEnv->YBufferSzShift, - h_con->nSfb[1], - sbrExtrEnv->time_step, - sbrExtrEnv->no_cols, - &hEnvChan->encEnvData.global_tonality); - - + FDKsbrEnc_frameSplitter( + sbrExtrEnv->YBuffer, sbrExtrEnv->YBufferScale, + &hEnvChan->sbrTransientDetector, h_con->freqBandTable[1], + eData->transient_info, sbrExtrEnv->YBufferWriteOffset, + sbrExtrEnv->YBufferSzShift, h_con->nSfb[1], sbrExtrEnv->time_step, + sbrExtrEnv->no_cols, &hEnvChan->encEnvData.global_tonality); } /***************************************************************************/ @@ -1128,53 +1139,45 @@ FDKsbrEnc_extractSbrEnvelope1 ( ****************************************************************************/ LNK_SECTION_CODE_L1 -void -FDKsbrEnc_extractSbrEnvelope2 ( - HANDLE_SBR_CONFIG_DATA h_con, /*! handle to config data */ - HANDLE_SBR_HEADER_DATA sbrHeaderData, - HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, - HANDLE_ENV_CHANNEL h_envChan0, - HANDLE_ENV_CHANNEL h_envChan1, - HANDLE_COMMON_DATA hCmonData, - SBR_ENV_TEMP_DATA *eData, - SBR_FRAME_TEMP_DATA *fData, - int clearOutput - ) -{ +void FDKsbrEnc_extractSbrEnvelope2( + HANDLE_SBR_CONFIG_DATA h_con, /*! handle to config data */ + HANDLE_SBR_HEADER_DATA sbrHeaderData, + HANDLE_PARAMETRIC_STEREO hParametricStereo, + HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, HANDLE_ENV_CHANNEL h_envChan0, + HANDLE_ENV_CHANNEL h_envChan1, HANDLE_COMMON_DATA hCmonData, + SBR_ENV_TEMP_DATA *eData, SBR_FRAME_TEMP_DATA *fData, int clearOutput) { HANDLE_ENV_CHANNEL h_envChan[MAX_NUM_CHANNELS] = {h_envChan0, h_envChan1}; int ch, i, j, c, YSzShift = h_envChan[0]->sbrExtractEnvelope.YBufferSzShift; SBR_STEREO_MODE stereoMode = h_con->stereoMode; int nChannels = h_con->nChannels; const int *v_tuning; - static const int v_tuningHEAAC[6] = { 0, 2, 4, 0, 0, 0 }; + static const int v_tuningHEAAC[6] = {0, 2, 4, 0, 0, 0}; - static const int v_tuningELD[6] = { 0, 2, 3, 0, 0, 0 }; + static const int v_tuningELD[6] = {0, 2, 3, 0, 0, 0}; if (h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) v_tuning = v_tuningELD; else v_tuning = v_tuningHEAAC; - /* Select stereo mode. */ if (stereoMode == SBR_COUPLING) { if (eData[0].transient_info[1] && eData[1].transient_info[1]) { - eData[0].transient_info[0] = fixMin(eData[1].transient_info[0], eData[0].transient_info[0]); + eData[0].transient_info[0] = + fixMin(eData[1].transient_info[0], eData[0].transient_info[0]); eData[1].transient_info[0] = eData[0].transient_info[0]; - } - else { + } else { if (eData[0].transient_info[1] && !eData[1].transient_info[1]) { eData[1].transient_info[0] = eData[0].transient_info[0]; - } - else { + } else { if (!eData[0].transient_info[1] && eData[1].transient_info[1]) eData[0].transient_info[0] = eData[1].transient_info[0]; else { - eData[0].transient_info[0] = fixMax(eData[1].transient_info[0], eData[0].transient_info[0]); + eData[0].transient_info[0] = + fixMax(eData[1].transient_info[0], eData[0].transient_info[0]); eData[1].transient_info[0] = eData[0].transient_info[0]; } } @@ -1184,183 +1187,171 @@ FDKsbrEnc_extractSbrEnvelope2 ( /* Determine time/frequency division of current granule */ - eData[0].frame_info = FDKsbrEnc_frameInfoGenerator(&h_envChan[0]->SbrEnvFrame, - eData[0].transient_info, - h_envChan[0]->sbrExtractEnvelope.pre_transient_info, - h_envChan[0]->encEnvData.ldGrid, - v_tuning); + eData[0].frame_info = FDKsbrEnc_frameInfoGenerator( + &h_envChan[0]->SbrEnvFrame, eData[0].transient_info, + sbrBitstreamData->rightBorderFIX, + h_envChan[0]->sbrExtractEnvelope.pre_transient_info, + h_envChan[0]->encEnvData.ldGrid, v_tuning); h_envChan[0]->encEnvData.hSbrBSGrid = &h_envChan[0]->SbrEnvFrame.SbrGrid; /* AAC LD patch for transient prediction */ if (h_envChan[0]->encEnvData.ldGrid && eData[0].transient_info[2]) { - /* if next frame will start with transient, set shortEnv to numEnvelopes(shortend Envelope = shortEnv-1)*/ - h_envChan[0]->SbrEnvFrame.SbrFrameInfo.shortEnv = h_envChan[0]->SbrEnvFrame.SbrFrameInfo.nEnvelopes; + /* if next frame will start with transient, set shortEnv to + * numEnvelopes(shortend Envelope = shortEnv-1)*/ + h_envChan[0]->SbrEnvFrame.SbrFrameInfo.shortEnv = + h_envChan[0]->SbrEnvFrame.SbrFrameInfo.nEnvelopes; } - switch (stereoMode) { - case SBR_LEFT_RIGHT: - case SBR_SWITCH_LRC: - eData[1].frame_info = FDKsbrEnc_frameInfoGenerator(&h_envChan[1]->SbrEnvFrame, - eData[1].transient_info, - h_envChan[1]->sbrExtractEnvelope.pre_transient_info, - h_envChan[1]->encEnvData.ldGrid, - v_tuning); - - h_envChan[1]->encEnvData.hSbrBSGrid = &h_envChan[1]->SbrEnvFrame.SbrGrid; - - if (h_envChan[1]->encEnvData.ldGrid && eData[1].transient_info[2]) { - /* if next frame will start with transient, set shortEnv to numEnvelopes(shortend Envelope = shortEnv-1)*/ - h_envChan[1]->SbrEnvFrame.SbrFrameInfo.shortEnv = h_envChan[1]->SbrEnvFrame.SbrFrameInfo.nEnvelopes; - } + case SBR_LEFT_RIGHT: + case SBR_SWITCH_LRC: + eData[1].frame_info = FDKsbrEnc_frameInfoGenerator( + &h_envChan[1]->SbrEnvFrame, eData[1].transient_info, + sbrBitstreamData->rightBorderFIX, + h_envChan[1]->sbrExtractEnvelope.pre_transient_info, + h_envChan[1]->encEnvData.ldGrid, v_tuning); + + h_envChan[1]->encEnvData.hSbrBSGrid = &h_envChan[1]->SbrEnvFrame.SbrGrid; + + if (h_envChan[1]->encEnvData.ldGrid && eData[1].transient_info[2]) { + /* if next frame will start with transient, set shortEnv to + * numEnvelopes(shortend Envelope = shortEnv-1)*/ + h_envChan[1]->SbrEnvFrame.SbrFrameInfo.shortEnv = + h_envChan[1]->SbrEnvFrame.SbrFrameInfo.nEnvelopes; + } - /* compare left and right frame_infos */ - if (eData[0].frame_info->nEnvelopes != eData[1].frame_info->nEnvelopes) { - stereoMode = SBR_LEFT_RIGHT; - } else { - for (i = 0; i < eData[0].frame_info->nEnvelopes + 1; i++) { - if (eData[0].frame_info->borders[i] != eData[1].frame_info->borders[i]) { - stereoMode = SBR_LEFT_RIGHT; - break; + /* compare left and right frame_infos */ + if (eData[0].frame_info->nEnvelopes != eData[1].frame_info->nEnvelopes) { + stereoMode = SBR_LEFT_RIGHT; + } else { + for (i = 0; i < eData[0].frame_info->nEnvelopes + 1; i++) { + if (eData[0].frame_info->borders[i] != + eData[1].frame_info->borders[i]) { + stereoMode = SBR_LEFT_RIGHT; + break; + } } - } - for (i = 0; i < eData[0].frame_info->nEnvelopes; i++) { - if (eData[0].frame_info->freqRes[i] != eData[1].frame_info->freqRes[i]) { + for (i = 0; i < eData[0].frame_info->nEnvelopes; i++) { + if (eData[0].frame_info->freqRes[i] != + eData[1].frame_info->freqRes[i]) { + stereoMode = SBR_LEFT_RIGHT; + break; + } + } + if (eData[0].frame_info->shortEnv != eData[1].frame_info->shortEnv) { stereoMode = SBR_LEFT_RIGHT; - break; } } - if (eData[0].frame_info->shortEnv != eData[1].frame_info->shortEnv) { - stereoMode = SBR_LEFT_RIGHT; - } - } - break; - case SBR_COUPLING: - eData[1].frame_info = eData[0].frame_info; - h_envChan[1]->encEnvData.hSbrBSGrid = &h_envChan[0]->SbrEnvFrame.SbrGrid; - break; - case SBR_MONO: - /* nothing to do */ - break; - default: - FDK_ASSERT (0); + break; + case SBR_COUPLING: + eData[1].frame_info = eData[0].frame_info; + h_envChan[1]->encEnvData.hSbrBSGrid = &h_envChan[0]->SbrEnvFrame.SbrGrid; + break; + case SBR_MONO: + /* nothing to do */ + break; + default: + FDK_ASSERT(0); } - - for (ch = 0; ch < nChannels;ch++) - { + for (ch = 0; ch < nChannels; ch++) { HANDLE_ENV_CHANNEL hEnvChan = h_envChan[ch]; HANDLE_SBR_EXTRACT_ENVELOPE sbrExtrEnv = &hEnvChan->sbrExtractEnvelope; SBR_ENV_TEMP_DATA *ed = &eData[ch]; - /* Send transient info to bitstream and store for next call */ - sbrExtrEnv->pre_transient_info[0] = ed->transient_info[0];/* tran_pos */ - sbrExtrEnv->pre_transient_info[1] = ed->transient_info[1];/* tran_flag */ - hEnvChan->encEnvData.noOfEnvelopes = ed->nEnvelopes = ed->frame_info->nEnvelopes; /* number of envelopes of current frame */ + sbrExtrEnv->pre_transient_info[0] = ed->transient_info[0]; /* tran_pos */ + sbrExtrEnv->pre_transient_info[1] = ed->transient_info[1]; /* tran_flag */ + hEnvChan->encEnvData.noOfEnvelopes = ed->nEnvelopes = + ed->frame_info->nEnvelopes; /* number of envelopes of current frame */ /* - Check if the current frame is divided into one envelope only. If so, set the amplitude - resolution to 1.5 dB, otherwise may set back to chosen value + Check if the current frame is divided into one envelope only. If so, set + the amplitude resolution to 1.5 dB, otherwise may set back to chosen value */ - if( ( hEnvChan->encEnvData.hSbrBSGrid->frameClass == FIXFIX ) - && ( ed->nEnvelopes == 1 ) ) - { - - if (h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) - { - /* Note: global_tonaliy_float_value == ((float)hEnvChan->encEnvData.global_tonality/((INT64)(1)<<(31-(19+2)))/0.524288*(2.0/3.0))); - threshold_float_value == ((float)h_con->thresholdAmpResFF_m/((INT64)(1)<<(31-(h_con->thresholdAmpResFF_e)))/0.524288*(2.0/3.0))); */ - /* decision of SBR_AMP_RES */ - if (fIsLessThan( /* global_tonality > threshold ? */ - h_con->thresholdAmpResFF_m, h_con->thresholdAmpResFF_e, - hEnvChan->encEnvData.global_tonality, RELAXATION_SHIFT+2 ) - ) - { - hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_1_5; - } - else { - hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_3_0; - } - } else { - hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_1_5; - } - - if ( hEnvChan->encEnvData.currentAmpResFF != hEnvChan->encEnvData.init_sbr_amp_res) { - - FDKsbrEnc_InitSbrHuffmanTables(&hEnvChan->encEnvData, - &hEnvChan->sbrCodeEnvelope, - &hEnvChan->sbrCodeNoiseFloor, - hEnvChan->encEnvData.currentAmpResFF); + if ((hEnvChan->encEnvData.hSbrBSGrid->frameClass == FIXFIX) && + (ed->nEnvelopes == 1)) { + if (h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { + /* Note: global_tonaliy_float_value == + ((float)hEnvChan->encEnvData.global_tonality/((INT64)(1)<<(31-(19+2)))/0.524288*(2.0/3.0))); + threshold_float_value == + ((float)h_con->thresholdAmpResFF_m/((INT64)(1)<<(31-(h_con->thresholdAmpResFF_e)))/0.524288*(2.0/3.0))); + */ + /* decision of SBR_AMP_RES */ + if (fIsLessThan(/* global_tonality > threshold ? */ + h_con->thresholdAmpResFF_m, h_con->thresholdAmpResFF_e, + hEnvChan->encEnvData.global_tonality, + RELAXATION_SHIFT + 2)) { + hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_1_5; + } else { + hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_3_0; + } + } else + hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_1_5; + + if (hEnvChan->encEnvData.currentAmpResFF != + hEnvChan->encEnvData.init_sbr_amp_res) { + FDKsbrEnc_InitSbrHuffmanTables( + &hEnvChan->encEnvData, &hEnvChan->sbrCodeEnvelope, + &hEnvChan->sbrCodeNoiseFloor, hEnvChan->encEnvData.currentAmpResFF); } - } - else { - if(sbrHeaderData->sbr_amp_res != hEnvChan->encEnvData.init_sbr_amp_res ) { - - FDKsbrEnc_InitSbrHuffmanTables(&hEnvChan->encEnvData, - &hEnvChan->sbrCodeEnvelope, - &hEnvChan->sbrCodeNoiseFloor, - sbrHeaderData->sbr_amp_res); + } else { + if (sbrHeaderData->sbr_amp_res != hEnvChan->encEnvData.init_sbr_amp_res) { + FDKsbrEnc_InitSbrHuffmanTables( + &hEnvChan->encEnvData, &hEnvChan->sbrCodeEnvelope, + &hEnvChan->sbrCodeNoiseFloor, sbrHeaderData->sbr_amp_res); } } if (!clearOutput) { - /* - Tonality correction parameter extraction (inverse filtering level, noise floor additional sines). + Tonality correction parameter extraction (inverse filtering level, noise + floor additional sines). */ - FDKsbrEnc_TonCorrParamExtr(&hEnvChan->TonCorr, - hEnvChan->encEnvData.sbr_invf_mode_vec, - ed->noiseFloor, - &hEnvChan->encEnvData.addHarmonicFlag, - hEnvChan->encEnvData.addHarmonic, - sbrExtrEnv->envelopeCompensation, - ed->frame_info, - ed->transient_info, - h_con->freqBandTable[HI], - h_con->nSfb[HI], - hEnvChan->encEnvData.sbr_xpos_mode, - h_con->sbrSyntaxFlags); - + FDKsbrEnc_TonCorrParamExtr( + &hEnvChan->TonCorr, hEnvChan->encEnvData.sbr_invf_mode_vec, + ed->noiseFloor, &hEnvChan->encEnvData.addHarmonicFlag, + hEnvChan->encEnvData.addHarmonic, sbrExtrEnv->envelopeCompensation, + ed->frame_info, ed->transient_info, h_con->freqBandTable[HI], + h_con->nSfb[HI], hEnvChan->encEnvData.sbr_xpos_mode, + h_con->sbrSyntaxFlags); } /* Low energy in low band fix */ - if ( hEnvChan->sbrTransientDetector.prevLowBandEnergy < hEnvChan->sbrTransientDetector.prevHighBandEnergy - && hEnvChan->sbrTransientDetector.prevHighBandEnergy > FL2FX_DBL(0.03) - /* The fix needs the non-fast transient detector running. - It sets prevLowBandEnergy and prevHighBandEnergy. */ - && !(h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) - ) - { - int i; - + if (hEnvChan->sbrTransientDetector.prevLowBandEnergy < + hEnvChan->sbrTransientDetector.prevHighBandEnergy && + hEnvChan->sbrTransientDetector.prevHighBandEnergy > FL2FX_DBL(0.03) + /* The fix needs the non-fast transient detector running. + It sets prevLowBandEnergy and prevHighBandEnergy. */ + && !(h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY)) { hEnvChan->fLevelProtect = 1; - for (i=0; i<MAX_NUM_NOISE_VALUES; i++) + for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) hEnvChan->encEnvData.sbr_invf_mode_vec[i] = INVF_HIGH_LEVEL; } else { hEnvChan->fLevelProtect = 0; } - hEnvChan->encEnvData.sbr_invf_mode = hEnvChan->encEnvData.sbr_invf_mode_vec[0]; - - hEnvChan->encEnvData.noOfnoisebands = hEnvChan->TonCorr.sbrNoiseFloorEstimate.noNoiseBands; + hEnvChan->encEnvData.sbr_invf_mode = + hEnvChan->encEnvData.sbr_invf_mode_vec[0]; + hEnvChan->encEnvData.noOfnoisebands = + hEnvChan->TonCorr.sbrNoiseFloorEstimate.noNoiseBands; } /* ch */ - - - /* - Save number of scf bands per envelope - */ - for (ch = 0; ch < nChannels;ch++) { - for (i = 0; i < eData[ch].nEnvelopes; i++){ + /* + Save number of scf bands per envelope + */ + for (ch = 0; ch < nChannels; ch++) { + for (i = 0; i < eData[ch].nEnvelopes; i++) { h_envChan[ch]->encEnvData.noScfBands[i] = - (eData[ch].frame_info->freqRes[i] == FREQ_RES_HIGH ? h_con->nSfb[FREQ_RES_HIGH] : h_con->nSfb[FREQ_RES_LOW]); + (eData[ch].frame_info->freqRes[i] == FREQ_RES_HIGH + ? h_con->nSfb[FREQ_RES_HIGH] + : h_con->nSfb[FREQ_RES_LOW]); } } @@ -1368,165 +1359,169 @@ FDKsbrEnc_extractSbrEnvelope2 ( Extract envelope of current frame. */ switch (stereoMode) { - case SBR_MONO: - calculateSbrEnvelope (h_envChan[0]->sbrExtractEnvelope.YBuffer, NULL, - h_envChan[0]->sbrExtractEnvelope.YBufferScale, NULL, - eData[0].frame_info, eData[0].sfb_nrg, NULL, - h_con, h_envChan[0], SBR_MONO, NULL, YSzShift); - break; - case SBR_LEFT_RIGHT: - calculateSbrEnvelope (h_envChan[0]->sbrExtractEnvelope.YBuffer, NULL, - h_envChan[0]->sbrExtractEnvelope.YBufferScale, NULL, - eData[0].frame_info, eData[0].sfb_nrg, NULL, - h_con, h_envChan[0], SBR_MONO, NULL, YSzShift); - calculateSbrEnvelope (h_envChan[1]->sbrExtractEnvelope.YBuffer, NULL, - h_envChan[1]->sbrExtractEnvelope.YBufferScale, NULL, - eData[1].frame_info,eData[1].sfb_nrg, NULL, - h_con, h_envChan[1], SBR_MONO, NULL, YSzShift); - break; - case SBR_COUPLING: - calculateSbrEnvelope (h_envChan[0]->sbrExtractEnvelope.YBuffer, h_envChan[1]->sbrExtractEnvelope.YBuffer, - h_envChan[0]->sbrExtractEnvelope.YBufferScale, h_envChan[1]->sbrExtractEnvelope.YBufferScale, - eData[0].frame_info, eData[0].sfb_nrg, eData[1].sfb_nrg, - h_con, h_envChan[0], SBR_COUPLING, &fData->maxQuantError, YSzShift); - break; - case SBR_SWITCH_LRC: - calculateSbrEnvelope (h_envChan[0]->sbrExtractEnvelope.YBuffer, NULL, - h_envChan[0]->sbrExtractEnvelope.YBufferScale, NULL, - eData[0].frame_info, eData[0].sfb_nrg, NULL, - h_con, h_envChan[0], SBR_MONO, NULL, YSzShift); - calculateSbrEnvelope (h_envChan[1]->sbrExtractEnvelope.YBuffer, NULL, - h_envChan[1]->sbrExtractEnvelope.YBufferScale, NULL, - eData[1].frame_info, eData[1].sfb_nrg, NULL, - h_con, h_envChan[1], SBR_MONO,NULL, YSzShift); - calculateSbrEnvelope (h_envChan[0]->sbrExtractEnvelope.YBuffer, h_envChan[1]->sbrExtractEnvelope.YBuffer, - h_envChan[0]->sbrExtractEnvelope.YBufferScale, h_envChan[1]->sbrExtractEnvelope.YBufferScale, - eData[0].frame_info, eData[0].sfb_nrg_coupling, eData[1].sfb_nrg_coupling, - h_con, h_envChan[0], SBR_COUPLING, &fData->maxQuantError, YSzShift); - break; + case SBR_MONO: + calculateSbrEnvelope(h_envChan[0]->sbrExtractEnvelope.YBuffer, NULL, + h_envChan[0]->sbrExtractEnvelope.YBufferScale, NULL, + eData[0].frame_info, eData[0].sfb_nrg, NULL, h_con, + h_envChan[0], SBR_MONO, NULL, YSzShift); + break; + case SBR_LEFT_RIGHT: + calculateSbrEnvelope(h_envChan[0]->sbrExtractEnvelope.YBuffer, NULL, + h_envChan[0]->sbrExtractEnvelope.YBufferScale, NULL, + eData[0].frame_info, eData[0].sfb_nrg, NULL, h_con, + h_envChan[0], SBR_MONO, NULL, YSzShift); + calculateSbrEnvelope(h_envChan[1]->sbrExtractEnvelope.YBuffer, NULL, + h_envChan[1]->sbrExtractEnvelope.YBufferScale, NULL, + eData[1].frame_info, eData[1].sfb_nrg, NULL, h_con, + h_envChan[1], SBR_MONO, NULL, YSzShift); + break; + case SBR_COUPLING: + calculateSbrEnvelope(h_envChan[0]->sbrExtractEnvelope.YBuffer, + h_envChan[1]->sbrExtractEnvelope.YBuffer, + h_envChan[0]->sbrExtractEnvelope.YBufferScale, + h_envChan[1]->sbrExtractEnvelope.YBufferScale, + eData[0].frame_info, eData[0].sfb_nrg, + eData[1].sfb_nrg, h_con, h_envChan[0], SBR_COUPLING, + &fData->maxQuantError, YSzShift); + break; + case SBR_SWITCH_LRC: + calculateSbrEnvelope(h_envChan[0]->sbrExtractEnvelope.YBuffer, NULL, + h_envChan[0]->sbrExtractEnvelope.YBufferScale, NULL, + eData[0].frame_info, eData[0].sfb_nrg, NULL, h_con, + h_envChan[0], SBR_MONO, NULL, YSzShift); + calculateSbrEnvelope(h_envChan[1]->sbrExtractEnvelope.YBuffer, NULL, + h_envChan[1]->sbrExtractEnvelope.YBufferScale, NULL, + eData[1].frame_info, eData[1].sfb_nrg, NULL, h_con, + h_envChan[1], SBR_MONO, NULL, YSzShift); + calculateSbrEnvelope(h_envChan[0]->sbrExtractEnvelope.YBuffer, + h_envChan[1]->sbrExtractEnvelope.YBuffer, + h_envChan[0]->sbrExtractEnvelope.YBufferScale, + h_envChan[1]->sbrExtractEnvelope.YBufferScale, + eData[0].frame_info, eData[0].sfb_nrg_coupling, + eData[1].sfb_nrg_coupling, h_con, h_envChan[0], + SBR_COUPLING, &fData->maxQuantError, YSzShift); + break; } - - /* Noise floor quantisation and coding. */ switch (stereoMode) { - case SBR_MONO: - sbrNoiseFloorLevelsQuantisation(eData[0].noise_level, eData[0].noiseFloor, 0); - - FDKsbrEnc_codeEnvelope(eData[0].noise_level, fData->res, - &h_envChan[0]->sbrCodeNoiseFloor, - h_envChan[0]->encEnvData.domain_vec_noise, 0, - (eData[0].frame_info->nEnvelopes > 1 ? 2 : 1), 0, - sbrBitstreamData->HeaderActive); - - break; - case SBR_LEFT_RIGHT: - sbrNoiseFloorLevelsQuantisation(eData[0].noise_level,eData[0].noiseFloor, 0); - - FDKsbrEnc_codeEnvelope (eData[0].noise_level, fData->res, - &h_envChan[0]->sbrCodeNoiseFloor, - h_envChan[0]->encEnvData.domain_vec_noise, 0, - (eData[0].frame_info->nEnvelopes > 1 ? 2 : 1), 0, - sbrBitstreamData->HeaderActive); - - sbrNoiseFloorLevelsQuantisation(eData[1].noise_level,eData[1].noiseFloor, 0); - - FDKsbrEnc_codeEnvelope (eData[1].noise_level, fData->res, - &h_envChan[1]->sbrCodeNoiseFloor, - h_envChan[1]->encEnvData.domain_vec_noise, 0, - (eData[1].frame_info->nEnvelopes > 1 ? 2 : 1), 0, - sbrBitstreamData->HeaderActive); - - break; - - case SBR_COUPLING: - coupleNoiseFloor(eData[0].noiseFloor,eData[1].noiseFloor); - - sbrNoiseFloorLevelsQuantisation(eData[0].noise_level,eData[0].noiseFloor, 0); - - FDKsbrEnc_codeEnvelope (eData[0].noise_level, fData->res, - &h_envChan[0]->sbrCodeNoiseFloor, - h_envChan[0]->encEnvData.domain_vec_noise, 1, - (eData[0].frame_info->nEnvelopes > 1 ? 2 : 1), 0, - sbrBitstreamData->HeaderActive); - - sbrNoiseFloorLevelsQuantisation(eData[1].noise_level,eData[1].noiseFloor, 1); - - FDKsbrEnc_codeEnvelope (eData[1].noise_level, fData->res, - &h_envChan[1]->sbrCodeNoiseFloor, - h_envChan[1]->encEnvData.domain_vec_noise, 1, - (eData[1].frame_info->nEnvelopes > 1 ? 2 : 1), 1, - sbrBitstreamData->HeaderActive); - - break; - case SBR_SWITCH_LRC: - sbrNoiseFloorLevelsQuantisation(eData[0].noise_level,eData[0].noiseFloor, 0); - sbrNoiseFloorLevelsQuantisation(eData[1].noise_level,eData[1].noiseFloor, 0); - coupleNoiseFloor(eData[0].noiseFloor,eData[1].noiseFloor); - sbrNoiseFloorLevelsQuantisation(eData[0].noise_level_coupling,eData[0].noiseFloor, 0); - sbrNoiseFloorLevelsQuantisation(eData[1].noise_level_coupling,eData[1].noiseFloor, 1); - break; + case SBR_MONO: + sbrNoiseFloorLevelsQuantisation(eData[0].noise_level, eData[0].noiseFloor, + 0); + + FDKsbrEnc_codeEnvelope(eData[0].noise_level, fData->res, + &h_envChan[0]->sbrCodeNoiseFloor, + h_envChan[0]->encEnvData.domain_vec_noise, 0, + (eData[0].frame_info->nEnvelopes > 1 ? 2 : 1), 0, + sbrBitstreamData->HeaderActive); + + break; + case SBR_LEFT_RIGHT: + sbrNoiseFloorLevelsQuantisation(eData[0].noise_level, eData[0].noiseFloor, + 0); + + FDKsbrEnc_codeEnvelope(eData[0].noise_level, fData->res, + &h_envChan[0]->sbrCodeNoiseFloor, + h_envChan[0]->encEnvData.domain_vec_noise, 0, + (eData[0].frame_info->nEnvelopes > 1 ? 2 : 1), 0, + sbrBitstreamData->HeaderActive); + + sbrNoiseFloorLevelsQuantisation(eData[1].noise_level, eData[1].noiseFloor, + 0); + + FDKsbrEnc_codeEnvelope(eData[1].noise_level, fData->res, + &h_envChan[1]->sbrCodeNoiseFloor, + h_envChan[1]->encEnvData.domain_vec_noise, 0, + (eData[1].frame_info->nEnvelopes > 1 ? 2 : 1), 0, + sbrBitstreamData->HeaderActive); + + break; + + case SBR_COUPLING: + coupleNoiseFloor(eData[0].noiseFloor, eData[1].noiseFloor); + + sbrNoiseFloorLevelsQuantisation(eData[0].noise_level, eData[0].noiseFloor, + 0); + + FDKsbrEnc_codeEnvelope(eData[0].noise_level, fData->res, + &h_envChan[0]->sbrCodeNoiseFloor, + h_envChan[0]->encEnvData.domain_vec_noise, 1, + (eData[0].frame_info->nEnvelopes > 1 ? 2 : 1), 0, + sbrBitstreamData->HeaderActive); + + sbrNoiseFloorLevelsQuantisation(eData[1].noise_level, eData[1].noiseFloor, + 1); + + FDKsbrEnc_codeEnvelope(eData[1].noise_level, fData->res, + &h_envChan[1]->sbrCodeNoiseFloor, + h_envChan[1]->encEnvData.domain_vec_noise, 1, + (eData[1].frame_info->nEnvelopes > 1 ? 2 : 1), 1, + sbrBitstreamData->HeaderActive); + + break; + case SBR_SWITCH_LRC: + sbrNoiseFloorLevelsQuantisation(eData[0].noise_level, eData[0].noiseFloor, + 0); + sbrNoiseFloorLevelsQuantisation(eData[1].noise_level, eData[1].noiseFloor, + 0); + coupleNoiseFloor(eData[0].noiseFloor, eData[1].noiseFloor); + sbrNoiseFloorLevelsQuantisation(eData[0].noise_level_coupling, + eData[0].noiseFloor, 0); + sbrNoiseFloorLevelsQuantisation(eData[1].noise_level_coupling, + eData[1].noiseFloor, 1); + break; } - - /* Encode envelope of current frame. */ switch (stereoMode) { - case SBR_MONO: - sbrHeaderData->coupling = 0; - h_envChan[0]->encEnvData.balance = 0; - FDKsbrEnc_codeEnvelope (eData[0].sfb_nrg, eData[0].frame_info->freqRes, - &h_envChan[0]->sbrCodeEnvelope, - h_envChan[0]->encEnvData.domain_vec, - sbrHeaderData->coupling, - eData[0].frame_info->nEnvelopes, 0, - sbrBitstreamData->HeaderActive); - break; - case SBR_LEFT_RIGHT: - sbrHeaderData->coupling = 0; - - h_envChan[0]->encEnvData.balance = 0; - h_envChan[1]->encEnvData.balance = 0; - - - FDKsbrEnc_codeEnvelope (eData[0].sfb_nrg, eData[0].frame_info->freqRes, - &h_envChan[0]->sbrCodeEnvelope, - h_envChan[0]->encEnvData.domain_vec, - sbrHeaderData->coupling, - eData[0].frame_info->nEnvelopes, 0, - sbrBitstreamData->HeaderActive); - FDKsbrEnc_codeEnvelope (eData[1].sfb_nrg, eData[1].frame_info->freqRes, - &h_envChan[1]->sbrCodeEnvelope, - h_envChan[1]->encEnvData.domain_vec, - sbrHeaderData->coupling, - eData[1].frame_info->nEnvelopes, 0, - sbrBitstreamData->HeaderActive); - break; - case SBR_COUPLING: - sbrHeaderData->coupling = 1; - h_envChan[0]->encEnvData.balance = 0; - h_envChan[1]->encEnvData.balance = 1; - - FDKsbrEnc_codeEnvelope (eData[0].sfb_nrg, eData[0].frame_info->freqRes, - &h_envChan[0]->sbrCodeEnvelope, - h_envChan[0]->encEnvData.domain_vec, - sbrHeaderData->coupling, - eData[0].frame_info->nEnvelopes, 0, - sbrBitstreamData->HeaderActive); - FDKsbrEnc_codeEnvelope (eData[1].sfb_nrg, eData[1].frame_info->freqRes, - &h_envChan[1]->sbrCodeEnvelope, - h_envChan[1]->encEnvData.domain_vec, - sbrHeaderData->coupling, - eData[1].frame_info->nEnvelopes, 1, - sbrBitstreamData->HeaderActive); - break; - case SBR_SWITCH_LRC: - { + case SBR_MONO: + sbrHeaderData->coupling = 0; + h_envChan[0]->encEnvData.balance = 0; + FDKsbrEnc_codeEnvelope( + eData[0].sfb_nrg, eData[0].frame_info->freqRes, + &h_envChan[0]->sbrCodeEnvelope, h_envChan[0]->encEnvData.domain_vec, + sbrHeaderData->coupling, eData[0].frame_info->nEnvelopes, 0, + sbrBitstreamData->HeaderActive); + break; + case SBR_LEFT_RIGHT: + sbrHeaderData->coupling = 0; + + h_envChan[0]->encEnvData.balance = 0; + h_envChan[1]->encEnvData.balance = 0; + + FDKsbrEnc_codeEnvelope( + eData[0].sfb_nrg, eData[0].frame_info->freqRes, + &h_envChan[0]->sbrCodeEnvelope, h_envChan[0]->encEnvData.domain_vec, + sbrHeaderData->coupling, eData[0].frame_info->nEnvelopes, 0, + sbrBitstreamData->HeaderActive); + FDKsbrEnc_codeEnvelope( + eData[1].sfb_nrg, eData[1].frame_info->freqRes, + &h_envChan[1]->sbrCodeEnvelope, h_envChan[1]->encEnvData.domain_vec, + sbrHeaderData->coupling, eData[1].frame_info->nEnvelopes, 0, + sbrBitstreamData->HeaderActive); + break; + case SBR_COUPLING: + sbrHeaderData->coupling = 1; + h_envChan[0]->encEnvData.balance = 0; + h_envChan[1]->encEnvData.balance = 1; + + FDKsbrEnc_codeEnvelope( + eData[0].sfb_nrg, eData[0].frame_info->freqRes, + &h_envChan[0]->sbrCodeEnvelope, h_envChan[0]->encEnvData.domain_vec, + sbrHeaderData->coupling, eData[0].frame_info->nEnvelopes, 0, + sbrBitstreamData->HeaderActive); + FDKsbrEnc_codeEnvelope( + eData[1].sfb_nrg, eData[1].frame_info->freqRes, + &h_envChan[1]->sbrCodeEnvelope, h_envChan[1]->encEnvData.domain_vec, + sbrHeaderData->coupling, eData[1].frame_info->nEnvelopes, 1, + sbrBitstreamData->HeaderActive); + break; + case SBR_SWITCH_LRC: { INT payloadbitsLR; INT payloadbitsCOUPLING; @@ -1541,15 +1536,18 @@ FDKsbrEnc_extractSbrEnvelope2 ( INT tempFlagLeft = 0; /* - Store previous values, in order to be able to "undo" what is being done. + Store previous values, in order to be able to "undo" what is being + done. */ - for(ch = 0; ch < nChannels;ch++){ - FDKmemcpy (sfbNrgPrevTemp[ch], h_envChan[ch]->sbrCodeEnvelope.sfb_nrg_prev, - MAX_FREQ_COEFFS * sizeof (SCHAR)); + for (ch = 0; ch < nChannels; ch++) { + FDKmemcpy(sfbNrgPrevTemp[ch], + h_envChan[ch]->sbrCodeEnvelope.sfb_nrg_prev, + MAX_FREQ_COEFFS * sizeof(SCHAR)); - FDKmemcpy (noisePrevTemp[ch], h_envChan[ch]->sbrCodeNoiseFloor.sfb_nrg_prev, - MAX_NUM_NOISE_COEFFS * sizeof (SCHAR)); + FDKmemcpy(noisePrevTemp[ch], + h_envChan[ch]->sbrCodeNoiseFloor.sfb_nrg_prev, + MAX_NUM_NOISE_COEFFS * sizeof(SCHAR)); upDateNrgTemp[ch] = h_envChan[ch]->sbrCodeEnvelope.upDate; upDateNoiseTemp[ch] = h_envChan[ch]->sbrCodeNoiseFloor.upDate; @@ -1558,247 +1556,233 @@ FDKsbrEnc_extractSbrEnvelope2 ( forbid time coding in the first envelope in case of a different previous stereomode */ - if(sbrHeaderData->prev_coupling){ + if (sbrHeaderData->prev_coupling) { h_envChan[ch]->sbrCodeEnvelope.upDate = 0; h_envChan[ch]->sbrCodeNoiseFloor.upDate = 0; } } /* ch */ - /* Code ordinary Left/Right stereo */ - FDKsbrEnc_codeEnvelope (eData[0].sfb_nrg, eData[0].frame_info->freqRes, - &h_envChan[0]->sbrCodeEnvelope, - h_envChan[0]->encEnvData.domain_vec, 0, - eData[0].frame_info->nEnvelopes, 0, - sbrBitstreamData->HeaderActive); - FDKsbrEnc_codeEnvelope (eData[1].sfb_nrg, eData[1].frame_info->freqRes, - &h_envChan[1]->sbrCodeEnvelope, - h_envChan[1]->encEnvData.domain_vec, 0, - eData[1].frame_info->nEnvelopes, 0, - sbrBitstreamData->HeaderActive); + FDKsbrEnc_codeEnvelope(eData[0].sfb_nrg, eData[0].frame_info->freqRes, + &h_envChan[0]->sbrCodeEnvelope, + h_envChan[0]->encEnvData.domain_vec, 0, + eData[0].frame_info->nEnvelopes, 0, + sbrBitstreamData->HeaderActive); + FDKsbrEnc_codeEnvelope(eData[1].sfb_nrg, eData[1].frame_info->freqRes, + &h_envChan[1]->sbrCodeEnvelope, + h_envChan[1]->encEnvData.domain_vec, 0, + eData[1].frame_info->nEnvelopes, 0, + sbrBitstreamData->HeaderActive); c = 0; for (i = 0; i < eData[0].nEnvelopes; i++) { - for (j = 0; j < h_envChan[0]->encEnvData.noScfBands[i]; j++) - { - h_envChan[0]->encEnvData.ienvelope[i][j] = eData[0].sfb_nrg[c]; - h_envChan[1]->encEnvData.ienvelope[i][j] = eData[1].sfb_nrg[c]; - c++; - } + for (j = 0; j < h_envChan[0]->encEnvData.noScfBands[i]; j++) { + h_envChan[0]->encEnvData.ienvelope[i][j] = eData[0].sfb_nrg[c]; + h_envChan[1]->encEnvData.ienvelope[i][j] = eData[1].sfb_nrg[c]; + c++; + } } - - - FDKsbrEnc_codeEnvelope (eData[0].noise_level, fData->res, - &h_envChan[0]->sbrCodeNoiseFloor, - h_envChan[0]->encEnvData.domain_vec_noise, 0, - (eData[0].frame_info->nEnvelopes > 1 ? 2 : 1), 0, - sbrBitstreamData->HeaderActive); - + FDKsbrEnc_codeEnvelope(eData[0].noise_level, fData->res, + &h_envChan[0]->sbrCodeNoiseFloor, + h_envChan[0]->encEnvData.domain_vec_noise, 0, + (eData[0].frame_info->nEnvelopes > 1 ? 2 : 1), 0, + sbrBitstreamData->HeaderActive); for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) h_envChan[0]->encEnvData.sbr_noise_levels[i] = eData[0].noise_level[i]; - - FDKsbrEnc_codeEnvelope (eData[1].noise_level, fData->res, - &h_envChan[1]->sbrCodeNoiseFloor, - h_envChan[1]->encEnvData.domain_vec_noise, 0, - (eData[1].frame_info->nEnvelopes > 1 ? 2 : 1), 0, - sbrBitstreamData->HeaderActive); + FDKsbrEnc_codeEnvelope(eData[1].noise_level, fData->res, + &h_envChan[1]->sbrCodeNoiseFloor, + h_envChan[1]->encEnvData.domain_vec_noise, 0, + (eData[1].frame_info->nEnvelopes > 1 ? 2 : 1), 0, + sbrBitstreamData->HeaderActive); for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) h_envChan[1]->encEnvData.sbr_noise_levels[i] = eData[1].noise_level[i]; - sbrHeaderData->coupling = 0; h_envChan[0]->encEnvData.balance = 0; h_envChan[1]->encEnvData.balance = 0; - payloadbitsLR = FDKsbrEnc_CountSbrChannelPairElement (sbrHeaderData, - hParametricStereo, - sbrBitstreamData, - &h_envChan[0]->encEnvData, - &h_envChan[1]->encEnvData, - hCmonData, - h_con->sbrSyntaxFlags); + payloadbitsLR = FDKsbrEnc_CountSbrChannelPairElement( + sbrHeaderData, hParametricStereo, sbrBitstreamData, + &h_envChan[0]->encEnvData, &h_envChan[1]->encEnvData, hCmonData, + h_con->sbrSyntaxFlags); /* swap saved stored with current values */ - for(ch = 0; ch < nChannels;ch++){ - INT itmp; - for(i=0;i<MAX_FREQ_COEFFS;i++){ + for (ch = 0; ch < nChannels; ch++) { + INT itmp; + for (i = 0; i < MAX_FREQ_COEFFS; i++) { /* swap sfb energies */ - itmp = h_envChan[ch]->sbrCodeEnvelope.sfb_nrg_prev[i]; - h_envChan[ch]->sbrCodeEnvelope.sfb_nrg_prev[i]=sfbNrgPrevTemp[ch][i]; - sfbNrgPrevTemp[ch][i]=itmp; + itmp = h_envChan[ch]->sbrCodeEnvelope.sfb_nrg_prev[i]; + h_envChan[ch]->sbrCodeEnvelope.sfb_nrg_prev[i] = + sfbNrgPrevTemp[ch][i]; + sfbNrgPrevTemp[ch][i] = itmp; } - for(i=0;i<MAX_NUM_NOISE_COEFFS;i++){ + for (i = 0; i < MAX_NUM_NOISE_COEFFS; i++) { /* swap noise energies */ - itmp = h_envChan[ch]->sbrCodeNoiseFloor.sfb_nrg_prev[i]; - h_envChan[ch]->sbrCodeNoiseFloor.sfb_nrg_prev[i]=noisePrevTemp[ch][i]; - noisePrevTemp[ch][i]=itmp; - } + itmp = h_envChan[ch]->sbrCodeNoiseFloor.sfb_nrg_prev[i]; + h_envChan[ch]->sbrCodeNoiseFloor.sfb_nrg_prev[i] = + noisePrevTemp[ch][i]; + noisePrevTemp[ch][i] = itmp; + } /* swap update flags */ - itmp = h_envChan[ch]->sbrCodeEnvelope.upDate; - h_envChan[ch]->sbrCodeEnvelope.upDate=upDateNrgTemp[ch]; + itmp = h_envChan[ch]->sbrCodeEnvelope.upDate; + h_envChan[ch]->sbrCodeEnvelope.upDate = upDateNrgTemp[ch]; upDateNrgTemp[ch] = itmp; - itmp = h_envChan[ch]->sbrCodeNoiseFloor.upDate; - h_envChan[ch]->sbrCodeNoiseFloor.upDate=upDateNoiseTemp[ch]; - upDateNoiseTemp[ch]=itmp; + itmp = h_envChan[ch]->sbrCodeNoiseFloor.upDate; + h_envChan[ch]->sbrCodeNoiseFloor.upDate = upDateNoiseTemp[ch]; + upDateNoiseTemp[ch] = itmp; /* save domain vecs */ - FDKmemcpy(domainVecTemp[ch],h_envChan[ch]->encEnvData.domain_vec,sizeof(INT)*MAX_ENVELOPES); - FDKmemcpy(domainVecNoiseTemp[ch],h_envChan[ch]->encEnvData.domain_vec_noise,sizeof(INT)*MAX_ENVELOPES); + FDKmemcpy(domainVecTemp[ch], h_envChan[ch]->encEnvData.domain_vec, + sizeof(INT) * MAX_ENVELOPES); + FDKmemcpy(domainVecNoiseTemp[ch], + h_envChan[ch]->encEnvData.domain_vec_noise, + sizeof(INT) * MAX_ENVELOPES); /* forbid time coding in the first envelope in case of a different previous stereomode */ - if(!sbrHeaderData->prev_coupling){ + if (!sbrHeaderData->prev_coupling) { h_envChan[ch]->sbrCodeEnvelope.upDate = 0; h_envChan[ch]->sbrCodeNoiseFloor.upDate = 0; } } /* ch */ - /* Coupling */ - FDKsbrEnc_codeEnvelope (eData[0].sfb_nrg_coupling, eData[0].frame_info->freqRes, - &h_envChan[0]->sbrCodeEnvelope, - h_envChan[0]->encEnvData.domain_vec, 1, - eData[0].frame_info->nEnvelopes, 0, - sbrBitstreamData->HeaderActive); - - FDKsbrEnc_codeEnvelope (eData[1].sfb_nrg_coupling, eData[1].frame_info->freqRes, - &h_envChan[1]->sbrCodeEnvelope, - h_envChan[1]->encEnvData.domain_vec, 1, - eData[1].frame_info->nEnvelopes, 1, - sbrBitstreamData->HeaderActive); + FDKsbrEnc_codeEnvelope( + eData[0].sfb_nrg_coupling, eData[0].frame_info->freqRes, + &h_envChan[0]->sbrCodeEnvelope, h_envChan[0]->encEnvData.domain_vec, + 1, eData[0].frame_info->nEnvelopes, 0, + sbrBitstreamData->HeaderActive); + FDKsbrEnc_codeEnvelope( + eData[1].sfb_nrg_coupling, eData[1].frame_info->freqRes, + &h_envChan[1]->sbrCodeEnvelope, h_envChan[1]->encEnvData.domain_vec, + 1, eData[1].frame_info->nEnvelopes, 1, + sbrBitstreamData->HeaderActive); c = 0; for (i = 0; i < eData[0].nEnvelopes; i++) { for (j = 0; j < h_envChan[0]->encEnvData.noScfBands[i]; j++) { - h_envChan[0]->encEnvData.ienvelope[i][j] = eData[0].sfb_nrg_coupling[c]; - h_envChan[1]->encEnvData.ienvelope[i][j] = eData[1].sfb_nrg_coupling[c]; + h_envChan[0]->encEnvData.ienvelope[i][j] = + eData[0].sfb_nrg_coupling[c]; + h_envChan[1]->encEnvData.ienvelope[i][j] = + eData[1].sfb_nrg_coupling[c]; c++; } } - FDKsbrEnc_codeEnvelope (eData[0].noise_level_coupling, fData->res, - &h_envChan[0]->sbrCodeNoiseFloor, - h_envChan[0]->encEnvData.domain_vec_noise, 1, - (eData[0].frame_info->nEnvelopes > 1 ? 2 : 1), 0, - sbrBitstreamData->HeaderActive); + FDKsbrEnc_codeEnvelope(eData[0].noise_level_coupling, fData->res, + &h_envChan[0]->sbrCodeNoiseFloor, + h_envChan[0]->encEnvData.domain_vec_noise, 1, + (eData[0].frame_info->nEnvelopes > 1 ? 2 : 1), 0, + sbrBitstreamData->HeaderActive); for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) - h_envChan[0]->encEnvData.sbr_noise_levels[i] = eData[0].noise_level_coupling[i]; - + h_envChan[0]->encEnvData.sbr_noise_levels[i] = + eData[0].noise_level_coupling[i]; - FDKsbrEnc_codeEnvelope (eData[1].noise_level_coupling, fData->res, - &h_envChan[1]->sbrCodeNoiseFloor, - h_envChan[1]->encEnvData.domain_vec_noise, 1, - (eData[1].frame_info->nEnvelopes > 1 ? 2 : 1), 1, - sbrBitstreamData->HeaderActive); + FDKsbrEnc_codeEnvelope(eData[1].noise_level_coupling, fData->res, + &h_envChan[1]->sbrCodeNoiseFloor, + h_envChan[1]->encEnvData.domain_vec_noise, 1, + (eData[1].frame_info->nEnvelopes > 1 ? 2 : 1), 1, + sbrBitstreamData->HeaderActive); for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) - h_envChan[1]->encEnvData.sbr_noise_levels[i] = eData[1].noise_level_coupling[i]; + h_envChan[1]->encEnvData.sbr_noise_levels[i] = + eData[1].noise_level_coupling[i]; sbrHeaderData->coupling = 1; - h_envChan[0]->encEnvData.balance = 0; - h_envChan[1]->encEnvData.balance = 1; + h_envChan[0]->encEnvData.balance = 0; + h_envChan[1]->encEnvData.balance = 1; - tempFlagLeft = h_envChan[0]->encEnvData.addHarmonicFlag; + tempFlagLeft = h_envChan[0]->encEnvData.addHarmonicFlag; tempFlagRight = h_envChan[1]->encEnvData.addHarmonicFlag; - payloadbitsCOUPLING = - FDKsbrEnc_CountSbrChannelPairElement (sbrHeaderData, - hParametricStereo, - sbrBitstreamData, - &h_envChan[0]->encEnvData, - &h_envChan[1]->encEnvData, - hCmonData, - h_con->sbrSyntaxFlags); - + payloadbitsCOUPLING = FDKsbrEnc_CountSbrChannelPairElement( + sbrHeaderData, hParametricStereo, sbrBitstreamData, + &h_envChan[0]->encEnvData, &h_envChan[1]->encEnvData, hCmonData, + h_con->sbrSyntaxFlags); h_envChan[0]->encEnvData.addHarmonicFlag = tempFlagLeft; h_envChan[1]->encEnvData.addHarmonicFlag = tempFlagRight; if (payloadbitsCOUPLING < payloadbitsLR) { + /* + copy coded coupling envelope and noise data to l/r + */ + for (ch = 0; ch < nChannels; ch++) { + SBR_ENV_TEMP_DATA *ed = &eData[ch]; + FDKmemcpy(ed->sfb_nrg, ed->sfb_nrg_coupling, + MAX_NUM_ENVELOPE_VALUES * sizeof(SCHAR)); + FDKmemcpy(ed->noise_level, ed->noise_level_coupling, + MAX_NUM_NOISE_VALUES * sizeof(SCHAR)); + } - /* - copy coded coupling envelope and noise data to l/r - */ - for(ch = 0; ch < nChannels;ch++){ - SBR_ENV_TEMP_DATA *ed = &eData[ch]; - FDKmemcpy (ed->sfb_nrg, ed->sfb_nrg_coupling, - MAX_NUM_ENVELOPE_VALUES * sizeof (SCHAR)); - FDKmemcpy (ed->noise_level, ed->noise_level_coupling, - MAX_NUM_NOISE_VALUES * sizeof (SCHAR)); - } - - sbrHeaderData->coupling = 1; - h_envChan[0]->encEnvData.balance = 0; - h_envChan[1]->encEnvData.balance = 1; - } - else{ - /* - restore saved l/r items - */ - for(ch = 0; ch < nChannels;ch++){ - - FDKmemcpy (h_envChan[ch]->sbrCodeEnvelope.sfb_nrg_prev, - sfbNrgPrevTemp[ch], MAX_FREQ_COEFFS * sizeof (SCHAR)); - - h_envChan[ch]->sbrCodeEnvelope.upDate = upDateNrgTemp[ch]; + sbrHeaderData->coupling = 1; + h_envChan[0]->encEnvData.balance = 0; + h_envChan[1]->encEnvData.balance = 1; + } else { + /* + restore saved l/r items + */ + for (ch = 0; ch < nChannels; ch++) { + FDKmemcpy(h_envChan[ch]->sbrCodeEnvelope.sfb_nrg_prev, + sfbNrgPrevTemp[ch], MAX_FREQ_COEFFS * sizeof(SCHAR)); - FDKmemcpy (h_envChan[ch]->sbrCodeNoiseFloor.sfb_nrg_prev, - noisePrevTemp[ch], MAX_NUM_NOISE_COEFFS * sizeof (SCHAR)); + h_envChan[ch]->sbrCodeEnvelope.upDate = upDateNrgTemp[ch]; - FDKmemcpy (h_envChan[ch]->encEnvData.domain_vec,domainVecTemp[ch],sizeof(INT)*MAX_ENVELOPES); - FDKmemcpy (h_envChan[ch]->encEnvData.domain_vec_noise,domainVecNoiseTemp[ch],sizeof(INT)*MAX_ENVELOPES); + FDKmemcpy(h_envChan[ch]->sbrCodeNoiseFloor.sfb_nrg_prev, + noisePrevTemp[ch], MAX_NUM_NOISE_COEFFS * sizeof(SCHAR)); - h_envChan[ch]->sbrCodeNoiseFloor.upDate = upDateNoiseTemp[ch]; - } + FDKmemcpy(h_envChan[ch]->encEnvData.domain_vec, domainVecTemp[ch], + sizeof(INT) * MAX_ENVELOPES); + FDKmemcpy(h_envChan[ch]->encEnvData.domain_vec_noise, + domainVecNoiseTemp[ch], sizeof(INT) * MAX_ENVELOPES); - sbrHeaderData->coupling = 0; - h_envChan[0]->encEnvData.balance = 0; - h_envChan[1]->encEnvData.balance = 0; + h_envChan[ch]->sbrCodeNoiseFloor.upDate = upDateNoiseTemp[ch]; } - } - break; - } /* switch */ + sbrHeaderData->coupling = 0; + h_envChan[0]->encEnvData.balance = 0; + h_envChan[1]->encEnvData.balance = 0; + } + } break; + } /* switch */ /* tell the envelope encoders how long it has been, since we last sent a frame starting with a dF-coded envelope */ - if (stereoMode == SBR_MONO ) { + if (stereoMode == SBR_MONO) { if (h_envChan[0]->encEnvData.domain_vec[0] == TIME) h_envChan[0]->sbrCodeEnvelope.dF_edge_incr_fac++; else h_envChan[0]->sbrCodeEnvelope.dF_edge_incr_fac = 0; - } - else { + } else { if (h_envChan[0]->encEnvData.domain_vec[0] == TIME || h_envChan[1]->encEnvData.domain_vec[0] == TIME) { h_envChan[0]->sbrCodeEnvelope.dF_edge_incr_fac++; h_envChan[1]->sbrCodeEnvelope.dF_edge_incr_fac++; - } - else { + } else { h_envChan[0]->sbrCodeEnvelope.dF_edge_incr_fac = 0; h_envChan[1]->sbrCodeEnvelope.dF_edge_incr_fac = 0; } @@ -1807,7 +1791,7 @@ FDKsbrEnc_extractSbrEnvelope2 ( /* Send the encoded data to the bitstream */ - for(ch = 0; ch < nChannels;ch++){ + for (ch = 0; ch < nChannels; ch++) { SBR_ENV_TEMP_DATA *ed = &eData[ch]; c = 0; for (i = 0; i < ed->nEnvelopes; i++) { @@ -1817,45 +1801,38 @@ FDKsbrEnc_extractSbrEnvelope2 ( c++; } } - for (i = 0; i < MAX_NUM_NOISE_VALUES; i++){ + for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) { h_envChan[ch]->encEnvData.sbr_noise_levels[i] = ed->noise_level[i]; } - }/* ch */ - + } /* ch */ /* Write bitstream */ if (nChannels == 2) { - FDKsbrEnc_WriteEnvChannelPairElement(sbrHeaderData, - hParametricStereo, - sbrBitstreamData, - &h_envChan[0]->encEnvData, - &h_envChan[1]->encEnvData, - hCmonData, - h_con->sbrSyntaxFlags); - } - else { - FDKsbrEnc_WriteEnvSingleChannelElement(sbrHeaderData, - hParametricStereo, - sbrBitstreamData, - &h_envChan[0]->encEnvData, - hCmonData, - h_con->sbrSyntaxFlags); + FDKsbrEnc_WriteEnvChannelPairElement( + sbrHeaderData, hParametricStereo, sbrBitstreamData, + &h_envChan[0]->encEnvData, &h_envChan[1]->encEnvData, hCmonData, + h_con->sbrSyntaxFlags); + } else { + FDKsbrEnc_WriteEnvSingleChannelElement( + sbrHeaderData, hParametricStereo, sbrBitstreamData, + &h_envChan[0]->encEnvData, hCmonData, h_con->sbrSyntaxFlags); } /* * Update buffers. */ - for (ch=0; ch<nChannels; ch++) - { - int YBufferLength = h_envChan[ch]->sbrExtractEnvelope.no_cols >> h_envChan[ch]->sbrExtractEnvelope.YBufferSzShift; - for (i = 0; i < h_envChan[ch]->sbrExtractEnvelope.YBufferWriteOffset; i++) { - FDKmemcpy(h_envChan[ch]->sbrExtractEnvelope.YBuffer[i], - h_envChan[ch]->sbrExtractEnvelope.YBuffer[i + YBufferLength], - sizeof(FIXP_DBL)*QMF_CHANNELS); - } - h_envChan[ch]->sbrExtractEnvelope.YBufferScale[0] = h_envChan[ch]->sbrExtractEnvelope.YBufferScale[1]; + for (ch = 0; ch < nChannels; ch++) { + int YBufferLength = h_envChan[ch]->sbrExtractEnvelope.no_cols >> + h_envChan[ch]->sbrExtractEnvelope.YBufferSzShift; + for (i = 0; i < h_envChan[ch]->sbrExtractEnvelope.YBufferWriteOffset; i++) { + FDKmemcpy(h_envChan[ch]->sbrExtractEnvelope.YBuffer[i], + h_envChan[ch]->sbrExtractEnvelope.YBuffer[i + YBufferLength], + sizeof(FIXP_DBL) * 64); + } + h_envChan[ch]->sbrExtractEnvelope.YBufferScale[0] = + h_envChan[ch]->sbrExtractEnvelope.YBufferScale[1]; } sbrHeaderData->prev_coupling = sbrHeaderData->coupling; @@ -1869,40 +1846,43 @@ FDKsbrEnc_extractSbrEnvelope2 ( \return error status ****************************************************************************/ -INT -FDKsbrEnc_CreateExtractSbrEnvelope (HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut, - INT channel - ,INT chInEl - ,UCHAR* dynamic_RAM - ) -{ +INT FDKsbrEnc_CreateExtractSbrEnvelope(HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut, + INT channel, INT chInEl, + UCHAR *dynamic_RAM) { INT i; - FIXP_DBL* YBuffer = GetRam_Sbr_envYBuffer(channel); + FIXP_DBL *rBuffer, *iBuffer; + INT n; + FIXP_DBL *YBufferDyn; - FDKmemclear(hSbrCut,sizeof(SBR_EXTRACT_ENVELOPE)); - hSbrCut->p_YBuffer = YBuffer; + FDKmemclear(hSbrCut, sizeof(SBR_EXTRACT_ENVELOPE)); + if (NULL == (hSbrCut->p_YBuffer = GetRam_Sbr_envYBuffer(channel))) { + goto bail; + } - for (i = 0; i < (QMF_MAX_TIME_SLOTS>>1); i++) { - hSbrCut->YBuffer[i] = YBuffer + (i*QMF_CHANNELS); + for (i = 0; i < (32 >> 1); i++) { + hSbrCut->YBuffer[i] = hSbrCut->p_YBuffer + (i * 64); } - FIXP_DBL *YBufferDyn = GetRam_Sbr_envYBuffer(chInEl, dynamic_RAM); - INT n=0; - for (; i < QMF_MAX_TIME_SLOTS; i++,n++) { - hSbrCut->YBuffer[i] = YBufferDyn + (n*QMF_CHANNELS); + YBufferDyn = GetRam_Sbr_envYBuffer(chInEl, dynamic_RAM); + for (n = 0; i < 32; i++, n++) { + hSbrCut->YBuffer[i] = YBufferDyn + (n * 64); } - FIXP_DBL* rBuffer = GetRam_Sbr_envRBuffer(0, dynamic_RAM); - FIXP_DBL* iBuffer = GetRam_Sbr_envIBuffer(0, dynamic_RAM); + rBuffer = GetRam_Sbr_envRBuffer(0, dynamic_RAM); + iBuffer = GetRam_Sbr_envIBuffer(0, dynamic_RAM); - for (i = 0; i < QMF_MAX_TIME_SLOTS; i++) { - hSbrCut->rBuffer[i] = rBuffer + (i*QMF_CHANNELS); - hSbrCut->iBuffer[i] = iBuffer + (i*QMF_CHANNELS); + for (i = 0; i < 32; i++) { + hSbrCut->rBuffer[i] = rBuffer + (i * 64); + hSbrCut->iBuffer[i] = iBuffer + (i * 64); } return 0; -} +bail: + FDKsbrEnc_deleteExtractSbrEnvelope(hSbrCut); + + return -1; +} /***************************************************************************/ /*! @@ -1912,36 +1892,22 @@ FDKsbrEnc_CreateExtractSbrEnvelope (HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut, \return error status ****************************************************************************/ -INT -FDKsbrEnc_InitExtractSbrEnvelope (HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut, - int no_cols, - int no_rows, - int start_index, - int time_slots, - int time_step, - int tran_off, - ULONG statesInitFlag - ,int chInEl - ,UCHAR* dynamic_RAM - ,UINT sbrSyntaxFlags - ) -{ +INT FDKsbrEnc_InitExtractSbrEnvelope(HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut, + int no_cols, int no_rows, int start_index, + int time_slots, int time_step, + int tran_off, ULONG statesInitFlag, + int chInEl, UCHAR *dynamic_RAM, + UINT sbrSyntaxFlags) { int YBufferLength, rBufferLength; int i; if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { int off = TRANSIENT_OFFSET_LD; -#ifndef FULL_DELAY - hSbrCut->YBufferWriteOffset = (no_cols>>1)+off*time_step; -#else - hSbrCut->YBufferWriteOffset = no_cols+off*time_step; -#endif - } else - { - hSbrCut->YBufferWriteOffset = tran_off*time_step; + hSbrCut->YBufferWriteOffset = (no_cols >> 1) + off * time_step; + } else { + hSbrCut->YBufferWriteOffset = tran_off * time_step; } - hSbrCut->rBufferReadOffset = 0; - + hSbrCut->rBufferReadOffset = 0; YBufferLength = hSbrCut->YBufferWriteOffset + no_cols; rBufferLength = no_cols; @@ -1949,7 +1915,6 @@ FDKsbrEnc_InitExtractSbrEnvelope (HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut, hSbrCut->pre_transient_info[0] = 0; hSbrCut->pre_transient_info[1] = 0; - hSbrCut->no_cols = no_cols; hSbrCut->no_rows = no_rows; hSbrCut->start_index = start_index; @@ -1957,7 +1922,7 @@ FDKsbrEnc_InitExtractSbrEnvelope (HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut, hSbrCut->time_slots = time_slots; hSbrCut->time_step = time_step; - FDK_ASSERT(no_rows <= QMF_CHANNELS); + FDK_ASSERT(no_rows <= 64); /* Use half the Energy values if time step is 2 or greater */ if (time_step >= 2) @@ -1965,40 +1930,37 @@ FDKsbrEnc_InitExtractSbrEnvelope (HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut, else hSbrCut->YBufferSzShift = 0; - YBufferLength >>= hSbrCut->YBufferSzShift; + YBufferLength >>= hSbrCut->YBufferSzShift; hSbrCut->YBufferWriteOffset >>= hSbrCut->YBufferSzShift; - FDK_ASSERT(YBufferLength<=QMF_MAX_TIME_SLOTS); + FDK_ASSERT(YBufferLength <= 32); FIXP_DBL *YBufferDyn = GetRam_Sbr_envYBuffer(chInEl, dynamic_RAM); - INT n=0; - for (i=(QMF_MAX_TIME_SLOTS>>1); i < QMF_MAX_TIME_SLOTS; i++,n++) { - hSbrCut->YBuffer[i] = YBufferDyn + (n*QMF_CHANNELS); + INT n = 0; + for (i = (32 >> 1); i < 32; i++, n++) { + hSbrCut->YBuffer[i] = YBufferDyn + (n * 64); } - if(statesInitFlag) { - for (i=0; i<YBufferLength; i++) { - FDKmemclear( hSbrCut->YBuffer[i],QMF_CHANNELS*sizeof(FIXP_DBL)); + if (statesInitFlag) { + for (i = 0; i < YBufferLength; i++) { + FDKmemclear(hSbrCut->YBuffer[i], 64 * sizeof(FIXP_DBL)); } } for (i = 0; i < rBufferLength; i++) { - FDKmemclear( hSbrCut->rBuffer[i],QMF_CHANNELS*sizeof(FIXP_DBL)); - FDKmemclear( hSbrCut->iBuffer[i],QMF_CHANNELS*sizeof(FIXP_DBL)); + FDKmemclear(hSbrCut->rBuffer[i], 64 * sizeof(FIXP_DBL)); + FDKmemclear(hSbrCut->iBuffer[i], 64 * sizeof(FIXP_DBL)); } - FDKmemclear (hSbrCut->envelopeCompensation,sizeof(UCHAR)*MAX_FREQ_COEFFS); + FDKmemclear(hSbrCut->envelopeCompensation, sizeof(UCHAR) * MAX_FREQ_COEFFS); - if(statesInitFlag) { - hSbrCut->YBufferScale[0] = hSbrCut->YBufferScale[1] = FRACT_BITS-1; + if (statesInitFlag) { + hSbrCut->YBufferScale[0] = hSbrCut->YBufferScale[1] = FRACT_BITS - 1; } return (0); } - - - /***************************************************************************/ /*! @@ -2008,23 +1970,16 @@ FDKsbrEnc_InitExtractSbrEnvelope (HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut, ****************************************************************************/ -void -FDKsbrEnc_deleteExtractSbrEnvelope (HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut) -{ - +void FDKsbrEnc_deleteExtractSbrEnvelope(HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut) { if (hSbrCut) { FreeRam_Sbr_envYBuffer(&hSbrCut->p_YBuffer); } } -INT -FDKsbrEnc_GetEnvEstDelay(HANDLE_SBR_EXTRACT_ENVELOPE hSbr) -{ - return hSbr->no_rows*((hSbr->YBufferWriteOffset)*2 /* mult 2 because nrg's are grouped half */ - - hSbr->rBufferReadOffset ); /* in reference hold half spec and calc nrg's on overlapped spec */ - +INT FDKsbrEnc_GetEnvEstDelay(HANDLE_SBR_EXTRACT_ENVELOPE hSbr) { + return hSbr->no_rows * + ((hSbr->YBufferWriteOffset) * + 2 /* mult 2 because nrg's are grouped half */ + - hSbr->rBufferReadOffset); /* in reference hold half spec and calc + nrg's on overlapped spec */ } - - - - diff --git a/libSBRenc/src/env_est.h b/libSBRenc/src/env_est.h index e17a974..006f55b 100644 --- a/libSBRenc/src/env_est.h +++ b/libSBRenc/src/env_est.h @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,14 +90,22 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ /*! \file - \brief Envelope estimation structs and prototypes + \brief Envelope estimation structs and prototypes $Revision: 92790 $ */ -#ifndef __ENV_EST_H -#define __ENV_EST_H +#ifndef ENV_EST_H +#define ENV_EST_H #include "sbr_def.h" #include "sbr_encoder.h" /* SBR econfig structs */ @@ -97,20 +116,18 @@ amm-info@iis.fraunhofer.de #include "code_env.h" #include "ton_corr.h" -typedef struct -{ - FIXP_DBL *rBuffer[QMF_MAX_TIME_SLOTS]; - FIXP_DBL *iBuffer[QMF_MAX_TIME_SLOTS]; +typedef struct { + FIXP_DBL *rBuffer[32]; + FIXP_DBL *iBuffer[32]; - FIXP_DBL *p_YBuffer; + FIXP_DBL *p_YBuffer; - FIXP_DBL *YBuffer[QMF_MAX_TIME_SLOTS]; - int YBufferScale[2]; + FIXP_DBL *YBuffer[32]; + int YBufferScale[2]; UCHAR envelopeCompensation[MAX_FREQ_COEFFS]; UCHAR pre_transient_info[2]; - int YBufferWriteOffset; int YBufferSzShift; int rBufferReadOffset; @@ -121,21 +138,18 @@ typedef struct int time_slots; int time_step; -} -SBR_EXTRACT_ENVELOPE; +} SBR_EXTRACT_ENVELOPE; typedef SBR_EXTRACT_ENVELOPE *HANDLE_SBR_EXTRACT_ENVELOPE; -struct ENV_CHANNEL -{ +struct ENV_CHANNEL { FAST_TRAN_DETECTOR sbrFastTransientDetector; SBR_TRANSIENT_DETECTOR sbrTransientDetector; SBR_CODE_ENVELOPE sbrCodeEnvelope; SBR_CODE_ENVELOPE sbrCodeNoiseFloor; SBR_EXTRACT_ENVELOPE sbrExtractEnvelope; - SBR_ENVELOPE_FRAME SbrEnvFrame; - SBR_TON_CORR_EST TonCorr; + SBR_TON_CORR_EST TonCorr; struct SBR_ENV_DATA encEnvData; @@ -146,80 +160,64 @@ typedef struct ENV_CHANNEL *HANDLE_ENV_CHANNEL; /************ Function Declarations ***************/ -INT -FDKsbrEnc_CreateExtractSbrEnvelope (HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut, - INT channel - ,INT chInEl - ,UCHAR* dynamic_RAM - ); - - -INT -FDKsbrEnc_InitExtractSbrEnvelope ( - HANDLE_SBR_EXTRACT_ENVELOPE hSbr, - int no_cols, - int no_rows, - int start_index, - int time_slots, int time_step, int tran_off, - ULONG statesInitFlag - ,int chInEl - ,UCHAR* dynamic_RAM - ,UINT sbrSyntaxFlags - ); - -void FDKsbrEnc_deleteExtractSbrEnvelope (HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut); +INT FDKsbrEnc_CreateExtractSbrEnvelope(HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut, + INT channel, INT chInEl, + UCHAR *dynamic_RAM); + +INT FDKsbrEnc_InitExtractSbrEnvelope(HANDLE_SBR_EXTRACT_ENVELOPE hSbr, + int no_cols, int no_rows, int start_index, + int time_slots, int time_step, + int tran_off, ULONG statesInitFlag, + int chInEl, UCHAR *dynamic_RAM, + UINT sbrSyntaxFlags); + +void FDKsbrEnc_deleteExtractSbrEnvelope(HANDLE_SBR_EXTRACT_ENVELOPE hSbrCut); typedef struct { - FREQ_RES res[MAX_NUM_NOISE_VALUES]; - int maxQuantError; + FREQ_RES res[MAX_NUM_NOISE_VALUES]; + int maxQuantError; } SBR_FRAME_TEMP_DATA; typedef struct { - const SBR_FRAME_INFO *frame_info; - FIXP_DBL noiseFloor[MAX_NUM_NOISE_VALUES]; - SCHAR sfb_nrg_coupling[MAX_NUM_ENVELOPE_VALUES]; /* only used if stereomode = SWITCH_L_R_C */ - SCHAR sfb_nrg[MAX_NUM_ENVELOPE_VALUES]; - SCHAR noise_level_coupling[MAX_NUM_NOISE_VALUES]; /* only used if stereomode = SWITCH_L_R_C */ - SCHAR noise_level[MAX_NUM_NOISE_VALUES]; - UCHAR transient_info[3]; - UCHAR nEnvelopes; + const SBR_FRAME_INFO *frame_info; + FIXP_DBL noiseFloor[MAX_NUM_NOISE_VALUES]; + SCHAR sfb_nrg_coupling + [MAX_NUM_ENVELOPE_VALUES]; /* only used if stereomode = SWITCH_L_R_C */ + SCHAR sfb_nrg[MAX_NUM_ENVELOPE_VALUES]; + SCHAR noise_level_coupling + [MAX_NUM_NOISE_VALUES]; /* only used if stereomode = SWITCH_L_R_C */ + SCHAR noise_level[MAX_NUM_NOISE_VALUES]; + UCHAR transient_info[3]; + UCHAR nEnvelopes; } SBR_ENV_TEMP_DATA; /* - * Extract features from QMF data. Afterwards, the QMF data is not required anymore. + * Extract features from QMF data. Afterwards, the QMF data is not required + * anymore. */ -void -FDKsbrEnc_extractSbrEnvelope1( - HANDLE_SBR_CONFIG_DATA h_con, - HANDLE_SBR_HEADER_DATA sbrHeaderData, - HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, - HANDLE_ENV_CHANNEL h_envChan, - HANDLE_COMMON_DATA cmonData, - SBR_ENV_TEMP_DATA *eData, - SBR_FRAME_TEMP_DATA *fData - ); - +void FDKsbrEnc_extractSbrEnvelope1(HANDLE_SBR_CONFIG_DATA h_con, + HANDLE_SBR_HEADER_DATA sbrHeaderData, + HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, + HANDLE_ENV_CHANNEL h_envChan, + HANDLE_COMMON_DATA cmonData, + SBR_ENV_TEMP_DATA *eData, + SBR_FRAME_TEMP_DATA *fData); /* * Process the previously features extracted by FDKsbrEnc_extractSbrEnvelope1 * and create/encode SBR envelopes. */ -void -FDKsbrEnc_extractSbrEnvelope2( - HANDLE_SBR_CONFIG_DATA h_con, - HANDLE_SBR_HEADER_DATA sbrHeaderData, - HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, - HANDLE_ENV_CHANNEL sbrEnvChannel0, - HANDLE_ENV_CHANNEL sbrEnvChannel1, - HANDLE_COMMON_DATA cmonData, - SBR_ENV_TEMP_DATA *eData, - SBR_FRAME_TEMP_DATA *fData, - int clearOutput - ); - -INT -FDKsbrEnc_GetEnvEstDelay(HANDLE_SBR_EXTRACT_ENVELOPE hSbr); +void FDKsbrEnc_extractSbrEnvelope2(HANDLE_SBR_CONFIG_DATA h_con, + HANDLE_SBR_HEADER_DATA sbrHeaderData, + HANDLE_PARAMETRIC_STEREO hParametricStereo, + HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData, + HANDLE_ENV_CHANNEL sbrEnvChannel0, + HANDLE_ENV_CHANNEL sbrEnvChannel1, + HANDLE_COMMON_DATA cmonData, + SBR_ENV_TEMP_DATA *eData, + SBR_FRAME_TEMP_DATA *fData, int clearOutput); + +INT FDKsbrEnc_GetEnvEstDelay(HANDLE_SBR_EXTRACT_ENVELOPE hSbr); #endif diff --git a/libSBRenc/src/fram_gen.cpp b/libSBRenc/src/fram_gen.cpp index 9a35111..7ed6e79 100644 --- a/libSBRenc/src/fram_gen.cpp +++ b/libSBRenc/src/fram_gen.cpp @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,272 +90,235 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ #include "fram_gen.h" #include "sbr_misc.h" #include "genericStds.h" -static const SBR_FRAME_INFO frameInfo1_2048 = { - 1, - { 0, 16}, - {FREQ_RES_HIGH}, - 0, - 1, - {0, 16} }; +static const SBR_FRAME_INFO frameInfo1_2048 = {1, {0, 16}, {FREQ_RES_HIGH}, + 0, 1, {0, 16}}; static const SBR_FRAME_INFO frameInfo2_2048 = { - 2, - { 0, 8, 16}, - {FREQ_RES_HIGH, FREQ_RES_HIGH}, - 0, - 2, - { 0, 8, 16} }; + 2, {0, 8, 16}, {FREQ_RES_HIGH, FREQ_RES_HIGH}, 0, 2, {0, 8, 16}}; static const SBR_FRAME_INFO frameInfo4_2048 = { - 4, - { 0, 4, 8, 12, 16}, - {FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH}, - 0, - 2, - { 0, 8, 16} }; - -static const SBR_FRAME_INFO frameInfo1_2304 = { - 1, - { 0, 18}, - {FREQ_RES_HIGH}, - 0, - 1, - { 0, 18} }; + 4, + {0, 4, 8, 12, 16}, + {FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH}, + 0, + 2, + {0, 8, 16}}; + +static const SBR_FRAME_INFO frameInfo1_2304 = {1, {0, 18}, {FREQ_RES_HIGH}, + 0, 1, {0, 18}}; static const SBR_FRAME_INFO frameInfo2_2304 = { - 2, - { 0, 9, 18}, - {FREQ_RES_HIGH, FREQ_RES_HIGH}, - 0, - 2, - { 0, 9, 18} }; + 2, {0, 9, 18}, {FREQ_RES_HIGH, FREQ_RES_HIGH}, 0, 2, {0, 9, 18}}; static const SBR_FRAME_INFO frameInfo4_2304 = { - 4, - { 0, 5, 9, 14, 18}, - {FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH}, - 0, - 2, - { 0, 9, 18} }; - -static const SBR_FRAME_INFO frameInfo1_1920 = { - 1, - { 0, 15}, - {FREQ_RES_HIGH}, - 0, - 1, - { 0, 15} }; + 4, + {0, 5, 9, 14, 18}, + {FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH}, + 0, + 2, + {0, 9, 18}}; + +static const SBR_FRAME_INFO frameInfo1_1920 = {1, {0, 15}, {FREQ_RES_HIGH}, + 0, 1, {0, 15}}; static const SBR_FRAME_INFO frameInfo2_1920 = { - 2, - { 0, 8, 15}, - {FREQ_RES_HIGH, FREQ_RES_HIGH}, - 0, - 2, - { 0, 8, 15} }; + 2, {0, 8, 15}, {FREQ_RES_HIGH, FREQ_RES_HIGH}, 0, 2, {0, 8, 15}}; static const SBR_FRAME_INFO frameInfo4_1920 = { - 4, - { 0, 4, 8, 12, 15}, - {FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH}, - 0, - 2, - { 0, 8, 15} }; - -static const SBR_FRAME_INFO frameInfo1_1152 = { - 1, - { 0, 9}, - {FREQ_RES_HIGH}, - 0, - 1, - { 0, 9} }; + 4, + {0, 4, 8, 12, 15}, + {FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH}, + 0, + 2, + {0, 8, 15}}; + +static const SBR_FRAME_INFO frameInfo1_1152 = {1, {0, 9}, {FREQ_RES_HIGH}, + 0, 1, {0, 9}}; static const SBR_FRAME_INFO frameInfo2_1152 = { - 2, - { 0, 5, 9}, - {FREQ_RES_HIGH, FREQ_RES_HIGH}, - 0, - 2, - { 0, 5, 9} }; + 2, {0, 5, 9}, {FREQ_RES_HIGH, FREQ_RES_HIGH}, 0, 2, {0, 5, 9}}; static const SBR_FRAME_INFO frameInfo4_1152 = { - 4, - { 0, 2, 5, - 7, 9}, - {FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH}, - 0, - 2, - { 0, 5, 9} }; - + 4, + {0, 2, 5, 7, 9}, + {FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH}, + 0, + 2, + {0, 5, 9}}; /* AACLD frame info */ -static const SBR_FRAME_INFO frameInfo1_512LD = { - 1, - {0, 8}, - {FREQ_RES_HIGH}, - 0, - 1, - {0, 8}}; +static const SBR_FRAME_INFO frameInfo1_512LD = {1, {0, 8}, {FREQ_RES_HIGH}, + 0, 1, {0, 8}}; static const SBR_FRAME_INFO frameInfo2_512LD = { - 2, - {0, 4, 8}, - {FREQ_RES_HIGH, FREQ_RES_HIGH}, - 0, - 2, - {0, 4, 8}}; + 2, {0, 4, 8}, {FREQ_RES_HIGH, FREQ_RES_HIGH}, 0, 2, {0, 4, 8}}; static const SBR_FRAME_INFO frameInfo4_512LD = { - 4, - {0, 2, 4, 6, 8}, - {FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH}, - 0, - 2, - {0, 4, 8}}; - -static int -calcFillLengthMax (int tranPos, /*!< input : transient position (ref: tran det) */ - int numberTimeSlots /*!< input : number of timeslots */ - ); - -static void -fillFrameTran (const int *v_tuningSegm, /*!< tuning: desired segment lengths */ - const int *v_tuningFreq, /*!< tuning: desired frequency resolutions */ - int tran, /*!< input : position of transient */ - int *v_bord, /*!< memNew: borders */ - int *length_v_bord, /*!< memNew: # borders */ - int *v_freq, /*!< memNew: frequency resolutions */ - int *length_v_freq, /*!< memNew: # frequency resolutions */ - int *bmin, /*!< hlpNew: first mandatory border */ - int *bmax /*!< hlpNew: last mandatory border */ - ); - -static void fillFramePre (INT dmax, INT *v_bord, INT *length_v_bord, - INT *v_freq, INT *length_v_freq, INT bmin, - INT rest); - -static void fillFramePost (INT *parts, INT *d, INT dmax, INT *v_bord, - INT *length_v_bord, INT *v_freq, - INT *length_v_freq, INT bmax, - INT bufferFrameStart, INT numberTimeSlots, INT fmax); - -static void fillFrameInter (INT *nL, const int *v_tuningSegm, INT *v_bord, - INT *length_v_bord, INT bmin, INT *v_freq, - INT *length_v_freq, INT *v_bordFollow, - INT *length_v_bordFollow, INT *v_freqFollow, - INT *length_v_freqFollow, INT i_fillFollow, - INT dmin, INT dmax, INT numberTimeSlots); - -static void calcFrameClass (FRAME_CLASS *frameClass, FRAME_CLASS *frameClassOld, INT tranFlag, - INT *spreadFlag); - -static void specialCase (INT *spreadFlag, INT allowSpread, INT *v_bord, - INT *length_v_bord, INT *v_freq, INT *length_v_freq, - INT *parts, INT d); - -static void calcCmonBorder (INT *i_cmon, INT *i_tran, INT *v_bord, - INT *length_v_bord, INT tran, - INT bufferFrameStart, INT numberTimeSlots); - -static void keepForFollowUp (INT *v_bordFollow, INT *length_v_bordFollow, - INT *v_freqFollow, INT *length_v_freqFollow, - INT *i_tranFollow, INT *i_fillFollow, - INT *v_bord, INT *length_v_bord, INT *v_freq, - INT i_cmon, INT i_tran, INT parts, INT numberTimeSlots); - -static void calcCtrlSignal (HANDLE_SBR_GRID hSbrGrid, FRAME_CLASS frameClass, - INT *v_bord, INT length_v_bord, INT *v_freq, - INT length_v_freq, INT i_cmon, INT i_tran, - INT spreadFlag, INT nL); - -static void ctrlSignal2FrameInfo (HANDLE_SBR_GRID hSbrGrid, - HANDLE_SBR_FRAME_INFO hFrameInfo, - FREQ_RES *freq_res_fixfix); - + 4, + {0, 2, 4, 6, 8}, + {FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH}, + 0, + 2, + {0, 4, 8}}; + +static int calcFillLengthMax( + int tranPos, /*!< input : transient position (ref: tran det) */ + int numberTimeSlots /*!< input : number of timeslots */ +); + +static void fillFrameTran( + const int *v_tuningSegm, /*!< tuning: desired segment lengths */ + const int *v_tuningFreq, /*!< tuning: desired frequency resolutions */ + int tran, /*!< input : position of transient */ + int *v_bord, /*!< memNew: borders */ + int *length_v_bord, /*!< memNew: # borders */ + int *v_freq, /*!< memNew: frequency resolutions */ + int *length_v_freq, /*!< memNew: # frequency resolutions */ + int *bmin, /*!< hlpNew: first mandatory border */ + int *bmax /*!< hlpNew: last mandatory border */ +); + +static void fillFramePre(INT dmax, INT *v_bord, INT *length_v_bord, INT *v_freq, + INT *length_v_freq, INT bmin, INT rest); + +static void fillFramePost(INT *parts, INT *d, INT dmax, INT *v_bord, + INT *length_v_bord, INT *v_freq, INT *length_v_freq, + INT bmax, INT bufferFrameStart, INT numberTimeSlots, + INT fmax); + +static void fillFrameInter(INT *nL, const int *v_tuningSegm, INT *v_bord, + INT *length_v_bord, INT bmin, INT *v_freq, + INT *length_v_freq, INT *v_bordFollow, + INT *length_v_bordFollow, INT *v_freqFollow, + INT *length_v_freqFollow, INT i_fillFollow, INT dmin, + INT dmax, INT numberTimeSlots); + +static void calcFrameClass(FRAME_CLASS *frameClass, FRAME_CLASS *frameClassOld, + INT tranFlag, INT *spreadFlag); + +static void specialCase(INT *spreadFlag, INT allowSpread, INT *v_bord, + INT *length_v_bord, INT *v_freq, INT *length_v_freq, + INT *parts, INT d); + +static void calcCmonBorder(INT *i_cmon, INT *i_tran, INT *v_bord, + INT *length_v_bord, INT tran, INT bufferFrameStart, + INT numberTimeSlots); + +static void keepForFollowUp(INT *v_bordFollow, INT *length_v_bordFollow, + INT *v_freqFollow, INT *length_v_freqFollow, + INT *i_tranFollow, INT *i_fillFollow, INT *v_bord, + INT *length_v_bord, INT *v_freq, INT i_cmon, + INT i_tran, INT parts, INT numberTimeSlots); + +static void calcCtrlSignal(HANDLE_SBR_GRID hSbrGrid, FRAME_CLASS frameClass, + INT *v_bord, INT length_v_bord, INT *v_freq, + INT length_v_freq, INT i_cmon, INT i_tran, + INT spreadFlag, INT nL); + +static void ctrlSignal2FrameInfo(HANDLE_SBR_GRID hSbrGrid, + HANDLE_SBR_FRAME_INFO hFrameInfo, + FREQ_RES *freq_res_fixfix); /* table for 8 time slot index */ -static const int 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 }, +static const int 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 16 time slot index */ -static const int envelopeTable_16 [16][6] = { +static const int 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}, + /*[|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}, }; /* table for 15 time slot index */ -static const int envelopeTable_15 [15][6] = { +static const int 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}, + /*[|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}, }; static const int minFrameTranDistance = 4; -static const FREQ_RES freqRes_table_8[] = {FREQ_RES_LOW, FREQ_RES_LOW, FREQ_RES_LOW, FREQ_RES_LOW, FREQ_RES_LOW, - FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH}; +static const FREQ_RES freqRes_table_8[] = { + FREQ_RES_LOW, FREQ_RES_LOW, FREQ_RES_LOW, FREQ_RES_LOW, FREQ_RES_LOW, + FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH}; static const FREQ_RES freqRes_table_16[16] = { /* size of envelope */ -/* 0-4 */ FREQ_RES_LOW, FREQ_RES_LOW, FREQ_RES_LOW, FREQ_RES_LOW, FREQ_RES_LOW, -/* 5-9 */ FREQ_RES_LOW, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, -/* 10-16 */ FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, FREQ_RES_HIGH, - FREQ_RES_HIGH }; - -static void generateFixFixOnly ( HANDLE_SBR_FRAME_INFO hSbrFrameInfo, - HANDLE_SBR_GRID hSbrGrid, - int tranPosInternal, - int numberTimeSlots, - UCHAR fResTransIsLow - ); - + /* 0-4 */ FREQ_RES_LOW, + FREQ_RES_LOW, + FREQ_RES_LOW, + FREQ_RES_LOW, + FREQ_RES_LOW, + /* 5-9 */ FREQ_RES_LOW, + FREQ_RES_HIGH, + FREQ_RES_HIGH, + FREQ_RES_HIGH, + FREQ_RES_HIGH, + /* 10-16 */ FREQ_RES_HIGH, + FREQ_RES_HIGH, + FREQ_RES_HIGH, + FREQ_RES_HIGH, + FREQ_RES_HIGH, + FREQ_RES_HIGH}; + +static void generateFixFixOnly(HANDLE_SBR_FRAME_INFO hSbrFrameInfo, + HANDLE_SBR_GRID hSbrGrid, int tranPosInternal, + int numberTimeSlots, UCHAR fResTransIsLow); /*! Functionname: FDKsbrEnc_frameInfoGenerator @@ -353,20 +327,19 @@ static void generateFixFixOnly ( HANDLE_SBR_FRAME_INFO hSbrFrameInfo, Arguments: hSbrEnvFrame - pointer to sbr envelope handle v_pre_transient_info - pointer to transient info vector - v_transient_info - pointer to previous transient info vector - v_tuning - pointer to tuning vector + v_transient_info - pointer to previous transient info +vector v_tuning - pointer to tuning vector Return: frame_info - pointer to SBR_FRAME_INFO struct *******************************************************************************/ HANDLE_SBR_FRAME_INFO -FDKsbrEnc_frameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame, - UCHAR *v_transient_info, - UCHAR *v_transient_info_pre, - int ldGrid, - const int *v_tuning) -{ - INT numEnv, tranPosInternal=0, bmin=0, bmax=0, parts, d, i_cmon=0, i_tran=0, nL; +FDKsbrEnc_frameInfoGenerator(HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame, + UCHAR *v_transient_info, const INT rightBorderFIX, + UCHAR *v_transient_info_pre, int ldGrid, + const int *v_tuning) { + INT numEnv, tranPosInternal = 0, bmin = 0, bmax = 0, parts, d, i_cmon = 0, + i_tran = 0, nL; INT fmax = 0; INT *v_bord = hSbrEnvFrame->v_bord; @@ -374,7 +347,6 @@ FDKsbrEnc_frameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame, INT *v_bordFollow = hSbrEnvFrame->v_bordFollow; INT *v_freqFollow = hSbrEnvFrame->v_freqFollow; - INT *length_v_bordFollow = &hSbrEnvFrame->length_v_bordFollow; INT *length_v_freqFollow = &hSbrEnvFrame->length_v_freqFollow; INT *length_v_bord = &hSbrEnvFrame->length_v_bord; @@ -385,7 +357,6 @@ FDKsbrEnc_frameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame, FRAME_CLASS *frameClassOld = &hSbrEnvFrame->frameClassOld; FRAME_CLASS frameClass = FIXFIX; - INT allowSpread = hSbrEnvFrame->allowSpread; INT numEnvStatic = hSbrEnvFrame->numEnvStatic; INT staticFraming = hSbrEnvFrame->staticFraming; @@ -405,10 +376,12 @@ FDKsbrEnc_frameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame, hSbrEnvFrame->v_tuningSegm = v_tuningSegm; if (ldGrid) { - /* in case there was a transient at the very end of the previous frame, start with a transient envelope */ - if ( !tranFlag && v_transient_info_pre[1] && (numberTimeSlots - v_transient_info_pre[0] < minFrameTranDistance) ){ + /* in case there was a transient at the very end of the previous frame, + * start with a transient envelope */ + if (!tranFlag && v_transient_info_pre[1] && + (numberTimeSlots - v_transient_info_pre[0] < minFrameTranDistance)) { tranFlag = 1; - tranPos = 0; + tranPos = 0; } } @@ -463,20 +436,23 @@ FDKsbrEnc_frameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame, ---------------------------------------------------------------------------*/ frameClass = FIXFIX; - numEnv = numEnvStatic; /* {1,2,4,8} */ - *frameClassOld = FIXFIX; /* for change to dyn */ + numEnv = numEnvStatic; /* {1,2,4,8} */ + *frameClassOld = FIXFIX; /* for change to dyn */ hSbrEnvFrame->SbrGrid.bs_num_env = numEnv; hSbrEnvFrame->SbrGrid.frameClass = frameClass; - } - else { + } else { /*-------------------------------------------------------------------------- Calculate frame class to use ---------------------------------------------------------------------------*/ - calcFrameClass (&frameClass, frameClassOld, tranFlag, spreadFlag); + if (rightBorderFIX) { + tranFlag = 0; + *spreadFlag = 0; + } + calcFrameClass(&frameClass, frameClassOld, tranFlag, spreadFlag); /* patch for new frame class FIXFIXonly for AAC LD */ if (tranFlag && ldGrid) { - frameClass = FIXFIXonly; + frameClass = FIXFIXonly; *frameClassOld = FIXFIX; } @@ -497,238 +473,226 @@ FDKsbrEnc_frameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame, /*-------------------------------------------------------------------------- Design frame (or follow-up old design) ---------------------------------------------------------------------------*/ - if (tranFlag) { /* Always for FixVar, often but not always for VarVar */ + if (tranFlag) { + /* Always for FixVar, often but not always for VarVar */ + /*-------------------------------------------------------------------------- Design part of T/F-grid around the new transient ---------------------------------------------------------------------------*/ - tranPosInternal = frameMiddleSlot + tranPos + bufferFrameStart ; /* FH 00-06-26 */ + tranPosInternal = + frameMiddleSlot + tranPos + bufferFrameStart; /* FH 00-06-26 */ /* add mandatory borders around transient */ - fillFrameTran ( v_tuningSegm, - v_tuningFreq, - tranPosInternal, - v_bord, - length_v_bord, - v_freq, - length_v_freq, - &bmin, - &bmax ); + fillFrameTran(v_tuningSegm, v_tuningFreq, tranPosInternal, v_bord, + length_v_bord, v_freq, length_v_freq, &bmin, &bmax); /* make sure we stay within the maximum SBR frame overlap */ fmax = calcFillLengthMax(tranPos, numberTimeSlots); } switch (frameClass) { + case FIXFIXonly: + FDK_ASSERT(ldGrid); + tranPosInternal = tranPos; + generateFixFixOnly(&(hSbrEnvFrame->SbrFrameInfo), + &(hSbrEnvFrame->SbrGrid), tranPosInternal, + numberTimeSlots, hSbrEnvFrame->fResTransIsLow); - case FIXFIXonly: - FDK_ASSERT(ldGrid); - tranPosInternal = tranPos; - generateFixFixOnly ( &(hSbrEnvFrame->SbrFrameInfo), - &(hSbrEnvFrame->SbrGrid), - tranPosInternal, - numberTimeSlots, - hSbrEnvFrame->fResTransIsLow - ); + return &(hSbrEnvFrame->SbrFrameInfo); - return &(hSbrEnvFrame->SbrFrameInfo); - - case FIXVAR: + case FIXVAR: - /*-------------------------------------------------------------------------- - Design remaining parts of T/F-grid (assuming next frame is VarFix) - ---------------------------------------------------------------------------*/ - - /*-------------------------------------------------------------------------- - Fill region before new transient: - ---------------------------------------------------------------------------*/ - fillFramePre (dmax, v_bord, length_v_bord, v_freq, length_v_freq, - bmin, bmin - bufferFrameStart); /* FH 00-06-26 */ - - /*-------------------------------------------------------------------------- - Fill region after new transient: - ---------------------------------------------------------------------------*/ - fillFramePost (&parts, &d, dmax, v_bord, length_v_bord, v_freq, - length_v_freq, bmax, bufferFrameStart, numberTimeSlots, fmax); - - /*-------------------------------------------------------------------------- - Take care of special case: - ---------------------------------------------------------------------------*/ - if (parts == 1 && d < dmin) /* no fill, short last envelope */ - specialCase (spreadFlag, allowSpread, v_bord, length_v_bord, - v_freq, length_v_freq, &parts, d); - - /*-------------------------------------------------------------------------- - Calculate common border (split-point) - ---------------------------------------------------------------------------*/ - calcCmonBorder (&i_cmon, &i_tran, v_bord, length_v_bord, tranPosInternal, - bufferFrameStart, numberTimeSlots); /* FH 00-06-26 */ - - /*-------------------------------------------------------------------------- - Extract data for proper follow-up in next frame - ---------------------------------------------------------------------------*/ - keepForFollowUp (v_bordFollow, length_v_bordFollow, v_freqFollow, - length_v_freqFollow, i_tranFollow, i_fillFollow, - v_bord, length_v_bord, v_freq, i_cmon, i_tran, parts, numberTimeSlots); /* FH 00-06-26 */ - - /*-------------------------------------------------------------------------- - Calculate control signal - ---------------------------------------------------------------------------*/ - calcCtrlSignal (&hSbrEnvFrame->SbrGrid, frameClass, - v_bord, *length_v_bord, v_freq, *length_v_freq, - i_cmon, i_tran, *spreadFlag, DC); - break; - case VARFIX: - /*-------------------------------------------------------------------------- - Follow-up old transient - calculate control signal - ---------------------------------------------------------------------------*/ - calcCtrlSignal (&hSbrEnvFrame->SbrGrid, frameClass, - v_bordFollow, *length_v_bordFollow, v_freqFollow, - *length_v_freqFollow, DC, *i_tranFollow, - *spreadFlag, DC); - break; - case VARVAR: - if (*spreadFlag) { /* spread across three frames */ /*-------------------------------------------------------------------------- - Follow-up old transient - calculate control signal + Design remaining parts of T/F-grid (assuming next frame is VarFix) ---------------------------------------------------------------------------*/ - calcCtrlSignal (&hSbrEnvFrame->SbrGrid, - frameClass, v_bordFollow, *length_v_bordFollow, - v_freqFollow, *length_v_freqFollow, DC, - *i_tranFollow, *spreadFlag, DC); - - *spreadFlag = 0; /*-------------------------------------------------------------------------- - Extract data for proper follow-up in next frame + Fill region before new transient: ---------------------------------------------------------------------------*/ - v_bordFollow[0] = hSbrEnvFrame->SbrGrid.bs_abs_bord_1 - numberTimeSlots; /* FH 00-06-26 */ - v_freqFollow[0] = 1; - *length_v_bordFollow = 1; - *length_v_freqFollow = 1; + fillFramePre(dmax, v_bord, length_v_bord, v_freq, length_v_freq, bmin, + bmin - bufferFrameStart); /* FH 00-06-26 */ - *i_tranFollow = -DC; - *i_fillFollow = -DC; - } - else { /*-------------------------------------------------------------------------- - Design remaining parts of T/F-grid (assuming next frame is VarFix) - adapt or fill region before new transient: + Fill region after new transient: ---------------------------------------------------------------------------*/ - fillFrameInter (&nL, v_tuningSegm, v_bord, length_v_bord, bmin, - v_freq, length_v_freq, v_bordFollow, - length_v_bordFollow, v_freqFollow, - length_v_freqFollow, *i_fillFollow, dmin, dmax, - numberTimeSlots); - - /*-------------------------------------------------------------------------- - Fill after transient: - ---------------------------------------------------------------------------*/ - fillFramePost (&parts, &d, dmax, v_bord, length_v_bord, v_freq, - length_v_freq, bmax, bufferFrameStart, numberTimeSlots, fmax); + fillFramePost(&parts, &d, dmax, v_bord, length_v_bord, v_freq, + length_v_freq, bmax, bufferFrameStart, numberTimeSlots, + fmax); /*-------------------------------------------------------------------------- Take care of special case: ---------------------------------------------------------------------------*/ - if (parts == 1 && d < dmin) /*% no fill, short last envelope */ - specialCase (spreadFlag, allowSpread, v_bord, length_v_bord, - v_freq, length_v_freq, &parts, d); + if (parts == 1 && d < dmin) /* no fill, short last envelope */ + specialCase(spreadFlag, allowSpread, v_bord, length_v_bord, v_freq, + length_v_freq, &parts, d); /*-------------------------------------------------------------------------- Calculate common border (split-point) ---------------------------------------------------------------------------*/ - calcCmonBorder (&i_cmon, &i_tran, v_bord, length_v_bord, tranPosInternal, - bufferFrameStart, numberTimeSlots); + calcCmonBorder(&i_cmon, &i_tran, v_bord, length_v_bord, tranPosInternal, + bufferFrameStart, numberTimeSlots); /* FH 00-06-26 */ /*-------------------------------------------------------------------------- Extract data for proper follow-up in next frame ---------------------------------------------------------------------------*/ - keepForFollowUp (v_bordFollow, length_v_bordFollow, - v_freqFollow, length_v_freqFollow, - i_tranFollow, i_fillFollow, v_bord, - length_v_bord, v_freq, i_cmon, i_tran, parts, numberTimeSlots); + keepForFollowUp(v_bordFollow, length_v_bordFollow, v_freqFollow, + length_v_freqFollow, i_tranFollow, i_fillFollow, v_bord, + length_v_bord, v_freq, i_cmon, i_tran, parts, + numberTimeSlots); /* FH 00-06-26 */ /*-------------------------------------------------------------------------- Calculate control signal ---------------------------------------------------------------------------*/ - calcCtrlSignal (&hSbrEnvFrame->SbrGrid, - frameClass, v_bord, *length_v_bord, v_freq, - *length_v_freq, i_cmon, i_tran, 0, nL); - } - break; - case FIXFIX: - if (tranPos == 0) - numEnv = 1; - else - numEnv = 2; + calcCtrlSignal(&hSbrEnvFrame->SbrGrid, frameClass, v_bord, + *length_v_bord, v_freq, *length_v_freq, i_cmon, i_tran, + *spreadFlag, DC); + break; + case VARFIX: + /*-------------------------------------------------------------------------- + Follow-up old transient - calculate control signal + ---------------------------------------------------------------------------*/ + calcCtrlSignal(&hSbrEnvFrame->SbrGrid, frameClass, v_bordFollow, + *length_v_bordFollow, v_freqFollow, *length_v_freqFollow, + DC, *i_tranFollow, *spreadFlag, DC); + break; + case VARVAR: + if (*spreadFlag) { /* spread across three frames */ + /*-------------------------------------------------------------------------- + Follow-up old transient - calculate control signal + ---------------------------------------------------------------------------*/ + calcCtrlSignal(&hSbrEnvFrame->SbrGrid, frameClass, v_bordFollow, + *length_v_bordFollow, v_freqFollow, + *length_v_freqFollow, DC, *i_tranFollow, *spreadFlag, + DC); + + *spreadFlag = 0; + + /*-------------------------------------------------------------------------- + Extract data for proper follow-up in next frame + ---------------------------------------------------------------------------*/ + v_bordFollow[0] = hSbrEnvFrame->SbrGrid.bs_abs_bord_1 - + numberTimeSlots; /* FH 00-06-26 */ + v_freqFollow[0] = 1; + *length_v_bordFollow = 1; + *length_v_freqFollow = 1; + + *i_tranFollow = -DC; + *i_fillFollow = -DC; + } else { + /*-------------------------------------------------------------------------- + Design remaining parts of T/F-grid (assuming next frame is VarFix) + adapt or fill region before new transient: + ---------------------------------------------------------------------------*/ + fillFrameInter(&nL, v_tuningSegm, v_bord, length_v_bord, bmin, v_freq, + length_v_freq, v_bordFollow, length_v_bordFollow, + v_freqFollow, length_v_freqFollow, *i_fillFollow, dmin, + dmax, numberTimeSlots); + + /*-------------------------------------------------------------------------- + Fill after transient: + ---------------------------------------------------------------------------*/ + fillFramePost(&parts, &d, dmax, v_bord, length_v_bord, v_freq, + length_v_freq, bmax, bufferFrameStart, numberTimeSlots, + fmax); + + /*-------------------------------------------------------------------------- + Take care of special case: + ---------------------------------------------------------------------------*/ + if (parts == 1 && d < dmin) /*% no fill, short last envelope */ + specialCase(spreadFlag, allowSpread, v_bord, length_v_bord, v_freq, + length_v_freq, &parts, d); + + /*-------------------------------------------------------------------------- + Calculate common border (split-point) + ---------------------------------------------------------------------------*/ + calcCmonBorder(&i_cmon, &i_tran, v_bord, length_v_bord, + tranPosInternal, bufferFrameStart, numberTimeSlots); + + /*-------------------------------------------------------------------------- + Extract data for proper follow-up in next frame + ---------------------------------------------------------------------------*/ + keepForFollowUp(v_bordFollow, length_v_bordFollow, v_freqFollow, + length_v_freqFollow, i_tranFollow, i_fillFollow, + v_bord, length_v_bord, v_freq, i_cmon, i_tran, parts, + numberTimeSlots); + + /*-------------------------------------------------------------------------- + Calculate control signal + ---------------------------------------------------------------------------*/ + calcCtrlSignal(&hSbrEnvFrame->SbrGrid, frameClass, v_bord, + *length_v_bord, v_freq, *length_v_freq, i_cmon, i_tran, + 0, nL); + } + break; + case FIXFIX: + if (tranPos == 0) + numEnv = 1; + else + numEnv = 2; - hSbrEnvFrame->SbrGrid.bs_num_env = numEnv; - hSbrEnvFrame->SbrGrid.frameClass = frameClass; + hSbrEnvFrame->SbrGrid.bs_num_env = numEnv; + hSbrEnvFrame->SbrGrid.frameClass = frameClass; - break; - default: - FDK_ASSERT(0); + break; + default: + FDK_ASSERT(0); } } /*------------------------------------------------------------------------- Convert control signal to frame info struct ---------------------------------------------------------------------------*/ - ctrlSignal2FrameInfo (&hSbrEnvFrame->SbrGrid, - &hSbrEnvFrame->SbrFrameInfo, - hSbrEnvFrame->freq_res_fixfix); + ctrlSignal2FrameInfo(&hSbrEnvFrame->SbrGrid, &hSbrEnvFrame->SbrFrameInfo, + hSbrEnvFrame->freq_res_fixfix); return &hSbrEnvFrame->SbrFrameInfo; } - /***************************************************************************/ /*! - \brief Gnerates frame info for FIXFIXonly frame class used for low delay version + \brief Gnerates frame info for FIXFIXonly frame class used for low delay + version \return nothing ****************************************************************************/ -static void generateFixFixOnly ( HANDLE_SBR_FRAME_INFO hSbrFrameInfo, - HANDLE_SBR_GRID hSbrGrid, - int tranPosInternal, - int numberTimeSlots, - UCHAR fResTransIsLow - ) -{ - int nEnv, i, k=0, tranIdx; +static void generateFixFixOnly(HANDLE_SBR_FRAME_INFO hSbrFrameInfo, + HANDLE_SBR_GRID hSbrGrid, int tranPosInternal, + int numberTimeSlots, UCHAR fResTransIsLow) { + int nEnv, i, k = 0, tranIdx; const int *pTable = NULL; const FREQ_RES *freqResTable = NULL; switch (numberTimeSlots) { - case 8: - pTable = envelopeTable_8[tranPosInternal]; - freqResTable = freqRes_table_8; - break; - case 15: - pTable = envelopeTable_15[tranPosInternal]; - freqResTable = freqRes_table_16; - break; - case 16: - pTable = envelopeTable_16[tranPosInternal]; - freqResTable = freqRes_table_16; - break; + case 8: { + pTable = envelopeTable_8[tranPosInternal]; + } + freqResTable = freqRes_table_8; + break; + case 15: + pTable = envelopeTable_15[tranPosInternal]; + freqResTable = freqRes_table_16; + break; + case 16: + pTable = envelopeTable_16[tranPosInternal]; + freqResTable = freqRes_table_16; + break; } /* look number of envolpes in table */ nEnv = pTable[0]; /* look up envolpe distribution in table */ - for (i=1; i<nEnv; i++) - hSbrFrameInfo->borders[i] = pTable[i+2]; + for (i = 1; i < nEnv; i++) hSbrFrameInfo->borders[i] = pTable[i + 2]; /* open and close frame border */ - hSbrFrameInfo->borders[0] = 0; + hSbrFrameInfo->borders[0] = 0; hSbrFrameInfo->borders[nEnv] = numberTimeSlots; /* adjust segment-frequency-resolution according to the segment-length */ - for (i=0; i<nEnv; i++){ - k = hSbrFrameInfo->borders[i+1] - hSbrFrameInfo->borders[i]; + for (i = 0; i < nEnv; i++) { + k = hSbrFrameInfo->borders[i + 1] - hSbrFrameInfo->borders[i]; if (!fResTransIsLow) hSbrFrameInfo->freqRes[i] = freqResTable[k]; else @@ -738,24 +702,22 @@ static void generateFixFixOnly ( HANDLE_SBR_FRAME_INFO hSbrFrameInfo, } hSbrFrameInfo->nEnvelopes = nEnv; - hSbrFrameInfo->shortEnv = pTable[2]; + hSbrFrameInfo->shortEnv = pTable[2]; /* transient idx */ tranIdx = pTable[1]; /* add noise floors */ hSbrFrameInfo->bordersNoise[0] = 0; - hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[tranIdx?tranIdx:1]; + hSbrFrameInfo->bordersNoise[1] = + hSbrFrameInfo->borders[tranIdx ? tranIdx : 1]; hSbrFrameInfo->bordersNoise[2] = numberTimeSlots; hSbrFrameInfo->nNoiseEnvelopes = 2; hSbrGrid->frameClass = FIXFIXonly; hSbrGrid->bs_abs_bord = tranPosInternal; hSbrGrid->bs_num_env = nEnv; - } - - /******************************************************************************* Functionname: FDKsbrEnc_initFrameInfoGenerator ******************************************************************************* @@ -770,21 +732,14 @@ static void generateFixFixOnly ( HANDLE_SBR_FRAME_INFO hSbrFrameInfo, Return: none *******************************************************************************/ -void -FDKsbrEnc_initFrameInfoGenerator ( - HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame, - INT allowSpread, - INT numEnvStatic, - INT staticFraming, - INT timeSlots, - const FREQ_RES* freq_res_fixfix - ,UCHAR fResTransIsLow, - INT ldGrid - ) -{ /* FH 00-06-26 */ - - FDKmemclear(hSbrEnvFrame,sizeof(SBR_ENVELOPE_FRAME )); +void FDKsbrEnc_initFrameInfoGenerator(HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame, + INT allowSpread, INT numEnvStatic, + INT staticFraming, INT timeSlots, + const FREQ_RES *freq_res_fixfix, + UCHAR fResTransIsLow, + INT ldGrid) { /* FH 00-06-26 */ + FDKmemclear(hSbrEnvFrame, sizeof(SBR_ENVELOPE_FRAME)); /* Initialisation */ hSbrEnvFrame->frameClassOld = FIXFIX; @@ -795,7 +750,7 @@ FDKsbrEnc_initFrameInfoGenerator ( hSbrEnvFrame->staticFraming = staticFraming; hSbrEnvFrame->freq_res_fixfix[0] = freq_res_fixfix[0]; hSbrEnvFrame->freq_res_fixfix[1] = freq_res_fixfix[1]; - hSbrEnvFrame->fResTransIsLow = fResTransIsLow; + hSbrEnvFrame->fResTransIsLow = fResTransIsLow; hSbrEnvFrame->length_v_bord = 0; hSbrEnvFrame->length_v_bordFollow = 0; @@ -810,43 +765,41 @@ FDKsbrEnc_initFrameInfoGenerator ( if (ldGrid) { /*case CODEC_AACLD:*/ - hSbrEnvFrame->dmin = 2; - hSbrEnvFrame->dmax = 16; - hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_512LD; - hSbrEnvFrame->SbrGrid.bufferFrameStart = 0; + hSbrEnvFrame->dmin = 2; + hSbrEnvFrame->dmax = 16; + hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_512LD; + hSbrEnvFrame->SbrGrid.bufferFrameStart = 0; } else - switch(timeSlots){ - case NUMBER_TIME_SLOTS_1920: - hSbrEnvFrame->dmin = 4; - hSbrEnvFrame->dmax = 12; - hSbrEnvFrame->SbrGrid.bufferFrameStart = 0; - hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_1920; - break; - case NUMBER_TIME_SLOTS_2048: - hSbrEnvFrame->dmin = 4; - hSbrEnvFrame->dmax = 12; - hSbrEnvFrame->SbrGrid.bufferFrameStart = 0; - hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_2048; - break; - case NUMBER_TIME_SLOTS_1152: - hSbrEnvFrame->dmin = 2; - hSbrEnvFrame->dmax = 8; - hSbrEnvFrame->SbrGrid.bufferFrameStart = 0; - hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_1152; - break; - case NUMBER_TIME_SLOTS_2304: - hSbrEnvFrame->dmin = 4; - hSbrEnvFrame->dmax = 15; - hSbrEnvFrame->SbrGrid.bufferFrameStart = 0; - hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_2304; - break; - default: - FDK_ASSERT(0); - } - + switch (timeSlots) { + case NUMBER_TIME_SLOTS_1920: + hSbrEnvFrame->dmin = 4; + hSbrEnvFrame->dmax = 12; + hSbrEnvFrame->SbrGrid.bufferFrameStart = 0; + hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_1920; + break; + case NUMBER_TIME_SLOTS_2048: + hSbrEnvFrame->dmin = 4; + hSbrEnvFrame->dmax = 12; + hSbrEnvFrame->SbrGrid.bufferFrameStart = 0; + hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_2048; + break; + case NUMBER_TIME_SLOTS_1152: + hSbrEnvFrame->dmin = 2; + hSbrEnvFrame->dmax = 8; + hSbrEnvFrame->SbrGrid.bufferFrameStart = 0; + hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_1152; + break; + case NUMBER_TIME_SLOTS_2304: + hSbrEnvFrame->dmin = 4; + hSbrEnvFrame->dmax = 15; + hSbrEnvFrame->SbrGrid.bufferFrameStart = 0; + hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_2304; + break; + default: + FDK_ASSERT(0); + } } - /******************************************************************************* Functionname: fillFrameTran ******************************************************************************* @@ -870,18 +823,17 @@ FDKsbrEnc_initFrameInfoGenerator ( Return: none *******************************************************************************/ -static void -fillFrameTran (const int *v_tuningSegm, /*!< tuning: desired segment lengths */ - const int *v_tuningFreq, /*!< tuning: desired frequency resolutions */ - int tran, /*!< input : position of transient */ - int *v_bord, /*!< memNew: borders */ - int *length_v_bord, /*!< memNew: # borders */ - int *v_freq, /*!< memNew: frequency resolutions */ - int *length_v_freq, /*!< memNew: # frequency resolutions */ - int *bmin, /*!< hlpNew: first mandatory border */ - int *bmax /*!< hlpNew: last mandatory border */ - ) -{ +static void fillFrameTran( + const int *v_tuningSegm, /*!< tuning: desired segment lengths */ + const int *v_tuningFreq, /*!< tuning: desired frequency resolutions */ + int tran, /*!< input : position of transient */ + int *v_bord, /*!< memNew: borders */ + int *length_v_bord, /*!< memNew: # borders */ + int *v_freq, /*!< memNew: frequency resolutions */ + int *length_v_freq, /*!< memNew: # frequency resolutions */ + int *bmin, /*!< hlpNew: first mandatory border */ + int *bmax /*!< hlpNew: last mandatory border */ +) { int bord, i; *length_v_bord = 0; @@ -890,25 +842,25 @@ fillFrameTran (const int *v_tuningSegm, /*!< tuning: desired segment length /* add attack env leading border (optional) */ if (v_tuningSegm[0]) { /* v_bord = [(Ba)] start of attack env */ - FDKsbrEnc_AddRight (v_bord, length_v_bord, (tran - v_tuningSegm[0])); + FDKsbrEnc_AddRight(v_bord, length_v_bord, (tran - v_tuningSegm[0])); /* v_freq = [(Fa)] res of attack env */ - FDKsbrEnc_AddRight (v_freq, length_v_freq, v_tuningFreq[0]); + FDKsbrEnc_AddRight(v_freq, length_v_freq, v_tuningFreq[0]); } /* add attack env trailing border/first decay env leading border */ bord = tran; - FDKsbrEnc_AddRight (v_bord, length_v_bord, tran); /* v_bord = [(Ba),Bd1] */ + FDKsbrEnc_AddRight(v_bord, length_v_bord, tran); /* v_bord = [(Ba),Bd1] */ /* add first decay env trailing border/2:nd decay env leading border */ if (v_tuningSegm[1]) { bord += v_tuningSegm[1]; /* v_bord = [(Ba),Bd1,Bd2] */ - FDKsbrEnc_AddRight (v_bord, length_v_bord, bord); + FDKsbrEnc_AddRight(v_bord, length_v_bord, bord); /* v_freq = [(Fa),Fd1] */ - FDKsbrEnc_AddRight (v_freq, length_v_freq, v_tuningFreq[1]); + FDKsbrEnc_AddRight(v_freq, length_v_freq, v_tuningFreq[1]); } /* add 2:nd decay env trailing border (optional) */ @@ -916,31 +868,25 @@ fillFrameTran (const int *v_tuningSegm, /*!< tuning: desired segment length bord += v_tuningSegm[2]; /* v_bord = [(Ba),Bd1, Bd2,(Bd3)] */ - FDKsbrEnc_AddRight (v_bord, length_v_bord, bord); + FDKsbrEnc_AddRight(v_bord, length_v_bord, bord); /* v_freq = [(Fa),Fd1,(Fd2)] */ - FDKsbrEnc_AddRight (v_freq, length_v_freq, v_tuningFreq[2]); + FDKsbrEnc_AddRight(v_freq, length_v_freq, v_tuningFreq[2]); } /* v_freq = [(Fa),Fd1,(Fd2),1] */ - FDKsbrEnc_AddRight (v_freq, length_v_freq, 1); - + FDKsbrEnc_AddRight(v_freq, length_v_freq, 1); /* calc min and max values of mandatory borders */ *bmin = v_bord[0]; for (i = 0; i < *length_v_bord; i++) - if (v_bord[i] < *bmin) - *bmin = v_bord[i]; + if (v_bord[i] < *bmin) *bmin = v_bord[i]; *bmax = v_bord[0]; for (i = 0; i < *length_v_bord; i++) - if (v_bord[i] > *bmax) - *bmax = v_bord[i]; - + if (v_bord[i] > *bmax) *bmax = v_bord[i]; } - - /******************************************************************************* Functionname: fillFramePre ******************************************************************************* @@ -961,12 +907,8 @@ fillFrameTran (const int *v_tuningSegm, /*!< tuning: desired segment length Return: none *******************************************************************************/ -static void -fillFramePre (INT dmax, - INT *v_bord, INT *length_v_bord, - INT *v_freq, INT *length_v_freq, - INT bmin, INT rest) -{ +static void fillFramePre(INT dmax, INT *v_bord, INT *length_v_bord, INT *v_freq, + INT *length_v_freq, INT bmin, INT rest) { /* input state: v_bord = [(Ba),Bd1, Bd2 ,(Bd3)] @@ -990,8 +932,8 @@ fillFramePre (INT dmax, parts++; segm = rest / parts; - S = (segm - 2)>>1; - s = fixMin (8, 2 * S + 2); + S = (segm - 2) >> 1; + s = fixMin(8, 2 * S + 2); d = rest - (parts - 1) * s; } @@ -1005,10 +947,10 @@ fillFramePre (INT dmax, bord = bord - s; /* v_bord = [...,(Bf),(Ba),Bd1, Bd2 ,(Bd3)] */ - FDKsbrEnc_AddLeft (v_bord, length_v_bord, bord); + FDKsbrEnc_AddLeft(v_bord, length_v_bord, bord); /* v_freq = [...,(1 ),(Fa),Fd1,(Fd2), 1 ] */ - FDKsbrEnc_AddLeft (v_freq, length_v_freq, 1); + FDKsbrEnc_AddLeft(v_freq, length_v_freq, 1); } } @@ -1022,39 +964,37 @@ fillFramePre (INT dmax, \return void ****************************************************************************/ -static int -calcFillLengthMax (int tranPos, /*!< input : transient position (ref: tran det) */ - int numberTimeSlots /*!< input : number of timeslots */ - ) -{ +static int calcFillLengthMax( + int tranPos, /*!< input : transient position (ref: tran det) */ + int numberTimeSlots /*!< input : number of timeslots */ +) { int fmax; /* calculate transient position within envelope buffer */ - switch (numberTimeSlots) - { + switch (numberTimeSlots) { case NUMBER_TIME_SLOTS_2048: - if (tranPos < 4) - fmax = 6; - else if (tranPos == 4 || tranPos == 5) - fmax = 4; - else - fmax = 8; - break; + if (tranPos < 4) + fmax = 6; + else if (tranPos == 4 || tranPos == 5) + fmax = 4; + else + fmax = 8; + break; case NUMBER_TIME_SLOTS_1920: - if (tranPos < 4) - fmax = 5; - else if (tranPos == 4 || tranPos == 5) - fmax = 3; - else - fmax = 7; - break; + if (tranPos < 4) + fmax = 5; + else if (tranPos == 4 || tranPos == 5) + fmax = 3; + else + fmax = 7; + break; default: - fmax = 8; - break; + fmax = 8; + break; } return fmax; @@ -1083,11 +1023,10 @@ calcFillLengthMax (int tranPos, /*!< input : transient position (ref: t Return: none *******************************************************************************/ -static void -fillFramePost (INT *parts, INT *d, INT dmax, INT *v_bord, INT *length_v_bord, - INT *v_freq, INT *length_v_freq, INT bmax, - INT bufferFrameStart, INT numberTimeSlots, INT fmax) -{ +static void fillFramePost(INT *parts, INT *d, INT dmax, INT *v_bord, + INT *length_v_bord, INT *v_freq, INT *length_v_freq, + INT bmax, INT bufferFrameStart, INT numberTimeSlots, + INT fmax) { INT j, rest, segm, S, s = 0, bord; /* @@ -1100,7 +1039,7 @@ fillFramePost (INT *parts, INT *d, INT dmax, INT *v_bord, INT *length_v_bord, *d = rest; if (*d > 0) { - *parts = 1; /* start with one envelope */ + *parts = 1; /* start with one envelope */ /* calc # of additional envelopes and corresponding lengths */ @@ -1108,8 +1047,8 @@ fillFramePost (INT *parts, INT *d, INT dmax, INT *v_bord, INT *length_v_bord, *parts = *parts + 1; segm = rest / (*parts); - S = (segm - 2)>>1; - s = fixMin (fmax, 2 * S + 2); + S = (segm - 2) >> 1; + s = fixMin(fmax, 2 * S + 2); *d = rest - (*parts - 1) * s; } @@ -1120,25 +1059,21 @@ fillFramePost (INT *parts, INT *d, INT dmax, INT *v_bord, INT *length_v_bord, bord += s; /* v_bord = [...,(Bf),(Ba),Bd1, Bd2 ,(Bd3),(Bf)] */ - FDKsbrEnc_AddRight (v_bord, length_v_bord, bord); + FDKsbrEnc_AddRight(v_bord, length_v_bord, bord); /* v_freq = [...,(1 ),(Fa),Fd1,(Fd2), 1 , 1! ,1] */ - FDKsbrEnc_AddRight (v_freq, length_v_freq, 1); + FDKsbrEnc_AddRight(v_freq, length_v_freq, 1); } - } - else { + } else { *parts = 1; /* remove last element from v_bord and v_freq */ *length_v_bord = *length_v_bord - 1; *length_v_freq = *length_v_freq - 1; - } } - - /******************************************************************************* Functionname: fillFrameInter ******************************************************************************* @@ -1163,17 +1098,15 @@ fillFramePost (INT *parts, INT *d, INT dmax, INT *v_bord, INT *length_v_bord, Return: none *******************************************************************************/ -static void -fillFrameInter (INT *nL, const int *v_tuningSegm, INT *v_bord, INT *length_v_bord, - INT bmin, INT *v_freq, INT *length_v_freq, INT *v_bordFollow, - INT *length_v_bordFollow, INT *v_freqFollow, - INT *length_v_freqFollow, INT i_fillFollow, INT dmin, - INT dmax, INT numberTimeSlots) -{ +static void fillFrameInter(INT *nL, const int *v_tuningSegm, INT *v_bord, + INT *length_v_bord, INT bmin, INT *v_freq, + INT *length_v_freq, INT *v_bordFollow, + INT *length_v_bordFollow, INT *v_freqFollow, + INT *length_v_freqFollow, INT i_fillFollow, INT dmin, + INT dmax, INT numberTimeSlots) { INT middle, b_new, numBordFollow, bordMaxFollow, i; if (numberTimeSlots != NUMBER_TIME_SLOTS_1152) { - /* % remove fill borders: */ if (i_fillFollow >= 1) { *length_v_bordFollow = i_fillFollow; @@ -1197,65 +1130,61 @@ fillFrameInter (INT *nL, const int *v_tuningSegm, INT *v_bord, INT *length_v_bor b_new = *length_v_bord; - if (middle <= dmax) { - if (middle >= dmin) { /* concatenate */ - FDKsbrEnc_AddVecLeft (v_bord, length_v_bord, v_bordFollow, *length_v_bordFollow); - FDKsbrEnc_AddVecLeft (v_freq, length_v_freq, v_freqFollow, *length_v_freqFollow); + if (middle >= dmin) { /* concatenate */ + FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow, + *length_v_bordFollow); + FDKsbrEnc_AddVecLeft(v_freq, length_v_freq, v_freqFollow, + *length_v_freqFollow); } else { - if (v_tuningSegm[0] != 0) { /* remove one new border and concatenate */ + if (v_tuningSegm[0] != 0) { /* remove one new border and concatenate */ *length_v_bord = b_new - 1; - FDKsbrEnc_AddVecLeft (v_bord, length_v_bord, v_bordFollow, - *length_v_bordFollow); + FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow, + *length_v_bordFollow); *length_v_freq = b_new - 1; - FDKsbrEnc_AddVecLeft (v_freq + 1, length_v_freq, v_freqFollow, - *length_v_freqFollow); - } - else { - if (*length_v_bordFollow > 1) { /* remove one old border and concatenate */ - FDKsbrEnc_AddVecLeft (v_bord, length_v_bord, v_bordFollow, - *length_v_bordFollow - 1); - FDKsbrEnc_AddVecLeft (v_freq, length_v_freq, v_freqFollow, - *length_v_bordFollow - 1); + FDKsbrEnc_AddVecLeft(v_freq + 1, length_v_freq, v_freqFollow, + *length_v_freqFollow); + } else { + if (*length_v_bordFollow > + 1) { /* remove one old border and concatenate */ + FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow, + *length_v_bordFollow - 1); + FDKsbrEnc_AddVecLeft(v_freq, length_v_freq, v_freqFollow, + *length_v_bordFollow - 1); *nL = *nL - 1; - } - else { /* remove new "transient" border and concatenate */ + } else { /* remove new "transient" border and concatenate */ - for (i = 0; i < *length_v_bord - 1; i++) - v_bord[i] = v_bord[i + 1]; + for (i = 0; i < *length_v_bord - 1; i++) v_bord[i] = v_bord[i + 1]; - for (i = 0; i < *length_v_freq - 1; i++) - v_freq[i] = v_freq[i + 1]; + for (i = 0; i < *length_v_freq - 1; i++) v_freq[i] = v_freq[i + 1]; *length_v_bord = b_new - 1; *length_v_freq = b_new - 1; - FDKsbrEnc_AddVecLeft (v_bord, length_v_bord, v_bordFollow, - *length_v_bordFollow); - FDKsbrEnc_AddVecLeft (v_freq, length_v_freq, v_freqFollow, - *length_v_freqFollow); + FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow, + *length_v_bordFollow); + FDKsbrEnc_AddVecLeft(v_freq, length_v_freq, v_freqFollow, + *length_v_freqFollow); } } } + } else { /* middle > dmax */ + + fillFramePre(dmax, v_bord, length_v_bord, v_freq, length_v_freq, bmin, + middle); + FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow, + *length_v_bordFollow); + FDKsbrEnc_AddVecLeft(v_freq, length_v_freq, v_freqFollow, + *length_v_freqFollow); } - else { /* middle > dmax */ - - fillFramePre (dmax, v_bord, length_v_bord, v_freq, length_v_freq, bmin, - middle); - FDKsbrEnc_AddVecLeft (v_bord, length_v_bord, v_bordFollow, *length_v_bordFollow); - FDKsbrEnc_AddVecLeft (v_freq, length_v_freq, v_freqFollow, *length_v_freqFollow); - } - - } - else { /* numberTimeSlots==NUMBER_TIME_SLOTS_1152 */ - - INT l,m; + } else { /* numberTimeSlots==NUMBER_TIME_SLOTS_1152 */ + INT l, m; /*------------------------------------------------------------------------ remove fill borders @@ -1277,17 +1206,15 @@ fillFrameInter (INT *nL, const int *v_tuningSegm, INT *v_bord, INT *length_v_bor /* intervals: i) middle < 0 : overlap, must remove borders - ii) 0 <= middle < dmin : no overlap but too tight, must remove borders - iii) dmin <= middle <= dmax : ok, just concatenate - iv) dmax <= middle : too wide, must add borders + ii) 0 <= middle < dmin : no overlap but too tight, must remove + borders iii) dmin <= middle <= dmax : ok, just concatenate iv) dmax + <= middle : too wide, must add borders */ /* first remove old non-fill-borders... */ while (middle < 0) { - /* ...but don't remove all of them */ - if (numBordFollow == 1) - break; + if (numBordFollow == 1) break; numBordFollow--; bordMaxFollow = v_bordFollow[numBordFollow - 1]; @@ -1295,12 +1222,9 @@ fillFrameInter (INT *nL, const int *v_tuningSegm, INT *v_bord, INT *length_v_bor } /* if this isn't enough, remove new non-fill borders */ - if (middle < 0) - { - for (l = 0, m = 0 ; l < *length_v_bord ; l++) - { - if(v_bord[l]> bordMaxFollow) - { + if (middle < 0) { + for (l = 0, m = 0; l < *length_v_bord; l++) { + if (v_bord[l] > bordMaxFollow) { v_bord[m] = v_bord[l]; v_freq[m] = v_freq[l]; m++; @@ -1311,7 +1235,6 @@ fillFrameInter (INT *nL, const int *v_tuningSegm, INT *v_bord, INT *length_v_bor *length_v_freq = l; bmin = v_bord[0]; - } /*------------------------------------------------------------------------ @@ -1331,69 +1254,62 @@ fillFrameInter (INT *nL, const int *v_tuningSegm, INT *v_bord, INT *length_v_bor /* now middle should be >= 0 */ middle = bmin - bordMaxFollow; - if (middle <= dmin) /* (ii) */ + if (middle <= dmin) /* (ii) */ { b_new = *length_v_bord; - if (v_tuningSegm[0] != 0) - { + if (v_tuningSegm[0] != 0) { /* remove new "luxury" border and concatenate */ *length_v_bord = b_new - 1; - FDKsbrEnc_AddVecLeft (v_bord, length_v_bord, v_bordFollow, - *length_v_bordFollow); + FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow, + *length_v_bordFollow); *length_v_freq = b_new - 1; - FDKsbrEnc_AddVecLeft (v_freq + 1, length_v_freq, v_freqFollow, - *length_v_freqFollow); + FDKsbrEnc_AddVecLeft(v_freq + 1, length_v_freq, v_freqFollow, + *length_v_freqFollow); - } - else if (*length_v_bordFollow > 1) - { + } else if (*length_v_bordFollow > 1) { /* remove old border and concatenate */ - FDKsbrEnc_AddVecLeft (v_bord, length_v_bord, v_bordFollow, - *length_v_bordFollow - 1); - FDKsbrEnc_AddVecLeft (v_freq, length_v_freq, v_freqFollow, - *length_v_bordFollow - 1); + FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow, + *length_v_bordFollow - 1); + FDKsbrEnc_AddVecLeft(v_freq, length_v_freq, v_freqFollow, + *length_v_bordFollow - 1); *nL = *nL - 1; - } - else - { + } else { /* remove new border and concatenate */ - for (i = 0; i < *length_v_bord - 1; i++) - v_bord[i] = v_bord[i + 1]; + for (i = 0; i < *length_v_bord - 1; i++) v_bord[i] = v_bord[i + 1]; - for (i = 0; i < *length_v_freq - 1; i++) - v_freq[i] = v_freq[i + 1]; + for (i = 0; i < *length_v_freq - 1; i++) v_freq[i] = v_freq[i + 1]; *length_v_bord = b_new - 1; *length_v_freq = b_new - 1; - FDKsbrEnc_AddVecLeft (v_bord, length_v_bord, v_bordFollow, - *length_v_bordFollow); - FDKsbrEnc_AddVecLeft (v_freq, length_v_freq, v_freqFollow, - *length_v_freqFollow); + FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow, + *length_v_bordFollow); + FDKsbrEnc_AddVecLeft(v_freq, length_v_freq, v_freqFollow, + *length_v_freqFollow); } - } - else if ((middle >= dmin) && (middle <= dmax)) /* (iii) */ + } else if ((middle >= dmin) && (middle <= dmax)) /* (iii) */ { /* concatenate */ - FDKsbrEnc_AddVecLeft (v_bord, length_v_bord, v_bordFollow, *length_v_bordFollow); - FDKsbrEnc_AddVecLeft (v_freq, length_v_freq, v_freqFollow, *length_v_freqFollow); + FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow, + *length_v_bordFollow); + FDKsbrEnc_AddVecLeft(v_freq, length_v_freq, v_freqFollow, + *length_v_freqFollow); - } - else /* (iv) */ + } else /* (iv) */ { - fillFramePre (dmax, v_bord, length_v_bord, v_freq, length_v_freq, bmin, - middle); - FDKsbrEnc_AddVecLeft (v_bord, length_v_bord, v_bordFollow, *length_v_bordFollow); - FDKsbrEnc_AddVecLeft (v_freq, length_v_freq, v_freqFollow, *length_v_freqFollow); + fillFramePre(dmax, v_bord, length_v_bord, v_freq, length_v_freq, bmin, + middle); + FDKsbrEnc_AddVecLeft(v_bord, length_v_bord, v_bordFollow, + *length_v_bordFollow); + FDKsbrEnc_AddVecLeft(v_freq, length_v_freq, v_freqFollow, + *length_v_freqFollow); } } } - - /******************************************************************************* Functionname: calcFrameClass ******************************************************************************* @@ -1405,42 +1321,49 @@ fillFrameInter (INT *nL, const int *v_tuningSegm, INT *v_bord, INT *length_v_bor Return: none *******************************************************************************/ -static void -calcFrameClass (FRAME_CLASS *frameClass, FRAME_CLASS *frameClassOld, INT tranFlag, - INT *spreadFlag) -{ - +static void calcFrameClass(FRAME_CLASS *frameClass, FRAME_CLASS *frameClassOld, + INT tranFlag, INT *spreadFlag) { switch (*frameClassOld) { - case FIXFIXonly: - case FIXFIX: - if (tranFlag) *frameClass = FIXVAR; - else *frameClass = FIXFIX; - break; - case FIXVAR: - if (tranFlag) { *frameClass = VARVAR; *spreadFlag = 0; } - else { - if (*spreadFlag) *frameClass = VARVAR; - else *frameClass = VARFIX; - } - break; - case VARFIX: - if (tranFlag) *frameClass = FIXVAR; - else *frameClass = FIXFIX; - break; - case VARVAR: - if (tranFlag) { *frameClass = VARVAR; *spreadFlag = 0; } - else { - if (*spreadFlag) *frameClass = VARVAR; - else *frameClass = VARFIX; - } - break; + case FIXFIXonly: + case FIXFIX: + if (tranFlag) + *frameClass = FIXVAR; + else + *frameClass = FIXFIX; + break; + case FIXVAR: + if (tranFlag) { + *frameClass = VARVAR; + *spreadFlag = 0; + } else { + if (*spreadFlag) + *frameClass = VARVAR; + else + *frameClass = VARFIX; + } + break; + case VARFIX: + if (tranFlag) + *frameClass = FIXVAR; + else + *frameClass = FIXFIX; + break; + case VARVAR: + if (tranFlag) { + *frameClass = VARVAR; + *spreadFlag = 0; + } else { + if (*spreadFlag) + *frameClass = VARVAR; + else + *frameClass = VARFIX; + } + break; }; *frameClassOld = *frameClass; } - - /******************************************************************************* Functionname: specialCase ******************************************************************************* @@ -1459,28 +1382,24 @@ calcFrameClass (FRAME_CLASS *frameClass, FRAME_CLASS *frameClassOld, INT tranFla Return: none *******************************************************************************/ -static void -specialCase (INT *spreadFlag, INT allowSpread, INT *v_bord, - INT *length_v_bord, INT *v_freq, INT *length_v_freq, INT *parts, - INT d) -{ +static void specialCase(INT *spreadFlag, INT allowSpread, INT *v_bord, + INT *length_v_bord, INT *v_freq, INT *length_v_freq, + INT *parts, INT d) { INT L; L = *length_v_bord; - if (allowSpread) { /* add one "step 8" */ + if (allowSpread) { /* add one "step 8" */ *spreadFlag = 1; - FDKsbrEnc_AddRight (v_bord, length_v_bord, v_bord[L - 1] + 8); - FDKsbrEnc_AddRight (v_freq, length_v_freq, 1); + FDKsbrEnc_AddRight(v_bord, length_v_bord, v_bord[L - 1] + 8); + FDKsbrEnc_AddRight(v_freq, length_v_freq, 1); (*parts)++; - } - else { - if (d == 1) { /* stretch one slot */ + } else { + if (d == 1) { /* stretch one slot */ *length_v_bord = L - 1; *length_v_freq = L - 1; - } - else { - if ((v_bord[L - 1] - v_bord[L - 2]) > 2) { /* compress one quant step */ + } else { + if ((v_bord[L - 1] - v_bord[L - 2]) > 2) { /* compress one quant step */ v_bord[L - 1] = v_bord[L - 1] - 2; v_freq[*length_v_freq - 1] = 0; /* use low res for short segment */ } @@ -1488,8 +1407,6 @@ specialCase (INT *spreadFlag, INT allowSpread, INT *v_bord, } } - - /******************************************************************************* Functionname: calcCmonBorder ******************************************************************************* @@ -1505,14 +1422,13 @@ specialCase (INT *spreadFlag, INT allowSpread, INT *v_bord, Return: none *******************************************************************************/ -static void -calcCmonBorder (INT *i_cmon, INT *i_tran, INT *v_bord, INT *length_v_bord, - INT tran, INT bufferFrameStart, INT numberTimeSlots) -{ /* FH 00-06-26 */ +static void calcCmonBorder(INT *i_cmon, INT *i_tran, INT *v_bord, + INT *length_v_bord, INT tran, INT bufferFrameStart, + INT numberTimeSlots) { /* FH 00-06-26 */ INT i; for (i = 0; i < *length_v_bord; i++) - if (v_bord[i] >= bufferFrameStart + numberTimeSlots) { /* FH 00-06-26 */ + if (v_bord[i] >= bufferFrameStart + numberTimeSlots) { /* FH 00-06-26 */ *i_cmon = i; break; } @@ -1522,8 +1438,7 @@ calcCmonBorder (INT *i_cmon, INT *i_tran, INT *v_bord, INT *length_v_bord, if (v_bord[i] >= tran) { *i_tran = i; break; - } - else + } else *i_tran = EMPTY; } @@ -1549,13 +1464,12 @@ calcCmonBorder (INT *i_cmon, INT *i_tran, INT *v_bord, INT *length_v_bord, Return: none *******************************************************************************/ -static void -keepForFollowUp (INT *v_bordFollow, INT *length_v_bordFollow, - INT *v_freqFollow, INT *length_v_freqFollow, - INT *i_tranFollow, INT *i_fillFollow, INT *v_bord, - INT *length_v_bord, INT *v_freq, INT i_cmon, INT i_tran, - INT parts, INT numberTimeSlots) -{ /* FH 00-06-26 */ +static void keepForFollowUp(INT *v_bordFollow, INT *length_v_bordFollow, + INT *v_freqFollow, INT *length_v_freqFollow, + INT *i_tranFollow, INT *i_fillFollow, INT *v_bord, + INT *length_v_bord, INT *v_freq, INT i_cmon, + INT i_tran, INT parts, + INT numberTimeSlots) { /* FH 00-06-26 */ INT L, i, j; L = *length_v_bord; @@ -1564,7 +1478,7 @@ keepForFollowUp (INT *v_bordFollow, INT *length_v_bordFollow, (*length_v_freqFollow) = 0; for (j = 0, i = i_cmon; i < L; i++, j++) { - v_bordFollow[j] = v_bord[i] - numberTimeSlots; /* FH 00-06-26 */ + v_bordFollow[j] = v_bord[i] - numberTimeSlots; /* FH 00-06-26 */ v_freqFollow[j] = v_freq[i]; (*length_v_bordFollow)++; (*length_v_freqFollow)++; @@ -1574,7 +1488,6 @@ keepForFollowUp (INT *v_bordFollow, INT *length_v_bordFollow, else *i_tranFollow = EMPTY; *i_fillFollow = L - (parts - 1) - i_cmon; - } /******************************************************************************* @@ -1597,14 +1510,10 @@ keepForFollowUp (INT *v_bordFollow, INT *length_v_bordFollow, Return: none *******************************************************************************/ -static void -calcCtrlSignal (HANDLE_SBR_GRID hSbrGrid, - FRAME_CLASS frameClass, INT *v_bord, INT length_v_bord, INT *v_freq, - INT length_v_freq, INT i_cmon, INT i_tran, INT spreadFlag, - INT nL) -{ - - +static void calcCtrlSignal(HANDLE_SBR_GRID hSbrGrid, FRAME_CLASS frameClass, + INT *v_bord, INT length_v_bord, INT *v_freq, + INT length_v_freq, INT i_cmon, INT i_tran, + INT spreadFlag, INT nL) { INT i, r, a, n, p, b, aL, aR, ntot, nmax, nR; INT *v_f = hSbrGrid->v_f; @@ -1618,164 +1527,152 @@ calcCtrlSignal (HANDLE_SBR_GRID hSbrGrid, INT length_v_rL = 0; switch (frameClass) { - case FIXVAR: - /* absolute border: */ - - a = v_bord[i_cmon]; - - /* relative borders: */ - length_v_r = 0; - i = i_cmon; - - while (i >= 1) { - r = v_bord[i] - v_bord[i - 1]; - FDKsbrEnc_AddRight (v_r, &length_v_r, r); - i--; - } - - - /* number of relative borders: */ - n = length_v_r; - - - /* freq res: */ - for (i = 0; i < i_cmon; i++) - v_f[i] = v_freq[i_cmon - 1 - i]; - v_f[i_cmon] = 1; + case FIXVAR: + /* absolute border: */ - /* pointer: */ - p = (i_cmon >= i_tran && i_tran != EMPTY) ? (i_cmon - i_tran + 1) : (0) ; + a = v_bord[i_cmon]; - hSbrGrid->frameClass = frameClass; - hSbrGrid->bs_abs_bord = a; - hSbrGrid->n = n; - hSbrGrid->p = p; + /* relative borders: */ + length_v_r = 0; + i = i_cmon; - break; - case VARFIX: - /* absolute border: */ - a = v_bord[0]; + while (i >= 1) { + r = v_bord[i] - v_bord[i - 1]; + FDKsbrEnc_AddRight(v_r, &length_v_r, r); + i--; + } - /* relative borders: */ - length_v_r = 0; + /* number of relative borders: */ + n = length_v_r; - for (i = 1; i < length_v_bord; i++) { - r = v_bord[i] - v_bord[i - 1]; - FDKsbrEnc_AddRight (v_r, &length_v_r, r); - } + /* freq res: */ + for (i = 0; i < i_cmon; i++) v_f[i] = v_freq[i_cmon - 1 - i]; + v_f[i_cmon] = 1; - /* number of relative borders: */ - n = length_v_r; + /* pointer: */ + p = (i_cmon >= i_tran && i_tran != EMPTY) ? (i_cmon - i_tran + 1) : (0); - /* freq res: */ - FDKmemcpy (v_f, v_freq, length_v_freq * sizeof (INT)); + hSbrGrid->frameClass = frameClass; + hSbrGrid->bs_abs_bord = a; + hSbrGrid->n = n; + hSbrGrid->p = p; + break; + case VARFIX: + /* absolute border: */ + a = v_bord[0]; - /* pointer: */ - p = (i_tran >= 0 && i_tran != EMPTY) ? (i_tran + 1) : (0) ; + /* relative borders: */ + length_v_r = 0; - hSbrGrid->frameClass = frameClass; - hSbrGrid->bs_abs_bord = a; - hSbrGrid->n = n; - hSbrGrid->p = p; + for (i = 1; i < length_v_bord; i++) { + r = v_bord[i] - v_bord[i - 1]; + FDKsbrEnc_AddRight(v_r, &length_v_r, r); + } - break; - case VARVAR: - if (spreadFlag) { - /* absolute borders: */ - b = length_v_bord; + /* number of relative borders: */ + n = length_v_r; - aL = v_bord[0]; - aR = v_bord[b - 1]; + /* freq res: */ + FDKmemcpy(v_f, v_freq, length_v_freq * sizeof(INT)); + /* pointer: */ + p = (i_tran >= 0 && i_tran != EMPTY) ? (i_tran + 1) : (0); - /* number of relative borders: */ - ntot = b - 2; + hSbrGrid->frameClass = frameClass; + hSbrGrid->bs_abs_bord = a; + hSbrGrid->n = n; + hSbrGrid->p = p; - nmax = 2; /* n: {0,1,2} */ - if (ntot > nmax) { - nL = nmax; - nR = ntot - nmax; - } - else { - nL = ntot; - nR = 0; - } + break; + case VARVAR: + if (spreadFlag) { + /* absolute borders: */ + b = length_v_bord; + + aL = v_bord[0]; + aR = v_bord[b - 1]; + + /* number of relative borders: */ + ntot = b - 2; + + nmax = 2; /* n: {0,1,2} */ + if (ntot > nmax) { + nL = nmax; + nR = ntot - nmax; + } else { + nL = ntot; + nR = 0; + } - /* relative borders: */ - length_v_rL = 0; - for (i = 1; i <= nL; i++) { - r = v_bord[i] - v_bord[i - 1]; - FDKsbrEnc_AddRight (v_rL, &length_v_rL, r); - } + /* relative borders: */ + length_v_rL = 0; + for (i = 1; i <= nL; i++) { + r = v_bord[i] - v_bord[i - 1]; + FDKsbrEnc_AddRight(v_rL, &length_v_rL, r); + } - length_v_rR = 0; - i = b - 1; - while (i >= b - nR) { - r = v_bord[i] - v_bord[i - 1]; - FDKsbrEnc_AddRight (v_rR, &length_v_rR, r); - i--; - } + length_v_rR = 0; + i = b - 1; + while (i >= b - nR) { + r = v_bord[i] - v_bord[i - 1]; + FDKsbrEnc_AddRight(v_rR, &length_v_rR, r); + i--; + } - /* pointer (only one due to constraint in frame info): */ - p = (i_tran > 0 && i_tran != EMPTY) ? (b - i_tran) : (0) ; + /* pointer (only one due to constraint in frame info): */ + p = (i_tran > 0 && i_tran != EMPTY) ? (b - i_tran) : (0); - /* freq res: */ + /* freq res: */ - for (i = 0; i < b - 1; i++) - v_fLR[i] = v_freq[i]; - } - else { + for (i = 0; i < b - 1; i++) v_fLR[i] = v_freq[i]; + } else { + length_v_bord = i_cmon + 1; - length_v_bord = i_cmon + 1; - length_v_freq = i_cmon + 1; + /* absolute borders: */ + b = length_v_bord; + aL = v_bord[0]; + aR = v_bord[b - 1]; - /* absolute borders: */ - b = length_v_bord; + /* number of relative borders: */ + ntot = b - 2; + nR = ntot - nL; - aL = v_bord[0]; - aR = v_bord[b - 1]; + /* relative borders: */ + length_v_rL = 0; + for (i = 1; i <= nL; i++) { + r = v_bord[i] - v_bord[i - 1]; + FDKsbrEnc_AddRight(v_rL, &length_v_rL, r); + } - /* number of relative borders: */ - ntot = b - 2; - nR = ntot - nL; + length_v_rR = 0; + i = b - 1; + while (i >= b - nR) { + r = v_bord[i] - v_bord[i - 1]; + FDKsbrEnc_AddRight(v_rR, &length_v_rR, r); + i--; + } - /* relative borders: */ - length_v_rL = 0; - for (i = 1; i <= nL; i++) { - r = v_bord[i] - v_bord[i - 1]; - FDKsbrEnc_AddRight (v_rL, &length_v_rL, r); - } + /* pointer (only one due to constraint in frame info): */ + p = (i_cmon >= i_tran && i_tran != EMPTY) ? (i_cmon - i_tran + 1) : (0); - length_v_rR = 0; - i = b - 1; - while (i >= b - nR) { - r = v_bord[i] - v_bord[i - 1]; - FDKsbrEnc_AddRight (v_rR, &length_v_rR, r); - i--; + /* freq res: */ + for (i = 0; i < b - 1; i++) v_fLR[i] = v_freq[i]; } - /* pointer (only one due to constraint in frame info): */ - p = (i_cmon >= i_tran && i_tran != EMPTY) ? (i_cmon - i_tran + 1) : (0) ; + hSbrGrid->frameClass = frameClass; + hSbrGrid->bs_abs_bord_0 = aL; + hSbrGrid->bs_abs_bord_1 = aR; + hSbrGrid->bs_num_rel_0 = nL; + hSbrGrid->bs_num_rel_1 = nR; + hSbrGrid->p = p; - /* freq res: */ - for (i = 0; i < b - 1; i++) - v_fLR[i] = v_freq[i]; - } - - hSbrGrid->frameClass = frameClass; - hSbrGrid->bs_abs_bord_0 = aL; - hSbrGrid->bs_abs_bord_1 = aR; - hSbrGrid->bs_num_rel_0 = nL; - hSbrGrid->bs_num_rel_1 = nR; - hSbrGrid->p = p; - - break; + break; - default: - /* do nothing */ - break; + default: + /* do nothing */ + break; } } @@ -1795,79 +1692,77 @@ calcCtrlSignal (HANDLE_SBR_GRID hSbrGrid, Written: Andreas Schneider Revised: *******************************************************************************/ -static void -createDefFrameInfo(HANDLE_SBR_FRAME_INFO hSbrFrameInfo, INT nEnv, INT nTimeSlots) -{ +static void createDefFrameInfo(HANDLE_SBR_FRAME_INFO hSbrFrameInfo, INT nEnv, + INT nTimeSlots) { switch (nEnv) { - case 1: - switch (nTimeSlots) { - case NUMBER_TIME_SLOTS_1920: - FDKmemcpy (hSbrFrameInfo, &frameInfo1_1920, sizeof (SBR_FRAME_INFO)); - break; - case NUMBER_TIME_SLOTS_2048: - FDKmemcpy (hSbrFrameInfo, &frameInfo1_2048, sizeof (SBR_FRAME_INFO)); - break; - case NUMBER_TIME_SLOTS_1152: - FDKmemcpy (hSbrFrameInfo, &frameInfo1_1152, sizeof (SBR_FRAME_INFO)); - break; - case NUMBER_TIME_SLOTS_2304: - FDKmemcpy (hSbrFrameInfo, &frameInfo1_2304, sizeof (SBR_FRAME_INFO)); - break; - case NUMBER_TIME_SLOTS_512LD: - FDKmemcpy (hSbrFrameInfo, &frameInfo1_512LD, sizeof (SBR_FRAME_INFO)); - break; - default: - FDK_ASSERT(0); - } - break; - case 2: - switch (nTimeSlots) { - case NUMBER_TIME_SLOTS_1920: - FDKmemcpy (hSbrFrameInfo, &frameInfo2_1920, sizeof (SBR_FRAME_INFO)); - break; - case NUMBER_TIME_SLOTS_2048: - FDKmemcpy (hSbrFrameInfo, &frameInfo2_2048, sizeof (SBR_FRAME_INFO)); - break; - case NUMBER_TIME_SLOTS_1152: - FDKmemcpy (hSbrFrameInfo, &frameInfo2_1152, sizeof (SBR_FRAME_INFO)); - break; - case NUMBER_TIME_SLOTS_2304: - FDKmemcpy (hSbrFrameInfo, &frameInfo2_2304, sizeof (SBR_FRAME_INFO)); - break; - case NUMBER_TIME_SLOTS_512LD: - FDKmemcpy (hSbrFrameInfo, &frameInfo2_512LD, sizeof (SBR_FRAME_INFO)); - break; - default: - FDK_ASSERT(0); - } - break; - case 4: - switch (nTimeSlots) { - case NUMBER_TIME_SLOTS_1920: - FDKmemcpy (hSbrFrameInfo, &frameInfo4_1920, sizeof (SBR_FRAME_INFO)); - break; - case NUMBER_TIME_SLOTS_2048: - FDKmemcpy (hSbrFrameInfo, &frameInfo4_2048, sizeof (SBR_FRAME_INFO)); - break; - case NUMBER_TIME_SLOTS_1152: - FDKmemcpy (hSbrFrameInfo, &frameInfo4_1152, sizeof (SBR_FRAME_INFO)); + case 1: + switch (nTimeSlots) { + case NUMBER_TIME_SLOTS_1920: + FDKmemcpy(hSbrFrameInfo, &frameInfo1_1920, sizeof(SBR_FRAME_INFO)); + break; + case NUMBER_TIME_SLOTS_2048: + FDKmemcpy(hSbrFrameInfo, &frameInfo1_2048, sizeof(SBR_FRAME_INFO)); + break; + case NUMBER_TIME_SLOTS_1152: + FDKmemcpy(hSbrFrameInfo, &frameInfo1_1152, sizeof(SBR_FRAME_INFO)); + break; + case NUMBER_TIME_SLOTS_2304: + FDKmemcpy(hSbrFrameInfo, &frameInfo1_2304, sizeof(SBR_FRAME_INFO)); + break; + case NUMBER_TIME_SLOTS_512LD: + FDKmemcpy(hSbrFrameInfo, &frameInfo1_512LD, sizeof(SBR_FRAME_INFO)); + break; + default: + FDK_ASSERT(0); + } break; - case NUMBER_TIME_SLOTS_2304: - FDKmemcpy (hSbrFrameInfo, &frameInfo4_2304, sizeof (SBR_FRAME_INFO)); + case 2: + switch (nTimeSlots) { + case NUMBER_TIME_SLOTS_1920: + FDKmemcpy(hSbrFrameInfo, &frameInfo2_1920, sizeof(SBR_FRAME_INFO)); + break; + case NUMBER_TIME_SLOTS_2048: + FDKmemcpy(hSbrFrameInfo, &frameInfo2_2048, sizeof(SBR_FRAME_INFO)); + break; + case NUMBER_TIME_SLOTS_1152: + FDKmemcpy(hSbrFrameInfo, &frameInfo2_1152, sizeof(SBR_FRAME_INFO)); + break; + case NUMBER_TIME_SLOTS_2304: + FDKmemcpy(hSbrFrameInfo, &frameInfo2_2304, sizeof(SBR_FRAME_INFO)); + break; + case NUMBER_TIME_SLOTS_512LD: + FDKmemcpy(hSbrFrameInfo, &frameInfo2_512LD, sizeof(SBR_FRAME_INFO)); + break; + default: + FDK_ASSERT(0); + } break; - case NUMBER_TIME_SLOTS_512LD: - FDKmemcpy (hSbrFrameInfo, &frameInfo4_512LD, sizeof (SBR_FRAME_INFO)); + case 4: + switch (nTimeSlots) { + case NUMBER_TIME_SLOTS_1920: + FDKmemcpy(hSbrFrameInfo, &frameInfo4_1920, sizeof(SBR_FRAME_INFO)); + break; + case NUMBER_TIME_SLOTS_2048: + FDKmemcpy(hSbrFrameInfo, &frameInfo4_2048, sizeof(SBR_FRAME_INFO)); + break; + case NUMBER_TIME_SLOTS_1152: + FDKmemcpy(hSbrFrameInfo, &frameInfo4_1152, sizeof(SBR_FRAME_INFO)); + break; + case NUMBER_TIME_SLOTS_2304: + FDKmemcpy(hSbrFrameInfo, &frameInfo4_2304, sizeof(SBR_FRAME_INFO)); + break; + case NUMBER_TIME_SLOTS_512LD: + FDKmemcpy(hSbrFrameInfo, &frameInfo4_512LD, sizeof(SBR_FRAME_INFO)); + break; + default: + FDK_ASSERT(0); + } break; default: FDK_ASSERT(0); - } - break; - default: - FDK_ASSERT(0); } } - /******************************************************************************* Functionname: ctrlSignal2FrameInfo ******************************************************************************* @@ -1886,171 +1781,177 @@ createDefFrameInfo(HANDLE_SBR_FRAME_INFO hSbrFrameInfo, INT nEnv, INT nTimeSlots Return: void; hSbrFrameInfo contains the updated FRAME_INFO struct *******************************************************************************/ -static void -ctrlSignal2FrameInfo ( - HANDLE_SBR_GRID hSbrGrid, /* input : the grid handle */ - HANDLE_SBR_FRAME_INFO hSbrFrameInfo, /* output: the frame info handle */ - FREQ_RES *freq_res_fixfix /* in/out: frequency resolution for FIXFIX frames */ - ) -{ +static void ctrlSignal2FrameInfo( + HANDLE_SBR_GRID hSbrGrid, /* input : the grid handle */ + HANDLE_SBR_FRAME_INFO hSbrFrameInfo, /* output: the frame info handle */ + FREQ_RES + *freq_res_fixfix /* in/out: frequency resolution for FIXFIX frames */ +) { INT frameSplit = 0; INT nEnv = 0, border = 0, i, k, p /*?*/; INT *v_r = hSbrGrid->bs_rel_bord; INT *v_f = hSbrGrid->v_f; FRAME_CLASS frameClass = hSbrGrid->frameClass; - INT bufferFrameStart = hSbrGrid->bufferFrameStart; - INT numberTimeSlots = hSbrGrid->numberTimeSlots; + INT bufferFrameStart = hSbrGrid->bufferFrameStart; + INT numberTimeSlots = hSbrGrid->numberTimeSlots; switch (frameClass) { - case FIXFIX: - createDefFrameInfo(hSbrFrameInfo, hSbrGrid->bs_num_env, numberTimeSlots); + case FIXFIX: + createDefFrameInfo(hSbrFrameInfo, hSbrGrid->bs_num_env, numberTimeSlots); - frameSplit = (hSbrFrameInfo->nEnvelopes > 1); - for (i = 0; i < hSbrFrameInfo->nEnvelopes; i++) { - hSbrGrid->v_f[i] = hSbrFrameInfo->freqRes[i] = freq_res_fixfix[frameSplit]; - } - break; + frameSplit = (hSbrFrameInfo->nEnvelopes > 1); + for (i = 0; i < hSbrFrameInfo->nEnvelopes; i++) { + hSbrGrid->v_f[i] = hSbrFrameInfo->freqRes[i] = + freq_res_fixfix[frameSplit]; + } + break; - case FIXVAR: - case VARFIX: - nEnv = hSbrGrid->n + 1; /* read n [SBR_NUM_BITS bits] */ /*? snd*/ - FDK_ASSERT(nEnv <= MAX_ENVELOPES_FIXVAR_VARFIX); + case FIXVAR: + case VARFIX: + nEnv = hSbrGrid->n + 1; /* read n [SBR_NUM_BITS bits] */ /*? snd*/ + FDK_ASSERT(nEnv <= MAX_ENVELOPES_FIXVAR_VARFIX); - hSbrFrameInfo->nEnvelopes = nEnv; + hSbrFrameInfo->nEnvelopes = nEnv; - border = hSbrGrid->bs_abs_bord; /* read the absolute border */ + border = hSbrGrid->bs_abs_bord; /* read the absolute border */ - if (nEnv == 1) - hSbrFrameInfo->nNoiseEnvelopes = 1; - else - hSbrFrameInfo->nNoiseEnvelopes = 2; + if (nEnv == 1) + hSbrFrameInfo->nNoiseEnvelopes = 1; + else + hSbrFrameInfo->nNoiseEnvelopes = 2; - break; + break; - default: - /* do nothing */ - break; + default: + /* do nothing */ + break; } switch (frameClass) { - case FIXVAR: - hSbrFrameInfo->borders[0] = bufferFrameStart; /* start-position of 1st envelope */ + case FIXVAR: + hSbrFrameInfo->borders[0] = + bufferFrameStart; /* start-position of 1st envelope */ - hSbrFrameInfo->borders[nEnv] = border; + hSbrFrameInfo->borders[nEnv] = border; - for (k = 0, i = nEnv - 1; k < nEnv - 1; k++, i--) { - border -= v_r[k]; - hSbrFrameInfo->borders[i] = border; - } + for (k = 0, i = nEnv - 1; k < nEnv - 1; k++, i--) { + border -= v_r[k]; + hSbrFrameInfo->borders[i] = border; + } - /* make either envelope nr. nEnv + 1 - p short; or don't shorten if p == 0 */ - p = hSbrGrid->p; - if (p == 0) { - hSbrFrameInfo->shortEnv = 0; - } else { - hSbrFrameInfo->shortEnv = nEnv + 1 - p; - } + /* make either envelope nr. nEnv + 1 - p short; or don't shorten if p == 0 + */ + p = hSbrGrid->p; + if (p == 0) { + hSbrFrameInfo->shortEnv = 0; + } else { + hSbrFrameInfo->shortEnv = nEnv + 1 - p; + } - for (k = 0, i = nEnv - 1; k < nEnv; k++, i--) { - hSbrFrameInfo->freqRes[i] = (FREQ_RES)v_f[k]; - } + for (k = 0, i = nEnv - 1; k < nEnv; k++, i--) { + hSbrFrameInfo->freqRes[i] = (FREQ_RES)v_f[k]; + } - /* if either there is no short envelope or the last envelope is short... */ - if (p == 0 || p == 1) { - hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[nEnv - 1]; - } else { - hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[hSbrFrameInfo->shortEnv]; - } + /* if either there is no short envelope or the last envelope is short... + */ + if (p == 0 || p == 1) { + hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[nEnv - 1]; + } else { + hSbrFrameInfo->bordersNoise[1] = + hSbrFrameInfo->borders[hSbrFrameInfo->shortEnv]; + } - break; + break; - case VARFIX: - /* in this case 'border' indicates the start of the 1st envelope */ - hSbrFrameInfo->borders[0] = border; + case VARFIX: + /* in this case 'border' indicates the start of the 1st envelope */ + hSbrFrameInfo->borders[0] = border; - for (k = 0; k < nEnv - 1; k++) { - border += v_r[k]; - hSbrFrameInfo->borders[k + 1] = border; - } + for (k = 0; k < nEnv - 1; k++) { + border += v_r[k]; + hSbrFrameInfo->borders[k + 1] = border; + } - hSbrFrameInfo->borders[nEnv] = bufferFrameStart + numberTimeSlots; + hSbrFrameInfo->borders[nEnv] = bufferFrameStart + numberTimeSlots; - p = hSbrGrid->p; - if (p == 0 || p == 1) { - hSbrFrameInfo->shortEnv = 0; - } else { - hSbrFrameInfo->shortEnv = p - 1; - } + p = hSbrGrid->p; + if (p == 0 || p == 1) { + hSbrFrameInfo->shortEnv = 0; + } else { + hSbrFrameInfo->shortEnv = p - 1; + } - for (k = 0; k < nEnv; k++) { - hSbrFrameInfo->freqRes[k] = (FREQ_RES)v_f[k]; - } + for (k = 0; k < nEnv; k++) { + hSbrFrameInfo->freqRes[k] = (FREQ_RES)v_f[k]; + } - switch (p) { - case 0: - hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[1]; - break; - case 1: - hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[nEnv - 1]; - break; - default: - hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[hSbrFrameInfo->shortEnv]; + switch (p) { + case 0: + hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[1]; + break; + case 1: + hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[nEnv - 1]; + break; + default: + hSbrFrameInfo->bordersNoise[1] = + hSbrFrameInfo->borders[hSbrFrameInfo->shortEnv]; + break; + } break; - } - break; - - case VARVAR: - nEnv = hSbrGrid->bs_num_rel_0 + hSbrGrid->bs_num_rel_1 + 1; - FDK_ASSERT(nEnv <= MAX_ENVELOPES_VARVAR); /* just to be sure */ - hSbrFrameInfo->nEnvelopes = nEnv; - hSbrFrameInfo->borders[0] = border = hSbrGrid->bs_abs_bord_0; + case VARVAR: + nEnv = hSbrGrid->bs_num_rel_0 + hSbrGrid->bs_num_rel_1 + 1; + FDK_ASSERT(nEnv <= MAX_ENVELOPES_VARVAR); /* just to be sure */ + hSbrFrameInfo->nEnvelopes = nEnv; - for (k = 0, i = 1; k < hSbrGrid->bs_num_rel_0; k++, i++) { - border += hSbrGrid->bs_rel_bord_0[k]; - hSbrFrameInfo->borders[i] = border; - } + hSbrFrameInfo->borders[0] = border = hSbrGrid->bs_abs_bord_0; - border = hSbrGrid->bs_abs_bord_1; - hSbrFrameInfo->borders[nEnv] = border; + for (k = 0, i = 1; k < hSbrGrid->bs_num_rel_0; k++, i++) { + border += hSbrGrid->bs_rel_bord_0[k]; + hSbrFrameInfo->borders[i] = border; + } - for (k = 0, i = nEnv - 1; k < hSbrGrid->bs_num_rel_1; k++, i--) { - border -= hSbrGrid->bs_rel_bord_1[k]; - hSbrFrameInfo->borders[i] = border; - } + border = hSbrGrid->bs_abs_bord_1; + hSbrFrameInfo->borders[nEnv] = border; - p = hSbrGrid->p; - if (p == 0) { - hSbrFrameInfo->shortEnv = 0; - } else { - hSbrFrameInfo->shortEnv = nEnv + 1 - p; - } + for (k = 0, i = nEnv - 1; k < hSbrGrid->bs_num_rel_1; k++, i--) { + border -= hSbrGrid->bs_rel_bord_1[k]; + hSbrFrameInfo->borders[i] = border; + } - for (k = 0; k < nEnv; k++) { - hSbrFrameInfo->freqRes[k] = (FREQ_RES)hSbrGrid->v_fLR[k]; - } + p = hSbrGrid->p; + if (p == 0) { + hSbrFrameInfo->shortEnv = 0; + } else { + hSbrFrameInfo->shortEnv = nEnv + 1 - p; + } - if (nEnv == 1) { - hSbrFrameInfo->nNoiseEnvelopes = 1; - hSbrFrameInfo->bordersNoise[0] = hSbrGrid->bs_abs_bord_0; - hSbrFrameInfo->bordersNoise[1] = hSbrGrid->bs_abs_bord_1; - } else { - hSbrFrameInfo->nNoiseEnvelopes = 2; - hSbrFrameInfo->bordersNoise[0] = hSbrGrid->bs_abs_bord_0; + for (k = 0; k < nEnv; k++) { + hSbrFrameInfo->freqRes[k] = (FREQ_RES)hSbrGrid->v_fLR[k]; + } - if (p == 0 || p == 1) { - hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[nEnv - 1]; + if (nEnv == 1) { + hSbrFrameInfo->nNoiseEnvelopes = 1; + hSbrFrameInfo->bordersNoise[0] = hSbrGrid->bs_abs_bord_0; + hSbrFrameInfo->bordersNoise[1] = hSbrGrid->bs_abs_bord_1; } else { - hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[hSbrFrameInfo->shortEnv]; + hSbrFrameInfo->nNoiseEnvelopes = 2; + hSbrFrameInfo->bordersNoise[0] = hSbrGrid->bs_abs_bord_0; + + if (p == 0 || p == 1) { + hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[nEnv - 1]; + } else { + hSbrFrameInfo->bordersNoise[1] = + hSbrFrameInfo->borders[hSbrFrameInfo->shortEnv]; + } + hSbrFrameInfo->bordersNoise[2] = hSbrGrid->bs_abs_bord_1; } - hSbrFrameInfo->bordersNoise[2] = hSbrGrid->bs_abs_bord_1; - } - break; + break; - default: - /* do nothing */ - break; + default: + /* do nothing */ + break; } if (frameClass == VARFIX || frameClass == FIXVAR) { @@ -2062,4 +1963,3 @@ ctrlSignal2FrameInfo ( } } } - diff --git a/libSBRenc/src/fram_gen.h b/libSBRenc/src/fram_gen.h index 00473d4..0c5edc3 100644 --- a/libSBRenc/src/fram_gen.h +++ b/libSBRenc/src/fram_gen.h @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,50 +90,64 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ /*! \file - \brief Framing generator prototypes and structs + \brief Framing generator prototypes and structs $Revision: 92790 $ */ -#ifndef _FRAM_GEN_H -#define _FRAM_GEN_H +#ifndef FRAM_GEN_H +#define FRAM_GEN_H #include "sbr_def.h" /* for MAX_ENVELOPES and MAX_NOISE_ENVELOPES in struct FRAME_INFO and CODEC_TYPE */ #include "sbr_encoder.h" /* for FREQ_RES */ -#define MAX_ENVELOPES_VARVAR MAX_ENVELOPES /*!< worst case number of envelopes in a VARVAR frame */ -#define MAX_ENVELOPES_FIXVAR_VARFIX 4 /*!< worst case number of envelopes in VARFIX and FIXVAR frames */ -#define MAX_NUM_REL 3 /*!< maximum number of relative borders in any VAR frame */ +#define MAX_ENVELOPES_VARVAR \ + MAX_ENVELOPES /*!< worst case number of envelopes in a VARVAR frame */ +#define MAX_ENVELOPES_FIXVAR_VARFIX \ + 4 /*!< worst case number of envelopes in VARFIX and FIXVAR frames */ +#define MAX_NUM_REL \ + 3 /*!< maximum number of relative borders in any VAR frame */ /* SBR frame class definitions */ typedef enum { - FIXFIX = 0, /*!< bs_frame_class: leading and trailing frame borders are fixed */ - FIXVAR, /*!< bs_frame_class: leading frame border is fixed, trailing frame border is variable */ - VARFIX, /*!< bs_frame_class: leading frame border is variable, trailing frame border is fixed */ - VARVAR /*!< bs_frame_class: leading and trailing frame borders are variable */ - ,FIXFIXonly /*!< bs_frame_class: leading border fixed (0), trailing border fixed (nrTimeSlots) and encased borders are dynamically derived from the tranPos */ -}FRAME_CLASS; - + FIXFIX = + 0, /*!< bs_frame_class: leading and trailing frame borders are fixed */ + FIXVAR, /*!< bs_frame_class: leading frame border is fixed, trailing frame + border is variable */ + VARFIX, /*!< bs_frame_class: leading frame border is variable, trailing frame + border is fixed */ + VARVAR /*!< bs_frame_class: leading and trailing frame borders are variable */ + , + FIXFIXonly /*!< bs_frame_class: leading border fixed (0), trailing border + fixed (nrTimeSlots) and encased borders are dynamically derived + from the tranPos */ +} FRAME_CLASS; /* helper constants */ -#define DC 4711 /*!< helper constant: don't care */ -#define EMPTY (-99) /*!< helper constant: empty */ - +#define DC 4711 /*!< helper constant: don't care */ +#define EMPTY (-99) /*!< helper constant: empty */ /* system constants: AAC+SBR, DRM Frame-Length */ -#define FRAME_MIDDLE_SLOT_1920 4 -#define NUMBER_TIME_SLOTS_1920 15 - -#define LD_PRETRAN_OFF 3 -#define FRAME_MIDDLE_SLOT_512LD 4 -#define NUMBER_TIME_SLOTS_512LD 8 -#define TRANSIENT_OFFSET_LD 0 - +#define FRAME_MIDDLE_SLOT_1920 4 +#define NUMBER_TIME_SLOTS_1920 15 +#define LD_PRETRAN_OFF 3 +#define FRAME_MIDDLE_SLOT_512LD 4 +#define NUMBER_TIME_SLOTS_512LD 8 +#define TRANSIENT_OFFSET_LD 0 /* -system constants: AAC+SBR or aacPRO (hybrid format), Standard Frame-Length, Multi-Rate +system constants: AAC+SBR or aacPRO (hybrid format), Standard Frame-Length, +Multi-Rate --------------------------------------------------------------------------- Number of slots (numberTimeSlots): 16 (NUMBER_TIME_SLOTS_2048) Detector-offset (frameMiddleSlot): 4 @@ -141,12 +166,11 @@ Buffer-offset : 8 (BUFFER_FRAME_START_2048 = 0) |-|-|-|-|-|-|-|-B-|-|-|-|-|-|-|-|-|-|-|-|-|-|-|-B-|-|-|-|-|-|-|-|-|-|-|-|-|-|-|-| -frame-generator:0 16 24 32 -analysis-buffer:8 24 32 40 +frame-generator:0 16 24 32 +analysis-buffer:8 24 32 40 */ -#define FRAME_MIDDLE_SLOT_2048 4 -#define NUMBER_TIME_SLOTS_2048 16 - +#define FRAME_MIDDLE_SLOT_2048 4 +#define NUMBER_TIME_SLOTS_2048 16 /* system constants: mp3PRO, Multi-Rate & Single-Rate @@ -171,14 +195,12 @@ Buffer-offset : 4.5 (BUFFER_FRAME_START_1152 = 0) frame-generator: 0 9 13 18 analysis-buffer: 4.5 13.5 22.5 */ -#define FRAME_MIDDLE_SLOT_1152 4 -#define NUMBER_TIME_SLOTS_1152 9 - +#define FRAME_MIDDLE_SLOT_1152 4 +#define NUMBER_TIME_SLOTS_1152 9 /* system constants: Layer2+SBR */ -#define FRAME_MIDDLE_SLOT_2304 8 -#define NUMBER_TIME_SLOTS_2304 18 - +#define FRAME_MIDDLE_SLOT_2304 8 +#define NUMBER_TIME_SLOTS_2304 18 /*! \struct SBR_GRID @@ -187,123 +209,135 @@ analysis-buffer: 4.5 13.5 22.5 The variables hold the signals (e.g. lengths and numbers) in "clear text" */ -typedef struct -{ +typedef struct { /* system constants */ - INT bufferFrameStart; /*!< frame generator vs analysis buffer time alignment (currently set to 0, offset added elsewhere) */ - INT numberTimeSlots; /*!< number of SBR timeslots per frame */ + INT bufferFrameStart; /*!< frame generator vs analysis buffer time alignment + (currently set to 0, offset added elsewhere) */ + INT numberTimeSlots; /*!< number of SBR timeslots per frame */ /* will be adjusted for every frame */ - FRAME_CLASS frameClass; /*!< SBR frame class */ - INT bs_num_env; /*!< bs_num_env, number of envelopes for FIXFIX */ - INT bs_abs_bord; /*!< bs_abs_bord, absolute border for VARFIX and FIXVAR */ - INT n; /*!< number of relative borders for VARFIX and FIXVAR */ - INT p; /*!< pointer-to-transient-border */ - INT bs_rel_bord[MAX_NUM_REL]; /*!< bs_rel_bord, relative borders for all VAR */ - INT v_f[MAX_ENVELOPES_FIXVAR_VARFIX]; /*!< envelope frequency resolutions for FIXVAR and VARFIX */ - - INT bs_abs_bord_0; /*!< bs_abs_bord_0, leading absolute border for VARVAR */ - INT bs_abs_bord_1; /*!< bs_abs_bord_1, trailing absolute border for VARVAR */ - INT bs_num_rel_0; /*!< bs_num_rel_0, number of relative borders associated with leading absolute border for VARVAR */ - INT bs_num_rel_1; /*!< bs_num_rel_1, number of relative borders associated with trailing absolute border for VARVAR */ - INT bs_rel_bord_0[MAX_NUM_REL]; /*!< bs_rel_bord_0, relative borders associated with leading absolute border for VARVAR */ - INT bs_rel_bord_1[MAX_NUM_REL]; /*!< bs_rel_bord_1, relative borders associated with trailing absolute border for VARVAR */ - INT v_fLR[MAX_ENVELOPES_VARVAR]; /*!< envelope frequency resolutions for VARVAR */ - -} -SBR_GRID; + FRAME_CLASS frameClass; /*!< SBR frame class */ + INT bs_num_env; /*!< bs_num_env, number of envelopes for FIXFIX */ + INT bs_abs_bord; /*!< bs_abs_bord, absolute border for VARFIX and FIXVAR */ + INT n; /*!< number of relative borders for VARFIX and FIXVAR */ + INT p; /*!< pointer-to-transient-border */ + INT bs_rel_bord[MAX_NUM_REL]; /*!< bs_rel_bord, relative borders for all VAR + */ + INT v_f[MAX_ENVELOPES_FIXVAR_VARFIX]; /*!< envelope frequency resolutions for + FIXVAR and VARFIX */ + + INT bs_abs_bord_0; /*!< bs_abs_bord_0, leading absolute border for VARVAR */ + INT bs_abs_bord_1; /*!< bs_abs_bord_1, trailing absolute border for VARVAR */ + INT bs_num_rel_0; /*!< bs_num_rel_0, number of relative borders associated + with leading absolute border for VARVAR */ + INT bs_num_rel_1; /*!< bs_num_rel_1, number of relative borders associated + with trailing absolute border for VARVAR */ + INT bs_rel_bord_0[MAX_NUM_REL]; /*!< bs_rel_bord_0, relative borders + associated with leading absolute border + for VARVAR */ + INT bs_rel_bord_1[MAX_NUM_REL]; /*!< bs_rel_bord_1, relative borders + associated with trailing absolute border + for VARVAR */ + INT v_fLR[MAX_ENVELOPES_VARVAR]; /*!< envelope frequency resolutions for + VARVAR */ + +} SBR_GRID; typedef SBR_GRID *HANDLE_SBR_GRID; - - /*! \struct SBR_FRAME_INFO \brief time/frequency grid description for one frame */ -typedef struct -{ - INT nEnvelopes; /*!< number of envelopes */ - INT borders[MAX_ENVELOPES+1]; /*!< envelope borders in SBR timeslots */ - FREQ_RES freqRes[MAX_ENVELOPES]; /*!< frequency resolution of each envelope */ - INT shortEnv; /*!< number of an envelope to be shortened (starting at 1) or 0 for no shortened envelope */ - INT nNoiseEnvelopes; /*!< number of noise floors */ - INT bordersNoise[MAX_NOISE_ENVELOPES+1];/*!< noise floor borders in SBR timeslots */ -} -SBR_FRAME_INFO; -/* WARNING: When rearranging the elements of this struct keep in mind that the static - * initializations in the corresponding C-file have to be rearranged as well! - * snd 2002/01/23 +typedef struct { + INT nEnvelopes; /*!< number of envelopes */ + INT borders[MAX_ENVELOPES + 1]; /*!< envelope borders in SBR timeslots */ + FREQ_RES freqRes[MAX_ENVELOPES]; /*!< frequency resolution of each envelope */ + INT shortEnv; /*!< number of an envelope to be shortened (starting at 1) or 0 + for no shortened envelope */ + INT nNoiseEnvelopes; /*!< number of noise floors */ + INT bordersNoise[MAX_NOISE_ENVELOPES + + 1]; /*!< noise floor borders in SBR timeslots */ +} SBR_FRAME_INFO; +/* WARNING: When rearranging the elements of this struct keep in mind that the + * static initializations in the corresponding C-file have to be rearranged as + * well! snd 2002/01/23 */ typedef SBR_FRAME_INFO *HANDLE_SBR_FRAME_INFO; - /*! \struct SBR_ENVELOPE_FRAME \brief frame generator main struct - Contains tuning parameters, time/frequency grid description, sbr_grid() bitstream elements, and generator internal signals + Contains tuning parameters, time/frequency grid description, sbr_grid() + bitstream elements, and generator internal signals */ -typedef struct -{ +typedef struct { /* system constants */ - INT frameMiddleSlot; /*!< transient detector offset in SBR timeslots */ + INT frameMiddleSlot; /*!< transient detector offset in SBR timeslots */ /* basic tuning parameters */ - INT staticFraming; /*!< 1: run static framing in time, i.e. exclusive use of bs_frame_class = FIXFIX */ - INT numEnvStatic; /*!< number of envelopes per frame for static framing */ - FREQ_RES freq_res_fixfix[2]; /*!< envelope frequency resolution to use for bs_frame_class = FIXFIX; single env and split */ - UCHAR fResTransIsLow; /*!< frequency resolution for transient frames - always low (0) or according to table (1) */ + INT staticFraming; /*!< 1: run static framing in time, i.e. exclusive use of + bs_frame_class = FIXFIX */ + INT numEnvStatic; /*!< number of envelopes per frame for static framing */ + FREQ_RES + freq_res_fixfix[2]; /*!< envelope frequency resolution to use for + bs_frame_class = FIXFIX; single env and split */ + UCHAR + fResTransIsLow; /*!< frequency resolution for transient frames - always + low (0) or according to table (1) */ /* expert tuning parameters */ - const int *v_tuningSegm; /*!< segment lengths to use around transient */ - const int *v_tuningFreq; /*!< frequency resolutions to use around transient */ - INT dmin; /*!< minimum length of dependent segments */ - INT dmax; /*!< maximum length of dependent segments */ - INT allowSpread; /*!< 1: allow isolated transient to influence grid of 3 consecutive frames */ + const int *v_tuningSegm; /*!< segment lengths to use around transient */ + const int *v_tuningFreq; /*!< frequency resolutions to use around transient */ + INT dmin; /*!< minimum length of dependent segments */ + INT dmax; /*!< maximum length of dependent segments */ + INT allowSpread; /*!< 1: allow isolated transient to influence grid of 3 + consecutive frames */ /* internally used signals */ - FRAME_CLASS frameClassOld; /*!< frame class used for previous frame */ - INT spreadFlag; /*!< 1: use VARVAR instead of VARFIX to follow up old transient */ - - INT v_bord[2 * MAX_ENVELOPES_VARVAR + 1]; /*!< borders for current frame and preliminary borders for next frame (fixed borders excluded) */ - INT length_v_bord; /*!< helper variable: length of v_bord */ - INT v_freq[2 * MAX_ENVELOPES_VARVAR + 1]; /*!< frequency resolutions for current frame and preliminary resolutions for next frame */ - INT length_v_freq; /*!< helper variable: length of v_freq */ - - INT v_bordFollow[MAX_ENVELOPES_VARVAR]; /*!< preliminary borders for current frame (calculated during previous frame) */ - INT length_v_bordFollow; /*!< helper variable: length of v_bordFollow */ - INT i_tranFollow; /*!< points to transient border in v_bordFollow (may be negative, see keepForFollowUp()) */ - INT i_fillFollow; /*!< points to first fill border in v_bordFollow */ - INT v_freqFollow[MAX_ENVELOPES_VARVAR]; /*!< preliminary frequency resolutions for current frame (calculated during previous frame) */ - INT length_v_freqFollow; /*!< helper variable: length of v_freqFollow */ - + FRAME_CLASS frameClassOld; /*!< frame class used for previous frame */ + INT spreadFlag; /*!< 1: use VARVAR instead of VARFIX to follow up old + transient */ + + INT v_bord[2 * MAX_ENVELOPES_VARVAR + 1]; /*!< borders for current frame and + preliminary borders for next + frame (fixed borders excluded) */ + INT length_v_bord; /*!< helper variable: length of v_bord */ + INT v_freq[2 * MAX_ENVELOPES_VARVAR + 1]; /*!< frequency resolutions for + current frame and preliminary + resolutions for next frame */ + INT length_v_freq; /*!< helper variable: length of v_freq */ + + INT v_bordFollow[MAX_ENVELOPES_VARVAR]; /*!< preliminary borders for current + frame (calculated during previous + frame) */ + INT length_v_bordFollow; /*!< helper variable: length of v_bordFollow */ + INT i_tranFollow; /*!< points to transient border in v_bordFollow (may be + negative, see keepForFollowUp()) */ + INT i_fillFollow; /*!< points to first fill border in v_bordFollow */ + INT v_freqFollow[MAX_ENVELOPES_VARVAR]; /*!< preliminary frequency resolutions + for current frame (calculated + during previous frame) */ + INT length_v_freqFollow; /*!< helper variable: length of v_freqFollow */ /* externally needed signals */ - SBR_GRID SbrGrid; /*!< sbr_grid() signals to be converted to bitstream elements */ - SBR_FRAME_INFO SbrFrameInfo; /*!< time/frequency grid description for one frame */ -} -SBR_ENVELOPE_FRAME; + SBR_GRID + SbrGrid; /*!< sbr_grid() signals to be converted to bitstream elements */ + SBR_FRAME_INFO + SbrFrameInfo; /*!< time/frequency grid description for one frame */ +} SBR_ENVELOPE_FRAME; typedef SBR_ENVELOPE_FRAME *HANDLE_SBR_ENVELOPE_FRAME; - - -void -FDKsbrEnc_initFrameInfoGenerator ( - HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame, - INT allowSpread, - INT numEnvStatic, - INT staticFraming, - INT timeSlots, - const FREQ_RES* freq_res_fixfix - ,UCHAR fResTransIsLow, - INT ldGrid - ); +void FDKsbrEnc_initFrameInfoGenerator(HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame, + INT allowSpread, INT numEnvStatic, + INT staticFraming, INT timeSlots, + const FREQ_RES *freq_res_fixfix, + UCHAR fResTransIsLow, INT ldGrid); HANDLE_SBR_FRAME_INFO -FDKsbrEnc_frameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame, - UCHAR *v_transient_info, - UCHAR *v_transient_info_pre, - int ldGrid, - const int *v_tuning); +FDKsbrEnc_frameInfoGenerator(HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame, + UCHAR *v_transient_info, const INT rightBorderFIX, + UCHAR *v_transient_info_pre, int ldGrid, + const int *v_tuning); #endif diff --git a/libSBRenc/src/invf_est.cpp b/libSBRenc/src/invf_est.cpp index 32df6d9..53b47ac 100644 --- a/libSBRenc/src/invf_est.cpp +++ b/libSBRenc/src/invf_est.cpp @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,7 +90,15 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ #include "invf_est.h" #include "sbr_misc.h" @@ -87,46 +106,66 @@ amm-info@iis.fraunhofer.de #include "genericStds.h" #define MAX_NUM_REGIONS 10 -#define SCALE_FAC_QUO 512.0f -#define SCALE_FAC_NRG 256.0f +#define SCALE_FAC_QUO 512.0f +#define SCALE_FAC_NRG 256.0f #ifndef min -#define min(a,b) ( a < b ? a:b) +#define min(a, b) (a < b ? a : b) #endif #ifndef max -#define max(a,b) ( a > b ? a:b) +#define max(a, b) (a > b ? a : b) #endif -static const FIXP_DBL quantStepsSbr[4] = { 0x00400000, 0x02800000, 0x03800000, 0x04c00000 } ; /* table scaled with SCALE_FAC_QUO */ -static const FIXP_DBL quantStepsOrig[4] = { 0x00000000, 0x00c00000, 0x01c00000, 0x02800000 } ; /* table scaled with SCALE_FAC_QUO */ -static const FIXP_DBL nrgBorders[4] = { 0x0c800000, 0x0f000000, 0x11800000, 0x14000000 } ; /* table scaled with SCALE_FAC_NRG */ +static const FIXP_DBL quantStepsSbr[4] = { + 0x00400000, 0x02800000, 0x03800000, + 0x04c00000}; /* table scaled with SCALE_FAC_QUO */ +static const FIXP_DBL quantStepsOrig[4] = { + 0x00000000, 0x00c00000, 0x01c00000, + 0x02800000}; /* table scaled with SCALE_FAC_QUO */ +static const FIXP_DBL nrgBorders[4] = { + 0x0c800000, 0x0f000000, 0x11800000, + 0x14000000}; /* table scaled with SCALE_FAC_NRG */ static const DETECTOR_PARAMETERS detectorParamsAAC = { quantStepsSbr, quantStepsOrig, nrgBorders, - 4, /* Number of borders SBR. */ - 4, /* Number of borders orig. */ - 4, /* Number of borders Nrg. */ - { /* Region space. */ - {INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF, INVF_OFF}, /* | */ - {INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF, INVF_OFF}, /* | */ - {INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF}, /* regionSbr */ - {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF}, /* | */ - {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF} /* | */ - },/*------------------------ regionOrig ---------------------------------*/ - { /* Region space transient. */ - {INVF_LOW_LEVEL, INVF_LOW_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF}, /* | */ - {INVF_LOW_LEVEL, INVF_LOW_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF}, /* | */ - {INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF}, /* regionSbr */ - {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF}, /* | */ - {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF} /* | */ - },/*------------------------ regionOrig ---------------------------------*/ - {-4, -3, -2, -1, 0} /* Reduction factor of the inverse filtering for low energies.*/ + 4, /* Number of borders SBR. */ + 4, /* Number of borders orig. */ + 4, /* Number of borders Nrg. */ + { + /* Region space. */ + {INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF, + INVF_OFF}, /* | */ + {INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF, + INVF_OFF}, /* | */ + {INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, + INVF_OFF}, /* regionSbr */ + {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, + INVF_OFF}, /* | */ + {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, + INVF_OFF} /* | */ + }, /*------------------------ regionOrig ---------------------------------*/ + { + /* Region space transient. */ + {INVF_LOW_LEVEL, INVF_LOW_LEVEL, INVF_LOW_LEVEL, INVF_OFF, + INVF_OFF}, /* | */ + {INVF_LOW_LEVEL, INVF_LOW_LEVEL, INVF_LOW_LEVEL, INVF_OFF, + INVF_OFF}, /* | */ + {INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_OFF, + INVF_OFF}, /* regionSbr */ + {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, + INVF_OFF}, /* | */ + {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, + INVF_OFF} /* | */ + }, /*------------------------ regionOrig ---------------------------------*/ + {-4, -3, -2, -1, + 0} /* Reduction factor of the inverse filtering for low energies.*/ }; -static const FIXP_DBL hysteresis = 0x00400000 ; /* Delta value for hysteresis. scaled with SCALE_FAC_QUO */ +static const FIXP_DBL hysteresis = + 0x00400000; /* Delta value for hysteresis. scaled with SCALE_FAC_QUO */ /* * AAC+SBR PARAMETERS for Speech @@ -135,24 +174,37 @@ static const DETECTOR_PARAMETERS detectorParamsAACSpeech = { quantStepsSbr, quantStepsOrig, nrgBorders, - 4, /* Number of borders SBR. */ - 4, /* Number of borders orig. */ - 4, /* Number of borders Nrg. */ - { /* Region space. */ - {INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF}, /* | */ - {INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF}, /* | */ - {INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF}, /* regionSbr */ - {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF}, /* | */ - {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF} /* | */ - },/*------------------------ regionOrig ---------------------------------*/ - { /* Region space transient. */ - {INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF}, /* | */ - {INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, INVF_OFF}, /* | */ - {INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF}, /* regionSbr */ - {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF}, /* | */ - {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, INVF_OFF} /* | */ - },/*------------------------ regionOrig ---------------------------------*/ - {-4, -3, -2, -1, 0} /* Reduction factor of the inverse filtering for low energies.*/ + 4, /* Number of borders SBR. */ + 4, /* Number of borders orig. */ + 4, /* Number of borders Nrg. */ + { + /* Region space. */ + {INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, + INVF_OFF}, /* | */ + {INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, + INVF_OFF}, /* | */ + {INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_OFF, + INVF_OFF}, /* regionSbr */ + {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, + INVF_OFF}, /* | */ + {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, + INVF_OFF} /* | */ + }, /*------------------------ regionOrig ---------------------------------*/ + { + /* Region space transient. */ + {INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, + INVF_OFF}, /* | */ + {INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_LOW_LEVEL, INVF_OFF, + INVF_OFF}, /* | */ + {INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_MID_LEVEL, INVF_OFF, + INVF_OFF}, /* regionSbr */ + {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, + INVF_OFF}, /* | */ + {INVF_HIGH_LEVEL, INVF_HIGH_LEVEL, INVF_MID_LEVEL, INVF_OFF, + INVF_OFF} /* | */ + }, /*------------------------ regionOrig ---------------------------------*/ + {-4, -3, -2, -1, + 0} /* Reduction factor of the inverse filtering for low energies.*/ }; /* @@ -160,20 +212,19 @@ static const DETECTOR_PARAMETERS detectorParamsAACSpeech = { ************************/ typedef const FIXP_DBL FIR_FILTER[5]; -static const FIR_FILTER fir_0 = { 0x7fffffff, 0x00000000, 0x00000000, 0x00000000, 0x00000000 } ; -static const FIR_FILTER fir_1 = { 0x2aaaaa80, 0x555554ff, 0x00000000, 0x00000000, 0x00000000 } ; -static const FIR_FILTER fir_2 = { 0x10000000, 0x30000000, 0x40000000, 0x00000000, 0x00000000 } ; -static const FIR_FILTER fir_3 = { 0x077f80e8, 0x199999a0, 0x2bb3b240, 0x33333340, 0x00000000 } ; -static const FIR_FILTER fir_4 = { 0x04130598, 0x0ebdb000, 0x1becfa60, 0x2697a4c0, 0x2aaaaa80 } ; - +static const FIR_FILTER fir_0 = {0x7fffffff, 0x00000000, 0x00000000, 0x00000000, + 0x00000000}; +static const FIR_FILTER fir_1 = {0x2aaaaa80, 0x555554ff, 0x00000000, 0x00000000, + 0x00000000}; +static const FIR_FILTER fir_2 = {0x10000000, 0x30000000, 0x40000000, 0x00000000, + 0x00000000}; +static const FIR_FILTER fir_3 = {0x077f80e8, 0x199999a0, 0x2bb3b240, 0x33333340, + 0x00000000}; +static const FIR_FILTER fir_4 = {0x04130598, 0x0ebdb000, 0x1becfa60, 0x2697a4c0, + 0x2aaaaa80}; -static const FIR_FILTER *const fir_table[5] = { - &fir_0, - &fir_1, - &fir_2, - &fir_3, - &fir_4 -}; +static const FIR_FILTER *const fir_table[5] = {&fir_0, &fir_1, &fir_2, &fir_3, + &fir_4}; /**************************************************************************/ /*! @@ -184,98 +235,111 @@ static const FIR_FILTER *const fir_table[5] = { */ /**************************************************************************/ -static void -calculateDetectorValues(FIXP_DBL **quotaMatrixOrig, /*!< Matrix holding the tonality values of the original. */ - SCHAR *indexVector, /*!< Index vector to obtain the patched data. */ - FIXP_DBL *nrgVector, /*!< Energy vector. */ - DETECTOR_VALUES *detectorValues, /*!< pointer to DETECTOR_VALUES struct. */ - INT startChannel, /*!< Start channel. */ - INT stopChannel, /*!< Stop channel. */ - INT startIndex, /*!< Start index. */ - INT stopIndex, /*!< Stop index. */ - INT numberOfStrongest /*!< The number of sorted tonal components to be considered. */ - ) -{ - INT i,temp, j; - - const FIXP_DBL* filter = *fir_table[INVF_SMOOTHING_LENGTH]; +static void calculateDetectorValues( + FIXP_DBL **quotaMatrixOrig, /*!< Matrix holding the tonality values of the + original. */ + SCHAR *indexVector, /*!< Index vector to obtain the patched data. */ + FIXP_DBL *nrgVector, /*!< Energy vector. */ + DETECTOR_VALUES *detectorValues, /*!< pointer to DETECTOR_VALUES struct. */ + INT startChannel, /*!< Start channel. */ + INT stopChannel, /*!< Stop channel. */ + INT startIndex, /*!< Start index. */ + INT stopIndex, /*!< Stop index. */ + INT numberOfStrongest /*!< The number of sorted tonal components to be + considered. */ +) { + INT i, temp, j; + + const FIXP_DBL *filter = *fir_table[INVF_SMOOTHING_LENGTH]; FIXP_DBL origQuotaMeanStrongest, sbrQuotaMeanStrongest; FIXP_DBL origQuota, sbrQuota; FIXP_DBL invIndex, invChannel, invTemp; FIXP_DBL quotaVecOrig[64], quotaVecSbr[64]; - FDKmemclear(quotaVecOrig,64*sizeof(FIXP_DBL)); - FDKmemclear(quotaVecSbr,64*sizeof(FIXP_DBL)); + FDKmemclear(quotaVecOrig, 64 * sizeof(FIXP_DBL)); + FDKmemclear(quotaVecSbr, 64 * sizeof(FIXP_DBL)); - invIndex = GetInvInt(stopIndex-startIndex); - invChannel = GetInvInt(stopChannel-startChannel); + invIndex = GetInvInt(stopIndex - startIndex); + invChannel = GetInvInt(stopChannel - startChannel); /* - Calculate the mean value, over the current time segment, for the original, the HFR - and the difference, over all channels in the current frequency range. + Calculate the mean value, over the current time segment, for the original, + the HFR and the difference, over all channels in the current frequency range. NOTE: the averaging is done on the values quota/(1 - quota + RELAXATION). */ /* The original, the sbr signal and the total energy */ detectorValues->avgNrg = FL2FXCONST_DBL(0.0f); - for(j=startIndex; j<stopIndex; j++) { - for(i=startChannel; i<stopChannel; i++) { + for (j = startIndex; j < stopIndex; j++) { + for (i = startChannel; i < stopChannel; i++) { quotaVecOrig[i] += fMult(quotaMatrixOrig[j][i], invIndex); - if(indexVector[i] != -1) + if (indexVector[i] != -1) quotaVecSbr[i] += fMult(quotaMatrixOrig[j][indexVector[i]], invIndex); } detectorValues->avgNrg += fMult(nrgVector[j], invIndex); } /* - Calculate the mean value, over the current frequency range, for the original, the HFR - and the difference. Also calculate the same mean values for the three vectors, but only - includeing the x strongest copmponents. + Calculate the mean value, over the current frequency range, for the original, + the HFR and the difference. Also calculate the same mean values for the three + vectors, but only includeing the x strongest copmponents. */ origQuota = FL2FXCONST_DBL(0.0f); - sbrQuota = FL2FXCONST_DBL(0.0f); - for(i=startChannel; i<stopChannel; i++) { + sbrQuota = FL2FXCONST_DBL(0.0f); + for (i = startChannel; i < stopChannel; i++) { origQuota += fMultDiv2(quotaVecOrig[i], invChannel); - sbrQuota += fMultDiv2(quotaVecSbr[i], invChannel); + sbrQuota += fMultDiv2(quotaVecSbr[i], invChannel); } /* Calculate the mean value for the x strongest components */ - FDKsbrEnc_Shellsort_fract(quotaVecOrig+startChannel,stopChannel-startChannel); - FDKsbrEnc_Shellsort_fract(quotaVecSbr+startChannel,stopChannel-startChannel); + FDKsbrEnc_Shellsort_fract(quotaVecOrig + startChannel, + stopChannel - startChannel); + FDKsbrEnc_Shellsort_fract(quotaVecSbr + startChannel, + stopChannel - startChannel); origQuotaMeanStrongest = FL2FXCONST_DBL(0.0f); - sbrQuotaMeanStrongest = FL2FXCONST_DBL(0.0f); + sbrQuotaMeanStrongest = FL2FXCONST_DBL(0.0f); temp = min(stopChannel - startChannel, numberOfStrongest); invTemp = GetInvInt(temp); - for(i=0; i<temp; i++) { - origQuotaMeanStrongest += fMultDiv2(quotaVecOrig[i + stopChannel - temp], invTemp); - sbrQuotaMeanStrongest += fMultDiv2(quotaVecSbr[i + stopChannel - temp], invTemp); + for (i = 0; i < temp; i++) { + origQuotaMeanStrongest += + fMultDiv2(quotaVecOrig[i + stopChannel - temp], invTemp); + sbrQuotaMeanStrongest += + fMultDiv2(quotaVecSbr[i + stopChannel - temp], invTemp); } /* The value for the strongest component */ detectorValues->origQuotaMax = quotaVecOrig[stopChannel - 1]; - detectorValues->sbrQuotaMax = quotaVecSbr[stopChannel - 1]; + detectorValues->sbrQuotaMax = quotaVecSbr[stopChannel - 1]; /* Buffer values */ - FDKmemmove(detectorValues->origQuotaMean, detectorValues->origQuotaMean + 1, INVF_SMOOTHING_LENGTH*sizeof(FIXP_DBL)); - FDKmemmove(detectorValues->sbrQuotaMean, detectorValues->sbrQuotaMean + 1, INVF_SMOOTHING_LENGTH*sizeof(FIXP_DBL)); - FDKmemmove(detectorValues->origQuotaMeanStrongest, detectorValues->origQuotaMeanStrongest + 1, INVF_SMOOTHING_LENGTH*sizeof(FIXP_DBL)); - FDKmemmove(detectorValues->sbrQuotaMeanStrongest, detectorValues->sbrQuotaMeanStrongest + 1, INVF_SMOOTHING_LENGTH*sizeof(FIXP_DBL)); - - detectorValues->origQuotaMean[INVF_SMOOTHING_LENGTH] = origQuota<<1; - detectorValues->sbrQuotaMean[INVF_SMOOTHING_LENGTH] = sbrQuota<<1; - detectorValues->origQuotaMeanStrongest[INVF_SMOOTHING_LENGTH] = origQuotaMeanStrongest<<1; - detectorValues->sbrQuotaMeanStrongest[INVF_SMOOTHING_LENGTH] = sbrQuotaMeanStrongest<<1; + FDKmemmove(detectorValues->origQuotaMean, detectorValues->origQuotaMean + 1, + INVF_SMOOTHING_LENGTH * sizeof(FIXP_DBL)); + FDKmemmove(detectorValues->sbrQuotaMean, detectorValues->sbrQuotaMean + 1, + INVF_SMOOTHING_LENGTH * sizeof(FIXP_DBL)); + FDKmemmove(detectorValues->origQuotaMeanStrongest, + detectorValues->origQuotaMeanStrongest + 1, + INVF_SMOOTHING_LENGTH * sizeof(FIXP_DBL)); + FDKmemmove(detectorValues->sbrQuotaMeanStrongest, + detectorValues->sbrQuotaMeanStrongest + 1, + INVF_SMOOTHING_LENGTH * sizeof(FIXP_DBL)); + + detectorValues->origQuotaMean[INVF_SMOOTHING_LENGTH] = origQuota << 1; + detectorValues->sbrQuotaMean[INVF_SMOOTHING_LENGTH] = sbrQuota << 1; + detectorValues->origQuotaMeanStrongest[INVF_SMOOTHING_LENGTH] = + origQuotaMeanStrongest << 1; + detectorValues->sbrQuotaMeanStrongest[INVF_SMOOTHING_LENGTH] = + sbrQuotaMeanStrongest << 1; /* Filter values @@ -285,11 +349,15 @@ calculateDetectorValues(FIXP_DBL **quotaMatrixOrig, /*!< Matrix holding th detectorValues->origQuotaMeanStrongestFilt = FL2FXCONST_DBL(0.0f); detectorValues->sbrQuotaMeanStrongestFilt = FL2FXCONST_DBL(0.0f); - for(i=0;i<INVF_SMOOTHING_LENGTH+1;i++) { - detectorValues->origQuotaMeanFilt += fMult(detectorValues->origQuotaMean[i], filter[i]); - detectorValues->sbrQuotaMeanFilt += fMult(detectorValues->sbrQuotaMean[i], filter[i]); - detectorValues->origQuotaMeanStrongestFilt += fMult(detectorValues->origQuotaMeanStrongest[i], filter[i]); - detectorValues->sbrQuotaMeanStrongestFilt += fMult(detectorValues->sbrQuotaMeanStrongest[i], filter[i]); + for (i = 0; i < INVF_SMOOTHING_LENGTH + 1; i++) { + detectorValues->origQuotaMeanFilt += + fMult(detectorValues->origQuotaMean[i], filter[i]); + detectorValues->sbrQuotaMeanFilt += + fMult(detectorValues->sbrQuotaMean[i], filter[i]); + detectorValues->origQuotaMeanStrongestFilt += + fMult(detectorValues->origQuotaMeanStrongest[i], filter[i]); + detectorValues->sbrQuotaMeanStrongestFilt += + fMult(detectorValues->sbrQuotaMeanStrongest[i], filter[i]); } } @@ -303,29 +371,28 @@ calculateDetectorValues(FIXP_DBL **quotaMatrixOrig, /*!< Matrix holding th */ /**************************************************************************/ -static INT -findRegion(FIXP_DBL currVal, /*!< The current value. */ - const FIXP_DBL *borders, /*!< The border of the regions. */ - const INT numBorders /*!< The number of borders. */ - ) -{ +static INT findRegion( + FIXP_DBL currVal, /*!< The current value. */ + const FIXP_DBL *borders, /*!< The border of the regions. */ + const INT numBorders /*!< The number of borders. */ +) { INT i; - if(currVal < borders[0]){ + if (currVal < borders[0]) { return 0; } - for(i = 1; i < numBorders; i++){ - if( currVal >= borders[i-1] && currVal < borders[i]){ + for (i = 1; i < numBorders; i++) { + if (currVal >= borders[i - 1] && currVal < borders[i]) { return i; } } - if(currVal >= borders[numBorders-1]){ + if (currVal >= borders[numBorders - 1]) { return numBorders; } - return 0; /* We never get here, it's just to avoid compiler warnings.*/ + return 0; /* We never get here, it's just to avoid compiler warnings.*/ } /**************************************************************************/ @@ -337,25 +404,22 @@ findRegion(FIXP_DBL currVal, /*!< The current value. */ */ /**************************************************************************/ -static INVF_MODE -decisionAlgorithm(const DETECTOR_PARAMETERS *detectorParams, /*!< Struct with the detector parameters. */ - DETECTOR_VALUES *detectorValues, /*!< Struct with the detector values. */ - INT transientFlag, /*!< Flag indicating if there is a transient present.*/ - INT* prevRegionSbr, /*!< The previous region in which the Sbr value was. */ - INT* prevRegionOrig /*!< The previous region in which the Orig value was. */ - ) -{ +static INVF_MODE decisionAlgorithm( + const DETECTOR_PARAMETERS + *detectorParams, /*!< Struct with the detector parameters. */ + DETECTOR_VALUES *detectorValues, /*!< Struct with the detector values. */ + INT transientFlag, /*!< Flag indicating if there is a transient present.*/ + INT *prevRegionSbr, /*!< The previous region in which the Sbr value was. */ + INT *prevRegionOrig /*!< The previous region in which the Orig value was. */ +) { INT invFiltLevel, regionSbr, regionOrig, regionNrg; /* Current thresholds. */ - const FIXP_DBL *quantStepsSbr = detectorParams->quantStepsSbr; - const FIXP_DBL *quantStepsOrig = detectorParams->quantStepsOrig; - const FIXP_DBL *nrgBorders = detectorParams->nrgBorders; - const INT numRegionsSbr = detectorParams->numRegionsSbr; - const INT numRegionsOrig = detectorParams->numRegionsOrig; - const INT numRegionsNrg = detectorParams->numRegionsNrg; + const INT numRegionsSbr = detectorParams->numRegionsSbr; + const INT numRegionsOrig = detectorParams->numRegionsOrig; + const INT numRegionsNrg = detectorParams->numRegionsNrg; FIXP_DBL quantStepsSbrTmp[MAX_NUM_REGIONS]; FIXP_DBL quantStepsOrigTmp[MAX_NUM_REGIONS]; @@ -367,40 +431,65 @@ decisionAlgorithm(const DETECTOR_PARAMETERS *detectorParams, /*!< Struct wit FIXP_DBL sbrQuotaMeanFilt; FIXP_DBL nrg; - /* 0.375 = 3.0 / 8.0; 0.31143075889 = log2(RELAXATION)/64.0; 0.625 = log(16)/64.0; 0.6875 = 44/64.0 */ - origQuotaMeanFilt = (fMultDiv2(FL2FXCONST_DBL(2.f*0.375f), (FIXP_DBL)(CalcLdData(max(detectorValues->origQuotaMeanFilt,(FIXP_DBL)1)) + FL2FXCONST_DBL(0.31143075889f)))) << 0; /* scaled by 1/2^9 */ - sbrQuotaMeanFilt = (fMultDiv2(FL2FXCONST_DBL(2.f*0.375f), (FIXP_DBL)(CalcLdData(max(detectorValues->sbrQuotaMeanFilt,(FIXP_DBL)1)) + FL2FXCONST_DBL(0.31143075889f)))) << 0; /* scaled by 1/2^9 */ - /* If energy is zero then we will get different results for different word lengths. */ - nrg = (fMultDiv2(FL2FXCONST_DBL(2.f*0.375f), (FIXP_DBL)(CalcLdData(detectorValues->avgNrg+(FIXP_DBL)1) + FL2FXCONST_DBL(0.0625f) + FL2FXCONST_DBL(0.6875f)))) << 0; /* scaled by 1/2^8; 2^44 -> qmf energy scale */ - - FDKmemcpy(quantStepsSbrTmp,quantStepsSbr,numRegionsSbr*sizeof(FIXP_DBL)); - FDKmemcpy(quantStepsOrigTmp,quantStepsOrig,numRegionsOrig*sizeof(FIXP_DBL)); - - if(*prevRegionSbr < numRegionsSbr) - quantStepsSbrTmp[*prevRegionSbr] = quantStepsSbr[*prevRegionSbr] + hysteresis; - if(*prevRegionSbr > 0) - quantStepsSbrTmp[*prevRegionSbr - 1] = quantStepsSbr[*prevRegionSbr - 1] - hysteresis; - - if(*prevRegionOrig < numRegionsOrig) - quantStepsOrigTmp[*prevRegionOrig] = quantStepsOrig[*prevRegionOrig] + hysteresis; - if(*prevRegionOrig > 0) - quantStepsOrigTmp[*prevRegionOrig - 1] = quantStepsOrig[*prevRegionOrig - 1] - hysteresis; - - regionSbr = findRegion(sbrQuotaMeanFilt, quantStepsSbrTmp, numRegionsSbr); + /* 0.375 = 3.0 / 8.0; 0.31143075889 = log2(RELAXATION)/64.0; 0.625 = + * log(16)/64.0; 0.6875 = 44/64.0 */ + origQuotaMeanFilt = + (fMultDiv2(FL2FXCONST_DBL(2.f * 0.375f), + (FIXP_DBL)(CalcLdData(max(detectorValues->origQuotaMeanFilt, + (FIXP_DBL)1)) + + FL2FXCONST_DBL(0.31143075889f)))) + << 0; /* scaled by 1/2^9 */ + sbrQuotaMeanFilt = + (fMultDiv2(FL2FXCONST_DBL(2.f * 0.375f), + (FIXP_DBL)(CalcLdData(max(detectorValues->sbrQuotaMeanFilt, + (FIXP_DBL)1)) + + FL2FXCONST_DBL(0.31143075889f)))) + << 0; /* scaled by 1/2^9 */ + /* If energy is zero then we will get different results for different word + * lengths. */ + nrg = + (fMultDiv2(FL2FXCONST_DBL(2.f * 0.375f), + (FIXP_DBL)(CalcLdData(detectorValues->avgNrg + (FIXP_DBL)1) + + FL2FXCONST_DBL(0.0625f) + FL2FXCONST_DBL(0.6875f)))) + << 0; /* scaled by 1/2^8; 2^44 -> qmf energy scale */ + + FDKmemcpy(quantStepsSbrTmp, detectorParams->quantStepsSbr, + numRegionsSbr * sizeof(FIXP_DBL)); + FDKmemcpy(quantStepsOrigTmp, detectorParams->quantStepsOrig, + numRegionsOrig * sizeof(FIXP_DBL)); + + if (*prevRegionSbr < numRegionsSbr) + quantStepsSbrTmp[*prevRegionSbr] = + detectorParams->quantStepsSbr[*prevRegionSbr] + hysteresis; + if (*prevRegionSbr > 0) + quantStepsSbrTmp[*prevRegionSbr - 1] = + detectorParams->quantStepsSbr[*prevRegionSbr - 1] - hysteresis; + + if (*prevRegionOrig < numRegionsOrig) + quantStepsOrigTmp[*prevRegionOrig] = + detectorParams->quantStepsOrig[*prevRegionOrig] + hysteresis; + if (*prevRegionOrig > 0) + quantStepsOrigTmp[*prevRegionOrig - 1] = + detectorParams->quantStepsOrig[*prevRegionOrig - 1] - hysteresis; + + regionSbr = findRegion(sbrQuotaMeanFilt, quantStepsSbrTmp, numRegionsSbr); regionOrig = findRegion(origQuotaMeanFilt, quantStepsOrigTmp, numRegionsOrig); - regionNrg = findRegion(nrg,nrgBorders,numRegionsNrg); + regionNrg = findRegion(nrg, detectorParams->nrgBorders, numRegionsNrg); *prevRegionSbr = regionSbr; *prevRegionOrig = regionOrig; /* Use different settings if a transient is present*/ - invFiltLevel = (transientFlag == 1) ? detectorParams->regionSpaceTransient[regionSbr][regionOrig] - : detectorParams->regionSpace[regionSbr][regionOrig]; + invFiltLevel = + (transientFlag == 1) + ? detectorParams->regionSpaceTransient[regionSbr][regionOrig] + : detectorParams->regionSpace[regionSbr][regionOrig]; /* Compensate for low energy.*/ - invFiltLevel = max(invFiltLevel + detectorParams->EnergyCompFactor[regionNrg],0); + invFiltLevel = + max(invFiltLevel + detectorParams->EnergyCompFactor[regionNrg], 0); - return (INVF_MODE) (invFiltLevel); + return (INVF_MODE)(invFiltLevel); } /**************************************************************************/ @@ -416,46 +505,38 @@ decisionAlgorithm(const DETECTOR_PARAMETERS *detectorParams, /*!< Struct wit */ /**************************************************************************/ -void -FDKsbrEnc_qmfInverseFilteringDetector(HANDLE_SBR_INV_FILT_EST hInvFilt, /*!< Handle to the SBR_INV_FILT_EST struct. */ - FIXP_DBL **quotaMatrix, /*!< The matrix holding the tonality values of the original. */ - FIXP_DBL *nrgVector, /*!< The energy vector. */ - SCHAR *indexVector, /*!< Index vector to obtain the patched data. */ - INT startIndex, /*!< Start index. */ - INT stopIndex, /*!< Stop index. */ - INT transientFlag, /*!< Flag indicating if a transient is present or not.*/ - INVF_MODE* infVec /*!< Vector holding the inverse filtering levels. */ - ) -{ +void FDKsbrEnc_qmfInverseFilteringDetector( + HANDLE_SBR_INV_FILT_EST + hInvFilt, /*!< Handle to the SBR_INV_FILT_EST struct. */ + FIXP_DBL **quotaMatrix, /*!< The matrix holding the tonality values of the + original. */ + FIXP_DBL *nrgVector, /*!< The energy vector. */ + SCHAR *indexVector, /*!< Index vector to obtain the patched data. */ + INT startIndex, /*!< Start index. */ + INT stopIndex, /*!< Stop index. */ + INT transientFlag, /*!< Flag indicating if a transient is present or not.*/ + INVF_MODE *infVec /*!< Vector holding the inverse filtering levels. */ +) { INT band; /* * Do the inverse filtering level estimation. *****************************************************/ - for(band = 0 ; band < hInvFilt->noDetectorBands; band++){ + for (band = 0; band < hInvFilt->noDetectorBands; band++) { INT startChannel = hInvFilt->freqBandTableInvFilt[band]; - INT stopChannel = hInvFilt->freqBandTableInvFilt[band+1]; - - - calculateDetectorValues( quotaMatrix, - indexVector, - nrgVector, - &hInvFilt->detectorValues[band], - startChannel, - stopChannel, - startIndex, - stopIndex, - hInvFilt->numberOfStrongest); - - infVec[band]= decisionAlgorithm( hInvFilt->detectorParams, - &hInvFilt->detectorValues[band], - transientFlag, - &hInvFilt->prevRegionSbr[band], - &hInvFilt->prevRegionOrig[band]); - } + INT stopChannel = hInvFilt->freqBandTableInvFilt[band + 1]; -} + calculateDetectorValues(quotaMatrix, indexVector, nrgVector, + &hInvFilt->detectorValues[band], startChannel, + stopChannel, startIndex, stopIndex, + hInvFilt->numberOfStrongest); + infVec[band] = decisionAlgorithm( + hInvFilt->detectorParams, &hInvFilt->detectorValues[band], + transientFlag, &hInvFilt->prevRegionSbr[band], + &hInvFilt->prevRegionOrig[band]); + } +} /**************************************************************************/ /*! @@ -466,43 +547,43 @@ FDKsbrEnc_qmfInverseFilteringDetector(HANDLE_SBR_INV_FILT_EST hInvFilt, /*!< Ha */ /**************************************************************************/ -INT -FDKsbrEnc_initInvFiltDetector (HANDLE_SBR_INV_FILT_EST hInvFilt, /*!< Pointer to a handle to the SBR_INV_FILT_EST struct. */ - INT* freqBandTableDetector, /*!< Frequency band table for the inverse filtering. */ - INT numDetectorBands, /*!< Number of inverse filtering bands. */ - UINT useSpeechConfig /*!< Flag: adapt tuning parameters according to speech*/ - ) -{ +INT FDKsbrEnc_initInvFiltDetector( + HANDLE_SBR_INV_FILT_EST + hInvFilt, /*!< Pointer to a handle to the SBR_INV_FILT_EST struct. */ + INT *freqBandTableDetector, /*!< Frequency band table for the inverse + filtering. */ + INT numDetectorBands, /*!< Number of inverse filtering bands. */ + UINT + useSpeechConfig /*!< Flag: adapt tuning parameters according to speech*/ +) { INT i; - FDKmemclear( hInvFilt,sizeof(SBR_INV_FILT_EST)); + FDKmemclear(hInvFilt, sizeof(SBR_INV_FILT_EST)); - hInvFilt->detectorParams = (useSpeechConfig) ? &detectorParamsAACSpeech - : &detectorParamsAAC ; + hInvFilt->detectorParams = + (useSpeechConfig) ? &detectorParamsAACSpeech : &detectorParamsAAC; hInvFilt->noDetectorBandsMax = numDetectorBands; /* Memory initialisation */ - for(i=0;i<hInvFilt->noDetectorBandsMax;i++){ + for (i = 0; i < hInvFilt->noDetectorBandsMax; i++) { FDKmemclear(&hInvFilt->detectorValues[i], sizeof(DETECTOR_VALUES)); - hInvFilt->prevInvfMode[i] = INVF_OFF; + hInvFilt->prevInvfMode[i] = INVF_OFF; hInvFilt->prevRegionOrig[i] = 0; - hInvFilt->prevRegionSbr[i] = 0; + hInvFilt->prevRegionSbr[i] = 0; } /* Reset the inverse fltering detector. */ - FDKsbrEnc_resetInvFiltDetector(hInvFilt, - freqBandTableDetector, - hInvFilt->noDetectorBandsMax); + FDKsbrEnc_resetInvFiltDetector(hInvFilt, freqBandTableDetector, + hInvFilt->noDetectorBandsMax); return (0); } - /**************************************************************************/ /*! \brief resets sbr inverse filtering structure. @@ -513,17 +594,17 @@ FDKsbrEnc_initInvFiltDetector (HANDLE_SBR_INV_FILT_EST hInvFilt, /*!< Pointer */ /**************************************************************************/ -INT -FDKsbrEnc_resetInvFiltDetector(HANDLE_SBR_INV_FILT_EST hInvFilt, /*!< Handle to the SBR_INV_FILT_EST struct. */ - INT* freqBandTableDetector, /*!< Frequency band table for the inverse filtering. */ - INT numDetectorBands) /*!< Number of inverse filtering bands. */ +INT FDKsbrEnc_resetInvFiltDetector( + HANDLE_SBR_INV_FILT_EST + hInvFilt, /*!< Handle to the SBR_INV_FILT_EST struct. */ + INT *freqBandTableDetector, /*!< Frequency band table for the inverse + filtering. */ + INT numDetectorBands) /*!< Number of inverse filtering bands. */ { - - hInvFilt->numberOfStrongest = 1; - FDKmemcpy(hInvFilt->freqBandTableInvFilt,freqBandTableDetector,(numDetectorBands+1)*sizeof(INT)); + hInvFilt->numberOfStrongest = 1; + FDKmemcpy(hInvFilt->freqBandTableInvFilt, freqBandTableDetector, + (numDetectorBands + 1) * sizeof(INT)); hInvFilt->noDetectorBands = numDetectorBands; return (0); } - - diff --git a/libSBRenc/src/invf_est.h b/libSBRenc/src/invf_est.h index 2bd2a78..3ab6726 100644 --- a/libSBRenc/src/invf_est.h +++ b/libSBRenc/src/invf_est.h @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,40 +90,46 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ /*! \file - \brief Inverse Filtering detection prototypes + \brief Inverse Filtering detection prototypes $Revision: 92790 $ */ -#ifndef _INV_FILT_DET_H -#define _INV_FILT_DET_H +#ifndef INVF_EST_H +#define INVF_EST_H #include "sbr_encoder.h" #include "sbr_def.h" #define INVF_SMOOTHING_LENGTH 2 -typedef struct -{ +typedef struct { const FIXP_DBL *quantStepsSbr; const FIXP_DBL *quantStepsOrig; const FIXP_DBL *nrgBorders; - INT numRegionsSbr; - INT numRegionsOrig; - INT numRegionsNrg; + INT numRegionsSbr; + INT numRegionsOrig; + INT numRegionsNrg; INVF_MODE regionSpace[5][5]; INVF_MODE regionSpaceTransient[5][5]; INT EnergyCompFactor[5]; -}DETECTOR_PARAMETERS; +} DETECTOR_PARAMETERS; -typedef struct -{ - FIXP_DBL origQuotaMean[INVF_SMOOTHING_LENGTH+1]; - FIXP_DBL sbrQuotaMean[INVF_SMOOTHING_LENGTH+1]; - FIXP_DBL origQuotaMeanStrongest[INVF_SMOOTHING_LENGTH+1]; - FIXP_DBL sbrQuotaMeanStrongest[INVF_SMOOTHING_LENGTH+1]; +typedef struct { + FIXP_DBL origQuotaMean[INVF_SMOOTHING_LENGTH + 1]; + FIXP_DBL sbrQuotaMean[INVF_SMOOTHING_LENGTH + 1]; + FIXP_DBL origQuotaMeanStrongest[INVF_SMOOTHING_LENGTH + 1]; + FIXP_DBL sbrQuotaMeanStrongest[INVF_SMOOTHING_LENGTH + 1]; FIXP_DBL origQuotaMeanFilt; FIXP_DBL sbrQuotaMeanFilt; @@ -123,12 +140,9 @@ typedef struct FIXP_DBL sbrQuotaMax; FIXP_DBL avgNrg; -}DETECTOR_VALUES; +} DETECTOR_VALUES; - - -typedef struct -{ +typedef struct { INT numberOfStrongest; INT prevRegionSbr[MAX_NUM_NOISE_VALUES]; @@ -145,31 +159,23 @@ typedef struct FIXP_DBL nrgAvg; FIXP_DBL wmQmf[MAX_NUM_NOISE_VALUES]; -} -SBR_INV_FILT_EST; +} SBR_INV_FILT_EST; typedef SBR_INV_FILT_EST *HANDLE_SBR_INV_FILT_EST; -void -FDKsbrEnc_qmfInverseFilteringDetector(HANDLE_SBR_INV_FILT_EST hInvFilt, - FIXP_DBL ** quotaMatrix, - FIXP_DBL *nrgVector, - SCHAR *indexVector, - INT startIndex, - INT stopIndex, - INT transientFlag, - INVF_MODE* infVec); - -INT -FDKsbrEnc_initInvFiltDetector (HANDLE_SBR_INV_FILT_EST hInvFilt, - INT* freqBandTableDetector, - INT numDetectorBands, - UINT useSpeechConfig); - -INT -FDKsbrEnc_resetInvFiltDetector(HANDLE_SBR_INV_FILT_EST hInvFilt, - INT* freqBandTableDetector, - INT numDetectorBands); +void FDKsbrEnc_qmfInverseFilteringDetector(HANDLE_SBR_INV_FILT_EST hInvFilt, + FIXP_DBL **quotaMatrix, + FIXP_DBL *nrgVector, + SCHAR *indexVector, INT startIndex, + INT stopIndex, INT transientFlag, + INVF_MODE *infVec); -#endif /* _QMF_INV_FILT_H */ +INT FDKsbrEnc_initInvFiltDetector(HANDLE_SBR_INV_FILT_EST hInvFilt, + INT *freqBandTableDetector, + INT numDetectorBands, UINT useSpeechConfig); +INT FDKsbrEnc_resetInvFiltDetector(HANDLE_SBR_INV_FILT_EST hInvFilt, + INT *freqBandTableDetector, + INT numDetectorBands); + +#endif /* _QMF_INV_FILT_H */ diff --git a/libSBRenc/src/mh_det.cpp b/libSBRenc/src/mh_det.cpp index bc80a15..2f3b386 100644 --- a/libSBRenc/src/mh_det.cpp +++ b/libSBRenc/src/mh_det.cpp @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,61 +90,78 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ #include "mh_det.h" -#include "sbr_ram.h" +#include "sbrenc_ram.h" #include "sbr_misc.h" - #include "genericStds.h" -#define SFM_SHIFT 2 /* Attention: SFM_SCALE depends on SFM_SHIFT */ -#define SFM_SCALE (MAXVAL_DBL >> SFM_SHIFT) /* 1.0 >> SFM_SHIFT */ - +#define SFM_SHIFT 2 /* Attention: SFM_SCALE depends on SFM_SHIFT */ +#define SFM_SCALE (MAXVAL_DBL >> SFM_SHIFT) /* 1.0 >> SFM_SHIFT */ /*!< Detector Parameters for AAC core codec. */ static const DETECTOR_PARAMETERS_MH paramsAac = { -9, /*!< deltaTime */ -{ -FL2FXCONST_DBL(20.0f*RELAXATION_FLOAT), /*!< thresHoldDiff */ -FL2FXCONST_DBL(1.26f*RELAXATION_FLOAT), /*!< thresHoldDiffGuide */ -FL2FXCONST_DBL(15.0f*RELAXATION_FLOAT), /*!< thresHoldTone */ -FL2FXCONST_DBL((1.0f/15.0f)*RELAXATION_FLOAT), /*!< invThresHoldTone */ -FL2FXCONST_DBL(1.26f*RELAXATION_FLOAT), /*!< thresHoldToneGuide */ -FL2FXCONST_DBL(0.3f)>>SFM_SHIFT, /*!< sfmThresSbr */ -FL2FXCONST_DBL(0.1f)>>SFM_SHIFT, /*!< sfmThresOrig */ -FL2FXCONST_DBL(0.3f), /*!< decayGuideOrig */ -FL2FXCONST_DBL(0.5f), /*!< decayGuideDiff */ -FL2FXCONST_DBL(-0.000112993269), /* LD64(FL2FXCONST_DBL(0.995f)) */ /*!< derivThresMaxLD64 */ -FL2FXCONST_DBL(-0.000112993269), /* LD64(FL2FXCONST_DBL(0.995f)) */ /*!< derivThresBelowLD64 */ -FL2FXCONST_DBL(-0.005030126483f) /* LD64(FL2FXCONST_DBL(0.8f)) */ /*!< derivThresAboveLD64 */ -}, -50 /*!< maxComp */ + 9, /*!< deltaTime */ + { + FL2FXCONST_DBL(20.0f * RELAXATION_FLOAT), /*!< thresHoldDiff */ + FL2FXCONST_DBL(1.26f * RELAXATION_FLOAT), /*!< thresHoldDiffGuide */ + FL2FXCONST_DBL(15.0f * RELAXATION_FLOAT), /*!< thresHoldTone */ + FL2FXCONST_DBL((1.0f / 15.0f) * + RELAXATION_FLOAT), /*!< invThresHoldTone */ + FL2FXCONST_DBL(1.26f * RELAXATION_FLOAT), /*!< thresHoldToneGuide */ + FL2FXCONST_DBL(0.3f) >> SFM_SHIFT, /*!< sfmThresSbr */ + FL2FXCONST_DBL(0.1f) >> SFM_SHIFT, /*!< sfmThresOrig */ + FL2FXCONST_DBL(0.3f), /*!< decayGuideOrig */ + FL2FXCONST_DBL(0.5f), /*!< decayGuideDiff */ + FL2FXCONST_DBL(-0.000112993269), + /* LD64(FL2FXCONST_DBL(0.995f)) */ /*!< derivThresMaxLD64 */ + FL2FXCONST_DBL(-0.000112993269), + /* LD64(FL2FXCONST_DBL(0.995f)) */ /*!< derivThresBelowLD64 */ + FL2FXCONST_DBL( + -0.005030126483f) /* LD64(FL2FXCONST_DBL(0.8f)) */ /*!< + derivThresAboveLD64 + */ + }, + 50 /*!< maxComp */ }; /*!< Detector Parameters for AAC LD core codec. */ static const DETECTOR_PARAMETERS_MH paramsAacLd = { -16, /*!< Delta time. */ -{ -FL2FXCONST_DBL(25.0f*RELAXATION_FLOAT), /*!< thresHoldDiff */ -FL2FXCONST_DBL(1.26f*RELAXATION_FLOAT), /*!< tresHoldDiffGuide */ -FL2FXCONST_DBL(15.0f*RELAXATION_FLOAT), /*!< thresHoldTone */ -FL2FXCONST_DBL((1.0f/15.0f)*RELAXATION_FLOAT), /*!< invThresHoldTone */ -FL2FXCONST_DBL(1.26f*RELAXATION_FLOAT), /*!< thresHoldToneGuide */ -FL2FXCONST_DBL(0.3f)>>SFM_SHIFT, /*!< sfmThresSbr */ -FL2FXCONST_DBL(0.1f)>>SFM_SHIFT, /*!< sfmThresOrig */ -FL2FXCONST_DBL(0.3f), /*!< decayGuideOrig */ -FL2FXCONST_DBL(0.2f), /*!< decayGuideDiff */ -FL2FXCONST_DBL(-0.000112993269), /* LD64(FL2FXCONST_DBL(0.995f)) */ /*!< derivThresMaxLD64 */ -FL2FXCONST_DBL(-0.000112993269), /* LD64(FL2FXCONST_DBL(0.995f)) */ /*!< derivThresBelowLD64 */ -FL2FXCONST_DBL(-0.005030126483f) /* LD64(FL2FXCONST_DBL(0.8f)) */ /*!< derivThresAboveLD64 */ -}, -50 /*!< maxComp */ + 16, /*!< Delta time. */ + { + FL2FXCONST_DBL(25.0f * RELAXATION_FLOAT), /*!< thresHoldDiff */ + FL2FXCONST_DBL(1.26f * RELAXATION_FLOAT), /*!< tresHoldDiffGuide */ + FL2FXCONST_DBL(15.0f * RELAXATION_FLOAT), /*!< thresHoldTone */ + FL2FXCONST_DBL((1.0f / 15.0f) * + RELAXATION_FLOAT), /*!< invThresHoldTone */ + FL2FXCONST_DBL(1.26f * RELAXATION_FLOAT), /*!< thresHoldToneGuide */ + FL2FXCONST_DBL(0.3f) >> SFM_SHIFT, /*!< sfmThresSbr */ + FL2FXCONST_DBL(0.1f) >> SFM_SHIFT, /*!< sfmThresOrig */ + FL2FXCONST_DBL(0.3f), /*!< decayGuideOrig */ + FL2FXCONST_DBL(0.2f), /*!< decayGuideDiff */ + FL2FXCONST_DBL(-0.000112993269), + /* LD64(FL2FXCONST_DBL(0.995f)) */ /*!< derivThresMaxLD64 */ + FL2FXCONST_DBL(-0.000112993269), + /* LD64(FL2FXCONST_DBL(0.995f)) */ /*!< derivThresBelowLD64 */ + FL2FXCONST_DBL( + -0.005030126483f) /* LD64(FL2FXCONST_DBL(0.8f)) */ /*!< + derivThresAboveLD64 + */ + }, + 50 /*!< maxComp */ }; - /**************************************************************************/ /*! \brief Calculates the difference in tonality between original and SBR @@ -145,39 +173,36 @@ FL2FXCONST_DBL(-0.005030126483f) /* LD64(FL2FXCONST_DBL(0.8f)) */ /*!< deriv */ /**************************************************************************/ -static void diff(FIXP_DBL *RESTRICT pTonalityOrig, - FIXP_DBL *pDiffMapped2Scfb, - const UCHAR *RESTRICT pFreqBandTable, - INT nScfb, - SCHAR *indexVector) -{ +static void diff(FIXP_DBL *RESTRICT pTonalityOrig, FIXP_DBL *pDiffMapped2Scfb, + const UCHAR *RESTRICT pFreqBandTable, INT nScfb, + SCHAR *indexVector) { UCHAR i, ll, lu, k; FIXP_DBL maxValOrig, maxValSbr, tmp; INT scale; - for(i=0; i < nScfb; i++){ + for (i = 0; i < nScfb; i++) { ll = pFreqBandTable[i]; - lu = pFreqBandTable[i+1]; + lu = pFreqBandTable[i + 1]; maxValOrig = FL2FXCONST_DBL(0.0f); maxValSbr = FL2FXCONST_DBL(0.0f); - for(k=ll;k<lu;k++){ + for (k = ll; k < lu; k++) { maxValOrig = fixMax(maxValOrig, pTonalityOrig[k]); maxValSbr = fixMax(maxValSbr, pTonalityOrig[indexVector[k]]); } if ((maxValSbr >= RELAXATION)) { - tmp = fDivNorm(maxValOrig, maxValSbr, &scale); - pDiffMapped2Scfb[i] = scaleValue(fMult(tmp,RELAXATION_FRACT), fixMax(-(DFRACT_BITS-1),(scale-RELAXATION_SHIFT))); - } - else { - pDiffMapped2Scfb[i] = maxValOrig; + tmp = fDivNorm(maxValOrig, maxValSbr, &scale); + pDiffMapped2Scfb[i] = + scaleValue(fMult(tmp, RELAXATION_FRACT), + fixMax(-(DFRACT_BITS - 1), (scale - RELAXATION_SHIFT))); + } else { + pDiffMapped2Scfb[i] = maxValOrig; } } } - /**************************************************************************/ /*! \brief Calculates a flatness measure of the tonality measures. @@ -199,87 +224,81 @@ static void diff(FIXP_DBL *RESTRICT pTonalityOrig, */ /**************************************************************************/ -static void calculateFlatnessMeasure(FIXP_DBL *pQuotaBuffer, - SCHAR *indexVector, +static void calculateFlatnessMeasure(FIXP_DBL *pQuotaBuffer, SCHAR *indexVector, FIXP_DBL *pSfmOrigVec, FIXP_DBL *pSfmSbrVec, - const UCHAR *pFreqBandTable, - INT nSfb) -{ - INT i,j; - FIXP_DBL invBands,tmp1,tmp2; - INT shiftFac0,shiftFacSum0; - INT shiftFac1,shiftFacSum1; + const UCHAR *pFreqBandTable, INT nSfb) { + INT i, j; + FIXP_DBL invBands, tmp1, tmp2; + INT shiftFac0, shiftFacSum0; + INT shiftFac1, shiftFacSum1; FIXP_DBL accu; - for(i=0;i<nSfb;i++) - { + for (i = 0; i < nSfb; i++) { INT ll = pFreqBandTable[i]; - INT lu = pFreqBandTable[i+1]; - pSfmOrigVec[i] = (FIXP_DBL)(MAXVAL_DBL>>2); - pSfmSbrVec[i] = (FIXP_DBL)(MAXVAL_DBL>>2); + INT lu = pFreqBandTable[i + 1]; + pSfmOrigVec[i] = (FIXP_DBL)(MAXVAL_DBL >> 2); + pSfmSbrVec[i] = (FIXP_DBL)(MAXVAL_DBL >> 2); - if(lu - ll > 1){ - FIXP_DBL amOrig,amTransp,gmOrig,gmTransp,sfmOrig,sfmTransp; - invBands = GetInvInt(lu-ll); + if (lu - ll > 1) { + FIXP_DBL amOrig, amTransp, gmOrig, gmTransp, sfmOrig, sfmTransp; + invBands = GetInvInt(lu - ll); shiftFacSum0 = 0; shiftFacSum1 = 0; amOrig = amTransp = FL2FXCONST_DBL(0.0f); gmOrig = gmTransp = (FIXP_DBL)MAXVAL_DBL; - for(j= ll; j<lu; j++) { - sfmOrig = pQuotaBuffer[j]; + for (j = ll; j < lu; j++) { + sfmOrig = pQuotaBuffer[j]; sfmTransp = pQuotaBuffer[indexVector[j]]; - amOrig += fMult(sfmOrig, invBands); + amOrig += fMult(sfmOrig, invBands); amTransp += fMult(sfmTransp, invBands); shiftFac0 = CountLeadingBits(sfmOrig); shiftFac1 = CountLeadingBits(sfmTransp); - gmOrig = fMult(gmOrig, sfmOrig<<shiftFac0); - gmTransp = fMult(gmTransp, sfmTransp<<shiftFac1); + gmOrig = fMult(gmOrig, sfmOrig << shiftFac0); + gmTransp = fMult(gmTransp, sfmTransp << shiftFac1); shiftFacSum0 += shiftFac0; shiftFacSum1 += shiftFac1; } if (gmOrig > FL2FXCONST_DBL(0.0f)) { - - tmp1 = CalcLdData(gmOrig); /* CalcLd64(x)/64 */ - tmp1 = fMult(invBands, tmp1); /* y*CalcLd64(x)/64 */ + tmp1 = CalcLdData(gmOrig); /* CalcLd64(x)/64 */ + tmp1 = fMult(invBands, tmp1); /* y*CalcLd64(x)/64 */ /* y*k/64 */ - accu = (FIXP_DBL)-shiftFacSum0 << (DFRACT_BITS-1-8); - tmp2 = fMultDiv2(invBands, accu) << (2+1); + accu = (FIXP_DBL)-shiftFacSum0 << (DFRACT_BITS - 1 - 8); + tmp2 = fMultDiv2(invBands, accu) << (2 + 1); - tmp2 = tmp1 + tmp2; /* y*CalcLd64(x)/64 + y*k/64 */ - gmOrig = CalcInvLdData(tmp2); /* CalcInvLd64(z'); */ - } - else { + tmp2 = tmp1 + tmp2; /* y*CalcLd64(x)/64 + y*k/64 */ + gmOrig = CalcInvLdData(tmp2); /* CalcInvLd64(z'); */ + } else { gmOrig = FL2FXCONST_DBL(0.0f); } if (gmTransp > FL2FXCONST_DBL(0.0f)) { - - tmp1 = CalcLdData(gmTransp); /* CalcLd64(x)/64 */ - tmp1 = fMult(invBands, tmp1); /* y*CalcLd64(x)/64 */ + tmp1 = CalcLdData(gmTransp); /* CalcLd64(x)/64 */ + tmp1 = fMult(invBands, tmp1); /* y*CalcLd64(x)/64 */ /* y*k/64 */ - accu = (FIXP_DBL)-shiftFacSum1 << (DFRACT_BITS-1-8); - tmp2 = fMultDiv2(invBands, accu) << (2+1); + accu = (FIXP_DBL)-shiftFacSum1 << (DFRACT_BITS - 1 - 8); + tmp2 = fMultDiv2(invBands, accu) << (2 + 1); tmp2 = tmp1 + tmp2; /* y*CalcLd64(x)/64 + y*k/64 */ gmTransp = CalcInvLdData(tmp2); /* CalcInvLd64(z'); */ - } - else { + } else { gmTransp = FL2FXCONST_DBL(0.0f); } - if ( amOrig != FL2FXCONST_DBL(0.0f) ) - pSfmOrigVec[i] = FDKsbrEnc_LSI_divide_scale_fract(gmOrig,amOrig,SFM_SCALE); + if (amOrig != FL2FXCONST_DBL(0.0f)) + pSfmOrigVec[i] = + FDKsbrEnc_LSI_divide_scale_fract(gmOrig, amOrig, SFM_SCALE); - if ( amTransp != FL2FXCONST_DBL(0.0f) ) - pSfmSbrVec[i] = FDKsbrEnc_LSI_divide_scale_fract(gmTransp,amTransp,SFM_SCALE); + if (amTransp != FL2FXCONST_DBL(0.0f)) + pSfmSbrVec[i] = + FDKsbrEnc_LSI_divide_scale_fract(gmTransp, amTransp, SFM_SCALE); } } } @@ -293,39 +312,26 @@ static void calculateFlatnessMeasure(FIXP_DBL *pQuotaBuffer, */ /**************************************************************************/ -static void calculateDetectorInput(FIXP_DBL **RESTRICT pQuotaBuffer, /*!< Pointer to tonality matrix. */ - SCHAR *RESTRICT indexVector, - FIXP_DBL **RESTRICT tonalityDiff, - FIXP_DBL **RESTRICT pSfmOrig, - FIXP_DBL **RESTRICT pSfmSbr, - const UCHAR *freqBandTable, - INT nSfb, - INT noEstPerFrame, - INT move) -{ +static void calculateDetectorInput( + FIXP_DBL **RESTRICT pQuotaBuffer, /*!< Pointer to tonality matrix. */ + SCHAR *RESTRICT indexVector, FIXP_DBL **RESTRICT tonalityDiff, + FIXP_DBL **RESTRICT pSfmOrig, FIXP_DBL **RESTRICT pSfmSbr, + const UCHAR *freqBandTable, INT nSfb, INT noEstPerFrame, INT move) { INT est; /* New estimate. */ - for (est=0; est < noEstPerFrame; est++) { - - diff(pQuotaBuffer[est+move], - tonalityDiff[est+move], - freqBandTable, - nSfb, - indexVector); - - calculateFlatnessMeasure(pQuotaBuffer[est+ move], - indexVector, - pSfmOrig[est + move], - pSfmSbr[est + move], - freqBandTable, - nSfb); + for (est = 0; est < noEstPerFrame; est++) { + diff(pQuotaBuffer[est + move], tonalityDiff[est + move], freqBandTable, + nSfb, indexVector); + + calculateFlatnessMeasure(pQuotaBuffer[est + move], indexVector, + pSfmOrig[est + move], pSfmSbr[est + move], + freqBandTable, nSfb); } } - /**************************************************************************/ /*! \brief Checks that the detection is not due to a LP filter @@ -340,97 +346,97 @@ static void calculateDetectorInput(FIXP_DBL **RESTRICT pQuotaBuffer, /*!< Point /**************************************************************************/ static void removeLowPassDetection(UCHAR *RESTRICT pAddHarmSfb, UCHAR **RESTRICT pDetectionVectors, - INT start, - INT stop, - INT nSfb, + INT start, INT stop, INT nSfb, const UCHAR *RESTRICT pFreqBandTable, FIXP_DBL *RESTRICT pNrgVector, THRES_HOLDS mhThresh) { - INT i,est; + INT i, est; INT maxDerivPos = pFreqBandTable[nSfb]; INT numBands = pFreqBandTable[nSfb]; - FIXP_DBL nrgLow,nrgHigh; - FIXP_DBL nrgLD64,nrgLowLD64,nrgHighLD64,nrgDiffLD64; - FIXP_DBL valLD64,maxValLD64,maxValAboveLD64; + FIXP_DBL nrgLow, nrgHigh; + FIXP_DBL nrgLD64, nrgLowLD64, nrgHighLD64, nrgDiffLD64; + FIXP_DBL valLD64, maxValLD64, maxValAboveLD64; INT bLPsignal = 0; maxValLD64 = FL2FXCONST_DBL(-1.0f); - for(i = numBands - 1 - 2; i > pFreqBandTable[0];i--){ - nrgLow = pNrgVector[i]; + for (i = numBands - 1 - 2; i > pFreqBandTable[0]; i--) { + nrgLow = pNrgVector[i]; nrgHigh = pNrgVector[i + 2]; - if(nrgLow != FL2FXCONST_DBL(0.0f) && nrgLow > nrgHigh){ - nrgLowLD64 = CalcLdData(nrgLow>>1); - nrgDiffLD64 = CalcLdData((nrgLow>>1)-(nrgHigh>>1)); - valLD64 = nrgDiffLD64-nrgLowLD64; - if(valLD64 > maxValLD64){ + if (nrgLow != FL2FXCONST_DBL(0.0f) && nrgLow > nrgHigh) { + nrgLowLD64 = CalcLdData(nrgLow >> 1); + nrgDiffLD64 = CalcLdData((nrgLow >> 1) - (nrgHigh >> 1)); + valLD64 = nrgDiffLD64 - nrgLowLD64; + if (valLD64 > maxValLD64) { maxDerivPos = i; maxValLD64 = valLD64; } - if(maxValLD64 > mhThresh.derivThresMaxLD64) { + if (maxValLD64 > mhThresh.derivThresMaxLD64) { break; } } } - /* Find the largest "gradient" above. (should be relatively flat, hence we expect a low value - if the signal is LP.*/ + /* Find the largest "gradient" above. (should be relatively flat, hence we + expect a low value if the signal is LP.*/ maxValAboveLD64 = FL2FXCONST_DBL(-1.0f); - for(i = numBands - 1 - 2; i > maxDerivPos + 2;i--){ - nrgLow = pNrgVector[i]; + for (i = numBands - 1 - 2; i > maxDerivPos + 2; i--) { + nrgLow = pNrgVector[i]; nrgHigh = pNrgVector[i + 2]; - if(nrgLow != FL2FXCONST_DBL(0.0f) && nrgLow > nrgHigh){ - nrgLowLD64 = CalcLdData(nrgLow>>1); - nrgDiffLD64 = CalcLdData((nrgLow>>1)-(nrgHigh>>1)); - valLD64 = nrgDiffLD64-nrgLowLD64; - if(valLD64 > maxValAboveLD64){ + if (nrgLow != FL2FXCONST_DBL(0.0f) && nrgLow > nrgHigh) { + nrgLowLD64 = CalcLdData(nrgLow >> 1); + nrgDiffLD64 = CalcLdData((nrgLow >> 1) - (nrgHigh >> 1)); + valLD64 = nrgDiffLD64 - nrgLowLD64; + if (valLD64 > maxValAboveLD64) { maxValAboveLD64 = valLD64; } - } - else { - if(nrgHigh != FL2FXCONST_DBL(0.0f) && nrgHigh > nrgLow){ - nrgHighLD64 = CalcLdData(nrgHigh>>1); - nrgDiffLD64 = CalcLdData((nrgHigh>>1)-(nrgLow>>1)); - valLD64 = nrgDiffLD64-nrgHighLD64; - if(valLD64 > maxValAboveLD64){ + } else { + if (nrgHigh != FL2FXCONST_DBL(0.0f) && nrgHigh > nrgLow) { + nrgHighLD64 = CalcLdData(nrgHigh >> 1); + nrgDiffLD64 = CalcLdData((nrgHigh >> 1) - (nrgLow >> 1)); + valLD64 = nrgDiffLD64 - nrgHighLD64; + if (valLD64 > maxValAboveLD64) { maxValAboveLD64 = valLD64; } } - } + } } - if(maxValLD64 > mhThresh.derivThresMaxLD64 && maxValAboveLD64 < mhThresh.derivThresAboveLD64){ + if (maxValLD64 > mhThresh.derivThresMaxLD64 && + maxValAboveLD64 < mhThresh.derivThresAboveLD64) { bLPsignal = 1; - for(i = maxDerivPos - 1; i > maxDerivPos - 5 && i >= 0 ; i--){ - if(pNrgVector[i] != FL2FXCONST_DBL(0.0f) && pNrgVector[i] > pNrgVector[maxDerivPos + 2]){ - nrgDiffLD64 = CalcLdData((pNrgVector[i]>>1)-(pNrgVector[maxDerivPos + 2]>>1)); - nrgLD64 = CalcLdData(pNrgVector[i]>>1); - valLD64 = nrgDiffLD64-nrgLD64; - if(valLD64 < mhThresh.derivThresBelowLD64) { + for (i = maxDerivPos - 1; i > maxDerivPos - 5 && i >= 0; i--) { + if (pNrgVector[i] != FL2FXCONST_DBL(0.0f) && + pNrgVector[i] > pNrgVector[maxDerivPos + 2]) { + nrgDiffLD64 = CalcLdData((pNrgVector[i] >> 1) - + (pNrgVector[maxDerivPos + 2] >> 1)); + nrgLD64 = CalcLdData(pNrgVector[i] >> 1); + valLD64 = nrgDiffLD64 - nrgLD64; + if (valLD64 < mhThresh.derivThresBelowLD64) { bLPsignal = 0; break; } - } - else{ + } else { bLPsignal = 0; break; } } } - if(bLPsignal){ - for(i=0;i<nSfb;i++){ - if(maxDerivPos >= pFreqBandTable[i] && maxDerivPos < pFreqBandTable[i+1]) + if (bLPsignal) { + for (i = 0; i < nSfb; i++) { + if (maxDerivPos >= pFreqBandTable[i] && + maxDerivPos < pFreqBandTable[i + 1]) break; } - if(pAddHarmSfb[i]){ + if (pAddHarmSfb[i]) { pAddHarmSfb[i] = 0; - for(est = start; est < stop ; est++){ + for (est = start; est < stop; est++) { pDetectionVectors[est][i] = 0; } } @@ -447,44 +453,37 @@ static void removeLowPassDetection(UCHAR *RESTRICT pAddHarmSfb, */ /**************************************************************************/ -static INT isDetectionOfNewToneAllowed(const SBR_FRAME_INFO *pFrameInfo, - INT *pDetectionStartPos, - INT noEstPerFrame, - INT prevTransientFrame, - INT prevTransientPos, - INT prevTransientFlag, - INT transientPosOffset, - INT transientFlag, - INT transientPos, - INT deltaTime, - HANDLE_SBR_MISSING_HARMONICS_DETECTOR h_sbrMissingHarmonicsDetector) -{ +static INT isDetectionOfNewToneAllowed( + const SBR_FRAME_INFO *pFrameInfo, INT *pDetectionStartPos, + INT noEstPerFrame, INT prevTransientFrame, INT prevTransientPos, + INT prevTransientFlag, INT transientPosOffset, INT transientFlag, + INT transientPos, INT deltaTime, + HANDLE_SBR_MISSING_HARMONICS_DETECTOR h_sbrMissingHarmonicsDetector) { INT transientFrame, newDetectionAllowed; - /* Determine if this is a frame where a transient starts... * If the transient flag was set the previous frame but not the * transient frame flag, the transient frame flag is set in the current frame. *****************************************************************************/ transientFrame = 0; - if(transientFlag){ - if(transientPos + transientPosOffset < pFrameInfo->borders[pFrameInfo->nEnvelopes]) + if (transientFlag) { + if (transientPos + transientPosOffset < + pFrameInfo->borders[pFrameInfo->nEnvelopes]) { transientFrame = 1; - if(noEstPerFrame > 1){ - if(transientPos + transientPosOffset > h_sbrMissingHarmonicsDetector->timeSlots >> 1){ + if (noEstPerFrame > 1) { + if (transientPos + transientPosOffset > + h_sbrMissingHarmonicsDetector->timeSlots >> 1) { *pDetectionStartPos = noEstPerFrame; - } - else{ + } else { *pDetectionStartPos = noEstPerFrame >> 1; } - } - else{ + } else { *pDetectionStartPos = noEstPerFrame; } - } - else{ - if(prevTransientFlag && !prevTransientFrame){ + } + } else { + if (prevTransientFlag && !prevTransientFrame) { transientFrame = 1; *pDetectionStartPos = 0; } @@ -497,25 +496,25 @@ static INT isDetectionOfNewToneAllowed(const SBR_FRAME_INFO *pFrameInfo, * to the start of the current frame. ****************************************************************/ newDetectionAllowed = 0; - if(transientFrame){ + if (transientFrame) { newDetectionAllowed = 1; - } - else { - if(prevTransientFrame && - fixp_abs(pFrameInfo->borders[0] - (prevTransientPos + transientPosOffset - - h_sbrMissingHarmonicsDetector->timeSlots)) < deltaTime) + } else { + if (prevTransientFrame && + fixp_abs(pFrameInfo->borders[0] - + (prevTransientPos + transientPosOffset - + h_sbrMissingHarmonicsDetector->timeSlots)) < deltaTime) { newDetectionAllowed = 1; *pDetectionStartPos = 0; + } } - h_sbrMissingHarmonicsDetector->previousTransientFlag = transientFlag; + h_sbrMissingHarmonicsDetector->previousTransientFlag = transientFlag; h_sbrMissingHarmonicsDetector->previousTransientFrame = transientFrame; - h_sbrMissingHarmonicsDetector->previousTransientPos = transientPos; + h_sbrMissingHarmonicsDetector->previousTransientPos = transientPos; return (newDetectionAllowed); } - /**************************************************************************/ /*! \brief Cleans up the detection after a transient. @@ -525,51 +524,41 @@ static INT isDetectionOfNewToneAllowed(const SBR_FRAME_INFO *pFrameInfo, */ /**************************************************************************/ -static void transientCleanUp(FIXP_DBL **quotaBuffer, - INT nSfb, - UCHAR **detectionVectors, - UCHAR *pAddHarmSfb, - UCHAR *pPrevAddHarmSfb, - INT ** signBuffer, - const UCHAR *pFreqBandTable, - INT start, - INT stop, - INT newDetectionAllowed, - FIXP_DBL *pNrgVector, - THRES_HOLDS mhThresh) -{ - INT i,j,li, ui,est; - - for(est=start; est < stop; est++) { - for(i=0; i<nSfb; i++) { +static void transientCleanUp(FIXP_DBL **quotaBuffer, INT nSfb, + UCHAR **detectionVectors, UCHAR *pAddHarmSfb, + UCHAR *pPrevAddHarmSfb, INT **signBuffer, + const UCHAR *pFreqBandTable, INT start, INT stop, + INT newDetectionAllowed, FIXP_DBL *pNrgVector, + THRES_HOLDS mhThresh) { + INT i, j, est; + + for (est = start; est < stop; est++) { + for (i = 0; i < nSfb; i++) { pAddHarmSfb[i] = pAddHarmSfb[i] || detectionVectors[est][i]; } } - if(newDetectionAllowed == 1){ + if (newDetectionAllowed == 1) { /* * Check for duplication of sines located * on the border of two scf-bands. *************************************************/ - for(i=0;i<nSfb-1;i++) { - li = pFreqBandTable[i]; - ui = pFreqBandTable[i+1]; - + for (i = 0; i < nSfb - 1; i++) { /* detection in adjacent channels.*/ - if(pAddHarmSfb[i] && pAddHarmSfb[i+1]) { + if (pAddHarmSfb[i] && pAddHarmSfb[i + 1]) { FIXP_DBL maxVal1, maxVal2; INT maxPos1, maxPos2, maxPosTime1, maxPosTime2; - li = pFreqBandTable[i]; - ui = pFreqBandTable[i+1]; + INT li = pFreqBandTable[i]; + INT ui = pFreqBandTable[i + 1]; /* Find maximum tonality in the the two scf bands.*/ maxPosTime1 = start; maxPos1 = li; maxVal1 = quotaBuffer[start][li]; - for(est = start; est < stop; est++){ - for(j = li; j<ui; j++){ - if(quotaBuffer[est][j] > maxVal1){ + for (est = start; est < stop; est++) { + for (j = li; j < ui; j++) { + if (quotaBuffer[est][j] > maxVal1) { maxVal1 = quotaBuffer[est][j]; maxPos1 = j; maxPosTime1 = est; @@ -577,16 +566,16 @@ static void transientCleanUp(FIXP_DBL **quotaBuffer, } } - li = pFreqBandTable[i+1]; - ui = pFreqBandTable[i+2]; + li = pFreqBandTable[i + 1]; + ui = pFreqBandTable[i + 2]; /* Find maximum tonality in the the two scf bands.*/ maxPosTime2 = start; maxPos2 = li; maxVal2 = quotaBuffer[start][li]; - for(est = start; est < stop; est++){ - for(j = li; j<ui; j++){ - if(quotaBuffer[est][j] > maxVal2){ + for (est = start; est < stop; est++) { + for (j = li; j < ui; j++) { + if (quotaBuffer[est][j] > maxVal2) { maxVal2 = quotaBuffer[est][j]; maxPos2 = j; maxPosTime2 = est; @@ -596,40 +585,39 @@ static void transientCleanUp(FIXP_DBL **quotaBuffer, /* If the maximum values are in adjacent QMF-channels, we need to remove the lowest of the two.*/ - if(maxPos2-maxPos1 < 2){ - - if(pPrevAddHarmSfb[i] == 1 && pPrevAddHarmSfb[i+1] == 0){ + if (maxPos2 - maxPos1 < 2) { + if (pPrevAddHarmSfb[i] == 1 && pPrevAddHarmSfb[i + 1] == 0) { /* Keep the lower, remove the upper.*/ - pAddHarmSfb[i+1] = 0; - for(est=start; est<stop; est++){ - detectionVectors[est][i+1] = 0; + pAddHarmSfb[i + 1] = 0; + for (est = start; est < stop; est++) { + detectionVectors[est][i + 1] = 0; } - } - else{ - if(pPrevAddHarmSfb[i] == 0 && pPrevAddHarmSfb[i+1] == 1){ + } else { + if (pPrevAddHarmSfb[i] == 0 && pPrevAddHarmSfb[i + 1] == 1) { /* Keep the upper, remove the lower.*/ pAddHarmSfb[i] = 0; - for(est=start; est<stop; est++){ + for (est = start; est < stop; est++) { detectionVectors[est][i] = 0; } - } - else{ - /* If the maximum values are in adjacent QMF-channels, and if the signs indicate that it is the same sine, - we need to remove the lowest of the two.*/ - if(maxVal1 > maxVal2){ - if(signBuffer[maxPosTime1][maxPos2] < 0 && signBuffer[maxPosTime1][maxPos1] > 0){ + } else { + /* If the maximum values are in adjacent QMF-channels, and if the + signs indicate that it is the same sine, we need to remove the + lowest of the two.*/ + if (maxVal1 > maxVal2) { + if (signBuffer[maxPosTime1][maxPos2] < 0 && + signBuffer[maxPosTime1][maxPos1] > 0) { /* Keep the lower, remove the upper.*/ - pAddHarmSfb[i+1] = 0; - for(est=start; est<stop; est++){ - detectionVectors[est][i+1] = 0; + pAddHarmSfb[i + 1] = 0; + for (est = start; est < stop; est++) { + detectionVectors[est][i + 1] = 0; } } - } - else{ - if(signBuffer[maxPosTime2][maxPos2] < 0 && signBuffer[maxPosTime2][maxPos1] > 0){ + } else { + if (signBuffer[maxPosTime2][maxPos2] < 0 && + signBuffer[maxPosTime2][maxPos1] > 0) { /* Keep the upper, remove the lower.*/ pAddHarmSfb[i] = 0; - for(est=start; est<stop; est++){ + for (est = start; est < stop; est++) { detectionVectors[est][i] = 0; } } @@ -641,28 +629,19 @@ static void transientCleanUp(FIXP_DBL **quotaBuffer, } /* Make sure that the detection is not the cut-off of a low pass filter. */ - removeLowPassDetection(pAddHarmSfb, - detectionVectors, - start, - stop, - nSfb, - pFreqBandTable, - pNrgVector, - mhThresh); - } - else { - /* - * If a missing harmonic wasn't missing the previous frame - * the transient-flag needs to be set in order to be allowed to detect it. - *************************************************************************/ - for(i=0;i<nSfb;i++){ - if(pAddHarmSfb[i] - pPrevAddHarmSfb[i] > 0) - pAddHarmSfb[i] = 0; + removeLowPassDetection(pAddHarmSfb, detectionVectors, start, stop, nSfb, + pFreqBandTable, pNrgVector, mhThresh); + } else { + /* + * If a missing harmonic wasn't missing the previous frame + * the transient-flag needs to be set in order to be allowed to detect it. + *************************************************************************/ + for (i = 0; i < nSfb; i++) { + if (pAddHarmSfb[i] - pPrevAddHarmSfb[i] > 0) pAddHarmSfb[i] = 0; } } } - /*****************************************************************************/ /*! \brief Detection for one tonality estimate. @@ -689,42 +668,35 @@ static void transientCleanUp(FIXP_DBL **quotaBuffer, */ /**************************************************************************/ -static void detection(FIXP_DBL *quotaBuffer, - FIXP_DBL *pDiffVecScfb, - INT nSfb, - UCHAR *pHarmVec, - const UCHAR *pFreqBandTable, - FIXP_DBL *sfmOrig, - FIXP_DBL *sfmSbr, - GUIDE_VECTORS guideVectors, - GUIDE_VECTORS newGuideVectors, - THRES_HOLDS mhThresh) -{ - - INT i,j,ll, lu; - FIXP_DBL thresTemp,thresOrig; +static void detection(FIXP_DBL *quotaBuffer, FIXP_DBL *pDiffVecScfb, INT nSfb, + UCHAR *pHarmVec, const UCHAR *pFreqBandTable, + FIXP_DBL *sfmOrig, FIXP_DBL *sfmSbr, + GUIDE_VECTORS guideVectors, GUIDE_VECTORS newGuideVectors, + THRES_HOLDS mhThresh) { + INT i, j, ll, lu; + FIXP_DBL thresTemp, thresOrig; /* * Do detection on the difference vector, i.e. the difference between * the original and the transposed. *********************************************************************/ - for(i=0;i<nSfb;i++){ - + for (i = 0; i < nSfb; i++) { thresTemp = (guideVectors.guideVectorDiff[i] != FL2FXCONST_DBL(0.0f)) - ? fMax(fMult(mhThresh.decayGuideDiff,guideVectors.guideVectorDiff[i]), mhThresh.thresHoldDiffGuide) - : mhThresh.thresHoldDiff; + ? fMax(fMult(mhThresh.decayGuideDiff, + guideVectors.guideVectorDiff[i]), + mhThresh.thresHoldDiffGuide) + : mhThresh.thresHoldDiff; thresTemp = fMin(thresTemp, mhThresh.thresHoldDiff); - if(pDiffVecScfb[i] > thresTemp){ + if (pDiffVecScfb[i] > thresTemp) { pHarmVec[i] = 1; newGuideVectors.guideVectorDiff[i] = pDiffVecScfb[i]; - } - else{ + } else { /* If the guide wasn't zero, but the current level is to low, start tracking the decay on the tone in the original rather than the difference.*/ - if(guideVectors.guideVectorDiff[i] != FL2FXCONST_DBL(0.0f)){ + if (guideVectors.guideVectorDiff[i] != FL2FXCONST_DBL(0.0f)) { guideVectors.guideVectorOrig[i] = mhThresh.thresHoldToneGuide; } } @@ -736,16 +708,18 @@ static void detection(FIXP_DBL *quotaBuffer, * multiple tones in the sbr signal. ****************************************************/ - for(i=0;i<nSfb;i++){ + for (i = 0; i < nSfb; i++) { ll = pFreqBandTable[i]; - lu = pFreqBandTable[i+1]; + lu = pFreqBandTable[i + 1]; - thresOrig = fixMax(fMult(guideVectors.guideVectorOrig[i], mhThresh.decayGuideOrig), mhThresh.thresHoldToneGuide); + thresOrig = + fixMax(fMult(guideVectors.guideVectorOrig[i], mhThresh.decayGuideOrig), + mhThresh.thresHoldToneGuide); thresOrig = fixMin(thresOrig, mhThresh.thresHoldTone); - if(guideVectors.guideVectorOrig[i] != FL2FXCONST_DBL(0.0f)){ - for(j= ll;j<lu;j++){ - if(quotaBuffer[j] > thresOrig){ + if (guideVectors.guideVectorOrig[i] != FL2FXCONST_DBL(0.0f)) { + for (j = ll; j < lu; j++) { + if (quotaBuffer[j] > thresOrig) { pHarmVec[i] = 1; newGuideVectors.guideVectorOrig[i] = quotaBuffer[j]; } @@ -759,33 +733,36 @@ static void detection(FIXP_DBL *quotaBuffer, ****************************************************/ thresOrig = mhThresh.thresHoldTone; - for(i=0;i<nSfb;i++){ + for (i = 0; i < nSfb; i++) { ll = pFreqBandTable[i]; - lu = pFreqBandTable[i+1]; - - if(pHarmVec[i] == 0){ - if(lu -ll > 1){ - for(j= ll;j<lu;j++){ - if(quotaBuffer[j] > thresOrig && (sfmSbr[i] > mhThresh.sfmThresSbr && sfmOrig[i] < mhThresh.sfmThresOrig)){ + lu = pFreqBandTable[i + 1]; + + if (pHarmVec[i] == 0) { + if (lu - ll > 1) { + for (j = ll; j < lu; j++) { + if (quotaBuffer[j] > thresOrig && + (sfmSbr[i] > mhThresh.sfmThresSbr && + sfmOrig[i] < mhThresh.sfmThresOrig)) { pHarmVec[i] = 1; newGuideVectors.guideVectorOrig[i] = quotaBuffer[j]; } } - } - else{ - if(i < nSfb -1){ + } else { + if (i < nSfb - 1) { ll = pFreqBandTable[i]; - if(i>0){ - if(quotaBuffer[ll] > mhThresh.thresHoldTone && (pDiffVecScfb[i+1] < mhThresh.invThresHoldTone || pDiffVecScfb[i-1] < mhThresh.invThresHoldTone)){ - pHarmVec[i] = 1; - newGuideVectors.guideVectorOrig[i] = quotaBuffer[ll]; + if (i > 0) { + if (quotaBuffer[ll] > mhThresh.thresHoldTone && + (pDiffVecScfb[i + 1] < mhThresh.invThresHoldTone || + pDiffVecScfb[i - 1] < mhThresh.invThresHoldTone)) { + pHarmVec[i] = 1; + newGuideVectors.guideVectorOrig[i] = quotaBuffer[ll]; } - } - else{ - if(quotaBuffer[ll] > mhThresh.thresHoldTone && pDiffVecScfb[i+1] < mhThresh.invThresHoldTone){ - pHarmVec[i] = 1; - newGuideVectors.guideVectorOrig[i] = quotaBuffer[ll]; + } else { + if (quotaBuffer[ll] > mhThresh.thresHoldTone && + pDiffVecScfb[i + 1] < mhThresh.invThresHoldTone) { + pHarmVec[i] = 1; + newGuideVectors.guideVectorOrig[i] = quotaBuffer[ll]; } } } @@ -794,7 +771,6 @@ static void detection(FIXP_DBL *quotaBuffer, } } - /**************************************************************************/ /*! \brief Do detection for every tonality estimate, using forward prediction. @@ -804,149 +780,116 @@ static void detection(FIXP_DBL *quotaBuffer, */ /**************************************************************************/ -static void detectionWithPrediction(FIXP_DBL **quotaBuffer, - FIXP_DBL **pDiffVecScfb, - INT ** signBuffer, - INT nSfb, - const UCHAR* pFreqBandTable, - FIXP_DBL **sfmOrig, - FIXP_DBL **sfmSbr, - UCHAR **detectionVectors, - UCHAR *pPrevAddHarmSfb, - GUIDE_VECTORS *guideVectors, - INT noEstPerFrame, - INT detectionStart, - INT totNoEst, - INT newDetectionAllowed, - INT *pAddHarmFlag, - UCHAR *pAddHarmSfb, - FIXP_DBL *pNrgVector, - const DETECTOR_PARAMETERS_MH *mhParams) -{ - INT est = 0,i; +static void detectionWithPrediction( + FIXP_DBL **quotaBuffer, FIXP_DBL **pDiffVecScfb, INT **signBuffer, INT nSfb, + const UCHAR *pFreqBandTable, FIXP_DBL **sfmOrig, FIXP_DBL **sfmSbr, + UCHAR **detectionVectors, UCHAR *pPrevAddHarmSfb, + GUIDE_VECTORS *guideVectors, INT noEstPerFrame, INT detectionStart, + INT totNoEst, INT newDetectionAllowed, INT *pAddHarmFlag, + UCHAR *pAddHarmSfb, FIXP_DBL *pNrgVector, + const DETECTOR_PARAMETERS_MH *mhParams) { + INT est = 0, i; INT start; - FDKmemclear(pAddHarmSfb,nSfb*sizeof(UCHAR)); + FDKmemclear(pAddHarmSfb, nSfb * sizeof(UCHAR)); - if(newDetectionAllowed){ - - /* Since we don't want to use the transient region for detection (since the tonality values - tend to be a bit unreliable for this region) the guide-values are copied to the current - starting point. */ - if(totNoEst > 1){ - start = detectionStart+1; + if (newDetectionAllowed) { + /* Since we don't want to use the transient region for detection (since the + tonality values tend to be a bit unreliable for this region) the + guide-values are copied to the current starting point. */ + if (totNoEst > 1) { + start = detectionStart + 1; if (start != 0) { - FDKmemcpy(guideVectors[start].guideVectorDiff,guideVectors[0].guideVectorDiff,nSfb*sizeof(FIXP_DBL)); - FDKmemcpy(guideVectors[start].guideVectorOrig,guideVectors[0].guideVectorOrig,nSfb*sizeof(FIXP_DBL)); - FDKmemclear(guideVectors[start-1].guideVectorDetected,nSfb*sizeof(UCHAR)); + FDKmemcpy(guideVectors[start].guideVectorDiff, + guideVectors[0].guideVectorDiff, nSfb * sizeof(FIXP_DBL)); + FDKmemcpy(guideVectors[start].guideVectorOrig, + guideVectors[0].guideVectorOrig, nSfb * sizeof(FIXP_DBL)); + FDKmemclear(guideVectors[start - 1].guideVectorDetected, + nSfb * sizeof(UCHAR)); } - } - else{ + } else { start = 0; } - } - else{ + } else { start = 0; } - - for(est = start; est < totNoEst; est++){ - + for (est = start; est < totNoEst; est++) { /* - * Do detection on the current frame using - * guide-info from the previous. - *******************************************/ - if(est > 0){ - FDKmemcpy(guideVectors[est].guideVectorDetected,detectionVectors[est-1],nSfb*sizeof(UCHAR)); + * Do detection on the current frame using + * guide-info from the previous. + *******************************************/ + if (est > 0) { + FDKmemcpy(guideVectors[est].guideVectorDetected, + detectionVectors[est - 1], nSfb * sizeof(UCHAR)); } - FDKmemclear(detectionVectors[est], nSfb*sizeof(UCHAR)); - - if(est < totNoEst-1){ - FDKmemclear(guideVectors[est+1].guideVectorDiff,nSfb*sizeof(FIXP_DBL)); - FDKmemclear(guideVectors[est+1].guideVectorOrig,nSfb*sizeof(FIXP_DBL)); - FDKmemclear(guideVectors[est+1].guideVectorDetected,nSfb*sizeof(UCHAR)); - - detection(quotaBuffer[est], - pDiffVecScfb[est], - nSfb, - detectionVectors[est], - pFreqBandTable, - sfmOrig[est], - sfmSbr[est], - guideVectors[est], - guideVectors[est+1], + FDKmemclear(detectionVectors[est], nSfb * sizeof(UCHAR)); + + if (est < totNoEst - 1) { + FDKmemclear(guideVectors[est + 1].guideVectorDiff, + nSfb * sizeof(FIXP_DBL)); + FDKmemclear(guideVectors[est + 1].guideVectorOrig, + nSfb * sizeof(FIXP_DBL)); + FDKmemclear(guideVectors[est + 1].guideVectorDetected, + nSfb * sizeof(UCHAR)); + + detection(quotaBuffer[est], pDiffVecScfb[est], nSfb, + detectionVectors[est], pFreqBandTable, sfmOrig[est], + sfmSbr[est], guideVectors[est], guideVectors[est + 1], mhParams->thresHolds); - } - else{ - FDKmemclear(guideVectors[est].guideVectorDiff,nSfb*sizeof(FIXP_DBL)); - FDKmemclear(guideVectors[est].guideVectorOrig,nSfb*sizeof(FIXP_DBL)); - FDKmemclear(guideVectors[est].guideVectorDetected,nSfb*sizeof(UCHAR)); - - detection(quotaBuffer[est], - pDiffVecScfb[est], - nSfb, - detectionVectors[est], - pFreqBandTable, - sfmOrig[est], - sfmSbr[est], - guideVectors[est], - guideVectors[est], + } else { + FDKmemclear(guideVectors[est].guideVectorDiff, nSfb * sizeof(FIXP_DBL)); + FDKmemclear(guideVectors[est].guideVectorOrig, nSfb * sizeof(FIXP_DBL)); + FDKmemclear(guideVectors[est].guideVectorDetected, nSfb * sizeof(UCHAR)); + + detection(quotaBuffer[est], pDiffVecScfb[est], nSfb, + detectionVectors[est], pFreqBandTable, sfmOrig[est], + sfmSbr[est], guideVectors[est], guideVectors[est], mhParams->thresHolds); } } - /* Clean up the detection.*/ - transientCleanUp(quotaBuffer, - nSfb, - detectionVectors, - pAddHarmSfb, - pPrevAddHarmSfb, - signBuffer, - pFreqBandTable, - start, - totNoEst, - newDetectionAllowed, - pNrgVector, - mhParams->thresHolds); - + transientCleanUp(quotaBuffer, nSfb, detectionVectors, pAddHarmSfb, + pPrevAddHarmSfb, signBuffer, pFreqBandTable, start, totNoEst, + newDetectionAllowed, pNrgVector, mhParams->thresHolds); /* Set flag... */ *pAddHarmFlag = 0; - for(i=0; i<nSfb; i++){ - if(pAddHarmSfb[i]){ + for (i = 0; i < nSfb; i++) { + if (pAddHarmSfb[i]) { *pAddHarmFlag = 1; break; } } - FDKmemcpy(pPrevAddHarmSfb, pAddHarmSfb, nSfb*sizeof(UCHAR)); - FDKmemcpy(guideVectors[0].guideVectorDetected,pAddHarmSfb,nSfb*sizeof(INT)); - - for(i=0; i<nSfb ; i++){ + FDKmemcpy(pPrevAddHarmSfb, pAddHarmSfb, nSfb * sizeof(UCHAR)); + FDKmemcpy(guideVectors[0].guideVectorDetected, pAddHarmSfb, + nSfb * sizeof(INT)); + for (i = 0; i < nSfb; i++) { guideVectors[0].guideVectorDiff[i] = FL2FXCONST_DBL(0.0f); guideVectors[0].guideVectorOrig[i] = FL2FXCONST_DBL(0.0f); - if(pAddHarmSfb[i] == 1){ - /* If we had a detection use the guide-value in the next frame from the last estimate were the detection - was done.*/ - for(est=start; est < totNoEst; est++){ - if(guideVectors[est].guideVectorDiff[i] != FL2FXCONST_DBL(0.0f)){ - guideVectors[0].guideVectorDiff[i] = guideVectors[est].guideVectorDiff[i]; + if (pAddHarmSfb[i] == 1) { + /* If we had a detection use the guide-value in the next frame from the + last estimate were the detection was done.*/ + for (est = start; est < totNoEst; est++) { + if (guideVectors[est].guideVectorDiff[i] != FL2FXCONST_DBL(0.0f)) { + guideVectors[0].guideVectorDiff[i] = + guideVectors[est].guideVectorDiff[i]; } - if(guideVectors[est].guideVectorOrig[i] != FL2FXCONST_DBL(0.0f)){ - guideVectors[0].guideVectorOrig[i] = guideVectors[est].guideVectorOrig[i]; + if (guideVectors[est].guideVectorOrig[i] != FL2FXCONST_DBL(0.0f)) { + guideVectors[0].guideVectorOrig[i] = + guideVectors[est].guideVectorOrig[i]; } } } } - } - /**************************************************************************/ /*! \brief Calculates a compensation vector for the energy data. @@ -963,38 +906,30 @@ static void detectionWithPrediction(FIXP_DBL **quotaBuffer, */ /**************************************************************************/ -static void calculateCompVector(UCHAR *pAddHarmSfb, - FIXP_DBL **pTonalityMatrix, - INT ** pSignMatrix, - UCHAR *pEnvComp, - INT nSfb, - const UCHAR *freqBandTable, - INT totNoEst, - INT maxComp, - UCHAR *pPrevEnvComp, - INT newDetectionAllowed) -{ - - INT scfBand,est,l,ll,lu,maxPosF,maxPosT; +static void calculateCompVector(UCHAR *pAddHarmSfb, FIXP_DBL **pTonalityMatrix, + INT **pSignMatrix, UCHAR *pEnvComp, INT nSfb, + const UCHAR *freqBandTable, INT totNoEst, + INT maxComp, UCHAR *pPrevEnvComp, + INT newDetectionAllowed) { + INT scfBand, est, l, ll, lu, maxPosF, maxPosT; FIXP_DBL maxVal; INT compValue; FIXP_DBL tmp; - FDKmemclear(pEnvComp,nSfb*sizeof(UCHAR)); + FDKmemclear(pEnvComp, nSfb * sizeof(UCHAR)); - for(scfBand=0; scfBand < nSfb; scfBand++){ - - if(pAddHarmSfb[scfBand]){ /* A missing sine was detected */ + for (scfBand = 0; scfBand < nSfb; scfBand++) { + if (pAddHarmSfb[scfBand]) { /* A missing sine was detected */ ll = freqBandTable[scfBand]; - lu = freqBandTable[scfBand+1]; + lu = freqBandTable[scfBand + 1]; - maxPosF = 0; /* First find the maximum*/ + maxPosF = 0; /* First find the maximum*/ maxPosT = 0; maxVal = FL2FXCONST_DBL(0.0f); - for(est=0;est<totNoEst;est++){ - for(l=ll; l<lu; l++){ - if(pTonalityMatrix[est][l] > maxVal){ + for (est = 0; est < totNoEst; est++) { + for (l = ll; l < lu; l++) { + if (pTonalityMatrix[est][l] > maxVal) { maxVal = pTonalityMatrix[est][l]; maxPosF = l; maxPosT = est; @@ -1010,57 +945,63 @@ static void calculateCompVector(UCHAR *pAddHarmSfb, * in the SBR data, which will cause problems in the decoder, when we * add a sine to just one of the channels. *********************************************************************/ - if(maxPosF == ll && scfBand){ - if(!pAddHarmSfb[scfBand - 1]) { /* No detection below*/ - if (pSignMatrix[maxPosT][maxPosF - 1] > 0 && pSignMatrix[maxPosT][maxPosF] < 0) { - /* The comp value is calulated as the tonallity value, i.e we want to - reduce the envelope data for this channel with as much as the tonality - that is spread from the channel above. (ld64(RELAXATION) = 0.31143075889) */ - tmp = fixp_abs((FIXP_DBL)CalcLdData(pTonalityMatrix[maxPosT][maxPosF - 1]) + RELAXATION_LD64); - tmp = (tmp >> (DFRACT_BITS-1-LD_DATA_SHIFT-1)) + (FIXP_DBL)1; /* shift one bit less for rounding */ + if (maxPosF == ll && scfBand) { + if (!pAddHarmSfb[scfBand - 1]) { /* No detection below*/ + if (pSignMatrix[maxPosT][maxPosF - 1] > 0 && + pSignMatrix[maxPosT][maxPosF] < 0) { + /* The comp value is calulated as the tonallity value, i.e we want + to reduce the envelope data for this channel with as much as the + tonality that is spread from the channel above. (ld64(RELAXATION) + = 0.31143075889) */ + tmp = fixp_abs( + (FIXP_DBL)CalcLdData(pTonalityMatrix[maxPosT][maxPosF - 1]) + + RELAXATION_LD64); + tmp = (tmp >> (DFRACT_BITS - 1 - LD_DATA_SHIFT - 1)) + + (FIXP_DBL)1; /* shift one bit less for rounding */ compValue = ((INT)(LONG)tmp) >> 1; - /* limit the comp-value*/ - if (compValue > maxComp) - compValue = maxComp; + /* limit the comp-value*/ + if (compValue > maxComp) compValue = maxComp; - pEnvComp[scfBand-1] = compValue; - } - } + pEnvComp[scfBand - 1] = compValue; + } + } } /* * Same as above, but for the upper end of the scalefactor-band. ***************************************************************/ - if(maxPosF == lu-1 && scfBand+1 < nSfb){ /* Upper border*/ - if(!pAddHarmSfb[scfBand + 1]) { - if (pSignMatrix[maxPosT][maxPosF] > 0 && pSignMatrix[maxPosT][maxPosF + 1] < 0) { - tmp = fixp_abs((FIXP_DBL)CalcLdData(pTonalityMatrix[maxPosT][maxPosF + 1]) + RELAXATION_LD64); - tmp = (tmp >> (DFRACT_BITS-1-LD_DATA_SHIFT-1)) + (FIXP_DBL)1; /* shift one bit less for rounding */ + if (maxPosF == lu - 1 && scfBand + 1 < nSfb) { /* Upper border*/ + if (!pAddHarmSfb[scfBand + 1]) { + if (pSignMatrix[maxPosT][maxPosF] > 0 && + pSignMatrix[maxPosT][maxPosF + 1] < 0) { + tmp = fixp_abs( + (FIXP_DBL)CalcLdData(pTonalityMatrix[maxPosT][maxPosF + 1]) + + RELAXATION_LD64); + tmp = (tmp >> (DFRACT_BITS - 1 - LD_DATA_SHIFT - 1)) + + (FIXP_DBL)1; /* shift one bit less for rounding */ compValue = ((INT)(LONG)tmp) >> 1; - if (compValue > maxComp) - compValue = maxComp; + if (compValue > maxComp) compValue = maxComp; - pEnvComp[scfBand+1] = compValue; - } - } + pEnvComp[scfBand + 1] = compValue; + } + } } - } - } + } + } - if(newDetectionAllowed == 0){ - for(scfBand=0;scfBand<nSfb;scfBand++){ - if(pEnvComp[scfBand] != 0 && pPrevEnvComp[scfBand] == 0) + if (newDetectionAllowed == 0) { + for (scfBand = 0; scfBand < nSfb; scfBand++) { + if (pEnvComp[scfBand] != 0 && pPrevEnvComp[scfBand] == 0) pEnvComp[scfBand] = 0; } } /* remember the value for the next frame.*/ - FDKmemcpy(pPrevEnvComp,pEnvComp,nSfb*sizeof(UCHAR)); + FDKmemcpy(pPrevEnvComp, pEnvComp, nSfb * sizeof(UCHAR)); } - /**************************************************************************/ /*! \brief Detects where strong tonal components will be missing after @@ -1071,34 +1012,26 @@ static void calculateCompVector(UCHAR *pAddHarmSfb, */ /**************************************************************************/ -void -FDKsbrEnc_SbrMissingHarmonicsDetectorQmf(HANDLE_SBR_MISSING_HARMONICS_DETECTOR h_sbrMHDet, - FIXP_DBL ** pQuotaBuffer, - INT ** pSignBuffer, - SCHAR* indexVector, - const SBR_FRAME_INFO *pFrameInfo, - const UCHAR* pTranInfo, - INT* pAddHarmonicsFlag, - UCHAR* pAddHarmonicsScaleFactorBands, - const UCHAR* freqBandTable, - INT nSfb, - UCHAR* envelopeCompensation, - FIXP_DBL *pNrgVector) -{ +void FDKsbrEnc_SbrMissingHarmonicsDetectorQmf( + HANDLE_SBR_MISSING_HARMONICS_DETECTOR h_sbrMHDet, FIXP_DBL **pQuotaBuffer, + INT **pSignBuffer, SCHAR *indexVector, const SBR_FRAME_INFO *pFrameInfo, + const UCHAR *pTranInfo, INT *pAddHarmonicsFlag, + UCHAR *pAddHarmonicsScaleFactorBands, const UCHAR *freqBandTable, INT nSfb, + UCHAR *envelopeCompensation, FIXP_DBL *pNrgVector) { INT transientFlag = pTranInfo[1]; - INT transientPos = pTranInfo[0]; + INT transientPos = pTranInfo[0]; INT newDetectionAllowed; INT transientDetStart = 0; - UCHAR ** detectionVectors = h_sbrMHDet->detectionVectors; - INT move = h_sbrMHDet->move; - INT noEstPerFrame = h_sbrMHDet->noEstPerFrame; - INT totNoEst = h_sbrMHDet->totNoEst; - INT prevTransientFlag = h_sbrMHDet->previousTransientFlag; - INT prevTransientFrame = h_sbrMHDet->previousTransientFrame; - INT transientPosOffset = h_sbrMHDet->transientPosOffset; - INT prevTransientPos = h_sbrMHDet->previousTransientPos; - GUIDE_VECTORS* guideVectors = h_sbrMHDet->guideVectors; + UCHAR **detectionVectors = h_sbrMHDet->detectionVectors; + INT move = h_sbrMHDet->move; + INT noEstPerFrame = h_sbrMHDet->noEstPerFrame; + INT totNoEst = h_sbrMHDet->totNoEst; + INT prevTransientFlag = h_sbrMHDet->previousTransientFlag; + INT prevTransientFrame = h_sbrMHDet->previousTransientFrame; + INT transientPosOffset = h_sbrMHDet->transientPosOffset; + INT prevTransientPos = h_sbrMHDet->previousTransientPos; + GUIDE_VECTORS *guideVectors = h_sbrMHDet->guideVectors; INT deltaTime = h_sbrMHDet->mhParams->deltaTime; INT maxComp = h_sbrMHDet->mhParams->maxComp; @@ -1107,96 +1040,70 @@ FDKsbrEnc_SbrMissingHarmonicsDetectorQmf(HANDLE_SBR_MISSING_HARMONICS_DETECTOR h /* Buffer values. */ - FDK_ASSERT(move<=(MAX_NO_OF_ESTIMATES>>1)); - FDK_ASSERT(noEstPerFrame<=(MAX_NO_OF_ESTIMATES>>1)); + FDK_ASSERT(move <= (MAX_NO_OF_ESTIMATES >> 1)); + FDK_ASSERT(noEstPerFrame <= (MAX_NO_OF_ESTIMATES >> 1)); FIXP_DBL *sfmSbr[MAX_NO_OF_ESTIMATES]; FIXP_DBL *sfmOrig[MAX_NO_OF_ESTIMATES]; FIXP_DBL *tonalityDiff[MAX_NO_OF_ESTIMATES]; - for (est=0; est < MAX_NO_OF_ESTIMATES/2; est++) { - sfmSbr[est] = h_sbrMHDet->sfmSbr[est]; - sfmOrig[est] = h_sbrMHDet->sfmOrig[est]; + for (est = 0; est < MAX_NO_OF_ESTIMATES / 2; est++) { + sfmSbr[est] = h_sbrMHDet->sfmSbr[est]; + sfmOrig[est] = h_sbrMHDet->sfmOrig[est]; tonalityDiff[est] = h_sbrMHDet->tonalityDiff[est]; } - C_ALLOC_SCRATCH_START(scratch_mem, FIXP_DBL, (3*MAX_NO_OF_ESTIMATES/2*MAX_FREQ_COEFFS)); - FIXP_DBL *scratch = scratch_mem; + C_ALLOC_SCRATCH_START(_scratch, FIXP_DBL, + 3 * MAX_NO_OF_ESTIMATES / 2 * MAX_FREQ_COEFFS) + FIXP_DBL *scratch = _scratch; for (; est < MAX_NO_OF_ESTIMATES; est++) { - sfmSbr[est] = scratch; scratch+=MAX_FREQ_COEFFS; - sfmOrig[est] = scratch; scratch+=MAX_FREQ_COEFFS; - tonalityDiff[est] = scratch; scratch+=MAX_FREQ_COEFFS; + sfmSbr[est] = scratch; + scratch += MAX_FREQ_COEFFS; + sfmOrig[est] = scratch; + scratch += MAX_FREQ_COEFFS; + tonalityDiff[est] = scratch; + scratch += MAX_FREQ_COEFFS; } + /* Determine if we're allowed to detect "missing harmonics" that wasn't + detected before. In order to be allowed to do new detection, there must be + a transient in the current frame, or a transient in the previous frame + sufficiently close to the current frame. */ + newDetectionAllowed = isDetectionOfNewToneAllowed( + pFrameInfo, &transientDetStart, noEstPerFrame, prevTransientFrame, + prevTransientPos, prevTransientFlag, transientPosOffset, transientFlag, + transientPos, deltaTime, h_sbrMHDet); - - /* Determine if we're allowed to detect "missing harmonics" that wasn't detected before. - In order to be allowed to do new detection, there must be a transient in the current - frame, or a transient in the previous frame sufficiently close to the current frame. */ - newDetectionAllowed = isDetectionOfNewToneAllowed(pFrameInfo, - &transientDetStart, - noEstPerFrame, - prevTransientFrame, - prevTransientPos, - prevTransientFlag, - transientPosOffset, - transientFlag, - transientPos, - deltaTime, - h_sbrMHDet); - - /* Calulate the variables that will be used subsequently for the actual detection */ - calculateDetectorInput(pQuotaBuffer, - indexVector, - tonalityDiff, - sfmOrig, - sfmSbr, - freqBandTable, - nSfb, - noEstPerFrame, - move); + /* Calulate the variables that will be used subsequently for the actual + * detection */ + calculateDetectorInput(pQuotaBuffer, indexVector, tonalityDiff, sfmOrig, + sfmSbr, freqBandTable, nSfb, noEstPerFrame, move); /* Do the actual detection using information from previous detections */ - detectionWithPrediction(pQuotaBuffer, - tonalityDiff, - pSignBuffer, - nSfb, - freqBandTable, - sfmOrig, - sfmSbr, - detectionVectors, - h_sbrMHDet->guideScfb, - guideVectors, - noEstPerFrame, - transientDetStart, - totNoEst, - newDetectionAllowed, - pAddHarmonicsFlag, - pAddHarmonicsScaleFactorBands, - pNrgVector, - h_sbrMHDet->mhParams); + detectionWithPrediction(pQuotaBuffer, tonalityDiff, pSignBuffer, nSfb, + freqBandTable, sfmOrig, sfmSbr, detectionVectors, + h_sbrMHDet->guideScfb, guideVectors, noEstPerFrame, + transientDetStart, totNoEst, newDetectionAllowed, + pAddHarmonicsFlag, pAddHarmonicsScaleFactorBands, + pNrgVector, h_sbrMHDet->mhParams); /* Calculate the comp vector, so that the energy can be compensated for a sine between two QMF-bands. */ - calculateCompVector(pAddHarmonicsScaleFactorBands, - pQuotaBuffer, - pSignBuffer, - envelopeCompensation, - nSfb, - freqBandTable, - totNoEst, - maxComp, - h_sbrMHDet->prevEnvelopeCompensation, + calculateCompVector(pAddHarmonicsScaleFactorBands, pQuotaBuffer, pSignBuffer, + envelopeCompensation, nSfb, freqBandTable, totNoEst, + maxComp, h_sbrMHDet->prevEnvelopeCompensation, newDetectionAllowed); - for (est=0; est < move; est++) { - FDKmemcpy(tonalityDiff[est], tonalityDiff[est + noEstPerFrame], sizeof(FIXP_DBL)*MAX_FREQ_COEFFS); - FDKmemcpy(sfmOrig[est], sfmOrig[est + noEstPerFrame], sizeof(FIXP_DBL)*MAX_FREQ_COEFFS); - FDKmemcpy(sfmSbr[est], sfmSbr[est + noEstPerFrame], sizeof(FIXP_DBL)*MAX_FREQ_COEFFS); + for (est = 0; est < move; est++) { + FDKmemcpy(tonalityDiff[est], tonalityDiff[est + noEstPerFrame], + sizeof(FIXP_DBL) * MAX_FREQ_COEFFS); + FDKmemcpy(sfmOrig[est], sfmOrig[est + noEstPerFrame], + sizeof(FIXP_DBL) * MAX_FREQ_COEFFS); + FDKmemcpy(sfmSbr[est], sfmSbr[est + noEstPerFrame], + sizeof(FIXP_DBL) * MAX_FREQ_COEFFS); } - C_ALLOC_SCRATCH_END(scratch, FIXP_DBL, (3*MAX_NO_OF_ESTIMATES/2*MAX_FREQ_COEFFS)); - - + C_ALLOC_SCRATCH_END(_scratch, FIXP_DBL, + 3 * MAX_NO_OF_ESTIMATES / 2 * MAX_FREQ_COEFFS) } /**************************************************************************/ @@ -1208,34 +1115,48 @@ FDKsbrEnc_SbrMissingHarmonicsDetectorQmf(HANDLE_SBR_MISSING_HARMONICS_DETECTOR h */ /**************************************************************************/ -INT -FDKsbrEnc_CreateSbrMissingHarmonicsDetector ( - HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMHDet, - INT chan) -{ +INT FDKsbrEnc_CreateSbrMissingHarmonicsDetector( + HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMHDet, INT chan) { HANDLE_SBR_MISSING_HARMONICS_DETECTOR hs = hSbrMHDet; INT i; - UCHAR* detectionVectors = GetRam_Sbr_detectionVectors(chan); - UCHAR* guideVectorDetected = GetRam_Sbr_guideVectorDetected(chan); - FIXP_DBL* guideVectorDiff = GetRam_Sbr_guideVectorDiff(chan); - FIXP_DBL* guideVectorOrig = GetRam_Sbr_guideVectorOrig(chan); + UCHAR *detectionVectors = GetRam_Sbr_detectionVectors(chan); + UCHAR *guideVectorDetected = GetRam_Sbr_guideVectorDetected(chan); + FIXP_DBL *guideVectorDiff = GetRam_Sbr_guideVectorDiff(chan); + FIXP_DBL *guideVectorOrig = GetRam_Sbr_guideVectorOrig(chan); - FDKmemclear (hs,sizeof(SBR_MISSING_HARMONICS_DETECTOR)); + FDKmemclear(hs, sizeof(SBR_MISSING_HARMONICS_DETECTOR)); hs->prevEnvelopeCompensation = GetRam_Sbr_prevEnvelopeCompensation(chan); - hs->guideScfb = GetRam_Sbr_guideScfb(chan); + hs->guideScfb = GetRam_Sbr_guideScfb(chan); + + if ((NULL == detectionVectors) || (NULL == guideVectorDetected) || + (NULL == guideVectorDiff) || (NULL == guideVectorOrig) || + (NULL == hs->prevEnvelopeCompensation) || (NULL == hs->guideScfb)) { + goto bail; + } - for(i=0; i<MAX_NO_OF_ESTIMATES; i++) { - hs->guideVectors[i].guideVectorDiff = guideVectorDiff + (i*MAX_FREQ_COEFFS); - hs->guideVectors[i].guideVectorOrig = guideVectorOrig + (i*MAX_FREQ_COEFFS); - hs->detectionVectors[i] = detectionVectors + (i*MAX_FREQ_COEFFS); - hs->guideVectors[i].guideVectorDetected = guideVectorDetected + (i*MAX_FREQ_COEFFS); + for (i = 0; i < MAX_NO_OF_ESTIMATES; i++) { + hs->guideVectors[i].guideVectorDiff = + guideVectorDiff + (i * MAX_FREQ_COEFFS); + hs->guideVectors[i].guideVectorOrig = + guideVectorOrig + (i * MAX_FREQ_COEFFS); + hs->detectionVectors[i] = detectionVectors + (i * MAX_FREQ_COEFFS); + hs->guideVectors[i].guideVectorDetected = + guideVectorDetected + (i * MAX_FREQ_COEFFS); } return 0; -} +bail: + hs->guideVectors[0].guideVectorDiff = guideVectorDiff; + hs->guideVectors[0].guideVectorOrig = guideVectorOrig; + hs->detectionVectors[0] = detectionVectors; + hs->guideVectors[0].guideVectorDetected = guideVectorDetected; + + FDKsbrEnc_DeleteSbrMissingHarmonicsDetector(hs); + return -1; +} /**************************************************************************/ /*! @@ -1246,54 +1167,43 @@ FDKsbrEnc_CreateSbrMissingHarmonicsDetector ( */ /**************************************************************************/ -INT -FDKsbrEnc_InitSbrMissingHarmonicsDetector ( - HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMHDet, - INT sampleFreq, - INT frameSize, - INT nSfb, - INT qmfNoChannels, - INT totNoEst, - INT move, - INT noEstPerFrame, - UINT sbrSyntaxFlags - ) -{ +INT FDKsbrEnc_InitSbrMissingHarmonicsDetector( + HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMHDet, INT sampleFreq, + INT frameSize, INT nSfb, INT qmfNoChannels, INT totNoEst, INT move, + INT noEstPerFrame, UINT sbrSyntaxFlags) { HANDLE_SBR_MISSING_HARMONICS_DETECTOR hs = hSbrMHDet; int i; FDK_ASSERT(totNoEst <= MAX_NO_OF_ESTIMATES); - if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) - { - switch(frameSize){ - case 1024: - case 512: + if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { + switch (frameSize) { + case 1024: + case 512: hs->transientPosOffset = FRAME_MIDDLE_SLOT_512LD; - hs->timeSlots = 16; + hs->timeSlots = 16; break; - case 960: - case 480: + case 960: + case 480: hs->transientPosOffset = FRAME_MIDDLE_SLOT_512LD; - hs->timeSlots = 15; + hs->timeSlots = 15; break; - default: + default: return -1; } - } else - { - switch(frameSize){ - case 2048: - case 1024: + } else { + switch (frameSize) { + case 2048: + case 1024: hs->transientPosOffset = FRAME_MIDDLE_SLOT_2048; - hs->timeSlots = NUMBER_TIME_SLOTS_2048; + hs->timeSlots = NUMBER_TIME_SLOTS_2048; break; - case 1920: - case 960: + case 1920: + case 960: hs->transientPosOffset = FRAME_MIDDLE_SLOT_1920; - hs->timeSlots = NUMBER_TIME_SLOTS_1920; + hs->timeSlots = NUMBER_TIME_SLOTS_1920; break; - default: + default: return -1; } } @@ -1301,7 +1211,7 @@ FDKsbrEnc_InitSbrMissingHarmonicsDetector ( if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { hs->mhParams = ¶msAacLd; } else - hs->mhParams = ¶msAac; + hs->mhParams = ¶msAac; hs->qmfNoChannels = qmfNoChannels; hs->sampleFreq = sampleFreq; @@ -1311,22 +1221,25 @@ FDKsbrEnc_InitSbrMissingHarmonicsDetector ( hs->move = move; hs->noEstPerFrame = noEstPerFrame; - for(i=0; i<totNoEst; i++) { - FDKmemclear (hs->guideVectors[i].guideVectorDiff,sizeof(FIXP_DBL)*MAX_FREQ_COEFFS); - FDKmemclear (hs->guideVectors[i].guideVectorOrig,sizeof(FIXP_DBL)*MAX_FREQ_COEFFS); - FDKmemclear (hs->detectionVectors[i],sizeof(UCHAR)*MAX_FREQ_COEFFS); - FDKmemclear (hs->guideVectors[i].guideVectorDetected,sizeof(UCHAR)*MAX_FREQ_COEFFS); + for (i = 0; i < totNoEst; i++) { + FDKmemclear(hs->guideVectors[i].guideVectorDiff, + sizeof(FIXP_DBL) * MAX_FREQ_COEFFS); + FDKmemclear(hs->guideVectors[i].guideVectorOrig, + sizeof(FIXP_DBL) * MAX_FREQ_COEFFS); + FDKmemclear(hs->detectionVectors[i], sizeof(UCHAR) * MAX_FREQ_COEFFS); + FDKmemclear(hs->guideVectors[i].guideVectorDetected, + sizeof(UCHAR) * MAX_FREQ_COEFFS); } - //for(i=0; i<totNoEst/2; i++) { - for(i=0; i<MAX_NO_OF_ESTIMATES/2; i++) { - FDKmemclear (hs->tonalityDiff[i],sizeof(FIXP_DBL)*MAX_FREQ_COEFFS); - FDKmemclear (hs->sfmOrig[i],sizeof(FIXP_DBL)*MAX_FREQ_COEFFS); - FDKmemclear (hs->sfmSbr[i],sizeof(FIXP_DBL)*MAX_FREQ_COEFFS); + // for(i=0; i<totNoEst/2; i++) { + for (i = 0; i < MAX_NO_OF_ESTIMATES / 2; i++) { + FDKmemclear(hs->tonalityDiff[i], sizeof(FIXP_DBL) * MAX_FREQ_COEFFS); + FDKmemclear(hs->sfmOrig[i], sizeof(FIXP_DBL) * MAX_FREQ_COEFFS); + FDKmemclear(hs->sfmSbr[i], sizeof(FIXP_DBL) * MAX_FREQ_COEFFS); } - FDKmemclear ( hs->prevEnvelopeCompensation, sizeof(UCHAR)*MAX_FREQ_COEFFS); - FDKmemclear ( hs->guideScfb, sizeof(UCHAR)*MAX_FREQ_COEFFS); + FDKmemclear(hs->prevEnvelopeCompensation, sizeof(UCHAR) * MAX_FREQ_COEFFS); + FDKmemclear(hs->guideScfb, sizeof(UCHAR) * MAX_FREQ_COEFFS); hs->previousTransientFlag = 0; hs->previousTransientFrame = 0; @@ -1344,9 +1257,8 @@ FDKsbrEnc_InitSbrMissingHarmonicsDetector ( */ /**************************************************************************/ -void -FDKsbrEnc_DeleteSbrMissingHarmonicsDetector(HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMHDet) -{ +void FDKsbrEnc_DeleteSbrMissingHarmonicsDetector( + HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMHDet) { if (hSbrMHDet) { HANDLE_SBR_MISSING_HARMONICS_DETECTOR hs = hSbrMHDet; @@ -1356,7 +1268,6 @@ FDKsbrEnc_DeleteSbrMissingHarmonicsDetector(HANDLE_SBR_MISSING_HARMONICS_DETECTO FreeRam_Sbr_guideVectorOrig(&hs->guideVectors[0].guideVectorOrig); FreeRam_Sbr_prevEnvelopeCompensation(&hs->prevEnvelopeCompensation); FreeRam_Sbr_guideScfb(&hs->guideScfb); - } } @@ -1369,10 +1280,9 @@ FDKsbrEnc_DeleteSbrMissingHarmonicsDetector(HANDLE_SBR_MISSING_HARMONICS_DETECTO */ /**************************************************************************/ -INT -FDKsbrEnc_ResetSbrMissingHarmonicsDetector (HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMissingHarmonicsDetector, - INT nSfb) -{ +INT FDKsbrEnc_ResetSbrMissingHarmonicsDetector( + HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMissingHarmonicsDetector, + INT nSfb) { int i; FIXP_DBL tempGuide[MAX_FREQ_COEFFS]; UCHAR tempGuideInt[MAX_FREQ_COEFFS]; @@ -1381,91 +1291,106 @@ FDKsbrEnc_ResetSbrMissingHarmonicsDetector (HANDLE_SBR_MISSING_HARMONICS_DETECTO nSfbPrev = hSbrMissingHarmonicsDetector->nSfb; hSbrMissingHarmonicsDetector->nSfb = nSfb; - FDKmemcpy( tempGuideInt, hSbrMissingHarmonicsDetector->guideScfb, nSfbPrev * sizeof(UCHAR) ); + FDKmemcpy(tempGuideInt, hSbrMissingHarmonicsDetector->guideScfb, + nSfbPrev * sizeof(UCHAR)); - if ( nSfb > nSfbPrev ) { - for ( i = 0; i < (nSfb - nSfbPrev); i++ ) { + if (nSfb > nSfbPrev) { + for (i = 0; i < (nSfb - nSfbPrev); i++) { hSbrMissingHarmonicsDetector->guideScfb[i] = 0; } - for ( i = 0; i < nSfbPrev; i++ ) { - hSbrMissingHarmonicsDetector->guideScfb[i + (nSfb - nSfbPrev)] = tempGuideInt[i]; + for (i = 0; i < nSfbPrev; i++) { + hSbrMissingHarmonicsDetector->guideScfb[i + (nSfb - nSfbPrev)] = + tempGuideInt[i]; } - } - else { - for ( i = 0; i < nSfb; i++ ) { - hSbrMissingHarmonicsDetector->guideScfb[i] = tempGuideInt[i + (nSfbPrev-nSfb)]; + } else { + for (i = 0; i < nSfb; i++) { + hSbrMissingHarmonicsDetector->guideScfb[i] = + tempGuideInt[i + (nSfbPrev - nSfb)]; } } - FDKmemcpy ( tempGuide, hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDiff, nSfbPrev * sizeof(FIXP_DBL) ); + FDKmemcpy(tempGuide, + hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDiff, + nSfbPrev * sizeof(FIXP_DBL)); - if (nSfb > nSfbPrev ) { - for ( i = 0; i < (nSfb - nSfbPrev); i++ ) { - hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDiff[i] = FL2FXCONST_DBL(0.0f); + if (nSfb > nSfbPrev) { + for (i = 0; i < (nSfb - nSfbPrev); i++) { + hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDiff[i] = + FL2FXCONST_DBL(0.0f); } - for ( i = 0; i < nSfbPrev; i++ ) { - hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDiff[i + (nSfb - nSfbPrev)] = tempGuide[i]; + for (i = 0; i < nSfbPrev; i++) { + hSbrMissingHarmonicsDetector->guideVectors[0] + .guideVectorDiff[i + (nSfb - nSfbPrev)] = tempGuide[i]; } - } - else { - for ( i = 0; i < nSfb; i++ ) { - hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDiff[i] = tempGuide[i + (nSfbPrev-nSfb)]; + } else { + for (i = 0; i < nSfb; i++) { + hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDiff[i] = + tempGuide[i + (nSfbPrev - nSfb)]; } } - FDKmemcpy ( tempGuide, hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorOrig, nSfbPrev * sizeof(FIXP_DBL) ); + FDKmemcpy(tempGuide, + hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorOrig, + nSfbPrev * sizeof(FIXP_DBL)); - if ( nSfb > nSfbPrev ) { - for ( i = 0; i< (nSfb - nSfbPrev); i++ ) { - hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorOrig[i] = FL2FXCONST_DBL(0.0f); + if (nSfb > nSfbPrev) { + for (i = 0; i < (nSfb - nSfbPrev); i++) { + hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorOrig[i] = + FL2FXCONST_DBL(0.0f); } - for ( i = 0; i < nSfbPrev; i++ ) { - hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorOrig[i + (nSfb - nSfbPrev)] = tempGuide[i]; + for (i = 0; i < nSfbPrev; i++) { + hSbrMissingHarmonicsDetector->guideVectors[0] + .guideVectorOrig[i + (nSfb - nSfbPrev)] = tempGuide[i]; } - } - else { - for ( i = 0; i < nSfb; i++ ) { - hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorOrig[i] = tempGuide[i + (nSfbPrev-nSfb)]; + } else { + for (i = 0; i < nSfb; i++) { + hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorOrig[i] = + tempGuide[i + (nSfbPrev - nSfb)]; } } - FDKmemcpy ( tempGuideInt, hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDetected, nSfbPrev * sizeof(UCHAR) ); + FDKmemcpy(tempGuideInt, + hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDetected, + nSfbPrev * sizeof(UCHAR)); - if ( nSfb > nSfbPrev ) { - for ( i = 0; i < (nSfb - nSfbPrev); i++ ) { + if (nSfb > nSfbPrev) { + for (i = 0; i < (nSfb - nSfbPrev); i++) { hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDetected[i] = 0; } - for ( i = 0; i < nSfbPrev; i++ ) { - hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDetected[i + (nSfb - nSfbPrev)] = tempGuideInt[i]; + for (i = 0; i < nSfbPrev; i++) { + hSbrMissingHarmonicsDetector->guideVectors[0] + .guideVectorDetected[i + (nSfb - nSfbPrev)] = tempGuideInt[i]; } - } - else { - for ( i = 0; i < nSfb; i++ ) { - hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDetected[i] = tempGuideInt[i + (nSfbPrev-nSfb)]; + } else { + for (i = 0; i < nSfb; i++) { + hSbrMissingHarmonicsDetector->guideVectors[0].guideVectorDetected[i] = + tempGuideInt[i + (nSfbPrev - nSfb)]; } } - FDKmemcpy ( tempGuideInt, hSbrMissingHarmonicsDetector->prevEnvelopeCompensation, nSfbPrev * sizeof(UCHAR) ); + FDKmemcpy(tempGuideInt, + hSbrMissingHarmonicsDetector->prevEnvelopeCompensation, + nSfbPrev * sizeof(UCHAR)); - if ( nSfb > nSfbPrev ) { - for ( i = 0; i < (nSfb - nSfbPrev); i++ ) { + if (nSfb > nSfbPrev) { + for (i = 0; i < (nSfb - nSfbPrev); i++) { hSbrMissingHarmonicsDetector->prevEnvelopeCompensation[i] = 0; } - for ( i = 0; i < nSfbPrev; i++ ) { - hSbrMissingHarmonicsDetector->prevEnvelopeCompensation[i + (nSfb - nSfbPrev)] = tempGuideInt[i]; + for (i = 0; i < nSfbPrev; i++) { + hSbrMissingHarmonicsDetector + ->prevEnvelopeCompensation[i + (nSfb - nSfbPrev)] = tempGuideInt[i]; } - } - else { - for ( i = 0; i < nSfb; i++ ) { - hSbrMissingHarmonicsDetector->prevEnvelopeCompensation[i] = tempGuideInt[i + (nSfbPrev-nSfb)]; + } else { + for (i = 0; i < nSfb; i++) { + hSbrMissingHarmonicsDetector->prevEnvelopeCompensation[i] = + tempGuideInt[i + (nSfbPrev - nSfb)]; } } return 0; } - diff --git a/libSBRenc/src/mh_det.h b/libSBRenc/src/mh_det.h index 74c2a99..89d81b5 100644 --- a/libSBRenc/src/mh_det.h +++ b/libSBRenc/src/mh_det.h @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,53 +90,67 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ /*! \file - \brief missing harmonics detection header file + \brief missing harmonics detection header file $Revision: 92790 $ */ -#ifndef __MH_DETECT_H -#define __MH_DETECT_H +#ifndef MH_DET_H +#define MH_DET_H #include "sbr_encoder.h" #include "fram_gen.h" -typedef struct -{ +typedef struct { FIXP_DBL thresHoldDiff; /*!< threshold for tonality difference */ - FIXP_DBL thresHoldDiffGuide; /*!< threshold for tonality difference for the guide */ + FIXP_DBL thresHoldDiffGuide; /*!< threshold for tonality difference for the + guide */ FIXP_DBL thresHoldTone; /*!< threshold for tonality for a sine */ FIXP_DBL invThresHoldTone; - FIXP_DBL thresHoldToneGuide; /*!< threshold for tonality for a sine for the guide */ - FIXP_DBL sfmThresSbr; /*!< tonality flatness measure threshold for the SBR signal.*/ - FIXP_DBL sfmThresOrig; /*!< tonality flatness measure threshold for the original signal.*/ - FIXP_DBL decayGuideOrig; /*!< decay value of the tonality value of the guide for the tone. */ - FIXP_DBL decayGuideDiff; /*!< decay value of the tonality value of the guide for the tonality difference. */ - FIXP_DBL derivThresMaxLD64; /*!< threshold for detecting LP character in a signal. */ - FIXP_DBL derivThresBelowLD64; /*!< threshold for detecting LP character in a signal. */ - FIXP_DBL derivThresAboveLD64; /*!< threshold for detecting LP character in a signal. */ -}THRES_HOLDS; - -typedef struct -{ - INT deltaTime; /*!< maximum allowed transient distance (from frame border in number of qmf subband sample) - for a frame to be considered a transient frame.*/ - THRES_HOLDS thresHolds; /*!< the thresholds used for detection. */ - INT maxComp; /*!< maximum alllowed compensation factor for the envelope data. */ -}DETECTOR_PARAMETERS_MH; - -typedef struct -{ + FIXP_DBL thresHoldToneGuide; /*!< threshold for tonality for a sine for the + guide */ + FIXP_DBL sfmThresSbr; /*!< tonality flatness measure threshold for the SBR + signal.*/ + FIXP_DBL sfmThresOrig; /*!< tonality flatness measure threshold for the + original signal.*/ + FIXP_DBL decayGuideOrig; /*!< decay value of the tonality value of the guide + for the tone. */ + FIXP_DBL decayGuideDiff; /*!< decay value of the tonality value of the guide + for the tonality difference. */ + FIXP_DBL derivThresMaxLD64; /*!< threshold for detecting LP character in a + signal. */ + FIXP_DBL derivThresBelowLD64; /*!< threshold for detecting LP character in a + signal. */ + FIXP_DBL derivThresAboveLD64; /*!< threshold for detecting LP character in a + signal. */ +} THRES_HOLDS; + +typedef struct { + INT deltaTime; /*!< maximum allowed transient distance (from frame border in + number of qmf subband sample) for a frame to be considered a + transient frame.*/ + THRES_HOLDS thresHolds; /*!< the thresholds used for detection. */ + INT maxComp; /*!< maximum alllowed compensation factor for the envelope data. + */ +} DETECTOR_PARAMETERS_MH; + +typedef struct { FIXP_DBL *guideVectorDiff; FIXP_DBL *guideVectorOrig; - UCHAR* guideVectorDetected; -}GUIDE_VECTORS; + UCHAR *guideVectorDetected; +} GUIDE_VECTORS; - -typedef struct -{ +typedef struct { INT qmfNoChannels; INT nSfb; INT sampleFreq; @@ -144,53 +169,36 @@ typedef struct UCHAR *guideScfb; UCHAR *prevEnvelopeCompensation; UCHAR *detectionVectors[MAX_NO_OF_ESTIMATES]; - FIXP_DBL tonalityDiff[MAX_NO_OF_ESTIMATES/2][MAX_FREQ_COEFFS]; - FIXP_DBL sfmOrig[MAX_NO_OF_ESTIMATES/2][MAX_FREQ_COEFFS]; - FIXP_DBL sfmSbr[MAX_NO_OF_ESTIMATES/2][MAX_FREQ_COEFFS]; + FIXP_DBL tonalityDiff[MAX_NO_OF_ESTIMATES / 2][MAX_FREQ_COEFFS]; + FIXP_DBL sfmOrig[MAX_NO_OF_ESTIMATES / 2][MAX_FREQ_COEFFS]; + FIXP_DBL sfmSbr[MAX_NO_OF_ESTIMATES / 2][MAX_FREQ_COEFFS]; const DETECTOR_PARAMETERS_MH *mhParams; GUIDE_VECTORS guideVectors[MAX_NO_OF_ESTIMATES]; -} -SBR_MISSING_HARMONICS_DETECTOR; +} SBR_MISSING_HARMONICS_DETECTOR; typedef SBR_MISSING_HARMONICS_DETECTOR *HANDLE_SBR_MISSING_HARMONICS_DETECTOR; -void -FDKsbrEnc_SbrMissingHarmonicsDetectorQmf(HANDLE_SBR_MISSING_HARMONICS_DETECTOR h_sbrMissingHarmonicsDetector, - FIXP_DBL ** pQuotaBuffer, - INT ** pSignBuffer, - SCHAR *indexVector, - const SBR_FRAME_INFO *pFrameInfo, - const UCHAR* pTranInfo, - INT* pAddHarmonicsFlag, - UCHAR* pAddHarmonicsScaleFactorBands, - const UCHAR* freqBandTable, - INT nSfb, - UCHAR * envelopeCompensation, - FIXP_DBL *pNrgVector); - -INT -FDKsbrEnc_CreateSbrMissingHarmonicsDetector ( - HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMHDet, - INT chan); - -INT -FDKsbrEnc_InitSbrMissingHarmonicsDetector( - HANDLE_SBR_MISSING_HARMONICS_DETECTOR h_sbrMissingHarmonicsDetector, - INT sampleFreq, - INT frameSize, - INT nSfb, - INT qmfNoChannels, - INT totNoEst, - INT move, - INT noEstPerFrame, - UINT sbrSyntaxFlags); - -void -FDKsbrEnc_DeleteSbrMissingHarmonicsDetector (HANDLE_SBR_MISSING_HARMONICS_DETECTOR h_sbrMissingHarmonicsDetector); - - -INT -FDKsbrEnc_ResetSbrMissingHarmonicsDetector (HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMissingHarmonicsDetector, - INT nSfb); +void FDKsbrEnc_SbrMissingHarmonicsDetectorQmf( + HANDLE_SBR_MISSING_HARMONICS_DETECTOR h_sbrMissingHarmonicsDetector, + FIXP_DBL **pQuotaBuffer, INT **pSignBuffer, SCHAR *indexVector, + const SBR_FRAME_INFO *pFrameInfo, const UCHAR *pTranInfo, + INT *pAddHarmonicsFlag, UCHAR *pAddHarmonicsScaleFactorBands, + const UCHAR *freqBandTable, INT nSfb, UCHAR *envelopeCompensation, + FIXP_DBL *pNrgVector); + +INT FDKsbrEnc_CreateSbrMissingHarmonicsDetector( + HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMHDet, INT chan); + +INT FDKsbrEnc_InitSbrMissingHarmonicsDetector( + HANDLE_SBR_MISSING_HARMONICS_DETECTOR h_sbrMissingHarmonicsDetector, + INT sampleFreq, INT frameSize, INT nSfb, INT qmfNoChannels, INT totNoEst, + INT move, INT noEstPerFrame, UINT sbrSyntaxFlags); + +void FDKsbrEnc_DeleteSbrMissingHarmonicsDetector( + HANDLE_SBR_MISSING_HARMONICS_DETECTOR h_sbrMissingHarmonicsDetector); + +INT FDKsbrEnc_ResetSbrMissingHarmonicsDetector( + HANDLE_SBR_MISSING_HARMONICS_DETECTOR hSbrMissingHarmonicsDetector, + INT nSfb); #endif diff --git a/libSBRenc/src/nf_est.cpp b/libSBRenc/src/nf_est.cpp index a4c5574..290ec35 100644 --- a/libSBRenc/src/nf_est.cpp +++ b/libSBRenc/src/nf_est.cpp @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,7 +90,15 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ #include "nf_est.h" @@ -88,23 +107,22 @@ amm-info@iis.fraunhofer.de #include "genericStds.h" /* smoothFilter[4] = {0.05857864376269f, 0.2f, 0.34142135623731f, 0.4f}; */ -static const FIXP_DBL smoothFilter[4] = { 0x077f813d, 0x19999995, 0x2bb3b1f5, 0x33333335 }; +static const FIXP_DBL smoothFilter[4] = {0x077f813d, 0x19999995, 0x2bb3b1f5, + 0x33333335}; /* static const INT smoothFilterLength = 4; */ -static const FIXP_DBL QuantOffset = (INT)0xfc000000; /* ld64(0.25) */ +static const FIXP_DBL QuantOffset = (INT)0xfc000000; /* ld64(0.25) */ #ifndef min -#define min(a,b) ( a < b ? a:b) +#define min(a, b) (a < b ? a : b) #endif #ifndef max -#define max(a,b) ( a > b ? a:b) +#define max(a, b) (a > b ? a : b) #endif -#define NOISE_FLOOR_OFFSET_SCALING (4) - - +#define NOISE_FLOOR_OFFSET_SCALING (4) /**************************************************************************/ /*! @@ -116,38 +134,45 @@ static const FIXP_DBL QuantOffset = (INT)0xfc000000; /* ld64(0.25) */ */ /**************************************************************************/ -static void -smoothingOfNoiseLevels(FIXP_DBL *NoiseLevels, /*!< pointer to noise-floor levels.*/ - INT nEnvelopes, /*!< Number of noise floor envelopes.*/ - INT noNoiseBands, /*!< Number of noise bands for every noise floor envelope. */ - FIXP_DBL prevNoiseLevels[NF_SMOOTHING_LENGTH][MAX_NUM_NOISE_VALUES],/*!< Previous noise floor envelopes. */ - const FIXP_DBL *smoothFilter, /*!< filter used for smoothing the noise floor levels. */ - INT transientFlag) /*!< flag indicating if a transient is present*/ +static void smoothingOfNoiseLevels( + FIXP_DBL *NoiseLevels, /*!< pointer to noise-floor levels.*/ + INT nEnvelopes, /*!< Number of noise floor envelopes.*/ + INT noNoiseBands, /*!< Number of noise bands for every noise floor envelope. + */ + FIXP_DBL prevNoiseLevels[NF_SMOOTHING_LENGTH] + [MAX_NUM_NOISE_VALUES], /*!< Previous noise floor + envelopes. */ + const FIXP_DBL * + pSmoothFilter, /*!< filter used for smoothing the noise floor levels. */ + INT transientFlag) /*!< flag indicating if a transient is present*/ { - INT i,band,env; + INT i, band, env; FIXP_DBL accu; - for(env = 0; env < nEnvelopes; env++){ - if(transientFlag){ - for (i = 0; i < NF_SMOOTHING_LENGTH; i++){ - FDKmemcpy(prevNoiseLevels[i],NoiseLevels+env*noNoiseBands,noNoiseBands*sizeof(FIXP_DBL)); + for (env = 0; env < nEnvelopes; env++) { + if (transientFlag) { + for (i = 0; i < NF_SMOOTHING_LENGTH; i++) { + FDKmemcpy(prevNoiseLevels[i], NoiseLevels + env * noNoiseBands, + noNoiseBands * sizeof(FIXP_DBL)); } - } - else { - for (i = 1; i < NF_SMOOTHING_LENGTH; i++){ - FDKmemcpy(prevNoiseLevels[i - 1],prevNoiseLevels[i],noNoiseBands*sizeof(FIXP_DBL)); + } else { + for (i = 1; i < NF_SMOOTHING_LENGTH; i++) { + FDKmemcpy(prevNoiseLevels[i - 1], prevNoiseLevels[i], + noNoiseBands * sizeof(FIXP_DBL)); } - FDKmemcpy(prevNoiseLevels[NF_SMOOTHING_LENGTH - 1],NoiseLevels+env*noNoiseBands,noNoiseBands*sizeof(FIXP_DBL)); + FDKmemcpy(prevNoiseLevels[NF_SMOOTHING_LENGTH - 1], + NoiseLevels + env * noNoiseBands, + noNoiseBands * sizeof(FIXP_DBL)); } - for (band = 0; band < noNoiseBands; band++){ + for (band = 0; band < noNoiseBands; band++) { accu = FL2FXCONST_DBL(0.0f); - for (i = 0; i < NF_SMOOTHING_LENGTH; i++){ - accu += fMultDiv2(smoothFilter[i], prevNoiseLevels[i][band]); + for (i = 0; i < NF_SMOOTHING_LENGTH; i++) { + accu += fMultDiv2(pSmoothFilter[i], prevNoiseLevels[i][band]); } - FDK_ASSERT( (band + env*noNoiseBands) < MAX_NUM_NOISE_VALUES); - NoiseLevels[band+ env*noNoiseBands] = accu<<1; + FDK_ASSERT((band + env * noNoiseBands) < MAX_NUM_NOISE_VALUES); + NoiseLevels[band + env * noNoiseBands] = accu << 1; } } } @@ -162,92 +187,100 @@ smoothingOfNoiseLevels(FIXP_DBL *NoiseLevels, /*!< pointer to noise-floor */ /**************************************************************************/ -static void -qmfBasedNoiseFloorDetection(FIXP_DBL *noiseLevel, /*!< Pointer to vector to store the noise levels in.*/ - FIXP_DBL ** quotaMatrixOrig, /*!< Matrix holding the quota values of the original. */ - SCHAR *indexVector, /*!< Index vector to obtain the patched data. */ - INT startIndex, /*!< Start index. */ - INT stopIndex, /*!< Stop index. */ - INT startChannel, /*!< Start channel of the current noise floor band.*/ - INT stopChannel, /*!< Stop channel of the current noise floor band. */ - FIXP_DBL ana_max_level, /*!< Maximum level of the adaptive noise.*/ - FIXP_DBL noiseFloorOffset, /*!< Noise floor offset. */ - INT missingHarmonicFlag, /*!< Flag indicating if a strong tonal component is missing.*/ - FIXP_DBL weightFac, /*!< Weightening factor for the difference between orig and sbr. */ - INVF_MODE diffThres, /*!< Threshold value to control the inverse filtering decision.*/ - INVF_MODE inverseFilteringLevel) /*!< Inverse filtering level of the current band.*/ +static void qmfBasedNoiseFloorDetection( + FIXP_DBL *noiseLevel, /*!< Pointer to vector to + store the noise levels + in.*/ + FIXP_DBL **quotaMatrixOrig, /*!< Matrix holding the quota + values of the original. */ + SCHAR *indexVector, /*!< Index vector to obtain the + patched data. */ + INT startIndex, /*!< Start index. */ + INT stopIndex, /*!< Stop index. */ + INT startChannel, /*!< Start channel of the current + noise floor band.*/ + INT stopChannel, /*!< Stop channel of the current + noise floor band. */ + FIXP_DBL ana_max_level, /*!< Maximum level of the + adaptive noise.*/ + FIXP_DBL noiseFloorOffset, /*!< Noise floor offset. */ + INT missingHarmonicFlag, /*!< Flag indicating if a + strong tonal component + is missing.*/ + FIXP_DBL weightFac, /*!< Weightening factor for the + difference between orig and sbr. + */ + INVF_MODE diffThres, /*!< Threshold value to control the + inverse filtering decision.*/ + INVF_MODE inverseFilteringLevel) /*!< Inverse filtering + level of the current + band.*/ { INT scale, l, k; - FIXP_DBL meanOrig=FL2FXCONST_DBL(0.0f), meanSbr=FL2FXCONST_DBL(0.0f), diff; - FIXP_DBL invIndex = GetInvInt(stopIndex-startIndex); - FIXP_DBL invChannel = GetInvInt(stopChannel-startChannel); + FIXP_DBL meanOrig = FL2FXCONST_DBL(0.0f), meanSbr = FL2FXCONST_DBL(0.0f), + diff; + FIXP_DBL invIndex = GetInvInt(stopIndex - startIndex); + FIXP_DBL invChannel = GetInvInt(stopChannel - startChannel); FIXP_DBL accu; - /* - Calculate the mean value, over the current time segment, for the original, the HFR - and the difference, over all channels in the current frequency range. - */ + /* + Calculate the mean value, over the current time segment, for the original, the + HFR and the difference, over all channels in the current frequency range. + */ - if(missingHarmonicFlag == 1){ - for(l = startChannel; l < stopChannel;l++){ + if (missingHarmonicFlag == 1) { + for (l = startChannel; l < stopChannel; l++) { /* tonalityOrig */ accu = FL2FXCONST_DBL(0.0f); - for(k = startIndex ; k < stopIndex; k++){ + for (k = startIndex; k < stopIndex; k++) { accu += fMultDiv2(quotaMatrixOrig[k][l], invIndex); } - meanOrig = fixMax(meanOrig,(accu<<1)); + meanOrig = fixMax(meanOrig, (accu << 1)); /* tonalitySbr */ accu = FL2FXCONST_DBL(0.0f); - for(k = startIndex ; k < stopIndex; k++){ + for (k = startIndex; k < stopIndex; k++) { accu += fMultDiv2(quotaMatrixOrig[k][indexVector[l]], invIndex); } - meanSbr = fixMax(meanSbr,(accu<<1)); - + meanSbr = fixMax(meanSbr, (accu << 1)); } - } - else{ - for(l = startChannel; l < stopChannel;l++){ + } else { + for (l = startChannel; l < stopChannel; l++) { /* tonalityOrig */ accu = FL2FXCONST_DBL(0.0f); - for(k = startIndex ; k < stopIndex; k++){ + for (k = startIndex; k < stopIndex; k++) { accu += fMultDiv2(quotaMatrixOrig[k][l], invIndex); } - meanOrig += fMult((accu<<1), invChannel); + meanOrig += fMult((accu << 1), invChannel); /* tonalitySbr */ accu = FL2FXCONST_DBL(0.0f); - for(k = startIndex ; k < stopIndex; k++){ + for (k = startIndex; k < stopIndex; k++) { accu += fMultDiv2(quotaMatrixOrig[k][indexVector[l]], invIndex); } - meanSbr += fMult((accu<<1), invChannel); + meanSbr += fMult((accu << 1), invChannel); } } /* Small fix to avoid noise during silent passages.*/ - if( meanOrig <= FL2FXCONST_DBL(0.000976562f*RELAXATION_FLOAT) && - meanSbr <= FL2FXCONST_DBL(0.000976562f*RELAXATION_FLOAT) ) - { - meanOrig = FL2FXCONST_DBL(101.5936673f*RELAXATION_FLOAT); - meanSbr = FL2FXCONST_DBL(101.5936673f*RELAXATION_FLOAT); + if (meanOrig <= FL2FXCONST_DBL(0.000976562f * RELAXATION_FLOAT) && + meanSbr <= FL2FXCONST_DBL(0.000976562f * RELAXATION_FLOAT)) { + meanOrig = FL2FXCONST_DBL(101.5936673f * RELAXATION_FLOAT); + meanSbr = FL2FXCONST_DBL(101.5936673f * RELAXATION_FLOAT); } - meanOrig = fixMax(meanOrig,RELAXATION); - meanSbr = fixMax(meanSbr,RELAXATION); + meanOrig = fixMax(meanOrig, RELAXATION); + meanSbr = fixMax(meanSbr, RELAXATION); - if (missingHarmonicFlag == 1 || - inverseFilteringLevel == INVF_MID_LEVEL || + if (missingHarmonicFlag == 1 || inverseFilteringLevel == INVF_MID_LEVEL || inverseFilteringLevel == INVF_LOW_LEVEL || - inverseFilteringLevel == INVF_OFF || - inverseFilteringLevel <= diffThres) - { + inverseFilteringLevel == INVF_OFF || inverseFilteringLevel <= diffThres) { diff = RELAXATION; - } - else { + } else { accu = fDivNorm(meanSbr, meanOrig, &scale); - diff = fixMax( RELAXATION, - fMult(RELAXATION_FRACT,fMult(weightFac,accu)) >>( RELAXATION_SHIFT-scale ) ) ; + diff = fixMax(RELAXATION, fMult(RELAXATION_FRACT, fMult(weightFac, accu)) >> + (RELAXATION_SHIFT - scale)); } /* @@ -258,24 +291,27 @@ qmfBasedNoiseFloorDetection(FIXP_DBL *noiseLevel, /*!< Pointer to v accu = fDivNorm(diff, meanOrig, &scale); scale -= 2; - if ( (scale>0) && (accu > ((FIXP_DBL)MAXVAL_DBL)>>scale) ) { + if ((scale > 0) && (accu > ((FIXP_DBL)MAXVAL_DBL) >> scale)) { *noiseLevel = (FIXP_DBL)MAXVAL_DBL; - } - else { + } else { *noiseLevel = scaleValue(accu, scale); } /* * Add a noise floor offset to compensate for bias in the detector *****************************************************************/ - if(!missingHarmonicFlag) { - *noiseLevel = fixMin(fMult(*noiseLevel, noiseFloorOffset), (FIXP_DBL)MAXVAL_DBL>>NOISE_FLOOR_OFFSET_SCALING) << NOISE_FLOOR_OFFSET_SCALING; + if (!missingHarmonicFlag) { + *noiseLevel = fixMin(fMult(*noiseLevel, noiseFloorOffset), + (FIXP_DBL)MAXVAL_DBL >> NOISE_FLOOR_OFFSET_SCALING) + << NOISE_FLOOR_OFFSET_SCALING; } /* * check to see that we don't exceed the maximum allowed level **************************************************************/ - *noiseLevel = fixMin(*noiseLevel, ana_max_level); /* ana_max_level is scaled with factor 0.25 */ + *noiseLevel = + fixMin(*noiseLevel, + ana_max_level); /* ana_max_level is scaled with factor 0.25 */ } /**************************************************************************/ @@ -290,85 +326,78 @@ qmfBasedNoiseFloorDetection(FIXP_DBL *noiseLevel, /*!< Pointer to v */ /**************************************************************************/ -void -FDKsbrEnc_sbrNoiseFloorEstimateQmf(HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ - const SBR_FRAME_INFO *frame_info, /*!< Time frequency grid of the current frame. */ - FIXP_DBL *noiseLevels, /*!< Pointer to vector to store the noise levels in.*/ - FIXP_DBL **quotaMatrixOrig, /*!< Matrix holding the quota values of the original. */ - SCHAR *indexVector, /*!< Index vector to obtain the patched data. */ - INT missingHarmonicsFlag, /*!< Flag indicating if a strong tonal component will be missing. */ - INT startIndex, /*!< Start index. */ - UINT numberOfEstimatesPerFrame, /*!< The number of tonality estimates per frame. */ - int transientFrame, /*!< A flag indicating if a transient is present. */ - INVF_MODE* pInvFiltLevels, /*!< Pointer to the vector holding the inverse filtering levels. */ - UINT sbrSyntaxFlags - ) +void FDKsbrEnc_sbrNoiseFloorEstimateQmf( + HANDLE_SBR_NOISE_FLOOR_ESTIMATE + h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct + */ + const SBR_FRAME_INFO + *frame_info, /*!< Time frequency grid of the current frame. */ + FIXP_DBL + *noiseLevels, /*!< Pointer to vector to store the noise levels in.*/ + FIXP_DBL **quotaMatrixOrig, /*!< Matrix holding the quota values of the + original. */ + SCHAR *indexVector, /*!< Index vector to obtain the patched data. */ + INT missingHarmonicsFlag, /*!< Flag indicating if a strong tonal component + will be missing. */ + INT startIndex, /*!< Start index. */ + UINT numberOfEstimatesPerFrame, /*!< The number of tonality estimates per + frame. */ + int transientFrame, /*!< A flag indicating if a transient is present. */ + INVF_MODE *pInvFiltLevels, /*!< Pointer to the vector holding the inverse + filtering levels. */ + UINT sbrSyntaxFlags) { - INT nNoiseEnvelopes, startPos[2], stopPos[2], env, band; - INT noNoiseBands = h_sbrNoiseFloorEstimate->noNoiseBands; - INT *freqBandTable = h_sbrNoiseFloorEstimate->freqBandTableQmf; + INT noNoiseBands = h_sbrNoiseFloorEstimate->noNoiseBands; + INT *freqBandTable = h_sbrNoiseFloorEstimate->freqBandTableQmf; nNoiseEnvelopes = frame_info->nNoiseEnvelopes; - if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { - nNoiseEnvelopes = 1; - startPos[0] = startIndex; - stopPos[0] = startIndex + min(numberOfEstimatesPerFrame,2); - } else - if(nNoiseEnvelopes == 1){ - startPos[0] = startIndex; - stopPos[0] = startIndex + 2; - } - else{ - startPos[0] = startIndex; - stopPos[0] = startIndex + 1; + startPos[0] = startIndex; + + if (nNoiseEnvelopes == 1) { + stopPos[0] = startIndex + min(numberOfEstimatesPerFrame, 2); + } else { + stopPos[0] = startIndex + 1; startPos[1] = startIndex + 1; - stopPos[1] = startIndex + 2; + stopPos[1] = startIndex + min(numberOfEstimatesPerFrame, 2); } /* * Estimate the noise floor. **************************************/ - for(env = 0; env < nNoiseEnvelopes; env++){ - for(band = 0; band < noNoiseBands; band++){ - FDK_ASSERT( (band + env*noNoiseBands) < MAX_NUM_NOISE_VALUES); - qmfBasedNoiseFloorDetection(&noiseLevels[band + env*noNoiseBands], - quotaMatrixOrig, - indexVector, - startPos[env], - stopPos[env], - freqBandTable[band], - freqBandTable[band+1], - h_sbrNoiseFloorEstimate->ana_max_level, - h_sbrNoiseFloorEstimate->noiseFloorOffset[band], - missingHarmonicsFlag, - h_sbrNoiseFloorEstimate->weightFac, - h_sbrNoiseFloorEstimate->diffThres, - pInvFiltLevels[band]); + for (env = 0; env < nNoiseEnvelopes; env++) { + for (band = 0; band < noNoiseBands; band++) { + FDK_ASSERT((band + env * noNoiseBands) < MAX_NUM_NOISE_VALUES); + qmfBasedNoiseFloorDetection( + &noiseLevels[band + env * noNoiseBands], quotaMatrixOrig, indexVector, + startPos[env], stopPos[env], freqBandTable[band], + freqBandTable[band + 1], h_sbrNoiseFloorEstimate->ana_max_level, + h_sbrNoiseFloorEstimate->noiseFloorOffset[band], missingHarmonicsFlag, + h_sbrNoiseFloorEstimate->weightFac, + h_sbrNoiseFloorEstimate->diffThres, pInvFiltLevels[band]); } } - /* * Smoothing of the values. **************************/ - smoothingOfNoiseLevels(noiseLevels, - nNoiseEnvelopes, + smoothingOfNoiseLevels(noiseLevels, nNoiseEnvelopes, h_sbrNoiseFloorEstimate->noNoiseBands, h_sbrNoiseFloorEstimate->prevNoiseLevels, - h_sbrNoiseFloorEstimate->smoothFilter, - transientFrame); - + h_sbrNoiseFloorEstimate->smoothFilter, transientFrame); /* quantisation*/ - for(env = 0; env < nNoiseEnvelopes; env++){ - for(band = 0; band < noNoiseBands; band++){ - FDK_ASSERT( (band + env*noNoiseBands) < MAX_NUM_NOISE_VALUES); - noiseLevels[band + env*noNoiseBands] = - (FIXP_DBL)NOISE_FLOOR_OFFSET_64 - (FIXP_DBL)CalcLdData(noiseLevels[band + env*noNoiseBands]+(FIXP_DBL)1) + QuantOffset; + for (env = 0; env < nNoiseEnvelopes; env++) { + for (band = 0; band < noNoiseBands; band++) { + FDK_ASSERT((band + env * noNoiseBands) < MAX_NUM_NOISE_VALUES); + noiseLevels[band + env * noNoiseBands] = + (FIXP_DBL)NOISE_FLOOR_OFFSET_64 - + (FIXP_DBL)CalcLdData(noiseLevels[band + env * noNoiseBands] + + (FIXP_DBL)1) + + QuantOffset; } } } @@ -382,39 +411,39 @@ FDKsbrEnc_sbrNoiseFloorEstimateQmf(HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFlo */ /**************************************************************************/ -static INT -downSampleLoRes(INT *v_result, /*!< */ - INT num_result, /*!< */ - const UCHAR *freqBandTableRef,/*!< */ - INT num_Ref) /*!< */ +static INT downSampleLoRes(INT *v_result, /*!< */ + INT num_result, /*!< */ + const UCHAR *freqBandTableRef, /*!< */ + INT num_Ref) /*!< */ { INT step; - INT i,j; - INT org_length,result_length; - INT v_index[MAX_FREQ_COEFFS/2]; + INT i, j; + INT org_length, result_length; + INT v_index[MAX_FREQ_COEFFS / 2]; /* 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; /* floor; */ - org_length=org_length - step; - result_length--; - v_index[i]=v_index[i-1]+step; - } + org_length = num_Ref; + result_length = num_result; - if(i != num_result ) /* Should never happen */ - return (1);/* error downsampling */ + v_index[0] = 0; /* Always use left border */ + i = 0; + while (org_length > 0) /* Create downsample vector */ + { + i++; + step = org_length / result_length; /* floor; */ + 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]]; - } + if (i != num_result) /* Should never happen */ + return (1); /* error downsampling */ + + for (j = 0; j <= i; + j++) /* Use downsample vector to index LoResolution vector. */ + { + v_result[j] = freqBandTableRef[v_index[j]]; + } return (0); } @@ -428,48 +457,48 @@ downSampleLoRes(INT *v_result, /*!< */ */ /**************************************************************************/ -INT -FDKsbrEnc_InitSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ - INT ana_max_level, /*!< Maximum level of the adaptive noise. */ - const UCHAR *freqBandTable, /*!< Frequany band table. */ - INT nSfb, /*!< Number of frequency bands. */ - INT noiseBands, /*!< Number of noise bands per octave. */ - INT noiseFloorOffset, /*!< Noise floor offset. */ - INT timeSlots, /*!< Number of time slots in a frame. */ - UINT useSpeechConfig /*!< Flag: adapt tuning parameters according to speech */ - ) -{ +INT FDKsbrEnc_InitSbrNoiseFloorEstimate( + HANDLE_SBR_NOISE_FLOOR_ESTIMATE + h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct + */ + INT ana_max_level, /*!< Maximum level of the adaptive noise. */ + const UCHAR *freqBandTable, /*!< Frequency band table. */ + INT nSfb, /*!< Number of frequency bands. */ + INT noiseBands, /*!< Number of noise bands per octave. */ + INT noiseFloorOffset, /*!< Noise floor offset. */ + INT timeSlots, /*!< Number of time slots in a frame. */ + UINT useSpeechConfig /*!< Flag: adapt tuning parameters according to speech + */ +) { INT i, qexp, qtmp; FIXP_DBL tmp, exp; - FDKmemclear(h_sbrNoiseFloorEstimate,sizeof(SBR_NOISE_FLOOR_ESTIMATE)); + FDKmemclear(h_sbrNoiseFloorEstimate, sizeof(SBR_NOISE_FLOOR_ESTIMATE)); h_sbrNoiseFloorEstimate->smoothFilter = smoothFilter; if (useSpeechConfig) { h_sbrNoiseFloorEstimate->weightFac = (FIXP_DBL)MAXVAL_DBL; h_sbrNoiseFloorEstimate->diffThres = INVF_LOW_LEVEL; - } - else { + } else { h_sbrNoiseFloorEstimate->weightFac = FL2FXCONST_DBL(0.25f); h_sbrNoiseFloorEstimate->diffThres = INVF_MID_LEVEL; } - h_sbrNoiseFloorEstimate->timeSlots = timeSlots; - h_sbrNoiseFloorEstimate->noiseBands = noiseBands; + h_sbrNoiseFloorEstimate->timeSlots = timeSlots; + h_sbrNoiseFloorEstimate->noiseBands = noiseBands; /* h_sbrNoiseFloorEstimate->ana_max_level is scaled by 0.25 */ - switch(ana_max_level) - { - case 6: + switch (ana_max_level) { + case 6: h_sbrNoiseFloorEstimate->ana_max_level = (FIXP_DBL)MAXVAL_DBL; break; - case 3: + case 3: h_sbrNoiseFloorEstimate->ana_max_level = FL2FXCONST_DBL(0.5); break; - case -3: + case -3: h_sbrNoiseFloorEstimate->ana_max_level = FL2FXCONST_DBL(0.125); break; - default: + default: /* Should not enter here */ h_sbrNoiseFloorEstimate->ana_max_level = (FIXP_DBL)MAXVAL_DBL; break; @@ -478,26 +507,26 @@ FDKsbrEnc_InitSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoise /* calculate number of noise bands and allocate */ - if(FDKsbrEnc_resetSbrNoiseFloorEstimate(h_sbrNoiseFloorEstimate,freqBandTable,nSfb)) - return(1); + if (FDKsbrEnc_resetSbrNoiseFloorEstimate(h_sbrNoiseFloorEstimate, + freqBandTable, nSfb)) + return (1); - if(noiseFloorOffset == 0) { - tmp = ((FIXP_DBL)MAXVAL_DBL)>>NOISE_FLOOR_OFFSET_SCALING; - } - else { + if (noiseFloorOffset == 0) { + tmp = ((FIXP_DBL)MAXVAL_DBL) >> NOISE_FLOOR_OFFSET_SCALING; + } else { /* noiseFloorOffset has to be smaller than 12, because the result of the calculation below must be smaller than 1: (2^(noiseFloorOffset/3))*2^4<1 */ - FDK_ASSERT(noiseFloorOffset<12); + FDK_ASSERT(noiseFloorOffset < 12); /* Assumes the noise floor offset in tuning table are in q31 */ /* Change the qformat here when non-zero values would be filled */ exp = fDivNorm((FIXP_DBL)noiseFloorOffset, 3, &qexp); - tmp = fPow(2, DFRACT_BITS-1, exp, qexp, &qtmp); - tmp = scaleValue(tmp, qtmp-NOISE_FLOOR_OFFSET_SCALING); + tmp = fPow(2, DFRACT_BITS - 1, exp, qexp, &qtmp); + tmp = scaleValue(tmp, qtmp - NOISE_FLOOR_OFFSET_SCALING); } - for(i=0;i<h_sbrNoiseFloorEstimate->noNoiseBands;i++) { + for (i = 0; i < h_sbrNoiseFloorEstimate->noNoiseBands; i++) { h_sbrNoiseFloorEstimate->noiseFloorOffset[i] = tmp; } @@ -514,52 +543,50 @@ FDKsbrEnc_InitSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoise */ /**************************************************************************/ -INT -FDKsbrEnc_resetSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ - const UCHAR *freqBandTable, /*!< Frequany band table. */ - INT nSfb) /*!< Number of bands in the frequency band table. */ -{ - INT k2,kx; +INT FDKsbrEnc_resetSbrNoiseFloorEstimate( + HANDLE_SBR_NOISE_FLOOR_ESTIMATE + h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct + */ + const UCHAR *freqBandTable, /*!< Frequency band table. */ + INT nSfb /*!< Number of bands in the frequency band table. */ +) { + INT k2, kx; + /* + * Calculate number of noise bands + ***********************************/ + k2 = freqBandTable[nSfb]; + kx = freqBandTable[0]; + if (h_sbrNoiseFloorEstimate->noiseBands == 0) { + h_sbrNoiseFloorEstimate->noNoiseBands = 1; + } else { /* - * Calculate number of noise bands - ***********************************/ - k2=freqBandTable[nSfb]; - kx=freqBandTable[0]; - if(h_sbrNoiseFloorEstimate->noiseBands == 0){ - h_sbrNoiseFloorEstimate->noNoiseBands = 1; - } - else{ - /* - * Calculate number of noise bands 1,2 or 3 bands/octave - ********************************************************/ - FIXP_DBL tmp, ratio, lg2; - INT ratio_e, qlg2, nNoiseBands; - - ratio = fDivNorm(k2, kx, &ratio_e); - lg2 = fLog2(ratio, ratio_e, &qlg2); - tmp = fMult((FIXP_DBL)(h_sbrNoiseFloorEstimate->noiseBands<<24), lg2); - tmp = scaleValue(tmp, qlg2-23); + * Calculate number of noise bands 1,2 or 3 bands/octave + ********************************************************/ + FIXP_DBL tmp, ratio, lg2; + INT ratio_e, qlg2, nNoiseBands; - nNoiseBands = (INT)((tmp + (FIXP_DBL)1) >> 1); + ratio = fDivNorm(k2, kx, &ratio_e); + lg2 = fLog2(ratio, ratio_e, &qlg2); + tmp = fMult((FIXP_DBL)(h_sbrNoiseFloorEstimate->noiseBands << 24), lg2); + tmp = scaleValue(tmp, qlg2 - 23); + nNoiseBands = (INT)((tmp + (FIXP_DBL)1) >> 1); - if (nNoiseBands > MAX_NUM_NOISE_COEFFS ) { - nNoiseBands = MAX_NUM_NOISE_COEFFS; - } - - if( nNoiseBands == 0 ) { - nNoiseBands = 1; - } - - h_sbrNoiseFloorEstimate->noNoiseBands = nNoiseBands; + if (nNoiseBands > MAX_NUM_NOISE_COEFFS) { + nNoiseBands = MAX_NUM_NOISE_COEFFS; + } + if (nNoiseBands == 0) { + nNoiseBands = 1; } + h_sbrNoiseFloorEstimate->noNoiseBands = nNoiseBands; + } - return(downSampleLoRes(h_sbrNoiseFloorEstimate->freqBandTableQmf, - h_sbrNoiseFloorEstimate->noNoiseBands, - freqBandTable,nSfb)); + return (downSampleLoRes(h_sbrNoiseFloorEstimate->freqBandTableQmf, + h_sbrNoiseFloorEstimate->noNoiseBands, freqBandTable, + nSfb)); } /**************************************************************************/ @@ -572,10 +599,11 @@ FDKsbrEnc_resetSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoise */ /**************************************************************************/ -void -FDKsbrEnc_deleteSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate) /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ +void FDKsbrEnc_deleteSbrNoiseFloorEstimate( + HANDLE_SBR_NOISE_FLOOR_ESTIMATE + h_sbrNoiseFloorEstimate) /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct + */ { - if (h_sbrNoiseFloorEstimate) { /* nothing to do diff --git a/libSBRenc/src/nf_est.h b/libSBRenc/src/nf_est.h index f26f74f..c2f16e9 100644 --- a/libSBRenc/src/nf_est.h +++ b/libSBRenc/src/nf_est.h @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,69 +90,96 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ /*! \file - \brief Noise floor estimation structs and prototypes + \brief Noise floor estimation structs and prototypes $Revision: 92790 $ */ -#ifndef __NF_EST_H -#define __NF_EST_H +#ifndef NF_EST_H +#define NF_EST_H #include "sbr_encoder.h" #include "fram_gen.h" -#define NF_SMOOTHING_LENGTH 4 /*!< Smoothing length of the noise floors. */ - -typedef struct -{ - FIXP_DBL prevNoiseLevels[NF_SMOOTHING_LENGTH][MAX_NUM_NOISE_VALUES]; /*!< The previous noise levels. */ - FIXP_DBL noiseFloorOffset[MAX_NUM_NOISE_VALUES]; /*!< Noise floor offset, scaled with NOISE_FLOOR_OFFSET_SCALING */ - const FIXP_DBL *smoothFilter; /*!< Smoothing filter to use. */ - FIXP_DBL ana_max_level; /*!< Max level allowed. */ - FIXP_DBL weightFac; /*!< Weightening factor for the difference between orig and sbr. */ - INT freqBandTableQmf[MAX_NUM_NOISE_VALUES + 1]; /*!< Frequncy band table for the noise floor bands.*/ - INT noNoiseBands; /*!< Number of noisebands. */ - INT noiseBands; /*!< NoiseBands switch 4 bit.*/ - INT timeSlots; /*!< Number of timeslots in a frame. */ - INVF_MODE diffThres; /*!< Threshold value to control the inverse filtering decision */ -} -SBR_NOISE_FLOOR_ESTIMATE; +#define NF_SMOOTHING_LENGTH 4 /*!< Smoothing length of the noise floors. */ + +typedef struct { + FIXP_DBL + prevNoiseLevels[NF_SMOOTHING_LENGTH] + [MAX_NUM_NOISE_VALUES]; /*!< The previous noise levels. */ + FIXP_DBL noiseFloorOffset + [MAX_NUM_NOISE_VALUES]; /*!< Noise floor offset, scaled with + NOISE_FLOOR_OFFSET_SCALING */ + const FIXP_DBL *smoothFilter; /*!< Smoothing filter to use. */ + FIXP_DBL ana_max_level; /*!< Max level allowed. */ + FIXP_DBL weightFac; /*!< Weightening factor for the difference between orig + and sbr. */ + INT freqBandTableQmf[MAX_NUM_NOISE_VALUES + + 1]; /*!< Frequncy band table for the noise floor bands.*/ + INT noNoiseBands; /*!< Number of noisebands. */ + INT noiseBands; /*!< NoiseBands switch 4 bit.*/ + INT timeSlots; /*!< Number of timeslots in a frame. */ + INVF_MODE diffThres; /*!< Threshold value to control the inverse filtering + decision */ +} SBR_NOISE_FLOOR_ESTIMATE; typedef SBR_NOISE_FLOOR_ESTIMATE *HANDLE_SBR_NOISE_FLOOR_ESTIMATE; -void -FDKsbrEnc_sbrNoiseFloorEstimateQmf(HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ - const SBR_FRAME_INFO *frame_info, /*!< Time frequency grid of the current frame. */ - FIXP_DBL *noiseLevels, /*!< Pointer to vector to store the noise levels in.*/ - FIXP_DBL **quotaMatrixOrig, /*!< Matrix holding the quota values of the original. */ - SCHAR* indexVector, /*!< Index vector to obtain the patched data. */ - INT missingHarmonicsFlag, /*!< Flag indicating if a strong tonal component will be missing. */ - INT startIndex, /*!< Start index. */ - UINT numberOfEstimatesPerFrame, /*!< The number of tonality estimates per frame. */ - INT transientFrame, /*!< A flag indicating if a transient is present. */ - INVF_MODE* pInvFiltLevels, /*!< Pointer to the vector holding the inverse filtering levels. */ - UINT sbrSyntaxFlags - ); - -INT -FDKsbrEnc_InitSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ - INT ana_max_level, /*!< Maximum level of the adaptive noise. */ - const UCHAR *freqBandTable, /*!< Frequany band table. */ - INT nSfb, /*!< Number of frequency bands. */ - INT noiseBands, /*!< Number of noise bands per octave. */ - INT noiseFloorOffset, /*!< Noise floor offset. */ - INT timeSlots, /*!< Number of time slots in a frame. */ - UINT useSpeechConfig /*!< Flag: adapt tuning parameters according to speech */ - ); - -INT -FDKsbrEnc_resetSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ - const UCHAR *freqBandTable, /*!< Frequany band table. */ - INT nSfb); /*!< Number of bands in the frequency band table. */ - -void -FDKsbrEnc_deleteSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate); /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */ +void FDKsbrEnc_sbrNoiseFloorEstimateQmf( + HANDLE_SBR_NOISE_FLOOR_ESTIMATE + h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct + */ + const SBR_FRAME_INFO + *frame_info, /*!< Time frequency grid of the current frame. */ + FIXP_DBL + *noiseLevels, /*!< Pointer to vector to store the noise levels in.*/ + FIXP_DBL **quotaMatrixOrig, /*!< Matrix holding the quota values of the + original. */ + SCHAR *indexVector, /*!< Index vector to obtain the patched data. */ + INT missingHarmonicsFlag, /*!< Flag indicating if a strong tonal component + will be missing. */ + INT startIndex, /*!< Start index. */ + UINT numberOfEstimatesPerFrame, /*!< The number of tonality estimates per + frame. */ + INT transientFrame, /*!< A flag indicating if a transient is present. */ + INVF_MODE *pInvFiltLevels, /*!< Pointer to the vector holding the inverse + filtering levels. */ + UINT sbrSyntaxFlags); + +INT FDKsbrEnc_InitSbrNoiseFloorEstimate( + HANDLE_SBR_NOISE_FLOOR_ESTIMATE + h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct + */ + INT ana_max_level, /*!< Maximum level of the adaptive noise. */ + const UCHAR *freqBandTable, /*!< Frequany band table. */ + INT nSfb, /*!< Number of frequency bands. */ + INT noiseBands, /*!< Number of noise bands per octave. */ + INT noiseFloorOffset, /*!< Noise floor offset. */ + INT timeSlots, /*!< Number of time slots in a frame. */ + UINT useSpeechConfig /*!< Flag: adapt tuning parameters according to speech + */ +); + +INT FDKsbrEnc_resetSbrNoiseFloorEstimate( + HANDLE_SBR_NOISE_FLOOR_ESTIMATE + h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct + */ + const UCHAR *freqBandTable, /*!< Frequany band table. */ + INT nSfb); /*!< Number of bands in the frequency band table. */ + +void FDKsbrEnc_deleteSbrNoiseFloorEstimate( + HANDLE_SBR_NOISE_FLOOR_ESTIMATE + h_sbrNoiseFloorEstimate); /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct + */ #endif diff --git a/libSBRenc/src/ps_bitenc.cpp b/libSBRenc/src/ps_bitenc.cpp index 420ea15..e30af2a 100644 --- a/libSBRenc/src/ps_bitenc.cpp +++ b/libSBRenc/src/ps_bitenc.cpp @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,37 +90,35 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ - -/***************************** MPEG Audio Encoder *************************** +----------------------------------------------------------------------------- */ - Initial author: N. Rettelbach - contents/description: Parametric Stereo bitstream encoder +/**************************** SBR encoder library ****************************** -******************************************************************************/ + Author(s): N. Rettelbach -#include "ps_main.h" + Description: Parametric Stereo bitstream encoder +*******************************************************************************/ -#include "ps_const.h" #include "ps_bitenc.h" -static -inline UCHAR FDKsbrEnc_WriteBits_ps(HANDLE_FDK_BITSTREAM hBitStream, UINT value, - const UINT numberOfBits) -{ +#include "ps_main.h" + +static inline UCHAR FDKsbrEnc_WriteBits_ps(HANDLE_FDK_BITSTREAM hBitStream, + UINT value, + const UINT numberOfBits) { /* hBitStream == NULL happens here intentionally */ - if(hBitStream!=NULL){ + if (hBitStream != NULL) { FDKwriteBits(hBitStream, value, numberOfBits); } return numberOfBits; } -#define SI_SBR_EXTENSION_SIZE_BITS 4 -#define SI_SBR_EXTENSION_ESC_COUNT_BITS 8 -#define SI_SBR_EXTENSION_ID_BITS 2 -#define EXTENSION_ID_PS_CODING 2 -#define PS_EXT_ID_V0 0 +#define SI_SBR_EXTENSION_SIZE_BITS 4 +#define SI_SBR_EXTENSION_ESC_COUNT_BITS 8 +#define SI_SBR_EXTENSION_ID_BITS 2 +#define EXTENSION_ID_PS_CODING 2 +#define PS_EXT_ID_V0 0 static const INT iidDeltaCoarse_Offset = 14; static const INT iidDeltaCoarse_MaxVal = 28; @@ -117,499 +126,425 @@ static const INT iidDeltaFine_Offset = 30; static const INT iidDeltaFine_MaxVal = 60; /* PS Stereo Huffmantable: iidDeltaFreqCoarse */ -static const UINT iidDeltaFreqCoarse_Length[] = -{ - 17, 17, 17, 17, 16, 15, 13, 10, 9, 7, - 6, 5, 4, 3, 1, 3, 4, 5, 6, 6, - 8, 11, 13, 14, 14, 15, 17, 18, 18 -}; -static const UINT iidDeltaFreqCoarse_Code[] = -{ - 0x0001fffb, 0x0001fffc, 0x0001fffd, 0x0001fffa, 0x0000fffc, 0x00007ffc, 0x00001ffd, 0x000003fe, 0x000001fe, 0x0000007e, - 0x0000003c, 0x0000001d, 0x0000000d, 0x00000005, 0000000000, 0x00000004, 0x0000000c, 0x0000001c, 0x0000003d, 0x0000003e, - 0x000000fe, 0x000007fe, 0x00001ffc, 0x00003ffc, 0x00003ffd, 0x00007ffd, 0x0001fffe, 0x0003fffe, 0x0003ffff -}; +static const UINT iidDeltaFreqCoarse_Length[] = { + 17, 17, 17, 17, 16, 15, 13, 10, 9, 7, 6, 5, 4, 3, 1, + 3, 4, 5, 6, 6, 8, 11, 13, 14, 14, 15, 17, 18, 18}; +static const UINT iidDeltaFreqCoarse_Code[] = { + 0x0001fffb, 0x0001fffc, 0x0001fffd, 0x0001fffa, 0x0000fffc, 0x00007ffc, + 0x00001ffd, 0x000003fe, 0x000001fe, 0x0000007e, 0x0000003c, 0x0000001d, + 0x0000000d, 0x00000005, 0000000000, 0x00000004, 0x0000000c, 0x0000001c, + 0x0000003d, 0x0000003e, 0x000000fe, 0x000007fe, 0x00001ffc, 0x00003ffc, + 0x00003ffd, 0x00007ffd, 0x0001fffe, 0x0003fffe, 0x0003ffff}; /* PS Stereo Huffmantable: iidDeltaFreqFine */ -static const UINT iidDeltaFreqFine_Length[] = -{ - 18, 18, 18, 18, 18, 18, 18, 18, 18, 17, - 18, 17, 17, 16, 16, 15, 14, 14, 13, 12, - 12, 11, 10, 10, 8, 7, 6, 5, 4, 3, - 1, 3, 4, 5, 6, 7, 8, 9, 10, 11, - 11, 12, 13, 14, 14, 15, 16, 16, 17, 17, - 18, 17, 18, 18, 18, 18, 18, 18, 18, 18, - 18 -}; -static const UINT iidDeltaFreqFine_Code[] = -{ - 0x0001feb4, 0x0001feb5, 0x0001fd76, 0x0001fd77, 0x0001fd74, 0x0001fd75, 0x0001fe8a, 0x0001fe8b, 0x0001fe88, 0x0000fe80, - 0x0001feb6, 0x0000fe82, 0x0000feb8, 0x00007f42, 0x00007fae, 0x00003faf, 0x00001fd1, 0x00001fe9, 0x00000fe9, 0x000007ea, - 0x000007fb, 0x000003fb, 0x000001fb, 0x000001ff, 0x0000007c, 0x0000003c, 0x0000001c, 0x0000000c, 0000000000, 0x00000001, - 0x00000001, 0x00000002, 0x00000001, 0x0000000d, 0x0000001d, 0x0000003d, 0x0000007d, 0x000000fc, 0x000001fc, 0x000003fc, - 0x000003f4, 0x000007eb, 0x00000fea, 0x00001fea, 0x00001fd6, 0x00003fd0, 0x00007faf, 0x00007f43, 0x0000feb9, 0x0000fe83, - 0x0001feb7, 0x0000fe81, 0x0001fe89, 0x0001fe8e, 0x0001fe8f, 0x0001fe8c, 0x0001fe8d, 0x0001feb2, 0x0001feb3, 0x0001feb0, - 0x0001feb1 -}; +static const UINT iidDeltaFreqFine_Length[] = { + 18, 18, 18, 18, 18, 18, 18, 18, 18, 17, 18, 17, 17, 16, 16, 15, + 14, 14, 13, 12, 12, 11, 10, 10, 8, 7, 6, 5, 4, 3, 1, 3, + 4, 5, 6, 7, 8, 9, 10, 11, 11, 12, 13, 14, 14, 15, 16, 16, + 17, 17, 18, 17, 18, 18, 18, 18, 18, 18, 18, 18, 18}; +static const UINT iidDeltaFreqFine_Code[] = { + 0x0001feb4, 0x0001feb5, 0x0001fd76, 0x0001fd77, 0x0001fd74, 0x0001fd75, + 0x0001fe8a, 0x0001fe8b, 0x0001fe88, 0x0000fe80, 0x0001feb6, 0x0000fe82, + 0x0000feb8, 0x00007f42, 0x00007fae, 0x00003faf, 0x00001fd1, 0x00001fe9, + 0x00000fe9, 0x000007ea, 0x000007fb, 0x000003fb, 0x000001fb, 0x000001ff, + 0x0000007c, 0x0000003c, 0x0000001c, 0x0000000c, 0000000000, 0x00000001, + 0x00000001, 0x00000002, 0x00000001, 0x0000000d, 0x0000001d, 0x0000003d, + 0x0000007d, 0x000000fc, 0x000001fc, 0x000003fc, 0x000003f4, 0x000007eb, + 0x00000fea, 0x00001fea, 0x00001fd6, 0x00003fd0, 0x00007faf, 0x00007f43, + 0x0000feb9, 0x0000fe83, 0x0001feb7, 0x0000fe81, 0x0001fe89, 0x0001fe8e, + 0x0001fe8f, 0x0001fe8c, 0x0001fe8d, 0x0001feb2, 0x0001feb3, 0x0001feb0, + 0x0001feb1}; /* PS Stereo Huffmantable: iidDeltaTimeCoarse */ -static const UINT iidDeltaTimeCoarse_Length[] = -{ - 19, 19, 19, 20, 20, 20, 17, 15, 12, 10, - 8, 6, 4, 2, 1, 3, 5, 7, 9, 11, - 13, 14, 17, 19, 20, 20, 20, 20, 20 -}; -static const UINT iidDeltaTimeCoarse_Code[] = -{ - 0x0007fff9, 0x0007fffa, 0x0007fffb, 0x000ffff8, 0x000ffff9, 0x000ffffa, 0x0001fffd, 0x00007ffe, 0x00000ffe, 0x000003fe, - 0x000000fe, 0x0000003e, 0x0000000e, 0x00000002, 0000000000, 0x00000006, 0x0000001e, 0x0000007e, 0x000001fe, 0x000007fe, - 0x00001ffe, 0x00003ffe, 0x0001fffc, 0x0007fff8, 0x000ffffb, 0x000ffffc, 0x000ffffd, 0x000ffffe, 0x000fffff -}; +static const UINT iidDeltaTimeCoarse_Length[] = { + 19, 19, 19, 20, 20, 20, 17, 15, 12, 10, 8, 6, 4, 2, 1, + 3, 5, 7, 9, 11, 13, 14, 17, 19, 20, 20, 20, 20, 20}; +static const UINT iidDeltaTimeCoarse_Code[] = { + 0x0007fff9, 0x0007fffa, 0x0007fffb, 0x000ffff8, 0x000ffff9, 0x000ffffa, + 0x0001fffd, 0x00007ffe, 0x00000ffe, 0x000003fe, 0x000000fe, 0x0000003e, + 0x0000000e, 0x00000002, 0000000000, 0x00000006, 0x0000001e, 0x0000007e, + 0x000001fe, 0x000007fe, 0x00001ffe, 0x00003ffe, 0x0001fffc, 0x0007fff8, + 0x000ffffb, 0x000ffffc, 0x000ffffd, 0x000ffffe, 0x000fffff}; /* PS Stereo Huffmantable: iidDeltaTimeFine */ -static const UINT iidDeltaTimeFine_Length[] = -{ - 16, 16, 16, 16, 16, 16, 16, 16, 16, 15, - 15, 15, 15, 15, 15, 14, 14, 13, 13, 13, - 12, 12, 11, 10, 9, 9, 7, 6, 5, 3, - 1, 2, 5, 6, 7, 8, 9, 10, 11, 11, - 12, 12, 13, 13, 14, 14, 15, 15, 15, 15, - 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, - 16 -}; -static const UINT iidDeltaTimeFine_Code[] = -{ - 0x00004ed4, 0x00004ed5, 0x00004ece, 0x00004ecf, 0x00004ecc, 0x00004ed6, 0x00004ed8, 0x00004f46, 0x00004f60, 0x00002718, - 0x00002719, 0x00002764, 0x00002765, 0x0000276d, 0x000027b1, 0x000013b7, 0x000013d6, 0x000009c7, 0x000009e9, 0x000009ed, - 0x000004ee, 0x000004f7, 0x00000278, 0x00000139, 0x0000009a, 0x0000009f, 0x00000020, 0x00000011, 0x0000000a, 0x00000003, - 0x00000001, 0000000000, 0x0000000b, 0x00000012, 0x00000021, 0x0000004c, 0x0000009b, 0x0000013a, 0x00000279, 0x00000270, - 0x000004ef, 0x000004e2, 0x000009ea, 0x000009d8, 0x000013d7, 0x000013d0, 0x000027b2, 0x000027a2, 0x0000271a, 0x0000271b, - 0x00004f66, 0x00004f67, 0x00004f61, 0x00004f47, 0x00004ed9, 0x00004ed7, 0x00004ecd, 0x00004ed2, 0x00004ed3, 0x00004ed0, - 0x00004ed1 -}; - -static const INT iccDelta_Offset = 7; +static const UINT iidDeltaTimeFine_Length[] = { + 16, 16, 16, 16, 16, 16, 16, 16, 16, 15, 15, 15, 15, 15, 15, 14, + 14, 13, 13, 13, 12, 12, 11, 10, 9, 9, 7, 6, 5, 3, 1, 2, + 5, 6, 7, 8, 9, 10, 11, 11, 12, 12, 13, 13, 14, 14, 15, 15, + 15, 15, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16}; +static const UINT iidDeltaTimeFine_Code[] = { + 0x00004ed4, 0x00004ed5, 0x00004ece, 0x00004ecf, 0x00004ecc, 0x00004ed6, + 0x00004ed8, 0x00004f46, 0x00004f60, 0x00002718, 0x00002719, 0x00002764, + 0x00002765, 0x0000276d, 0x000027b1, 0x000013b7, 0x000013d6, 0x000009c7, + 0x000009e9, 0x000009ed, 0x000004ee, 0x000004f7, 0x00000278, 0x00000139, + 0x0000009a, 0x0000009f, 0x00000020, 0x00000011, 0x0000000a, 0x00000003, + 0x00000001, 0000000000, 0x0000000b, 0x00000012, 0x00000021, 0x0000004c, + 0x0000009b, 0x0000013a, 0x00000279, 0x00000270, 0x000004ef, 0x000004e2, + 0x000009ea, 0x000009d8, 0x000013d7, 0x000013d0, 0x000027b2, 0x000027a2, + 0x0000271a, 0x0000271b, 0x00004f66, 0x00004f67, 0x00004f61, 0x00004f47, + 0x00004ed9, 0x00004ed7, 0x00004ecd, 0x00004ed2, 0x00004ed3, 0x00004ed0, + 0x00004ed1}; + +static const INT iccDelta_Offset = 7; static const INT iccDelta_MaxVal = 14; /* PS Stereo Huffmantable: iccDeltaFreq */ -static const UINT iccDeltaFreq_Length[] = -{ - 14, 14, 12, 10, 7, 5, 3, 1, 2, 4, - 6, 8, 9, 11, 13 -}; -static const UINT iccDeltaFreq_Code[] = -{ - 0x00003fff, 0x00003ffe, 0x00000ffe, 0x000003fe, 0x0000007e, 0x0000001e, 0x00000006, 0000000000, 0x00000002, 0x0000000e, - 0x0000003e, 0x000000fe, 0x000001fe, 0x000007fe, 0x00001ffe -}; +static const UINT iccDeltaFreq_Length[] = {14, 14, 12, 10, 7, 5, 3, 1, + 2, 4, 6, 8, 9, 11, 13}; +static const UINT iccDeltaFreq_Code[] = { + 0x00003fff, 0x00003ffe, 0x00000ffe, 0x000003fe, 0x0000007e, + 0x0000001e, 0x00000006, 0000000000, 0x00000002, 0x0000000e, + 0x0000003e, 0x000000fe, 0x000001fe, 0x000007fe, 0x00001ffe}; /* PS Stereo Huffmantable: iccDeltaTime */ -static const UINT iccDeltaTime_Length[] = -{ - 14, 13, 11, 9, 7, 5, 3, 1, 2, 4, - 6, 8, 10, 12, 14 -}; -static const UINT iccDeltaTime_Code[] = -{ - 0x00003ffe, 0x00001ffe, 0x000007fe, 0x000001fe, 0x0000007e, 0x0000001e, 0x00000006, 0000000000, 0x00000002, 0x0000000e, - 0x0000003e, 0x000000fe, 0x000003fe, 0x00000ffe, 0x00003fff -}; - - +static const UINT iccDeltaTime_Length[] = {14, 13, 11, 9, 7, 5, 3, 1, + 2, 4, 6, 8, 10, 12, 14}; +static const UINT iccDeltaTime_Code[] = { + 0x00003ffe, 0x00001ffe, 0x000007fe, 0x000001fe, 0x0000007e, + 0x0000001e, 0x00000006, 0000000000, 0x00000002, 0x0000000e, + 0x0000003e, 0x000000fe, 0x000003fe, 0x00000ffe, 0x00003fff}; static const INT ipdDelta_Offset = 0; static const INT ipdDelta_MaxVal = 7; /* PS Stereo Huffmantable: ipdDeltaFreq */ -static const UINT ipdDeltaFreq_Length[] = -{ - 1, 3, 4, 4, 4, 4, 4, 4 -}; -static const UINT ipdDeltaFreq_Code[] = -{ - 0x00000001, 0000000000, 0x00000006, 0x00000004, 0x00000002, 0x00000003, 0x00000005, 0x00000007 -}; +static const UINT ipdDeltaFreq_Length[] = {1, 3, 4, 4, 4, 4, 4, 4}; +static const UINT ipdDeltaFreq_Code[] = {0x00000001, 0000000000, 0x00000006, + 0x00000004, 0x00000002, 0x00000003, + 0x00000005, 0x00000007}; /* PS Stereo Huffmantable: ipdDeltaTime */ -static const UINT ipdDeltaTime_Length[] = -{ - 1, 3, 4, 5, 5, 4, 4, 3 -}; -static const UINT ipdDeltaTime_Code[] = -{ - 0x00000001, 0x00000002, 0x00000002, 0x00000003, 0x00000002, 0000000000, 0x00000003, 0x00000003 -}; - +static const UINT ipdDeltaTime_Length[] = {1, 3, 4, 5, 5, 4, 4, 3}; +static const UINT ipdDeltaTime_Code[] = {0x00000001, 0x00000002, 0x00000002, + 0x00000003, 0x00000002, 0000000000, + 0x00000003, 0x00000003}; static const INT opdDelta_Offset = 0; static const INT opdDelta_MaxVal = 7; /* PS Stereo Huffmantable: opdDeltaFreq */ -static const UINT opdDeltaFreq_Length[] = -{ - 1, 3, 4, 4, 5, 5, 4, 3 -}; -static const UINT opdDeltaFreq_Code[] = -{ - 0x00000001, 0x00000001, 0x00000006, 0x00000004, 0x0000000f, 0x0000000e, 0x00000005, 0000000000, +static const UINT opdDeltaFreq_Length[] = {1, 3, 4, 4, 5, 5, 4, 3}; +static const UINT opdDeltaFreq_Code[] = { + 0x00000001, 0x00000001, 0x00000006, 0x00000004, + 0x0000000f, 0x0000000e, 0x00000005, 0000000000, }; /* PS Stereo Huffmantable: opdDeltaTime */ -static const UINT opdDeltaTime_Length[] = -{ - 1, 3, 4, 5, 5, 4, 4, 3 -}; -static const UINT opdDeltaTime_Code[] = -{ - 0x00000001, 0x00000002, 0x00000001, 0x00000007, 0x00000006, 0000000000, 0x00000002, 0x00000003 -}; +static const UINT opdDeltaTime_Length[] = {1, 3, 4, 5, 5, 4, 4, 3}; +static const UINT opdDeltaTime_Code[] = {0x00000001, 0x00000002, 0x00000001, + 0x00000007, 0x00000006, 0000000000, + 0x00000002, 0x00000003}; -static INT getNoBands(const INT mode) -{ +static INT getNoBands(const INT mode) { INT noBands = 0; switch (mode) { - case 0: case 3: /* coarse */ + case 0: + case 3: /* coarse */ noBands = PS_BANDS_COARSE; break; - case 1: case 4: /* mid */ + case 1: + case 4: /* mid */ noBands = PS_BANDS_MID; break; - case 2: case 5: /* fine not supported */ - default: /* coarse as default */ + case 2: + case 5: /* fine not supported */ + default: /* coarse as default */ noBands = PS_BANDS_COARSE; } return noBands; } -static INT getIIDRes(INT iidMode) -{ - if(iidMode<3) +static INT getIIDRes(INT iidMode) { + if (iidMode < 3) return PS_IID_RES_COARSE; else return PS_IID_RES_FINE; } -static INT -encodeDeltaFreq(HANDLE_FDK_BITSTREAM hBitBuf, - const INT *val, - const INT nBands, - const UINT *codeTable, - const UINT *lengthTable, - const INT tableOffset, - const INT maxVal, - INT *error) -{ +static INT encodeDeltaFreq(HANDLE_FDK_BITSTREAM hBitBuf, const INT *val, + const INT nBands, const UINT *codeTable, + const UINT *lengthTable, const INT tableOffset, + const INT maxVal, INT *error) { INT bitCnt = 0; INT lastVal = 0; INT band; - for(band=0;band<nBands;band++) { + for (band = 0; band < nBands; band++) { INT delta = (val[band] - lastVal) + tableOffset; lastVal = val[band]; - if( (delta>maxVal) || (delta<0) ) { + if ((delta > maxVal) || (delta < 0)) { *error = 1; - delta = delta>0?maxVal:0; + delta = delta > 0 ? maxVal : 0; } - bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, codeTable[delta], lengthTable[delta]); + bitCnt += + FDKsbrEnc_WriteBits_ps(hBitBuf, codeTable[delta], lengthTable[delta]); } return bitCnt; } -static INT -encodeDeltaTime(HANDLE_FDK_BITSTREAM hBitBuf, - const INT *val, - const INT *valLast, - const INT nBands, - const UINT *codeTable, - const UINT *lengthTable, - const INT tableOffset, - const INT maxVal, - INT *error) -{ +static INT encodeDeltaTime(HANDLE_FDK_BITSTREAM hBitBuf, const INT *val, + const INT *valLast, const INT nBands, + const UINT *codeTable, const UINT *lengthTable, + const INT tableOffset, const INT maxVal, + INT *error) { INT bitCnt = 0; INT band; - for(band=0;band<nBands;band++) { + for (band = 0; band < nBands; band++) { INT delta = (val[band] - valLast[band]) + tableOffset; - if( (delta>maxVal) || (delta<0) ) { + if ((delta > maxVal) || (delta < 0)) { *error = 1; - delta = delta>0?maxVal:0; + delta = delta > 0 ? maxVal : 0; } - bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, codeTable[delta], lengthTable[delta]); + bitCnt += + FDKsbrEnc_WriteBits_ps(hBitBuf, codeTable[delta], lengthTable[delta]); } return bitCnt; } -INT FDKsbrEnc_EncodeIid(HANDLE_FDK_BITSTREAM hBitBuf, - const INT *iidVal, - const INT *iidValLast, - const INT nBands, - const PS_IID_RESOLUTION res, - const PS_DELTA mode, - INT *error) -{ +INT FDKsbrEnc_EncodeIid(HANDLE_FDK_BITSTREAM hBitBuf, const INT *iidVal, + const INT *iidValLast, const INT nBands, + const PS_IID_RESOLUTION res, const PS_DELTA mode, + INT *error) { const UINT *codeTable; const UINT *lengthTable; INT bitCnt = 0; bitCnt = 0; - switch(mode) { - case PS_DELTA_FREQ: - switch(res) { - case PS_IID_RES_COARSE: - codeTable = iidDeltaFreqCoarse_Code; - lengthTable = iidDeltaFreqCoarse_Length; - bitCnt += encodeDeltaFreq(hBitBuf, iidVal, nBands, codeTable, - lengthTable, iidDeltaCoarse_Offset, - iidDeltaCoarse_MaxVal, error); - break; - case PS_IID_RES_FINE: - codeTable = iidDeltaFreqFine_Code; - lengthTable = iidDeltaFreqFine_Length; - bitCnt += encodeDeltaFreq(hBitBuf, iidVal, nBands, codeTable, - lengthTable, iidDeltaFine_Offset, - iidDeltaFine_MaxVal, error); - break; - default: - *error = 1; - } - break; - - case PS_DELTA_TIME: - switch(res) { - case PS_IID_RES_COARSE: - codeTable = iidDeltaTimeCoarse_Code; - lengthTable = iidDeltaTimeCoarse_Length; - bitCnt += encodeDeltaTime(hBitBuf, iidVal, iidValLast, nBands, codeTable, - lengthTable, iidDeltaCoarse_Offset, - iidDeltaCoarse_MaxVal, error); - break; - case PS_IID_RES_FINE: - codeTable = iidDeltaTimeFine_Code; - lengthTable = iidDeltaTimeFine_Length; - bitCnt += encodeDeltaTime(hBitBuf, iidVal, iidValLast, nBands, codeTable, - lengthTable, iidDeltaFine_Offset, - iidDeltaFine_MaxVal, error); - break; + switch (mode) { + case PS_DELTA_FREQ: + switch (res) { + case PS_IID_RES_COARSE: + codeTable = iidDeltaFreqCoarse_Code; + lengthTable = iidDeltaFreqCoarse_Length; + bitCnt += encodeDeltaFreq(hBitBuf, iidVal, nBands, codeTable, + lengthTable, iidDeltaCoarse_Offset, + iidDeltaCoarse_MaxVal, error); + break; + case PS_IID_RES_FINE: + codeTable = iidDeltaFreqFine_Code; + lengthTable = iidDeltaFreqFine_Length; + bitCnt += + encodeDeltaFreq(hBitBuf, iidVal, nBands, codeTable, lengthTable, + iidDeltaFine_Offset, iidDeltaFine_MaxVal, error); + break; + default: + *error = 1; + } + break; + + case PS_DELTA_TIME: + switch (res) { + case PS_IID_RES_COARSE: + codeTable = iidDeltaTimeCoarse_Code; + lengthTable = iidDeltaTimeCoarse_Length; + bitCnt += encodeDeltaTime( + hBitBuf, iidVal, iidValLast, nBands, codeTable, lengthTable, + iidDeltaCoarse_Offset, iidDeltaCoarse_MaxVal, error); + break; + case PS_IID_RES_FINE: + codeTable = iidDeltaTimeFine_Code; + lengthTable = iidDeltaTimeFine_Length; + bitCnt += encodeDeltaTime(hBitBuf, iidVal, iidValLast, nBands, + codeTable, lengthTable, iidDeltaFine_Offset, + iidDeltaFine_MaxVal, error); + break; + default: + *error = 1; + } + break; + default: *error = 1; - } - break; - - default: - *error = 1; } return bitCnt; } - -INT FDKsbrEnc_EncodeIcc(HANDLE_FDK_BITSTREAM hBitBuf, - const INT *iccVal, - const INT *iccValLast, - const INT nBands, - const PS_DELTA mode, - INT *error) -{ +INT FDKsbrEnc_EncodeIcc(HANDLE_FDK_BITSTREAM hBitBuf, const INT *iccVal, + const INT *iccValLast, const INT nBands, + const PS_DELTA mode, INT *error) { const UINT *codeTable; const UINT *lengthTable; INT bitCnt = 0; - switch(mode) { - case PS_DELTA_FREQ: - codeTable = iccDeltaFreq_Code; - lengthTable = iccDeltaFreq_Length; - bitCnt += encodeDeltaFreq(hBitBuf, iccVal, nBands, codeTable, - lengthTable, iccDelta_Offset, iccDelta_MaxVal, error); - break; + switch (mode) { + case PS_DELTA_FREQ: + codeTable = iccDeltaFreq_Code; + lengthTable = iccDeltaFreq_Length; + bitCnt += encodeDeltaFreq(hBitBuf, iccVal, nBands, codeTable, lengthTable, + iccDelta_Offset, iccDelta_MaxVal, error); + break; - case PS_DELTA_TIME: - codeTable = iccDeltaTime_Code; - lengthTable = iccDeltaTime_Length; + case PS_DELTA_TIME: + codeTable = iccDeltaTime_Code; + lengthTable = iccDeltaTime_Length; - bitCnt += encodeDeltaTime(hBitBuf, iccVal, iccValLast, nBands, codeTable, - lengthTable, iccDelta_Offset, iccDelta_MaxVal, error); - break; + bitCnt += + encodeDeltaTime(hBitBuf, iccVal, iccValLast, nBands, codeTable, + lengthTable, iccDelta_Offset, iccDelta_MaxVal, error); + break; - default: - *error = 1; + default: + *error = 1; } return bitCnt; } -INT FDKsbrEnc_EncodeIpd(HANDLE_FDK_BITSTREAM hBitBuf, - const INT *ipdVal, - const INT *ipdValLast, - const INT nBands, - const PS_DELTA mode, - INT *error) -{ +INT FDKsbrEnc_EncodeIpd(HANDLE_FDK_BITSTREAM hBitBuf, const INT *ipdVal, + const INT *ipdValLast, const INT nBands, + const PS_DELTA mode, INT *error) { const UINT *codeTable; const UINT *lengthTable; INT bitCnt = 0; - switch(mode) { - case PS_DELTA_FREQ: - codeTable = ipdDeltaFreq_Code; - lengthTable = ipdDeltaFreq_Length; - bitCnt += encodeDeltaFreq(hBitBuf, ipdVal, nBands, codeTable, - lengthTable, ipdDelta_Offset, ipdDelta_MaxVal, error); - break; + switch (mode) { + case PS_DELTA_FREQ: + codeTable = ipdDeltaFreq_Code; + lengthTable = ipdDeltaFreq_Length; + bitCnt += encodeDeltaFreq(hBitBuf, ipdVal, nBands, codeTable, lengthTable, + ipdDelta_Offset, ipdDelta_MaxVal, error); + break; - case PS_DELTA_TIME: - codeTable = ipdDeltaTime_Code; - lengthTable = ipdDeltaTime_Length; + case PS_DELTA_TIME: + codeTable = ipdDeltaTime_Code; + lengthTable = ipdDeltaTime_Length; - bitCnt += encodeDeltaTime(hBitBuf, ipdVal, ipdValLast, nBands, codeTable, - lengthTable, ipdDelta_Offset, ipdDelta_MaxVal, error); - break; + bitCnt += + encodeDeltaTime(hBitBuf, ipdVal, ipdValLast, nBands, codeTable, + lengthTable, ipdDelta_Offset, ipdDelta_MaxVal, error); + break; - default: - *error = 1; + default: + *error = 1; } return bitCnt; } -INT FDKsbrEnc_EncodeOpd(HANDLE_FDK_BITSTREAM hBitBuf, - const INT *opdVal, - const INT *opdValLast, - const INT nBands, - const PS_DELTA mode, - INT *error) -{ +INT FDKsbrEnc_EncodeOpd(HANDLE_FDK_BITSTREAM hBitBuf, const INT *opdVal, + const INT *opdValLast, const INT nBands, + const PS_DELTA mode, INT *error) { const UINT *codeTable; const UINT *lengthTable; INT bitCnt = 0; - switch(mode) { - case PS_DELTA_FREQ: - codeTable = opdDeltaFreq_Code; - lengthTable = opdDeltaFreq_Length; - bitCnt += encodeDeltaFreq(hBitBuf, opdVal, nBands, codeTable, - lengthTable, opdDelta_Offset, opdDelta_MaxVal, error); - break; + switch (mode) { + case PS_DELTA_FREQ: + codeTable = opdDeltaFreq_Code; + lengthTable = opdDeltaFreq_Length; + bitCnt += encodeDeltaFreq(hBitBuf, opdVal, nBands, codeTable, lengthTable, + opdDelta_Offset, opdDelta_MaxVal, error); + break; - case PS_DELTA_TIME: - codeTable = opdDeltaTime_Code; - lengthTable = opdDeltaTime_Length; + case PS_DELTA_TIME: + codeTable = opdDeltaTime_Code; + lengthTable = opdDeltaTime_Length; - bitCnt += encodeDeltaTime(hBitBuf, opdVal, opdValLast, nBands, codeTable, - lengthTable, opdDelta_Offset, opdDelta_MaxVal, error); - break; + bitCnt += + encodeDeltaTime(hBitBuf, opdVal, opdValLast, nBands, codeTable, + lengthTable, opdDelta_Offset, opdDelta_MaxVal, error); + break; - default: - *error = 1; + default: + *error = 1; } return bitCnt; } -static INT encodeIpdOpd(HANDLE_PS_OUT psOut, - HANDLE_FDK_BITSTREAM hBitBuf ) -{ +static INT encodeIpdOpd(HANDLE_PS_OUT psOut, HANDLE_FDK_BITSTREAM hBitBuf) { INT bitCnt = 0; - INT error = 0; + INT error = 0; INT env; FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->enableIpdOpd, 1); - if(psOut->enableIpdOpd==1) { + if (psOut->enableIpdOpd == 1) { INT *ipdLast = psOut->ipdLast; INT *opdLast = psOut->opdLast; - for(env=0; env<psOut->nEnvelopes; env++) { - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, psOut->deltaIPD[env], 1); - bitCnt += FDKsbrEnc_EncodeIpd( hBitBuf, - psOut->ipd[env], - ipdLast, - getNoBands(psOut->iidMode), - psOut->deltaIPD[env], - &error); - - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, psOut->deltaOPD[env], 1); - bitCnt += FDKsbrEnc_EncodeOpd( hBitBuf, - psOut->opd[env], - opdLast, - getNoBands(psOut->iidMode), - psOut->deltaOPD[env], - &error ); + for (env = 0; env < psOut->nEnvelopes; env++) { + bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->deltaIPD[env], 1); + bitCnt += FDKsbrEnc_EncodeIpd(hBitBuf, psOut->ipd[env], ipdLast, + getNoBands(psOut->iidMode), + psOut->deltaIPD[env], &error); + + bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->deltaOPD[env], 1); + bitCnt += FDKsbrEnc_EncodeOpd(hBitBuf, psOut->opd[env], opdLast, + getNoBands(psOut->iidMode), + psOut->deltaOPD[env], &error); } /* reserved bit */ - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, 0, 1); + bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, 0, 1); } - return bitCnt; } -static INT getEnvIdx(const INT nEnvelopes, const INT frameClass) -{ +static INT getEnvIdx(const INT nEnvelopes, const INT frameClass) { INT envIdx = 0; - switch(nEnvelopes) { - case 0: - envIdx = 0; - break; - - case 1: - if (frameClass==0) - envIdx = 1; - else + switch (nEnvelopes) { + case 0: envIdx = 0; - break; + break; - case 2: - if (frameClass==0) - envIdx = 2; - else - envIdx = 1; - break; + case 1: + if (frameClass == 0) + envIdx = 1; + else + envIdx = 0; + break; + + case 2: + if (frameClass == 0) + envIdx = 2; + else + envIdx = 1; + break; - case 3: - envIdx = 2; - break; + case 3: + envIdx = 2; + break; - case 4: - envIdx = 3; - break; + case 4: + envIdx = 3; + break; - default: - /* unsupported number of envelopes */ - envIdx = 0; + default: + /* unsupported number of envelopes */ + envIdx = 0; } return envIdx; } - -static INT encodePSExtension(const HANDLE_PS_OUT psOut, - HANDLE_FDK_BITSTREAM hBitBuf ) -{ +static INT encodePSExtension(const HANDLE_PS_OUT psOut, + HANDLE_FDK_BITSTREAM hBitBuf) { INT bitCnt = 0; - if(psOut->enableIpdOpd==1) { + if (psOut->enableIpdOpd == 1) { INT ipdOpdBits = 0; - INT extSize = (2 + encodeIpdOpd(psOut,NULL)+7)>>3; + INT extSize = (2 + encodeIpdOpd(psOut, NULL) + 7) >> 3; - if(extSize<15) { - bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, extSize, 4); - } - else { - bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, 15 , 4); - bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, (extSize-15), 8); + if (extSize < 15) { + bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, extSize, 4); + } else { + bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, 15, 4); + bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, (extSize - 15), 8); } /* write ipd opd data */ ipdOpdBits += FDKsbrEnc_WriteBits_ps(hBitBuf, PS_EXT_ID_V0, 2); - ipdOpdBits += encodeIpdOpd(psOut, hBitBuf ); + ipdOpdBits += encodeIpdOpd(psOut, hBitBuf); /* byte align the ipd opd data */ - if(ipdOpdBits%8) - ipdOpdBits += FDKsbrEnc_WriteBits_ps(hBitBuf, 0, (8-(ipdOpdBits%8)) ); + if (ipdOpdBits % 8) + ipdOpdBits += FDKsbrEnc_WriteBits_ps(hBitBuf, 0, (8 - (ipdOpdBits % 8))); bitCnt += ipdOpdBits; } @@ -617,77 +552,69 @@ static INT encodePSExtension(const HANDLE_PS_OUT psOut, return (bitCnt); } -INT FDKsbrEnc_WritePSBitstream(const HANDLE_PS_OUT psOut, - HANDLE_FDK_BITSTREAM hBitBuf ) -{ +INT FDKsbrEnc_WritePSBitstream(const HANDLE_PS_OUT psOut, + HANDLE_FDK_BITSTREAM hBitBuf) { INT psExtEnable = 0; INT bitCnt = 0; INT error = 0; INT env; - if(psOut != NULL){ - + if (psOut != NULL) { /* PS HEADER */ - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, psOut->enablePSHeader, 1); + bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->enablePSHeader, 1); - if(psOut->enablePSHeader) { - - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, psOut->enableIID, 1); - if(psOut->enableIID) { - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, psOut->iidMode, 3); + if (psOut->enablePSHeader) { + bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->enableIID, 1); + if (psOut->enableIID) { + bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->iidMode, 3); } - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, psOut->enableICC, 1); - if(psOut->enableICC) { - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, psOut->iccMode, 3); + bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->enableICC, 1); + if (psOut->enableICC) { + bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->iccMode, 3); } - if(psOut->enableIpdOpd) { + if (psOut->enableIpdOpd) { psExtEnable = 1; } - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, psExtEnable, 1); + bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psExtEnable, 1); } /* Frame class, number of envelopes */ - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, psOut->frameClass, 1); - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, getEnvIdx(psOut->nEnvelopes, psOut->frameClass), 2); + bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->frameClass, 1); + bitCnt += FDKsbrEnc_WriteBits_ps( + hBitBuf, getEnvIdx(psOut->nEnvelopes, psOut->frameClass), 2); - if(psOut->frameClass==1) { - for(env=0; env<psOut->nEnvelopes; env++) { - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, psOut->frameBorder[env], 5); + if (psOut->frameClass == 1) { + for (env = 0; env < psOut->nEnvelopes; env++) { + bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->frameBorder[env], 5); } } - if(psOut->enableIID==1) { + if (psOut->enableIID == 1) { INT *iidLast = psOut->iidLast; - for(env=0; env<psOut->nEnvelopes; env++) { - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, psOut->deltaIID[env], 1); - bitCnt += FDKsbrEnc_EncodeIid( hBitBuf, - psOut->iid[env], - iidLast, - getNoBands(psOut->iidMode), - (PS_IID_RESOLUTION)getIIDRes(psOut->iidMode), - psOut->deltaIID[env], - &error ); + for (env = 0; env < psOut->nEnvelopes; env++) { + bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->deltaIID[env], 1); + bitCnt += FDKsbrEnc_EncodeIid( + hBitBuf, psOut->iid[env], iidLast, getNoBands(psOut->iidMode), + (PS_IID_RESOLUTION)getIIDRes(psOut->iidMode), psOut->deltaIID[env], + &error); iidLast = psOut->iid[env]; } } - if(psOut->enableICC==1) { + if (psOut->enableICC == 1) { INT *iccLast = psOut->iccLast; - for(env=0; env<psOut->nEnvelopes; env++) { - bitCnt += FDKsbrEnc_WriteBits_ps( hBitBuf, psOut->deltaICC[env], 1); - bitCnt += FDKsbrEnc_EncodeIcc( hBitBuf, - psOut->icc[env], - iccLast, - getNoBands(psOut->iccMode), - psOut->deltaICC[env], - &error); + for (env = 0; env < psOut->nEnvelopes; env++) { + bitCnt += FDKsbrEnc_WriteBits_ps(hBitBuf, psOut->deltaICC[env], 1); + bitCnt += FDKsbrEnc_EncodeIcc(hBitBuf, psOut->icc[env], iccLast, + getNoBands(psOut->iccMode), + psOut->deltaICC[env], &error); iccLast = psOut->icc[env]; } } - if(psExtEnable!=0) { + if (psExtEnable != 0) { bitCnt += encodePSExtension(psOut, hBitBuf); } @@ -695,4 +622,3 @@ INT FDKsbrEnc_WritePSBitstream(const HANDLE_PS_OUT psOut, return bitCnt; } - diff --git a/libSBRenc/src/ps_bitenc.h b/libSBRenc/src/ps_bitenc.h index e98fe58..1d383e3 100644 --- a/libSBRenc/src/ps_bitenc.h +++ b/libSBRenc/src/ps_bitenc.h @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,14 +90,15 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** -/***************************** MPEG Audio Encoder *************************** + Author(s): N. Rettelbach - Initial author: N. Rettelbach - contents/description: Parametric Stereo bitstream encoder + Description: Parametric Stereo bitstream encoder -******************************************************************************/ +*******************************************************************************/ #include "ps_main.h" #include "ps_const.h" @@ -96,82 +108,66 @@ amm-info@iis.fraunhofer.de #define PS_BITENC_H typedef struct T_PS_OUT { - - INT enablePSHeader; - INT enableIID; - INT iidMode; - INT enableICC; - INT iccMode; - INT enableIpdOpd; - - INT frameClass; - INT nEnvelopes; + INT enablePSHeader; + INT enableIID; + INT iidMode; + INT enableICC; + INT iccMode; + INT enableIpdOpd; + + INT frameClass; + INT nEnvelopes; /* ENV data */ - INT frameBorder[PS_MAX_ENVELOPES]; + INT frameBorder[PS_MAX_ENVELOPES]; /* iid data */ - PS_DELTA deltaIID[PS_MAX_ENVELOPES]; - INT iid[PS_MAX_ENVELOPES][PS_MAX_BANDS]; - INT iidLast[PS_MAX_BANDS]; + PS_DELTA deltaIID[PS_MAX_ENVELOPES]; + INT iid[PS_MAX_ENVELOPES][PS_MAX_BANDS]; + INT iidLast[PS_MAX_BANDS]; /* icc data */ - PS_DELTA deltaICC[PS_MAX_ENVELOPES]; - INT icc[PS_MAX_ENVELOPES][PS_MAX_BANDS]; - INT iccLast[PS_MAX_BANDS]; + PS_DELTA deltaICC[PS_MAX_ENVELOPES]; + INT icc[PS_MAX_ENVELOPES][PS_MAX_BANDS]; + INT iccLast[PS_MAX_BANDS]; /* ipd data */ - PS_DELTA deltaIPD[PS_MAX_ENVELOPES]; - INT ipd[PS_MAX_ENVELOPES][PS_MAX_BANDS]; - INT ipdLast[PS_MAX_BANDS]; + PS_DELTA deltaIPD[PS_MAX_ENVELOPES]; + INT ipd[PS_MAX_ENVELOPES][PS_MAX_BANDS]; + INT ipdLast[PS_MAX_BANDS]; /* opd data */ - PS_DELTA deltaOPD[PS_MAX_ENVELOPES]; - INT opd[PS_MAX_ENVELOPES][PS_MAX_BANDS]; - INT opdLast[PS_MAX_BANDS]; + PS_DELTA deltaOPD[PS_MAX_ENVELOPES]; + INT opd[PS_MAX_ENVELOPES][PS_MAX_BANDS]; + INT opdLast[PS_MAX_BANDS]; } PS_OUT, *HANDLE_PS_OUT; - #ifdef __cplusplus extern "C" { #endif /* __cplusplus */ -INT FDKsbrEnc_EncodeIid(HANDLE_FDK_BITSTREAM hBitBuf, - const INT *iidVal, - const INT *iidValLast, - const INT nBands, - const PS_IID_RESOLUTION res, - const PS_DELTA mode, - INT *error); - -INT FDKsbrEnc_EncodeIcc(HANDLE_FDK_BITSTREAM hBitBuf, - const INT *iccVal, - const INT *iccValLast, - const INT nBands, - const PS_DELTA mode, - INT *error); - -INT FDKsbrEnc_EncodeIpd(HANDLE_FDK_BITSTREAM hBitBuf, - const INT *ipdVal, - const INT *ipdValLast, - const INT nBands, - const PS_DELTA mode, - INT *error); - -INT FDKsbrEnc_EncodeOpd(HANDLE_FDK_BITSTREAM hBitBuf, - const INT *opdVal, - const INT *opdValLast, - const INT nBands, - const PS_DELTA mode, - INT *error); - -INT FDKsbrEnc_WritePSBitstream(const HANDLE_PS_OUT psOut, - HANDLE_FDK_BITSTREAM hBitBuf); +INT FDKsbrEnc_EncodeIid(HANDLE_FDK_BITSTREAM hBitBuf, const INT *iidVal, + const INT *iidValLast, const INT nBands, + const PS_IID_RESOLUTION res, const PS_DELTA mode, + INT *error); +INT FDKsbrEnc_EncodeIcc(HANDLE_FDK_BITSTREAM hBitBuf, const INT *iccVal, + const INT *iccValLast, const INT nBands, + const PS_DELTA mode, INT *error); + +INT FDKsbrEnc_EncodeIpd(HANDLE_FDK_BITSTREAM hBitBuf, const INT *ipdVal, + const INT *ipdValLast, const INT nBands, + const PS_DELTA mode, INT *error); + +INT FDKsbrEnc_EncodeOpd(HANDLE_FDK_BITSTREAM hBitBuf, const INT *opdVal, + const INT *opdValLast, const INT nBands, + const PS_DELTA mode, INT *error); + +INT FDKsbrEnc_WritePSBitstream(const HANDLE_PS_OUT psOut, + HANDLE_FDK_BITSTREAM hBitBuf); #ifdef __cplusplus } #endif /* __cplusplus */ - -#endif /* #ifndef PS_BITENC_H */ +#endif /* defined(PSENC_ENABLE) */ diff --git a/libSBRenc/src/ps_const.h b/libSBRenc/src/ps_const.h index 633d210..b9a33f9 100644 --- a/libSBRenc/src/ps_const.h +++ b/libSBRenc/src/ps_const.h @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,56 +90,46 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** -/***************************** MPEG Audio Encoder *************************** + Author(s): N. Rettelbach - Initial author: N. Rettelbach - contents/description: Parametric Stereo constants + Description: Parametric Stereo constants -******************************************************************************/ +*******************************************************************************/ #ifndef PS_CONST_H #define PS_CONST_H -#define MAX_PS_CHANNELS ( 2 ) -#define HYBRID_MAX_QMF_BANDS ( 3 ) -#define HYBRID_FILTER_LENGTH ( 13 ) -#define HYBRID_FILTER_DELAY ( (HYBRID_FILTER_LENGTH-1)/2 ) +#define MAX_PS_CHANNELS (2) +#define HYBRID_MAX_QMF_BANDS (3) +#define HYBRID_FILTER_LENGTH (13) +#define HYBRID_FILTER_DELAY ((HYBRID_FILTER_LENGTH - 1) / 2) -#define HYBRID_FRAMESIZE ( QMF_MAX_TIME_SLOTS ) -#define HYBRID_READ_OFFSET ( 10 ) - -#define MAX_HYBRID_BANDS ( (QMF_CHANNELS-HYBRID_MAX_QMF_BANDS+10) ) +#define HYBRID_FRAMESIZE (32) +#define HYBRID_READ_OFFSET (10) +#define MAX_HYBRID_BANDS ((64 - HYBRID_MAX_QMF_BANDS + 10)) typedef enum { - PS_RES_COARSE = 0, - PS_RES_MID = 1, - PS_RES_FINE = 2 + PS_RES_COARSE = 0, + PS_RES_MID = 1, + PS_RES_FINE = 2 } PS_RESOLUTION; typedef enum { - PS_BANDS_COARSE = 10, - PS_BANDS_MID = 20, - PS_MAX_BANDS = PS_BANDS_MID + PS_BANDS_COARSE = 10, + PS_BANDS_MID = 20, + PS_MAX_BANDS = PS_BANDS_MID } PS_BANDS; -typedef enum { - PS_IID_RES_COARSE=0, - PS_IID_RES_FINE -} PS_IID_RESOLUTION; - -typedef enum { - PS_ICC_ROT_A=0, - PS_ICC_ROT_B -} PS_ICC_ROTATION_MODE; +typedef enum { PS_IID_RES_COARSE = 0, PS_IID_RES_FINE } PS_IID_RESOLUTION; -typedef enum { - PS_DELTA_FREQ, - PS_DELTA_TIME -} PS_DELTA; +typedef enum { PS_ICC_ROT_A = 0, PS_ICC_ROT_B } PS_ICC_ROTATION_MODE; +typedef enum { PS_DELTA_FREQ, PS_DELTA_TIME } PS_DELTA; typedef enum { PS_MAX_ENVELOPES = 4 @@ -136,13 +137,14 @@ typedef enum { } PS_CONSTS; typedef enum { - PSENC_OK = 0x0000, /*!< No error happened. All fine. */ - PSENC_INVALID_HANDLE = 0x0020, /*!< Handle passed to function call was invalid. */ - PSENC_MEMORY_ERROR = 0x0021, /*!< Memory allocation failed. */ - PSENC_INIT_ERROR = 0x0040, /*!< General initialization error. */ - PSENC_ENCODE_ERROR = 0x0060 /*!< The encoding process was interrupted by an unexpected error. */ + PSENC_OK = 0x0000, /*!< No error happened. All fine. */ + PSENC_INVALID_HANDLE = + 0x0020, /*!< Handle passed to function call was invalid. */ + PSENC_MEMORY_ERROR = 0x0021, /*!< Memory allocation failed. */ + PSENC_INIT_ERROR = 0x0040, /*!< General initialization error. */ + PSENC_ENCODE_ERROR = 0x0060 /*!< The encoding process was interrupted by an + unexpected error. */ } FDK_PSENC_ERROR; - #endif diff --git a/libSBRenc/src/ps_encode.cpp b/libSBRenc/src/ps_encode.cpp index fec39e8..88d3131 100644 --- a/libSBRenc/src/ps_encode.cpp +++ b/libSBRenc/src/ps_encode.cpp @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,121 +90,109 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): M. Neuendorf, N. Rettelbach, M. Multrus -/***************************** MPEG Audio Encoder *************************** + Description: PS parameter extraction, encoding - Initial Authors: M. Neuendorf, N. Rettelbach, M. Multrus - Contents/Description: PS parameter extraction, encoding +*******************************************************************************/ -******************************************************************************/ /*! \file - \brief PS parameter extraction, encoding functions + \brief PS parameter extraction, encoding functions $Revision: 96441 $ */ #include "ps_main.h" - - -#include "sbr_ram.h" #include "ps_encode.h" - #include "qmf.h" - -#include "ps_const.h" #include "sbr_misc.h" +#include "sbrenc_ram.h" #include "genericStds.h" -inline void FDKsbrEnc_addFIXP_DBL(const FIXP_DBL *X, const FIXP_DBL *Y, FIXP_DBL *Z, INT n) -{ - for (INT i=0; i<n; i++) - Z[i] = (X[i]>>1) + (Y[i]>>1); +inline void FDKsbrEnc_addFIXP_DBL(const FIXP_DBL *X, const FIXP_DBL *Y, + FIXP_DBL *Z, INT n) { + for (INT i = 0; i < n; i++) Z[i] = (X[i] >> 1) + (Y[i] >> 1); } -#define LOG10_2_10 3.01029995664f /* 10.0f*log10(2.f) */ - -static const INT iidGroupBordersLoRes[QMF_GROUPS_LO_RES + SUBQMF_GROUPS_LO_RES + 1] = -{ - 0, 1, 2, 3, 4, 5, /* 6 subqmf subbands - 0th qmf subband */ - 6, 7, /* 2 subqmf subbands - 1st qmf subband */ - 8, 9, /* 2 subqmf subbands - 2nd qmf subband */ - 10, 11, 12, 13, 14, 15, 16, 18, 21, 25, 30, 42, 71 -}; +#define LOG10_2_10 3.01029995664f /* 10.0f*log10(2.f) */ -static const UCHAR iidGroupWidthLdLoRes[QMF_GROUPS_LO_RES + SUBQMF_GROUPS_LO_RES] = -{ - 0, 0, 0, 0, 0, 0, - 0, 0, - 0, 0, - 0, 0, 0, 0, 0, 0, 1, 2, 2, 3, 4, 5 -}; +static const INT + iidGroupBordersLoRes[QMF_GROUPS_LO_RES + SUBQMF_GROUPS_LO_RES + 1] = { + 0, 1, 2, 3, 4, 5, /* 6 subqmf subbands - 0th qmf subband */ + 6, 7, /* 2 subqmf subbands - 1st qmf subband */ + 8, 9, /* 2 subqmf subbands - 2nd qmf subband */ + 10, 11, 12, 13, 14, 15, 16, 18, 21, 25, 30, 42, 71}; +static const UCHAR + iidGroupWidthLdLoRes[QMF_GROUPS_LO_RES + SUBQMF_GROUPS_LO_RES] = { + 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 1, 2, 2, 3, 4, 5}; static const INT subband2parameter20[QMF_GROUPS_LO_RES + SUBQMF_GROUPS_LO_RES] = -{ - 1, 0, 0, 1, 2, 3, /* 6 subqmf subbands - 0th qmf subband */ - 4, 5, /* 2 subqmf subbands - 1st qmf subband */ - 6, 7, /* 2 subqmf subbands - 2nd qmf subband */ - 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19 -}; - + {1, 0, 0, 1, 2, 3, /* 6 subqmf subbands - 0th qmf subband */ + 4, 5, /* 2 subqmf subbands - 1st qmf subband */ + 6, 7, /* 2 subqmf subbands - 2nd qmf subband */ + 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19}; typedef enum { MAX_TIME_DIFF_FRAMES = 20, - MAX_PS_NOHEADER_CNT = 10, - MAX_NOENV_CNT = 10, + MAX_PS_NOHEADER_CNT = 10, + MAX_NOENV_CNT = 10, DO_NOT_USE_THIS_MODE = 0x7FFFFF } __PS_CONSTANTS; - - static const FIXP_DBL iidQuant_fx[15] = { - (FIXP_DBL)0xce000000, (FIXP_DBL)0xdc000000, (FIXP_DBL)0xe4000000, (FIXP_DBL)0xec000000, (FIXP_DBL)0xf2000000, (FIXP_DBL)0xf8000000, (FIXP_DBL)0xfc000000, (FIXP_DBL)0x00000000, - (FIXP_DBL)0x04000000, (FIXP_DBL)0x08000000, (FIXP_DBL)0x0e000000, (FIXP_DBL)0x14000000, (FIXP_DBL)0x1c000000, (FIXP_DBL)0x24000000, (FIXP_DBL)0x32000000 -}; + (FIXP_DBL)0xce000000, (FIXP_DBL)0xdc000000, (FIXP_DBL)0xe4000000, + (FIXP_DBL)0xec000000, (FIXP_DBL)0xf2000000, (FIXP_DBL)0xf8000000, + (FIXP_DBL)0xfc000000, (FIXP_DBL)0x00000000, (FIXP_DBL)0x04000000, + (FIXP_DBL)0x08000000, (FIXP_DBL)0x0e000000, (FIXP_DBL)0x14000000, + (FIXP_DBL)0x1c000000, (FIXP_DBL)0x24000000, (FIXP_DBL)0x32000000}; static const FIXP_DBL iidQuantFine_fx[31] = { - (FIXP_DBL)0x9c000001, (FIXP_DBL)0xa6000001, (FIXP_DBL)0xb0000001, (FIXP_DBL)0xba000001, (FIXP_DBL)0xc4000000, (FIXP_DBL)0xce000000, (FIXP_DBL)0xd4000000, (FIXP_DBL)0xda000000, - (FIXP_DBL)0xe0000000, (FIXP_DBL)0xe6000000, (FIXP_DBL)0xec000000, (FIXP_DBL)0xf0000000, (FIXP_DBL)0xf4000000, (FIXP_DBL)0xf8000000, (FIXP_DBL)0xfc000000, (FIXP_DBL)0x00000000, - (FIXP_DBL)0x04000000, (FIXP_DBL)0x08000000, (FIXP_DBL)0x0c000000, (FIXP_DBL)0x10000000, (FIXP_DBL)0x14000000, (FIXP_DBL)0x1a000000, (FIXP_DBL)0x20000000, (FIXP_DBL)0x26000000, - (FIXP_DBL)0x2c000000, (FIXP_DBL)0x32000000, (FIXP_DBL)0x3c000000, (FIXP_DBL)0x45ffffff, (FIXP_DBL)0x4fffffff, (FIXP_DBL)0x59ffffff, (FIXP_DBL)0x63ffffff -}; - - + (FIXP_DBL)0x9c000001, (FIXP_DBL)0xa6000001, (FIXP_DBL)0xb0000001, + (FIXP_DBL)0xba000001, (FIXP_DBL)0xc4000000, (FIXP_DBL)0xce000000, + (FIXP_DBL)0xd4000000, (FIXP_DBL)0xda000000, (FIXP_DBL)0xe0000000, + (FIXP_DBL)0xe6000000, (FIXP_DBL)0xec000000, (FIXP_DBL)0xf0000000, + (FIXP_DBL)0xf4000000, (FIXP_DBL)0xf8000000, (FIXP_DBL)0xfc000000, + (FIXP_DBL)0x00000000, (FIXP_DBL)0x04000000, (FIXP_DBL)0x08000000, + (FIXP_DBL)0x0c000000, (FIXP_DBL)0x10000000, (FIXP_DBL)0x14000000, + (FIXP_DBL)0x1a000000, (FIXP_DBL)0x20000000, (FIXP_DBL)0x26000000, + (FIXP_DBL)0x2c000000, (FIXP_DBL)0x32000000, (FIXP_DBL)0x3c000000, + (FIXP_DBL)0x45ffffff, (FIXP_DBL)0x4fffffff, (FIXP_DBL)0x59ffffff, + (FIXP_DBL)0x63ffffff}; static const FIXP_DBL iccQuant[8] = { - (FIXP_DBL)0x7fffffff, (FIXP_DBL)0x77ef9d7f, (FIXP_DBL)0x6babc97f, (FIXP_DBL)0x4ceaf27f, (FIXP_DBL)0x2f0ed3c0, (FIXP_DBL)0x00000000, (FIXP_DBL)0xb49ba601, (FIXP_DBL)0x80000000 -}; + (FIXP_DBL)0x7fffffff, (FIXP_DBL)0x77ef9d7f, (FIXP_DBL)0x6babc97f, + (FIXP_DBL)0x4ceaf27f, (FIXP_DBL)0x2f0ed3c0, (FIXP_DBL)0x00000000, + (FIXP_DBL)0xb49ba601, (FIXP_DBL)0x80000000}; -static FDK_PSENC_ERROR InitPSData( - HANDLE_PS_DATA hPsData - ) -{ +static FDK_PSENC_ERROR InitPSData(HANDLE_PS_DATA hPsData) { FDK_PSENC_ERROR error = PSENC_OK; - if(hPsData == NULL) { + if (hPsData == NULL) { error = PSENC_INVALID_HANDLE; - } - else { + } else { int i, env; - FDKmemclear(hPsData,sizeof(PS_DATA)); + FDKmemclear(hPsData, sizeof(PS_DATA)); - for (i=0; i<PS_MAX_BANDS; i++) { + for (i = 0; i < PS_MAX_BANDS; i++) { hPsData->iidIdxLast[i] = 0; hPsData->iccIdxLast[i] = 0; } - hPsData->iidEnable = hPsData->iidEnableLast = 0; - hPsData->iccEnable = hPsData->iccEnableLast = 0; + hPsData->iidEnable = hPsData->iidEnableLast = 0; + hPsData->iccEnable = hPsData->iccEnableLast = 0; hPsData->iidQuantMode = hPsData->iidQuantModeLast = PS_IID_RES_COARSE; hPsData->iccQuantMode = hPsData->iccQuantModeLast = PS_ICC_ROT_A; - for(env=0; env<PS_MAX_ENVELOPES; env++) { + for (env = 0; env < PS_MAX_ENVELOPES; env++) { hPsData->iccDiffMode[env] = PS_DELTA_FREQ; hPsData->iccDiffMode[env] = PS_DELTA_FREQ; - for (i=0; i<PS_MAX_BANDS; i++) { + for (i = 0; i < PS_MAX_BANDS; i++) { hPsData->iidIdx[env][i] = 0; hPsData->iccIdx[env][i] = 0; } @@ -201,94 +200,84 @@ static FDK_PSENC_ERROR InitPSData( hPsData->nEnvelopesLast = 0; - hPsData->headerCnt = MAX_PS_NOHEADER_CNT; + hPsData->headerCnt = MAX_PS_NOHEADER_CNT; hPsData->iidTimeCnt = MAX_TIME_DIFF_FRAMES; hPsData->iccTimeCnt = MAX_TIME_DIFF_FRAMES; - hPsData->noEnvCnt = MAX_NOENV_CNT; + hPsData->noEnvCnt = MAX_NOENV_CNT; } return error; } -static FIXP_DBL quantizeCoef( const FIXP_DBL *RESTRICT input, - const INT nBands, - const FIXP_DBL *RESTRICT quantTable, - const INT idxOffset, - const INT nQuantSteps, - INT *RESTRICT quantOut) -{ +static FIXP_DBL quantizeCoef(const FIXP_DBL *RESTRICT input, const INT nBands, + const FIXP_DBL *RESTRICT quantTable, + const INT idxOffset, const INT nQuantSteps, + INT *RESTRICT quantOut) { INT idx, band; FIXP_DBL quantErr = FL2FXCONST_DBL(0.f); - for (band=0; band<nBands;band++) { - for(idx=0; idx<nQuantSteps-1; idx++){ - if( fixp_abs((input[band]>>1)-(quantTable[idx+1]>>1)) > - fixp_abs((input[band]>>1)-(quantTable[idx]>>1)) ) - { + for (band = 0; band < nBands; band++) { + for (idx = 0; idx < nQuantSteps - 1; idx++) { + if (fixp_abs((input[band] >> 1) - (quantTable[idx + 1] >> 1)) > + fixp_abs((input[band] >> 1) - (quantTable[idx] >> 1))) { break; } } - quantErr += (fixp_abs(input[band]-quantTable[idx])>>PS_QUANT_SCALE); /* don't scale before subtraction; diff smaller (64-25)/64 */ + quantErr += (fixp_abs(input[band] - quantTable[idx]) >> + PS_QUANT_SCALE); /* don't scale before subtraction; diff + smaller (64-25)/64 */ quantOut[band] = idx - idxOffset; } return quantErr; } -static INT getICCMode(const INT nBands, - const INT rotType) -{ +static INT getICCMode(const INT nBands, const INT rotType) { INT mode = 0; - switch(nBands) { - case PS_BANDS_COARSE: - mode = PS_RES_COARSE; - break; - case PS_BANDS_MID: - mode = PS_RES_MID; - break; - default: - mode = 0; + switch (nBands) { + case PS_BANDS_COARSE: + mode = PS_RES_COARSE; + break; + case PS_BANDS_MID: + mode = PS_RES_MID; + break; + default: + mode = 0; } - if(rotType==PS_ICC_ROT_B){ + if (rotType == PS_ICC_ROT_B) { mode += 3; } return mode; } - -static INT getIIDMode(const INT nBands, - const INT iidRes) -{ +static INT getIIDMode(const INT nBands, const INT iidRes) { INT mode = 0; - switch(nBands) { - case PS_BANDS_COARSE: - mode = PS_RES_COARSE; - break; - case PS_BANDS_MID: - mode = PS_RES_MID; - break; - default: - mode = 0; - break; + switch (nBands) { + case PS_BANDS_COARSE: + mode = PS_RES_COARSE; + break; + case PS_BANDS_MID: + mode = PS_RES_MID; + break; + default: + mode = 0; + break; } - if(iidRes == PS_IID_RES_FINE){ + if (iidRes == PS_IID_RES_FINE) { mode += 3; } return mode; } - static INT envelopeReducible(FIXP_DBL iid[PS_MAX_ENVELOPES][PS_MAX_BANDS], FIXP_DBL icc[PS_MAX_ENVELOPES][PS_MAX_BANDS], - INT psBands, - INT nEnvelopes) -{ - #define THRESH_SCALE 7 + INT psBands, INT nEnvelopes) { +#define THRESH_SCALE 7 INT reducible = 1; /* true */ INT e = 0, b = 0; @@ -299,31 +288,36 @@ static INT envelopeReducible(FIXP_DBL iid[PS_MAX_ENVELOPES][PS_MAX_BANDS], FIXP_DBL iidMeanError, iccMeanError; /* square values to prevent sqrt, - multiply bands to prevent division; bands shifted DFRACT_BITS instead (DFRACT_BITS-1) because fMultDiv2 used*/ - iidErrThreshold = fMultDiv2 ( FL2FXCONST_DBL(6.5f*6.5f/(IID_SCALE_FT*IID_SCALE_FT)), (FIXP_DBL)(psBands<<((DFRACT_BITS)-THRESH_SCALE)) ); - iccErrThreshold = fMultDiv2 ( FL2FXCONST_DBL(0.75f*0.75f), (FIXP_DBL)(psBands<<((DFRACT_BITS)-THRESH_SCALE)) ); + multiply bands to prevent division; bands shifted DFRACT_BITS instead + (DFRACT_BITS-1) because fMultDiv2 used*/ + iidErrThreshold = + fMultDiv2(FL2FXCONST_DBL(6.5f * 6.5f / (IID_SCALE_FT * IID_SCALE_FT)), + (FIXP_DBL)(psBands << ((DFRACT_BITS)-THRESH_SCALE))); + iccErrThreshold = + fMultDiv2(FL2FXCONST_DBL(0.75f * 0.75f), + (FIXP_DBL)(psBands << ((DFRACT_BITS)-THRESH_SCALE))); if (nEnvelopes <= 1) { reducible = 0; } else { - /* mean error criterion */ - for (e=0; (e < nEnvelopes/2) && (reducible!=0 ) ; e++) { + for (e = 0; (e < nEnvelopes / 2) && (reducible != 0); e++) { iidMeanError = iccMeanError = FL2FXCONST_DBL(0.f); - for(b=0; b<psBands; b++) { - dIid = (iid[2*e][b]>>1) - (iid[2*e+1][b]>>1); /* scale 1 bit; squared -> 2 bit */ - dIcc = (icc[2*e][b]>>1) - (icc[2*e+1][b]>>1); - iidMeanError += fPow2Div2(dIid)>>(5-1); /* + (bands=20) scale = 5 */ - iccMeanError += fPow2Div2(dIcc)>>(5-1); - } /* --> scaling = 7 bit = THRESH_SCALE !! */ + for (b = 0; b < psBands; b++) { + dIid = (iid[2 * e][b] >> 1) - + (iid[2 * e + 1][b] >> 1); /* scale 1 bit; squared -> 2 bit */ + dIcc = (icc[2 * e][b] >> 1) - (icc[2 * e + 1][b] >> 1); + iidMeanError += fPow2Div2(dIid) >> (5 - 1); /* + (bands=20) scale = 5 */ + iccMeanError += fPow2Div2(dIcc) >> (5 - 1); + } /* --> scaling = 7 bit = THRESH_SCALE !! */ /* instead sqrt values are squared! instead of division, multiply threshold with psBands scaling necessary!! */ /* quit as soon as threshold is reached */ - if ( (iidMeanError > (iidErrThreshold)) || - (iccMeanError > (iccErrThreshold)) ) { + if ((iidMeanError > (iidErrThreshold)) || + (iccMeanError > (iccErrThreshold))) { reducible = 0; } } @@ -332,305 +326,314 @@ static INT envelopeReducible(FIXP_DBL iid[PS_MAX_ENVELOPES][PS_MAX_BANDS], return reducible; } - -static void processIidData(PS_DATA *psData, - FIXP_DBL iid[PS_MAX_ENVELOPES][PS_MAX_BANDS], - const INT psBands, - const INT nEnvelopes, - const FIXP_DBL quantErrorThreshold) -{ - INT iidIdxFine [PS_MAX_ENVELOPES][PS_MAX_BANDS]; +static void processIidData(PS_DATA *psData, + FIXP_DBL iid[PS_MAX_ENVELOPES][PS_MAX_BANDS], + const INT psBands, const INT nEnvelopes, + const FIXP_DBL quantErrorThreshold) { + INT iidIdxFine[PS_MAX_ENVELOPES][PS_MAX_BANDS]; INT iidIdxCoarse[PS_MAX_ENVELOPES][PS_MAX_BANDS]; FIXP_DBL errIID = FL2FXCONST_DBL(0.f); FIXP_DBL errIIDFine = FL2FXCONST_DBL(0.f); - INT bitsIidFreq = 0; - INT bitsIidTime = 0; - INT bitsFineTot = 0; - INT bitsCoarseTot = 0; - INT error = 0; - INT env, band; - INT diffMode[PS_MAX_ENVELOPES], diffModeFine[PS_MAX_ENVELOPES]; + INT bitsIidFreq = 0; + INT bitsIidTime = 0; + INT bitsFineTot = 0; + INT bitsCoarseTot = 0; + INT error = 0; + INT env, band; + INT diffMode[PS_MAX_ENVELOPES], diffModeFine[PS_MAX_ENVELOPES]; INT loudnDiff = 0; INT iidTransmit = 0; - bitsIidFreq = bitsIidTime = 0; - /* Quantize IID coefficients */ - for(env=0;env<nEnvelopes; env++) { - errIID += quantizeCoef(iid[env], psBands, iidQuant_fx, 7, 15, iidIdxCoarse[env]); - errIIDFine += quantizeCoef(iid[env], psBands, iidQuantFine_fx, 15, 31, iidIdxFine[env]); + for (env = 0; env < nEnvelopes; env++) { + errIID += + quantizeCoef(iid[env], psBands, iidQuant_fx, 7, 15, iidIdxCoarse[env]); + errIIDFine += quantizeCoef(iid[env], psBands, iidQuantFine_fx, 15, 31, + iidIdxFine[env]); } /* normalize error to number of envelopes, ps bands errIID /= psBands*nEnvelopes; errIIDFine /= psBands*nEnvelopes; */ - /* Check if IID coefficients should be used in this frame */ psData->iidEnable = 0; - for(env=0;env<nEnvelopes; env++) { - for(band=0;band<psBands;band++) { - loudnDiff += fixp_abs(iidIdxCoarse[env][band]); - iidTransmit ++; + for (env = 0; env < nEnvelopes; env++) { + for (band = 0; band < psBands; band++) { + loudnDiff += fixp_abs(iidIdxCoarse[env][band]); + iidTransmit++; } } - if(loudnDiff > fMultI(FL2FXCONST_DBL(0.7f),iidTransmit)){ /* 0.7f empiric value */ + if (loudnDiff > + fMultI(FL2FXCONST_DBL(0.7f), iidTransmit)) { /* 0.7f empiric value */ psData->iidEnable = 1; } /* if iid not active -> RESET data */ - if(psData->iidEnable==0) { + if (psData->iidEnable == 0) { psData->iidTimeCnt = MAX_TIME_DIFF_FRAMES; - for(env=0;env<nEnvelopes; env++) { + for (env = 0; env < nEnvelopes; env++) { psData->iidDiffMode[env] = PS_DELTA_FREQ; - FDKmemclear(psData->iidIdx[env], sizeof(INT)*psBands); + FDKmemclear(psData->iidIdx[env], sizeof(INT) * psBands); } return; } /* count COARSE quantization bits for first envelope*/ - bitsIidFreq = FDKsbrEnc_EncodeIid(NULL, iidIdxCoarse[0], NULL, psBands, PS_IID_RES_COARSE, PS_DELTA_FREQ, &error); + bitsIidFreq = FDKsbrEnc_EncodeIid(NULL, iidIdxCoarse[0], NULL, psBands, + PS_IID_RES_COARSE, PS_DELTA_FREQ, &error); - if( (psData->iidTimeCnt>=MAX_TIME_DIFF_FRAMES) || (psData->iidQuantModeLast==PS_IID_RES_FINE) ) { - bitsIidTime = DO_NOT_USE_THIS_MODE; - } - else { - bitsIidTime = FDKsbrEnc_EncodeIid(NULL, iidIdxCoarse[0], psData->iidIdxLast, psBands, PS_IID_RES_COARSE, PS_DELTA_TIME, &error); + if ((psData->iidTimeCnt >= MAX_TIME_DIFF_FRAMES) || + (psData->iidQuantModeLast == PS_IID_RES_FINE)) { + bitsIidTime = DO_NOT_USE_THIS_MODE; + } else { + bitsIidTime = + FDKsbrEnc_EncodeIid(NULL, iidIdxCoarse[0], psData->iidIdxLast, psBands, + PS_IID_RES_COARSE, PS_DELTA_TIME, &error); } /* decision DELTA_FREQ vs DELTA_TIME */ - if(bitsIidTime>bitsIidFreq) { - diffMode[0] = PS_DELTA_FREQ; + if (bitsIidTime > bitsIidFreq) { + diffMode[0] = PS_DELTA_FREQ; bitsCoarseTot = bitsIidFreq; - } - else { - diffMode[0] = PS_DELTA_TIME; + } else { + diffMode[0] = PS_DELTA_TIME; bitsCoarseTot = bitsIidTime; } /* count COARSE quantization bits for following envelopes*/ - for(env=1;env<nEnvelopes; env++) { - bitsIidFreq = FDKsbrEnc_EncodeIid(NULL, iidIdxCoarse[env], NULL, psBands, PS_IID_RES_COARSE, PS_DELTA_FREQ, &error); - bitsIidTime = FDKsbrEnc_EncodeIid(NULL, iidIdxCoarse[env], iidIdxCoarse[env-1], psBands, PS_IID_RES_COARSE, PS_DELTA_TIME, &error); + for (env = 1; env < nEnvelopes; env++) { + bitsIidFreq = FDKsbrEnc_EncodeIid(NULL, iidIdxCoarse[env], NULL, psBands, + PS_IID_RES_COARSE, PS_DELTA_FREQ, &error); + bitsIidTime = + FDKsbrEnc_EncodeIid(NULL, iidIdxCoarse[env], iidIdxCoarse[env - 1], + psBands, PS_IID_RES_COARSE, PS_DELTA_TIME, &error); /* decision DELTA_FREQ vs DELTA_TIME */ - if(bitsIidTime>bitsIidFreq) { - diffMode[env] = PS_DELTA_FREQ; + if (bitsIidTime > bitsIidFreq) { + diffMode[env] = PS_DELTA_FREQ; bitsCoarseTot += bitsIidFreq; - } - else { - diffMode[env] = PS_DELTA_TIME; + } else { + diffMode[env] = PS_DELTA_TIME; bitsCoarseTot += bitsIidTime; } } - /* count FINE quantization bits for first envelope*/ - bitsIidFreq = FDKsbrEnc_EncodeIid(NULL, iidIdxFine[0], NULL, psBands, PS_IID_RES_FINE, PS_DELTA_FREQ, &error); + bitsIidFreq = FDKsbrEnc_EncodeIid(NULL, iidIdxFine[0], NULL, psBands, + PS_IID_RES_FINE, PS_DELTA_FREQ, &error); - if( (psData->iidTimeCnt>=MAX_TIME_DIFF_FRAMES) || (psData->iidQuantModeLast==PS_IID_RES_COARSE) ) { + if ((psData->iidTimeCnt >= MAX_TIME_DIFF_FRAMES) || + (psData->iidQuantModeLast == PS_IID_RES_COARSE)) { bitsIidTime = DO_NOT_USE_THIS_MODE; - } - else { - bitsIidTime = FDKsbrEnc_EncodeIid(NULL, iidIdxFine[0], psData->iidIdxLast, psBands, PS_IID_RES_FINE, PS_DELTA_TIME, &error); + } else { + bitsIidTime = + FDKsbrEnc_EncodeIid(NULL, iidIdxFine[0], psData->iidIdxLast, psBands, + PS_IID_RES_FINE, PS_DELTA_TIME, &error); } /* decision DELTA_FREQ vs DELTA_TIME */ - if(bitsIidTime>bitsIidFreq) { - diffModeFine[0] = PS_DELTA_FREQ; - bitsFineTot = bitsIidFreq; - } - else { - diffModeFine[0] = PS_DELTA_TIME; - bitsFineTot = bitsIidTime; + if (bitsIidTime > bitsIidFreq) { + diffModeFine[0] = PS_DELTA_FREQ; + bitsFineTot = bitsIidFreq; + } else { + diffModeFine[0] = PS_DELTA_TIME; + bitsFineTot = bitsIidTime; } /* count FINE quantization bits for following envelopes*/ - for(env=1;env<nEnvelopes; env++) { - bitsIidFreq = FDKsbrEnc_EncodeIid(NULL, iidIdxFine[env], NULL, psBands, PS_IID_RES_FINE, PS_DELTA_FREQ, &error); - bitsIidTime = FDKsbrEnc_EncodeIid(NULL, iidIdxFine[env], iidIdxFine[env-1], psBands, PS_IID_RES_FINE, PS_DELTA_TIME, &error); + for (env = 1; env < nEnvelopes; env++) { + bitsIidFreq = FDKsbrEnc_EncodeIid(NULL, iidIdxFine[env], NULL, psBands, + PS_IID_RES_FINE, PS_DELTA_FREQ, &error); + bitsIidTime = + FDKsbrEnc_EncodeIid(NULL, iidIdxFine[env], iidIdxFine[env - 1], psBands, + PS_IID_RES_FINE, PS_DELTA_TIME, &error); /* decision DELTA_FREQ vs DELTA_TIME */ - if(bitsIidTime>bitsIidFreq) { - diffModeFine[env] = PS_DELTA_FREQ; + if (bitsIidTime > bitsIidFreq) { + diffModeFine[env] = PS_DELTA_FREQ; bitsFineTot += bitsIidFreq; - } - else { - diffModeFine[env] = PS_DELTA_TIME; - bitsFineTot += bitsIidTime; + } else { + diffModeFine[env] = PS_DELTA_TIME; + bitsFineTot += bitsIidTime; } } - if(bitsFineTot == bitsCoarseTot){ - /* if same number of bits is needed, use the quantization with lower error */ - if(errIIDFine < errIID){ + if (bitsFineTot == bitsCoarseTot) { + /* if same number of bits is needed, use the quantization with lower error + */ + if (errIIDFine < errIID) { bitsCoarseTot = DO_NOT_USE_THIS_MODE; } else { bitsFineTot = DO_NOT_USE_THIS_MODE; } } else { - /* const FIXP_DBL minThreshold = FL2FXCONST_DBL(0.2f/(IID_SCALE_FT*PS_QUANT_SCALE_FT)*(psBands*nEnvelopes)); */ - const FIXP_DBL minThreshold = (FIXP_DBL)((LONG)0x00019999 * (psBands*nEnvelopes)); + /* const FIXP_DBL minThreshold = + * FL2FXCONST_DBL(0.2f/(IID_SCALE_FT*PS_QUANT_SCALE_FT)*(psBands*nEnvelopes)); + */ + const FIXP_DBL minThreshold = + (FIXP_DBL)((LONG)0x00019999 * (psBands * nEnvelopes)); /* decision RES_FINE vs RES_COARSE */ /* test if errIIDFine*quantErrorThreshold < errIID */ /* shiftVal 2 comes from scaling of quantErrorThreshold */ - if(fixMax(((errIIDFine>>1)+(minThreshold>>1))>>1, fMult(quantErrorThreshold,errIIDFine)) < (errIID>>2) ) { + if (fixMax(((errIIDFine >> 1) + (minThreshold >> 1)) >> 1, + fMult(quantErrorThreshold, errIIDFine)) < (errIID >> 2)) { bitsCoarseTot = DO_NOT_USE_THIS_MODE; - } - else if(fixMax(((errIID>>1)+(minThreshold>>1))>>1, fMult(quantErrorThreshold,errIID)) < (errIIDFine>>2) ) { + } else if (fixMax(((errIID >> 1) + (minThreshold >> 1)) >> 1, + fMult(quantErrorThreshold, errIID)) < (errIIDFine >> 2)) { bitsFineTot = DO_NOT_USE_THIS_MODE; } } /* decision RES_FINE vs RES_COARSE */ - if(bitsFineTot<bitsCoarseTot) { + if (bitsFineTot < bitsCoarseTot) { psData->iidQuantMode = PS_IID_RES_FINE; - for(env=0;env<nEnvelopes; env++) { + for (env = 0; env < nEnvelopes; env++) { psData->iidDiffMode[env] = diffModeFine[env]; - FDKmemcpy(psData->iidIdx[env], iidIdxFine[env], psBands*sizeof(INT)); + FDKmemcpy(psData->iidIdx[env], iidIdxFine[env], psBands * sizeof(INT)); } - } - else { + } else { psData->iidQuantMode = PS_IID_RES_COARSE; - for(env=0;env<nEnvelopes; env++) { + for (env = 0; env < nEnvelopes; env++) { psData->iidDiffMode[env] = diffMode[env]; - FDKmemcpy(psData->iidIdx[env], iidIdxCoarse[env], psBands*sizeof(INT)); + FDKmemcpy(psData->iidIdx[env], iidIdxCoarse[env], psBands * sizeof(INT)); } } /* Count DELTA_TIME encoding streaks */ - for(env=0;env<nEnvelopes; env++) { - if(psData->iidDiffMode[env]==PS_DELTA_TIME) + for (env = 0; env < nEnvelopes; env++) { + if (psData->iidDiffMode[env] == PS_DELTA_TIME) psData->iidTimeCnt++; else - psData->iidTimeCnt=0; + psData->iidTimeCnt = 0; } } - -static INT similarIid(PS_DATA *psData, - const INT psBands, - const INT nEnvelopes) -{ +static INT similarIid(PS_DATA *psData, const INT psBands, + const INT nEnvelopes) { const INT diffThr = (psData->iidQuantMode == PS_IID_RES_COARSE) ? 2 : 3; - const INT sumDiffThr = diffThr * psBands/4; + const INT sumDiffThr = diffThr * psBands / 4; INT similar = 0; - INT diff = 0; + INT diff = 0; INT sumDiff = 0; INT env = 0; - INT b = 0; - if ((nEnvelopes == psData->nEnvelopesLast) && (nEnvelopes==1)) { + INT b = 0; + if ((nEnvelopes == psData->nEnvelopesLast) && (nEnvelopes == 1)) { similar = 1; - for (env=0; env<nEnvelopes; env++) { + for (env = 0; env < nEnvelopes; env++) { sumDiff = 0; b = 0; do { diff = fixp_abs(psData->iidIdx[env][b] - psData->iidIdxLast[b]); sumDiff += diff; - if ( (diff > diffThr) /* more than x quantization steps in any band */ - || (sumDiff > sumDiffThr) ) { /* more than x quantisations steps overall difference */ + if ((diff > diffThr) /* more than x quantization steps in any band */ + || (sumDiff > sumDiffThr)) { /* more than x quantisations steps + overall difference */ similar = 0; } b++; - } while ((b<psBands) && (similar>0)); + } while ((b < psBands) && (similar > 0)); } } /* nEnvelopes==1 */ return similar; } - -static INT similarIcc(PS_DATA *psData, - const INT psBands, - const INT nEnvelopes) -{ +static INT similarIcc(PS_DATA *psData, const INT psBands, + const INT nEnvelopes) { const INT diffThr = 2; - const INT sumDiffThr = diffThr * psBands/4; + const INT sumDiffThr = diffThr * psBands / 4; INT similar = 0; - INT diff = 0; + INT diff = 0; INT sumDiff = 0; INT env = 0; - INT b = 0; - if ((nEnvelopes == psData->nEnvelopesLast) && (nEnvelopes==1)) { + INT b = 0; + if ((nEnvelopes == psData->nEnvelopesLast) && (nEnvelopes == 1)) { similar = 1; - for (env=0; env<nEnvelopes; env++) { + for (env = 0; env < nEnvelopes; env++) { sumDiff = 0; b = 0; do { diff = fixp_abs(psData->iccIdx[env][b] - psData->iccIdxLast[b]); sumDiff += diff; - if ( (diff > diffThr) /* more than x quantisation step in any band */ - || (sumDiff > sumDiffThr) ) { /* more than x quantisations steps overall difference */ + if ((diff > diffThr) /* more than x quantisation step in any band */ + || (sumDiff > sumDiffThr)) { /* more than x quantisations steps + overall difference */ similar = 0; } b++; - } while ((b<psBands) && (similar>0)); + } while ((b < psBands) && (similar > 0)); } } /* nEnvelopes==1 */ return similar; } -static void processIccData(PS_DATA *psData, - FIXP_DBL icc[PS_MAX_ENVELOPES][PS_MAX_BANDS], /* const input values: unable to declare as const, since it does not poINT to const memory */ - const INT psBands, - const INT nEnvelopes) -{ +static void processIccData( + PS_DATA *psData, + FIXP_DBL icc[PS_MAX_ENVELOPES][PS_MAX_BANDS], /* const input values: + unable to declare as + const, since it does + not poINT to const + memory */ + const INT psBands, const INT nEnvelopes) { FIXP_DBL errICC = FL2FXCONST_DBL(0.f); - INT env, band; - INT bitsIccFreq, bitsIccTime; - INT error = 0; - INT inCoherence=0, iccTransmit=0; - INT *iccIdxLast; + INT env, band; + INT bitsIccFreq, bitsIccTime; + INT error = 0; + INT inCoherence = 0, iccTransmit = 0; + INT *iccIdxLast; iccIdxLast = psData->iccIdxLast; /* Quantize ICC coefficients */ - for(env=0;env<nEnvelopes; env++) { - errICC += quantizeCoef(icc[env], psBands, iccQuant, 0, 8, psData->iccIdx[env]); + for (env = 0; env < nEnvelopes; env++) { + errICC += + quantizeCoef(icc[env], psBands, iccQuant, 0, 8, psData->iccIdx[env]); } /* Check if ICC coefficients should be used */ psData->iccEnable = 0; - for(env=0;env<nEnvelopes; env++) { - for(band=0;band<psBands;band++) { + for (env = 0; env < nEnvelopes; env++) { + for (band = 0; band < psBands; band++) { inCoherence += psData->iccIdx[env][band]; - iccTransmit ++; + iccTransmit++; } } - if(inCoherence > fMultI(FL2FXCONST_DBL(0.5f),iccTransmit)){ /* 0.5f empiric value */ + if (inCoherence > + fMultI(FL2FXCONST_DBL(0.5f), iccTransmit)) { /* 0.5f empiric value */ psData->iccEnable = 1; } - if(psData->iccEnable==0) { + if (psData->iccEnable == 0) { psData->iccTimeCnt = MAX_TIME_DIFF_FRAMES; - for(env=0;env<nEnvelopes; env++) { + for (env = 0; env < nEnvelopes; env++) { psData->iccDiffMode[env] = PS_DELTA_FREQ; - FDKmemclear(psData->iccIdx[env], sizeof(INT)*psBands); + FDKmemclear(psData->iccIdx[env], sizeof(INT) * psBands); } return; } - for(env=0;env<nEnvelopes; env++) { - bitsIccFreq = FDKsbrEnc_EncodeIcc(NULL, psData->iccIdx[env], NULL, psBands, PS_DELTA_FREQ, &error); + for (env = 0; env < nEnvelopes; env++) { + bitsIccFreq = FDKsbrEnc_EncodeIcc(NULL, psData->iccIdx[env], NULL, psBands, + PS_DELTA_FREQ, &error); - if(psData->iccTimeCnt<MAX_TIME_DIFF_FRAMES) { - bitsIccTime = FDKsbrEnc_EncodeIcc(NULL, psData->iccIdx[env], iccIdxLast, psBands, PS_DELTA_TIME, &error); - } - else { - bitsIccTime = DO_NOT_USE_THIS_MODE; + if (psData->iccTimeCnt < MAX_TIME_DIFF_FRAMES) { + bitsIccTime = FDKsbrEnc_EncodeIcc(NULL, psData->iccIdx[env], iccIdxLast, + psBands, PS_DELTA_TIME, &error); + } else { + bitsIccTime = DO_NOT_USE_THIS_MODE; } - if(bitsIccFreq>bitsIccTime) { + if (bitsIccFreq > bitsIccTime) { psData->iccDiffMode[env] = PS_DELTA_TIME; psData->iccTimeCnt++; - } - else { + } else { psData->iccDiffMode[env] = PS_DELTA_FREQ; - psData->iccTimeCnt=0; + psData->iccTimeCnt = 0; } iccIdxLast = psData->iccIdx[env]; } @@ -639,163 +642,148 @@ static void processIccData(PS_DATA *psData, static void calculateIID(FIXP_DBL ldPwrL[PS_MAX_ENVELOPES][PS_MAX_BANDS], FIXP_DBL ldPwrR[PS_MAX_ENVELOPES][PS_MAX_BANDS], FIXP_DBL iid[PS_MAX_ENVELOPES][PS_MAX_BANDS], - INT nEnvelopes, - INT psBands) -{ - INT i=0; - INT env=0; - for(env=0; env<nEnvelopes;env++) { - for (i=0; i<psBands; i++) { - + INT nEnvelopes, INT psBands) { + INT i = 0; + INT env = 0; + for (env = 0; env < nEnvelopes; env++) { + for (i = 0; i < psBands; i++) { /* iid[env][i] = 10.0f*(float)log10(pwrL[env][i]/pwrR[env][i]); - */ - FIXP_DBL IID = fMultDiv2( FL2FXCONST_DBL(LOG10_2_10/IID_SCALE_FT), (ldPwrL[env][i]-ldPwrR[env][i]) ); + */ + FIXP_DBL IID = fMultDiv2(FL2FXCONST_DBL(LOG10_2_10 / IID_SCALE_FT), + (ldPwrL[env][i] - ldPwrR[env][i])); - IID = fixMin( IID, (FIXP_DBL)(MAXVAL_DBL>>(LD_DATA_SHIFT+1)) ); - IID = fixMax( IID, (FIXP_DBL)(MINVAL_DBL>>(LD_DATA_SHIFT+1)) ); - iid[env][i] = IID << (LD_DATA_SHIFT+1); + IID = fixMin(IID, (FIXP_DBL)(MAXVAL_DBL >> (LD_DATA_SHIFT + 1))); + IID = fixMax(IID, (FIXP_DBL)(MINVAL_DBL >> (LD_DATA_SHIFT + 1))); + iid[env][i] = IID << (LD_DATA_SHIFT + 1); } } } -static void calculateICC(FIXP_DBL ldPwrL[PS_MAX_ENVELOPES][PS_MAX_BANDS], - FIXP_DBL ldPwrR[PS_MAX_ENVELOPES][PS_MAX_BANDS], +static void calculateICC(FIXP_DBL pwrL[PS_MAX_ENVELOPES][PS_MAX_BANDS], + FIXP_DBL pwrR[PS_MAX_ENVELOPES][PS_MAX_BANDS], FIXP_DBL pwrCr[PS_MAX_ENVELOPES][PS_MAX_BANDS], FIXP_DBL pwrCi[PS_MAX_ENVELOPES][PS_MAX_BANDS], FIXP_DBL icc[PS_MAX_ENVELOPES][PS_MAX_BANDS], - INT nEnvelopes, - INT psBands) -{ + INT nEnvelopes, INT psBands) { INT i = 0; INT env = 0; INT border = psBands; switch (psBands) { - case PS_BANDS_COARSE: - border = 5; - break; - case PS_BANDS_MID: - border = 11; - break; - default: - break; + case PS_BANDS_COARSE: + border = 5; + break; + case PS_BANDS_MID: + border = 11; + break; + default: + break; } - for(env=0; env<nEnvelopes;env++) { - for (i=0; i<border; i++) { - - /* icc[env][i] = min( pwrCr[env][i] / (float) sqrt(pwrL[env][i] * pwrR[env][i]) , 1.f); - */ - FIXP_DBL ICC, invNrg = CalcInvLdData ( -((ldPwrL[env][i]>>1) + (ldPwrR[env][i]>>1) + (FIXP_DBL)1) ); - INT scale, invScale = CountLeadingBits(invNrg); - - scale = (DFRACT_BITS-1) - invScale; - ICC = fMult(pwrCr[env][i], invNrg<<invScale) ; - icc[env][i] = SATURATE_LEFT_SHIFT(ICC, scale, DFRACT_BITS); + for (env = 0; env < nEnvelopes; env++) { + for (i = 0; i < border; i++) { + /* icc[env][i] = min( pwrCr[env][i] / (float) sqrt(pwrL[env][i] * + * pwrR[env][i]) , 1.f); + */ + int scale; + FIXP_DBL invNrg = invSqrtNorm2( + fMax(fMult(pwrL[env][i], pwrR[env][i]), (FIXP_DBL)1), &scale); + icc[env][i] = + SATURATE_LEFT_SHIFT(fMult(pwrCr[env][i], invNrg), scale, DFRACT_BITS); } - for (; i<psBands; i++) { - INT sc1, sc2; - FIXP_DBL cNrgR, cNrgI, ICC; - - sc1 = CountLeadingBits( fixMax(fixp_abs(pwrCr[env][i]),fixp_abs(pwrCi[env][i])) ) ; - cNrgR = fPow2Div2((pwrCr[env][i]<<sc1)); /* squared nrg's expect explicit scaling */ - cNrgI = fPow2Div2((pwrCi[env][i]<<sc1)); - - ICC = CalcInvLdData( (CalcLdData((cNrgR + cNrgI)>>1)>>1) - (FIXP_DBL)((sc1-1)<<(DFRACT_BITS-1-LD_DATA_SHIFT)) ); - - FIXP_DBL invNrg = CalcInvLdData ( -((ldPwrL[env][i]>>1) + (ldPwrR[env][i]>>1) + (FIXP_DBL)1) ); - sc1 = CountLeadingBits(invNrg); - invNrg <<= sc1; - - sc2 = CountLeadingBits(ICC); - ICC = fMult(ICC<<sc2,invNrg); - - sc1 = ( (DFRACT_BITS-1) - sc1 - sc2 ); - if (sc1 < 0) { - ICC >>= -sc1; + for (; i < psBands; i++) { + int denom_e; + FIXP_DBL denom_m = fMultNorm(pwrL[env][i], pwrR[env][i], &denom_e); + + if (denom_m == (FIXP_DBL)0) { + icc[env][i] = (FIXP_DBL)MAXVAL_DBL; + } else { + int num_e, result_e; + FIXP_DBL num_m, result_m; + + num_e = CountLeadingBits( + fixMax(fixp_abs(pwrCr[env][i]), fixp_abs(pwrCi[env][i]))); + num_m = fPow2Div2((pwrCr[env][i] << num_e)) + + fPow2Div2((pwrCi[env][i] << num_e)); + + result_m = fDivNorm(num_m, denom_m, &result_e); + result_e += (-2 * num_e + 1) - denom_e; + icc[env][i] = scaleValueSaturate(sqrtFixp(result_m >> (result_e & 1)), + (result_e + (result_e & 1)) >> 1); } - else { - if (ICC >= ((FIXP_DBL)MAXVAL_DBL>>sc1) ) - ICC = (FIXP_DBL)MAXVAL_DBL; - else - ICC <<= sc1; - } - - icc[env][i] = ICC; } } } -void FDKsbrEnc_initPsBandNrgScale(HANDLE_PS_ENCODE hPsEncode) -{ +void FDKsbrEnc_initPsBandNrgScale(HANDLE_PS_ENCODE hPsEncode) { INT group, bin; - INT nIidGroups = hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups; + INT nIidGroups = hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups; - FDKmemclear(hPsEncode->psBandNrgScale, PS_MAX_BANDS*sizeof(SCHAR)); + FDKmemclear(hPsEncode->psBandNrgScale, PS_MAX_BANDS * sizeof(SCHAR)); - for (group=0; group < nIidGroups; group++) { + for (group = 0; group < nIidGroups; group++) { /* Translate group to bin */ bin = hPsEncode->subband2parameterIndex[group]; /* Translate from 20 bins to 10 bins */ if (hPsEncode->psEncMode == PS_BANDS_COARSE) { - bin = bin>>1; + bin = bin >> 1; } - hPsEncode->psBandNrgScale[bin] = (hPsEncode->psBandNrgScale[bin]==0) - ? (hPsEncode->iidGroupWidthLd[group] + 5) - : (fixMax(hPsEncode->iidGroupWidthLd[group],hPsEncode->psBandNrgScale[bin]) + 1) ; - + hPsEncode->psBandNrgScale[bin] = + (hPsEncode->psBandNrgScale[bin] == 0) + ? (hPsEncode->iidGroupWidthLd[group] + 5) + : (fixMax(hPsEncode->iidGroupWidthLd[group], + hPsEncode->psBandNrgScale[bin]) + + 1); } } -FDK_PSENC_ERROR FDKsbrEnc_CreatePSEncode( - HANDLE_PS_ENCODE *phPsEncode - ) -{ +FDK_PSENC_ERROR FDKsbrEnc_CreatePSEncode(HANDLE_PS_ENCODE *phPsEncode) { FDK_PSENC_ERROR error = PSENC_OK; - if (phPsEncode==NULL) { + if (phPsEncode == NULL) { error = PSENC_INVALID_HANDLE; - } - else { + } else { HANDLE_PS_ENCODE hPsEncode = NULL; - if (NULL==(hPsEncode = GetRam_PsEncode())) { + if (NULL == (hPsEncode = GetRam_PsEncode())) { error = PSENC_MEMORY_ERROR; goto bail; } - FDKmemclear(hPsEncode,sizeof(PS_ENCODE)); + FDKmemclear(hPsEncode, sizeof(PS_ENCODE)); *phPsEncode = hPsEncode; /* return allocated handle */ } bail: return error; } -FDK_PSENC_ERROR FDKsbrEnc_InitPSEncode( - HANDLE_PS_ENCODE hPsEncode, - const PS_BANDS psEncMode, - const FIXP_DBL iidQuantErrorThreshold - ) -{ +FDK_PSENC_ERROR FDKsbrEnc_InitPSEncode(HANDLE_PS_ENCODE hPsEncode, + const PS_BANDS psEncMode, + const FIXP_DBL iidQuantErrorThreshold) { FDK_PSENC_ERROR error = PSENC_OK; - if (NULL==hPsEncode) { + if (NULL == hPsEncode) { error = PSENC_INVALID_HANDLE; - } - else { + } else { if (PSENC_OK != (InitPSData(&hPsEncode->psData))) { goto bail; } - switch(psEncMode){ + switch (psEncMode) { case PS_BANDS_COARSE: case PS_BANDS_MID: - hPsEncode->nQmfIidGroups = QMF_GROUPS_LO_RES; + hPsEncode->nQmfIidGroups = QMF_GROUPS_LO_RES; hPsEncode->nSubQmfIidGroups = SUBQMF_GROUPS_LO_RES; - FDKmemcpy(hPsEncode->iidGroupBorders, iidGroupBordersLoRes, (hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups + 1)*sizeof(INT)); - FDKmemcpy(hPsEncode->subband2parameterIndex, subband2parameter20, (hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups) *sizeof(INT)); - FDKmemcpy(hPsEncode->iidGroupWidthLd, iidGroupWidthLdLoRes, (hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups) *sizeof(UCHAR)); + FDKmemcpy(hPsEncode->iidGroupBorders, iidGroupBordersLoRes, + (hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups + 1) * + sizeof(INT)); + FDKmemcpy(hPsEncode->subband2parameterIndex, subband2parameter20, + (hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups) * + sizeof(INT)); + FDKmemcpy(hPsEncode->iidGroupWidthLd, iidGroupWidthLdLoRes, + (hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups) * + sizeof(UCHAR)); break; default: error = PSENC_INIT_ERROR; @@ -810,14 +798,10 @@ bail: return error; } - -FDK_PSENC_ERROR FDKsbrEnc_DestroyPSEncode( - HANDLE_PS_ENCODE *phPsEncode - ) -{ +FDK_PSENC_ERROR FDKsbrEnc_DestroyPSEncode(HANDLE_PS_ENCODE *phPsEncode) { FDK_PSENC_ERROR error = PSENC_OK; - if (NULL !=phPsEncode) { + if (NULL != phPsEncode) { FreeRam_PsEncode(phPsEncode); } @@ -834,49 +818,43 @@ typedef struct { } PS_PWR_DATA; - FDK_PSENC_ERROR FDKsbrEnc_PSEncode( - HANDLE_PS_ENCODE hPsEncode, - HANDLE_PS_OUT hPsOut, - UCHAR *dynBandScale, - UINT maxEnvelopes, - FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2], - const INT frameSize, - const INT sendHeader - ) -{ + HANDLE_PS_ENCODE hPsEncode, HANDLE_PS_OUT hPsOut, UCHAR *dynBandScale, + UINT maxEnvelopes, + FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2], + const INT frameSize, const INT sendHeader) { FDK_PSENC_ERROR error = PSENC_OK; HANDLE_PS_DATA hPsData = &hPsEncode->psData; - FIXP_DBL iid [PS_MAX_ENVELOPES][PS_MAX_BANDS]; - FIXP_DBL icc [PS_MAX_ENVELOPES][PS_MAX_BANDS]; - int envBorder[PS_MAX_ENVELOPES+1]; + FIXP_DBL iid[PS_MAX_ENVELOPES][PS_MAX_BANDS]; + FIXP_DBL icc[PS_MAX_ENVELOPES][PS_MAX_BANDS]; + int envBorder[PS_MAX_ENVELOPES + 1]; int group, bin, col, subband, band; int i = 0; int env = 0; - int psBands = (int) hPsEncode->psEncMode; - int nIidGroups = hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups; - int nEnvelopes = fixMin(maxEnvelopes, (UINT)PS_MAX_ENVELOPES); + int psBands = (int)hPsEncode->psEncMode; + int nIidGroups = hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups; + int nEnvelopes = fixMin(maxEnvelopes, (UINT)PS_MAX_ENVELOPES); - C_ALLOC_SCRATCH_START(pwrData, PS_PWR_DATA, 1); + C_ALLOC_SCRATCH_START(pwrData, PS_PWR_DATA, 1) - for(env=0; env<nEnvelopes+1;env++) { - envBorder[env] = fMultI(GetInvInt(nEnvelopes),frameSize*env); + for (env = 0; env < nEnvelopes + 1; env++) { + envBorder[env] = fMultI(GetInvInt(nEnvelopes), frameSize * env); } - for(env=0; env<nEnvelopes;env++) { - + for (env = 0; env < nEnvelopes; env++) { /* clear energy array */ - for (band=0; band<psBands; band++) { - pwrData->pwrL[env][band] = pwrData->pwrR[env][band] = pwrData->pwrCr[env][band] = pwrData->pwrCi[env][band] = FIXP_DBL(1); + for (band = 0; band < psBands; band++) { + pwrData->pwrL[env][band] = pwrData->pwrR[env][band] = + pwrData->pwrCr[env][band] = pwrData->pwrCi[env][band] = FIXP_DBL(1); } /**** calculate energies and correlation ****/ /* start with hybrid data */ - for (group=0; group < nIidGroups; group++) { + for (group = 0; group < nIidGroups; group++) { /* Translate group to bin */ bin = hPsEncode->subband2parameterIndex[group]; @@ -894,22 +872,26 @@ FDK_PSENC_ERROR FDKsbrEnc_PSEncode( FIXP_DBL pwrCi_env_bin = pwrData->pwrCi[env][bin]; int scale = (int)dynBandScale[bin]; - for (col=envBorder[env]; col<envBorder[env+1]; col++) { - for (subband = hPsEncode->iidGroupBorders[group]; subband < hPsEncode->iidGroupBorders[group+1]; subband++) { - FIXP_QMF l_real = (hybridData[col][0][0][subband]) << scale; - FIXP_QMF l_imag = (hybridData[col][0][1][subband]) << scale; - FIXP_QMF r_real = (hybridData[col][1][0][subband]) << scale; - FIXP_QMF r_imag = (hybridData[col][1][1][subband]) << scale; - - pwrL_env_bin += (fPow2Div2(l_real) + fPow2Div2(l_imag)) >> bScale; - pwrR_env_bin += (fPow2Div2(r_real) + fPow2Div2(r_imag)) >> bScale; - pwrCr_env_bin += (fMultDiv2(l_real, r_real) + fMultDiv2(l_imag, r_imag)) >> bScale; - pwrCi_env_bin += (fMultDiv2(r_real, l_imag) - fMultDiv2(l_real, r_imag)) >> bScale; + for (col = envBorder[env]; col < envBorder[env + 1]; col++) { + for (subband = hPsEncode->iidGroupBorders[group]; + subband < hPsEncode->iidGroupBorders[group + 1]; subband++) { + FIXP_DBL l_real = (hybridData[col][0][0][subband]) << scale; + FIXP_DBL l_imag = (hybridData[col][0][1][subband]) << scale; + FIXP_DBL r_real = (hybridData[col][1][0][subband]) << scale; + FIXP_DBL r_imag = (hybridData[col][1][1][subband]) << scale; + + pwrL_env_bin += (fPow2Div2(l_real) + fPow2Div2(l_imag)) >> bScale; + pwrR_env_bin += (fPow2Div2(r_real) + fPow2Div2(r_imag)) >> bScale; + pwrCr_env_bin += + (fMultDiv2(l_real, r_real) + fMultDiv2(l_imag, r_imag)) >> bScale; + pwrCi_env_bin += + (fMultDiv2(r_real, l_imag) - fMultDiv2(l_real, r_imag)) >> bScale; } } - /* assure, nrg's of left and right channel are not negative; necessary on 16 bit multiply units */ - pwrData->pwrL[env][bin] = fixMax((FIXP_DBL)0,pwrL_env_bin); - pwrData->pwrR[env][bin] = fixMax((FIXP_DBL)0,pwrR_env_bin); + /* assure, nrg's of left and right channel are not negative; necessary on + * 16 bit multiply units */ + pwrData->pwrL[env][bin] = fixMax((FIXP_DBL)0, pwrL_env_bin); + pwrData->pwrR[env][bin] = fixMax((FIXP_DBL)0, pwrR_env_bin); pwrData->pwrCr[env][bin] = pwrCr_env_bin; pwrData->pwrCi[env][bin] = pwrCi_env_bin; @@ -924,126 +906,122 @@ FDK_PSENC_ERROR FDKsbrEnc_PSEncode( /* calculate iid and icc */ calculateIID(pwrData->ldPwrL, pwrData->ldPwrR, iid, nEnvelopes, psBands); - calculateICC(pwrData->ldPwrL, pwrData->ldPwrR, pwrData->pwrCr, pwrData->pwrCi, icc, nEnvelopes, psBands); + calculateICC(pwrData->pwrL, pwrData->pwrR, pwrData->pwrCr, pwrData->pwrCi, + icc, nEnvelopes, psBands); /*** Envelope Reduction ***/ - while (envelopeReducible(iid,icc,psBands,nEnvelopes)) { - int e=0; + while (envelopeReducible(iid, icc, psBands, nEnvelopes)) { + int e = 0; /* sum energies of two neighboring envelopes */ nEnvelopes >>= 1; - for (e=0; e<nEnvelopes; e++) { - FDKsbrEnc_addFIXP_DBL(pwrData->pwrL[2*e], pwrData->pwrL[2*e+1], pwrData->pwrL[e], psBands); - FDKsbrEnc_addFIXP_DBL(pwrData->pwrR[2*e], pwrData->pwrR[2*e+1], pwrData->pwrR[e], psBands); - FDKsbrEnc_addFIXP_DBL(pwrData->pwrCr[2*e],pwrData->pwrCr[2*e+1],pwrData->pwrCr[e],psBands); - FDKsbrEnc_addFIXP_DBL(pwrData->pwrCi[2*e],pwrData->pwrCi[2*e+1],pwrData->pwrCi[e],psBands); + for (e = 0; e < nEnvelopes; e++) { + FDKsbrEnc_addFIXP_DBL(pwrData->pwrL[2 * e], pwrData->pwrL[2 * e + 1], + pwrData->pwrL[e], psBands); + FDKsbrEnc_addFIXP_DBL(pwrData->pwrR[2 * e], pwrData->pwrR[2 * e + 1], + pwrData->pwrR[e], psBands); + FDKsbrEnc_addFIXP_DBL(pwrData->pwrCr[2 * e], pwrData->pwrCr[2 * e + 1], + pwrData->pwrCr[e], psBands); + FDKsbrEnc_addFIXP_DBL(pwrData->pwrCi[2 * e], pwrData->pwrCi[2 * e + 1], + pwrData->pwrCi[e], psBands); /* calc logarithmic energy */ LdDataVector(pwrData->pwrL[e], pwrData->ldPwrL[e], psBands); LdDataVector(pwrData->pwrR[e], pwrData->ldPwrR[e], psBands); /* reduce number of envelopes and adjust borders */ - envBorder[e] = envBorder[2*e]; + envBorder[e] = envBorder[2 * e]; } - envBorder[nEnvelopes] = envBorder[2*nEnvelopes]; + envBorder[nEnvelopes] = envBorder[2 * nEnvelopes]; /* re-calculate iid and icc */ calculateIID(pwrData->ldPwrL, pwrData->ldPwrR, iid, nEnvelopes, psBands); - calculateICC(pwrData->ldPwrL, pwrData->ldPwrR, pwrData->pwrCr, pwrData->pwrCi, icc, nEnvelopes, psBands); + calculateICC(pwrData->pwrL, pwrData->pwrR, pwrData->pwrCr, pwrData->pwrCi, + icc, nEnvelopes, psBands); } - /* */ - if(sendHeader) { - hPsData->headerCnt = MAX_PS_NOHEADER_CNT; + if (sendHeader) { + hPsData->headerCnt = MAX_PS_NOHEADER_CNT; hPsData->iidTimeCnt = MAX_TIME_DIFF_FRAMES; hPsData->iccTimeCnt = MAX_TIME_DIFF_FRAMES; - hPsData->noEnvCnt = MAX_NOENV_CNT; + hPsData->noEnvCnt = MAX_NOENV_CNT; } /*** Parameter processing, quantisation etc ***/ - processIidData(hPsData, iid, psBands, nEnvelopes, hPsEncode->iidQuantErrorThreshold); + processIidData(hPsData, iid, psBands, nEnvelopes, + hPsEncode->iidQuantErrorThreshold); processIccData(hPsData, icc, psBands, nEnvelopes); - /*** Initialize output struct ***/ /* PS Header on/off ? */ - if( (hPsData->headerCnt<MAX_PS_NOHEADER_CNT) - && ( (hPsData->iidQuantMode == hPsData->iidQuantModeLast) && (hPsData->iccQuantMode == hPsData->iccQuantModeLast) ) - && ( (hPsData->iidEnable == hPsData->iidEnableLast) && (hPsData->iccEnable == hPsData->iccEnableLast) ) ) { + if ((hPsData->headerCnt < MAX_PS_NOHEADER_CNT) && + ((hPsData->iidQuantMode == hPsData->iidQuantModeLast) && + (hPsData->iccQuantMode == hPsData->iccQuantModeLast)) && + ((hPsData->iidEnable == hPsData->iidEnableLast) && + (hPsData->iccEnable == hPsData->iccEnableLast))) { hPsOut->enablePSHeader = 0; - } - else { + } else { hPsOut->enablePSHeader = 1; hPsData->headerCnt = 0; } /* nEnvelopes = 0 ? */ - if ( (hPsData->noEnvCnt < MAX_NOENV_CNT) - && (similarIid(hPsData, psBands, nEnvelopes)) - && (similarIcc(hPsData, psBands, nEnvelopes)) ) { + if ((hPsData->noEnvCnt < MAX_NOENV_CNT) && + (similarIid(hPsData, psBands, nEnvelopes)) && + (similarIcc(hPsData, psBands, nEnvelopes))) { hPsOut->nEnvelopes = nEnvelopes = 0; hPsData->noEnvCnt++; } else { hPsData->noEnvCnt = 0; } + if (nEnvelopes > 0) { + hPsOut->enableIID = hPsData->iidEnable; + hPsOut->iidMode = getIIDMode(psBands, hPsData->iidQuantMode); - if (nEnvelopes>0) { - - hPsOut->enableIID = hPsData->iidEnable; - hPsOut->iidMode = getIIDMode(psBands, hPsData->iidQuantMode); + hPsOut->enableICC = hPsData->iccEnable; + hPsOut->iccMode = getICCMode(psBands, hPsData->iccQuantMode); - hPsOut->enableICC = hPsData->iccEnable; - hPsOut->iccMode = getICCMode(psBands, hPsData->iccQuantMode); + hPsOut->enableIpdOpd = 0; + hPsOut->frameClass = 0; + hPsOut->nEnvelopes = nEnvelopes; - hPsOut->enableIpdOpd = 0; - hPsOut->frameClass = 0; - hPsOut->nEnvelopes = nEnvelopes; - - for(env=0; env<nEnvelopes; env++) { - hPsOut->frameBorder[env] = envBorder[env+1]; - } - - for(env=0; env<hPsOut->nEnvelopes; env++) { + for (env = 0; env < nEnvelopes; env++) { + hPsOut->frameBorder[env] = envBorder[env + 1]; hPsOut->deltaIID[env] = (PS_DELTA)hPsData->iidDiffMode[env]; - - for(band=0; band<psBands; band++) { - hPsOut->iid[env][band] = hPsData->iidIdx[env][band]; - } - } - - for(env=0; env<hPsOut->nEnvelopes; env++) { hPsOut->deltaICC[env] = (PS_DELTA)hPsData->iccDiffMode[env]; - for(band=0; band<psBands; band++) { + for (band = 0; band < psBands; band++) { + hPsOut->iid[env][band] = hPsData->iidIdx[env][band]; hPsOut->icc[env][band] = hPsData->iccIdx[env][band]; } } /* IPD OPD not supported right now */ - FDKmemclear(hPsOut->ipd, PS_MAX_ENVELOPES*PS_MAX_BANDS*sizeof(PS_DELTA)); - for(env=0; env<PS_MAX_ENVELOPES; env++) { + FDKmemclear(hPsOut->ipd, + PS_MAX_ENVELOPES * PS_MAX_BANDS * sizeof(PS_DELTA)); + for (env = 0; env < PS_MAX_ENVELOPES; env++) { hPsOut->deltaIPD[env] = PS_DELTA_FREQ; hPsOut->deltaOPD[env] = PS_DELTA_FREQ; } - FDKmemclear(hPsOut->ipdLast, PS_MAX_BANDS*sizeof(INT)); - FDKmemclear(hPsOut->opdLast, PS_MAX_BANDS*sizeof(INT)); + FDKmemclear(hPsOut->ipdLast, PS_MAX_BANDS * sizeof(INT)); + FDKmemclear(hPsOut->opdLast, PS_MAX_BANDS * sizeof(INT)); - for(band=0; band<PS_MAX_BANDS; band++) { + for (band = 0; band < PS_MAX_BANDS; band++) { hPsOut->iidLast[band] = hPsData->iidIdxLast[band]; hPsOut->iccLast[band] = hPsData->iccIdxLast[band]; } /* save iids and iccs for differential time coding in the next frame */ - hPsData->nEnvelopesLast = nEnvelopes; - hPsData->iidEnableLast = hPsData->iidEnable; - hPsData->iccEnableLast = hPsData->iccEnable; + hPsData->nEnvelopesLast = nEnvelopes; + hPsData->iidEnableLast = hPsData->iidEnable; + hPsData->iccEnableLast = hPsData->iccEnable; hPsData->iidQuantModeLast = hPsData->iidQuantMode; hPsData->iccQuantModeLast = hPsData->iccQuantMode; - for (i=0; i<psBands; i++) { - hPsData->iidIdxLast[i] = hPsData->iidIdx[nEnvelopes-1][i]; - hPsData->iccIdxLast[i] = hPsData->iccIdx[nEnvelopes-1][i]; + for (i = 0; i < psBands; i++) { + hPsData->iidIdxLast[i] = hPsData->iidIdx[nEnvelopes - 1][i]; + hPsData->iccIdxLast[i] = hPsData->iccIdx[nEnvelopes - 1][i]; } } /* Envelope > 0 */ @@ -1051,4 +1029,3 @@ FDK_PSENC_ERROR FDKsbrEnc_PSEncode( return error; } - diff --git a/libSBRenc/src/ps_encode.h b/libSBRenc/src/ps_encode.h index f728d47..4237a00 100644 --- a/libSBRenc/src/ps_encode.h +++ b/libSBRenc/src/ps_encode.h @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,57 +90,57 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): M. Neuendorf, N. Rettelbach, M. Multrus -/***************************** MPEG Audio Encoder *************************** + Description: PS Parameter extraction, encoding - Initial author: M. Neuendorf, N. Rettelbach, M. Multrus - contents/description: PS Parameter extraction, encoding +*******************************************************************************/ -******************************************************************************/ /*! \file - \brief PS parameter extraction, encoding functions + \brief PS parameter extraction, encoding functions $Revision: 92790 $ */ -#ifndef __INCLUDED_PS_ENCODE_H -#define __INCLUDED_PS_ENCODE_H +#ifndef PS_ENCODE_H +#define PS_ENCODE_H #include "ps_const.h" #include "ps_bitenc.h" +#define IID_SCALE_FT (64.f) /* maxVal in Quant tab is +/- 50 */ +#define IID_SCALE 6 /* maxVal in Quant tab is +/- 50 */ +#define IID_MAXVAL (1 << IID_SCALE) -#define IID_SCALE_FT (64.f) /* maxVal in Quant tab is +/- 50 */ -#define IID_SCALE 6 /* maxVal in Quant tab is +/- 50 */ -#define IID_MAXVAL (1<<IID_SCALE) - -#define PS_QUANT_SCALE_FT (64.f) /* error smaller (64-25)/64 * 20 bands * 4 env -> QuantScale 64 */ -#define PS_QUANT_SCALE 6 /* error smaller (64-25)/64 * 20 bands * 4 env -> QuantScale 6 bit */ - - -#define QMF_GROUPS_LO_RES 12 -#define SUBQMF_GROUPS_LO_RES 10 -#define QMF_GROUPS_HI_RES 18 -#define SUBQMF_GROUPS_HI_RES 30 +#define PS_QUANT_SCALE_FT \ + (64.f) /* error smaller (64-25)/64 * 20 bands * 4 env -> QuantScale 64 */ +#define PS_QUANT_SCALE \ + 6 /* error smaller (64-25)/64 * 20 bands * 4 env -> QuantScale 6 bit */ +#define QMF_GROUPS_LO_RES 12 +#define SUBQMF_GROUPS_LO_RES 10 +#define QMF_GROUPS_HI_RES 18 +#define SUBQMF_GROUPS_HI_RES 30 typedef struct T_PS_DATA { - INT iidEnable; INT iidEnableLast; INT iidQuantMode; INT iidQuantModeLast; INT iidDiffMode[PS_MAX_ENVELOPES]; - INT iidIdx [PS_MAX_ENVELOPES][PS_MAX_BANDS]; - INT iidIdxLast [PS_MAX_BANDS]; + INT iidIdx[PS_MAX_ENVELOPES][PS_MAX_BANDS]; + INT iidIdxLast[PS_MAX_BANDS]; INT iccEnable; INT iccEnableLast; INT iccQuantMode; INT iccQuantModeLast; INT iccDiffMode[PS_MAX_ENVELOPES]; - INT iccIdx [PS_MAX_ENVELOPES][PS_MAX_BANDS]; - INT iccIdxLast [PS_MAX_BANDS]; + INT iccIdx[PS_MAX_ENVELOPES][PS_MAX_BANDS]; + INT iccIdxLast[PS_MAX_BANDS]; INT nEnvelopesLast; @@ -140,48 +151,35 @@ typedef struct T_PS_DATA { } PS_DATA, *HANDLE_PS_DATA; +typedef struct T_PS_ENCODE { + PS_DATA psData; -typedef struct T_PS_ENCODE{ - - PS_DATA psData; + PS_BANDS psEncMode; + INT nQmfIidGroups; + INT nSubQmfIidGroups; + INT iidGroupBorders[QMF_GROUPS_HI_RES + SUBQMF_GROUPS_HI_RES + 1]; + INT subband2parameterIndex[QMF_GROUPS_HI_RES + SUBQMF_GROUPS_HI_RES]; + UCHAR iidGroupWidthLd[QMF_GROUPS_HI_RES + SUBQMF_GROUPS_HI_RES]; + FIXP_DBL iidQuantErrorThreshold; - PS_BANDS psEncMode; - INT nQmfIidGroups; - INT nSubQmfIidGroups; - INT iidGroupBorders[QMF_GROUPS_HI_RES + SUBQMF_GROUPS_HI_RES + 1]; - INT subband2parameterIndex[QMF_GROUPS_HI_RES + SUBQMF_GROUPS_HI_RES]; - UCHAR iidGroupWidthLd[QMF_GROUPS_HI_RES + SUBQMF_GROUPS_HI_RES]; - FIXP_DBL iidQuantErrorThreshold; - - UCHAR psBandNrgScale [PS_MAX_BANDS]; + UCHAR psBandNrgScale[PS_MAX_BANDS]; } PS_ENCODE; - typedef struct T_PS_ENCODE *HANDLE_PS_ENCODE; -FDK_PSENC_ERROR FDKsbrEnc_CreatePSEncode( - HANDLE_PS_ENCODE *phPsEncode - ); +FDK_PSENC_ERROR FDKsbrEnc_CreatePSEncode(HANDLE_PS_ENCODE *phPsEncode); -FDK_PSENC_ERROR FDKsbrEnc_InitPSEncode( - HANDLE_PS_ENCODE hPsEncode, - const PS_BANDS psEncMode, - const FIXP_DBL iidQuantErrorThreshold - ); +FDK_PSENC_ERROR FDKsbrEnc_InitPSEncode(HANDLE_PS_ENCODE hPsEncode, + const PS_BANDS psEncMode, + const FIXP_DBL iidQuantErrorThreshold); -FDK_PSENC_ERROR FDKsbrEnc_DestroyPSEncode( - HANDLE_PS_ENCODE *phPsEncode - ); +FDK_PSENC_ERROR FDKsbrEnc_DestroyPSEncode(HANDLE_PS_ENCODE *phPsEncode); FDK_PSENC_ERROR FDKsbrEnc_PSEncode( - HANDLE_PS_ENCODE hPsEncode, - HANDLE_PS_OUT hPsOut, - UCHAR *dynBandScale, - UINT maxEnvelopes, - FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2], - const INT frameSize, - const INT sendHeader - ); + HANDLE_PS_ENCODE hPsEncode, HANDLE_PS_OUT hPsOut, UCHAR *dynBandScale, + UINT maxEnvelopes, + FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2], + const INT frameSize, const INT sendHeader); #endif diff --git a/libSBRenc/src/ps_main.cpp b/libSBRenc/src/ps_main.cpp index ab183e2..4d7a7a5 100644 --- a/libSBRenc/src/ps_main.cpp +++ b/libSBRenc/src/ps_main.cpp @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,92 +90,82 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ -/***************************** MPEG Audio Encoder *************************** +/**************************** SBR encoder library ****************************** - Initial Authors: M. Multrus - Contents/Description: PS Wrapper, Downmix + Author(s): M. Multrus -******************************************************************************/ + Description: PS Wrapper, Downmix -#include "ps_main.h" +*******************************************************************************/ +#include "ps_main.h" /* Includes ******************************************************************/ - -#include "ps_const.h" #include "ps_bitenc.h" - -#include "sbr_ram.h" +#include "sbrenc_ram.h" /*--------------- function declarations --------------------*/ static void psFindBestScaling( - HANDLE_PARAMETRIC_STEREO hParametricStereo, - FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2], - UCHAR *dynBandScale, - FIXP_QMF *maxBandValue, - SCHAR *dmxScale - ); + HANDLE_PARAMETRIC_STEREO hParametricStereo, + FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2], + UCHAR *dynBandScale, FIXP_DBL *maxBandValue, SCHAR *dmxScale); /*------------- function definitions ----------------*/ -FDK_PSENC_ERROR PSEnc_Create( - HANDLE_PARAMETRIC_STEREO *phParametricStereo - ) -{ +FDK_PSENC_ERROR PSEnc_Create(HANDLE_PARAMETRIC_STEREO *phParametricStereo) { FDK_PSENC_ERROR error = PSENC_OK; + HANDLE_PARAMETRIC_STEREO hParametricStereo = NULL; - if (phParametricStereo==NULL) { + if (phParametricStereo == NULL) { error = PSENC_INVALID_HANDLE; - } - else { + } else { int i; - HANDLE_PARAMETRIC_STEREO hParametricStereo = NULL; - if (NULL==(hParametricStereo = GetRam_ParamStereo())) { + if (NULL == (hParametricStereo = GetRam_ParamStereo())) { error = PSENC_MEMORY_ERROR; goto bail; } FDKmemclear(hParametricStereo, sizeof(PARAMETRIC_STEREO)); - if (PSENC_OK != (error = FDKsbrEnc_CreatePSEncode(&hParametricStereo->hPsEncode))) { + if (PSENC_OK != + (error = FDKsbrEnc_CreatePSEncode(&hParametricStereo->hPsEncode))) { + error = PSENC_MEMORY_ERROR; goto bail; } - for (i=0; i<MAX_PS_CHANNELS; i++) { + for (i = 0; i < MAX_PS_CHANNELS; i++) { if (FDKhybridAnalysisOpen( - &hParametricStereo->fdkHybAnaFilter[i], - hParametricStereo->__staticHybAnaStatesLF[i], - sizeof(hParametricStereo->__staticHybAnaStatesLF[i]), - hParametricStereo->__staticHybAnaStatesHF[i], - sizeof(hParametricStereo->__staticHybAnaStatesHF[i]) - ) !=0 ) - { + &hParametricStereo->fdkHybAnaFilter[i], + hParametricStereo->__staticHybAnaStatesLF[i], + sizeof(hParametricStereo->__staticHybAnaStatesLF[i]), + hParametricStereo->__staticHybAnaStatesHF[i], + sizeof(hParametricStereo->__staticHybAnaStatesHF[i])) != 0) { error = PSENC_MEMORY_ERROR; goto bail; } } + } +bail: + if (phParametricStereo != NULL) { *phParametricStereo = hParametricStereo; /* return allocated handle */ } -bail: + + if (error != PSENC_OK) { + PSEnc_Destroy(phParametricStereo); + } return error; } -FDK_PSENC_ERROR PSEnc_Init( - HANDLE_PARAMETRIC_STEREO hParametricStereo, - const HANDLE_PSENC_CONFIG hPsEncConfig, - INT noQmfSlots, - INT noQmfBands - ,UCHAR *dynamic_RAM - ) -{ +FDK_PSENC_ERROR PSEnc_Init(HANDLE_PARAMETRIC_STEREO hParametricStereo, + const HANDLE_PSENC_CONFIG hPsEncConfig, + INT noQmfSlots, INT noQmfBands, UCHAR *dynamic_RAM) { FDK_PSENC_ERROR error = PSENC_OK; - if ( (NULL==hParametricStereo) || (NULL==hPsEncConfig) ) { + if ((NULL == hParametricStereo) || (NULL == hPsEncConfig)) { error = PSENC_INVALID_HANDLE; - } - else { + } else { int ch, i; hParametricStereo->initPS = 1; @@ -172,82 +173,83 @@ FDK_PSENC_ERROR PSEnc_Init( hParametricStereo->noQmfBands = noQmfBands; /* clear delay lines */ - FDKmemclear(hParametricStereo->qmfDelayLines, sizeof(hParametricStereo->qmfDelayLines)); + FDKmemclear(hParametricStereo->qmfDelayLines, + sizeof(hParametricStereo->qmfDelayLines)); - hParametricStereo->qmfDelayScale = FRACT_BITS-1; + hParametricStereo->qmfDelayScale = FRACT_BITS - 1; /* create configuration for hybrid filter bank */ - for (ch=0; ch<MAX_PS_CHANNELS; ch++) { - FDKhybridAnalysisInit( - &hParametricStereo->fdkHybAnaFilter[ch], - THREE_TO_TEN, - QMF_CHANNELS, - QMF_CHANNELS, - 1 - ); + for (ch = 0; ch < MAX_PS_CHANNELS; ch++) { + FDKhybridAnalysisInit(&hParametricStereo->fdkHybAnaFilter[ch], + THREE_TO_TEN, 64, 64, 1); } /* ch */ - FDKhybridSynthesisInit( - &hParametricStereo->fdkHybSynFilter, - THREE_TO_TEN, - QMF_CHANNELS, - QMF_CHANNELS - ); + FDKhybridSynthesisInit(&hParametricStereo->fdkHybSynFilter, THREE_TO_TEN, + 64, 64); /* determine average delay */ - hParametricStereo->psDelay = (HYBRID_FILTER_DELAY*hParametricStereo->noQmfBands); + hParametricStereo->psDelay = + (HYBRID_FILTER_DELAY * hParametricStereo->noQmfBands); - if ( (hPsEncConfig->maxEnvelopes < PSENC_NENV_1) || (hPsEncConfig->maxEnvelopes > PSENC_NENV_MAX) ) { + if ((hPsEncConfig->maxEnvelopes < PSENC_NENV_1) || + (hPsEncConfig->maxEnvelopes > PSENC_NENV_MAX)) { hPsEncConfig->maxEnvelopes = PSENC_NENV_DEFAULT; } hParametricStereo->maxEnvelopes = hPsEncConfig->maxEnvelopes; - if (PSENC_OK != (error = FDKsbrEnc_InitPSEncode(hParametricStereo->hPsEncode, (PS_BANDS) hPsEncConfig->nStereoBands, hPsEncConfig->iidQuantErrorThreshold))){ + if (PSENC_OK != + (error = FDKsbrEnc_InitPSEncode( + hParametricStereo->hPsEncode, (PS_BANDS)hPsEncConfig->nStereoBands, + hPsEncConfig->iidQuantErrorThreshold))) { goto bail; } - for (ch = 0; ch<MAX_PS_CHANNELS; ch ++) { - FIXP_DBL *pDynReal = GetRam_Sbr_envRBuffer (ch, dynamic_RAM); - FIXP_DBL *pDynImag = GetRam_Sbr_envIBuffer (ch, dynamic_RAM); + for (ch = 0; ch < MAX_PS_CHANNELS; ch++) { + FIXP_DBL *pDynReal = GetRam_Sbr_envRBuffer(ch, dynamic_RAM); + FIXP_DBL *pDynImag = GetRam_Sbr_envIBuffer(ch, dynamic_RAM); - for (i=0; i<HYBRID_FRAMESIZE; i++) { - hParametricStereo->pHybridData[i+HYBRID_READ_OFFSET][ch][0] = &pDynReal[i*MAX_HYBRID_BANDS]; - hParametricStereo->pHybridData[i+HYBRID_READ_OFFSET][ch][1] = &pDynImag[i*MAX_HYBRID_BANDS];; + for (i = 0; i < HYBRID_FRAMESIZE; i++) { + hParametricStereo->pHybridData[i + HYBRID_READ_OFFSET][ch][0] = + &pDynReal[i * MAX_HYBRID_BANDS]; + hParametricStereo->pHybridData[i + HYBRID_READ_OFFSET][ch][1] = + &pDynImag[i * MAX_HYBRID_BANDS]; + ; } - for (i=0; i<HYBRID_READ_OFFSET; i++) { - hParametricStereo->pHybridData[i][ch][0] = hParametricStereo->__staticHybridData[i][ch][0]; - hParametricStereo->pHybridData[i][ch][1] = hParametricStereo->__staticHybridData[i][ch][1]; + for (i = 0; i < HYBRID_READ_OFFSET; i++) { + hParametricStereo->pHybridData[i][ch][0] = + hParametricStereo->__staticHybridData[i][ch][0]; + hParametricStereo->pHybridData[i][ch][1] = + hParametricStereo->__staticHybridData[i][ch][1]; } } /* ch */ /* clear static hybrid buffer */ - FDKmemclear(hParametricStereo->__staticHybridData, sizeof(hParametricStereo->__staticHybridData)); + FDKmemclear(hParametricStereo->__staticHybridData, + sizeof(hParametricStereo->__staticHybridData)); /* clear bs buffer */ FDKmemclear(hParametricStereo->psOut, sizeof(hParametricStereo->psOut)); - hParametricStereo->psOut[0].enablePSHeader = 1; /* write ps header in first frame */ + hParametricStereo->psOut[0].enablePSHeader = + 1; /* write ps header in first frame */ /* clear scaling buffer */ - FDKmemclear(hParametricStereo->dynBandScale, sizeof(UCHAR)*PS_MAX_BANDS); - FDKmemclear(hParametricStereo->maxBandValue, sizeof(FIXP_QMF)*PS_MAX_BANDS); + FDKmemclear(hParametricStereo->dynBandScale, sizeof(UCHAR) * PS_MAX_BANDS); + FDKmemclear(hParametricStereo->maxBandValue, + sizeof(FIXP_DBL) * PS_MAX_BANDS); } /* valid handle */ bail: return error; } - -FDK_PSENC_ERROR PSEnc_Destroy( - HANDLE_PARAMETRIC_STEREO *phParametricStereo - ) -{ +FDK_PSENC_ERROR PSEnc_Destroy(HANDLE_PARAMETRIC_STEREO *phParametricStereo) { FDK_PSENC_ERROR error = PSENC_OK; - if (NULL!=phParametricStereo) { + if (NULL != phParametricStereo) { HANDLE_PARAMETRIC_STEREO hParametricStereo = *phParametricStereo; - if(hParametricStereo != NULL){ + if (hParametricStereo != NULL) { FDKsbrEnc_DestroyPSEncode(&hParametricStereo->hPsEncode); FreeRam_ParamStereo(phParametricStereo); } @@ -257,32 +259,24 @@ FDK_PSENC_ERROR PSEnc_Destroy( } static FDK_PSENC_ERROR ExtractPSParameters( - HANDLE_PARAMETRIC_STEREO hParametricStereo, - const int sendHeader, - FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2] - ) -{ + HANDLE_PARAMETRIC_STEREO hParametricStereo, const int sendHeader, + FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2]) { FDK_PSENC_ERROR error = PSENC_OK; if (hParametricStereo == NULL) { error = PSENC_INVALID_HANDLE; - } - else { + } else { /* call ps encode function */ - if (hParametricStereo->initPS){ + if (hParametricStereo->initPS) { hParametricStereo->psOut[1] = hParametricStereo->psOut[0]; } hParametricStereo->psOut[0] = hParametricStereo->psOut[1]; - if (PSENC_OK != (error = FDKsbrEnc_PSEncode( - hParametricStereo->hPsEncode, - &hParametricStereo->psOut[1], - hParametricStereo->dynBandScale, - hParametricStereo->maxEnvelopes, - hybridData, - hParametricStereo->noQmfSlots, - sendHeader))) - { + if (PSENC_OK != + (error = FDKsbrEnc_PSEncode( + hParametricStereo->hPsEncode, &hParametricStereo->psOut[1], + hParametricStereo->dynBandScale, hParametricStereo->maxEnvelopes, + hybridData, hParametricStereo->noQmfSlots, sendHeader))) { goto bail; } @@ -295,209 +289,201 @@ bail: return error; } - static FDK_PSENC_ERROR DownmixPSQmfData( - HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_QMF_FILTER_BANK sbrSynthQmf, - FIXP_QMF **RESTRICT mixRealQmfData, - FIXP_QMF **RESTRICT mixImagQmfData, - INT_PCM *downsampledOutSignal, - FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2], - const INT noQmfSlots, - const INT psQmfScale[MAX_PS_CHANNELS], - SCHAR *qmfScale - ) -{ + HANDLE_PARAMETRIC_STEREO hParametricStereo, + HANDLE_QMF_FILTER_BANK sbrSynthQmf, FIXP_DBL **RESTRICT mixRealQmfData, + FIXP_DBL **RESTRICT mixImagQmfData, INT_PCM *downsampledOutSignal, + const UINT downsampledOutSignalBufSize, + FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2], + const INT noQmfSlots, const INT psQmfScale[MAX_PS_CHANNELS], + SCHAR *qmfScale) { FDK_PSENC_ERROR error = PSENC_OK; - if(hParametricStereo == NULL){ + if (hParametricStereo == NULL) { error = PSENC_INVALID_HANDLE; - } - else { + } else { int n, k; - C_AALLOC_SCRATCH_START(pWorkBuffer, FIXP_QMF, 2*QMF_CHANNELS) + C_AALLOC_SCRATCH_START(pWorkBuffer, FIXP_DBL, 2 * 64) /* define scalings */ - int dynQmfScale = fixMax(0, hParametricStereo->dmxScale-1); /* scale one bit more for addition of left and right */ + int dynQmfScale = fixMax( + 0, hParametricStereo->dmxScale - + 1); /* scale one bit more for addition of left and right */ int downmixScale = psQmfScale[0] - dynQmfScale; const FIXP_DBL maxStereoScaleFactor = MAXVAL_DBL; /* 2.f/2.f */ - for (n = 0; n<noQmfSlots; n++) { - + for (n = 0; n < noQmfSlots; n++) { FIXP_DBL tmpHybrid[2][MAX_HYBRID_BANDS]; - for(k = 0; k<71; k++){ - int dynScale, sc; /* scaling */ - FIXP_QMF tmpLeftReal, tmpRightReal, tmpLeftImag, tmpRightImag; - FIXP_DBL tmpScaleFactor, stereoScaleFactor; - - tmpLeftReal = hybridData[n][0][0][k]; - tmpLeftImag = hybridData[n][0][1][k]; - tmpRightReal = hybridData[n][1][0][k]; - tmpRightImag = hybridData[n][1][1][k]; - - sc = fixMax(0,CntLeadingZeros( fixMax(fixMax(fixp_abs(tmpLeftReal),fixp_abs(tmpLeftImag)),fixMax(fixp_abs(tmpRightReal),fixp_abs(tmpRightImag))) )-2); - - tmpLeftReal <<= sc; tmpLeftImag <<= sc; - tmpRightReal <<= sc; tmpRightImag <<= sc; - dynScale = fixMin(sc-dynQmfScale,DFRACT_BITS-1); - - /* calc stereo scale factor to avoid loss of energy in bands */ - /* stereo scale factor = min(2.0f, sqrt( (abs(l(k, n)^2 + abs(r(k, n)^2 )))/(0.5f*abs(l(k, n) + r(k, n))) )) */ - stereoScaleFactor = fPow2Div2(tmpLeftReal) + fPow2Div2(tmpLeftImag) - + fPow2Div2(tmpRightReal) + fPow2Div2(tmpRightImag) ; - - /* might be that tmpScaleFactor becomes negative, so fabs(.) */ - tmpScaleFactor = fixp_abs(stereoScaleFactor + fMult(tmpLeftReal,tmpRightReal) + fMult(tmpLeftImag,tmpRightImag)); - - /* min(2.0f, sqrt(stereoScaleFactor/(0.5f*tmpScaleFactor))) */ - if ( (stereoScaleFactor>>1) < fMult(maxStereoScaleFactor,tmpScaleFactor) ) { - - int sc_num = CountLeadingBits(stereoScaleFactor) ; - int sc_denum = CountLeadingBits(tmpScaleFactor) ; - sc = -(sc_num-sc_denum); - - tmpScaleFactor = schur_div((stereoScaleFactor<<(sc_num))>>1, - tmpScaleFactor<<sc_denum, - 16) ; - - /* prevent odd scaling for next sqrt calculation */ - if (sc&0x1) { - sc++; - tmpScaleFactor>>=1; - } - stereoScaleFactor = sqrtFixp(tmpScaleFactor); - stereoScaleFactor <<= (sc>>1); - } - else { - stereoScaleFactor = maxStereoScaleFactor; + for (k = 0; k < 71; k++) { + int dynScale, sc; /* scaling */ + FIXP_DBL tmpLeftReal, tmpRightReal, tmpLeftImag, tmpRightImag; + FIXP_DBL tmpScaleFactor, stereoScaleFactor; + + tmpLeftReal = hybridData[n][0][0][k]; + tmpLeftImag = hybridData[n][0][1][k]; + tmpRightReal = hybridData[n][1][0][k]; + tmpRightImag = hybridData[n][1][1][k]; + + sc = fixMax( + 0, CntLeadingZeros(fixMax( + fixMax(fixp_abs(tmpLeftReal), fixp_abs(tmpLeftImag)), + fixMax(fixp_abs(tmpRightReal), fixp_abs(tmpRightImag)))) - + 2); + + tmpLeftReal <<= sc; + tmpLeftImag <<= sc; + tmpRightReal <<= sc; + tmpRightImag <<= sc; + dynScale = fixMin(sc - dynQmfScale, DFRACT_BITS - 1); + + /* calc stereo scale factor to avoid loss of energy in bands */ + /* stereo scale factor = min(2.0f, sqrt( (abs(l(k, n)^2 + abs(r(k, n)^2 + * )))/(0.5f*abs(l(k, n) + r(k, n))) )) */ + stereoScaleFactor = fPow2Div2(tmpLeftReal) + fPow2Div2(tmpLeftImag) + + fPow2Div2(tmpRightReal) + fPow2Div2(tmpRightImag); + + /* might be that tmpScaleFactor becomes negative, so fabs(.) */ + tmpScaleFactor = + fixp_abs(stereoScaleFactor + fMult(tmpLeftReal, tmpRightReal) + + fMult(tmpLeftImag, tmpRightImag)); + + /* min(2.0f, sqrt(stereoScaleFactor/(0.5f*tmpScaleFactor))) */ + if ((stereoScaleFactor >> 1) < + fMult(maxStereoScaleFactor, tmpScaleFactor)) { + int sc_num = CountLeadingBits(stereoScaleFactor); + int sc_denum = CountLeadingBits(tmpScaleFactor); + sc = -(sc_num - sc_denum); + + tmpScaleFactor = schur_div((stereoScaleFactor << (sc_num)) >> 1, + tmpScaleFactor << sc_denum, 16); + + /* prevent odd scaling for next sqrt calculation */ + if (sc & 0x1) { + sc++; + tmpScaleFactor >>= 1; } + stereoScaleFactor = sqrtFixp(tmpScaleFactor); + stereoScaleFactor <<= (sc >> 1); + } else { + stereoScaleFactor = maxStereoScaleFactor; + } - /* write data to hybrid output */ - tmpHybrid[0][k] = fMultDiv2(stereoScaleFactor, (FIXP_QMF)(tmpLeftReal + tmpRightReal))>>dynScale; - tmpHybrid[1][k] = fMultDiv2(stereoScaleFactor, (FIXP_QMF)(tmpLeftImag + tmpRightImag))>>dynScale; + /* write data to hybrid output */ + tmpHybrid[0][k] = fMultDiv2(stereoScaleFactor, + (FIXP_DBL)(tmpLeftReal + tmpRightReal)) >> + dynScale; + tmpHybrid[1][k] = fMultDiv2(stereoScaleFactor, + (FIXP_DBL)(tmpLeftImag + tmpRightImag)) >> + dynScale; } /* hybrid bands - k */ - FDKhybridSynthesisApply( - &hParametricStereo->fdkHybSynFilter, - tmpHybrid[0], - tmpHybrid[1], - mixRealQmfData[n], - mixImagQmfData[n]); + FDKhybridSynthesisApply(&hParametricStereo->fdkHybSynFilter, tmpHybrid[0], + tmpHybrid[1], mixRealQmfData[n], + mixImagQmfData[n]); qmfSynthesisFilteringSlot( - sbrSynthQmf, - mixRealQmfData[n], - mixImagQmfData[n], - downmixScale-7, - downmixScale-7, - downsampledOutSignal+(n*sbrSynthQmf->no_channels), - 1, - pWorkBuffer); + sbrSynthQmf, mixRealQmfData[n], mixImagQmfData[n], downmixScale - 7, + downmixScale - 7, + downsampledOutSignal + (n * sbrSynthQmf->no_channels), 1, + pWorkBuffer); } /* slots */ *qmfScale = -downmixScale + 7; - C_AALLOC_SCRATCH_END(pWorkBuffer, FIXP_QMF, 2*QMF_CHANNELS) + C_AALLOC_SCRATCH_END(pWorkBuffer, FIXP_DBL, 2 * 64) - { - const INT noQmfSlots2 = hParametricStereo->noQmfSlots>>1; - const int noQmfBands = hParametricStereo->noQmfBands; + { + const INT noQmfSlots2 = hParametricStereo->noQmfSlots >> 1; + const int noQmfBands = hParametricStereo->noQmfBands; - INT scale, i, j, slotOffset; + INT scale, i, j, slotOffset; - FIXP_QMF tmp[2][QMF_CHANNELS]; + FIXP_DBL tmp[2][64]; - for (i=0; i<noQmfSlots2; i++) { - FDKmemcpy(tmp[0], hParametricStereo->qmfDelayLines[0][i], noQmfBands*sizeof(FIXP_QMF)); - FDKmemcpy(tmp[1], hParametricStereo->qmfDelayLines[1][i], noQmfBands*sizeof(FIXP_QMF)); + for (i = 0; i < noQmfSlots2; i++) { + FDKmemcpy(tmp[0], hParametricStereo->qmfDelayLines[0][i], + noQmfBands * sizeof(FIXP_DBL)); + FDKmemcpy(tmp[1], hParametricStereo->qmfDelayLines[1][i], + noQmfBands * sizeof(FIXP_DBL)); - FDKmemcpy(hParametricStereo->qmfDelayLines[0][i], mixRealQmfData[i+noQmfSlots2], noQmfBands*sizeof(FIXP_QMF)); - FDKmemcpy(hParametricStereo->qmfDelayLines[1][i], mixImagQmfData[i+noQmfSlots2], noQmfBands*sizeof(FIXP_QMF)); + FDKmemcpy(hParametricStereo->qmfDelayLines[0][i], + mixRealQmfData[i + noQmfSlots2], + noQmfBands * sizeof(FIXP_DBL)); + FDKmemcpy(hParametricStereo->qmfDelayLines[1][i], + mixImagQmfData[i + noQmfSlots2], + noQmfBands * sizeof(FIXP_DBL)); - FDKmemcpy(mixRealQmfData[i+noQmfSlots2], mixRealQmfData[i], noQmfBands*sizeof(FIXP_QMF)); - FDKmemcpy(mixImagQmfData[i+noQmfSlots2], mixImagQmfData[i], noQmfBands*sizeof(FIXP_QMF)); + FDKmemcpy(mixRealQmfData[i + noQmfSlots2], mixRealQmfData[i], + noQmfBands * sizeof(FIXP_DBL)); + FDKmemcpy(mixImagQmfData[i + noQmfSlots2], mixImagQmfData[i], + noQmfBands * sizeof(FIXP_DBL)); - FDKmemcpy(mixRealQmfData[i], tmp[0], noQmfBands*sizeof(FIXP_QMF)); - FDKmemcpy(mixImagQmfData[i], tmp[1], noQmfBands*sizeof(FIXP_QMF)); - } + FDKmemcpy(mixRealQmfData[i], tmp[0], noQmfBands * sizeof(FIXP_DBL)); + FDKmemcpy(mixImagQmfData[i], tmp[1], noQmfBands * sizeof(FIXP_DBL)); + } - if (hParametricStereo->qmfDelayScale > *qmfScale) { - scale = hParametricStereo->qmfDelayScale - *qmfScale; - slotOffset = 0; - } - else { - scale = *qmfScale - hParametricStereo->qmfDelayScale; - slotOffset = noQmfSlots2; - } + if (hParametricStereo->qmfDelayScale > *qmfScale) { + scale = hParametricStereo->qmfDelayScale - *qmfScale; + slotOffset = 0; + } else { + scale = *qmfScale - hParametricStereo->qmfDelayScale; + slotOffset = noQmfSlots2; + } - for (i=0; i<noQmfSlots2; i++) { - for (j=0; j<noQmfBands; j++) { - mixRealQmfData[i+slotOffset][j] >>= scale; - mixImagQmfData[i+slotOffset][j] >>= scale; + for (i = 0; i < noQmfSlots2; i++) { + for (j = 0; j < noQmfBands; j++) { + mixRealQmfData[i + slotOffset][j] >>= scale; + mixImagQmfData[i + slotOffset][j] >>= scale; + } } - } - scale = *qmfScale; - *qmfScale = FDKmin(*qmfScale, hParametricStereo->qmfDelayScale); - hParametricStereo->qmfDelayScale = scale; - } + scale = *qmfScale; + *qmfScale = fMin(*qmfScale, hParametricStereo->qmfDelayScale); + hParametricStereo->qmfDelayScale = scale; + } } /* valid handle */ return error; } - -INT FDKsbrEnc_PSEnc_WritePSData( - HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_FDK_BITSTREAM hBitstream - ) -{ - return ( (hParametricStereo!=NULL) ? FDKsbrEnc_WritePSBitstream(&hParametricStereo->psOut[0], hBitstream) : 0 ); +INT FDKsbrEnc_PSEnc_WritePSData(HANDLE_PARAMETRIC_STEREO hParametricStereo, + HANDLE_FDK_BITSTREAM hBitstream) { + return ( + (hParametricStereo != NULL) + ? FDKsbrEnc_WritePSBitstream(&hParametricStereo->psOut[0], hBitstream) + : 0); } - FDK_PSENC_ERROR FDKsbrEnc_PSEnc_ParametricStereoProcessing( - HANDLE_PARAMETRIC_STEREO hParametricStereo, - INT_PCM *samples[2], - UINT timeInStride, - QMF_FILTER_BANK **hQmfAnalysis, - FIXP_QMF **RESTRICT downmixedRealQmfData, - FIXP_QMF **RESTRICT downmixedImagQmfData, - INT_PCM *downsampledOutSignal, - HANDLE_QMF_FILTER_BANK sbrSynthQmf, - SCHAR *qmfScale, - const int sendHeader - ) -{ + HANDLE_PARAMETRIC_STEREO hParametricStereo, INT_PCM *samples[2], + UINT samplesBufSize, QMF_FILTER_BANK **hQmfAnalysis, + FIXP_DBL **RESTRICT downmixedRealQmfData, + FIXP_DBL **RESTRICT downmixedImagQmfData, INT_PCM *downsampledOutSignal, + HANDLE_QMF_FILTER_BANK sbrSynthQmf, SCHAR *qmfScale, const int sendHeader) { FDK_PSENC_ERROR error = PSENC_OK; INT psQmfScale[MAX_PS_CHANNELS] = {0}; int psCh, i; - C_AALLOC_SCRATCH_START(pWorkBuffer, FIXP_QMF, 4*QMF_CHANNELS) - - for (psCh = 0; psCh<MAX_PS_CHANNELS; psCh ++) { + C_AALLOC_SCRATCH_START(pWorkBuffer, FIXP_DBL, 4 * 64) + for (psCh = 0; psCh < MAX_PS_CHANNELS; psCh++) { for (i = 0; i < hQmfAnalysis[psCh]->no_col; i++) { - qmfAnalysisFilteringSlot( - hQmfAnalysis[psCh], - &pWorkBuffer[2*QMF_CHANNELS], /* qmfReal[QMF_CHANNELS] */ - &pWorkBuffer[3*QMF_CHANNELS], /* qmfImag[QMF_CHANNELS] */ - samples[psCh]+i*(hQmfAnalysis[psCh]->no_channels*timeInStride), - timeInStride, - &pWorkBuffer[0*QMF_CHANNELS] /* qmf workbuffer 2*QMF_CHANNELS */ - ); + hQmfAnalysis[psCh], &pWorkBuffer[2 * 64], /* qmfReal[64] */ + &pWorkBuffer[3 * 64], /* qmfImag[64] */ + samples[psCh] + i * hQmfAnalysis[psCh]->no_channels, 1, + &pWorkBuffer[0 * 64] /* qmf workbuffer 2*64 */ + ); FDKhybridAnalysisApply( - &hParametricStereo->fdkHybAnaFilter[psCh], - &pWorkBuffer[2*QMF_CHANNELS], /* qmfReal[QMF_CHANNELS] */ - &pWorkBuffer[3*QMF_CHANNELS], /* qmfImag[QMF_CHANNELS] */ - hParametricStereo->pHybridData[i+HYBRID_READ_OFFSET][psCh][0], - hParametricStereo->pHybridData[i+HYBRID_READ_OFFSET][psCh][1] - ); + &hParametricStereo->fdkHybAnaFilter[psCh], + &pWorkBuffer[2 * 64], /* qmfReal[64] */ + &pWorkBuffer[3 * 64], /* qmfImag[64] */ + hParametricStereo->pHybridData[i + HYBRID_READ_OFFSET][psCh][0], + hParametricStereo->pHybridData[i + HYBRID_READ_OFFSET][psCh][1]); } /* no_col loop i */ @@ -505,31 +491,48 @@ FDK_PSENC_ERROR FDKsbrEnc_PSEnc_ParametricStereoProcessing( } /* for psCh */ - C_AALLOC_SCRATCH_END(pWorkBuffer, FIXP_QMF, 4*QMF_CHANNELS) + C_AALLOC_SCRATCH_END(pWorkBuffer, FIXP_DBL, 4 * 64) /* find best scaling in new QMF and Hybrid data */ - psFindBestScaling( hParametricStereo, - &hParametricStereo->pHybridData[HYBRID_READ_OFFSET], - hParametricStereo->dynBandScale, - hParametricStereo->maxBandValue, - &hParametricStereo->dmxScale ) ; - + psFindBestScaling( + hParametricStereo, &hParametricStereo->pHybridData[HYBRID_READ_OFFSET], + hParametricStereo->dynBandScale, hParametricStereo->maxBandValue, + &hParametricStereo->dmxScale); /* extract the ps parameters */ - if(PSENC_OK != (error = ExtractPSParameters(hParametricStereo, sendHeader, &hParametricStereo->pHybridData[0]))){ + if (PSENC_OK != + (error = ExtractPSParameters(hParametricStereo, sendHeader, + &hParametricStereo->pHybridData[0]))) { goto bail; } /* save hybrid date for next frame */ - for (i=0; i<HYBRID_READ_OFFSET; i++) { - FDKmemcpy(hParametricStereo->pHybridData[i][0][0], hParametricStereo->pHybridData[HYBRID_FRAMESIZE+i][0][0], MAX_HYBRID_BANDS*sizeof(FIXP_DBL)); /* left, real */ - FDKmemcpy(hParametricStereo->pHybridData[i][0][1], hParametricStereo->pHybridData[HYBRID_FRAMESIZE+i][0][1], MAX_HYBRID_BANDS*sizeof(FIXP_DBL)); /* left, imag */ - FDKmemcpy(hParametricStereo->pHybridData[i][1][0], hParametricStereo->pHybridData[HYBRID_FRAMESIZE+i][1][0], MAX_HYBRID_BANDS*sizeof(FIXP_DBL)); /* right, real */ - FDKmemcpy(hParametricStereo->pHybridData[i][1][1], hParametricStereo->pHybridData[HYBRID_FRAMESIZE+i][1][1], MAX_HYBRID_BANDS*sizeof(FIXP_DBL)); /* right, imag */ + for (i = 0; i < HYBRID_READ_OFFSET; i++) { + FDKmemcpy( + hParametricStereo->pHybridData[i][0][0], + hParametricStereo->pHybridData[hParametricStereo->noQmfSlots + i][0][0], + MAX_HYBRID_BANDS * sizeof(FIXP_DBL)); /* left, real */ + FDKmemcpy( + hParametricStereo->pHybridData[i][0][1], + hParametricStereo->pHybridData[hParametricStereo->noQmfSlots + i][0][1], + MAX_HYBRID_BANDS * sizeof(FIXP_DBL)); /* left, imag */ + FDKmemcpy( + hParametricStereo->pHybridData[i][1][0], + hParametricStereo->pHybridData[hParametricStereo->noQmfSlots + i][1][0], + MAX_HYBRID_BANDS * sizeof(FIXP_DBL)); /* right, real */ + FDKmemcpy( + hParametricStereo->pHybridData[i][1][1], + hParametricStereo->pHybridData[hParametricStereo->noQmfSlots + i][1][1], + MAX_HYBRID_BANDS * sizeof(FIXP_DBL)); /* right, imag */ } /* downmix and hybrid synthesis */ - if (PSENC_OK != (error = DownmixPSQmfData(hParametricStereo, sbrSynthQmf, downmixedRealQmfData, downmixedImagQmfData, downsampledOutSignal, &hParametricStereo->pHybridData[HYBRID_READ_OFFSET], hParametricStereo->noQmfSlots, psQmfScale, qmfScale))) { + if (PSENC_OK != + (error = DownmixPSQmfData( + hParametricStereo, sbrSynthQmf, downmixedRealQmfData, + downmixedImagQmfData, downsampledOutSignal, samplesBufSize, + &hParametricStereo->pHybridData[HYBRID_READ_OFFSET], + hParametricStereo->noQmfSlots, psQmfScale, qmfScale))) { goto bail; } @@ -539,28 +542,24 @@ bail: } static void psFindBestScaling( - HANDLE_PARAMETRIC_STEREO hParametricStereo, - FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2], - UCHAR *dynBandScale, - FIXP_QMF *maxBandValue, - SCHAR *dmxScale - ) -{ - HANDLE_PS_ENCODE hPsEncode = hParametricStereo->hPsEncode; + HANDLE_PARAMETRIC_STEREO hParametricStereo, + FIXP_DBL *hybridData[HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2], + UCHAR *dynBandScale, FIXP_DBL *maxBandValue, SCHAR *dmxScale) { + HANDLE_PS_ENCODE hPsEncode = hParametricStereo->hPsEncode; INT group, bin, col, band; - const INT frameSize = hParametricStereo->noQmfSlots; - const INT psBands = (INT) hPsEncode->psEncMode; + const INT frameSize = hParametricStereo->noQmfSlots; + const INT psBands = (INT)hPsEncode->psEncMode; const INT nIidGroups = hPsEncode->nQmfIidGroups + hPsEncode->nSubQmfIidGroups; /* group wise scaling */ - FIXP_QMF maxVal [2][PS_MAX_BANDS]; - FIXP_QMF maxValue = FL2FXCONST_DBL(0.f); + FIXP_DBL maxVal[2][PS_MAX_BANDS]; + FIXP_DBL maxValue = FL2FXCONST_DBL(0.f); FDKmemclear(maxVal, sizeof(maxVal)); /* start with hybrid data */ - for (group=0; group < nIidGroups; group++) { + for (group = 0; group < nIidGroups; group++) { /* Translate group to bin */ bin = hPsEncode->subband2parameterIndex[group]; @@ -570,49 +569,38 @@ static void psFindBestScaling( } /* QMF downmix scaling */ - { - FIXP_QMF tmp = maxVal[0][bin]; - int i; - for (col=0; col<frameSize-HYBRID_READ_OFFSET; col++) { - for (i = hPsEncode->iidGroupBorders[group]; i < hPsEncode->iidGroupBorders[group+1]; i++) { - tmp = fixMax(tmp, (FIXP_QMF)fixp_abs(hybridData[col][0][0][i])); - tmp = fixMax(tmp, (FIXP_QMF)fixp_abs(hybridData[col][0][1][i])); - tmp = fixMax(tmp, (FIXP_QMF)fixp_abs(hybridData[col][1][0][i])); - tmp = fixMax(tmp, (FIXP_QMF)fixp_abs(hybridData[col][1][1][i])); - } - } - maxVal[0][bin] = tmp; - - tmp = maxVal[1][bin]; - for (col=frameSize-HYBRID_READ_OFFSET; col<frameSize; col++) { - for (i = hPsEncode->iidGroupBorders[group]; i < hPsEncode->iidGroupBorders[group+1]; i++) { - tmp = fixMax(tmp, (FIXP_QMF)fixp_abs(hybridData[col][0][0][i])); - tmp = fixMax(tmp, (FIXP_QMF)fixp_abs(hybridData[col][0][1][i])); - tmp = fixMax(tmp, (FIXP_QMF)fixp_abs(hybridData[col][1][0][i])); - tmp = fixMax(tmp, (FIXP_QMF)fixp_abs(hybridData[col][1][1][i])); - } + for (col = 0; col < frameSize; col++) { + int i, section = (col < frameSize - HYBRID_READ_OFFSET) ? 0 : 1; + FIXP_DBL tmp = maxVal[section][bin]; + for (i = hPsEncode->iidGroupBorders[group]; + i < hPsEncode->iidGroupBorders[group + 1]; i++) { + tmp = fixMax(tmp, (FIXP_DBL)fixp_abs(hybridData[col][0][0][i])); + tmp = fixMax(tmp, (FIXP_DBL)fixp_abs(hybridData[col][0][1][i])); + tmp = fixMax(tmp, (FIXP_DBL)fixp_abs(hybridData[col][1][0][i])); + tmp = fixMax(tmp, (FIXP_DBL)fixp_abs(hybridData[col][1][1][i])); } - maxVal[1][bin] = tmp; + maxVal[section][bin] = tmp; } } /* nIidGroups */ /* convert maxSpec to maxScaling, find scaling space */ - for (band=0; band<psBands; band++) { + for (band = 0; band < psBands; band++) { #ifndef MULT_16x16 - dynBandScale[band] = CountLeadingBits(fixMax(maxVal[0][band],maxBandValue[band])); + dynBandScale[band] = + CountLeadingBits(fixMax(maxVal[0][band], maxBandValue[band])); #else - dynBandScale[band] = fixMax(0,CountLeadingBits(fixMax(maxVal[0][band],maxBandValue[band]))-FRACT_BITS); + dynBandScale[band] = fixMax( + 0, CountLeadingBits(fixMax(maxVal[0][band], maxBandValue[band])) - + FRACT_BITS); #endif - maxValue = fixMax(maxValue,fixMax(maxVal[0][band],maxVal[1][band])); + maxValue = fixMax(maxValue, fixMax(maxVal[0][band], maxVal[1][band])); maxBandValue[band] = fixMax(maxVal[0][band], maxVal[1][band]); } - /* calculate maximal scaling for QMF downmix */ + /* calculate maximal scaling for QMF downmix */ #ifndef MULT_16x16 *dmxScale = fixMin(DFRACT_BITS, CountLeadingBits(maxValue)); #else - *dmxScale = fixMax(0,fixMin(FRACT_BITS, CountLeadingBits(FX_QMF2FX_DBL(maxValue)))); + *dmxScale = fixMax(0, fixMin(FRACT_BITS, CountLeadingBits((maxValue)))); #endif - } - diff --git a/libSBRenc/src/ps_main.h b/libSBRenc/src/ps_main.h index 21b32ff..88b2993 100644 --- a/libSBRenc/src/ps_main.h +++ b/libSBRenc/src/ps_main.h @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,113 +90,116 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** -/***************************** MPEG Audio Encoder *************************** + Author(s): Markus Multrus - Initial Authors: Markus Multrus - Contents/Description: PS Wrapper, Downmix header file + Description: PS Wrapper, Downmix header file -******************************************************************************/ +*******************************************************************************/ -#ifndef __INCLUDED_PS_MAIN_H -#define __INCLUDED_PS_MAIN_H +#ifndef PS_MAIN_H +#define PS_MAIN_H /* Includes ******************************************************************/ + #include "sbr_def.h" #include "qmf.h" #include "ps_encode.h" #include "FDK_bitstream.h" #include "FDK_hybrid.h" - /* Data Types ****************************************************************/ typedef enum { PSENC_STEREO_BANDS_INVALID = 0, - PSENC_STEREO_BANDS_10 = 10, - PSENC_STEREO_BANDS_20 = 20 + PSENC_STEREO_BANDS_10 = 10, + PSENC_STEREO_BANDS_20 = 20 } PSENC_STEREO_BANDS_CONFIG; typedef enum { - PSENC_NENV_1 = 1, - PSENC_NENV_2 = 2, - PSENC_NENV_4 = 4, - PSENC_NENV_DEFAULT = PSENC_NENV_2, - PSENC_NENV_MAX = PSENC_NENV_4 + PSENC_NENV_1 = 1, + PSENC_NENV_2 = 2, + PSENC_NENV_4 = 4, + PSENC_NENV_DEFAULT = PSENC_NENV_2, + PSENC_NENV_MAX = PSENC_NENV_4 } PSENC_NENV_CONFIG; typedef struct { - UINT bitrateFrom; /* inclusive */ - UINT bitrateTo; /* exclusive */ - PSENC_STEREO_BANDS_CONFIG nStereoBands; - PSENC_NENV_CONFIG nEnvelopes; - LONG iidQuantErrorThreshold; /* quantization threshold to switch between coarse and fine iid quantization */ + UINT bitrateFrom; /* inclusive */ + UINT bitrateTo; /* exclusive */ + PSENC_STEREO_BANDS_CONFIG nStereoBands; + PSENC_NENV_CONFIG nEnvelopes; + LONG iidQuantErrorThreshold; /* quantization threshold to switch between + coarse and fine iid quantization */ } psTuningTable_t; /* Function / Class Declarations *********************************************/ typedef struct T_PARAMETRIC_STEREO { - HANDLE_PS_ENCODE hPsEncode; - PS_OUT psOut[2]; - - FIXP_DBL __staticHybridData[HYBRID_READ_OFFSET][MAX_PS_CHANNELS][2][MAX_HYBRID_BANDS]; - FIXP_DBL *pHybridData[HYBRID_READ_OFFSET+HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2]; - - FIXP_QMF qmfDelayLines[2][QMF_MAX_TIME_SLOTS>>1][QMF_CHANNELS]; - int qmfDelayScale; - - INT psDelay; - UINT maxEnvelopes; - UCHAR dynBandScale[PS_MAX_BANDS]; - FIXP_DBL maxBandValue[PS_MAX_BANDS]; - SCHAR dmxScale; - INT initPS; - INT noQmfSlots; - INT noQmfBands; - - FIXP_DBL __staticHybAnaStatesLF[MAX_PS_CHANNELS][2*HYBRID_FILTER_LENGTH*HYBRID_MAX_QMF_BANDS]; - FIXP_DBL __staticHybAnaStatesHF[MAX_PS_CHANNELS][2*HYBRID_FILTER_DELAY*(QMF_CHANNELS-HYBRID_MAX_QMF_BANDS)]; - FDK_ANA_HYB_FILTER fdkHybAnaFilter[MAX_PS_CHANNELS]; - FDK_SYN_HYB_FILTER fdkHybSynFilter; + HANDLE_PS_ENCODE hPsEncode; + PS_OUT psOut[2]; + + FIXP_DBL __staticHybridData[HYBRID_READ_OFFSET][MAX_PS_CHANNELS][2] + [MAX_HYBRID_BANDS]; + FIXP_DBL + *pHybridData[HYBRID_READ_OFFSET + HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2]; + + FIXP_DBL qmfDelayLines[2][32 >> 1][64]; + int qmfDelayScale; + + INT psDelay; + UINT maxEnvelopes; + UCHAR dynBandScale[PS_MAX_BANDS]; + FIXP_DBL maxBandValue[PS_MAX_BANDS]; + SCHAR dmxScale; + INT initPS; + INT noQmfSlots; + INT noQmfBands; + + FIXP_DBL __staticHybAnaStatesLF[MAX_PS_CHANNELS][2 * HYBRID_FILTER_LENGTH * + HYBRID_MAX_QMF_BANDS]; + FIXP_DBL __staticHybAnaStatesHF[MAX_PS_CHANNELS][2 * HYBRID_FILTER_DELAY * + (64 - HYBRID_MAX_QMF_BANDS)]; + FDK_ANA_HYB_FILTER fdkHybAnaFilter[MAX_PS_CHANNELS]; + FDK_SYN_HYB_FILTER fdkHybSynFilter; } PARAMETRIC_STEREO; - typedef struct T_PSENC_CONFIG { - INT frameSize; - INT qmfFilterMode; - INT sbrPsDelay; - PSENC_STEREO_BANDS_CONFIG nStereoBands; - PSENC_NENV_CONFIG maxEnvelopes; - FIXP_DBL iidQuantErrorThreshold; + INT frameSize; + INT qmfFilterMode; + INT sbrPsDelay; + PSENC_STEREO_BANDS_CONFIG nStereoBands; + PSENC_NENV_CONFIG maxEnvelopes; + FIXP_DBL iidQuantErrorThreshold; } PSENC_CONFIG, *HANDLE_PSENC_CONFIG; typedef struct T_PARAMETRIC_STEREO *HANDLE_PARAMETRIC_STEREO; - /** * \brief Create a parametric stereo encoder instance. * - * \param phParametricStereo A pointer to a parametric stereo handle to be allocated. Initialized on return. + * \param phParametricStereo A pointer to a parametric stereo handle to be + * allocated. Initialized on return. * * \return * - PSENC_OK, on succes. * - PSENC_INVALID_HANDLE, PSENC_MEMORY_ERROR, on failure. */ -FDK_PSENC_ERROR PSEnc_Create( - HANDLE_PARAMETRIC_STEREO *phParametricStereo - ); - +FDK_PSENC_ERROR PSEnc_Create(HANDLE_PARAMETRIC_STEREO *phParametricStereo); /** * \brief Initialize a parametric stereo encoder instance. * * \param hParametricStereo Meta Data handle. - * \param hPsEncConfig Filled parametric stereo configuration structure. + * \param hPsEncConfig Filled parametric stereo configuration + * structure. * \param noQmfSlots Number of slots within one audio frame. * \param noQmfBands Number of QMF bands. * \param dynamic_RAM Pointer to preallocated workbuffer. @@ -194,30 +208,23 @@ FDK_PSENC_ERROR PSEnc_Create( * - PSENC_OK, on succes. * - PSENC_INVALID_HANDLE, PSENC_INIT_ERROR, on failure. */ -FDK_PSENC_ERROR PSEnc_Init( - HANDLE_PARAMETRIC_STEREO hParametricStereo, - const HANDLE_PSENC_CONFIG hPsEncConfig, - INT noQmfSlots, - INT noQmfBands - ,UCHAR *dynamic_RAM - ); - +FDK_PSENC_ERROR PSEnc_Init(HANDLE_PARAMETRIC_STEREO hParametricStereo, + const HANDLE_PSENC_CONFIG hPsEncConfig, + INT noQmfSlots, INT noQmfBands, UCHAR *dynamic_RAM); /** * \brief Destroy parametric stereo encoder instance. * * Deallocate instance and free whole memory. * - * \param phParametricStereo Pointer to the parametric stereo handle to be deallocated. + * \param phParametricStereo Pointer to the parametric stereo handle to be + * deallocated. * * \return * - PSENC_OK, on succes. * - PSENC_INVALID_HANDLE, on failure. */ -FDK_PSENC_ERROR PSEnc_Destroy( - HANDLE_PARAMETRIC_STEREO *phParametricStereo - ); - +FDK_PSENC_ERROR PSEnc_Destroy(HANDLE_PARAMETRIC_STEREO *phParametricStereo); /** * \brief Apply parametric stereo processing. @@ -228,7 +235,8 @@ FDK_PSENC_ERROR PSEnc_Destroy( * \param hQmfAnalysis, Pointer to QMF analysis filterbanks. * \param downmixedRealQmfData Pointer to real QMF buffer to be written to. * \param downmixedImagQmfData Pointer to imag QMF buffer to be written to. - * \param downsampledOutSignal Pointer to buffer where to write downmixed timesignal. + * \param downsampledOutSignal Pointer to buffer where to write downmixed + * timesignal. * \param sbrSynthQmf Pointer to QMF synthesis filterbank. * \param qmfScale Return scaling factor of the qmf data. * \param sendHeader Signal whether to write header data. @@ -238,18 +246,11 @@ FDK_PSENC_ERROR PSEnc_Destroy( * - PSENC_INVALID_HANDLE, PSENC_ENCODE_ERROR, on failure. */ FDK_PSENC_ERROR FDKsbrEnc_PSEnc_ParametricStereoProcessing( - HANDLE_PARAMETRIC_STEREO hParametricStereo, - INT_PCM *samples[2], - UINT timeInStride, - QMF_FILTER_BANK **hQmfAnalysis, - FIXP_QMF **RESTRICT downmixedRealQmfData, - FIXP_QMF **RESTRICT downmixedImagQmfData, - INT_PCM *downsampledOutSignal, - HANDLE_QMF_FILTER_BANK sbrSynthQmf, - SCHAR *qmfScale, - const int sendHeader - ); - + HANDLE_PARAMETRIC_STEREO hParametricStereo, INT_PCM *samples[2], + UINT timeInStride, QMF_FILTER_BANK **hQmfAnalysis, + FIXP_DBL **RESTRICT downmixedRealQmfData, + FIXP_DBL **RESTRICT downmixedImagQmfData, INT_PCM *downsampledOutSignal, + HANDLE_QMF_FILTER_BANK sbrSynthQmf, SCHAR *qmfScale, const int sendHeader); /** * \brief Write parametric stereo bitstream. @@ -263,9 +264,7 @@ FDK_PSENC_ERROR FDKsbrEnc_PSEnc_ParametricStereoProcessing( * \return * - number of written bits. */ -INT FDKsbrEnc_PSEnc_WritePSData( - HANDLE_PARAMETRIC_STEREO hParametricStereo, - HANDLE_FDK_BITSTREAM hBitstream - ); +INT FDKsbrEnc_PSEnc_WritePSData(HANDLE_PARAMETRIC_STEREO hParametricStereo, + HANDLE_FDK_BITSTREAM hBitstream); -#endif /* __INCLUDED_PS_MAIN_H */ +#endif /* PS_MAIN_H */ diff --git a/libSBRenc/src/resampler.cpp b/libSBRenc/src/resampler.cpp index 4adb243..b1781a7 100644 --- a/libSBRenc/src/resampler.cpp +++ b/libSBRenc/src/resampler.cpp @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,11 +90,19 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ /*! \file - \brief FDK resampler tool box: + \brief FDK resampler tool box:$Revision: 91655 $ \author M. Werner */ @@ -91,7 +110,6 @@ amm-info@iis.fraunhofer.de #include "genericStds.h" - /**************************************************************************/ /* BIQUAD Filter Specifications */ /**************************************************************************/ @@ -101,92 +119,93 @@ amm-info@iis.fraunhofer.de #define A1 2 #define A2 3 -#define BQC(x) FL2FXCONST_SGL(x/2) - +#define BQC(x) FL2FXCONST_SGL(x / 2) struct FILTER_PARAM { - const FIXP_SGL *coeffa; /*! SOS matrix One row/section. Scaled using BQC(). Order of coefficients: B1,B2,A1,A2. B0=A0=1.0 */ - FIXP_DBL g; /*! overall gain */ - int Wc; /*! normalized passband bandwidth at input samplerate * 1000 */ - int noCoeffs; /*! number of filter coeffs */ - int delay; /*! delay in samples at input samplerate */ + const FIXP_SGL *coeffa; /*! SOS matrix One row/section. Scaled using BQC(). + Order of coefficients: B1,B2,A1,A2. B0=A0=1.0 */ + FIXP_DBL g; /*! overall gain */ + int Wc; /*! normalized passband bandwidth at input samplerate * 1000 */ + int noCoeffs; /*! number of filter coeffs */ + int delay; /*! delay in samples at input samplerate */ }; #define BIQUAD_COEFSTEP 4 /** *\brief Low Pass - Wc = 0,5, order 30, Stop Band -96dB. Wc criteria is "almost 0dB passband", not the usual -3db gain point. - [b,a]=cheby2(30,96,0.505) - [sos,g]=tf2sos(b,a) + Wc = 0,5, order 30, Stop Band -96dB. Wc criteria is "almost 0dB passband", not + the usual -3db gain point. [b,a]=cheby2(30,96,0.505) [sos,g]=tf2sos(b,a) bandwidth 0.48 */ static const FIXP_SGL sos48[] = { - BQC(1.98941075681938), BQC(0.999999996890811), BQC(0.863264527201963), BQC( 0.189553799960663), - BQC(1.90733804822445), BQC(1.00000001736189), BQC(0.836321575841691), BQC( 0.203505809266564), - BQC(1.75616665495325), BQC(0.999999946079721), BQC(0.784699225121588), BQC( 0.230471265506986), - BQC(1.55727745512726), BQC(1.00000011737815), BQC(0.712515423588351), BQC( 0.268752723900498), - BQC(1.33407591943643), BQC(0.999999795953228), BQC(0.625059117330989), BQC( 0.316194685288965), - BQC(1.10689898412458), BQC(1.00000035057114), BQC(0.52803514366398), BQC( 0.370517843224669), - BQC(0.89060371078454), BQC(0.999999343962822), BQC(0.426920462165257), BQC( 0.429608200207746), - BQC(0.694438261209433), BQC( 1.0000008629792), BQC(0.326530699561716), BQC( 0.491714450654174), - BQC(0.523237800935322), BQC(1.00000101349782), BQC(0.230829556274851), BQC( 0.555559034843281), - BQC(0.378631165929563), BQC(0.99998986482665), BQC(0.142906422036095), BQC( 0.620338874442411), - BQC(0.260786911308437), BQC(1.00003261460178), BQC(0.0651008576256505), BQC( 0.685759923926262), - BQC(0.168409429188098), BQC(0.999933049695828), BQC(-0.000790067789975562), BQC( 0.751905896602325), - BQC(0.100724533818628), BQC(1.00009472669872), BQC(-0.0533772830257041), BQC( 0.81930744384525), - BQC(0.0561434357867363), BQC(0.999911636304276), BQC(-0.0913550299236405), BQC( 0.88883625875915), - BQC(0.0341680678662057), BQC(1.00003667508676), BQC(-0.113405185536697), BQC( 0.961756638268446) -}; - -#ifdef RS_BIQUAD_SCATTERGAIN -static const FIXP_DBL g48 = FL2FXCONST_DBL(0.67436532061161992682404480717671 - 0.001); -#else -static const FIXP_DBL g48 = FL2FXCONST_DBL(0.002712866530047) - (FIXP_DBL)0x8000; -#endif + BQC(1.98941075681938), BQC(0.999999996890811), + BQC(0.863264527201963), BQC(0.189553799960663), + BQC(1.90733804822445), BQC(1.00000001736189), + BQC(0.836321575841691), BQC(0.203505809266564), + BQC(1.75616665495325), BQC(0.999999946079721), + BQC(0.784699225121588), BQC(0.230471265506986), + BQC(1.55727745512726), BQC(1.00000011737815), + BQC(0.712515423588351), BQC(0.268752723900498), + BQC(1.33407591943643), BQC(0.999999795953228), + BQC(0.625059117330989), BQC(0.316194685288965), + BQC(1.10689898412458), BQC(1.00000035057114), + BQC(0.52803514366398), BQC(0.370517843224669), + BQC(0.89060371078454), BQC(0.999999343962822), + BQC(0.426920462165257), BQC(0.429608200207746), + BQC(0.694438261209433), BQC(1.0000008629792), + BQC(0.326530699561716), BQC(0.491714450654174), + BQC(0.523237800935322), BQC(1.00000101349782), + BQC(0.230829556274851), BQC(0.555559034843281), + BQC(0.378631165929563), BQC(0.99998986482665), + BQC(0.142906422036095), BQC(0.620338874442411), + BQC(0.260786911308437), BQC(1.00003261460178), + BQC(0.0651008576256505), BQC(0.685759923926262), + BQC(0.168409429188098), BQC(0.999933049695828), + BQC(-0.000790067789975562), BQC(0.751905896602325), + BQC(0.100724533818628), BQC(1.00009472669872), + BQC(-0.0533772830257041), BQC(0.81930744384525), + BQC(0.0561434357867363), BQC(0.999911636304276), + BQC(-0.0913550299236405), BQC(0.88883625875915), + BQC(0.0341680678662057), BQC(1.00003667508676), + BQC(-0.113405185536697), BQC(0.961756638268446)}; + +static const FIXP_DBL g48 = + FL2FXCONST_DBL(0.002712866530047) - (FIXP_DBL)0x8000; static const struct FILTER_PARAM param_set48 = { - sos48, - g48, - 480, - 15, - 4 /* LF 2 */ + sos48, g48, 480, 15, 4 /* LF 2 */ }; /** *\brief Low Pass - Wc = 0,5, order 24, Stop Band -96dB. Wc criteria is "almost 0dB passband", not the usual -3db gain point. - [b,a]=cheby2(24,96,0.5) - [sos,g]=tf2sos(b,a) + Wc = 0,5, order 24, Stop Band -96dB. Wc criteria is "almost 0dB passband", not + the usual -3db gain point. [b,a]=cheby2(24,96,0.5) [sos,g]=tf2sos(b,a) bandwidth 0.45 */ static const FIXP_SGL sos45[] = { - BQC(1.982962601444), BQC(1.00000000007504), BQC(0.646113303737836), BQC( 0.10851149979981), - BQC(1.85334094281111), BQC(0.999999999677192), BQC(0.612073220102006), BQC( 0.130022141698044), - BQC(1.62541051415425), BQC(1.00000000080398), BQC(0.547879702855959), BQC( 0.171165825133192), - BQC(1.34554656923247), BQC(0.9999999980169), BQC(0.460373914508491), BQC( 0.228677463376354), - BQC(1.05656568503116), BQC(1.00000000569363), BQC(0.357891894038287), BQC( 0.298676843912185), - BQC(0.787967587877312), BQC(0.999999984415017), BQC(0.248826893211877), BQC( 0.377441803512978), - BQC(0.555480971120497), BQC(1.00000003583307), BQC(0.140614263345315), BQC( 0.461979302213679), - BQC(0.364986207070964), BQC(0.999999932084303), BQC(0.0392669446074516), BQC( 0.55033451180825), - BQC(0.216827267631558), BQC(1.00000010534682), BQC(-0.0506232228865103), BQC( 0.641691581560946), - BQC(0.108951672277119), BQC(0.999999871167516), BQC(-0.125584840183225), BQC( 0.736367748771803), - BQC(0.0387988607229035), BQC(1.00000011205574), BQC(-0.182814849097974), BQC( 0.835802108714964), - BQC(0.0042866175809225), BQC(0.999999954830813), BQC(-0.21965740617151), BQC( 0.942623047782363) -}; - -#ifdef RS_BIQUAD_SCATTERGAIN -static const FIXP_DBL g45 = FL2FXCONST_DBL(0.60547428891341319051142629706723 - 0.001); -#else -static const FIXP_DBL g45 = FL2FXCONST_DBL(0.00242743980909524) - (FIXP_DBL)0x8000; -#endif + BQC(1.982962601444), BQC(1.00000000007504), BQC(0.646113303737836), + BQC(0.10851149979981), BQC(1.85334094281111), BQC(0.999999999677192), + BQC(0.612073220102006), BQC(0.130022141698044), BQC(1.62541051415425), + BQC(1.00000000080398), BQC(0.547879702855959), BQC(0.171165825133192), + BQC(1.34554656923247), BQC(0.9999999980169), BQC(0.460373914508491), + BQC(0.228677463376354), BQC(1.05656568503116), BQC(1.00000000569363), + BQC(0.357891894038287), BQC(0.298676843912185), BQC(0.787967587877312), + BQC(0.999999984415017), BQC(0.248826893211877), BQC(0.377441803512978), + BQC(0.555480971120497), BQC(1.00000003583307), BQC(0.140614263345315), + BQC(0.461979302213679), BQC(0.364986207070964), BQC(0.999999932084303), + BQC(0.0392669446074516), BQC(0.55033451180825), BQC(0.216827267631558), + BQC(1.00000010534682), BQC(-0.0506232228865103), BQC(0.641691581560946), + BQC(0.108951672277119), BQC(0.999999871167516), BQC(-0.125584840183225), + BQC(0.736367748771803), BQC(0.0387988607229035), BQC(1.00000011205574), + BQC(-0.182814849097974), BQC(0.835802108714964), BQC(0.0042866175809225), + BQC(0.999999954830813), BQC(-0.21965740617151), BQC(0.942623047782363)}; + +static const FIXP_DBL g45 = + FL2FXCONST_DBL(0.00242743980909524) - (FIXP_DBL)0x8000; static const struct FILTER_PARAM param_set45 = { - sos45, - g45, - 450, - 12, - 4 /* LF 2 */ + sos45, g45, 450, 12, 4 /* LF 2 */ }; /* @@ -197,30 +216,23 @@ static const struct FILTER_PARAM param_set45 = { bandwidth = 0.41 */ -static const FIXP_SGL sos41[] = -{ - BQC(1.96193625292), BQC(0.999999999999964), BQC(0.169266178786789), BQC(0.0128823300475907), - BQC(1.68913437662092), BQC(1.00000000000053), BQC(0.124751503206552), BQC(0.0537472273950989), - BQC(1.27274692366017), BQC(0.999999999995674), BQC(0.0433108625178357), BQC(0.131015753236317), - BQC(0.85214175088395), BQC(1.00000000001813), BQC(-0.0625658152550408), BQC(0.237763778993806), - BQC(0.503841579939009), BQC(0.999999999953223), BQC(-0.179176128722865), BQC(0.367475236424474), - BQC(0.249990711986162), BQC(1.00000000007952), BQC(-0.294425165824676), BQC(0.516594857170212), - BQC(0.087971668680286), BQC(0.999999999915528), BQC(-0.398956566777928), BQC(0.686417767801123), - BQC(0.00965373325350294), BQC(1.00000000003744), BQC(-0.48579173764817), BQC(0.884931534239068) -}; +static const FIXP_SGL sos41[] = { + BQC(1.96193625292), BQC(0.999999999999964), BQC(0.169266178786789), + BQC(0.0128823300475907), BQC(1.68913437662092), BQC(1.00000000000053), + BQC(0.124751503206552), BQC(0.0537472273950989), BQC(1.27274692366017), + BQC(0.999999999995674), BQC(0.0433108625178357), BQC(0.131015753236317), + BQC(0.85214175088395), BQC(1.00000000001813), BQC(-0.0625658152550408), + BQC(0.237763778993806), BQC(0.503841579939009), BQC(0.999999999953223), + BQC(-0.179176128722865), BQC(0.367475236424474), BQC(0.249990711986162), + BQC(1.00000000007952), BQC(-0.294425165824676), BQC(0.516594857170212), + BQC(0.087971668680286), BQC(0.999999999915528), BQC(-0.398956566777928), + BQC(0.686417767801123), BQC(0.00965373325350294), BQC(1.00000000003744), + BQC(-0.48579173764817), BQC(0.884931534239068)}; -#ifdef RS_BIQUAD_SCATTERGAIN -static const FIXP_DBL g41 = FL2FXCONST_DBL(0.44578514476476679750811222123569); -#else static const FIXP_DBL g41 = FL2FXCONST_DBL(0.00155956951169248); -#endif static const struct FILTER_PARAM param_set41 = { - sos41, - g41, - 410, - 8, - 5 /* LF 3 */ + sos41, g41, 410, 8, 5 /* LF 3 */ }; /* @@ -229,29 +241,19 @@ static const struct FILTER_PARAM param_set41 = { [b,a]=cheby2(12,96,0.5); [sos,g]=tf2sos(b,a) */ -static const FIXP_SGL sos35[] = -{ - BQC(1.93299325235762), BQC(0.999999999999985), BQC(-0.140733187246596), BQC(0.0124139497836062), - BQC(1.4890416764109), BQC(1.00000000000011), BQC(-0.198215402588504), BQC(0.0746730616584138), - BQC(0.918450161309795), BQC(0.999999999999619), BQC(-0.30133912791941), BQC(0.192276468839529), - BQC(0.454877024246818), BQC(1.00000000000086), BQC(-0.432337328809815), BQC(0.356852933642815), - BQC(0.158017147118507), BQC(0.999999999998876), BQC(-0.574817494249777), BQC(0.566380436970833), - BQC(0.0171834649478749), BQC(1.00000000000055), BQC(-0.718581178041165), BQC(0.83367484487889) -}; +static const FIXP_SGL sos35[] = { + BQC(1.93299325235762), BQC(0.999999999999985), BQC(-0.140733187246596), + BQC(0.0124139497836062), BQC(1.4890416764109), BQC(1.00000000000011), + BQC(-0.198215402588504), BQC(0.0746730616584138), BQC(0.918450161309795), + BQC(0.999999999999619), BQC(-0.30133912791941), BQC(0.192276468839529), + BQC(0.454877024246818), BQC(1.00000000000086), BQC(-0.432337328809815), + BQC(0.356852933642815), BQC(0.158017147118507), BQC(0.999999999998876), + BQC(-0.574817494249777), BQC(0.566380436970833), BQC(0.0171834649478749), + BQC(1.00000000000055), BQC(-0.718581178041165), BQC(0.83367484487889)}; -#ifdef RS_BIQUAD_SCATTERGAIN -static const FIXP_DBL g35 = FL2FXCONST_DBL(0.34290853574973898694521267606792); -#else static const FIXP_DBL g35 = FL2FXCONST_DBL(0.00162580994125131); -#endif -static const struct FILTER_PARAM param_set35 = { - sos35, - g35, - 350, - 6, - 4 -}; +static const struct FILTER_PARAM param_set35 = {sos35, g35, 350, 6, 4}; /* # Created by Octave 2.1.73, Mon Oct 13 18:15:38 2008 CEST @@ -259,66 +261,53 @@ static const struct FILTER_PARAM param_set35 = { [b,a]=cheby2(8,96,0.5); [sos,g]=tf2sos(b,a) */ -static const FIXP_SGL sos25[] = -{ - BQC(1.85334094301225), BQC(1.0), BQC(-0.702127214212663), BQC(0.132452403998767), - BQC(1.056565682167), BQC(0.999999999999997), BQC(-0.789503667880785), BQC(0.236328693569128), - BQC(0.364986307455489), BQC(0.999999999999996), BQC(-0.955191189843375), BQC(0.442966457936379), - BQC(0.0387985751642125), BQC(1.0), BQC(-1.19817786088084), BQC(0.770493895456328) -}; +static const FIXP_SGL sos25[] = { + BQC(1.85334094301225), BQC(1.0), + BQC(-0.702127214212663), BQC(0.132452403998767), + BQC(1.056565682167), BQC(0.999999999999997), + BQC(-0.789503667880785), BQC(0.236328693569128), + BQC(0.364986307455489), BQC(0.999999999999996), + BQC(-0.955191189843375), BQC(0.442966457936379), + BQC(0.0387985751642125), BQC(1.0), + BQC(-1.19817786088084), BQC(0.770493895456328)}; -#ifdef RS_BIQUAD_SCATTERGAIN -static const FIXP_DBL g25 = FL2FXCONST_DBL(0.17533917408936346960080259950471); -#else static const FIXP_DBL g25 = FL2FXCONST_DBL(0.000945182835294559); -#endif -static const struct FILTER_PARAM param_set25 = { - sos25, - g25, - 250, - 4, - 5 -}; +static const struct FILTER_PARAM param_set25 = {sos25, g25, 250, 4, 5}; /* Must be sorted in descending order */ static const struct FILTER_PARAM *const filter_paramSet[] = { - ¶m_set48, - ¶m_set45, - ¶m_set41, - ¶m_set35, - ¶m_set25 -}; - + ¶m_set48, ¶m_set45, ¶m_set41, ¶m_set35, ¶m_set25}; /**************************************************************************/ /* Resampler Functions */ /**************************************************************************/ - /*! \brief Reset downsampler instance and clear delay lines \return success of operation */ -INT FDKaacEnc_InitDownsampler(DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */ - int Wc, /*!< normalized cutoff freq * 1000* */ - int ratio) /*!< downsampler ratio (only 2 supported at the momment) */ +INT FDKaacEnc_InitDownsampler( + DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */ + int Wc, /*!< normalized cutoff freq * 1000* */ + int ratio) /*!< downsampler ratio */ { UINT i; - const struct FILTER_PARAM *currentSet=NULL; + const struct FILTER_PARAM *currentSet = NULL; - FDK_ASSERT(ratio == 2); - FDKmemclear(DownSampler->downFilter.states, sizeof(DownSampler->downFilter.states)); - DownSampler->downFilter.ptr = 0; + FDKmemclear(DownSampler->downFilter.states, + sizeof(DownSampler->downFilter.states)); + DownSampler->downFilter.ptr = 0; /* find applicable parameter set */ currentSet = filter_paramSet[0]; - for(i=1;i<sizeof(filter_paramSet)/sizeof(struct FILTER_PARAM *);i++){ + for (i = 1; i < sizeof(filter_paramSet) / sizeof(struct FILTER_PARAM *); + i++) { if (filter_paramSet[i]->Wc <= Wc) { break; } @@ -327,20 +316,18 @@ INT FDKaacEnc_InitDownsampler(DOWNSAMPLER *DownSampler, /*!< pointer to downsamp DownSampler->downFilter.coeffa = currentSet->coeffa; - DownSampler->downFilter.gain = currentSet->g; - FDK_ASSERT(currentSet->noCoeffs <= MAXNR_SECTIONS*2); + FDK_ASSERT(currentSet->noCoeffs <= MAXNR_SECTIONS * 2); DownSampler->downFilter.noCoeffs = currentSet->noCoeffs; DownSampler->delay = currentSet->delay; DownSampler->downFilter.Wc = currentSet->Wc; - DownSampler->ratio = ratio; - DownSampler->pending = ratio-1; - return(1); + DownSampler->ratio = ratio; + DownSampler->pending = ratio - 1; + return (1); } - /*! \brief faster simple folding operation Filter: @@ -351,64 +338,54 @@ INT FDKaacEnc_InitDownsampler(DOWNSAMPLER *DownSampler, /*!< pointer to downsamp \return filtered value */ -static inline INT_PCM AdvanceFilter(LP_FILTER *downFilter, /*!< pointer to iir filter instance */ - INT_PCM *pInput, /*!< input of filter */ - int downRatio, - int inStride) -{ +static inline INT_PCM AdvanceFilter( + LP_FILTER *downFilter, /*!< pointer to iir filter instance */ + INT_PCM *pInput, /*!< input of filter */ + int downRatio) { INT_PCM output; int i, n; - -#ifdef RS_BIQUAD_SCATTERGAIN -#define BIQUAD_SCALE 3 -#else #define BIQUAD_SCALE 12 -#endif FIXP_DBL y = FL2FXCONST_DBL(0.0f); FIXP_DBL input; - for (n=0; n<downRatio; n++) - { - FIXP_BQS (*states)[2] = downFilter->states; + for (n = 0; n < downRatio; n++) { + FIXP_BQS(*states)[2] = downFilter->states; const FIXP_SGL *coeff = downFilter->coeffa; - int s1,s2; + int s1, s2; s1 = downFilter->ptr; s2 = s1 ^ 1; #if (SAMPLE_BITS == 16) - input = ((FIXP_DBL)pInput[n*inStride]) << (DFRACT_BITS-SAMPLE_BITS-BIQUAD_SCALE); + input = ((FIXP_DBL)pInput[n]) << (DFRACT_BITS - SAMPLE_BITS - BIQUAD_SCALE); #elif (SAMPLE_BITS == 32) - input = pInput[n*inStride] >> BIQUAD_SCALE; + input = pInput[n] >> BIQUAD_SCALE; #else #error NOT IMPLEMENTED #endif -#ifndef RS_BIQUAD_SCATTERGAIN /* Merged Direct form I */ - FIXP_BQS state1, state2, state1b, state2b; state1 = states[0][s1]; state2 = states[0][s2]; /* Loop over sections */ - for (i=0; i<downFilter->noCoeffs; i++) - { + for (i = 0; i < downFilter->noCoeffs; i++) { FIXP_DBL state0; /* Load merged states (from next section) */ - state1b = states[i+1][s1]; - state2b = states[i+1][s2]; + state1b = states[i + 1][s1]; + state2b = states[i + 1][s2]; - state0 = input + fMult(state1, coeff[B1]) + fMult(state2, coeff[B2]); - y = state0 - fMult(state1b, coeff[A1]) - fMult(state2b, coeff[A2]); + state0 = input + fMult(state1, coeff[B1]) + fMult(state2, coeff[B2]); + y = state0 - fMult(state1b, coeff[A1]) - fMult(state2b, coeff[A2]); /* Store new feed forward merge state */ - states[i+1][s2] = y<<1; + states[i + 1][s2] = y << 1; /* Store new feed backward state */ - states[i][s2] = input<<1; + states[i][s2] = input << 1; /* Feedback output to next section. */ input = y; @@ -425,57 +402,20 @@ static inline INT_PCM AdvanceFilter(LP_FILTER *downFilter, /*!< pointer to iir /* Apply global gain */ y = fMult(y, downFilter->gain); -#else /* Direct form II */ - - /* Loop over sections */ - for (i=0; i<downFilter->noCoeffs; i++) - { - FIXP_BQS state1, state2; - FIXP_DBL state0; - - /* Load states */ - state1 = states[i][s1]; - state2 = states[i][s2]; - - state0 = input - fMult(state1, coeff[A1]) - fMult(state2, coeff[A2]); - y = state0 + fMult(state1, coeff[B1]) + fMult(state2, coeff[B2]); - /* Apply scattered gain */ - y = fMult(y, downFilter->gain); - - /* Store new state in normalized form */ -#ifdef RS_BIQUAD_STATES16 - /* Do not saturate any state value ! The result would be unacceptable. Rounding makes SNR around 10dB better. */ - states[i][s2] = (FIXP_BQS)(LONG)((state0 + (FIXP_DBL)(1<<(DFRACT_BITS-FRACT_BITS-2))) >> (DFRACT_BITS-FRACT_BITS-1)); -#else - states[i][s2] = state0<<1; -#endif - - /* Feedback output to next section. */ - input=y; - - /* Step to next coef set */ - coeff += BIQUAD_COEFSTEP; - } - downFilter->ptr ^= 1; - } - -#endif - /* Apply final gain/scaling to output */ #if (SAMPLE_BITS == 16) - output = (INT_PCM) SATURATE_RIGHT_SHIFT(y+(FIXP_DBL)(1<<(DFRACT_BITS-SAMPLE_BITS-BIQUAD_SCALE-1)), DFRACT_BITS-SAMPLE_BITS-BIQUAD_SCALE, SAMPLE_BITS); - //output = (INT_PCM) SATURATE_RIGHT_SHIFT(y, DFRACT_BITS-SAMPLE_BITS-BIQUAD_SCALE, SAMPLE_BITS); + output = (INT_PCM)SATURATE_RIGHT_SHIFT( + y + (FIXP_DBL)(1 << (DFRACT_BITS - SAMPLE_BITS - BIQUAD_SCALE - 1)), + DFRACT_BITS - SAMPLE_BITS - BIQUAD_SCALE, SAMPLE_BITS); + // output = (INT_PCM) SATURATE_RIGHT_SHIFT(y, + // DFRACT_BITS-SAMPLE_BITS-BIQUAD_SCALE, SAMPLE_BITS); #else output = SATURATE_LEFT_SHIFT(y, BIQUAD_SCALE, SAMPLE_BITS); #endif - return output; } - - - /*! \brief FDKaacEnc_Downsample numInSamples of type INT_PCM Returns number of output samples in numOutSamples @@ -483,25 +423,22 @@ static inline INT_PCM AdvanceFilter(LP_FILTER *downFilter, /*!< pointer to iir \return success of operation */ -INT FDKaacEnc_Downsample(DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */ - INT_PCM *inSamples, /*!< pointer to input samples */ - INT numInSamples, /*!< number of input samples */ - INT inStride, /*!< increment of input samples */ - INT_PCM *outSamples, /*!< pointer to output samples */ - INT *numOutSamples, /*!< pointer tp number of output samples */ - INT outStride /*!< increment of output samples */ - ) -{ - INT i; - *numOutSamples=0; - - for(i=0; i<numInSamples; i+=DownSampler->ratio) - { - *outSamples = AdvanceFilter(&(DownSampler->downFilter), &inSamples[i*inStride], DownSampler->ratio, inStride); - outSamples += outStride; - } - *numOutSamples = numInSamples/DownSampler->ratio; +INT FDKaacEnc_Downsample( + DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */ + INT_PCM *inSamples, /*!< pointer to input samples */ + INT numInSamples, /*!< number of input samples */ + INT_PCM *outSamples, /*!< pointer to output samples */ + INT *numOutSamples /*!< pointer tp number of output samples */ +) { + INT i; + *numOutSamples = 0; + + for (i = 0; i < numInSamples; i += DownSampler->ratio) { + *outSamples = AdvanceFilter(&(DownSampler->downFilter), &inSamples[i], + DownSampler->ratio); + outSamples++; + } + *numOutSamples = numInSamples / DownSampler->ratio; - return 0; + return 0; } - diff --git a/libSBRenc/src/resampler.h b/libSBRenc/src/resampler.h index 0192970..7aa1cae 100644 --- a/libSBRenc/src/resampler.h +++ b/libSBRenc/src/resampler.h @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,73 +90,70 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): -#ifndef __RESAMPLER_H -#define __RESAMPLER_H + Description: + +*******************************************************************************/ + +#ifndef RESAMPLER_H +#define RESAMPLER_H /*! \file - \brief Fixed Point Resampler Tool Box + \brief Fixed Point Resampler Tool Box $Revision: 92790 $ */ #include "common_fix.h" - /**************************************************************************/ /* BIQUAD Filter Structure */ /**************************************************************************/ -#define MAXNR_SECTIONS (15) +#define MAXNR_SECTIONS (15) -#ifdef RS_BIQUAD_STATES16 -typedef FIXP_SGL FIXP_BQS; -#else typedef FIXP_DBL FIXP_BQS; -#endif - -typedef struct -{ - FIXP_BQS states[MAXNR_SECTIONS+1][2]; /*! state buffer */ - const FIXP_SGL *coeffa; /*! pointer to filter coeffs */ - FIXP_DBL gain; /*! overall gain factor */ - int Wc; /*! normalized cutoff freq * 1000 */ - int noCoeffs; /*! number of filter coeffs sets */ - int ptr; /*! index to rinbuffers */ -} LP_FILTER; +typedef struct { + FIXP_BQS states[MAXNR_SECTIONS + 1][2]; /*! state buffer */ + const FIXP_SGL *coeffa; /*! pointer to filter coeffs */ + FIXP_DBL gain; /*! overall gain factor */ + int Wc; /*! normalized cutoff freq * 1000 */ + int noCoeffs; /*! number of filter coeffs sets */ + int ptr; /*! index to rinbuffers */ +} LP_FILTER; /**************************************************************************/ /* Downsampler Structure */ /**************************************************************************/ -typedef struct -{ - LP_FILTER downFilter; /*! filter instance */ - int ratio; /*! downsampling ration */ - int delay; /*! downsampling delay (source fs) */ - int pending; /*! number of pending output samples */ +typedef struct { + LP_FILTER downFilter; /*! filter instance */ + int ratio; /*! downsampling ration */ + int delay; /*! downsampling delay (source fs) */ + int pending; /*! number of pending output samples */ } DOWNSAMPLER; - /** * \brief Initialized a given downsampler structure. */ -INT FDKaacEnc_InitDownsampler(DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */ - INT Wc, /*!< normalized cutoff freq * 1000 */ - INT ratio); /*!< downsampler ratio */ +INT FDKaacEnc_InitDownsampler( + DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */ + INT Wc, /*!< normalized cutoff freq * 1000 */ + INT ratio); /*!< downsampler ratio */ /** - * \brief Downsample a set of audio samples. numInSamples must be at least equal to the - * downsampler ratio. + * \brief Downsample a set of audio samples. numInSamples must be at least equal + * to the downsampler ratio. */ -INT FDKaacEnc_Downsample(DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */ - INT_PCM *inSamples, /*!< pointer to input samples */ - INT numInSamples, /*!< number of input samples */ - INT inStride, /*!< increment of input samples */ - INT_PCM *outSamples, /*!< pointer to output samples */ - INT *numOutSamples, /*!< pointer tp number of output samples */ - INT outstride); /*!< increment of output samples */ - - - -#endif /* __RESAMPLER_H */ +INT FDKaacEnc_Downsample( + DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */ + INT_PCM *inSamples, /*!< pointer to input samples */ + INT numInSamples, /*!< number of input samples */ + INT_PCM *outSamples, /*!< pointer to output samples */ + INT *numOutSamples); /*!< pointer tp number of output samples */ + +#endif /* RESAMPLER_H */ diff --git a/libSBRenc/src/sbr.h b/libSBRenc/src/sbr.h index c74ad2a..341dcab 100644 --- a/libSBRenc/src/sbr.h +++ b/libSBRenc/src/sbr.h @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,15 +90,23 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ /*! \file - \brief Main SBR structs definitions + \brief Main SBR structs definitions $Revision: 92790 $ */ -#ifndef __SBR_H -#define __SBR_H +#ifndef SBR_H +#define SBR_H #include "fram_gen.h" #include "bit_sbr.h" @@ -101,66 +120,75 @@ amm-info@iis.fraunhofer.de #include "ton_corr.h" - /* SBR bitstream delay */ - #define DELAY_FRAMES 2 +#define MAX_DELAY_FRAMES 2 +/* sbr encoder downsampling type */ +typedef enum { SBRENC_DS_NONE, SBRENC_DS_TIME, SBRENC_DS_QMF } SBRENC_DS_TYPE; typedef struct SBR_CHANNEL { - struct ENV_CHANNEL hEnvChannel; - //INT_PCM *pDSOutBuffer; /**< Pointer to downsampled audio output of SBR encoder */ - DOWNSAMPLER downSampler; + struct ENV_CHANNEL hEnvChannel; + // INT_PCM *pDSOutBuffer; /**< Pointer to + // downsampled audio output of SBR encoder */ + DOWNSAMPLER downSampler; } SBR_CHANNEL; typedef SBR_CHANNEL* HANDLE_SBR_CHANNEL; typedef struct SBR_ELEMENT { - HANDLE_SBR_CHANNEL sbrChannel[2]; - QMF_FILTER_BANK *hQmfAnalysis[2]; - SBR_CONFIG_DATA sbrConfigData; - SBR_HEADER_DATA sbrHeaderData; - SBR_BITSTREAM_DATA sbrBitstreamData; - COMMON_DATA CmonData; - INT dynXOverFreqDelay[5]; /**< to delay a frame (I don't like it that much that way - hrc) */ - SBR_ELEMENT_INFO elInfo; - - UCHAR payloadDelayLine[1+DELAY_FRAMES][MAX_PAYLOAD_SIZE]; - UINT payloadDelayLineSize[1+DELAY_FRAMES]; /* Sizes in bits */ + HANDLE_SBR_CHANNEL sbrChannel[2]; + QMF_FILTER_BANK* hQmfAnalysis[2]; + SBR_CONFIG_DATA sbrConfigData; + SBR_HEADER_DATA sbrHeaderData; + SBR_BITSTREAM_DATA sbrBitstreamData; + COMMON_DATA CmonData; + INT dynXOverFreqDelay[5]; /**< to delay a frame (I don't like it that much + that way - hrc) */ + SBR_ELEMENT_INFO elInfo; + + UCHAR payloadDelayLine[1 + MAX_DELAY_FRAMES][MAX_PAYLOAD_SIZE]; + UINT payloadDelayLineSize[1 + MAX_DELAY_FRAMES]; /* Sizes in bits */ } SBR_ELEMENT, *HANDLE_SBR_ELEMENT; -typedef struct SBR_ENCODER -{ - HANDLE_SBR_ELEMENT sbrElement[(8)]; - HANDLE_SBR_CHANNEL pSbrChannel[(8)]; - QMF_FILTER_BANK QmfAnalysis[(8)]; - DOWNSAMPLER lfeDownSampler; - int lfeChIdx; /* -1 default for no lfe, else assign channel index */ - int noElements; /* Number of elements */ - int nChannels; /* Total channel count across all elements. */ - int frameSize; /* SBR framelength. */ - int bufferOffset; /* Offset for SBR parameter extraction in time domain input buffer. */ - int downsampledOffset; /* Offset of downsampled/mixed output for core encoder. */ - int downmixSize; /* Size in samples of downsampled/mixed output for core encoder. */ - INT downSampleFactor; /* Sampling rate relation between the SBR and the core encoder. */ - int fTimeDomainDownsampling; /* Flag signalling time domain downsampling instead of QMF downsampling. */ - int nBitstrDelay; /* Amount of SBR frames to be delayed in bitstream domain. */ - INT estimateBitrate; /* estimate bitrate of SBR encoder */ - INT inputDataDelay; /* delay caused by downsampler, in/out buffer at sbrEncoder_EncodeFrame */ +typedef struct SBR_ENCODER { + HANDLE_SBR_ELEMENT sbrElement[(8)]; + HANDLE_SBR_CHANNEL pSbrChannel[(8)]; + QMF_FILTER_BANK QmfAnalysis[(8)]; + DOWNSAMPLER lfeDownSampler; + int lfeChIdx; /* -1 default for no lfe, else assign channel index. */ + int noElements; /* Number of elements. */ + int nChannels; /* Total channel count across all elements. */ + int frameSize; /* SBR framelength. */ + int bufferOffset; /* Offset for SBR parameter extraction in time domain input + buffer. */ + int downsampledOffset; /* Offset of downsampled/mixed output for core encoder. + */ + int downmixSize; /* Size in samples of downsampled/mixed output for core + encoder. */ + INT downSampleFactor; /* Sampling rate relation between the SBR and the core + encoder. */ + SBRENC_DS_TYPE + downsamplingMethod; /* Method of downsmapling, time-domain, QMF or none. + */ + int nBitstrDelay; /* Amount of SBR frames to be delayed in bitstream domain. + */ + int sbrDecDelay; /* SBR decoder delay in samples */ + INT estimateBitrate; /* Estimate bitrate of SBR encoder. */ + INT inputDataDelay; /* Delay caused by downsampler, in/out buffer at + sbrEncoder_EncodeFrame. */ UCHAR* dynamicRam; UCHAR* pSBRdynamic_RAM; - HANDLE_PARAMETRIC_STEREO hParametricStereo; - QMF_FILTER_BANK qmfSynthesisPS; + HANDLE_PARAMETRIC_STEREO hParametricStereo; + QMF_FILTER_BANK qmfSynthesisPS; /* parameters describing allocation volume of present instance */ - INT maxElements; - INT maxChannels; - INT supportPS; - + INT maxElements; + INT maxChannels; + INT supportPS; } SBR_ENCODER; - -#endif /* __SBR_H */ +#endif /* SBR_H */ diff --git a/libSBRenc/src/sbr_def.h b/libSBRenc/src/sbr_def.h index 85ac587..53eba71 100644 --- a/libSBRenc/src/sbr_def.h +++ b/libSBRenc/src/sbr_def.h @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,21 +90,28 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ /*! \file - \brief SBR main definitions + \brief SBR main definitions $Revision: 92790 $ */ -#ifndef __SBR_DEF_H -#define __SBR_DEF_H +#ifndef SBR_DEF_H +#define SBR_DEF_H #include "common_fix.h" #define noError 0 #define HANDLE_ERROR_INFO INT -#define ERROR(a,b) 1 -#define handBack +#define ERROR(a, b) 1 /* #define SBR_ENV_STATISTICS_BITRATE */ #undef SBR_ENV_STATISTICS_BITRATE @@ -104,172 +122,155 @@ amm-info@iis.fraunhofer.de /* #define SBR_PAYLOAD_MONITOR */ #undef SBR_PAYLOAD_MONITOR -#define SWAP(a,b) tempr=a, a=b, b=tempr -#define TRUE 1 +#define SWAP(a, b) tempr = a, a = b, b = tempr +#define TRUE 1 #define FALSE 0 - /* Constants */ -#define EPS 1e-12 -#define LOG2 0.69314718056f /* natural logarithm of 2 */ -#define ILOG2 1.442695041f /* 1/LOG2 */ -#define RELAXATION_FLOAT (1e-6f) -#define RELAXATION (FL2FXCONST_DBL(RELAXATION_FLOAT)) -#define RELAXATION_FRACT (FL2FXCONST_DBL(0.524288f)) /* 0.524288f is fractional part of RELAXATION */ -#define RELAXATION_SHIFT (19) -#define RELAXATION_LD64 (FL2FXCONST_DBL(0.31143075889f))/* (ld64(RELAXATION) */ +#define EPS 1e-12 +#define LOG2 0.69314718056f /* natural logarithm of 2 */ +#define ILOG2 1.442695041f /* 1/LOG2 */ +#define RELAXATION_FLOAT (1e-6f) +#define RELAXATION (FL2FXCONST_DBL(RELAXATION_FLOAT)) +#define RELAXATION_FRACT \ + (FL2FXCONST_DBL(0.524288f)) /* 0.524288f is fractional part of RELAXATION */ +#define RELAXATION_SHIFT (19) +#define RELAXATION_LD64 \ + (FL2FXCONST_DBL(0.31143075889f)) /* (ld64(RELAXATION) \ + */ /************ Definitions ***************/ -#define SBR_COMP_MODE_DELTA 0 -#define SBR_COMP_MODE_CTS 1 -#define SBR_MAX_ENERGY_VALUES 5 -#define SBR_GLOBAL_TONALITY_VALUES 2 - -#define MAX_NUM_CHANNELS 2 +#define SBR_COMP_MODE_DELTA 0 +#define SBR_COMP_MODE_CTS 1 +#define SBR_MAX_ENERGY_VALUES 5 +#define SBR_GLOBAL_TONALITY_VALUES 2 -#define MAX_NOISE_ENVELOPES 2 -#define MAX_NUM_NOISE_COEFFS 5 -#define MAX_NUM_NOISE_VALUES (MAX_NUM_NOISE_COEFFS*MAX_NOISE_ENVELOPES) +#define MAX_NUM_CHANNELS 2 -#define MAX_NUM_ENVELOPE_VALUES (MAX_ENVELOPES * MAX_FREQ_COEFFS) -#define MAX_ENVELOPES 5 -#define MAX_FREQ_COEFFS 48 +#define MAX_NOISE_ENVELOPES 2 +#define MAX_NUM_NOISE_COEFFS 5 +#define MAX_NUM_NOISE_VALUES (MAX_NUM_NOISE_COEFFS * MAX_NOISE_ENVELOPES) -#define MAX_FREQ_COEFFS_FS44100 35 -#define MAX_FREQ_COEFFS_FS48000 32 +#define MAX_NUM_ENVELOPE_VALUES (MAX_ENVELOPES * MAX_FREQ_COEFFS) +#define MAX_ENVELOPES 5 +#define MAX_FREQ_COEFFS 48 +#define MAX_FREQ_COEFFS_FS44100 35 +#define MAX_FREQ_COEFFS_FS48000 32 -#define QMF_CHANNELS 64 -#define QMF_FILTER_LENGTH 640 -#define QMF_MAX_TIME_SLOTS 32 -#define NO_OF_ESTIMATES_LC 4 -#define NO_OF_ESTIMATES_LD 3 -#define MAX_NO_OF_ESTIMATES 4 +#define NO_OF_ESTIMATES_LC 4 +#define NO_OF_ESTIMATES_LD 3 +#define MAX_NO_OF_ESTIMATES 4 +#define NOISE_FLOOR_OFFSET 6 +#define NOISE_FLOOR_OFFSET_64 (FL2FXCONST_DBL(0.09375f)) -#define NOISE_FLOOR_OFFSET 6 -#define NOISE_FLOOR_OFFSET_64 (FL2FXCONST_DBL(0.09375f)) +#define LOW_RES 0 +#define HIGH_RES 1 -#define LOW_RES 0 -#define HIGH_RES 1 +#define LO 0 +#define HI 1 -#define LO 0 -#define HI 1 +#define LENGTH_SBR_FRAME_INFO 35 /* 19 */ -#define LENGTH_SBR_FRAME_INFO 35 /* 19 */ +#define SBR_NSFB_LOW_RES 9 /* 8 */ +#define SBR_NSFB_HIGH_RES 18 /* 16 */ -#define SBR_NSFB_LOW_RES 9 /* 8 */ -#define SBR_NSFB_HIGH_RES 18 /* 16 */ +#define SBR_XPOS_CTRL_DEFAULT 2 +#define SBR_FREQ_SCALE_DEFAULT 2 +#define SBR_ALTER_SCALE_DEFAULT 1 +#define SBR_NOISE_BANDS_DEFAULT 2 -#define SBR_XPOS_CTRL_DEFAULT 2 - -#define SBR_FREQ_SCALE_DEFAULT 2 -#define SBR_ALTER_SCALE_DEFAULT 1 -#define SBR_NOISE_BANDS_DEFAULT 2 - -#define SBR_LIMITER_BANDS_DEFAULT 2 -#define SBR_LIMITER_GAINS_DEFAULT 2 -#define SBR_LIMITER_GAINS_INFINITE 3 -#define SBR_INTERPOL_FREQ_DEFAULT 1 -#define SBR_SMOOTHING_LENGTH_DEFAULT 0 - +#define SBR_LIMITER_BANDS_DEFAULT 2 +#define SBR_LIMITER_GAINS_DEFAULT 2 +#define SBR_LIMITER_GAINS_INFINITE 3 +#define SBR_INTERPOL_FREQ_DEFAULT 1 +#define SBR_SMOOTHING_LENGTH_DEFAULT 0 /* sbr_header */ -#define SI_SBR_AMP_RES_BITS 1 -#define SI_SBR_COUPLING_BITS 1 -#define SI_SBR_START_FREQ_BITS 4 -#define SI_SBR_STOP_FREQ_BITS 4 -#define SI_SBR_XOVER_BAND_BITS 3 -#define SI_SBR_RESERVED_BITS 2 -#define SI_SBR_DATA_EXTRA_BITS 1 -#define SI_SBR_HEADER_EXTRA_1_BITS 1 -#define SI_SBR_HEADER_EXTRA_2_BITS 1 +#define SI_SBR_AMP_RES_BITS 1 +#define SI_SBR_COUPLING_BITS 1 +#define SI_SBR_START_FREQ_BITS 4 +#define SI_SBR_STOP_FREQ_BITS 4 +#define SI_SBR_XOVER_BAND_BITS 3 +#define SI_SBR_RESERVED_BITS 2 +#define SI_SBR_DATA_EXTRA_BITS 1 +#define SI_SBR_HEADER_EXTRA_1_BITS 1 +#define SI_SBR_HEADER_EXTRA_2_BITS 1 /* sbr_header extra 1 */ -#define SI_SBR_FREQ_SCALE_BITS 2 -#define SI_SBR_ALTER_SCALE_BITS 1 -#define SI_SBR_NOISE_BANDS_BITS 2 +#define SI_SBR_FREQ_SCALE_BITS 2 +#define SI_SBR_ALTER_SCALE_BITS 1 +#define SI_SBR_NOISE_BANDS_BITS 2 /* sbr_header extra 2 */ -#define SI_SBR_LIMITER_BANDS_BITS 2 -#define SI_SBR_LIMITER_GAINS_BITS 2 -#define SI_SBR_INTERPOL_FREQ_BITS 1 -#define SI_SBR_SMOOTHING_LENGTH_BITS 1 +#define SI_SBR_LIMITER_BANDS_BITS 2 +#define SI_SBR_LIMITER_GAINS_BITS 2 +#define SI_SBR_INTERPOL_FREQ_BITS 1 +#define SI_SBR_SMOOTHING_LENGTH_BITS 1 /* sbr_grid */ -#define SBR_CLA_BITS 2 /*!< size of bs_frame_class */ -#define SBR_CLA_BITS_LD 1 /*!< size of bs_frame_class */ -#define SBR_ENV_BITS 2 /*!< size of bs_num_env_raw */ -#define SBR_ABS_BITS 2 /*!< size of bs_abs_bord_raw for HE-AAC */ -#define SBR_NUM_BITS 2 /*!< size of bs_num_rel */ -#define SBR_REL_BITS 2 /*!< size of bs_rel_bord_raw */ -#define SBR_RES_BITS 1 /*!< size of bs_freq_res_flag */ -#define SBR_DIR_BITS 1 /*!< size of bs_df_flag */ - +#define SBR_CLA_BITS 2 /*!< size of bs_frame_class */ +#define SBR_CLA_BITS_LD 1 /*!< size of bs_frame_class */ +#define SBR_ENV_BITS 2 /*!< size of bs_num_env_raw */ +#define SBR_ABS_BITS 2 /*!< size of bs_abs_bord_raw for HE-AAC */ +#define SBR_NUM_BITS 2 /*!< size of bs_num_rel */ +#define SBR_REL_BITS 2 /*!< size of bs_rel_bord_raw */ +#define SBR_RES_BITS 1 /*!< size of bs_freq_res_flag */ +#define SBR_DIR_BITS 1 /*!< size of bs_df_flag */ /* sbr_data */ -#define SI_SBR_INVF_MODE_BITS 2 - +#define SI_SBR_INVF_MODE_BITS 2 -#define SI_SBR_START_ENV_BITS_AMP_RES_3_0 6 -#define SI_SBR_START_ENV_BITS_BALANCE_AMP_RES_3_0 5 -#define SI_SBR_START_NOISE_BITS_AMP_RES_3_0 5 +#define SI_SBR_START_ENV_BITS_AMP_RES_3_0 6 +#define SI_SBR_START_ENV_BITS_BALANCE_AMP_RES_3_0 5 +#define SI_SBR_START_NOISE_BITS_AMP_RES_3_0 5 #define SI_SBR_START_NOISE_BITS_BALANCE_AMP_RES_3_0 5 -#define SI_SBR_START_ENV_BITS_AMP_RES_1_5 7 -#define SI_SBR_START_ENV_BITS_BALANCE_AMP_RES_1_5 6 - +#define SI_SBR_START_ENV_BITS_AMP_RES_1_5 7 +#define SI_SBR_START_ENV_BITS_BALANCE_AMP_RES_1_5 6 -#define SI_SBR_EXTENDED_DATA_BITS 1 -#define SI_SBR_EXTENSION_SIZE_BITS 4 -#define SI_SBR_EXTENSION_ESC_COUNT_BITS 8 -#define SI_SBR_EXTENSION_ID_BITS 2 +#define SI_SBR_EXTENDED_DATA_BITS 1 +#define SI_SBR_EXTENSION_SIZE_BITS 4 +#define SI_SBR_EXTENSION_ESC_COUNT_BITS 8 +#define SI_SBR_EXTENSION_ID_BITS 2 -#define SBR_EXTENDED_DATA_MAX_CNT (15+255) +#define SBR_EXTENDED_DATA_MAX_CNT (15 + 255) -#define EXTENSION_ID_PS_CODING 2 +#define EXTENSION_ID_PS_CODING 2 /* Envelope coding constants */ -#define FREQ 0 -#define TIME 1 +#define FREQ 0 +#define TIME 1 /* qmf data scaling */ -#define QMF_SCALE_OFFSET 7 +#define QMF_SCALE_OFFSET 7 /* huffman tables */ -#define CODE_BOOK_SCF_LAV00 60 -#define CODE_BOOK_SCF_LAV01 31 -#define CODE_BOOK_SCF_LAV10 60 -#define CODE_BOOK_SCF_LAV11 31 +#define CODE_BOOK_SCF_LAV00 60 +#define CODE_BOOK_SCF_LAV01 31 +#define CODE_BOOK_SCF_LAV10 60 +#define CODE_BOOK_SCF_LAV11 31 #define CODE_BOOK_SCF_LAV_BALANCE11 12 #define CODE_BOOK_SCF_LAV_BALANCE10 24 -typedef enum -{ - SBR_AMP_RES_1_5=0, - SBR_AMP_RES_3_0 -} -AMP_RES; +typedef enum { SBR_AMP_RES_1_5 = 0, SBR_AMP_RES_3_0 } AMP_RES; -typedef enum -{ +typedef enum { XPOS_MDCT, XPOS_MDCT_CROSS, XPOS_LC, XPOS_RESERVED, XPOS_SWITCHED /* not a real choice but used here to control behaviour */ -} -XPOS_MODE; +} XPOS_MODE; -typedef enum -{ +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; +} INVF_MODE; #endif diff --git a/libSBRenc/src/sbr_encoder.cpp b/libSBRenc/src/sbr_encoder.cpp index 71aab78..df9e996 100644 --- a/libSBRenc/src/sbr_encoder.cpp +++ b/libSBRenc/src/sbr_encoder.cpp @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,19 +90,20 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** -/*************************** Fraunhofer IIS FDK Tools *********************** + Author(s): Andreas Ehret, Tobias Chalupka - Author(s): Andreas Ehret, Tobias Chalupka Description: SBR encoder top level processing. -******************************************************************************/ +*******************************************************************************/ #include "sbr_encoder.h" -#include "sbr_ram.h" -#include "sbr_rom.h" +#include "sbrenc_ram.h" +#include "sbrenc_rom.h" #include "sbrenc_freq_sca.h" #include "env_bit.h" #include "cmondata.h" @@ -101,11 +113,9 @@ amm-info@iis.fraunhofer.de #include "ps_main.h" -#define SBRENCODER_LIB_VL0 3 -#define SBRENCODER_LIB_VL1 3 -#define SBRENCODER_LIB_VL2 12 - - +#define SBRENCODER_LIB_VL0 4 +#define SBRENCODER_LIB_VL1 0 +#define SBRENCODER_LIB_VL2 0 /***************************************************************************/ /* @@ -119,34 +129,83 @@ amm-info@iis.fraunhofer.de (core2sbr delay ) ds (read, core and ds area) */ -#define SFB(dwnsmp) (32 << (dwnsmp-1)) /* SBR Frequency bands: 64 for dual-rate, 32 for single-rate */ -#define STS(fl) (((fl)==1024)?32:30) /* SBR Time Slots: 32 for core frame length 1024, 30 for core frame length 960 */ - -#define DELAY_QMF_ANA(dwnsmp) ((320<<((dwnsmp)-1)) - (32<<((dwnsmp)-1))) /* Full bandwidth */ -#define DELAY_HYB_ANA (10*64) /* + 0.5 */ /* */ -#define DELAY_HYB_SYN (6*64 - 32) /* */ -#define DELAY_QMF_POSTPROC(dwnsmp) (32*(dwnsmp)) /* QMF postprocessing delay */ -#define DELAY_DEC_QMF(dwnsmp) (6 * SFB(dwnsmp) ) /* Decoder QMF overlap */ -#define DELAY_QMF_SYN (2) /* NO_POLY/2=2.5, rounded down to 2 */ -#define DELAY_QMF_DS (32) /* QMF synthesis for downsampled time signal */ +#define SFB(dwnsmp) \ + (32 << (dwnsmp - \ + 1)) /* SBR Frequency bands: 64 for dual-rate, 32 for single-rate */ +#define STS(fl) \ + (((fl) == 1024) ? 32 \ + : 30) /* SBR Time Slots: 32 for core frame length 1024, 30 \ + for core frame length 960 */ + +#define DELAY_QMF_ANA(dwnsmp) \ + ((320 << ((dwnsmp)-1)) - (32 << ((dwnsmp)-1))) /* Full bandwidth */ +#define DELAY_HYB_ANA (10 * 64) /* + 0.5 */ /* */ +#define DELAY_HYB_SYN (6 * 64 - 32) /* */ +#define DELAY_QMF_POSTPROC(dwnsmp) \ + (32 * (dwnsmp)) /* QMF postprocessing delay */ +#define DELAY_DEC_QMF(dwnsmp) (6 * SFB(dwnsmp)) /* Decoder QMF overlap */ +#define DELAY_QMF_SYN(dwnsmp) \ + (1 << (dwnsmp - \ + 1)) /* QMF_NO_POLY/2=2.5, rounded down to 2, half for single-rate */ +#define DELAY_QMF_DS (32) /* QMF synthesis for downsampled time signal */ /* Delay in QMF paths */ -#define DELAY_SBR(fl,dwnsmp) (DELAY_QMF_ANA(dwnsmp) + (SFB(dwnsmp)*STS(fl) - 1) + DELAY_QMF_SYN) -#define DELAY_PS(fl,dwnsmp) (DELAY_QMF_ANA(dwnsmp) + DELAY_HYB_ANA + DELAY_DEC_QMF(dwnsmp) + (SFB(dwnsmp)*STS(fl)-1) + DELAY_HYB_SYN + DELAY_QMF_SYN) -#define DELAY_ELDSBR(fl,dwnsmp) ( ( ((fl)/2)*(dwnsmp) ) - 1 + DELAY_QMF_POSTPROC(dwnsmp) ) - -/* Delay differences for SBR and SBR+PS */ -#define MAX_DS_FILTER_DELAY (5) /* the additional max downsampler filter delay (source fs) */ -#define DELAY_AAC2SBR(fl,dwnsmp) ((DELAY_QMF_ANA(dwnsmp) + DELAY_DEC_QMF(dwnsmp) + DELAY_QMF_SYN) - DELAY_SBR((fl),(dwnsmp))) -#define DELAY_ELD2SBR(fl,dwnsmp) ((DELAY_QMF_POSTPROC(dwnsmp)) - DELAY_ELDSBR(fl,dwnsmp)) -#define DELAY_AAC2PS(fl,dwnsmp) ((DELAY_QMF_ANA(dwnsmp) + DELAY_QMF_DS + /*(DELAY_AAC(fl)*2) + */ DELAY_QMF_ANA(dwnsmp) + DELAY_DEC_QMF(dwnsmp) + DELAY_HYB_SYN + DELAY_QMF_SYN) - DELAY_PS(fl,dwnsmp)) /* 2048 - 463*2 */ - -/* Assumption: The sample delay resulting of of DELAY_AAC2PS is always smaller than the sample delay implied by DELAY_AAC2SBR */ -#define MAX_SAMPLE_DELAY (DELAY_AAC2SBR(1024,2) + MAX_DS_FILTER_DELAY) /* maximum delay: frame length of 1024 and dual-rate sbr */ +#define DELAY_SBR(fl, dwnsmp) \ + (DELAY_QMF_ANA(dwnsmp) + (SFB(dwnsmp) * STS(fl) - 1) + DELAY_QMF_SYN(dwnsmp)) +#define DELAY_PS(fl, dwnsmp) \ + (DELAY_QMF_ANA(dwnsmp) + DELAY_HYB_ANA + DELAY_DEC_QMF(dwnsmp) + \ + (SFB(dwnsmp) * STS(fl) - 1) + DELAY_HYB_SYN + DELAY_QMF_SYN(dwnsmp)) +#define DELAY_ELDSBR(fl, dwnsmp) \ + ((((fl) / 2) * (dwnsmp)) - 1 + DELAY_QMF_POSTPROC(dwnsmp)) +#define DELAY_ELDv2SBR(fl, dwnsmp) \ + ((((fl) / 2) * (dwnsmp)) - 1 + 80 * (dwnsmp)) /* 80 is the delay caused \ + by the sum of the CLD \ + analysis and the MPSLD \ + synthesis filterbank */ + +/* Delay in core path (core and downsampler not taken into account) */ +#define DELAY_COREPATH_SBR(fl, dwnsmp) \ + ((DELAY_QMF_ANA(dwnsmp) + DELAY_DEC_QMF(dwnsmp) + DELAY_QMF_SYN(dwnsmp))) +#define DELAY_COREPATH_ELDSBR(fl, dwnsmp) ((DELAY_QMF_POSTPROC(dwnsmp))) +#define DELAY_COREPATH_ELDv2SBR(fl, dwnsmp) (128 * (dwnsmp)) /* 4 slots */ +#define DELAY_COREPATH_PS(fl, dwnsmp) \ + ((DELAY_QMF_ANA(dwnsmp) + DELAY_QMF_DS + \ + /*(DELAY_AAC(fl)*2) + */ DELAY_QMF_ANA(dwnsmp) + DELAY_DEC_QMF(dwnsmp) + \ + DELAY_HYB_SYN + DELAY_QMF_SYN(dwnsmp))) /* 2048 - 463*2 */ + +/* Delay differences between SBR- and downsampled path for SBR and SBR+PS */ +#define DELAY_AAC2SBR(fl, dwnsmp) \ + ((DELAY_COREPATH_SBR(fl, dwnsmp)) - DELAY_SBR((fl), (dwnsmp))) +#define DELAY_ELD2SBR(fl, dwnsmp) \ + ((DELAY_COREPATH_ELDSBR(fl, dwnsmp)) - DELAY_ELDSBR(fl, dwnsmp)) +#define DELAY_AAC2PS(fl, dwnsmp) \ + ((DELAY_COREPATH_PS(fl, dwnsmp)) - DELAY_PS(fl, dwnsmp)) /* 2048 - 463*2 */ + +/* Assumption: The sample delay resulting of of DELAY_AAC2PS is always smaller + * than the sample delay implied by DELAY_AAC2SBR */ +#define MAX_DS_FILTER_DELAY \ + (5) /* the additional max downsampler filter delay (source fs) */ +#define MAX_SAMPLE_DELAY \ + (DELAY_AAC2SBR(1024, 2) + MAX_DS_FILTER_DELAY) /* maximum delay: frame \ + length of 1024 and \ + dual-rate sbr */ /***************************************************************************/ - +/*************** Delay parameters for sbrEncoder_Init_delay() **************/ +typedef struct { + int dsDelay; /* the delay of the (time-domain) downsampler itself */ + int delay; /* overall delay / samples */ + int sbrDecDelay; /* SBR decoder's delay */ + int corePathOffset; /* core path offset / samples; added by + sbrEncoder_Init_delay() */ + int sbrPathOffset; /* SBR path offset / samples; added by + sbrEncoder_Init_delay() */ + int bitstrDelay; /* bitstream delay / frames; added by sbrEncoder_Init_delay() + */ + int delayInput2Core; /* delay of the input to the core / samples */ +} DELAY_PARAM; +/***************************************************************************/ #define INVALID_TABLE_IDX -1 @@ -160,44 +219,38 @@ amm-info@iis.fraunhofer.de ****************************************************************************/ #define DISTANCE_CEIL_VALUE 5000000 -static INT -getSbrTuningTableIndex(UINT bitrate, /*! the total bitrate in bits/sec */ - UINT numChannels,/*! the number of channels for the core coder */ - UINT sampleRate, /*! the sampling rate of the core coder */ - AUDIO_OBJECT_TYPE core, - UINT *pBitRateClosest - ) -{ - int i, bitRateClosestLowerIndex=-1, bitRateClosestUpperIndex=-1, found = 0; - UINT bitRateClosestUpper = 0, bitRateClosestLower=DISTANCE_CEIL_VALUE; - - #define isForThisCore(i) \ - ( ( sbrTuningTable[i].coreCoder == CODEC_AACLD && core == AOT_ER_AAC_ELD ) || \ - ( sbrTuningTable[i].coreCoder == CODEC_AAC && core != AOT_ER_AAC_ELD ) ) - - for (i=0; i < sbrTuningTableSize ; i++) { - if ( isForThisCore(i) ) /* tuning table is for this core codec */ +static INT getSbrTuningTableIndex( + UINT bitrate, /*! the total bitrate in bits/sec */ + UINT numChannels, /*! the number of channels for the core coder */ + UINT sampleRate, /*! the sampling rate of the core coder */ + AUDIO_OBJECT_TYPE core, UINT *pBitRateClosest) { + int i, bitRateClosestLowerIndex = -1, bitRateClosestUpperIndex = -1, + found = 0; + UINT bitRateClosestUpper = 0, bitRateClosestLower = DISTANCE_CEIL_VALUE; + +#define isForThisCore(i) \ + ((sbrTuningTable[i].coreCoder == CODEC_AACLD && core == AOT_ER_AAC_ELD) || \ + (sbrTuningTable[i].coreCoder == CODEC_AAC && core != AOT_ER_AAC_ELD)) + + for (i = 0; i < sbrTuningTableSize; i++) { + if (isForThisCore(i)) /* tuning table is for this core codec */ { - if ( numChannels == sbrTuningTable [i].numChannels - && sampleRate == sbrTuningTable [i].sampleRate ) - { + if (numChannels == sbrTuningTable[i].numChannels && + sampleRate == sbrTuningTable[i].sampleRate) { found = 1; - if ((bitrate >= sbrTuningTable [i].bitrateFrom) && - (bitrate < sbrTuningTable [i].bitrateTo)) { - bitRateClosestLower = bitrate; - bitRateClosestUpper = bitrate; - //FDKprintf("entry %d\n", i); - return i ; + if ((bitrate >= sbrTuningTable[i].bitrateFrom) && + (bitrate < sbrTuningTable[i].bitrateTo)) { + return i; } else { - if ( sbrTuningTable [i].bitrateFrom > bitrate ) { - if (sbrTuningTable [i].bitrateFrom < bitRateClosestLower) { - bitRateClosestLower = sbrTuningTable [i].bitrateFrom; + if (sbrTuningTable[i].bitrateFrom > bitrate) { + if (sbrTuningTable[i].bitrateFrom < bitRateClosestLower) { + bitRateClosestLower = sbrTuningTable[i].bitrateFrom; bitRateClosestLowerIndex = i; } } - if ( sbrTuningTable [i].bitrateTo <= bitrate ) { - if (sbrTuningTable [i].bitrateTo > bitRateClosestUpper) { - bitRateClosestUpper = sbrTuningTable [i].bitrateTo-1; + if (sbrTuningTable[i].bitrateTo <= bitrate) { + if (sbrTuningTable[i].bitrateTo > bitRateClosestUpper) { + bitRateClosestUpper = sbrTuningTable[i].bitrateTo - 1; bitRateClosestUpperIndex = i; } } @@ -206,20 +259,25 @@ getSbrTuningTableIndex(UINT bitrate, /*! the total bitrate in bits/sec */ } } - if (pBitRateClosest != NULL) - { - /* If there was at least one matching tuning entry found then pick the least distance bit rate */ - if (found) - { - int distanceUpper=DISTANCE_CEIL_VALUE, distanceLower=DISTANCE_CEIL_VALUE; + if (bitRateClosestUpperIndex >= 0) { + return bitRateClosestUpperIndex; + } + + if (pBitRateClosest != NULL) { + /* If there was at least one matching tuning entry pick the least distance + * bit rate */ + if (found) { + int distanceUpper = DISTANCE_CEIL_VALUE, + distanceLower = DISTANCE_CEIL_VALUE; if (bitRateClosestLowerIndex >= 0) { - distanceLower = sbrTuningTable [bitRateClosestLowerIndex].bitrateFrom - bitrate; + distanceLower = + sbrTuningTable[bitRateClosestLowerIndex].bitrateFrom - bitrate; } if (bitRateClosestUpperIndex >= 0) { - distanceUpper = bitrate - sbrTuningTable [bitRateClosestUpperIndex].bitrateTo; + distanceUpper = + bitrate - sbrTuningTable[bitRateClosestUpperIndex].bitrateTo; } - if ( distanceUpper < distanceLower ) - { + if (distanceUpper < distanceLower) { *pBitRateClosest = bitRateClosestUpper; } else { *pBitRateClosest = bitRateClosestLower; @@ -241,44 +299,47 @@ getSbrTuningTableIndex(UINT bitrate, /*! the total bitrate in bits/sec */ \return Index to the appropriate table ****************************************************************************/ -static INT -getPsTuningTableIndex(UINT bitrate, UINT *pBitRateClosest){ - - INT i, paramSets = sizeof (psTuningTable) / sizeof (psTuningTable [0]); - int bitRateClosestLowerIndex=-1, bitRateClosestUpperIndex=-1; - UINT bitRateClosestUpper = 0, bitRateClosestLower=DISTANCE_CEIL_VALUE; - - for (i = 0 ; i < paramSets ; i++) { - if ((bitrate >= psTuningTable [i].bitrateFrom) && - (bitrate < psTuningTable [i].bitrateTo)) { - return i ; +static INT getPsTuningTableIndex(UINT bitrate, UINT *pBitRateClosest) { + INT i, paramSets = sizeof(psTuningTable) / sizeof(psTuningTable[0]); + int bitRateClosestLowerIndex = -1, bitRateClosestUpperIndex = -1; + UINT bitRateClosestUpper = 0, bitRateClosestLower = DISTANCE_CEIL_VALUE; + + for (i = 0; i < paramSets; i++) { + if ((bitrate >= psTuningTable[i].bitrateFrom) && + (bitrate < psTuningTable[i].bitrateTo)) { + return i; } else { - if ( psTuningTable [i].bitrateFrom > bitrate ) { - if (psTuningTable [i].bitrateFrom < bitRateClosestLower) { - bitRateClosestLower = psTuningTable [i].bitrateFrom; + if (psTuningTable[i].bitrateFrom > bitrate) { + if (psTuningTable[i].bitrateFrom < bitRateClosestLower) { + bitRateClosestLower = psTuningTable[i].bitrateFrom; bitRateClosestLowerIndex = i; } } - if ( psTuningTable [i].bitrateTo <= bitrate ) { - if (psTuningTable [i].bitrateTo > bitRateClosestUpper) { - bitRateClosestUpper = psTuningTable [i].bitrateTo-1; + if (psTuningTable[i].bitrateTo <= bitrate) { + if (psTuningTable[i].bitrateTo > bitRateClosestUpper) { + bitRateClosestUpper = psTuningTable[i].bitrateTo - 1; bitRateClosestUpperIndex = i; } } } } - if (pBitRateClosest != NULL) - { - int distanceUpper=DISTANCE_CEIL_VALUE, distanceLower=DISTANCE_CEIL_VALUE; + if (bitRateClosestUpperIndex >= 0) { + return bitRateClosestUpperIndex; + } + + if (pBitRateClosest != NULL) { + int distanceUpper = DISTANCE_CEIL_VALUE, + distanceLower = DISTANCE_CEIL_VALUE; if (bitRateClosestLowerIndex >= 0) { - distanceLower = sbrTuningTable [bitRateClosestLowerIndex].bitrateFrom - bitrate; + distanceLower = + sbrTuningTable[bitRateClosestLowerIndex].bitrateFrom - bitrate; } if (bitRateClosestUpperIndex >= 0) { - distanceUpper = bitrate - sbrTuningTable [bitRateClosestUpperIndex].bitrateTo; + distanceUpper = + bitrate - sbrTuningTable[bitRateClosestUpperIndex].bitrateTo; } - if ( distanceUpper < distanceLower ) - { + if (distanceUpper < distanceLower) { *pBitRateClosest = bitRateClosestUpper; } else { *pBitRateClosest = bitRateClosestLower; @@ -300,41 +361,29 @@ getPsTuningTableIndex(UINT bitrate, UINT *pBitRateClosest){ \ingroup SbrEncCfg ****************************************************************************/ -static INT -FDKsbrEnc_GetDownsampledStopFreq ( - const INT sampleRateCore, - const INT startFreq, - INT stopFreq, - const INT downSampleFactor - ) -{ +static INT FDKsbrEnc_GetDownsampledStopFreq(const INT sampleRateCore, + const INT startFreq, INT stopFreq, + const INT downSampleFactor) { INT maxStopFreqRaw = sampleRateCore / 2; INT startBand, stopBand; HANDLE_ERROR_INFO err; - while (stopFreq > 0 && FDKsbrEnc_getSbrStopFreqRAW(stopFreq, sampleRateCore) > maxStopFreqRaw) { + while (stopFreq > 0 && FDKsbrEnc_getSbrStopFreqRAW(stopFreq, sampleRateCore) > + maxStopFreqRaw) { stopFreq--; } - if (FDKsbrEnc_getSbrStopFreqRAW( stopFreq, sampleRateCore) > maxStopFreqRaw) + if (FDKsbrEnc_getSbrStopFreqRAW(stopFreq, sampleRateCore) > maxStopFreqRaw) return -1; - err = FDKsbrEnc_FindStartAndStopBand ( - sampleRateCore<<(downSampleFactor-1), - sampleRateCore, - 32<<(downSampleFactor-1), - startFreq, - stopFreq, - &startBand, - &stopBand - ); - if (err) - return -1; + err = FDKsbrEnc_FindStartAndStopBand( + sampleRateCore << (downSampleFactor - 1), sampleRateCore, + 32 << (downSampleFactor - 1), startFreq, stopFreq, &startBand, &stopBand); + if (err) return -1; return stopFreq; } - /***************************************************************************/ /*! @@ -345,22 +394,18 @@ FDKsbrEnc_GetDownsampledStopFreq ( \return a flag indicating success: yes (1) or no (0) ****************************************************************************/ -static UINT -FDKsbrEnc_IsSbrSettingAvail ( - UINT bitrate, /*! the total bitrate in bits/sec */ - UINT vbrMode, /*! the vbr paramter, 0 means constant bitrate */ - UINT numOutputChannels, /*! the number of channels for the core coder */ - UINT sampleRateInput, /*! the input sample rate [in Hz] */ - UINT sampleRateCore, /*! the core's sampling rate */ - AUDIO_OBJECT_TYPE core - ) -{ +static UINT FDKsbrEnc_IsSbrSettingAvail( + UINT bitrate, /*! the total bitrate in bits/sec */ + UINT vbrMode, /*! the vbr paramter, 0 means constant bitrate */ + UINT numOutputChannels, /*! the number of channels for the core coder */ + UINT sampleRateInput, /*! the input sample rate [in Hz] */ + UINT sampleRateCore, /*! the core's sampling rate */ + AUDIO_OBJECT_TYPE core) { INT idx = INVALID_TABLE_IDX; - if (sampleRateInput < 16000) - return 0; + if (sampleRateInput < 16000) return 0; - if (bitrate==0) { + if (bitrate == 0) { /* map vbr quality to bitrate */ if (vbrMode < 30) bitrate = 24000; @@ -375,12 +420,12 @@ FDKsbrEnc_IsSbrSettingAvail ( bitrate *= numOutputChannels; } - idx = getSbrTuningTableIndex(bitrate, numOutputChannels, sampleRateCore, core, NULL); + idx = getSbrTuningTableIndex(bitrate, numOutputChannels, sampleRateCore, core, + NULL); return (idx == INVALID_TABLE_IDX ? 0 : 1); } - /***************************************************************************/ /*! @@ -390,46 +435,46 @@ FDKsbrEnc_IsSbrSettingAvail ( \return A flag indicating success: yes (1) or no (0) ****************************************************************************/ -static UINT -FDKsbrEnc_AdjustSbrSettings (const sbrConfigurationPtr config, /*! output, modified */ - UINT bitRate, /*! the total bitrate in bits/sec */ - UINT numChannels, /*! the core coder number of channels */ - UINT sampleRateCore, /*! the core coder sampling rate in Hz */ - UINT sampleRateSbr, /*! the sbr coder sampling rate in Hz */ - UINT transFac, /*! the short block to long block ratio */ - UINT standardBitrate, /*! the standard bitrate per channel in bits/sec */ - UINT vbrMode, /*! the vbr paramter, 0 poor quality .. 100 high quality*/ - UINT useSpeechConfig, /*!< adapt tuning parameters for speech ? */ - UINT lcsMode, /*! the low complexity stereo mode */ - UINT bParametricStereo, /*!< use parametric stereo */ - AUDIO_OBJECT_TYPE core) /* Core audio codec object type */ +static UINT FDKsbrEnc_AdjustSbrSettings( + const sbrConfigurationPtr config, /*! output, modified */ + UINT bitRate, /*! the total bitrate in bits/sec */ + UINT numChannels, /*! the core coder number of channels */ + UINT sampleRateCore, /*! the core coder sampling rate in Hz */ + UINT sampleRateSbr, /*! the sbr coder sampling rate in Hz */ + UINT transFac, /*! the short block to long block ratio */ + UINT standardBitrate, /*! the standard bitrate per channel in bits/sec */ + UINT vbrMode, /*! the vbr paramter, 0 poor quality .. 100 high quality*/ + UINT useSpeechConfig, /*!< adapt tuning parameters for speech ? */ + UINT lcsMode, /*! the low complexity stereo mode */ + UINT bParametricStereo, /*!< use parametric stereo */ + AUDIO_OBJECT_TYPE core) /* Core audio codec object type */ { INT idx = INVALID_TABLE_IDX; /* set the core codec settings */ - config->codecSettings.bitRate = bitRate; - config->codecSettings.nChannels = numChannels; - config->codecSettings.sampleFreq = sampleRateCore; - config->codecSettings.transFac = transFac; + config->codecSettings.bitRate = bitRate; + config->codecSettings.nChannels = numChannels; + config->codecSettings.sampleFreq = sampleRateCore; + config->codecSettings.transFac = transFac; config->codecSettings.standardBitrate = standardBitrate; if (bitRate < 28000) { config->threshold_AmpRes_FF_m = (FIXP_DBL)MAXVAL_DBL; config->threshold_AmpRes_FF_e = 7; - } - else if (bitRate >= 28000 && bitRate <= 48000) { + } else if (bitRate >= 28000 && bitRate <= 48000) { /* The float threshold is 75 - 0.524288f is fractional part of RELAXATION, the quotaMatrix and therefore tonality are scaled by this - 2/3 is because the original implementation divides the tonality values by 3, here it's divided by 2 - 128 compensates the necessary shiftfactor of 7 */ - config->threshold_AmpRes_FF_m = FL2FXCONST_DBL(75.0f*0.524288f/(2.0f/3.0f)/128.0f); + 0.524288f is fractional part of RELAXATION, the quotaMatrix and therefore + tonality are scaled by this 2/3 is because the original implementation + divides the tonality values by 3, here it's divided by 2 128 compensates + the necessary shiftfactor of 7 */ + config->threshold_AmpRes_FF_m = + FL2FXCONST_DBL(75.0f * 0.524288f / (2.0f / 3.0f) / 128.0f); config->threshold_AmpRes_FF_e = 7; - } - else if (bitRate > 48000) { + } else if (bitRate > 48000) { config->threshold_AmpRes_FF_m = FL2FXCONST_DBL(0); config->threshold_AmpRes_FF_e = 0; } - if (bitRate==0) { + if (bitRate == 0) { /* map vbr quality to bitrate */ if (vbrMode < 30) bitRate = 24000; @@ -443,31 +488,29 @@ FDKsbrEnc_AdjustSbrSettings (const sbrConfigurationPtr config, /*! output, modif bitRate = 48000; bitRate *= numChannels; /* fix to enable mono vbrMode<40 @ 44.1 of 48kHz */ - if (numChannels==1) { - if (sampleRateSbr==44100 || sampleRateSbr==48000) { - if (vbrMode<40) bitRate = 32000; + if (numChannels == 1) { + if (sampleRateSbr == 44100 || sampleRateSbr == 48000) { + if (vbrMode < 40) bitRate = 32000; } } } - idx = getSbrTuningTableIndex(bitRate,numChannels,sampleRateCore, core, NULL); + idx = + getSbrTuningTableIndex(bitRate, numChannels, sampleRateCore, core, NULL); if (idx != INVALID_TABLE_IDX) { - config->startFreq = sbrTuningTable[idx].startFreq ; - config->stopFreq = sbrTuningTable[idx].stopFreq ; + config->startFreq = sbrTuningTable[idx].startFreq; + config->stopFreq = sbrTuningTable[idx].stopFreq; if (useSpeechConfig) { - config->startFreq = sbrTuningTable[idx].startFreqSpeech; - config->stopFreq = sbrTuningTable[idx].stopFreqSpeech; + config->startFreq = sbrTuningTable[idx].startFreqSpeech; + config->stopFreq = sbrTuningTable[idx].stopFreqSpeech; } /* Adapt stop frequency in case of downsampled SBR - only 32 bands then */ if (1 == config->downSampleFactor) { INT dsStopFreq = FDKsbrEnc_GetDownsampledStopFreq( - sampleRateCore, - config->startFreq, - config->stopFreq, - config->downSampleFactor - ); + sampleRateCore, config->startFreq, config->stopFreq, + config->downSampleFactor); if (dsStopFreq < 0) { return 0; } @@ -475,52 +518,68 @@ FDKsbrEnc_AdjustSbrSettings (const sbrConfigurationPtr config, /*! output, modif config->stopFreq = dsStopFreq; } - config->sbr_noise_bands = sbrTuningTable[idx].numNoiseBands ; - if (core == AOT_ER_AAC_ELD) - config->init_amp_res_FF = SBR_AMP_RES_1_5; - config->noiseFloorOffset= sbrTuningTable[idx].noiseFloorOffset; + config->sbr_noise_bands = sbrTuningTable[idx].numNoiseBands; + if (core == AOT_ER_AAC_ELD) config->init_amp_res_FF = SBR_AMP_RES_1_5; + config->noiseFloorOffset = sbrTuningTable[idx].noiseFloorOffset; - config->ana_max_level = sbrTuningTable[idx].noiseMaxLevel ; - config->stereoMode = sbrTuningTable[idx].stereoMode ; - config->freqScale = sbrTuningTable[idx].freqScale ; + config->ana_max_level = sbrTuningTable[idx].noiseMaxLevel; + config->stereoMode = sbrTuningTable[idx].stereoMode; + config->freqScale = sbrTuningTable[idx].freqScale; if (numChannels == 1) { /* stereo case */ switch (core) { case AOT_AAC_LC: - if (bitRate <= (useSpeechConfig?24000U:20000U)) { - config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency resolution for non-split frames */ - config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency resolution for split frames */ + if (bitRate <= (useSpeechConfig ? 24000U : 20000U)) { + config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency + resolution for + non-split frames */ + config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency + resolution for split + frames */ } break; case AOT_ER_AAC_ELD: if (bitRate < 36000) - config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency resolution for split frames */ + config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency + resolution for split + frames */ if (bitRate < 26000) { - config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency resolution for non-split frames */ - config->fResTransIsLow = 1; /* for transient frames, set low frequency resolution */ + config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency + resolution for + non-split frames */ + config->fResTransIsLow = + 1; /* for transient frames, set low frequency resolution */ } break; default: break; } - } - else { + } else { /* stereo case */ switch (core) { case AOT_AAC_LC: if (bitRate <= 28000) { - config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency resolution for non-split frames */ - config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency resolution for split frames */ + config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency + resolution for + non-split frames */ + config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency + resolution for split + frames */ } break; case AOT_ER_AAC_ELD: if (bitRate < 72000) { - config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency resolution for split frames */ + config->freq_res_fixfix[1] = FREQ_RES_LOW; /* set low frequency + resolution for split + frames */ } if (bitRate < 52000) { - config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency resolution for non-split frames */ - config->fResTransIsLow = 1; /* for transient frames, set low frequency resolution */ + config->freq_res_fixfix[0] = FREQ_RES_LOW; /* set low frequency + resolution for + non-split frames */ + config->fResTransIsLow = + 1; /* for transient frames, set low frequency resolution */ } break; default: @@ -535,24 +594,22 @@ FDKsbrEnc_AdjustSbrSettings (const sbrConfigurationPtr config, /*! output, modif } } - /* adjust usage of parametric coding dependent on bitrate and speech config flag */ - if (useSpeechConfig) - config->parametricCoding = 0; + /* adjust usage of parametric coding dependent on bitrate and speech config + * flag */ + if (useSpeechConfig) config->parametricCoding = 0; if (core == AOT_ER_AAC_ELD) { - if (bitRate < 28000) - config->init_amp_res_FF = SBR_AMP_RES_3_0; + if (bitRate < 28000) config->init_amp_res_FF = SBR_AMP_RES_3_0; config->SendHeaderDataTime = -1; } if (numChannels == 1) { if (bitRate < 16000) { - config->parametricCoding = 0; + config->parametricCoding = 0; } - } - else { + } else { if (bitRate < 20000) { - config->parametricCoding = 0; + config->parametricCoding = 0; } } @@ -561,17 +618,16 @@ FDKsbrEnc_AdjustSbrSettings (const sbrConfigurationPtr config, /*! output, modif /* PS settings */ config->bParametricStereo = bParametricStereo; - return 1 ; - } - else { - return 0 ; + return 1; + } else { + return 0; } } /***************************************************************************** functionname: FDKsbrEnc_InitializeSbrDefaults - description: initializes the SBR confifuration + description: initializes the SBR configuration returns: error status input: - core codec type, - factor of SBR to core frame length, @@ -579,76 +635,73 @@ FDKsbrEnc_AdjustSbrSettings (const sbrConfigurationPtr config, /*! output, modif output: initialized SBR configuration *****************************************************************************/ -static UINT -FDKsbrEnc_InitializeSbrDefaults (sbrConfigurationPtr config, - INT downSampleFactor, - UINT codecGranuleLen - ,const INT isLowDelay - ) -{ - if ( (downSampleFactor < 1 || downSampleFactor > 2) || - (codecGranuleLen*downSampleFactor > QMF_CHANNELS*QMF_MAX_TIME_SLOTS) ) - return(0); /* error */ - - config->SendHeaderDataTime = 1000; - config->useWaveCoding = 0; - config->crcSbr = 0; - config->dynBwSupported = 1; - if (isLowDelay) - config->tran_thr = 6000; - else - config->tran_thr = 13000; - - config->parametricCoding = 1; - - config->sbrFrameSize = codecGranuleLen * downSampleFactor; - config->downSampleFactor = downSampleFactor; - - /* sbr default parameters */ - config->sbr_data_extra = 0; - config->amp_res = SBR_AMP_RES_3_0 ; - config->tran_fc = 0 ; - config->tran_det_mode = 1 ; - config->spread = 1 ; - config->stat = 0 ; - config->e = 1 ; - config->deltaTAcrossFrames = 1 ; - config->dF_edge_1stEnv = FL2FXCONST_DBL(0.3f) ; - config->dF_edge_incr = FL2FXCONST_DBL(0.3f) ; - - config->sbr_invf_mode = INVF_SWITCHED; - config->sbr_xpos_mode = XPOS_LC; - config->sbr_xpos_ctrl = SBR_XPOS_CTRL_DEFAULT; - config->sbr_xpos_level = 0; - config->useSaPan = 0; - config->dynBwEnabled = 0; - - - /* the following parameters are overwritten by the FDKsbrEnc_AdjustSbrSettings() function since - they are included in the tuning table */ - config->stereoMode = SBR_SWITCH_LRC; - config->ana_max_level = 6; - config->noiseFloorOffset = 0; - config->startFreq = 5; /* 5.9 respectively 6.0 kHz at fs = 44.1/48 kHz */ - config->stopFreq = 9; /* 16.2 respectively 16.8 kHz at fs = 44.1/48 kHz */ - config->freq_res_fixfix[0] = FREQ_RES_HIGH; /* non-split case */ - config->freq_res_fixfix[1] = FREQ_RES_HIGH; /* split case */ - config->fResTransIsLow = 0; /* for transient frames, set variable frequency resolution according to freqResTable */ - - /* header_extra_1 */ - config->freqScale = SBR_FREQ_SCALE_DEFAULT; - config->alterScale = SBR_ALTER_SCALE_DEFAULT; - config->sbr_noise_bands = SBR_NOISE_BANDS_DEFAULT; - - /* header_extra_2 */ - config->sbr_limiter_bands = SBR_LIMITER_BANDS_DEFAULT; - config->sbr_limiter_gains = SBR_LIMITER_GAINS_DEFAULT; - config->sbr_interpol_freq = SBR_INTERPOL_FREQ_DEFAULT; - config->sbr_smoothing_length = SBR_SMOOTHING_LENGTH_DEFAULT; +static UINT FDKsbrEnc_InitializeSbrDefaults(sbrConfigurationPtr config, + INT downSampleFactor, + UINT codecGranuleLen, + const INT isLowDelay) { + if ((downSampleFactor < 1 || downSampleFactor > 2) || + (codecGranuleLen * downSampleFactor > 64 * 32)) + return (0); /* error */ + + config->SendHeaderDataTime = 1000; + config->useWaveCoding = 0; + config->crcSbr = 0; + config->dynBwSupported = 1; + if (isLowDelay) + config->tran_thr = 6000; + else + config->tran_thr = 13000; + + config->parametricCoding = 1; + + config->sbrFrameSize = codecGranuleLen * downSampleFactor; + config->downSampleFactor = downSampleFactor; + + /* sbr default parameters */ + config->sbr_data_extra = 0; + config->amp_res = SBR_AMP_RES_3_0; + config->tran_fc = 0; + config->tran_det_mode = 1; + config->spread = 1; + config->stat = 0; + config->e = 1; + config->deltaTAcrossFrames = 1; + config->dF_edge_1stEnv = FL2FXCONST_DBL(0.3f); + config->dF_edge_incr = FL2FXCONST_DBL(0.3f); + + config->sbr_invf_mode = INVF_SWITCHED; + config->sbr_xpos_mode = XPOS_LC; + config->sbr_xpos_ctrl = SBR_XPOS_CTRL_DEFAULT; + config->sbr_xpos_level = 0; + config->useSaPan = 0; + config->dynBwEnabled = 0; + + /* the following parameters are overwritten by the + FDKsbrEnc_AdjustSbrSettings() function since they are included in the + tuning table */ + config->stereoMode = SBR_SWITCH_LRC; + config->ana_max_level = 6; + config->noiseFloorOffset = 0; + config->startFreq = 5; /* 5.9 respectively 6.0 kHz at fs = 44.1/48 kHz */ + config->stopFreq = 9; /* 16.2 respectively 16.8 kHz at fs = 44.1/48 kHz */ + config->freq_res_fixfix[0] = FREQ_RES_HIGH; /* non-split case */ + config->freq_res_fixfix[1] = FREQ_RES_HIGH; /* split case */ + config->fResTransIsLow = 0; /* for transient frames, set variable frequency + resolution according to freqResTable */ - return 1; -} + /* header_extra_1 */ + config->freqScale = SBR_FREQ_SCALE_DEFAULT; + config->alterScale = SBR_ALTER_SCALE_DEFAULT; + config->sbr_noise_bands = SBR_NOISE_BANDS_DEFAULT; + /* header_extra_2 */ + config->sbr_limiter_bands = SBR_LIMITER_BANDS_DEFAULT; + config->sbr_limiter_gains = SBR_LIMITER_GAINS_DEFAULT; + config->sbr_interpol_freq = SBR_INTERPOL_FREQ_DEFAULT; + config->sbr_smoothing_length = SBR_SMOOTHING_LENGTH_DEFAULT; + + return 1; +} /***************************************************************************** @@ -659,19 +712,14 @@ FDKsbrEnc_InitializeSbrDefaults (sbrConfigurationPtr config, output: released handle *****************************************************************************/ -static void -deleteEnvChannel (HANDLE_ENV_CHANNEL hEnvCut) -{ +static void deleteEnvChannel(HANDLE_ENV_CHANNEL hEnvCut) { if (hEnvCut) { - FDKsbrEnc_DeleteTonCorrParamExtr(&hEnvCut->TonCorr); - FDKsbrEnc_deleteExtractSbrEnvelope (&hEnvCut->sbrExtractEnvelope); + FDKsbrEnc_deleteExtractSbrEnvelope(&hEnvCut->sbrExtractEnvelope); } - } - /***************************************************************************** functionname: sbrEncoder_ChannelClose @@ -681,12 +729,9 @@ deleteEnvChannel (HANDLE_ENV_CHANNEL hEnvCut) output: *****************************************************************************/ -static void -sbrEncoder_ChannelClose(HANDLE_SBR_CHANNEL hSbrChannel) -{ - if (hSbrChannel != NULL) - { - deleteEnvChannel (&hSbrChannel->hEnvChannel); +static void sbrEncoder_ChannelClose(HANDLE_SBR_CHANNEL hSbrChannel) { + if (hSbrChannel != NULL) { + deleteEnvChannel(&hSbrChannel->hEnvChannel); } } @@ -699,67 +744,60 @@ sbrEncoder_ChannelClose(HANDLE_SBR_CHANNEL hSbrChannel) output: *****************************************************************************/ -static void -sbrEncoder_ElementClose(HANDLE_SBR_ELEMENT *phSbrElement) -{ +static void sbrEncoder_ElementClose(HANDLE_SBR_ELEMENT *phSbrElement) { HANDLE_SBR_ELEMENT hSbrElement = *phSbrElement; - if (hSbrElement!=NULL) { + if (hSbrElement != NULL) { if (hSbrElement->sbrConfigData.v_k_master) FreeRam_Sbr_v_k_master(&hSbrElement->sbrConfigData.v_k_master); if (hSbrElement->sbrConfigData.freqBandTable[LO]) - FreeRam_Sbr_freqBandTableLO(&hSbrElement->sbrConfigData.freqBandTable[LO]); + FreeRam_Sbr_freqBandTableLO( + &hSbrElement->sbrConfigData.freqBandTable[LO]); if (hSbrElement->sbrConfigData.freqBandTable[HI]) - FreeRam_Sbr_freqBandTableHI(&hSbrElement->sbrConfigData.freqBandTable[HI]); + FreeRam_Sbr_freqBandTableHI( + &hSbrElement->sbrConfigData.freqBandTable[HI]); FreeRam_SbrElement(phSbrElement); } - return ; - + return; } - -void sbrEncoder_Close (HANDLE_SBR_ENCODER *phSbrEncoder) -{ +void sbrEncoder_Close(HANDLE_SBR_ENCODER *phSbrEncoder) { HANDLE_SBR_ENCODER hSbrEncoder = *phSbrEncoder; - if (hSbrEncoder != NULL) - { + if (hSbrEncoder != NULL) { int el, ch; - for (el=0; el<(8); el++) - { - if (hSbrEncoder->sbrElement[el]!=NULL) { + for (el = 0; el < (8); el++) { + if (hSbrEncoder->sbrElement[el] != NULL) { sbrEncoder_ElementClose(&hSbrEncoder->sbrElement[el]); } } /* Close sbr Channels */ - for (ch=0; ch<(8); ch++) - { + for (ch = 0; ch < (8); ch++) { if (hSbrEncoder->pSbrChannel[ch]) { sbrEncoder_ChannelClose(hSbrEncoder->pSbrChannel[ch]); FreeRam_SbrChannel(&hSbrEncoder->pSbrChannel[ch]); } if (hSbrEncoder->QmfAnalysis[ch].FilterStates) - FreeRam_Sbr_QmfStatesAnalysis((FIXP_QAS**)&hSbrEncoder->QmfAnalysis[ch].FilterStates); - - + FreeRam_Sbr_QmfStatesAnalysis( + (FIXP_QAS **)&hSbrEncoder->QmfAnalysis[ch].FilterStates); } if (hSbrEncoder->hParametricStereo) PSEnc_Destroy(&hSbrEncoder->hParametricStereo); if (hSbrEncoder->qmfSynthesisPS.FilterStates) - FreeRam_PsQmfStatesSynthesis((FIXP_DBL**)&hSbrEncoder->qmfSynthesisPS.FilterStates); + FreeRam_PsQmfStatesSynthesis( + (FIXP_DBL **)&hSbrEncoder->qmfSynthesisPS.FilterStates); /* Release Overlay */ - FreeRam_SbrDynamic_RAM((FIXP_DBL**)&hSbrEncoder->pSBRdynamic_RAM); - + if (hSbrEncoder->pSBRdynamic_RAM) + FreeRam_SbrDynamic_RAM((FIXP_DBL **)&hSbrEncoder->pSBRdynamic_RAM); FreeRam_SbrEncoder(phSbrEncoder); } - } /***************************************************************************** @@ -771,67 +809,44 @@ void sbrEncoder_Close (HANDLE_SBR_ENCODER *phSbrEncoder) output: error info *****************************************************************************/ -static INT updateFreqBandTable( - HANDLE_SBR_CONFIG_DATA sbrConfigData, - HANDLE_SBR_HEADER_DATA sbrHeaderData, - const INT downSampleFactor - ) -{ +static INT updateFreqBandTable(HANDLE_SBR_CONFIG_DATA sbrConfigData, + HANDLE_SBR_HEADER_DATA sbrHeaderData, + const INT downSampleFactor) { INT k0, k2; - if( FDKsbrEnc_FindStartAndStopBand ( - sbrConfigData->sampleFreq, - sbrConfigData->sampleFreq >> (downSampleFactor-1), - sbrConfigData->noQmfBands, - sbrHeaderData->sbr_start_frequency, - sbrHeaderData->sbr_stop_frequency, - &k0, - &k2 - ) - ) - return(1); - - - if( FDKsbrEnc_UpdateFreqScale( - sbrConfigData->v_k_master, - &sbrConfigData->num_Master, - k0, - k2, - sbrHeaderData->freqScale, - sbrHeaderData->alterScale - ) - ) - return(1); - - - sbrHeaderData->sbr_xover_band=0; - - - if( FDKsbrEnc_UpdateHiRes( - sbrConfigData->freqBandTable[HI], - &sbrConfigData->nSfb[HI], - sbrConfigData->v_k_master, - sbrConfigData->num_Master, - &sbrHeaderData->sbr_xover_band - ) - ) - return(1); + if (FDKsbrEnc_FindStartAndStopBand( + sbrConfigData->sampleFreq, + sbrConfigData->sampleFreq >> (downSampleFactor - 1), + sbrConfigData->noQmfBands, sbrHeaderData->sbr_start_frequency, + sbrHeaderData->sbr_stop_frequency, &k0, &k2)) + return (1); + if (FDKsbrEnc_UpdateFreqScale( + sbrConfigData->v_k_master, &sbrConfigData->num_Master, k0, k2, + sbrHeaderData->freqScale, sbrHeaderData->alterScale)) + return (1); - FDKsbrEnc_UpdateLoRes( - sbrConfigData->freqBandTable[LO], - &sbrConfigData->nSfb[LO], - sbrConfigData->freqBandTable[HI], - sbrConfigData->nSfb[HI] - ); + sbrHeaderData->sbr_xover_band = 0; + if (FDKsbrEnc_UpdateHiRes(sbrConfigData->freqBandTable[HI], + &sbrConfigData->nSfb[HI], sbrConfigData->v_k_master, + sbrConfigData->num_Master, + &sbrHeaderData->sbr_xover_band)) + return (1); - sbrConfigData->xOverFreq = (sbrConfigData->freqBandTable[LOW_RES][0] * sbrConfigData->sampleFreq / sbrConfigData->noQmfBands+1)>>1; + FDKsbrEnc_UpdateLoRes( + sbrConfigData->freqBandTable[LO], &sbrConfigData->nSfb[LO], + sbrConfigData->freqBandTable[HI], sbrConfigData->nSfb[HI]); + + sbrConfigData->xOverFreq = + (sbrConfigData->freqBandTable[LOW_RES][0] * sbrConfigData->sampleFreq / + sbrConfigData->noQmfBands + + 1) >> + 1; return (0); } - /***************************************************************************** functionname: resetEnvChannel @@ -841,27 +856,26 @@ static INT updateFreqBandTable( output: hEnv *****************************************************************************/ -static INT resetEnvChannel (HANDLE_SBR_CONFIG_DATA sbrConfigData, - HANDLE_SBR_HEADER_DATA sbrHeaderData, - HANDLE_ENV_CHANNEL hEnv) -{ - /* note !!! hEnv->encEnvData.noOfnoisebands will be updated later in function FDKsbrEnc_extractSbrEnvelope !!!*/ - hEnv->TonCorr.sbrNoiseFloorEstimate.noiseBands = sbrHeaderData->sbr_noise_bands; - - - if(FDKsbrEnc_ResetTonCorrParamExtr(&hEnv->TonCorr, - sbrConfigData->xposCtrlSwitch, - sbrConfigData->freqBandTable[HI][0], - sbrConfigData->v_k_master, - sbrConfigData->num_Master, - sbrConfigData->sampleFreq, - sbrConfigData->freqBandTable, - sbrConfigData->nSfb, - sbrConfigData->noQmfBands)) - return(1); - - hEnv->sbrCodeNoiseFloor.nSfb[LO] = hEnv->TonCorr.sbrNoiseFloorEstimate.noNoiseBands; - hEnv->sbrCodeNoiseFloor.nSfb[HI] = hEnv->TonCorr.sbrNoiseFloorEstimate.noNoiseBands; +static INT resetEnvChannel(HANDLE_SBR_CONFIG_DATA sbrConfigData, + HANDLE_SBR_HEADER_DATA sbrHeaderData, + HANDLE_ENV_CHANNEL hEnv) { + /* note !!! hEnv->encEnvData.noOfnoisebands will be updated later in function + * FDKsbrEnc_extractSbrEnvelope !!!*/ + hEnv->TonCorr.sbrNoiseFloorEstimate.noiseBands = + sbrHeaderData->sbr_noise_bands; + + if (FDKsbrEnc_ResetTonCorrParamExtr( + &hEnv->TonCorr, sbrConfigData->xposCtrlSwitch, + sbrConfigData->freqBandTable[HI][0], sbrConfigData->v_k_master, + sbrConfigData->num_Master, sbrConfigData->sampleFreq, + sbrConfigData->freqBandTable, sbrConfigData->nSfb, + sbrConfigData->noQmfBands)) + return (1); + + hEnv->sbrCodeNoiseFloor.nSfb[LO] = + hEnv->TonCorr.sbrNoiseFloorEstimate.noNoiseBands; + hEnv->sbrCodeNoiseFloor.nSfb[HI] = + hEnv->TonCorr.sbrNoiseFloorEstimate.noNoiseBands; hEnv->sbrCodeEnvelope.nSfb[LO] = sbrConfigData->nSfb[LO]; hEnv->sbrCodeEnvelope.nSfb[HI] = sbrConfigData->nSfb[HI]; @@ -874,16 +888,17 @@ static INT resetEnvChannel (HANDLE_SBR_CONFIG_DATA sbrConfigData, return (0); } -/* ****************************** FDKsbrEnc_SbrGetXOverFreq ******************************/ +/* ****************************** FDKsbrEnc_SbrGetXOverFreq + * ******************************/ /** * @fn * @brief calculates the closest possible crossover frequency * @return the crossover frequency SBR accepts * */ -static INT -FDKsbrEnc_SbrGetXOverFreq(HANDLE_SBR_ELEMENT hEnv, /*!< handle to SBR encoder instance */ - INT xoverFreq) /*!< from core coder suggested crossover frequency */ +static INT FDKsbrEnc_SbrGetXOverFreq( + HANDLE_SBR_ELEMENT hEnv, /*!< handle to SBR encoder instance */ + INT xoverFreq) /*!< from core coder suggested crossover frequency */ { INT band; INT lastDiff, newDiff; @@ -892,13 +907,15 @@ FDKsbrEnc_SbrGetXOverFreq(HANDLE_SBR_ELEMENT hEnv, /*!< handle to SBR en UCHAR *RESTRICT pVKMaster = hEnv->sbrConfigData.v_k_master; /* Check if there is a matching cutoff frequency in the master table */ - cutoffSb = (4*xoverFreq * hEnv->sbrConfigData.noQmfBands / hEnv->sbrConfigData.sampleFreq + 1)>>1; + cutoffSb = (4 * xoverFreq * hEnv->sbrConfigData.noQmfBands / + hEnv->sbrConfigData.sampleFreq + + 1) >> + 1; lastDiff = cutoffSb; for (band = 0; band < hEnv->sbrConfigData.num_Master; band++) { - newDiff = fixp_abs((INT)pVKMaster[band] - cutoffSb); - if(newDiff >= lastDiff) { + if (newDiff >= lastDiff) { band--; break; } @@ -906,7 +923,10 @@ FDKsbrEnc_SbrGetXOverFreq(HANDLE_SBR_ELEMENT hEnv, /*!< handle to SBR en lastDiff = newDiff; } - return ((pVKMaster[band] * hEnv->sbrConfigData.sampleFreq/hEnv->sbrConfigData.noQmfBands+1)>>1); + return ((pVKMaster[band] * hEnv->sbrConfigData.sampleFreq / + hEnv->sbrConfigData.noQmfBands + + 1) >> + 1); } /***************************************************************************** @@ -918,32 +938,27 @@ FDKsbrEnc_SbrGetXOverFreq(HANDLE_SBR_ELEMENT hEnv, /*!< handle to SBR en output: *****************************************************************************/ -INT -FDKsbrEnc_EnvEncodeFrame(HANDLE_SBR_ENCODER hEnvEncoder, - int iElement, - INT_PCM *samples, /*!< time samples, always interleaved */ - UINT timeInStride, /*!< time buffer channel interleaving stride */ - UINT *sbrDataBits, /*!< Size of SBR payload */ - UCHAR *sbrData, /*!< SBR payload */ - int clearOutput /*!< Do not consider any input signal */ - ) -{ +INT FDKsbrEnc_EnvEncodeFrame( + HANDLE_SBR_ENCODER hEnvEncoder, int iElement, + INT_PCM *samples, /*!< time samples, always deinterleaved */ + UINT samplesBufSize, /*!< time buffer channel stride */ + UINT *sbrDataBits, /*!< Size of SBR payload */ + UCHAR *sbrData, /*!< SBR payload */ + int clearOutput /*!< Do not consider any input signal */ +) { HANDLE_SBR_ELEMENT hSbrElement = NULL; - FDK_CRCINFO crcInfo; - INT crcReg; - INT ch; - INT band; - INT cutoffSb; - INT newXOver; - - if (hEnvEncoder == NULL) - return -1; + FDK_CRCINFO crcInfo; + INT crcReg; + INT ch; + INT band; + INT cutoffSb; + INT newXOver; - hSbrElement = hEnvEncoder->sbrElement[iElement]; + if (hEnvEncoder == NULL) return -1; - if (hSbrElement == NULL) - return -1; + hSbrElement = hEnvEncoder->sbrElement[iElement]; + if (hSbrElement == NULL) return -1; /* header bitstream handling */ HANDLE_SBR_BITSTREAM_DATA sbrBitstreamData = &hSbrElement->sbrBitstreamData; @@ -951,33 +966,33 @@ FDKsbrEnc_EnvEncodeFrame(HANDLE_SBR_ENCODER hEnvEncoder, INT psHeaderActive = 0; sbrBitstreamData->HeaderActive = 0; - /* Anticipate PS header because of internal PS bitstream delay in order to be in sync with SBR header. */ - if ( sbrBitstreamData->CountSendHeaderData==(sbrBitstreamData->NrSendHeaderData-1) ) - { - psHeaderActive = 1; + /* Anticipate PS header because of internal PS bitstream delay in order to be + * in sync with SBR header. */ + if (sbrBitstreamData->CountSendHeaderData == + (sbrBitstreamData->NrSendHeaderData - 1)) { + psHeaderActive = 1; } /* Signal SBR header to be written into bitstream */ - if ( sbrBitstreamData->CountSendHeaderData==0 ) - { - sbrBitstreamData->HeaderActive = 1; + if (sbrBitstreamData->CountSendHeaderData == 0) { + sbrBitstreamData->HeaderActive = 1; } /* Increment header interval counter */ if (sbrBitstreamData->NrSendHeaderData == 0) { sbrBitstreamData->CountSendHeaderData = 1; - } - else { + } else { if (sbrBitstreamData->CountSendHeaderData >= 0) { sbrBitstreamData->CountSendHeaderData++; - sbrBitstreamData->CountSendHeaderData %= sbrBitstreamData->NrSendHeaderData; + sbrBitstreamData->CountSendHeaderData %= + sbrBitstreamData->NrSendHeaderData; } } - if (hSbrElement->CmonData.dynBwEnabled ) { + if (hSbrElement->CmonData.dynBwEnabled) { INT i; - for ( i = 4; i > 0; i-- ) - hSbrElement->dynXOverFreqDelay[i] = hSbrElement->dynXOverFreqDelay[i-1]; + for (i = 4; i > 0; i--) + hSbrElement->dynXOverFreqDelay[i] = hSbrElement->dynXOverFreqDelay[i - 1]; hSbrElement->dynXOverFreqDelay[0] = hSbrElement->CmonData.dynXOverFreqEnc; if (hSbrElement->dynXOverFreqDelay[1] > hSbrElement->dynXOverFreqDelay[2]) @@ -986,41 +1001,38 @@ FDKsbrEnc_EnvEncodeFrame(HANDLE_SBR_ENCODER hEnvEncoder, newXOver = hSbrElement->dynXOverFreqDelay[1]; /* has the crossover frequency changed? */ - if ( hSbrElement->sbrConfigData.dynXOverFreq != newXOver ) { - + if (hSbrElement->sbrConfigData.dynXOverFreq != newXOver) { /* get corresponding master band */ - cutoffSb = ((4* newXOver * hSbrElement->sbrConfigData.noQmfBands - / hSbrElement->sbrConfigData.sampleFreq)+1)>>1; + cutoffSb = ((4 * newXOver * hSbrElement->sbrConfigData.noQmfBands / + hSbrElement->sbrConfigData.sampleFreq) + + 1) >> + 1; - for ( band = 0; band < hSbrElement->sbrConfigData.num_Master; band++ ) { - if ( cutoffSb == hSbrElement->sbrConfigData.v_k_master[band] ) - break; + for (band = 0; band < hSbrElement->sbrConfigData.num_Master; band++) { + if (cutoffSb == hSbrElement->sbrConfigData.v_k_master[band]) break; } - FDK_ASSERT( band < hSbrElement->sbrConfigData.num_Master ); + FDK_ASSERT(band < hSbrElement->sbrConfigData.num_Master); hSbrElement->sbrConfigData.dynXOverFreq = newXOver; hSbrElement->sbrHeaderData.sbr_xover_band = band; - hSbrElement->sbrBitstreamData.HeaderActive=1; + hSbrElement->sbrBitstreamData.HeaderActive = 1; psHeaderActive = 1; /* ps header is one frame delayed */ /* update vk_master table */ - if(updateFreqBandTable(&hSbrElement->sbrConfigData, - &hSbrElement->sbrHeaderData, - hEnvEncoder->downSampleFactor - )) - return(1); - + if (updateFreqBandTable(&hSbrElement->sbrConfigData, + &hSbrElement->sbrHeaderData, + hEnvEncoder->downSampleFactor)) + return (1); /* reset SBR channels */ INT nEnvCh = hSbrElement->sbrConfigData.nChannels; - for ( ch = 0; ch < nEnvCh; ch++ ) { - if(resetEnvChannel (&hSbrElement->sbrConfigData, + for (ch = 0; ch < nEnvCh; ch++) { + if (resetEnvChannel(&hSbrElement->sbrConfigData, &hSbrElement->sbrHeaderData, &hSbrElement->sbrChannel[ch]->hEnvChannel)) - return(1); - + return (1); } } } @@ -1028,11 +1040,11 @@ FDKsbrEnc_EnvEncodeFrame(HANDLE_SBR_ENCODER hEnvEncoder, /* allocate space for dummy header and crc */ - crcReg = FDKsbrEnc_InitSbrBitstream(&hSbrElement->CmonData, - hSbrElement->payloadDelayLine[hEnvEncoder->nBitstrDelay], - MAX_PAYLOAD_SIZE*sizeof(UCHAR), - &crcInfo, - hSbrElement->sbrConfigData.sbrSyntaxFlags); + crcReg = FDKsbrEnc_InitSbrBitstream( + &hSbrElement->CmonData, + hSbrElement->payloadDelayLine[hEnvEncoder->nBitstrDelay], + MAX_PAYLOAD_SIZE * sizeof(UCHAR), &crcInfo, + hSbrElement->sbrConfigData.sbrSyntaxFlags); /* Temporal Envelope Data */ SBR_FRAME_TEMP_DATA _fData; @@ -1047,61 +1059,47 @@ FDKsbrEnc_EnvEncodeFrame(HANDLE_SBR_ENCODER hEnvEncoder, FDKmemclear(&eData[1], sizeof(SBR_ENV_TEMP_DATA)); FDKmemclear(fData, sizeof(SBR_FRAME_TEMP_DATA)); - for(i=0; i<MAX_NUM_NOISE_VALUES; i++) - fData->res[i] = FREQ_RES_HIGH; + for (i = 0; i < MAX_NUM_NOISE_VALUES; i++) fData->res[i] = FREQ_RES_HIGH; } - - if (!clearOutput) - { + if (!clearOutput) { /* * Transform audio data into QMF domain */ - for(ch = 0; ch < hSbrElement->sbrConfigData.nChannels; ch++) - { + for (ch = 0; ch < hSbrElement->sbrConfigData.nChannels; ch++) { HANDLE_ENV_CHANNEL h_envChan = &hSbrElement->sbrChannel[ch]->hEnvChannel; HANDLE_SBR_EXTRACT_ENVELOPE sbrExtrEnv = &h_envChan->sbrExtractEnvelope; - if(hSbrElement->elInfo.fParametricStereo == 0) - { + if (hSbrElement->elInfo.fParametricStereo == 0) { QMF_SCALE_FACTOR tmpScale; FIXP_DBL **pQmfReal, **pQmfImag; - C_AALLOC_SCRATCH_START(qmfWorkBuffer, FIXP_DBL, QMF_CHANNELS*2) - + C_AALLOC_SCRATCH_START(qmfWorkBuffer, FIXP_DBL, 64 * 2) /* Obtain pointers to QMF buffers. */ pQmfReal = sbrExtrEnv->rBuffer; pQmfImag = sbrExtrEnv->iBuffer; - qmfAnalysisFiltering( hSbrElement->hQmfAnalysis[ch], - pQmfReal, - pQmfImag, - &tmpScale, - samples + hSbrElement->elInfo.ChannelIndex[ch], - timeInStride, - qmfWorkBuffer ); + qmfAnalysisFiltering( + hSbrElement->hQmfAnalysis[ch], pQmfReal, pQmfImag, &tmpScale, + samples + hSbrElement->elInfo.ChannelIndex[ch] * samplesBufSize, 0, + 1, qmfWorkBuffer); h_envChan->qmfScale = tmpScale.lb_scale + 7; - - C_AALLOC_SCRATCH_END(qmfWorkBuffer, FIXP_DBL, QMF_CHANNELS*2) + C_AALLOC_SCRATCH_END(qmfWorkBuffer, FIXP_DBL, 64 * 2) } /* fParametricStereo == 0 */ - /* Parametric Stereo processing */ - if (hSbrElement->elInfo.fParametricStereo) - { + if (hSbrElement->elInfo.fParametricStereo) { INT error = noError; - /* Limit Parametric Stereo to one instance */ FDK_ASSERT(ch == 0); - - if(error == noError){ + if (error == noError) { /* parametric stereo processing: - input: o left and right time domain samples @@ -1111,28 +1109,22 @@ FDKsbrEnc_EnvEncodeFrame(HANDLE_SBR_ENCODER hEnvEncoder, o ps parameter extraction o downmix + hybrid synthesis - output: - o downmixed qmf data is written to sbrExtrEnv->rBuffer and sbrExtrEnv->iBuffer + o downmixed qmf data is written to sbrExtrEnv->rBuffer and + sbrExtrEnv->iBuffer */ SCHAR qmfScale; - INT_PCM* pSamples[2] = {samples + hSbrElement->elInfo.ChannelIndex[0],samples + hSbrElement->elInfo.ChannelIndex[1]}; - error = FDKsbrEnc_PSEnc_ParametricStereoProcessing( hEnvEncoder->hParametricStereo, - pSamples, - timeInStride, - hSbrElement->hQmfAnalysis, - sbrExtrEnv->rBuffer, - sbrExtrEnv->iBuffer, - samples + hSbrElement->elInfo.ChannelIndex[ch], - &hEnvEncoder->qmfSynthesisPS, - &qmfScale, - psHeaderActive ); - if (noError != error) - { - error = handBack(error); - } + INT_PCM *pSamples[2] = { + samples + hSbrElement->elInfo.ChannelIndex[0] * samplesBufSize, + samples + hSbrElement->elInfo.ChannelIndex[1] * samplesBufSize}; + error = FDKsbrEnc_PSEnc_ParametricStereoProcessing( + hEnvEncoder->hParametricStereo, pSamples, samplesBufSize, + hSbrElement->hQmfAnalysis, sbrExtrEnv->rBuffer, + sbrExtrEnv->iBuffer, + samples + hSbrElement->elInfo.ChannelIndex[ch] * samplesBufSize, + &hEnvEncoder->qmfSynthesisPS, &qmfScale, psHeaderActive); h_envChan->qmfScale = (int)qmfScale; } - } /* if (hEnvEncoder->hParametricStereo) */ /* @@ -1140,80 +1132,146 @@ FDKsbrEnc_EnvEncodeFrame(HANDLE_SBR_ENCODER hEnvEncoder, Extract Envelope relevant things from QMF data */ - FDKsbrEnc_extractSbrEnvelope1( - &hSbrElement->sbrConfigData, - &hSbrElement->sbrHeaderData, - &hSbrElement->sbrBitstreamData, - h_envChan, - &hSbrElement->CmonData, - &eData[ch], - fData - ); + FDKsbrEnc_extractSbrEnvelope1(&hSbrElement->sbrConfigData, + &hSbrElement->sbrHeaderData, + &hSbrElement->sbrBitstreamData, h_envChan, + &hSbrElement->CmonData, &eData[ch], fData); } /* hEnvEncoder->sbrConfigData.nChannels */ - } + } /* - Process Envelope relevant things and calculate envelope data and write payload + Process Envelope relevant things and calculate envelope data and write + payload */ FDKsbrEnc_extractSbrEnvelope2( - &hSbrElement->sbrConfigData, - &hSbrElement->sbrHeaderData, - (hSbrElement->elInfo.fParametricStereo) ? hEnvEncoder->hParametricStereo : NULL, - &hSbrElement->sbrBitstreamData, - &hSbrElement->sbrChannel[0]->hEnvChannel, - &hSbrElement->sbrChannel[1]->hEnvChannel, - &hSbrElement->CmonData, - eData, - fData, - clearOutput - ); + &hSbrElement->sbrConfigData, &hSbrElement->sbrHeaderData, + (hSbrElement->elInfo.fParametricStereo) ? hEnvEncoder->hParametricStereo + : NULL, + &hSbrElement->sbrBitstreamData, &hSbrElement->sbrChannel[0]->hEnvChannel, + (hSbrElement->sbrConfigData.stereoMode != SBR_MONO) + ? &hSbrElement->sbrChannel[1]->hEnvChannel + : NULL, + &hSbrElement->CmonData, eData, fData, clearOutput); + + hSbrElement->sbrBitstreamData.rightBorderFIX = 0; /* format payload, calculate crc */ - FDKsbrEnc_AssembleSbrBitstream(&hSbrElement->CmonData, &crcInfo, crcReg, hSbrElement->sbrConfigData.sbrSyntaxFlags); + FDKsbrEnc_AssembleSbrBitstream(&hSbrElement->CmonData, &crcInfo, crcReg, + hSbrElement->sbrConfigData.sbrSyntaxFlags); /* save new payload, set to zero length if greater than MAX_PAYLOAD_SIZE */ - hSbrElement->payloadDelayLineSize[hEnvEncoder->nBitstrDelay] = FDKgetValidBits(&hSbrElement->CmonData.sbrBitbuf); + hSbrElement->payloadDelayLineSize[hEnvEncoder->nBitstrDelay] = + FDKgetValidBits(&hSbrElement->CmonData.sbrBitbuf); - if(hSbrElement->payloadDelayLineSize[hEnvEncoder->nBitstrDelay] > (MAX_PAYLOAD_SIZE<<3)) - hSbrElement->payloadDelayLineSize[hEnvEncoder->nBitstrDelay]=0; + if (hSbrElement->payloadDelayLineSize[hEnvEncoder->nBitstrDelay] > + (MAX_PAYLOAD_SIZE << 3)) + hSbrElement->payloadDelayLineSize[hEnvEncoder->nBitstrDelay] = 0; /* While filling the Delay lines, sbrData is NULL */ if (sbrData) { *sbrDataBits = hSbrElement->payloadDelayLineSize[0]; - FDKmemcpy(sbrData, hSbrElement->payloadDelayLine[0], (hSbrElement->payloadDelayLineSize[0]+7)>>3); - + FDKmemcpy(sbrData, hSbrElement->payloadDelayLine[0], + (hSbrElement->payloadDelayLineSize[0] + 7) >> 3); + } + /* delay header active flag */ + if (hSbrElement->sbrBitstreamData.HeaderActive == 1) { + hSbrElement->sbrBitstreamData.HeaderActiveDelay = + 1 + hEnvEncoder->nBitstrDelay; + } else { + if (hSbrElement->sbrBitstreamData.HeaderActiveDelay > 0) { + hSbrElement->sbrBitstreamData.HeaderActiveDelay--; + } } + return (0); +} -/*******************************/ +/***************************************************************************** - if (hEnvEncoder->fTimeDomainDownsampling) - { - int ch; - int nChannels = hSbrElement->sbrConfigData.nChannels; + functionname: FDKsbrEnc_Downsample + description: performs downsampling and delay compensation of the core path + returns: + input: + output: - for (ch=0; ch < nChannels; ch++) - { - INT nOutSamples; - - FDKaacEnc_Downsample(&hSbrElement->sbrChannel[ch]->downSampler, - samples + hSbrElement->elInfo.ChannelIndex[ch] + hEnvEncoder->bufferOffset, - hSbrElement->sbrConfigData.frameSize, - timeInStride, - samples + hSbrElement->elInfo.ChannelIndex[ch], - &nOutSamples, - hEnvEncoder->nChannels); +*****************************************************************************/ +INT FDKsbrEnc_Downsample( + HANDLE_SBR_ENCODER hSbrEncoder, + INT_PCM *samples, /*!< time samples, always deinterleaved */ + UINT samplesBufSize, /*!< time buffer size per channel */ + UINT numChannels, /*!< number of channels */ + UINT *sbrDataBits, /*!< Size of SBR payload */ + UCHAR *sbrData, /*!< SBR payload */ + int clearOutput /*!< Do not consider any input signal */ +) { + HANDLE_SBR_ELEMENT hSbrElement = NULL; + INT nOutSamples; + int el; + if (hSbrEncoder->downSampleFactor > 1) { + /* Do downsampling */ + + /* Loop over elements (LFE is handled later) */ + for (el = 0; el < hSbrEncoder->noElements; el++) { + hSbrElement = hSbrEncoder->sbrElement[el]; + if (hSbrEncoder->sbrElement[el] != NULL) { + if (hSbrEncoder->downsamplingMethod == SBRENC_DS_TIME) { + int ch; + int nChannels = hSbrElement->sbrConfigData.nChannels; + + for (ch = 0; ch < nChannels; ch++) { + FDKaacEnc_Downsample( + &hSbrElement->sbrChannel[ch]->downSampler, + samples + + hSbrElement->elInfo.ChannelIndex[ch] * samplesBufSize + + hSbrEncoder->bufferOffset / numChannels, + hSbrElement->sbrConfigData.frameSize, + samples + hSbrElement->elInfo.ChannelIndex[ch] * samplesBufSize, + &nOutSamples); + } + } + } } - } /* downsample */ + /* Handle LFE (if existing) */ + if (hSbrEncoder->lfeChIdx != -1) { /* lfe downsampler */ + FDKaacEnc_Downsample(&hSbrEncoder->lfeDownSampler, + samples + hSbrEncoder->lfeChIdx * samplesBufSize + + hSbrEncoder->bufferOffset / numChannels, + hSbrEncoder->frameSize, + samples + hSbrEncoder->lfeChIdx * samplesBufSize, + &nOutSamples); + } + } else { + /* No downsampling. Still, some buffer shifting for correct delay */ + int samples2Copy = hSbrEncoder->frameSize; + if (hSbrEncoder->bufferOffset / (int)numChannels < samples2Copy) { + for (int c = 0; c < (int)numChannels; c++) { + /* Do memmove while taking care of overlapping memory areas. (memcpy + does not necessarily take care) Distinguish between oeverlapping and + non overlapping version due to reasons of complexity. */ + FDKmemmove(samples + c * samplesBufSize, + samples + c * samplesBufSize + + hSbrEncoder->bufferOffset / numChannels, + samples2Copy * sizeof(INT_PCM)); + } + } else { + for (int c = 0; c < (int)numChannels; c++) { + /* Simple memcpy since the memory areas are not overlapping */ + FDKmemcpy(samples + c * samplesBufSize, + samples + c * samplesBufSize + + hSbrEncoder->bufferOffset / numChannels, + samples2Copy * sizeof(INT_PCM)); + } + } + } - return (0); + return 0; } /***************************************************************************** @@ -1226,27 +1284,17 @@ FDKsbrEnc_EnvEncodeFrame(HANDLE_SBR_ENCODER hEnvEncoder, *****************************************************************************/ -static INT -createEnvChannel (HANDLE_ENV_CHANNEL hEnv, - INT channel - ,UCHAR* dynamic_RAM - ) -{ - FDKmemclear(hEnv,sizeof (struct ENV_CHANNEL)); +static INT createEnvChannel(HANDLE_ENV_CHANNEL hEnv, INT channel, + UCHAR *dynamic_RAM) { + FDKmemclear(hEnv, sizeof(struct ENV_CHANNEL)); - if ( FDKsbrEnc_CreateTonCorrParamExtr(&hEnv->TonCorr, - channel) ) - { - return(1); + if (FDKsbrEnc_CreateTonCorrParamExtr(&hEnv->TonCorr, channel)) { + return (1); } - if ( FDKsbrEnc_CreateExtractSbrEnvelope (&hEnv->sbrExtractEnvelope, - channel - ,/*chan*/0 - ,dynamic_RAM - ) ) - { - return(1); + if (FDKsbrEnc_CreateExtractSbrEnvelope(&hEnv->sbrExtractEnvelope, channel, + /*chan*/ 0, dynamic_RAM)) { + return (1); } return 0; @@ -1261,21 +1309,16 @@ createEnvChannel (HANDLE_ENV_CHANNEL hEnv, output: *****************************************************************************/ -static INT -initEnvChannel ( HANDLE_SBR_CONFIG_DATA sbrConfigData, - HANDLE_SBR_HEADER_DATA sbrHeaderData, - HANDLE_ENV_CHANNEL hEnv, - sbrConfigurationPtr params, - ULONG statesInitFlag - ,INT chanInEl - ,UCHAR* dynamic_RAM - ) -{ - int frameShift, tran_off=0; +static INT initEnvChannel(HANDLE_SBR_CONFIG_DATA sbrConfigData, + HANDLE_SBR_HEADER_DATA sbrHeaderData, + HANDLE_ENV_CHANNEL hEnv, sbrConfigurationPtr params, + ULONG statesInitFlag, INT chanInEl, + UCHAR *dynamic_RAM) { + int frameShift, tran_off = 0; INT e; INT tran_fc; INT timeSlots, timeStep, startIndex; - INT noiseBands[2] = { 3, 3 }; + INT noiseBands[2] = {3, 3}; e = 1 << params->e; @@ -1283,11 +1326,12 @@ initEnvChannel ( HANDLE_SBR_CONFIG_DATA sbrConfigData, hEnv->encEnvData.freq_res_fixfix[0] = params->freq_res_fixfix[0]; hEnv->encEnvData.freq_res_fixfix[1] = params->freq_res_fixfix[1]; - hEnv->encEnvData.fResTransIsLow = params->fResTransIsLow; + hEnv->encEnvData.fResTransIsLow = params->fResTransIsLow; hEnv->fLevelProtect = 0; - hEnv->encEnvData.ldGrid = (sbrConfigData->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) ? 1 : 0; + hEnv->encEnvData.ldGrid = + (sbrConfigData->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) ? 1 : 0; hEnv->encEnvData.sbr_xpos_mode = (XPOS_MODE)params->sbr_xpos_mode; @@ -1298,19 +1342,16 @@ initEnvChannel ( HANDLE_SBR_CONFIG_DATA sbrConfigData, */ sbrConfigData->switchTransposers = TRUE; hEnv->encEnvData.sbr_xpos_mode = XPOS_MDCT; - } - else { + } else { sbrConfigData->switchTransposers = FALSE; } hEnv->encEnvData.sbr_xpos_ctrl = params->sbr_xpos_ctrl; - /* extended data */ - if(params->parametricCoding) { + if (params->parametricCoding) { hEnv->encEnvData.extended_data = 1; - } - else { + } else { hEnv->encEnvData.extended_data = 0; } @@ -1319,40 +1360,37 @@ initEnvChannel ( HANDLE_SBR_CONFIG_DATA sbrConfigData, startIndex = QMF_FILTER_PROTOTYPE_SIZE - sbrConfigData->noQmfBands; switch (params->sbrFrameSize) { - case 2304: - timeSlots = 18; - break; - case 2048: - case 1024: - case 512: - timeSlots = 16; - break; - case 1920: - case 960: - case 480: - timeSlots = 15; - break; - case 1152: - timeSlots = 9; - break; - default: - return (1); /* Illegal frame size */ + case 2304: + timeSlots = 18; + break; + case 2048: + case 1024: + case 512: + timeSlots = 16; + break; + case 1920: + case 960: + case 480: + timeSlots = 15; + break; + case 1152: + timeSlots = 9; + break; + default: + return (1); /* Illegal frame size */ } timeStep = sbrConfigData->noQmfSlots / timeSlots; - if ( FDKsbrEnc_InitTonCorrParamExtr(params->sbrFrameSize, - &hEnv->TonCorr, - sbrConfigData, - timeSlots, - params->sbr_xpos_ctrl, - params->ana_max_level, - sbrHeaderData->sbr_noise_bands, - params->noiseFloorOffset, - params->useSpeechConfig) ) - return(1); + if (FDKsbrEnc_InitTonCorrParamExtr( + params->sbrFrameSize, &hEnv->TonCorr, sbrConfigData, timeSlots, + params->sbr_xpos_ctrl, params->ana_max_level, + sbrHeaderData->sbr_noise_bands, params->noiseFloorOffset, + params->useSpeechConfig)) + return (1); - hEnv->encEnvData.noOfnoisebands = hEnv->TonCorr.sbrNoiseFloorEstimate.noNoiseBands; + hEnv->encEnvData.noOfnoisebands = + hEnv->TonCorr.sbrNoiseFloorEstimate.noNoiseBands; noiseBands[0] = hEnv->encEnvData.noOfnoisebands; noiseBands[1] = hEnv->encEnvData.noOfnoisebands; @@ -1362,106 +1400,90 @@ initEnvChannel ( HANDLE_SBR_CONFIG_DATA sbrConfigData, if (hEnv->encEnvData.sbr_invf_mode == INVF_SWITCHED) { hEnv->encEnvData.sbr_invf_mode = INVF_MID_LEVEL; hEnv->TonCorr.switchInverseFilt = TRUE; - } - else { + } else { hEnv->TonCorr.switchInverseFilt = FALSE; } - - tran_fc = params->tran_fc; + tran_fc = params->tran_fc; if (tran_fc == 0) { - tran_fc = fixMin (5000, FDKsbrEnc_getSbrStartFreqRAW (sbrHeaderData->sbr_start_frequency,params->codecSettings.sampleFreq)); + tran_fc = fixMin( + 5000, FDKsbrEnc_getSbrStartFreqRAW(sbrHeaderData->sbr_start_frequency, + params->codecSettings.sampleFreq)); } - tran_fc = (tran_fc*4*sbrConfigData->noQmfBands/sbrConfigData->sampleFreq + 1)>>1; + tran_fc = + (tran_fc * 4 * sbrConfigData->noQmfBands / sbrConfigData->sampleFreq + + 1) >> + 1; if (sbrConfigData->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { frameShift = LD_PRETRAN_OFF; - tran_off = LD_PRETRAN_OFF + FRAME_MIDDLE_SLOT_512LD*timeStep; - } else - { + tran_off = LD_PRETRAN_OFF + FRAME_MIDDLE_SLOT_512LD * timeStep; + } else { frameShift = 0; switch (timeSlots) { /* The factor of 2 is by definition. */ - case NUMBER_TIME_SLOTS_2048: tran_off = 8 + FRAME_MIDDLE_SLOT_2048 * timeStep; break; - case NUMBER_TIME_SLOTS_1920: tran_off = 7 + FRAME_MIDDLE_SLOT_1920 * timeStep; break; - default: return 1; + case NUMBER_TIME_SLOTS_2048: + tran_off = 8 + FRAME_MIDDLE_SLOT_2048 * timeStep; + break; + case NUMBER_TIME_SLOTS_1920: + tran_off = 7 + FRAME_MIDDLE_SLOT_1920 * timeStep; + break; + default: + return 1; } } - if ( FDKsbrEnc_InitExtractSbrEnvelope (&hEnv->sbrExtractEnvelope, - sbrConfigData->noQmfSlots, - sbrConfigData->noQmfBands, startIndex, - timeSlots, timeStep, tran_off, - statesInitFlag - ,chanInEl - ,dynamic_RAM - ,sbrConfigData->sbrSyntaxFlags - ) ) - return(1); - - if(FDKsbrEnc_InitSbrCodeEnvelope (&hEnv->sbrCodeEnvelope, - sbrConfigData->nSfb, - params->deltaTAcrossFrames, - params->dF_edge_1stEnv, - params->dF_edge_incr)) - return(1); - - if(FDKsbrEnc_InitSbrCodeEnvelope (&hEnv->sbrCodeNoiseFloor, - noiseBands, - params->deltaTAcrossFrames, - 0,0)) - return(1); + if (FDKsbrEnc_InitExtractSbrEnvelope( + &hEnv->sbrExtractEnvelope, sbrConfigData->noQmfSlots, + sbrConfigData->noQmfBands, startIndex, timeSlots, timeStep, tran_off, + statesInitFlag, chanInEl, dynamic_RAM, sbrConfigData->sbrSyntaxFlags)) + return (1); + + if (FDKsbrEnc_InitSbrCodeEnvelope(&hEnv->sbrCodeEnvelope, sbrConfigData->nSfb, + params->deltaTAcrossFrames, + params->dF_edge_1stEnv, + params->dF_edge_incr)) + return (1); + + if (FDKsbrEnc_InitSbrCodeEnvelope(&hEnv->sbrCodeNoiseFloor, noiseBands, + params->deltaTAcrossFrames, 0, 0)) + return (1); sbrConfigData->initAmpResFF = params->init_amp_res_FF; - if(FDKsbrEnc_InitSbrHuffmanTables (&hEnv->encEnvData, - &hEnv->sbrCodeEnvelope, - &hEnv->sbrCodeNoiseFloor, - sbrHeaderData->sbr_amp_res)) - return(1); - - FDKsbrEnc_initFrameInfoGenerator (&hEnv->SbrEnvFrame, - params->spread, - e, - params->stat, - timeSlots, - hEnv->encEnvData.freq_res_fixfix, - hEnv->encEnvData.fResTransIsLow, - hEnv->encEnvData.ldGrid - ); - - if(sbrConfigData->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) + if (FDKsbrEnc_InitSbrHuffmanTables(&hEnv->encEnvData, &hEnv->sbrCodeEnvelope, + &hEnv->sbrCodeNoiseFloor, + sbrHeaderData->sbr_amp_res)) + return (1); + + FDKsbrEnc_initFrameInfoGenerator( + &hEnv->SbrEnvFrame, params->spread, e, params->stat, timeSlots, + hEnv->encEnvData.freq_res_fixfix, hEnv->encEnvData.fResTransIsLow, + hEnv->encEnvData.ldGrid); + + if (sbrConfigData->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) + { - INT bandwidth_qmf_slot = (sbrConfigData->sampleFreq>>1) / (sbrConfigData->noQmfBands); - if(FDKsbrEnc_InitSbrFastTransientDetector( - &hEnv->sbrFastTransientDetector, - sbrConfigData->noQmfSlots, - bandwidth_qmf_slot, - sbrConfigData->noQmfBands, - sbrConfigData->freqBandTable[0][0] - )) - return(1); + INT bandwidth_qmf_slot = + (sbrConfigData->sampleFreq >> 1) / (sbrConfigData->noQmfBands); + if (FDKsbrEnc_InitSbrFastTransientDetector( + &hEnv->sbrFastTransientDetector, sbrConfigData->noQmfSlots, + bandwidth_qmf_slot, sbrConfigData->noQmfBands, + sbrConfigData->freqBandTable[0][0])) + return (1); } /* The transient detector has to be initialized also if the fast transient detector was active, because the values from the transient detector structure are used. */ - if(FDKsbrEnc_InitSbrTransientDetector (&hEnv->sbrTransientDetector, - sbrConfigData->sbrSyntaxFlags, - sbrConfigData->frameSize, - sbrConfigData->sampleFreq, - params, - tran_fc, - sbrConfigData->noQmfSlots, - sbrConfigData->noQmfBands, - hEnv->sbrExtractEnvelope.YBufferWriteOffset, - hEnv->sbrExtractEnvelope.YBufferSzShift, - frameShift, - tran_off - )) - return(1); - + if (FDKsbrEnc_InitSbrTransientDetector( + &hEnv->sbrTransientDetector, sbrConfigData->sbrSyntaxFlags, + sbrConfigData->frameSize, sbrConfigData->sampleFreq, params, tran_fc, + sbrConfigData->noQmfSlots, sbrConfigData->noQmfBands, + hEnv->sbrExtractEnvelope.YBufferWriteOffset, + hEnv->sbrExtractEnvelope.YBufferSzShift, frameShift, tran_off)) + return (1); sbrConfigData->xposCtrlSwitch = params->sbr_xpos_ctrl; @@ -1471,83 +1493,80 @@ initEnvChannel ( HANDLE_SBR_CONFIG_DATA sbrConfigData, return (0); } -INT sbrEncoder_Open( - HANDLE_SBR_ENCODER *phSbrEncoder, - INT nElements, - INT nChannels, - INT supportPS - ) -{ +INT sbrEncoder_Open(HANDLE_SBR_ENCODER *phSbrEncoder, INT nElements, + INT nChannels, INT supportPS) { INT i; INT errorStatus = 1; HANDLE_SBR_ENCODER hSbrEncoder = NULL; - if (phSbrEncoder==NULL - ) - { + if (phSbrEncoder == NULL) { goto bail; } hSbrEncoder = GetRam_SbrEncoder(); - if (hSbrEncoder==NULL) { + if (hSbrEncoder == NULL) { goto bail; } FDKmemclear(hSbrEncoder, sizeof(SBR_ENCODER)); - hSbrEncoder->pSBRdynamic_RAM = (UCHAR*)GetRam_SbrDynamic_RAM(); - hSbrEncoder->dynamicRam = hSbrEncoder->pSBRdynamic_RAM; + if (NULL == + (hSbrEncoder->pSBRdynamic_RAM = (UCHAR *)GetRam_SbrDynamic_RAM())) { + goto bail; + } + hSbrEncoder->dynamicRam = hSbrEncoder->pSBRdynamic_RAM; - for (i=0; i<nElements; i++) { + /* Create SBR elements */ + for (i = 0; i < nElements; i++) { hSbrEncoder->sbrElement[i] = GetRam_SbrElement(i); - if (hSbrEncoder->sbrElement[i]==NULL) { - goto bail; + if (hSbrEncoder->sbrElement[i] == NULL) { + goto bail; } FDKmemclear(hSbrEncoder->sbrElement[i], sizeof(SBR_ELEMENT)); - hSbrEncoder->sbrElement[i]->sbrConfigData.freqBandTable[LO] = GetRam_Sbr_freqBandTableLO(i); - hSbrEncoder->sbrElement[i]->sbrConfigData.freqBandTable[HI] = GetRam_Sbr_freqBandTableHI(i); - hSbrEncoder->sbrElement[i]->sbrConfigData.v_k_master = GetRam_Sbr_v_k_master(i); - if ( (hSbrEncoder->sbrElement[i]->sbrConfigData.freqBandTable[LO]==NULL) || - (hSbrEncoder->sbrElement[i]->sbrConfigData.freqBandTable[HI]==NULL) || - (hSbrEncoder->sbrElement[i]->sbrConfigData.v_k_master==NULL) ) - { - goto bail; + hSbrEncoder->sbrElement[i]->sbrConfigData.freqBandTable[LO] = + GetRam_Sbr_freqBandTableLO(i); + hSbrEncoder->sbrElement[i]->sbrConfigData.freqBandTable[HI] = + GetRam_Sbr_freqBandTableHI(i); + hSbrEncoder->sbrElement[i]->sbrConfigData.v_k_master = + GetRam_Sbr_v_k_master(i); + if ((hSbrEncoder->sbrElement[i]->sbrConfigData.freqBandTable[LO] == NULL) || + (hSbrEncoder->sbrElement[i]->sbrConfigData.freqBandTable[HI] == NULL) || + (hSbrEncoder->sbrElement[i]->sbrConfigData.v_k_master == NULL)) { + goto bail; } } - for (i=0; i<nChannels; i++) { + /* Create SBR channels */ + for (i = 0; i < nChannels; i++) { hSbrEncoder->pSbrChannel[i] = GetRam_SbrChannel(i); - if (hSbrEncoder->pSbrChannel[i]==NULL) { - goto bail; + if (hSbrEncoder->pSbrChannel[i] == NULL) { + goto bail; } - if ( createEnvChannel(&hSbrEncoder->pSbrChannel[i]->hEnvChannel, - i - ,hSbrEncoder->dynamicRam - ) ) - { - goto bail; + if (createEnvChannel(&hSbrEncoder->pSbrChannel[i]->hEnvChannel, i, + hSbrEncoder->dynamicRam)) { + goto bail; } - } - for (i=0; i<fixMax(nChannels,(supportPS)?2:0); i++) { + /* Create QMF States */ + for (i = 0; i < fixMax(nChannels, (supportPS) ? 2 : 0); i++) { hSbrEncoder->QmfAnalysis[i].FilterStates = GetRam_Sbr_QmfStatesAnalysis(i); - if (hSbrEncoder->QmfAnalysis[i].FilterStates==NULL) { - goto bail; + if (hSbrEncoder->QmfAnalysis[i].FilterStates == NULL) { + goto bail; } } + /* Create Parametric Stereo handle */ if (supportPS) { - if (PSEnc_Create(&hSbrEncoder->hParametricStereo)) - { + if (PSEnc_Create(&hSbrEncoder->hParametricStereo)) { goto bail; } hSbrEncoder->qmfSynthesisPS.FilterStates = GetRam_PsQmfStatesSynthesis(); - if (hSbrEncoder->qmfSynthesisPS.FilterStates==NULL) { + if (hSbrEncoder->qmfSynthesisPS.FilterStates == NULL) { goto bail; } - } /* supportPS */ + } /* supportPS */ *phSbrEncoder = hSbrEncoder; @@ -1560,56 +1579,74 @@ bail: return errorStatus; } -static -INT FDKsbrEnc_Reallocate( - HANDLE_SBR_ENCODER hSbrEncoder, - SBR_ELEMENT_INFO elInfo[(8)], - const INT noElements) -{ +static INT FDKsbrEnc_Reallocate(HANDLE_SBR_ENCODER hSbrEncoder, + SBR_ELEMENT_INFO elInfo[(8)], + const INT noElements) { INT totalCh = 0; INT totalQmf = 0; INT coreEl; - INT el=-1; + INT el = -1; hSbrEncoder->lfeChIdx = -1; /* default value, until lfe found */ - for (coreEl=0; coreEl<noElements; coreEl++) - { + for (coreEl = 0; coreEl < noElements; coreEl++) { /* SBR only handles SCE and CPE's */ if (elInfo[coreEl].elType == ID_SCE || elInfo[coreEl].elType == ID_CPE) { el++; } else { if (elInfo[coreEl].elType == ID_LFE) { - hSbrEncoder->lfeChIdx = elInfo[coreEl].ChannelIndex[0]; + hSbrEncoder->lfeChIdx = elInfo[coreEl].ChannelIndex[0]; } continue; } - SBR_ELEMENT_INFO *pelInfo = &elInfo[coreEl]; - HANDLE_SBR_ELEMENT hSbrElement = hSbrEncoder->sbrElement[el]; + SBR_ELEMENT_INFO *pelInfo = &elInfo[coreEl]; + HANDLE_SBR_ELEMENT hSbrElement = hSbrEncoder->sbrElement[el]; int ch; - for ( ch = 0; ch < pelInfo->nChannelsInEl; ch++ ) { + for (ch = 0; ch < pelInfo->nChannelsInEl; ch++) { hSbrElement->sbrChannel[ch] = hSbrEncoder->pSbrChannel[totalCh]; totalCh++; } /* analysis QMF */ - for ( ch = 0; ch < ((pelInfo->fParametricStereo)?2:pelInfo->nChannelsInEl); ch++ ) { + for (ch = 0; + ch < ((pelInfo->fParametricStereo) ? 2 : pelInfo->nChannelsInEl); + ch++) { hSbrElement->elInfo.ChannelIndex[ch] = pelInfo->ChannelIndex[ch]; hSbrElement->hQmfAnalysis[ch] = &hSbrEncoder->QmfAnalysis[totalQmf++]; } /* Copy Element info */ - hSbrElement->elInfo.elType = pelInfo->elType; - hSbrElement->elInfo.instanceTag = pelInfo->instanceTag; - hSbrElement->elInfo.nChannelsInEl = pelInfo->nChannelsInEl; + hSbrElement->elInfo.elType = pelInfo->elType; + hSbrElement->elInfo.instanceTag = pelInfo->instanceTag; + hSbrElement->elInfo.nChannelsInEl = pelInfo->nChannelsInEl; hSbrElement->elInfo.fParametricStereo = pelInfo->fParametricStereo; + hSbrElement->elInfo.fDualMono = pelInfo->fDualMono; } /* coreEl */ return 0; } +/***************************************************************************** + + functionname: FDKsbrEnc_bsBufInit + description: initializes bitstream buffer + returns: initialized bitstream buffer in env encoder + input: + output: hEnv + +*****************************************************************************/ +static INT FDKsbrEnc_bsBufInit(HANDLE_SBR_ELEMENT hSbrElement, + int nBitstrDelay) { + UCHAR *bitstreamBuffer; + + /* initialize the bitstream buffer */ + bitstreamBuffer = hSbrElement->payloadDelayLine[nBitstrDelay]; + FDKinitBitStream(&hSbrElement->CmonData.sbrBitbuf, bitstreamBuffer, + MAX_PAYLOAD_SIZE * sizeof(UCHAR), 0, BS_WRITER); + return (0); +} /***************************************************************************** @@ -1620,113 +1657,109 @@ INT FDKsbrEnc_Reallocate( output: hEnv *****************************************************************************/ -static -INT FDKsbrEnc_EnvInit ( - HANDLE_SBR_ELEMENT hSbrElement, - sbrConfigurationPtr params, - INT *coreBandWith, - AUDIO_OBJECT_TYPE aot, - int nBitstrDelay, - int nElement, - const int headerPeriod, - ULONG statesInitFlag, - int fTimeDomainDownsampling - ,UCHAR *dynamic_RAM - ) -{ - UCHAR *bitstreamBuffer; +static INT FDKsbrEnc_EnvInit(HANDLE_SBR_ELEMENT hSbrElement, + sbrConfigurationPtr params, INT *coreBandWith, + AUDIO_OBJECT_TYPE aot, int nElement, + const int headerPeriod, ULONG statesInitFlag, + const SBRENC_DS_TYPE downsamplingMethod, + UCHAR *dynamic_RAM) { int ch, i; - if ((params->codecSettings.nChannels < 1) || (params->codecSettings.nChannels > MAX_NUM_CHANNELS)){ - return(1); + if ((params->codecSettings.nChannels < 1) || + (params->codecSettings.nChannels > MAX_NUM_CHANNELS)) { + return (1); } - /* initialize the encoder handle and structs*/ - bitstreamBuffer = hSbrElement->payloadDelayLine[nBitstrDelay]; - /* init and set syntax flags */ hSbrElement->sbrConfigData.sbrSyntaxFlags = 0; switch (aot) { - case AOT_ER_AAC_ELD: - hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_LOW_DELAY; - break; - default: - break; + case AOT_ER_AAC_ELD: + hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_LOW_DELAY; + break; + default: + break; } if (params->crcSbr) { hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_CRC; } - hSbrElement->sbrConfigData.noQmfBands = QMF_CHANNELS>>(2-params->downSampleFactor); - switch (hSbrElement->sbrConfigData.noQmfBands) - { - case 64: hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize>>6; - break; - case 32: hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize>>5; - break; - default: hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize>>6; - return(2); + hSbrElement->sbrConfigData.noQmfBands = 64 >> (2 - params->downSampleFactor); + switch (hSbrElement->sbrConfigData.noQmfBands) { + case 64: + hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize >> 6; + break; + case 32: + hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize >> 5; + break; + default: + hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize >> 6; + return (2); } - FDKinitBitStream(&hSbrElement->CmonData.sbrBitbuf, bitstreamBuffer, MAX_PAYLOAD_SIZE*sizeof(UCHAR), 0, BS_WRITER); - /* now initialize sbrConfigData, sbrHeaderData and sbrBitstreamData, */ hSbrElement->sbrConfigData.nChannels = params->codecSettings.nChannels; - if(params->codecSettings.nChannels == 2) - hSbrElement->sbrConfigData.stereoMode = params->stereoMode; - else - hSbrElement->sbrConfigData.stereoMode = SBR_MONO; + if (params->codecSettings.nChannels == 2) { + if ((hSbrElement->elInfo.elType == ID_CPE) && + ((hSbrElement->elInfo.fDualMono == 1))) { + hSbrElement->sbrConfigData.stereoMode = SBR_LEFT_RIGHT; + } else { + hSbrElement->sbrConfigData.stereoMode = params->stereoMode; + } + } else { + hSbrElement->sbrConfigData.stereoMode = SBR_MONO; + } - hSbrElement->sbrConfigData.frameSize = params->sbrFrameSize; + hSbrElement->sbrConfigData.frameSize = params->sbrFrameSize; - hSbrElement->sbrConfigData.sampleFreq = params->downSampleFactor * params->codecSettings.sampleFreq; + hSbrElement->sbrConfigData.sampleFreq = + params->downSampleFactor * params->codecSettings.sampleFreq; hSbrElement->sbrBitstreamData.CountSendHeaderData = 0; - if (params->SendHeaderDataTime > 0 ) { - - if (headerPeriod==-1) { - - hSbrElement->sbrBitstreamData.NrSendHeaderData = (INT)(params->SendHeaderDataTime * hSbrElement->sbrConfigData.sampleFreq - / (1000 * hSbrElement->sbrConfigData.frameSize)); - hSbrElement->sbrBitstreamData.NrSendHeaderData = fixMax(hSbrElement->sbrBitstreamData.NrSendHeaderData,1); - } - else { + if (params->SendHeaderDataTime > 0) { + if (headerPeriod == -1) { + hSbrElement->sbrBitstreamData.NrSendHeaderData = (INT)( + params->SendHeaderDataTime * hSbrElement->sbrConfigData.sampleFreq / + (1000 * hSbrElement->sbrConfigData.frameSize)); + hSbrElement->sbrBitstreamData.NrSendHeaderData = + fixMax(hSbrElement->sbrBitstreamData.NrSendHeaderData, 1); + } else { /* assure header period at least once per second */ - hSbrElement->sbrBitstreamData.NrSendHeaderData = fixMin(fixMax(headerPeriod,1),(hSbrElement->sbrConfigData.sampleFreq/hSbrElement->sbrConfigData.frameSize)); + hSbrElement->sbrBitstreamData.NrSendHeaderData = fixMin( + fixMax(headerPeriod, 1), (hSbrElement->sbrConfigData.sampleFreq / + hSbrElement->sbrConfigData.frameSize)); } - } - else { - hSbrElement->sbrBitstreamData.NrSendHeaderData = 0; + } else { + hSbrElement->sbrBitstreamData.NrSendHeaderData = 0; } hSbrElement->sbrHeaderData.sbr_data_extra = params->sbr_data_extra; hSbrElement->sbrBitstreamData.HeaderActive = 0; + hSbrElement->sbrBitstreamData.rightBorderFIX = 0; hSbrElement->sbrHeaderData.sbr_start_frequency = params->startFreq; - hSbrElement->sbrHeaderData.sbr_stop_frequency = params->stopFreq; + hSbrElement->sbrHeaderData.sbr_stop_frequency = params->stopFreq; hSbrElement->sbrHeaderData.sbr_xover_band = 0; hSbrElement->sbrHeaderData.sbr_lc_stereo_mode = 0; /* data_extra */ - if (params->sbr_xpos_ctrl!= SBR_XPOS_CTRL_DEFAULT) - hSbrElement->sbrHeaderData.sbr_data_extra = 1; + if (params->sbr_xpos_ctrl != SBR_XPOS_CTRL_DEFAULT) + hSbrElement->sbrHeaderData.sbr_data_extra = 1; hSbrElement->sbrHeaderData.sbr_amp_res = (AMP_RES)params->amp_res; /* header_extra_1 */ - hSbrElement->sbrHeaderData.freqScale = params->freqScale; + hSbrElement->sbrHeaderData.freqScale = params->freqScale; hSbrElement->sbrHeaderData.alterScale = params->alterScale; hSbrElement->sbrHeaderData.sbr_noise_bands = params->sbr_noise_bands; hSbrElement->sbrHeaderData.header_extra_1 = 0; if ((params->freqScale != SBR_FREQ_SCALE_DEFAULT) || (params->alterScale != SBR_ALTER_SCALE_DEFAULT) || - (params->sbr_noise_bands != SBR_NOISE_BANDS_DEFAULT)) - { - hSbrElement->sbrHeaderData.header_extra_1 = 1; + (params->sbr_noise_bands != SBR_NOISE_BANDS_DEFAULT)) { + hSbrElement->sbrHeaderData.header_extra_1 = 1; } /* header_extra_2 */ @@ -1734,95 +1767,92 @@ INT FDKsbrEnc_EnvInit ( hSbrElement->sbrHeaderData.sbr_limiter_gains = params->sbr_limiter_gains; if ((hSbrElement->sbrConfigData.sampleFreq > 48000) && - (hSbrElement->sbrHeaderData.sbr_start_frequency >= 9)) - { + (hSbrElement->sbrHeaderData.sbr_start_frequency >= 9)) { hSbrElement->sbrHeaderData.sbr_limiter_gains = SBR_LIMITER_GAINS_INFINITE; } hSbrElement->sbrHeaderData.sbr_interpol_freq = params->sbr_interpol_freq; - hSbrElement->sbrHeaderData.sbr_smoothing_length = params->sbr_smoothing_length; + hSbrElement->sbrHeaderData.sbr_smoothing_length = + params->sbr_smoothing_length; hSbrElement->sbrHeaderData.header_extra_2 = 0; if ((params->sbr_limiter_bands != SBR_LIMITER_BANDS_DEFAULT) || (params->sbr_limiter_gains != SBR_LIMITER_GAINS_DEFAULT) || (params->sbr_interpol_freq != SBR_INTERPOL_FREQ_DEFAULT) || - (params->sbr_smoothing_length != SBR_SMOOTHING_LENGTH_DEFAULT)) - { - hSbrElement->sbrHeaderData.header_extra_2 = 1; + (params->sbr_smoothing_length != SBR_SMOOTHING_LENGTH_DEFAULT)) { + hSbrElement->sbrHeaderData.header_extra_2 = 1; } - /* other switches */ - hSbrElement->sbrConfigData.useWaveCoding = params->useWaveCoding; - hSbrElement->sbrConfigData.useParametricCoding = params->parametricCoding; - hSbrElement->sbrConfigData.thresholdAmpResFF_m = params->threshold_AmpRes_FF_m; - hSbrElement->sbrConfigData.thresholdAmpResFF_e = params->threshold_AmpRes_FF_e; + /* other switches */ + hSbrElement->sbrConfigData.useWaveCoding = params->useWaveCoding; + hSbrElement->sbrConfigData.useParametricCoding = params->parametricCoding; + hSbrElement->sbrConfigData.thresholdAmpResFF_m = + params->threshold_AmpRes_FF_m; + hSbrElement->sbrConfigData.thresholdAmpResFF_e = + params->threshold_AmpRes_FF_e; /* init freq band table */ - if(updateFreqBandTable(&hSbrElement->sbrConfigData, - &hSbrElement->sbrHeaderData, - params->downSampleFactor - )) - { - return(1); + if (updateFreqBandTable(&hSbrElement->sbrConfigData, + &hSbrElement->sbrHeaderData, + params->downSampleFactor)) { + return (1); } /* now create envelope ext and QMF for each available channel */ - for ( ch = 0; ch < hSbrElement->sbrConfigData.nChannels; ch++ ) { - - if ( initEnvChannel(&hSbrElement->sbrConfigData, - &hSbrElement->sbrHeaderData, - &hSbrElement->sbrChannel[ch]->hEnvChannel, - params, - statesInitFlag - ,ch - ,dynamic_RAM - ) ) - { - return(1); - } - + for (ch = 0; ch < hSbrElement->sbrConfigData.nChannels; ch++) { + if (initEnvChannel(&hSbrElement->sbrConfigData, &hSbrElement->sbrHeaderData, + &hSbrElement->sbrChannel[ch]->hEnvChannel, params, + statesInitFlag, ch, dynamic_RAM)) { + return (1); + } } /* nChannels */ /* reset and intialize analysis qmf */ - for ( ch = 0; ch < ((hSbrElement->elInfo.fParametricStereo)?2:hSbrElement->sbrConfigData.nChannels); ch++ ) - { + for (ch = 0; ch < ((hSbrElement->elInfo.fParametricStereo) + ? 2 + : hSbrElement->sbrConfigData.nChannels); + ch++) { int err; - UINT qmfFlags = (hSbrElement->sbrConfigData.sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) ? QMF_FLAG_CLDFB : 0; + UINT qmfFlags = + (hSbrElement->sbrConfigData.sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) + ? QMF_FLAG_CLDFB + : 0; if (statesInitFlag) qmfFlags &= ~QMF_FLAG_KEEP_STATES; else - qmfFlags |= QMF_FLAG_KEEP_STATES; - - err = qmfInitAnalysisFilterBank( hSbrElement->hQmfAnalysis[ch], - (FIXP_QAS*)hSbrElement->hQmfAnalysis[ch]->FilterStates, - hSbrElement->sbrConfigData.noQmfSlots, - hSbrElement->sbrConfigData.noQmfBands, - hSbrElement->sbrConfigData.noQmfBands, - hSbrElement->sbrConfigData.noQmfBands, - qmfFlags ); - if (0!=err) { + qmfFlags |= QMF_FLAG_KEEP_STATES; + + err = qmfInitAnalysisFilterBank( + hSbrElement->hQmfAnalysis[ch], + (FIXP_QAS *)hSbrElement->hQmfAnalysis[ch]->FilterStates, + hSbrElement->sbrConfigData.noQmfSlots, + hSbrElement->sbrConfigData.noQmfBands, + hSbrElement->sbrConfigData.noQmfBands, + hSbrElement->sbrConfigData.noQmfBands, qmfFlags); + if (0 != err) { return err; } } /* */ hSbrElement->CmonData.xOverFreq = hSbrElement->sbrConfigData.xOverFreq; - hSbrElement->CmonData.dynBwEnabled = (params->dynBwSupported && params->dynBwEnabled); - hSbrElement->CmonData.dynXOverFreqEnc = FDKsbrEnc_SbrGetXOverFreq( hSbrElement, hSbrElement->CmonData.xOverFreq); - for ( i = 0; i < 5; i++ ) - hSbrElement->dynXOverFreqDelay[i] = hSbrElement->CmonData.dynXOverFreqEnc; - hSbrElement->CmonData.sbrNumChannels = hSbrElement->sbrConfigData.nChannels; + hSbrElement->CmonData.dynBwEnabled = + (params->dynBwSupported && params->dynBwEnabled); + hSbrElement->CmonData.dynXOverFreqEnc = + FDKsbrEnc_SbrGetXOverFreq(hSbrElement, hSbrElement->CmonData.xOverFreq); + for (i = 0; i < 5; i++) + hSbrElement->dynXOverFreqDelay[i] = hSbrElement->CmonData.dynXOverFreqEnc; + hSbrElement->CmonData.sbrNumChannels = hSbrElement->sbrConfigData.nChannels; hSbrElement->sbrConfigData.dynXOverFreq = hSbrElement->CmonData.xOverFreq; /* Update Bandwith to be passed to the core encoder */ *coreBandWith = hSbrElement->CmonData.xOverFreq; - return(0); - } + return (0); +} -INT sbrEncoder_GetInBufferSize(int noChannels) -{ +INT sbrEncoder_GetInBufferSize(int noChannels) { INT temp; temp = (2048); @@ -1835,53 +1865,42 @@ INT sbrEncoder_GetInBufferSize(int noChannels) /* * Encode Dummy SBR payload frames to fill the delay lines. */ -static -INT FDKsbrEnc_DelayCompensation ( - HANDLE_SBR_ENCODER hEnvEnc, - INT_PCM *timeBuffer - ) -{ - int n, el; - - for (n=hEnvEnc->nBitstrDelay; n>0; n--) - { - for (el=0; el<hEnvEnc->noElements; el++) - { - if (FDKsbrEnc_EnvEncodeFrame( - hEnvEnc, - el, - timeBuffer + hEnvEnc->downsampledOffset, - hEnvEnc->sbrElement[el]->sbrConfigData.nChannels, - NULL, - NULL, - 1 - )) - return -1; - } - sbrEncoder_UpdateBuffers(hEnvEnc, timeBuffer); +static INT FDKsbrEnc_DelayCompensation(HANDLE_SBR_ENCODER hEnvEnc, + INT_PCM *timeBuffer, + UINT timeBufferBufSize) { + int n, el; + + for (n = hEnvEnc->nBitstrDelay; n > 0; n--) { + for (el = 0; el < hEnvEnc->noElements; el++) { + if (FDKsbrEnc_EnvEncodeFrame( + hEnvEnc, el, + timeBuffer + hEnvEnc->downsampledOffset / hEnvEnc->nChannels, + timeBufferBufSize, NULL, NULL, 1)) + return -1; } - return 0; + sbrEncoder_UpdateBuffers(hEnvEnc, timeBuffer, timeBufferBufSize); + } + return 0; } -UINT sbrEncoder_LimitBitRate(UINT bitRate, UINT numChannels, UINT coreSampleRate, AUDIO_OBJECT_TYPE aot) -{ - UINT newBitRate; +UINT sbrEncoder_LimitBitRate(UINT bitRate, UINT numChannels, + UINT coreSampleRate, AUDIO_OBJECT_TYPE aot) { + UINT newBitRate = bitRate; INT index; FDK_ASSERT(numChannels > 0 && numChannels <= 2); if (aot == AOT_PS) { - if (numChannels == 2) { + if (numChannels == 1) { index = getPsTuningTableIndex(bitRate, &newBitRate); if (index == INVALID_TABLE_IDX) { bitRate = newBitRate; } - /* Set numChannels to 1 because for PS we need a SBR SCE (mono) element. */ - numChannels = 1; } else { return 0; } } - index = getSbrTuningTableIndex(bitRate, numChannels, coreSampleRate, aot, &newBitRate); + index = getSbrTuningTableIndex(bitRate, numChannels, coreSampleRate, aot, + &newBitRate); if (index != INVALID_TABLE_IDX) { newBitRate = bitRate; } @@ -1889,523 +1908,640 @@ UINT sbrEncoder_LimitBitRate(UINT bitRate, UINT numChannels, UINT coreSampleRate return newBitRate; } -UINT sbrEncoder_IsSingleRatePossible(AUDIO_OBJECT_TYPE aot) -{ - UINT isPossible=(AOT_PS==aot)?0:1; +UINT sbrEncoder_IsSingleRatePossible(AUDIO_OBJECT_TYPE aot) { + UINT isPossible = (AOT_PS == aot) ? 0 : 1; return isPossible; } -INT sbrEncoder_Init( - HANDLE_SBR_ENCODER hSbrEncoder, - SBR_ELEMENT_INFO elInfo[(8)], - int noElements, - INT_PCM *inputBuffer, - INT *coreBandwidth, - INT *inputBufferOffset, - INT *numChannels, - INT *coreSampleRate, - UINT *downSampleFactor, - INT *frameLength, - AUDIO_OBJECT_TYPE aot, - int *delay, - int transformFactor, - const int headerPeriod, - ULONG statesInitFlag - ) -{ - HANDLE_ERROR_INFO errorInfo = noError; - sbrConfiguration sbrConfig[(8)]; - INT error = 0; - INT lowestBandwidth; - /* Save input parameters */ - INT inputSampleRate = *coreSampleRate; - int coreFrameLength = *frameLength; - int inputBandWidth = *coreBandwidth; - int inputChannels = *numChannels; - - int downsampledOffset = 0; - int sbrOffset = 0; - int downsamplerDelay = 0; - int timeDomainDownsample = 0; - int nBitstrDelay = 0; - int highestSbrStartFreq, highestSbrStopFreq; - int lowDelay = 0; - int usePs = 0; - - /* check whether SBR setting is available for the current encoder configuration (bitrate, samplerate) */ - if (!sbrEncoder_IsSingleRatePossible(aot)) { - *downSampleFactor = 2; +/*****************************************************************************/ +/* */ +/*functionname: sbrEncoder_Init_delay */ +/*description: Determine Delay balancing and new encoder delay */ +/* */ +/*returns: - error status */ +/*input: - frame length of the core (i.e. e.g. AAC) */ +/* - number of channels */ +/* - downsample factor (1 for downsampled, 2 for dual-rate SBR) */ +/* - low delay presence */ +/* - ps presence */ +/* - downsampling method: QMF-, time domain or no downsampling */ +/* - various delay values (see DELAY_PARAM struct description) */ +/* */ +/*Example: Delay balancing for a HE-AACv1 encoder (time-domain downsampling) */ +/*========================================================================== */ +/* */ +/* +--------+ +--------+ +--------+ +--------+ +--------+ */ +/* |core | |ds 2:1 | |AAC | |QMF | |QMF | */ +/* +-+path +------------+ +-+core +-+analysis+-+overlap +-+ */ +/* | |offset | | | | | |32 bands| | | | */ +/* | +--------+ +--------+ +--------+ +--------+ +--------+ | */ +/* | core path +-------++ */ +/* | |QMF | */ +/*->+ +synth. +-> */ +/* | |64 bands| */ +/* | +-------++ */ +/* | +--------+ +--------+ +--------+ +--------+ | */ +/* | |SBR path| |QMF | |subband | |bs delay| | */ +/* +-+offset +-+analysis+-+sample +-+(full +-----------------------+ */ +/* | | |64 bands| |buffer | | frames)| */ +/* +--------+ +--------+ +--------+ +--------+ */ +/* SBR path */ +/* */ +/*****************************************************************************/ +static INT sbrEncoder_Init_delay( + const int coreFrameLength, /* input */ + const int numChannels, /* input */ + const int downSampleFactor, /* input */ + const int lowDelay, /* input */ + const int usePs, /* input */ + const int is212, /* input */ + const SBRENC_DS_TYPE downsamplingMethod, /* input */ + DELAY_PARAM *hDelayParam /* input/output */ +) { + int delayCorePath = 0; /* delay in core path */ + int delaySbrPath = 0; /* delay difference in QMF aka SBR path */ + int delayInput2Core = 0; /* delay from the input to the core */ + int delaySbrDec = 0; /* delay of the decoder's SBR module */ + + int delayCore = hDelayParam->delay; /* delay of the core */ + + /* Added delay by the SBR delay initialization */ + int corePathOffset = 0; /* core path */ + int sbrPathOffset = 0; /* sbr path */ + int bitstreamDelay = 0; /* sbr path, framewise */ + + int flCore = coreFrameLength; /* core frame length */ + + int returnValue = 0; /* return value - 0 means: no error */ + + /* 1) Calculate actual delay for core and SBR path */ + if (is212) { + delayCorePath = DELAY_COREPATH_ELDv2SBR(flCore, downSampleFactor); + delaySbrPath = DELAY_ELDv2SBR(flCore, downSampleFactor); + delaySbrDec = ((flCore) / 2) * (downSampleFactor); + } else if (lowDelay) { + delayCorePath = DELAY_COREPATH_ELDSBR(flCore, downSampleFactor); + delaySbrPath = DELAY_ELDSBR(flCore, downSampleFactor); + delaySbrDec = DELAY_QMF_POSTPROC(downSampleFactor); + } else if (usePs) { + delayCorePath = DELAY_COREPATH_PS(flCore, downSampleFactor); + delaySbrPath = DELAY_PS(flCore, downSampleFactor); + delaySbrDec = DELAY_COREPATH_SBR(flCore, downSampleFactor); + } else { + delayCorePath = DELAY_COREPATH_SBR(flCore, downSampleFactor); + delaySbrPath = DELAY_SBR(flCore, downSampleFactor); + delaySbrDec = DELAY_COREPATH_SBR(flCore, downSampleFactor); + } + delayCorePath += delayCore * downSampleFactor; + delayCorePath += + (downsamplingMethod == SBRENC_DS_TIME) ? hDelayParam->dsDelay : 0; + + /* 2) Manage coupling of paths */ + if (downsamplingMethod == SBRENC_DS_QMF && delayCorePath > delaySbrPath) { + /* In case of QMF downsampling, both paths are coupled, i.e. the SBR path + offset would be added to both the SBR path and to the core path + as well, thus making it impossible to achieve delay balancing. + To overcome that problem, a framewise delay is added to the SBR path + first, until the overall delay of the core path is shorter than + the delay of the SBR path. When this is achieved, the missing delay + difference can be added as downsampled offset to the core path. + */ + while (delayCorePath > delaySbrPath) { + /* Add one frame delay to SBR path */ + delaySbrPath += flCore * downSampleFactor; + bitstreamDelay += 1; } + } + /* 3) Calculate necessary additional delay to balance the paths */ + if (delayCorePath > delaySbrPath) { + /* Delay QMF input */ + while (delayCorePath > delaySbrPath + (int)flCore * (int)downSampleFactor) { + /* Do bitstream frame-wise delay balancing if there are + more than SBR framelength samples delay difference */ + delaySbrPath += flCore * downSampleFactor; + bitstreamDelay += 1; + } + /* Multiply input offset by input channels */ + corePathOffset = 0; + sbrPathOffset = (delayCorePath - delaySbrPath) * numChannels; + } else { + /* Delay AAC data */ + /* Multiply downsampled offset by AAC core channels. Divide by 2 because of + half samplerate of downsampled data. */ + corePathOffset = ((delaySbrPath - delayCorePath) * numChannels) >> + (downSampleFactor - 1); + sbrPathOffset = 0; + } + /* 4) Calculate delay from input to core */ + if (usePs) { + delayInput2Core = + (DELAY_QMF_ANA(downSampleFactor) + DELAY_QMF_DS + DELAY_HYB_SYN) + + (downSampleFactor * corePathOffset) + 1; + } else if (downsamplingMethod == SBRENC_DS_TIME) { + delayInput2Core = corePathOffset + hDelayParam->dsDelay; + } else { + delayInput2Core = corePathOffset; + } - if ( aot==AOT_PS ) { - usePs = 1; - } - if ( aot==AOT_ER_AAC_ELD ) { - lowDelay = 1; - } - else if ( aot==AOT_ER_AAC_LD ) { - error = 1; - goto bail; - } + /* 6) Set output parameters */ + hDelayParam->delay = FDKmax(delayCorePath, delaySbrPath); /* overall delay */ + hDelayParam->sbrDecDelay = delaySbrDec; /* SBR decoder delay */ + hDelayParam->delayInput2Core = delayInput2Core; /* delay input - core */ + hDelayParam->bitstrDelay = bitstreamDelay; /* bitstream delay, in frames */ + hDelayParam->corePathOffset = corePathOffset; /* offset added to core path */ + hDelayParam->sbrPathOffset = sbrPathOffset; /* offset added to SBR path */ - /* Parametric Stereo */ - if ( usePs ) { - if ( *numChannels == 2 && noElements == 1) { - /* Override Element type in case of Parametric stereo */ - elInfo[0].elType = ID_SCE; - elInfo[0].fParametricStereo = 1; - elInfo[0].nChannelsInEl = 1; - /* core encoder gets downmixed mono signal */ - *numChannels = 1; - } else { - error = 1; - goto bail; - } - } /* usePs */ + return returnValue; +} - /* set the core's sample rate */ - switch (*downSampleFactor) { +/***************************************************************************** + + functionname: sbrEncoder_Init + description: initializes the SBR encoder + returns: error status + +*****************************************************************************/ +INT sbrEncoder_Init(HANDLE_SBR_ENCODER hSbrEncoder, + SBR_ELEMENT_INFO elInfo[(8)], int noElements, + INT_PCM *inputBuffer, UINT inputBufferBufSize, + INT *coreBandwidth, INT *inputBufferOffset, + INT *numChannels, const UINT syntaxFlags, + INT *coreSampleRate, UINT *downSampleFactor, + INT *frameLength, AUDIO_OBJECT_TYPE aot, int *delay, + int transformFactor, const int headerPeriod, + ULONG statesInitFlag) { + HANDLE_ERROR_INFO errorInfo = noError; + sbrConfiguration sbrConfig[(8)]; + INT error = 0; + INT lowestBandwidth; + /* Save input parameters */ + INT inputSampleRate = *coreSampleRate; + int coreFrameLength = *frameLength; + int inputBandWidth = *coreBandwidth; + int inputChannels = *numChannels; + + SBRENC_DS_TYPE downsamplingMethod = SBRENC_DS_NONE; + int highestSbrStartFreq, highestSbrStopFreq; + int lowDelay = 0; + int usePs = 0; + int is212 = 0; + + DELAY_PARAM delayParam; + + /* check whether SBR setting is available for the current encoder + * configuration (bitrate, samplerate) */ + if (!sbrEncoder_IsSingleRatePossible(aot)) { + *downSampleFactor = 2; + } + + if (aot == AOT_PS) { + usePs = 1; + } + if (aot == AOT_ER_AAC_ELD) { + lowDelay = 1; + } else if (aot == AOT_ER_AAC_LD) { + error = 1; + goto bail; + } + + /* Parametric Stereo */ + if (usePs) { + if (*numChannels == 2 && noElements == 1) { + /* Override Element type in case of Parametric stereo */ + elInfo[0].elType = ID_SCE; + elInfo[0].fParametricStereo = 1; + elInfo[0].nChannelsInEl = 1; + /* core encoder gets downmixed mono signal */ + *numChannels = 1; + } else { + error = 1; + goto bail; + } + } /* usePs */ + + /* set the core's sample rate */ + switch (*downSampleFactor) { case 1: *coreSampleRate = inputSampleRate; + downsamplingMethod = SBRENC_DS_NONE; break; case 2: - *coreSampleRate = inputSampleRate>>1; + *coreSampleRate = inputSampleRate >> 1; + downsamplingMethod = usePs ? SBRENC_DS_QMF : SBRENC_DS_TIME; break; default: - *coreSampleRate = inputSampleRate>>1; + *coreSampleRate = inputSampleRate >> 1; return 0; /* return error */ - } + } - /* check whether SBR setting is available for the current encoder configuration (bitrate, coreSampleRate) */ - { - int delayDiff = 0; - int el, coreEl; - - /* Check if every element config is feasible */ - for (coreEl=0; coreEl<noElements; coreEl++) - { - /* SBR only handles SCE and CPE's */ - if (elInfo[coreEl].elType != ID_SCE && elInfo[coreEl].elType != ID_CPE) { - continue; - } - /* check if desired configuration is available */ - if ( !FDKsbrEnc_IsSbrSettingAvail (elInfo[coreEl].bitRate, 0, elInfo[coreEl].nChannelsInEl, inputSampleRate, *coreSampleRate, aot) ) - { - error = 1; - goto bail; - } - } + /* check whether SBR setting is available for the current encoder + * configuration (bitrate, coreSampleRate) */ + { + int el, coreEl; - /* Determine Delay balancing and new encoder delay */ - if (lowDelay) { - { - delayDiff = (*delay * *downSampleFactor) + DELAY_ELD2SBR(coreFrameLength,*downSampleFactor); - *delay = DELAY_ELDSBR(coreFrameLength,*downSampleFactor); - } - } - else if (usePs) { - delayDiff = (*delay * *downSampleFactor) + DELAY_AAC2PS(coreFrameLength,*downSampleFactor); - *delay = DELAY_PS(coreFrameLength,*downSampleFactor); - } - else { - delayDiff = DELAY_AAC2SBR(coreFrameLength,*downSampleFactor); - delayDiff += (*delay * *downSampleFactor); - *delay = DELAY_SBR(coreFrameLength,*downSampleFactor); + /* Check if every element config is feasible */ + for (coreEl = 0; coreEl < noElements; coreEl++) { + /* SBR only handles SCE and CPE's */ + if (elInfo[coreEl].elType != ID_SCE && elInfo[coreEl].elType != ID_CPE) { + continue; } - - if (!usePs) { - timeDomainDownsample = *downSampleFactor-1; /* activate time domain downsampler when downSampleFactor is != 1 */ + /* check if desired configuration is available */ + if (!FDKsbrEnc_IsSbrSettingAvail(elInfo[coreEl].bitRate, 0, + elInfo[coreEl].nChannelsInEl, + inputSampleRate, *coreSampleRate, aot)) { + error = 1; + goto bail; } + } - - /* Take care about downsampled data bound to the SBR path */ - if (!timeDomainDownsample && delayDiff > 0) { - /* - * We must tweak the balancing into a situation where the downsampled path - * is the one to be delayed, because delaying the QMF domain input, also delays - * the downsampled audio, counteracting to the purpose of delay balancing. - */ - while ( delayDiff > 0 ) - { - /* Encoder delay increases */ - { - *delay += coreFrameLength * *downSampleFactor; - /* Add one frame delay to SBR path */ - delayDiff -= coreFrameLength * *downSampleFactor; - } - nBitstrDelay += 1; - } - } else - { - *delay += fixp_abs(delayDiff); + hSbrEncoder->nChannels = *numChannels; + hSbrEncoder->frameSize = coreFrameLength * *downSampleFactor; + hSbrEncoder->downsamplingMethod = downsamplingMethod; + hSbrEncoder->downSampleFactor = *downSampleFactor; + hSbrEncoder->estimateBitrate = 0; + hSbrEncoder->inputDataDelay = 0; + is212 = ((aot == AOT_ER_AAC_ELD) && (syntaxFlags & AC_LD_MPS)) ? 1 : 0; + + /* Open SBR elements */ + el = -1; + highestSbrStartFreq = highestSbrStopFreq = 0; + lowestBandwidth = 99999; + + /* Loop through each core encoder element and get a matching SBR element + * config */ + for (coreEl = 0; coreEl < noElements; coreEl++) { + /* SBR only handles SCE and CPE's */ + if (elInfo[coreEl].elType == ID_SCE || elInfo[coreEl].elType == ID_CPE) { + el++; + } else { + continue; } - if (delayDiff < 0) { - /* Delay AAC data */ - delayDiff = -delayDiff; - /* Multiply downsampled offset by AAC core channels. Divide by 2 because of half samplerate of downsampled data. */ - FDK_ASSERT(*downSampleFactor>0 && *downSampleFactor<=2); - downsampledOffset = (delayDiff*(*numChannels))>>(*downSampleFactor-1); - sbrOffset = 0; + /* Set parametric Stereo Flag. */ + if (usePs) { + elInfo[coreEl].fParametricStereo = 1; } else { - /* Delay SBR input */ - if ( delayDiff > (int)coreFrameLength * (int)*downSampleFactor ) - { - /* Do bitstream frame-wise delay balancing if we have more than SBR framelength samples delay difference */ - delayDiff -= coreFrameLength * *downSampleFactor; - nBitstrDelay = 1; - } - /* Multiply input offset by input channels */ - sbrOffset = delayDiff*(*numChannels); - downsampledOffset = 0; + elInfo[coreEl].fParametricStereo = 0; } - hSbrEncoder->nBitstrDelay = nBitstrDelay; - hSbrEncoder->nChannels = *numChannels; - hSbrEncoder->frameSize = coreFrameLength * *downSampleFactor; - hSbrEncoder->fTimeDomainDownsampling = timeDomainDownsample; - hSbrEncoder->downSampleFactor = *downSampleFactor; - hSbrEncoder->estimateBitrate = 0; - hSbrEncoder->inputDataDelay = 0; - - - /* Open SBR elements */ - el = -1; - highestSbrStartFreq = highestSbrStopFreq = 0; - lowestBandwidth = 99999; - - /* Loop through each core encoder element and get a matching SBR element config */ - for (coreEl=0; coreEl<noElements; coreEl++) - { - /* SBR only handles SCE and CPE's */ - if (elInfo[coreEl].elType == ID_SCE || elInfo[coreEl].elType == ID_CPE) { - el++; - } else { - continue; - } - /* Set parametric Stereo Flag. */ - if (usePs) { - elInfo[coreEl].fParametricStereo = 1; - } else { - elInfo[coreEl].fParametricStereo = 0; - } - - /* - * Init sbrConfig structure - */ - if ( ! FDKsbrEnc_InitializeSbrDefaults ( &sbrConfig[el], - *downSampleFactor, - coreFrameLength, - IS_LOWDELAY(aot) - ) ) - { - error = 1; - goto bail; - } - - /* - * Modify sbrConfig structure according to Element parameters - */ - if ( ! FDKsbrEnc_AdjustSbrSettings (&sbrConfig[el], - elInfo[coreEl].bitRate, - elInfo[coreEl].nChannelsInEl, - *coreSampleRate, - inputSampleRate, - transformFactor, - 24000, - 0, - 0, /* useSpeechConfig */ - 0, /* lcsMode */ - usePs, /* bParametricStereo */ - aot) ) - { - error = 1; - goto bail; - } - - /* Find common frequency border for all SBR elements */ - highestSbrStartFreq = fixMax(highestSbrStartFreq, sbrConfig[el].startFreq); - highestSbrStopFreq = fixMax(highestSbrStopFreq, sbrConfig[el].stopFreq); - - } /* first element loop */ - - /* Set element count (can be less than core encoder element count) */ - hSbrEncoder->noElements = el+1; + /* + * Init sbrConfig structure + */ + if (!FDKsbrEnc_InitializeSbrDefaults(&sbrConfig[el], *downSampleFactor, + coreFrameLength, IS_LOWDELAY(aot))) { + error = 1; + goto bail; + } - FDKsbrEnc_Reallocate(hSbrEncoder, - elInfo, - noElements); + /* + * Modify sbrConfig structure according to Element parameters + */ + if (!FDKsbrEnc_AdjustSbrSettings( + &sbrConfig[el], elInfo[coreEl].bitRate, + elInfo[coreEl].nChannelsInEl, *coreSampleRate, inputSampleRate, + transformFactor, 24000, 0, 0, /* useSpeechConfig */ + 0, /* lcsMode */ + usePs, /* bParametricStereo */ + aot)) { + error = 1; + goto bail; + } - for (el=0; el<hSbrEncoder->noElements; el++) { + /* Find common frequency border for all SBR elements */ + highestSbrStartFreq = + fixMax(highestSbrStartFreq, sbrConfig[el].startFreq); + highestSbrStopFreq = fixMax(highestSbrStopFreq, sbrConfig[el].stopFreq); - int bandwidth = *coreBandwidth; + } /* first element loop */ - /* Use lowest common bandwidth */ - sbrConfig[el].startFreq = highestSbrStartFreq; - sbrConfig[el].stopFreq = highestSbrStopFreq; + /* Set element count (can be less than core encoder element count) */ + hSbrEncoder->noElements = el + 1; - /* initialize SBR element, and get core bandwidth */ - error = FDKsbrEnc_EnvInit(hSbrEncoder->sbrElement[el], - &sbrConfig[el], - &bandwidth, - aot, - nBitstrDelay, - el, - headerPeriod, - statesInitFlag, - hSbrEncoder->fTimeDomainDownsampling - ,hSbrEncoder->dynamicRam - ); + FDKsbrEnc_Reallocate(hSbrEncoder, elInfo, noElements); - if (error != 0) { - error = 2; - goto bail; - } + for (el = 0; el < hSbrEncoder->noElements; el++) { + int bandwidth = *coreBandwidth; - /* Get lowest core encoder bandwidth to be returned later. */ - lowestBandwidth = fixMin(lowestBandwidth, bandwidth); + /* Use lowest common bandwidth */ + sbrConfig[el].startFreq = highestSbrStartFreq; + sbrConfig[el].stopFreq = highestSbrStopFreq; - } /* second element loop */ + /* initialize SBR element, and get core bandwidth */ + error = FDKsbrEnc_EnvInit(hSbrEncoder->sbrElement[el], &sbrConfig[el], + &bandwidth, aot, el, headerPeriod, + statesInitFlag, hSbrEncoder->downsamplingMethod, + hSbrEncoder->dynamicRam); - /* Initialize a downsampler for each channel in each SBR element */ - if (hSbrEncoder->fTimeDomainDownsampling) - { - for (el=0; el<hSbrEncoder->noElements; el++) - { - HANDLE_SBR_ELEMENT hSbrEl = hSbrEncoder->sbrElement[el]; - INT Wc, ch; + if (error != 0) { + error = 2; + goto bail; + } - /* Calculated required normalized cutoff frequency (Wc = 1.0 -> lowestBandwidth = inputSampleRate/2) */ - Wc = (2*lowestBandwidth)*1000 / inputSampleRate; + /* Get lowest core encoder bandwidth to be returned later. */ + lowestBandwidth = fixMin(lowestBandwidth, bandwidth); - for (ch=0; ch<hSbrEl->elInfo.nChannelsInEl; ch++) - { - FDKaacEnc_InitDownsampler (&hSbrEl->sbrChannel[ch]->downSampler, Wc, *downSampleFactor); - FDK_ASSERT (hSbrEl->sbrChannel[ch]->downSampler.delay <=MAX_DS_FILTER_DELAY); - } + } /* second element loop */ - downsamplerDelay = hSbrEl->sbrChannel[0]->downSampler.delay; - } /* third element loop */ + /* Initialize a downsampler for each channel in each SBR element */ + if (hSbrEncoder->downsamplingMethod == SBRENC_DS_TIME) { + for (el = 0; el < hSbrEncoder->noElements; el++) { + HANDLE_SBR_ELEMENT hSbrEl = hSbrEncoder->sbrElement[el]; + INT Wc, ch; - /* lfe */ - FDKaacEnc_InitDownsampler (&hSbrEncoder->lfeDownSampler, 0, *downSampleFactor); + Wc = 500; /* Cutoff frequency with full bandwidth */ - /* Add the resampler additional delay to get the final delay and buffer offset values. */ - if (sbrOffset > 0 || downsampledOffset <= ((downsamplerDelay * (*numChannels))>>(*downSampleFactor-1))) { - sbrOffset += (downsamplerDelay - downsampledOffset) * (*numChannels) ; - *delay += downsamplerDelay - downsampledOffset; - downsampledOffset = 0; - } else { - downsampledOffset -= (downsamplerDelay * (*numChannels))>>(*downSampleFactor-1); - sbrOffset = 0; + for (ch = 0; ch < hSbrEl->elInfo.nChannelsInEl; ch++) { + FDKaacEnc_InitDownsampler(&hSbrEl->sbrChannel[ch]->downSampler, Wc, + *downSampleFactor); + FDK_ASSERT(hSbrEl->sbrChannel[ch]->downSampler.delay <= + MAX_DS_FILTER_DELAY); } + } /* third element loop */ - hSbrEncoder->inputDataDelay = downsamplerDelay; - } + /* lfe */ + FDKaacEnc_InitDownsampler(&hSbrEncoder->lfeDownSampler, 0, + *downSampleFactor); + } - /* Assign core encoder Bandwidth */ - *coreBandwidth = lowestBandwidth; + /* Get delay information */ + delayParam.dsDelay = + hSbrEncoder->sbrElement[0]->sbrChannel[0]->downSampler.delay; + delayParam.delay = *delay; - /* Estimate sbr bitrate, 2.5 kBit/s per sbr channel */ - hSbrEncoder->estimateBitrate += 2500 * (*numChannels); + error = sbrEncoder_Init_delay(coreFrameLength, *numChannels, + *downSampleFactor, lowDelay, usePs, is212, + downsamplingMethod, &delayParam); - /* initialize parametric stereo */ - if (usePs) - { - PSENC_CONFIG psEncConfig; - FDK_ASSERT(hSbrEncoder->noElements == 1); - INT psTuningTableIdx = getPsTuningTableIndex(elInfo[0].bitRate, NULL); + if (error != 0) { + error = 3; + goto bail; + } - psEncConfig.frameSize = coreFrameLength; //sbrConfig.sbrFrameSize; - psEncConfig.qmfFilterMode = 0; - psEncConfig.sbrPsDelay = 0; + hSbrEncoder->nBitstrDelay = delayParam.bitstrDelay; + hSbrEncoder->sbrDecDelay = delayParam.sbrDecDelay; + hSbrEncoder->inputDataDelay = delayParam.delayInput2Core; - /* tuning parameters */ - if (psTuningTableIdx != INVALID_TABLE_IDX) { - psEncConfig.nStereoBands = psTuningTable[psTuningTableIdx].nStereoBands; - psEncConfig.maxEnvelopes = psTuningTable[psTuningTableIdx].nEnvelopes; - psEncConfig.iidQuantErrorThreshold = (FIXP_DBL)psTuningTable[psTuningTableIdx].iidQuantErrorThreshold; + /* Assign core encoder Bandwidth */ + *coreBandwidth = lowestBandwidth; - /* calculation is not quite linear, increased number of envelopes causes more bits */ - /* assume avg. 50 bits per frame for 10 stereo bands / 1 envelope configuration */ - hSbrEncoder->estimateBitrate += ( (((*coreSampleRate) * 5 * psEncConfig.nStereoBands * psEncConfig.maxEnvelopes) / hSbrEncoder->frameSize)); + /* Estimate sbr bitrate, 2.5 kBit/s per sbr channel */ + hSbrEncoder->estimateBitrate += 2500 * (*numChannels); - } else { - error = ERROR(CDI, "Invalid ps tuning table index."); - goto bail; - } - - qmfInitSynthesisFilterBank(&hSbrEncoder->qmfSynthesisPS, - (FIXP_DBL*)hSbrEncoder->qmfSynthesisPS.FilterStates, - hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfSlots, - hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfBands>>1, - hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfBands>>1, - hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfBands>>1, - (statesInitFlag) ? 0 : QMF_FLAG_KEEP_STATES); - - if(errorInfo == noError){ - /* update delay */ - psEncConfig.sbrPsDelay = FDKsbrEnc_GetEnvEstDelay(&hSbrEncoder->sbrElement[0]->sbrChannel[0]->hEnvChannel.sbrExtractEnvelope); - - if(noError != (errorInfo = PSEnc_Init( hSbrEncoder->hParametricStereo, - &psEncConfig, - hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfSlots, - hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfBands - ,hSbrEncoder->dynamicRam - ))) - { - errorInfo = handBack(errorInfo); - } - } + /* Initialize bitstream buffer for each element */ + for (el = 0; el < hSbrEncoder->noElements; el++) { + FDKsbrEnc_bsBufInit(hSbrEncoder->sbrElement[el], delayParam.bitstrDelay); + } - /* QMF analysis + Hybrid analysis + Hybrid synthesis + QMF synthesis + downsampled input buffer delay */ - hSbrEncoder->inputDataDelay = (64*10/2) + (6*64) + (0) + (64*10/2-64+1) + ((*downSampleFactor)*downsampledOffset); - } + /* initialize parametric stereo */ + if (usePs) { + PSENC_CONFIG psEncConfig; + FDK_ASSERT(hSbrEncoder->noElements == 1); + INT psTuningTableIdx = getPsTuningTableIndex(elInfo[0].bitRate, NULL); + + psEncConfig.frameSize = coreFrameLength; // sbrConfig.sbrFrameSize; + psEncConfig.qmfFilterMode = 0; + psEncConfig.sbrPsDelay = 0; + + /* tuning parameters */ + if (psTuningTableIdx != INVALID_TABLE_IDX) { + psEncConfig.nStereoBands = psTuningTable[psTuningTableIdx].nStereoBands; + psEncConfig.maxEnvelopes = psTuningTable[psTuningTableIdx].nEnvelopes; + psEncConfig.iidQuantErrorThreshold = + (FIXP_DBL)psTuningTable[psTuningTableIdx].iidQuantErrorThreshold; + + /* calculation is not quite linear, increased number of envelopes causes + * more bits */ + /* assume avg. 50 bits per frame for 10 stereo bands / 1 envelope + * configuration */ + hSbrEncoder->estimateBitrate += + ((((*coreSampleRate) * 5 * psEncConfig.nStereoBands * + psEncConfig.maxEnvelopes) / + hSbrEncoder->frameSize)); - hSbrEncoder->downsampledOffset = downsampledOffset; - { - hSbrEncoder->downmixSize = coreFrameLength*(*numChannels); + } else { + error = ERROR(CDI, "Invalid ps tuning table index."); + goto bail; } - hSbrEncoder->bufferOffset = sbrOffset; - /* Delay Compensation: fill bitstream delay buffer with zero input signal */ - if ( hSbrEncoder->nBitstrDelay > 0 ) - { - error = FDKsbrEnc_DelayCompensation (hSbrEncoder, inputBuffer); - if (error != 0) - goto bail; + qmfInitSynthesisFilterBank( + &hSbrEncoder->qmfSynthesisPS, + (FIXP_DBL *)hSbrEncoder->qmfSynthesisPS.FilterStates, + hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfSlots, + hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfBands >> 1, + hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfBands >> 1, + hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfBands >> 1, + (statesInitFlag) ? 0 : QMF_FLAG_KEEP_STATES); + + if (errorInfo == noError) { + /* update delay */ + psEncConfig.sbrPsDelay = + FDKsbrEnc_GetEnvEstDelay(&hSbrEncoder->sbrElement[0] + ->sbrChannel[0] + ->hEnvChannel.sbrExtractEnvelope); + + errorInfo = + PSEnc_Init(hSbrEncoder->hParametricStereo, &psEncConfig, + hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfSlots, + hSbrEncoder->sbrElement[0]->sbrConfigData.noQmfBands, + hSbrEncoder->dynamicRam); } + } - /* Set Output frame length */ - *frameLength = coreFrameLength * *downSampleFactor; - /* Input buffer offset */ - *inputBufferOffset = fixMax(sbrOffset, downsampledOffset); + hSbrEncoder->downsampledOffset = delayParam.corePathOffset; + hSbrEncoder->bufferOffset = delayParam.sbrPathOffset; + *delay = delayParam.delay; + { hSbrEncoder->downmixSize = coreFrameLength * (*numChannels); } + /* Delay Compensation: fill bitstream delay buffer with zero input signal */ + if (hSbrEncoder->nBitstrDelay > 0) { + error = FDKsbrEnc_DelayCompensation(hSbrEncoder, inputBuffer, + inputBufferBufSize); + if (error != 0) goto bail; } - return error; + /* Set Output frame length */ + *frameLength = coreFrameLength * *downSampleFactor; + /* Input buffer offset */ + *inputBufferOffset = + fixMax(delayParam.sbrPathOffset, delayParam.corePathOffset); + } + + return error; bail: - /* Restore input settings */ - *coreSampleRate = inputSampleRate; - *frameLength = coreFrameLength; - *numChannels = inputChannels; - *coreBandwidth = inputBandWidth; - - return error; - } - - -INT -sbrEncoder_EncodeFrame( HANDLE_SBR_ENCODER hSbrEncoder, - INT_PCM *samples, - UINT timeInStride, - UINT sbrDataBits[(8)], - UCHAR sbrData[(8)][MAX_PAYLOAD_SIZE] - ) -{ + /* Restore input settings */ + *coreSampleRate = inputSampleRate; + *frameLength = coreFrameLength; + *numChannels = inputChannels; + *coreBandwidth = inputBandWidth; + + return error; +} + +INT sbrEncoder_EncodeFrame(HANDLE_SBR_ENCODER hSbrEncoder, INT_PCM *samples, + UINT samplesBufSize, UINT sbrDataBits[(8)], + UCHAR sbrData[(8)][MAX_PAYLOAD_SIZE]) { INT error; int el; - for (el=0; el<hSbrEncoder->noElements; el++) - { - if (hSbrEncoder->sbrElement[el] != NULL) - { + for (el = 0; el < hSbrEncoder->noElements; el++) { + if (hSbrEncoder->sbrElement[el] != NULL) { error = FDKsbrEnc_EnvEncodeFrame( - hSbrEncoder, - el, - samples + hSbrEncoder->downsampledOffset, - timeInStride, - &sbrDataBits[el], - sbrData[el], - 0 - ); - if (error) - return error; + hSbrEncoder, el, + samples + hSbrEncoder->downsampledOffset / hSbrEncoder->nChannels, + samplesBufSize, &sbrDataBits[el], sbrData[el], 0); + if (error) return error; } } - if ( ( hSbrEncoder->lfeChIdx!=-1) && (hSbrEncoder->downSampleFactor > 1) ) - { /* lfe downsampler */ - INT nOutSamples; + error = FDKsbrEnc_Downsample( + hSbrEncoder, + samples + hSbrEncoder->downsampledOffset / hSbrEncoder->nChannels, + samplesBufSize, hSbrEncoder->nChannels, &sbrDataBits[el], sbrData[el], 0); + if (error) return error; - FDKaacEnc_Downsample(&hSbrEncoder->lfeDownSampler, - samples + hSbrEncoder->downsampledOffset + hSbrEncoder->bufferOffset + hSbrEncoder->lfeChIdx, - hSbrEncoder->frameSize, - timeInStride, - samples + hSbrEncoder->downsampledOffset + hSbrEncoder->lfeChIdx, - &nOutSamples, - hSbrEncoder->nChannels); + return 0; +} +INT sbrEncoder_UpdateBuffers(HANDLE_SBR_ENCODER hSbrEncoder, + INT_PCM *timeBuffer, UINT timeBufferBufSize) { + if (hSbrEncoder->downsampledOffset > 0) { + int c; + int nd = hSbrEncoder->downmixSize / hSbrEncoder->nChannels; - } + for (c = 0; c < hSbrEncoder->nChannels; c++) { + /* Move delayed downsampled data */ + FDKmemcpy(timeBuffer + timeBufferBufSize * c, + timeBuffer + timeBufferBufSize * c + nd, + sizeof(INT_PCM) * + (hSbrEncoder->downsampledOffset / hSbrEncoder->nChannels)); + } + } else { + int c; + for (c = 0; c < hSbrEncoder->nChannels; c++) { + /* Move delayed input data */ + FDKmemcpy( + timeBuffer + timeBufferBufSize * c, + timeBuffer + timeBufferBufSize * c + hSbrEncoder->frameSize, + sizeof(INT_PCM) * hSbrEncoder->bufferOffset / hSbrEncoder->nChannels); + } + } + if (hSbrEncoder->nBitstrDelay > 0) { + int el; + + for (el = 0; el < hSbrEncoder->noElements; el++) { + FDKmemmove( + hSbrEncoder->sbrElement[el]->payloadDelayLine[0], + hSbrEncoder->sbrElement[el]->payloadDelayLine[1], + sizeof(UCHAR) * (hSbrEncoder->nBitstrDelay * MAX_PAYLOAD_SIZE)); + + FDKmemmove(&hSbrEncoder->sbrElement[el]->payloadDelayLineSize[0], + &hSbrEncoder->sbrElement[el]->payloadDelayLineSize[1], + sizeof(UINT) * (hSbrEncoder->nBitstrDelay)); + } + } return 0; } +INT sbrEncoder_SendHeader(HANDLE_SBR_ENCODER hSbrEncoder) { + INT error = -1; + if (hSbrEncoder) { + int el; + for (el = 0; el < hSbrEncoder->noElements; el++) { + if ((hSbrEncoder->noElements == 1) && + (hSbrEncoder->sbrElement[0]->elInfo.fParametricStereo == 1)) { + hSbrEncoder->sbrElement[el]->sbrBitstreamData.CountSendHeaderData = + hSbrEncoder->sbrElement[el]->sbrBitstreamData.NrSendHeaderData - 1; + } else { + hSbrEncoder->sbrElement[el]->sbrBitstreamData.CountSendHeaderData = 0; + } + } + error = 0; + } + return error; +} -INT sbrEncoder_UpdateBuffers( - HANDLE_SBR_ENCODER hSbrEncoder, - INT_PCM *timeBuffer - ) - { - if ( hSbrEncoder->downsampledOffset > 0 ) { - /* Move delayed downsampled data */ - FDKmemcpy ( timeBuffer, - timeBuffer + hSbrEncoder->downmixSize, - sizeof(INT_PCM) * (hSbrEncoder->downsampledOffset) ); - } else { - /* Move delayed input data */ - FDKmemcpy ( timeBuffer, - timeBuffer + hSbrEncoder->nChannels * hSbrEncoder->frameSize, - sizeof(INT_PCM) * hSbrEncoder->bufferOffset ); +INT sbrEncoder_ContainsHeader(HANDLE_SBR_ENCODER hSbrEncoder) { + INT sbrHeader = 1; + if (hSbrEncoder) { + int el; + for (el = 0; el < hSbrEncoder->noElements; el++) { + sbrHeader &= + (hSbrEncoder->sbrElement[el]->sbrBitstreamData.HeaderActiveDelay == 1) + ? 1 + : 0; } - if ( hSbrEncoder->nBitstrDelay > 0 ) - { - int el; + } + return sbrHeader; +} - for (el=0; el<hSbrEncoder->noElements; el++) - { - FDKmemmove ( hSbrEncoder->sbrElement[el]->payloadDelayLine[0], - hSbrEncoder->sbrElement[el]->payloadDelayLine[1], - sizeof(UCHAR) * (hSbrEncoder->nBitstrDelay*MAX_PAYLOAD_SIZE) ); +INT sbrEncoder_GetHeaderDelay(HANDLE_SBR_ENCODER hSbrEncoder) { + INT delay = -1; - FDKmemmove( &hSbrEncoder->sbrElement[el]->payloadDelayLineSize[0], - &hSbrEncoder->sbrElement[el]->payloadDelayLineSize[1], - sizeof(UINT) * (hSbrEncoder->nBitstrDelay) ); - } + if (hSbrEncoder) { + if ((hSbrEncoder->noElements == 1) && + (hSbrEncoder->sbrElement[0]->elInfo.fParametricStereo == 1)) { + delay = hSbrEncoder->nBitstrDelay + 1; + } else { + delay = hSbrEncoder->nBitstrDelay; } - return 0; - } + } + return delay; +} +INT sbrEncoder_GetBsDelay(HANDLE_SBR_ENCODER hSbrEncoder) { + INT delay = -1; + if (hSbrEncoder) { + delay = hSbrEncoder->nBitstrDelay; + } + return delay; +} -INT sbrEncoder_GetEstimateBitrate(HANDLE_SBR_ENCODER hSbrEncoder) -{ +INT sbrEncoder_SAPPrepare(HANDLE_SBR_ENCODER hSbrEncoder) { + INT error = -1; + if (hSbrEncoder) { + int el; + for (el = 0; el < hSbrEncoder->noElements; el++) { + hSbrEncoder->sbrElement[el]->sbrBitstreamData.rightBorderFIX = 1; + } + error = 0; + } + return error; +} + +INT sbrEncoder_GetEstimateBitrate(HANDLE_SBR_ENCODER hSbrEncoder) { INT estimateBitrate = 0; - if(hSbrEncoder) { + if (hSbrEncoder) { estimateBitrate += hSbrEncoder->estimateBitrate; } return estimateBitrate; } -INT sbrEncoder_GetInputDataDelay(HANDLE_SBR_ENCODER hSbrEncoder) -{ +INT sbrEncoder_GetInputDataDelay(HANDLE_SBR_ENCODER hSbrEncoder) { INT delay = -1; - if(hSbrEncoder) { + if (hSbrEncoder) { delay = hSbrEncoder->inputDataDelay; } return delay; } +INT sbrEncoder_GetSbrDecDelay(HANDLE_SBR_ENCODER hSbrEncoder) { + INT delay = -1; -INT sbrEncoder_GetLibInfo( LIB_INFO *info ) -{ + if (hSbrEncoder) { + delay = hSbrEncoder->sbrDecDelay; + } + return delay; +} + +INT sbrEncoder_GetLibInfo(LIB_INFO *info) { int i; if (info == NULL) { @@ -2421,7 +2557,8 @@ INT sbrEncoder_GetLibInfo( LIB_INFO *info ) info += i; info->module_id = FDK_SBRENC; - info->version = LIB_VERSION(SBRENCODER_LIB_VL0, SBRENCODER_LIB_VL1, SBRENCODER_LIB_VL2); + info->version = + LIB_VERSION(SBRENCODER_LIB_VL0, SBRENCODER_LIB_VL1, SBRENCODER_LIB_VL2); LIB_VERSION_STRING(info); #ifdef __ANDROID__ info->build_date = ""; @@ -2433,10 +2570,7 @@ INT sbrEncoder_GetLibInfo( LIB_INFO *info ) info->title = "SBR Encoder"; /* Set flags */ - info->flags = 0 - | CAPF_SBR_HQ - | CAPF_SBR_PS_MPEG - ; + info->flags = 0 | CAPF_SBR_HQ | CAPF_SBR_PS_MPEG; /* End of flags */ return 0; diff --git a/libSBRenc/src/sbr_misc.cpp b/libSBRenc/src/sbr_misc.cpp index c673b81..83d7e36 100644 --- a/libSBRenc/src/sbr_misc.cpp +++ b/libSBRenc/src/sbr_misc.cpp @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,17 +90,23 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ /*! \file - \brief Sbr miscellaneous helper functions + \brief Sbr miscellaneous helper functions $Revision: 36750 $ */ #include "sbr_misc.h" - -void FDKsbrEnc_Shellsort_fract (FIXP_DBL *in, INT n) -{ +void FDKsbrEnc_Shellsort_fract(FIXP_DBL *in, INT n) { FIXP_DBL v; INT i, j; INT inc = 1; @@ -101,24 +118,20 @@ void FDKsbrEnc_Shellsort_fract (FIXP_DBL *in, INT n) do { inc = inc / 3; for (i = inc + 1; i <= n; i++) { - v = in[i-1]; + v = in[i - 1]; j = i; - while (in[j-inc-1] > v) { - in[j-1] = in[j-inc-1]; + while (in[j - inc - 1] > v) { + in[j - 1] = in[j - inc - 1]; j -= inc; - if (j <= inc) - break; + if (j <= inc) break; } - in[j-1] = v; + in[j - 1] = v; } } while (inc > 1); - } /* Sorting routine */ -void FDKsbrEnc_Shellsort_int (INT *in, INT n) -{ - +void FDKsbrEnc_Shellsort_int(INT *in, INT n) { INT i, j, v; INT inc = 1; @@ -129,22 +142,18 @@ void FDKsbrEnc_Shellsort_int (INT *in, INT n) do { inc = inc / 3; for (i = inc + 1; i <= n; i++) { - v = in[i-1]; + v = in[i - 1]; j = i; - while (in[j-inc-1] > v) { - in[j-1] = in[j-inc-1]; + while (in[j - inc - 1] > v) { + in[j - 1] = in[j - inc - 1]; j -= inc; - if (j <= inc) - break; + if (j <= inc) break; } - in[j-1] = v; + in[j - 1] = v; } } while (inc > 1); - } - - /******************************************************************************* Functionname: FDKsbrEnc_AddVecLeft ******************************************************************************* @@ -156,16 +165,13 @@ void FDKsbrEnc_Shellsort_int (INT *in, INT n) Return: none *******************************************************************************/ -void -FDKsbrEnc_AddVecLeft (INT *dst, INT *length_dst, INT *src, INT length_src) -{ +void FDKsbrEnc_AddVecLeft(INT *dst, INT *length_dst, INT *src, INT length_src) { INT i; for (i = length_src - 1; i >= 0; i--) - FDKsbrEnc_AddLeft (dst, length_dst, src[i]); + FDKsbrEnc_AddLeft(dst, length_dst, src[i]); } - /******************************************************************************* Functionname: FDKsbrEnc_AddLeft ******************************************************************************* @@ -177,18 +183,14 @@ FDKsbrEnc_AddVecLeft (INT *dst, INT *length_dst, INT *src, INT length_src) Return: none *******************************************************************************/ -void -FDKsbrEnc_AddLeft (INT *vector, INT *length_vector, INT value) -{ +void FDKsbrEnc_AddLeft(INT *vector, INT *length_vector, INT value) { INT i; - for (i = *length_vector; i > 0; i--) - vector[i] = vector[i - 1]; + for (i = *length_vector; i > 0; i--) vector[i] = vector[i - 1]; vector[0] = value; (*length_vector)++; } - /******************************************************************************* Functionname: FDKsbrEnc_AddRight ******************************************************************************* @@ -200,15 +202,11 @@ FDKsbrEnc_AddLeft (INT *vector, INT *length_vector, INT value) Return: none *******************************************************************************/ -void -FDKsbrEnc_AddRight (INT *vector, INT *length_vector, INT value) -{ +void FDKsbrEnc_AddRight(INT *vector, INT *length_vector, INT value) { vector[*length_vector] = value; (*length_vector)++; } - - /******************************************************************************* Functionname: FDKsbrEnc_AddVecRight ******************************************************************************* @@ -220,15 +218,12 @@ FDKsbrEnc_AddRight (INT *vector, INT *length_vector, INT value) Return: none *******************************************************************************/ -void -FDKsbrEnc_AddVecRight (INT *dst, INT *length_dst, INT *src, INT length_src) -{ +void FDKsbrEnc_AddVecRight(INT *dst, INT *length_dst, INT *src, + INT length_src) { INT i; - for (i = 0; i < length_src; i++) - FDKsbrEnc_AddRight (dst, length_dst, src[i]); + for (i = 0; i < length_src; i++) FDKsbrEnc_AddRight(dst, length_dst, src[i]); } - /***************************************************************************** functionname: FDKsbrEnc_LSI_divide_scale_fract @@ -238,35 +233,33 @@ FDKsbrEnc_AddVecRight (INT *dst, INT *length_dst, INT *src, INT length_src) return: num*scale/denom *****************************************************************************/ -FIXP_DBL FDKsbrEnc_LSI_divide_scale_fract(FIXP_DBL num, FIXP_DBL denom, FIXP_DBL scale) -{ +FIXP_DBL FDKsbrEnc_LSI_divide_scale_fract(FIXP_DBL num, FIXP_DBL denom, + FIXP_DBL scale) { FIXP_DBL tmp = FL2FXCONST_DBL(0.0f); if (num != FL2FXCONST_DBL(0.0f)) { - INT shiftCommon; - INT shiftNum = CountLeadingBits(num); + INT shiftNum = CountLeadingBits(num); INT shiftDenom = CountLeadingBits(denom); INT shiftScale = CountLeadingBits(scale); - num = num << shiftNum; + num = num << shiftNum; scale = scale << shiftScale; - tmp = fMultDiv2(num,scale); + tmp = fMultDiv2(num, scale); - if ( denom > (tmp >> fixMin(shiftNum+shiftScale-1,(DFRACT_BITS-1))) ) { + if (denom > (tmp >> fixMin(shiftNum + shiftScale - 1, (DFRACT_BITS - 1)))) { denom = denom << shiftDenom; - tmp = schur_div(tmp,denom,15); - shiftCommon = fixMin((shiftNum-shiftDenom+shiftScale-1),(DFRACT_BITS-1)); + tmp = schur_div(tmp, denom, 15); + shiftCommon = + fixMin((shiftNum - shiftDenom + shiftScale - 1), (DFRACT_BITS - 1)); if (shiftCommon < 0) tmp <<= -shiftCommon; else - tmp >>= shiftCommon; - } - else { + tmp >>= shiftCommon; + } else { tmp = /*FL2FXCONST_DBL(1.0)*/ (FIXP_DBL)MAXVAL_DBL; } } return (tmp); } - diff --git a/libSBRenc/src/sbr_misc.h b/libSBRenc/src/sbr_misc.h index f471974..fad853f 100644 --- a/libSBRenc/src/sbr_misc.h +++ b/libSBRenc/src/sbr_misc.h @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,28 +90,38 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ /*! \file - \brief Sbr miscellaneous helper functions prototypes + \brief Sbr miscellaneous helper functions prototypes $Revision: 92790 $ \author */ -#ifndef _SBR_MISC_H -#define _SBR_MISC_H +#ifndef SBR_MISC_H +#define SBR_MISC_H #include "sbr_encoder.h" /* Sorting routines */ -void FDKsbrEnc_Shellsort_fract (FIXP_DBL *in, INT n); -void FDKsbrEnc_Shellsort_int (INT *in, INT n); +void FDKsbrEnc_Shellsort_fract(FIXP_DBL *in, INT n); +void FDKsbrEnc_Shellsort_int(INT *in, INT n); -void FDKsbrEnc_AddLeft (INT *vector, INT *length_vector, INT value); -void FDKsbrEnc_AddRight (INT *vector, INT *length_vector, INT value); -void FDKsbrEnc_AddVecLeft (INT *dst, INT *length_dst, INT *src, INT length_src); -void FDKsbrEnc_AddVecRight (INT *dst, INT *length_vector_dst, INT *src, INT length_src); +void FDKsbrEnc_AddLeft(INT *vector, INT *length_vector, INT value); +void FDKsbrEnc_AddRight(INT *vector, INT *length_vector, INT value); +void FDKsbrEnc_AddVecLeft(INT *dst, INT *length_dst, INT *src, INT length_src); +void FDKsbrEnc_AddVecRight(INT *dst, INT *length_vector_dst, INT *src, + INT length_src); -FIXP_DBL FDKsbrEnc_LSI_divide_scale_fract(FIXP_DBL num, FIXP_DBL denom, FIXP_DBL scale); +FIXP_DBL FDKsbrEnc_LSI_divide_scale_fract(FIXP_DBL num, FIXP_DBL denom, + FIXP_DBL scale); #endif diff --git a/libSBRenc/src/sbr_ram.cpp b/libSBRenc/src/sbr_ram.cpp deleted file mode 100644 index ee6c37f..0000000 --- a/libSBRenc/src/sbr_ram.cpp +++ /dev/null @@ -1,222 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 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 ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Memory layout - - - This module declares all static and dynamic memory spaces -*/ -#include "sbr_ram.h" - -#include "sbr.h" -#include "genericStds.h" - -C_ALLOC_MEM (Ram_SbrDynamic_RAM, FIXP_DBL, ((SBR_ENC_DYN_RAM_SIZE)/sizeof(FIXP_DBL))) - -/*! - \name StaticSbrData - - Static memory areas, must not be overwritten in other sections of the encoder -*/ -/* @{ */ - -/*! static sbr encoder instance for one encoder (2 channels) - all major static and dynamic memory areas are located - in module sbr_ram and sbr rom -*/ -C_ALLOC_MEM (Ram_SbrEncoder, SBR_ENCODER, 1) -C_ALLOC_MEM2(Ram_SbrChannel, SBR_CHANNEL, 1, (8)) -C_ALLOC_MEM2(Ram_SbrElement, SBR_ELEMENT, 1, (8)) - -/*! Filter states for QMF-analysis. <br> - Dimension: #MAXNRSBRCHANNELS * #SBR_QMF_FILTER_LENGTH -*/ -C_AALLOC_MEM2_L (Ram_Sbr_QmfStatesAnalysis, FIXP_QAS, QMF_FILTER_LENGTH, (8), SECT_DATA_L1) - - -/*! Matrix holding the quota values for all estimates, all channels - Dimension #MAXNRSBRCHANNELS * +#SBR_QMF_CHANNELS* #MAX_NO_OF_ESTIMATES -*/ -C_ALLOC_MEM2_L (Ram_Sbr_quotaMatrix, FIXP_DBL, (MAX_NO_OF_ESTIMATES*QMF_CHANNELS), (8), SECT_DATA_L1) - -/*! Matrix holding the sign values for all estimates, all channels - Dimension #MAXNRSBRCHANNELS * +#SBR_QMF_CHANNELS* #MAX_NO_OF_ESTIMATES -*/ -C_ALLOC_MEM2 (Ram_Sbr_signMatrix, INT, (MAX_NO_OF_ESTIMATES*QMF_CHANNELS), (8)) - -/*! Frequency band table (low res) <br> - Dimension #MAX_FREQ_COEFFS/2+1 -*/ -C_ALLOC_MEM2 (Ram_Sbr_freqBandTableLO, UCHAR, (MAX_FREQ_COEFFS/2+1), (8)) - -/*! Frequency band table (high res) <br> - Dimension #MAX_FREQ_COEFFS +1 -*/ -C_ALLOC_MEM2 (Ram_Sbr_freqBandTableHI, UCHAR, (MAX_FREQ_COEFFS+1), (8)) - -/*! vk matser table <br> - Dimension #MAX_FREQ_COEFFS +1 -*/ -C_ALLOC_MEM2 (Ram_Sbr_v_k_master, UCHAR, (MAX_FREQ_COEFFS+1), (8)) - - -/* - Missing harmonics detection -*/ - -/*! sbr_detectionVectors <br> - Dimension #MAX_NUM_CHANNELS*#MAX_NO_OF_ESTIMATES*#MAX_FREQ_COEFFS] -*/ -C_ALLOC_MEM2 (Ram_Sbr_detectionVectors, UCHAR, (MAX_NO_OF_ESTIMATES*MAX_FREQ_COEFFS), (8)) - -/*! sbr_prevCompVec[ <br> - Dimension #MAX_NUM_CHANNELS*#MAX_FREQ_COEFFS] -*/ -C_ALLOC_MEM2 (Ram_Sbr_prevEnvelopeCompensation, UCHAR, MAX_FREQ_COEFFS, (8)) -/*! sbr_guideScfb[ <br> - Dimension #MAX_NUM_CHANNELS*#MAX_FREQ_COEFFS] -*/ -C_ALLOC_MEM2 (Ram_Sbr_guideScfb, UCHAR, MAX_FREQ_COEFFS, (8)) - -/*! sbr_guideVectorDetected <br> - Dimension #MAX_NUM_CHANNELS*#MAX_NO_OF_ESTIMATES*#MAX_FREQ_COEFFS] -*/ -C_ALLOC_MEM2 (Ram_Sbr_guideVectorDetected, UCHAR, (MAX_NO_OF_ESTIMATES*MAX_FREQ_COEFFS), (8)) -C_ALLOC_MEM2 (Ram_Sbr_guideVectorDiff, FIXP_DBL, (MAX_NO_OF_ESTIMATES*MAX_FREQ_COEFFS), (8)) -C_ALLOC_MEM2 (Ram_Sbr_guideVectorOrig, FIXP_DBL, (MAX_NO_OF_ESTIMATES*MAX_FREQ_COEFFS), (8)) - -/* - Static Parametric Stereo memory -*/ -C_AALLOC_MEM_L(Ram_PsQmfStatesSynthesis, FIXP_DBL, QMF_FILTER_LENGTH/2, SECT_DATA_L1) - -C_ALLOC_MEM_L (Ram_PsEncode, PS_ENCODE, 1, SECT_DATA_L1) -C_ALLOC_MEM (Ram_ParamStereo, PARAMETRIC_STEREO, 1) - - - -/* @} */ - - -/*! - \name DynamicSbrData - - Dynamic memory areas, might be reused in other algorithm sections, - e.g. the core encoder. -*/ -/* @{ */ - - /*! Energy buffer for envelope extraction <br> - Dimension #MAXNRSBRCHANNELS * +#SBR_QMF_SLOTS * #SBR_QMF_CHANNELS - */ - C_ALLOC_MEM2 (Ram_Sbr_envYBuffer, FIXP_DBL, (QMF_MAX_TIME_SLOTS/2 * QMF_CHANNELS), (8)) - - FIXP_DBL* GetRam_Sbr_envYBuffer (int n, UCHAR* dynamic_RAM) { - FDK_ASSERT(dynamic_RAM!=0); - return ((FIXP_DBL*) (dynamic_RAM + OFFSET_NRG + (n*Y_2_BUF_BYTE) )); - } - - /* - * QMF data - */ - /* The SBR encoder uses a single channel overlapping buffer set (always n=0), but PS does not. */ - FIXP_DBL* GetRam_Sbr_envRBuffer (int n, UCHAR* dynamic_RAM) { - FDK_ASSERT(dynamic_RAM!=0); - return ((FIXP_DBL*) (dynamic_RAM + OFFSET_QMF + (n*(ENV_R_BUFF_BYTE+ENV_I_BUFF_BYTE)) )); - } - FIXP_DBL* GetRam_Sbr_envIBuffer (int n, UCHAR* dynamic_RAM) { - FDK_ASSERT(dynamic_RAM!=0); - return ((FIXP_DBL*) (dynamic_RAM + OFFSET_QMF + (ENV_R_BUFF_BYTE) + (n*(ENV_R_BUFF_BYTE+ENV_I_BUFF_BYTE)))); - } - - - - -/* @} */ - - - - - diff --git a/libSBRenc/src/sbr_ram.h b/libSBRenc/src/sbr_ram.h deleted file mode 100644 index 7e3d0c8..0000000 --- a/libSBRenc/src/sbr_ram.h +++ /dev/null @@ -1,187 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 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 ------------------------------------------------------------------------------------------------------------ */ - -/*! -\file -\brief Memory layout - -*/ -#ifndef __SBR_RAM_H -#define __SBR_RAM_H - -#include "sbr_def.h" -#include "env_est.h" -#include "sbr_encoder.h" -#include "sbr.h" - - - -#include "ps_main.h" -#include "ps_encode.h" - - -#define ENV_TRANSIENTS_BYTE ( (sizeof(FIXP_DBL)*(MAX_NUM_CHANNELS*3*QMF_MAX_TIME_SLOTS)) ) - - #define ENV_R_BUFF_BYTE ( (sizeof(FIXP_DBL)*((QMF_MAX_TIME_SLOTS) * MAX_HYBRID_BANDS)) ) - #define ENV_I_BUFF_BYTE ( (sizeof(FIXP_DBL)*((QMF_MAX_TIME_SLOTS) * MAX_HYBRID_BANDS)) ) - #define Y_BUF_CH_BYTE ( (2*sizeof(FIXP_DBL)*((QMF_MAX_TIME_SLOTS) * MAX_HYBRID_BANDS)) ) - - -#define ENV_R_BUF_PS_BYTE ( (sizeof(FIXP_DBL)*QMF_MAX_TIME_SLOTS * QMF_CHANNELS / 2) ) -#define ENV_I_BUF_PS_BYTE ( (sizeof(FIXP_DBL)*QMF_MAX_TIME_SLOTS * QMF_CHANNELS / 2) ) - -#define TON_BUF_CH_BYTE ( (sizeof(FIXP_DBL)*(MAX_NO_OF_ESTIMATES*MAX_FREQ_COEFFS)) ) - -#define Y_2_BUF_BYTE ( Y_BUF_CH_BYTE>>1 ) - - -/* Workbuffer RAM - Allocation */ -/* - ++++++++++++++++++++++++++++++++++++++++++++++++++++ - | OFFSET_QMF | OFFSET_NRG | - ++++++++++++++++++++++++++++++++++++++++++++++++++++ - ------------------------- ------------------------- - | | 0.5 * | - | sbr_envRBuffer | sbr_envYBuffer_size | - | sbr_envIBuffer | | - ------------------------- ------------------------- - -*/ - #define BUF_NRG_SIZE ( (MAX_NUM_CHANNELS * Y_2_BUF_BYTE) ) - #define BUF_QMF_SIZE (ENV_R_BUFF_BYTE + ENV_I_BUFF_BYTE) - - /* Size of the shareable memory region than can be reused */ - #define SBR_ENC_DYN_RAM_SIZE ( BUF_QMF_SIZE + BUF_NRG_SIZE ) - - #define OFFSET_QMF ( 0 ) - #define OFFSET_NRG ( OFFSET_QMF + BUF_QMF_SIZE ) - - -/* - ***************************************************************************************************** - */ - - H_ALLOC_MEM(Ram_SbrDynamic_RAM, FIXP_DBL) - - H_ALLOC_MEM(Ram_SbrEncoder, SBR_ENCODER) - H_ALLOC_MEM(Ram_SbrChannel, SBR_CHANNEL) - H_ALLOC_MEM(Ram_SbrElement, SBR_ELEMENT) - - H_ALLOC_MEM(Ram_Sbr_quotaMatrix, FIXP_DBL) - H_ALLOC_MEM(Ram_Sbr_signMatrix, INT) - - H_ALLOC_MEM(Ram_Sbr_QmfStatesAnalysis, FIXP_QAS) - - H_ALLOC_MEM(Ram_Sbr_freqBandTableLO, UCHAR) - H_ALLOC_MEM(Ram_Sbr_freqBandTableHI, UCHAR) - H_ALLOC_MEM(Ram_Sbr_v_k_master, UCHAR) - - H_ALLOC_MEM(Ram_Sbr_detectionVectors, UCHAR) - H_ALLOC_MEM(Ram_Sbr_prevEnvelopeCompensation, UCHAR) - H_ALLOC_MEM(Ram_Sbr_guideScfb, UCHAR) - H_ALLOC_MEM(Ram_Sbr_guideVectorDetected, UCHAR) - - /* Dynamic Memory Allocation */ - - H_ALLOC_MEM(Ram_Sbr_envYBuffer, FIXP_DBL) - FIXP_DBL* GetRam_Sbr_envYBuffer (int n, UCHAR* dynamic_RAM); - FIXP_DBL* GetRam_Sbr_envRBuffer (int n, UCHAR* dynamic_RAM); - FIXP_DBL* GetRam_Sbr_envIBuffer (int n, UCHAR* dynamic_RAM); - - H_ALLOC_MEM(Ram_Sbr_guideVectorDiff, FIXP_DBL) - H_ALLOC_MEM(Ram_Sbr_guideVectorOrig, FIXP_DBL) - - - H_ALLOC_MEM(Ram_PsQmfStatesSynthesis, FIXP_DBL) - - H_ALLOC_MEM(Ram_PsEncode, PS_ENCODE) - - FIXP_DBL* FDKsbrEnc_SliceRam_PsRqmf (FIXP_DBL* rQmfData, UCHAR* dynamic_RAM, int n, int i, int qmfSlots); - FIXP_DBL* FDKsbrEnc_SliceRam_PsIqmf (FIXP_DBL* iQmfData, UCHAR* dynamic_RAM, int n, int i, int qmfSlots); - - H_ALLOC_MEM(Ram_ParamStereo, PARAMETRIC_STEREO) - - - -#endif - diff --git a/libSBRenc/src/sbr_rom.cpp b/libSBRenc/src/sbr_rom.cpp deleted file mode 100644 index 7a51668..0000000 --- a/libSBRenc/src/sbr_rom.cpp +++ /dev/null @@ -1,795 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2015 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 ------------------------------------------------------------------------------------------------------------ */ - -/*! - \file - \brief Definition of constant tables - - - This module contains most of the constant data that can be stored in ROM. -*/ - -#include "sbr_rom.h" -#include "genericStds.h" - -//@{ -/******************************************************************************* - - Table Overview: - - o envelope level, 1.5 dB: - 1a) v_Huff_envelopeLevelC10T[121] - 1b) v_Huff_envelopeLevelL10T[121] - 2a) v_Huff_envelopeLevelC10F[121] - 2b) v_Huff_envelopeLevelL10F[121] - - o envelope balance, 1.5 dB: - 3a) bookSbrEnvBalanceC10T[49] - 3b) bookSbrEnvBalanceL10T[49] - 4a) bookSbrEnvBalanceC10F[49] - 4b) bookSbrEnvBalanceL10F[49] - - o envelope level, 3.0 dB: - 5a) v_Huff_envelopeLevelC11T[63] - 5b) v_Huff_envelopeLevelL11T[63] - 6a) v_Huff_envelopeLevelC11F[63] - 6b) v_Huff_envelopeLevelC11F[63] - - o envelope balance, 3.0 dB: - 7a) bookSbrEnvBalanceC11T[25] - 7b) bookSbrEnvBalanceL11T[25] - 8a) bookSbrEnvBalanceC11F[25] - 8b) bookSbrEnvBalanceL11F[25] - - o noise level, 3.0 dB: - 9a) v_Huff_NoiseLevelC11T[63] - 9b) v_Huff_NoiseLevelL11T[63] - - ) (v_Huff_envelopeLevelC11F[63] is used for freq dir) - - ) (v_Huff_envelopeLevelL11F[63] is used for freq dir) - - o noise balance, 3.0 dB: - 10a) bookSbrNoiseBalanceC11T[25] - 10b) bookSbrNoiseBalanceL11T[25] - - ) (bookSbrEnvBalanceC11F[25] is used for freq dir) - - ) (bookSbrEnvBalanceL11F[25] is used for freq dir) - - - (1.5 dB is never used for noise) - -********************************************************************************/ - - -/*******************************************************************************/ -/* table : envelope level, 1.5 dB */ -/* theor range : [-58,58], CODE_BOOK_SCF_LAV = 58 */ -/* implem range: [-60,60], CODE_BOOK_SCF_LAV10 = 60 */ -/* raw stats : envelopeLevel_00 (yes, wrong suffix in name) KK 01-03-09 */ -/*******************************************************************************/ - -/* direction: time - contents : codewords - raw table: HuffCode3C2FIX.m/envelopeLevel_00T_cF.mat/v_nChex_cF - built by : FH 01-07-05 */ - -const INT v_Huff_envelopeLevelC10T[121] = -{ - 0x0003FFD6, 0x0003FFD7, 0x0003FFD8, 0x0003FFD9, 0x0003FFDA, 0x0003FFDB, 0x0007FFB8, 0x0007FFB9, - 0x0007FFBA, 0x0007FFBB, 0x0007FFBC, 0x0007FFBD, 0x0007FFBE, 0x0007FFBF, 0x0007FFC0, 0x0007FFC1, - 0x0007FFC2, 0x0007FFC3, 0x0007FFC4, 0x0007FFC5, 0x0007FFC6, 0x0007FFC7, 0x0007FFC8, 0x0007FFC9, - 0x0007FFCA, 0x0007FFCB, 0x0007FFCC, 0x0007FFCD, 0x0007FFCE, 0x0007FFCF, 0x0007FFD0, 0x0007FFD1, - 0x0007FFD2, 0x0007FFD3, 0x0001FFE6, 0x0003FFD4, 0x0000FFF0, 0x0001FFE9, 0x0003FFD5, 0x0001FFE7, - 0x0000FFF1, 0x0000FFEC, 0x0000FFED, 0x0000FFEE, 0x00007FF4, 0x00003FF9, 0x00003FF7, 0x00001FFA, - 0x00001FF9, 0x00000FFB, 0x000007FC, 0x000003FC, 0x000001FD, 0x000000FD, 0x0000007D, 0x0000003D, - 0x0000001D, 0x0000000D, 0x00000005, 0x00000001, 0x00000000, 0x00000004, 0x0000000C, 0x0000001C, - 0x0000003C, 0x0000007C, 0x000000FC, 0x000001FC, 0x000003FD, 0x00000FFA, 0x00001FF8, 0x00003FF6, - 0x00003FF8, 0x00007FF5, 0x0000FFEF, 0x0001FFE8, 0x0000FFF2, 0x0007FFD4, 0x0007FFD5, 0x0007FFD6, - 0x0007FFD7, 0x0007FFD8, 0x0007FFD9, 0x0007FFDA, 0x0007FFDB, 0x0007FFDC, 0x0007FFDD, 0x0007FFDE, - 0x0007FFDF, 0x0007FFE0, 0x0007FFE1, 0x0007FFE2, 0x0007FFE3, 0x0007FFE4, 0x0007FFE5, 0x0007FFE6, - 0x0007FFE7, 0x0007FFE8, 0x0007FFE9, 0x0007FFEA, 0x0007FFEB, 0x0007FFEC, 0x0007FFED, 0x0007FFEE, - 0x0007FFEF, 0x0007FFF0, 0x0007FFF1, 0x0007FFF2, 0x0007FFF3, 0x0007FFF4, 0x0007FFF5, 0x0007FFF6, - 0x0007FFF7, 0x0007FFF8, 0x0007FFF9, 0x0007FFFA, 0x0007FFFB, 0x0007FFFC, 0x0007FFFD, 0x0007FFFE, - 0x0007FFFF -}; - - -/* direction: time - contents : codeword lengths - raw table: HuffCode3C2FIX.m/envelopeLevel_00T_cF.mat/v_nLhex_cF - built by : FH 01-07-05 */ - -const UCHAR v_Huff_envelopeLevelL10T[121] = -{ - 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, - 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, - 0x13, 0x13, 0x11, 0x12, 0x10, 0x11, 0x12, 0x11, 0x10, 0x10, 0x10, 0x10, 0x0F, 0x0E, 0x0E, 0x0D, - 0x0D, 0x0C, 0x0B, 0x0A, 0x09, 0x08, 0x07, 0x06, 0x05, 0x04, 0x03, 0x02, 0x02, 0x03, 0x04, 0x05, - 0x06, 0x07, 0x08, 0x09, 0x0A, 0x0C, 0x0D, 0x0E, 0x0E, 0x0F, 0x10, 0x11, 0x10, 0x13, 0x13, 0x13, - 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, - 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, - 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13 -}; - - -/* direction: freq - contents : codewords - raw table: HuffCode3C2FIX.m/envelopeLevel_00F_cF.mat/v_nChex_cF - built by : FH 01-07-05 */ - -const INT v_Huff_envelopeLevelC10F[121] = -{ - 0x0007FFE7, 0x0007FFE8, 0x000FFFD2, 0x000FFFD3, 0x000FFFD4, 0x000FFFD5, 0x000FFFD6, 0x000FFFD7, - 0x000FFFD8, 0x0007FFDA, 0x000FFFD9, 0x000FFFDA, 0x000FFFDB, 0x000FFFDC, 0x0007FFDB, 0x000FFFDD, - 0x0007FFDC, 0x0007FFDD, 0x000FFFDE, 0x0003FFE4, 0x000FFFDF, 0x000FFFE0, 0x000FFFE1, 0x0007FFDE, - 0x000FFFE2, 0x000FFFE3, 0x000FFFE4, 0x0007FFDF, 0x000FFFE5, 0x0007FFE0, 0x0003FFE8, 0x0007FFE1, - 0x0003FFE0, 0x0003FFE9, 0x0001FFEF, 0x0003FFE5, 0x0001FFEC, 0x0001FFED, 0x0001FFEE, 0x0000FFF4, - 0x0000FFF3, 0x0000FFF0, 0x00007FF7, 0x00007FF6, 0x00003FFA, 0x00001FFA, 0x00001FF9, 0x00000FFA, - 0x00000FF8, 0x000007F9, 0x000003FB, 0x000001FC, 0x000001FA, 0x000000FB, 0x0000007C, 0x0000003C, - 0x0000001C, 0x0000000C, 0x00000005, 0x00000001, 0x00000000, 0x00000004, 0x0000000D, 0x0000001D, - 0x0000003D, 0x000000FA, 0x000000FC, 0x000001FB, 0x000003FA, 0x000007F8, 0x000007FA, 0x000007FB, - 0x00000FF9, 0x00000FFB, 0x00001FF8, 0x00001FFB, 0x00003FF8, 0x00003FF9, 0x0000FFF1, 0x0000FFF2, - 0x0001FFEA, 0x0001FFEB, 0x0003FFE1, 0x0003FFE2, 0x0003FFEA, 0x0003FFE3, 0x0003FFE6, 0x0003FFE7, - 0x0003FFEB, 0x000FFFE6, 0x0007FFE2, 0x000FFFE7, 0x000FFFE8, 0x000FFFE9, 0x000FFFEA, 0x000FFFEB, - 0x000FFFEC, 0x0007FFE3, 0x000FFFED, 0x000FFFEE, 0x000FFFEF, 0x000FFFF0, 0x0007FFE4, 0x000FFFF1, - 0x0003FFEC, 0x000FFFF2, 0x000FFFF3, 0x0007FFE5, 0x0007FFE6, 0x000FFFF4, 0x000FFFF5, 0x000FFFF6, - 0x000FFFF7, 0x000FFFF8, 0x000FFFF9, 0x000FFFFA, 0x000FFFFB, 0x000FFFFC, 0x000FFFFD, 0x000FFFFE, - 0x000FFFFF -}; - - -/* direction: freq - contents : codeword lengths - raw table: HuffCode3C2FIX.m/envelopeLevel_00F_cF.mat/v_nLhex_cF - built by : FH 01-07-05 */ - -const UCHAR v_Huff_envelopeLevelL10F[121] = -{ - 0x13, 0x13, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x13, 0x14, 0x14, 0x14, 0x14, 0x13, 0x14, - 0x13, 0x13, 0x14, 0x12, 0x14, 0x14, 0x14, 0x13, 0x14, 0x14, 0x14, 0x13, 0x14, 0x13, 0x12, 0x13, - 0x12, 0x12, 0x11, 0x12, 0x11, 0x11, 0x11, 0x10, 0x10, 0x10, 0x0F, 0x0F, 0x0E, 0x0D, 0x0D, 0x0C, - 0x0C, 0x0B, 0x0A, 0x09, 0x09, 0x08, 0x07, 0x06, 0x05, 0x04, 0x03, 0x02, 0x02, 0x03, 0x04, 0x05, - 0x06, 0x08, 0x08, 0x09, 0x0A, 0x0B, 0x0B, 0x0B, 0x0C, 0x0C, 0x0D, 0x0D, 0x0E, 0x0E, 0x10, 0x10, - 0x11, 0x11, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x14, 0x13, 0x14, 0x14, 0x14, 0x14, 0x14, - 0x14, 0x13, 0x14, 0x14, 0x14, 0x14, 0x13, 0x14, 0x12, 0x14, 0x14, 0x13, 0x13, 0x14, 0x14, 0x14, - 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14 -}; - - -/*******************************************************************************/ -/* table : envelope balance, 1.5 dB */ -/* theor range : [-48,48], CODE_BOOK_SCF_LAV = 48 */ -/* implem range: same but mapped to [-24,24], CODE_BOOK_SCF_LAV_BALANCE10 = 24 */ -/* raw stats : envelopePan_00 (yes, wrong suffix in name) KK 01-03-09 */ -/*******************************************************************************/ - -/* direction: time - contents : codewords - raw table: HuffCode3C.m/envelopePan_00T.mat/v_nBhex - built by : FH 01-05-15 */ - -const INT bookSbrEnvBalanceC10T[49] = -{ - 0x0000FFE4, 0x0000FFE5, 0x0000FFE6, 0x0000FFE7, 0x0000FFE8, 0x0000FFE9, 0x0000FFEA, 0x0000FFEB, - 0x0000FFEC, 0x0000FFED, 0x0000FFEE, 0x0000FFEF, 0x0000FFF0, 0x0000FFF1, 0x0000FFF2, 0x0000FFF3, - 0x0000FFF4, 0x0000FFE2, 0x00000FFC, 0x000007FC, 0x000001FE, 0x0000007E, 0x0000001E, 0x00000006, - 0x00000000, 0x00000002, 0x0000000E, 0x0000003E, 0x000000FE, 0x000007FD, 0x00000FFD, 0x00007FF0, - 0x0000FFE3, 0x0000FFF5, 0x0000FFF6, 0x0000FFF7, 0x0000FFF8, 0x0000FFF9, 0x0000FFFA, 0x0001FFF6, - 0x0001FFF7, 0x0001FFF8, 0x0001FFF9, 0x0001FFFA, 0x0001FFFB, 0x0001FFFC, 0x0001FFFD, 0x0001FFFE, - 0x0001FFFF -}; - - -/* direction: time - contents : codeword lengths - raw table: HuffCode3C.m/envelopePan_00T.mat/v_nLhex - built by : FH 01-05-15 */ - -const UCHAR bookSbrEnvBalanceL10T[49] = -{ - 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, - 0x10, 0x10, 0x0C, 0x0B, 0x09, 0x07, 0x05, 0x03, 0x01, 0x02, 0x04, 0x06, 0x08, 0x0B, 0x0C, 0x0F, - 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x11, 0x11, 0x11, 0x11, 0x11, 0x11, 0x11, 0x11, 0x11, - 0x11 -}; - - -/* direction: freq - contents : codewords - raw table: HuffCode3C.m/envelopePan_00F.mat/v_nBhex - built by : FH 01-05-15 */ - -const INT bookSbrEnvBalanceC10F[49] = -{ - 0x0003FFE2, 0x0003FFE3, 0x0003FFE4, 0x0003FFE5, 0x0003FFE6, 0x0003FFE7, 0x0003FFE8, 0x0003FFE9, - 0x0003FFEA, 0x0003FFEB, 0x0003FFEC, 0x0003FFED, 0x0003FFEE, 0x0003FFEF, 0x0003FFF0, 0x0000FFF7, - 0x0001FFF0, 0x00003FFC, 0x000007FE, 0x000007FC, 0x000000FE, 0x0000007E, 0x0000000E, 0x00000002, - 0x00000000, 0x00000006, 0x0000001E, 0x0000003E, 0x000001FE, 0x000007FD, 0x00000FFE, 0x00007FFA, - 0x0000FFF6, 0x0003FFF1, 0x0003FFF2, 0x0003FFF3, 0x0003FFF4, 0x0003FFF5, 0x0003FFF6, 0x0003FFF7, - 0x0003FFF8, 0x0003FFF9, 0x0003FFFA, 0x0003FFFB, 0x0003FFFC, 0x0003FFFD, 0x0003FFFE, 0x0007FFFE, - 0x0007FFFF -}; - - -/* direction: freq - contents : codeword lengths - raw table: HuffCode3C.m/envelopePan_00F.mat/v_nLhex - built by : FH 01-05-15 */ - -const UCHAR bookSbrEnvBalanceL10F[49] = -{ - 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x10, - 0x11, 0x0E, 0x0B, 0x0B, 0x08, 0x07, 0x04, 0x02, 0x01, 0x03, 0x05, 0x06, 0x09, 0x0B, 0x0C, 0x0F, - 0x10, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x13, - 0x13 -}; - - -/*******************************************************************************/ -/* table : envelope level, 3.0 dB */ -/* theor range : [-29,29], CODE_BOOK_SCF_LAV = 29 */ -/* implem range: [-31,31], CODE_BOOK_SCF_LAV11 = 31 */ -/* raw stats : envelopeLevel_11 KK 00-02-03 */ -/*******************************************************************************/ - -/* direction: time - contents : codewords - raw table: HuffCode2.m - built by : FH 00-02-04 */ - -const INT v_Huff_envelopeLevelC11T[63] = { - 0x0003FFED, 0x0003FFEE, 0x0007FFDE, 0x0007FFDF, 0x0007FFE0, 0x0007FFE1, 0x0007FFE2, 0x0007FFE3, - 0x0007FFE4, 0x0007FFE5, 0x0007FFE6, 0x0007FFE7, 0x0007FFE8, 0x0007FFE9, 0x0007FFEA, 0x0007FFEB, - 0x0007FFEC, 0x0001FFF4, 0x0000FFF7, 0x0000FFF9, 0x0000FFF8, 0x00003FFB, 0x00003FFA, 0x00003FF8, - 0x00001FFA, 0x00000FFC, 0x000007FC, 0x000000FE, 0x0000003E, 0x0000000E, 0x00000002, 0x00000000, - 0x00000006, 0x0000001E, 0x0000007E, 0x000001FE, 0x000007FD, 0x00001FFB, 0x00003FF9, 0x00003FFC, - 0x00007FFA, 0x0000FFF6, 0x0001FFF5, 0x0003FFEC, 0x0007FFED, 0x0007FFEE, 0x0007FFEF, 0x0007FFF0, - 0x0007FFF1, 0x0007FFF2, 0x0007FFF3, 0x0007FFF4, 0x0007FFF5, 0x0007FFF6, 0x0007FFF7, 0x0007FFF8, - 0x0007FFF9, 0x0007FFFA, 0x0007FFFB, 0x0007FFFC, 0x0007FFFD, 0x0007FFFE, 0x0007FFFF -}; - - -/* direction: time - contents : codeword lengths - raw table: HuffCode2.m - built by : FH 00-02-04 */ - -const UCHAR v_Huff_envelopeLevelL11T[63] = { - 0x12, 0x12, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, - 0x13, 0x11, 0x10, 0x10, 0x10, 0x0E, 0x0E, 0x0E, 0x0D, 0x0C, 0x0B, 0x08, 0x06, 0x04, 0x02, 0x01, - 0x03, 0x05, 0x07, 0x09, 0x0B, 0x0D, 0x0E, 0x0E, 0x0F, 0x10, 0x11, 0x12, 0x13, 0x13, 0x13, 0x13, - 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13 -}; - - -/* direction: freq - contents : codewords - raw table: HuffCode2.m - built by : FH 00-02-04 */ - -const INT v_Huff_envelopeLevelC11F[63] = { - 0x000FFFF0, 0x000FFFF1, 0x000FFFF2, 0x000FFFF3, 0x000FFFF4, 0x000FFFF5, 0x000FFFF6, 0x0003FFF3, - 0x0007FFF5, 0x0007FFEE, 0x0007FFEF, 0x0007FFF6, 0x0003FFF4, 0x0003FFF2, 0x000FFFF7, 0x0007FFF0, - 0x0001FFF5, 0x0003FFF0, 0x0001FFF4, 0x0000FFF7, 0x0000FFF6, 0x00007FF8, 0x00003FFB, 0x00000FFD, - 0x000007FD, 0x000003FD, 0x000001FD, 0x000000FD, 0x0000003E, 0x0000000E, 0x00000002, 0x00000000, - 0x00000006, 0x0000001E, 0x000000FC, 0x000001FC, 0x000003FC, 0x000007FC, 0x00000FFC, 0x00001FFC, - 0x00003FFA, 0x00007FF9, 0x00007FFA, 0x0000FFF8, 0x0000FFF9, 0x0001FFF6, 0x0001FFF7, 0x0003FFF5, - 0x0003FFF6, 0x0003FFF1, 0x000FFFF8, 0x0007FFF1, 0x0007FFF2, 0x0007FFF3, 0x000FFFF9, 0x0007FFF7, - 0x0007FFF4, 0x000FFFFA, 0x000FFFFB, 0x000FFFFC, 0x000FFFFD, 0x000FFFFE, 0x000FFFFF -}; - - -/* direction: freq - contents : codeword lengths - raw table: HuffCode2.m - built by : FH 00-02-04 */ - -const UCHAR v_Huff_envelopeLevelL11F[63] = { - 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x12, 0x13, 0x13, 0x13, 0x13, 0x12, 0x12, 0x14, 0x13, - 0x11, 0x12, 0x11, 0x10, 0x10, 0x0F, 0x0E, 0x0C, 0x0B, 0x0A, 0x09, 0x08, 0x06, 0x04, 0x02, 0x01, - 0x03, 0x05, 0x08, 0x09, 0x0A, 0x0B, 0x0C, 0x0D, 0x0E, 0x0F, 0x0F, 0x10, 0x10, 0x11, 0x11, 0x12, - 0x12, 0x12, 0x14, 0x13, 0x13, 0x13, 0x14, 0x13, 0x13, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14 -}; - - - -/*******************************************************************************/ -/* table : envelope balance, 3.0 dB */ -/* theor range : [-24,24], CODE_BOOK_SCF_LAV = 24 */ -/* implem range: same but mapped to [-12,12], CODE_BOOK_SCF_LAV_BALANCE11 = 12 */ -/* raw stats : envelopeBalance_11 KK 00-02-03 */ -/*******************************************************************************/ - -/* direction: time - contents : codewords - raw table: HuffCode3C.m/envelopeBalance_11T.mat/v_nBhex - built by : FH 01-05-15 */ - -const INT bookSbrEnvBalanceC11T[25] = -{ - 0x00001FF2, 0x00001FF3, 0x00001FF4, 0x00001FF5, 0x00001FF6, 0x00001FF7, 0x00001FF8, 0x00000FF8, - 0x000000FE, 0x0000007E, 0x0000000E, 0x00000006, 0x00000000, 0x00000002, 0x0000001E, 0x0000003E, - 0x000001FE, 0x00001FF9, 0x00001FFA, 0x00001FFB, 0x00001FFC, 0x00001FFD, 0x00001FFE, 0x00003FFE, - 0x00003FFF -}; - - -/* direction: time - contents : codeword lengths - raw table: HuffCode3C.m/envelopeBalance_11T.mat/v_nLhex - built by : FH 01-05-15 */ - -const UCHAR bookSbrEnvBalanceL11T[25] = -{ - 0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0C, 0x08, 0x07, 0x04, 0x03, 0x01, 0x02, 0x05, 0x06, - 0x09, 0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0E, 0x0E -}; - - -/* direction: freq - contents : codewords - raw table: HuffCode3C.m/envelopeBalance_11F.mat/v_nBhex - built by : FH 01-05-15 */ - -const INT bookSbrEnvBalanceC11F[25] = -{ - 0x00001FF7, 0x00001FF8, 0x00001FF9, 0x00001FFA, 0x00001FFB, 0x00003FF8, 0x00003FF9, 0x000007FC, - 0x000000FE, 0x0000007E, 0x0000000E, 0x00000002, 0x00000000, 0x00000006, 0x0000001E, 0x0000003E, - 0x000001FE, 0x00000FFA, 0x00001FF6, 0x00003FFA, 0x00003FFB, 0x00003FFC, 0x00003FFD, 0x00003FFE, - 0x00003FFF -}; - - -/* direction: freq - contents : codeword lengths - raw table: HuffCode3C.m/envelopeBalance_11F.mat/v_nLhex - built by : FH 01-05-15 */ - -const UCHAR bookSbrEnvBalanceL11F[25] = -{ - 0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0E, 0x0E, 0x0B, 0x08, 0x07, 0x04, 0x02, 0x01, 0x03, 0x05, 0x06, - 0x09, 0x0C, 0x0D, 0x0E, 0x0E, 0x0E, 0x0E, 0x0E, 0x0E -}; - - -/*******************************************************************************/ -/* table : noise level, 3.0 dB */ -/* theor range : [-29,29], CODE_BOOK_SCF_LAV = 29 */ -/* implem range: [-31,31], CODE_BOOK_SCF_LAV11 = 31 */ -/* raw stats : noiseLevel_11 KK 00-02-03 */ -/*******************************************************************************/ - -/* direction: time - contents : codewords - raw table: HuffCode2.m - built by : FH 00-02-04 */ - -const INT v_Huff_NoiseLevelC11T[63] = { - 0x00001FCE, 0x00001FCF, 0x00001FD0, 0x00001FD1, 0x00001FD2, 0x00001FD3, 0x00001FD4, 0x00001FD5, - 0x00001FD6, 0x00001FD7, 0x00001FD8, 0x00001FD9, 0x00001FDA, 0x00001FDB, 0x00001FDC, 0x00001FDD, - 0x00001FDE, 0x00001FDF, 0x00001FE0, 0x00001FE1, 0x00001FE2, 0x00001FE3, 0x00001FE4, 0x00001FE5, - 0x00001FE6, 0x00001FE7, 0x000007F2, 0x000000FD, 0x0000003E, 0x0000000E, 0x00000006, 0x00000000, - 0x00000002, 0x0000001E, 0x000000FC, 0x000003F8, 0x00001FCC, 0x00001FE8, 0x00001FE9, 0x00001FEA, - 0x00001FEB, 0x00001FEC, 0x00001FCD, 0x00001FED, 0x00001FEE, 0x00001FEF, 0x00001FF0, 0x00001FF1, - 0x00001FF2, 0x00001FF3, 0x00001FF4, 0x00001FF5, 0x00001FF6, 0x00001FF7, 0x00001FF8, 0x00001FF9, - 0x00001FFA, 0x00001FFB, 0x00001FFC, 0x00001FFD, 0x00001FFE, 0x00003FFE, 0x00003FFF -}; - - -/* direction: time - contents : codeword lengths - raw table: HuffCode2.m - built by : FH 00-02-04 */ - -const UCHAR v_Huff_NoiseLevelL11T[63] = { - 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, - 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, - 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, - 0x0000000D, 0x0000000D, 0x0000000B, 0x00000008, 0x00000006, 0x00000004, 0x00000003, 0x00000001, - 0x00000002, 0x00000005, 0x00000008, 0x0000000A, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, - 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, - 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, - 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000E, 0x0000000E -}; - - -/*******************************************************************************/ -/* table : noise balance, 3.0 dB */ -/* theor range : [-24,24], CODE_BOOK_SCF_LAV = 24 */ -/* implem range: same but mapped to [-12,12], CODE_BOOK_SCF_LAV_BALANCE11 = 12 */ -/* raw stats : noiseBalance_11 KK 00-02-03 */ -/*******************************************************************************/ - -/* direction: time - contents : codewords - raw table: HuffCode3C.m/noiseBalance_11.mat/v_nBhex - built by : FH 01-05-15 */ - -const INT bookSbrNoiseBalanceC11T[25] = -{ - 0x000000EC, 0x000000ED, 0x000000EE, 0x000000EF, 0x000000F0, 0x000000F1, 0x000000F2, 0x000000F3, - 0x000000F4, 0x000000F5, 0x0000001C, 0x00000002, 0x00000000, 0x00000006, 0x0000003A, 0x000000F6, - 0x000000F7, 0x000000F8, 0x000000F9, 0x000000FA, 0x000000FB, 0x000000FC, 0x000000FD, 0x000000FE, - 0x000000FF -}; - - -/* direction: time - contents : codeword lengths - raw table: HuffCode3C.m/noiseBalance_11.mat/v_nLhex - built by : FH 01-05-15 */ - -const UCHAR bookSbrNoiseBalanceL11T[25] = -{ - 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x05, 0x02, 0x01, 0x03, 0x06, 0x08, - 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08 -}; - -/* - tuningTable -*/ -const sbrTuningTable_t sbrTuningTable[] = -{ - /* Some of the low bitrates are commented out here, this is because the - encoder could lose frames at those bitrates and throw an error because - it has insufficient bits to encode for some test items. - */ - - /*** HE-AAC section ***/ - /* sf,sfsp,sf,sfsp,nnb,nfo,saml,SM,FS*/ - - /*** mono ***/ - - /* 8/16 kHz dual rate */ - { CODEC_AAC, 8000, 10000, 8000, 1, 7, 6, 11,10, 1, 0, 6, SBR_MONO, 3 }, - { CODEC_AAC, 10000, 12000, 8000, 1, 11, 7, 13,12, 1, 0, 6, SBR_MONO, 3 }, - { CODEC_AAC, 12000, 16001, 8000, 1, 14,10, 13,13, 1, 0, 6, SBR_MONO, 3 }, - { CODEC_AAC, 16000, 24000, 8000, 1, 14,10, 14,14, 2, 0, 3, SBR_MONO, 2 }, /* placebo */ - { CODEC_AAC, 24000, 32000, 8000, 1, 14,10, 14,14, 2, 0, 3, SBR_MONO, 2 }, /* placebo */ - { CODEC_AAC, 32000, 48001, 8000, 1, 14,11, 15,15, 2, 0, 3, SBR_MONO, 2 }, /* placebo */ /* bitrates higher than 48000 not supported by AAC core */ - - /* 11/22 kHz dual rate */ - { CODEC_AAC, 8000, 10000, 11025, 1, 5, 4, 6, 6, 1, 0, 6, SBR_MONO, 3 }, - { CODEC_AAC, 10000, 12000, 11025, 1, 8, 5, 12, 9, 1, 0, 6, SBR_MONO, 3 }, - { CODEC_AAC, 12000, 16000, 11025, 1, 12, 8, 13, 8, 1, 0, 6, SBR_MONO, 3 }, - { CODEC_AAC, 16000, 20000, 11025, 1, 12, 8, 13, 8, 1, 0, 6, SBR_MONO, 3 }, /* at such "high" bitrates it's better to upsample the input */ - { CODEC_AAC, 20000, 24001, 11025, 1, 13, 9, 13, 8, 1, 0, 6, SBR_MONO, 3 }, /* signal by a factor of 2 before sending it into the encoder */ - { CODEC_AAC, 24000, 32000, 11025, 1, 14,10, 14, 9, 2, 0, 3, SBR_MONO, 2 }, /* placebo */ - { CODEC_AAC, 32000, 48000, 11025, 1, 15,11, 15,10, 2, 0, 3, SBR_MONO, 2 }, /* placebo */ - { CODEC_AAC, 48000, 64001, 11025, 1, 15,11, 15,10, 2, 0, 3, SBR_MONO, 1 }, /* placebo */ - - /* 12/24 kHz dual rate */ - { CODEC_AAC, 8000, 10000, 12000, 1, 4, 3, 6, 6, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 8 kbit/s */ - { CODEC_AAC, 10000, 12000, 12000, 1, 7, 4, 11, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 10 kbit/s */ - { CODEC_AAC, 12000, 16000, 12000, 1, 11, 7, 12, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */ - { CODEC_AAC, 16000, 20000, 12000, 1, 11, 7, 12, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */ /* at such "high" bitrates it's better to upsample the input */ - { CODEC_AAC, 20000, 24001, 12000, 1, 12, 8, 12, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 20 kbit/s */ /* signal by a factor of 2 before sending it into the encoder */ - { CODEC_AAC, 24000, 32000, 12000, 1, 13, 9, 13, 9, 2, 0, 3, SBR_MONO, 2 }, /* placebo */ - { CODEC_AAC, 32000, 48000, 12000, 1, 14,10, 14,10, 2, 0, 3, SBR_MONO, 2 }, /* placebo */ - { CODEC_AAC, 48000, 64001, 12000, 1, 14,11, 15,11, 2, 0, 3, SBR_MONO, 1 }, /* placebo */ - - /* 16/32 kHz dual rate */ - { CODEC_AAC, 8000, 10000, 16000, 1, 1, 1, 0, 0, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 8 kbit/s */ - { CODEC_AAC, 10000, 12000, 16000, 1, 2, 1, 6, 0, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 10 kbit/s */ - { CODEC_AAC, 12000, 16000, 16000, 1, 4, 2, 6, 0, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */ - { CODEC_AAC, 16000, 18000, 16000, 1, 4, 2, 8, 3, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */ - { CODEC_AAC, 18000, 22000, 16000, 1, 6, 5,11, 7, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */ - { CODEC_AAC, 22000, 28000, 16000, 1, 10, 9,12, 8, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */ - { CODEC_AAC, 28000, 36000, 16000, 1, 12,12,13,13, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AAC, 36000, 44000, 16000, 1, 14,14,13,13, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */ - { CODEC_AAC, 44000, 64001, 16000, 1, 14,14,13,13, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */ - - /* 22.05/44.1 kHz dual rate */ - /* { CODEC_AAC, 8000, 11369, 22050, 1, 1, 1, 1, 1, 1, 0, 6, SBR_MONO, 3 }, */ /* nominal: 8 kbit/s */ /* encoder can not work stable at this extremely low bitrate */ - { CODEC_AAC, 11369, 16000, 22050, 1, 3, 1, 4, 4, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */ - { CODEC_AAC, 16000, 18000, 22050, 1, 3, 1, 5, 4, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */ - { CODEC_AAC, 18000, 22000, 22050, 1, 4, 4, 8, 5, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */ - { CODEC_AAC, 22000, 28000, 22050, 1, 7, 6, 8, 6, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */ - { CODEC_AAC, 28000, 36000, 22050, 1, 10,10, 9, 9, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AAC, 36000, 44000, 22050, 1, 11,11,10,10, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */ - { CODEC_AAC, 44000, 64001, 22050, 1, 13,13,12,12, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */ - - /* 24/48 kHz dual rate */ - /* { CODEC_AAC, 8000, 12000, 24000, 1, 1, 1, 1, 1, 1, 0, 6, SBR_MONO, 3 }, */ /* nominal: 8 kbit/s */ /* encoder can not work stable at this extremely low bitrate */ - { CODEC_AAC, 12000, 16000, 24000, 1, 3, 1, 4, 4, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */ - { CODEC_AAC, 16000, 18000, 24000, 1, 3, 1, 5, 4, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */ - { CODEC_AAC, 18000, 22000, 24000, 1, 4, 3, 8, 5, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */ - { CODEC_AAC, 22000, 28000, 24000, 1, 7, 6, 8, 6, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */ - { CODEC_AAC, 28000, 36000, 24000, 1, 10,10, 9, 9, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AAC, 36000, 44000, 24000, 1, 11,11,10,10, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */ - { CODEC_AAC, 44000, 64001, 24000, 1, 13,13,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */ - - /* 32/64 kHz dual rate */ /* placebo settings */ - { CODEC_AAC, 24000, 36000, 32000, 1, 4, 4, 4, 4, 2, 0, 3, SBR_MONO, 3 }, /* lowest range */ - { CODEC_AAC, 36000, 60000, 32000, 1, 7, 7, 6, 6, 2, 0, 3, SBR_MONO, 2 }, /* lowest range */ - { CODEC_AAC, 60000, 72000, 32000, 1, 9, 9, 8, 8, 2, 0, 3, SBR_MONO, 1 }, /* low range */ - { CODEC_AAC, 72000,100000, 32000, 1, 11,11,10,10, 2, 0, 3, SBR_MONO, 1 }, /* SBR sweet spot */ - { CODEC_AAC, 100000,160001, 32000, 1, 13,13,11,11, 2, 0, 3, SBR_MONO, 1 }, /* backwards compatible */ - - /* 44.1/88.2 kHz dual rate */ /* placebo settings */ - { CODEC_AAC, 24000, 36000, 44100, 1, 4, 4, 4, 4, 2, 0, 3, SBR_MONO, 3 }, /* lowest range (multichannel rear) */ - { CODEC_AAC, 36000, 60000, 44100, 1, 7, 7, 6, 6, 2, 0, 3, SBR_MONO, 2 }, /* lowest range (multichannel rear) */ - { CODEC_AAC, 60000, 72000, 44100, 1, 9, 9, 8, 8, 2, 0, 3, SBR_MONO, 1 }, /* low range */ - { CODEC_AAC, 72000,100000, 44100, 1, 11,11,10,10, 2, 0, 3, SBR_MONO, 1 }, /* SBR sweet spot */ - { CODEC_AAC, 100000,160001, 44100, 1, 13,13,11,11, 2, 0, 3, SBR_MONO, 1 }, /* backwards compatible */ - - /* 48/96 kHz dual rate */ /* not yet finally tuned */ - { CODEC_AAC, 32000, 36000, 48000, 1, 4, 4, 9, 9, 2, 0, 3, SBR_MONO, 3 }, /* lowest range (multichannel rear) */ - { CODEC_AAC, 36000, 60000, 48000, 1, 7, 7,10,10, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 40 */ - { CODEC_AAC, 60000, 72000, 48000, 1, 9, 9,10,10, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 64 */ - { CODEC_AAC, 72000,100000, 48000, 1, 11,11,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 80 */ - { CODEC_AAC, 100000,160001, 48000, 1, 13,13,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 128 */ - - /*** stereo ***/ - /* 08/16 kHz dual rate */ - { CODEC_AAC, 16000, 24000, 8000, 2, 6, 6, 9, 7, 1, 0,-3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ /* placebo */ - { CODEC_AAC, 24000, 28000, 8000, 2, 9, 9, 11, 9, 1, 0,-3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */ - { CODEC_AAC, 28000, 36000, 8000, 2, 11, 9, 11, 9, 2, 0,-3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AAC, 36000, 44000, 8000, 2, 13,11, 13,11, 2, 0,-3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */ - { CODEC_AAC, 44000, 52000, 8000, 2, 14,12, 13,12, 2, 0,-3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */ - { CODEC_AAC, 52000, 60000, 8000, 2, 14,14, 13,13, 3, 0,-3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */ - { CODEC_AAC, 60000, 76000, 8000, 2, 14,14, 13,13, 3, 0,-3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */ - { CODEC_AAC, 76000,128001, 8000, 2, 14,14, 13,13, 3, 0,-3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */ - - /* 11/22 kHz dual rate */ - { CODEC_AAC, 16000, 24000, 11025, 2, 7, 5, 9, 7, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ /* placebo */ - { CODEC_AAC, 24000, 28000, 11025, 2, 10, 8,10, 8, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */ - { CODEC_AAC, 28000, 36000, 11025, 2, 12, 8,12, 8, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AAC, 36000, 44000, 11025, 2, 13, 9,13, 9, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */ - { CODEC_AAC, 44000, 52000, 11025, 2, 14,11,13,11, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */ - { CODEC_AAC, 52000, 60000, 11025, 2, 15,15,13,13, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */ - { CODEC_AAC, 60000, 76000, 11025, 2, 15,15,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */ - { CODEC_AAC, 76000,128001, 11025, 2, 15,15,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */ - - /* 12/24 kHz dual rate */ - { CODEC_AAC, 16000, 24000, 12000, 2, 6, 4, 9, 7, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ /* placebo */ - { CODEC_AAC, 24000, 28000, 12000, 2, 9, 7,10, 8, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */ - { CODEC_AAC, 28000, 36000, 12000, 2, 11, 7,12, 8, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AAC, 36000, 44000, 12000, 2, 12, 9,12, 9, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */ - { CODEC_AAC, 44000, 52000, 12000, 2, 13,12,13,12, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */ - { CODEC_AAC, 52000, 60000, 12000, 2, 14,14,13,13, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */ - { CODEC_AAC, 60000, 76000, 12000, 2, 14,14,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */ - { CODEC_AAC, 76000,128001, 12000, 2, 14,14,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */ - - /* 16/32 kHz dual rate */ - { CODEC_AAC, 16000, 24000, 16000, 2, 4, 2, 1, 0, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ - { CODEC_AAC, 24000, 28000, 16000, 2, 8, 7,10, 8, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */ - { CODEC_AAC, 28000, 36000, 16000, 2, 10, 9,12,11, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AAC, 36000, 44000, 16000, 2, 13,13,13,13, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */ - { CODEC_AAC, 44000, 52000, 16000, 2, 14,14,13,13, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */ - { CODEC_AAC, 52000, 60000, 16000, 2, 14,14,13,13, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */ - { CODEC_AAC, 60000, 76000, 16000, 2, 14,14,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */ - { CODEC_AAC, 76000,128001, 16000, 2, 14,14,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */ - - /* 22.05/44.1 kHz dual rate */ - { CODEC_AAC, 16000, 24000, 22050, 2, 2, 1, 1, 0, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ - { CODEC_AAC, 24000, 28000, 22050, 2, 5, 4, 6, 5, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */ - { CODEC_AAC, 28000, 32000, 22050, 2, 5, 4, 8, 7, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 28 kbit/s */ - { CODEC_AAC, 32000, 36000, 22050, 2, 7, 6, 8, 7, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AAC, 36000, 44000, 22050, 2, 10,10, 9, 9, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */ - { CODEC_AAC, 44000, 52000, 22050, 2, 12,12, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */ - { CODEC_AAC, 52000, 60000, 22050, 2, 13,13,10,10, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */ - { CODEC_AAC, 60000, 76000, 22050, 2, 14,14,12,12, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */ - { CODEC_AAC, 76000,128001, 22050, 2, 14,14,12,12, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */ - - /* 24/48 kHz dual rate */ - { CODEC_AAC, 16000, 24000, 24000, 2, 2, 1, 1, 0, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ - { CODEC_AAC, 24000, 28000, 24000, 2, 5, 5, 6, 6, 1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */ - { CODEC_AAC, 28000, 36000, 24000, 2, 7, 6, 8, 7, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AAC, 36000, 44000, 24000, 2, 10,10, 9, 9, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */ - { CODEC_AAC, 44000, 52000, 24000, 2, 12,12, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */ - { CODEC_AAC, 52000, 60000, 24000, 2, 13,13,10,10, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */ - { CODEC_AAC, 60000, 76000, 24000, 2, 14,14,12,12, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */ - { CODEC_AAC, 76000,128001, 24000, 2, 14,14,12,12, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */ - - /* 32/64 kHz dual rate */ /* placebo settings */ - { CODEC_AAC, 32000, 60000, 32000, 2, 4, 4, 4, 4, 2, 0, -3, SBR_SWITCH_LRC, 3 }, /* lowest range (multichannel rear) */ - { CODEC_AAC, 60000, 80000, 32000, 2, 7, 7, 6, 6, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* lowest range (multichannel rear) */ - { CODEC_AAC, 80000,112000, 32000, 2, 9, 9, 8, 8, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* low range */ - { CODEC_AAC, 112000,144000, 32000, 2, 11,11,10,10, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* SBR sweet spot */ - { CODEC_AAC, 144000,256001, 32000, 2, 13,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* backwards compatible */ - - /* 44.1/88.2 kHz dual rate */ /* placebo settings */ - { CODEC_AAC, 32000, 60000, 44100, 2, 4, 4, 4, 4, 2, 0, -3, SBR_SWITCH_LRC, 3 }, /* lowest range (multichannel rear) */ - { CODEC_AAC, 60000, 80000, 44100, 2, 7, 7, 6, 6, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* lowest range (multichannel rear) */ - { CODEC_AAC, 80000,112000, 44100, 2, 9, 9, 8, 8, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* low range */ - { CODEC_AAC, 112000,144000, 44100, 2, 11,11,10,10, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* SBR sweet spot */ - { CODEC_AAC, 144000,256001, 44100, 2, 13,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* backwards compatible */ - - /* 48/96 kHz dual rate */ /* not yet finally tuned */ - { CODEC_AAC, 36000, 60000, 48000, 2, 4, 4, 9, 9, 2, 0, -3, SBR_SWITCH_LRC, 3 }, /* lowest range (multichannel rear) */ - { CODEC_AAC, 60000, 80000, 48000, 2, 7, 7, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 64 */ - { CODEC_AAC, 80000,112000, 48000, 2, 9, 9,10,10, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 96 */ - { CODEC_AAC, 112000,144000, 48000, 2, 11,11,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 128 */ - { CODEC_AAC, 144000,256001, 48000, 2, 13,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 192 */ - - - /** AAC LOW DELAY SECTION **/ - - /* 24 kHz dual rate - 12kHz singlerate is not allowed (deactivated in FDKsbrEnc_IsSbrSettingAvail()) */ - { CODEC_AACLD, 8000, 32000, 12000, 1, 1, 1, 0, 0, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 8 kbit/s */ - - /*** mono ***/ - /* 16/32 kHz dual rate not yet tuned ->alb copied from non LD tables*/ - { CODEC_AACLD, 16000, 18000, 16000, 1, 4, 5, 9, 7, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s wrr: tuned */ - { CODEC_AACLD, 18000, 22000, 16000, 1, 7, 7,12,12, 1, 6, 9, SBR_MONO, 3 }, /* nominal: 20 kbit/s wrr: tuned */ - { CODEC_AACLD, 22000, 28000, 16000, 1, 6, 6, 9, 9, 2, 3, 6, SBR_MONO, 3 }, /* nominal: 24 kbit/s wrr: tuned */ - { CODEC_AACLD, 28000, 36000, 16000, 1, 8, 8,12, 7, 2, 9,12, SBR_MONO, 3 }, /* jgr: special */ /* wrr: tuned */ - { CODEC_AACLD, 36000, 44000, 16000, 1, 10,14,12,13, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */ - { CODEC_AACLD, 44000, 64001, 16000, 1, 11,14,13,13, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */ - - /* 22.05/44.1 kHz dual rate */ - { CODEC_AACLD, 18000, 22000, 22050, 1, 4, 4, 5, 5, 2, 0, 6, SBR_MONO, 3 }, /* nominal: 20 kbit/s */ - { CODEC_AACLD, 22000, 28000, 22050, 1, 5, 5, 6, 6, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */ - { CODEC_AACLD, 28000, 36000, 22050, 1, 7, 8, 8, 8, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AACLD, 36000, 44000, 22050, 1, 9, 9, 9, 9, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */ - { CODEC_AACLD, 44000, 52000, 22050, 1, 12,11,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */ - { CODEC_AACLD, 52000, 64001, 22050, 1, 13,11,11,10, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 56 kbit/s */ - - /* 24/48 kHz dual rate */ - { CODEC_AACLD, 20000, 22000, 24000, 1, 3, 4, 8, 8, 2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */ - { CODEC_AACLD, 22000, 28000, 24000, 1, 3, 8, 8, 7, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 24 kbit/s */ - { CODEC_AACLD, 28000, 36000, 24000, 1, 4, 8, 8, 7, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AACLD, 36000, 56000, 24000, 1, 8, 9, 9, 8, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */ - { CODEC_AACLD, 56000, 64001, 24000, 1, 13,11,11,10, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 64 kbit/s */ - - /* 32/64 kHz dual rate */ /* placebo settings */ /*jgr: new, copy from CODEC_AAC */ - { CODEC_AACLD, 24000, 36000, 32000, 1, 4, 4, 4, 4, 2, 0, 3, SBR_MONO, 3 }, /* lowest range */ - { CODEC_AACLD, 36000, 60000, 32000, 1, 7, 7, 6, 6, 2, 0, 3, SBR_MONO, 2 }, /* lowest range */ - { CODEC_AACLD, 60000, 72000, 32000, 1, 9, 9, 8, 8, 2, 0, 3, SBR_MONO, 1 }, /* low range */ - { CODEC_AACLD, 72000,100000, 32000, 1, 11,11,10,10, 2, 0, 3, SBR_MONO, 1 }, /* SBR sweet spot */ - { CODEC_AACLD, 100000,160001, 32000, 1, 13,13,11,11, 2, 0, 3, SBR_MONO, 1 }, /* backwards compatible */ - - /* 44/88 kHz dual rate */ /* not yet finally tuned */ - { CODEC_AACLD, 36000, 60000, 44100, 1, 8, 7, 6, 9, 2, 0, 3, SBR_MONO, 2 }, /* nominal: 40 */ - { CODEC_AACLD, 60000, 72000, 44100, 1, 9, 9,10,10, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 64 */ - { CODEC_AACLD, 72000,100000, 44100, 1, 11,11,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 80 */ - { CODEC_AACLD, 100000,160001, 44100, 1, 13,13,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 128 */ - - /* 48/96 kHz dual rate */ /* 32 and 40kbps line tuned for dual-rate SBR */ - { CODEC_AACLD, 36000, 60000, 48000, 1, 4, 7, 4, 4, 2, 0, 3, SBR_MONO, 3 }, /* nominal: 40 */ - { CODEC_AACLD, 60000, 72000, 48000, 1, 9, 9,10,10, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 64 */ - { CODEC_AACLD, 72000,100000, 48000, 1, 11,11,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 80 */ - { CODEC_AACLD, 100000,160001, 48000, 1, 13,13,11,11, 2, 0, 3, SBR_MONO, 1 }, /* nominal: 128 */ - - /*** stereo ***/ - /* 16/32 kHz dual rate not yet tuned ->alb copied from non LD tables*/ - { CODEC_AACLD, 32000, 36000, 16000, 2, 10, 9,12,11, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AACLD, 36000, 44000, 16000, 2, 13,13,13,13, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */ - { CODEC_AACLD, 44000, 52000, 16000, 2, 10, 9,11, 9, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* tune12 nominal: 48 kbit/s */ - { CODEC_AACLD, 52000, 60000, 16000, 2, 14,14,13,13, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */ - { CODEC_AACLD, 60000, 76000, 16000, 2, 14,14,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */ - { CODEC_AACLD, 76000,128001, 16000, 2, 14,14,13,13, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */ - - /* 22.05/44.1 kHz dual rate */ - { CODEC_AACLD, 32000, 36000, 22050, 2, 5, 4, 7, 6, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AACLD, 36000, 44000, 22050, 2, 5, 8, 8, 8, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */ - { CODEC_AACLD, 44000, 52000, 22050, 2, 7,10, 8, 8, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */ - { CODEC_AACLD, 52000, 60000, 22050, 2, 9,11, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */ - { CODEC_AACLD, 60000, 76000, 22050, 2, 10,12,10,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */ - { CODEC_AACLD, 76000, 82000, 22050, 2, 12,12,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */ - { CODEC_AACLD, 82000,128001, 22050, 2, 13,12,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */ - - /* 24/48 kHz dual rate */ - { CODEC_AACLD, 32000, 36000, 24000, 2, 5, 4, 7, 6, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */ - { CODEC_AACLD, 36000, 44000, 24000, 2, 4, 8, 8, 8, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */ - { CODEC_AACLD, 44000, 52000, 24000, 2, 6,10, 8, 8, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */ - { CODEC_AACLD, 52000, 60000, 24000, 2, 9,11, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */ - { CODEC_AACLD, 60000, 76000, 24000, 2, 11,12,10,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */ - { CODEC_AACLD, 76000, 88000, 24000, 2, 12,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */ - { CODEC_AACLD, 88000,128001, 24000, 2, 13,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 92 kbit/s */ - - /* 32/64 kHz dual rate */ /* placebo settings */ /*jgr: new, copy from CODEC_AAC */ - { CODEC_AACLD, 60000, 80000, 32000, 2, 7, 7, 6, 6, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* lowest range (multichannel rear) */ - { CODEC_AACLD, 80000,112000, 32000, 2, 9, 9, 8, 8, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* low range */ - { CODEC_AACLD, 112000,144000, 32000, 2, 11,11,10,10, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* SBR sweet spot */ - { CODEC_AACLD, 144000,256001, 32000, 2, 13,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* backwards compatible */ - - /* 44.1/88.2 kHz dual rate */ /* placebo settings */ /*wrr: new, copy from CODEC_AAC */ - { CODEC_AACLD, 60000, 80000, 44100, 2, 7, 7, 6, 6, 3, 0, -3, SBR_SWITCH_LRC, 2 }, /* lowest range (multichannel rear) */ - { CODEC_AACLD, 80000,112000, 44100, 2, 10,10, 8, 8, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* hlm 11-08-29 */ - { CODEC_AACLD, 112000,144000, 44100, 2, 12,12,10,10, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* hlm 11-08-29 */ - { CODEC_AACLD, 144000,256001, 44100, 2, 13,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* backwards compatible */ - - /* 48/96 kHz dual rate */ /* not yet finally tuned */ /*wrr: new, copy from CODEC_AAC */ - { CODEC_AACLD, 60000, 80000, 48000, 2, 7, 7,10,10, 2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 64 */ - { CODEC_AACLD, 80000,112000, 48000, 2, 9, 9,10,10, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 96 */ - { CODEC_AACLD, 112000,144000, 48000, 2, 11,11,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 128 */ - { CODEC_AACLD, 144000,176000, 48000, 2, 12,12,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* hlm 09-10-19 */ - { CODEC_AACLD, 176000,256001, 48000, 2, 13,13,11,11, 3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* hlm 09-10-19 */ - -}; - -const int sbrTuningTableSize = sizeof(sbrTuningTable)/sizeof(sbrTuningTable[0]); - -const psTuningTable_t psTuningTable[4] = -{ - { 8000, 22000, PSENC_STEREO_BANDS_10, PSENC_NENV_1, FL2FXCONST_DBL(3.0f/4.0f) }, - { 22000, 28000, PSENC_STEREO_BANDS_20, PSENC_NENV_1, FL2FXCONST_DBL(2.0f/4.0f) }, - { 28000, 36000, PSENC_STEREO_BANDS_20, PSENC_NENV_2, FL2FXCONST_DBL(1.5f/4.0f) }, - { 36000, 160001, PSENC_STEREO_BANDS_20, PSENC_NENV_4, FL2FXCONST_DBL(1.1f/4.0f) }, -}; - - -//@} - - - diff --git a/libSBRenc/src/sbr_rom.h b/libSBRenc/src/sbr_rom.h deleted file mode 100644 index afa924e..0000000 --- a/libSBRenc/src/sbr_rom.h +++ /dev/null @@ -1,127 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2013 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 ------------------------------------------------------------------------------------------------------------ */ - -/*! -\file -\brief Declaration of constant tables - -*/ -#ifndef __SBR_ROM_H -#define __SBR_ROM_H - -#include "sbr_def.h" -#include "sbr_encoder.h" - -#include "ps_main.h" - -/* - huffman tables -*/ -extern const INT v_Huff_envelopeLevelC10T[121]; -extern const UCHAR v_Huff_envelopeLevelL10T[121]; -extern const INT v_Huff_envelopeLevelC10F[121]; -extern const UCHAR v_Huff_envelopeLevelL10F[121]; -extern const INT bookSbrEnvBalanceC10T[49]; -extern const UCHAR bookSbrEnvBalanceL10T[49]; -extern const INT bookSbrEnvBalanceC10F[49]; -extern const UCHAR bookSbrEnvBalanceL10F[49]; -extern const INT v_Huff_envelopeLevelC11T[63]; -extern const UCHAR v_Huff_envelopeLevelL11T[63]; -extern const INT v_Huff_envelopeLevelC11F[63]; -extern const UCHAR v_Huff_envelopeLevelL11F[63]; -extern const INT bookSbrEnvBalanceC11T[25]; -extern const UCHAR bookSbrEnvBalanceL11T[25]; -extern const INT bookSbrEnvBalanceC11F[25]; -extern const UCHAR bookSbrEnvBalanceL11F[25]; -extern const INT v_Huff_NoiseLevelC11T[63]; -extern const UCHAR v_Huff_NoiseLevelL11T[63]; -extern const INT bookSbrNoiseBalanceC11T[25]; -extern const UCHAR bookSbrNoiseBalanceL11T[25]; - -extern const sbrTuningTable_t sbrTuningTable[]; -extern const int sbrTuningTableSize; - -extern const psTuningTable_t psTuningTable[4]; - - -#endif diff --git a/libSBRenc/src/sbrenc_freq_sca.cpp b/libSBRenc/src/sbrenc_freq_sca.cpp index 30bc5ca..c86e047 100644 --- a/libSBRenc/src/sbrenc_freq_sca.cpp +++ b/libSBRenc/src/sbrenc_freq_sca.cpp @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,12 +90,19 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ /*! \file - \brief frequency scale - \author Tobias Chalupka + \brief frequency scale $Revision: 95225 $ */ #include "sbrenc_freq_sca.h" @@ -98,12 +116,10 @@ static INT getStartFreq(INT fsCore, const INT start_freq); /* StopFreq */ static INT getStopFreq(INT fsCore, const INT stop_freq); -static INT numberOfBands(INT b_p_o, INT start, INT stop, FIXP_DBL warp_factor); -static void CalcBands(INT * diff, INT start , INT stop , INT num_bands); -static INT modifyBands(INT max_band, INT * diff, INT length); -static void cumSum(INT start_value, INT* diff, INT length, UCHAR *start_adress); - - +static INT numberOfBands(INT b_p_o, INT start, INT stop, FIXP_DBL warp_factor); +static void CalcBands(INT *diff, INT start, INT stop, INT num_bands); +static INT modifyBands(INT max_band, INT *diff, INT length); +static void cumSum(INT start_value, INT *diff, INT length, UCHAR *start_adress); /******************************************************************************* Functionname: FDKsbrEnc_getSbrStartFreqRAW @@ -115,24 +131,22 @@ static void cumSum(INT start_value, INT* diff, INT length, UCHAR *start_adress) Return: *******************************************************************************/ -INT -FDKsbrEnc_getSbrStartFreqRAW (INT startFreq, INT fsCore) -{ +INT FDKsbrEnc_getSbrStartFreqRAW(INT startFreq, INT fsCore) { INT result; - if ( startFreq < 0 || startFreq > 15) { + if (startFreq < 0 || startFreq > 15) { return -1; } /* Update startFreq struct */ result = getStartFreq(fsCore, startFreq); - result = (result*(fsCore>>5)+1)>>1; /* (result*fsSBR/QMFbands+1)>>1; */ + result = + (result * (fsCore >> 5) + 1) >> 1; /* (result*fsSBR/QMFbands+1)>>1; */ return (result); } /* End FDKsbrEnc_getSbrStartFreqRAW */ - /******************************************************************************* Functionname: getSbrStopFreq ******************************************************************************* @@ -142,21 +156,19 @@ FDKsbrEnc_getSbrStartFreqRAW (INT startFreq, INT fsCore) Return: *******************************************************************************/ -INT FDKsbrEnc_getSbrStopFreqRAW (INT stopFreq, INT fsCore) -{ +INT FDKsbrEnc_getSbrStopFreqRAW(INT stopFreq, INT fsCore) { INT result; - if ( stopFreq < 0 || stopFreq > 13) - return -1; + if (stopFreq < 0 || stopFreq > 13) return -1; /* Uppdate stopFreq struct */ result = getStopFreq(fsCore, stopFreq); - result = (result*(fsCore>>5)+1)>>1; /* (result*fsSBR/QMFbands+1)>>1; */ + result = + (result * (fsCore >> 5) + 1) >> 1; /* (result*fsSBR/QMFbands+1)>>1; */ return (result); } /* End getSbrStopFreq */ - /******************************************************************************* Functionname: getStartFreq ******************************************************************************* @@ -167,82 +179,80 @@ INT FDKsbrEnc_getSbrStopFreqRAW (INT stopFreq, INT fsCore) Return: *******************************************************************************/ -static INT -getStartFreq(INT fsCore, const INT start_freq) -{ +static INT getStartFreq(INT fsCore, const INT start_freq) { INT k0_min; - switch(fsCore){ - case 8000: k0_min = 24; /* (3000 * nQmfChannels / fsSBR ) + 0.5 */ - break; - case 11025: k0_min = 17; /* (3000 * nQmfChannels / fsSBR ) + 0.5 */ - break; - case 12000: k0_min = 16; /* (3000 * nQmfChannels / fsSBR ) + 0.5 */ - break; - case 16000: k0_min = 16; /* (4000 * nQmfChannels / fsSBR ) + 0.5 */ - break; - case 22050: k0_min = 12; /* (4000 * nQmfChannels / fsSBR ) + 0.5 */ - break; - case 24000: k0_min = 11; /* (4000 * nQmfChannels / fsSBR ) + 0.5 */ - break; - case 32000: k0_min = 10; /* (5000 * nQmfChannels / fsSBR ) + 0.5 */ - break; - case 44100: k0_min = 7; /* (5000 * nQmfChannels / fsSBR ) + 0.5 */ - break; - case 48000: k0_min = 7; /* (5000 * nQmfChannels / fsSBR ) + 0.5 */ - break; - case 96000: k0_min = 3; /* (5000 * nQmfChannels / fsSBR ) + 0.5 */ - break; - default: - k0_min=11; /* illegal fs */ + switch (fsCore) { + case 8000: + k0_min = 24; /* (3000 * nQmfChannels / fsSBR ) + 0.5 */ + break; + case 11025: + k0_min = 17; /* (3000 * nQmfChannels / fsSBR ) + 0.5 */ + break; + case 12000: + k0_min = 16; /* (3000 * nQmfChannels / fsSBR ) + 0.5 */ + break; + case 16000: + k0_min = 16; /* (4000 * nQmfChannels / fsSBR ) + 0.5 */ + break; + case 22050: + k0_min = 12; /* (4000 * nQmfChannels / fsSBR ) + 0.5 */ + break; + case 24000: + k0_min = 11; /* (4000 * nQmfChannels / fsSBR ) + 0.5 */ + break; + case 32000: + k0_min = 10; /* (5000 * nQmfChannels / fsSBR ) + 0.5 */ + break; + case 44100: + k0_min = 7; /* (5000 * nQmfChannels / fsSBR ) + 0.5 */ + break; + case 48000: + k0_min = 7; /* (5000 * nQmfChannels / fsSBR ) + 0.5 */ + break; + case 96000: + k0_min = 3; /* (5000 * nQmfChannels / fsSBR ) + 0.5 */ + break; + default: + k0_min = 11; /* illegal fs */ } - switch (fsCore) { - - case 8000: - { - INT v_offset[]= {-8, -7, -6, -5, -4, -3, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7}; + case 8000: { + INT v_offset[] = {-8, -7, -6, -5, -4, -3, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7}; return (k0_min + v_offset[start_freq]); } - case 11025: - { - INT v_offset[]= {-5, -4, -3, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13}; + case 11025: { + INT v_offset[] = {-5, -4, -3, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13}; return (k0_min + v_offset[start_freq]); } - case 12000: - { - INT v_offset[]= {-5, -3, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16}; + case 12000: { + INT v_offset[] = {-5, -3, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16}; return (k0_min + v_offset[start_freq]); } - case 16000: - { - INT v_offset[]= {-6, -4, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16}; + case 16000: { + INT v_offset[] = {-6, -4, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16}; return (k0_min + v_offset[start_freq]); } - case 22050: - case 24000: - case 32000: - { - INT v_offset[]= {-4, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16, 20}; + case 22050: + case 24000: + case 32000: { + INT v_offset[] = {-4, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16, 20}; return (k0_min + v_offset[start_freq]); } - case 44100: - case 48000: - case 96000: - { - INT v_offset[]= {-2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16, 20, 24}; + case 44100: + case 48000: + case 96000: { + INT v_offset[] = {-2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16, 20, 24}; return (k0_min + v_offset[start_freq]); } - default: - { - INT v_offset[]= {0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16, 20, 24, 28, 33}; + default: { + INT v_offset[] = {0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16, 20, 24, 28, 33}; return (k0_min + v_offset[start_freq]); } } } /* End getStartFreq */ - /******************************************************************************* Functionname: getStopFreq ******************************************************************************* @@ -252,78 +262,93 @@ getStartFreq(INT fsCore, const INT start_freq) Return: *******************************************************************************/ - static INT -getStopFreq(INT fsCore, const INT stop_freq) -{ - INT result,i; +static INT getStopFreq(INT fsCore, const INT stop_freq) { + INT result, i; INT k1_min; INT v_dstop[13]; INT *v_stop_freq = NULL; - INT v_stop_freq_16[14] = {48,49,50,51,52,54,55,56,57,59,60,61,63,64}; - INT v_stop_freq_22[14] = {35,37,38,40,42,44,46,48,51,53,56,58,61,64}; - INT v_stop_freq_24[14] = {32,34,36,38,40,42,44,46,49,52,55,58,61,64}; - INT v_stop_freq_32[14] = {32,34,36,38,40,42,44,46,49,52,55,58,61,64}; - INT v_stop_freq_44[14] = {23,25,27,29,32,34,37,40,43,47,51,55,59,64}; - INT v_stop_freq_48[14] = {21,23,25,27,30,32,35,38,42,45,49,54,59,64}; - INT v_stop_freq_64[14] = {20,22,24,26,29,31,34,37,41,45,49,54,59,64}; - INT v_stop_freq_88[14] = {15,17,19,21,23,26,29,33,37,41,46,51,57,64}; - INT v_stop_freq_96[14] = {13,15,17,19,21,24,27,31,35,39,44,50,57,64}; - INT v_stop_freq_192[14] = {7, 8,10,12,14,16,19,23,27,32,38,46,54,64}; - - switch(fsCore){ - case 8000: k1_min = 48; - v_stop_freq =v_stop_freq_16; - break; - case 11025: k1_min = 35; - v_stop_freq =v_stop_freq_22; - break; - case 12000: k1_min = 32; - v_stop_freq =v_stop_freq_24; - break; - case 16000: k1_min = 32; - v_stop_freq =v_stop_freq_32; - break; - case 22050: k1_min = 23; - v_stop_freq =v_stop_freq_44; - break; - case 24000: k1_min = 21; - v_stop_freq =v_stop_freq_48; - break; - case 32000: k1_min = 20; - v_stop_freq =v_stop_freq_64; - break; - case 44100: k1_min = 15; - v_stop_freq =v_stop_freq_88; - break; - case 48000: k1_min = 13; - v_stop_freq =v_stop_freq_96; - break; - case 96000: k1_min = 7; - v_stop_freq =v_stop_freq_192; - break; - default: - k1_min = 21; /* illegal fs */ + INT v_stop_freq_16[14] = {48, 49, 50, 51, 52, 54, 55, + 56, 57, 59, 60, 61, 63, 64}; + INT v_stop_freq_22[14] = {35, 37, 38, 40, 42, 44, 46, + 48, 51, 53, 56, 58, 61, 64}; + INT v_stop_freq_24[14] = {32, 34, 36, 38, 40, 42, 44, + 46, 49, 52, 55, 58, 61, 64}; + INT v_stop_freq_32[14] = {32, 34, 36, 38, 40, 42, 44, + 46, 49, 52, 55, 58, 61, 64}; + INT v_stop_freq_44[14] = {23, 25, 27, 29, 32, 34, 37, + 40, 43, 47, 51, 55, 59, 64}; + INT v_stop_freq_48[14] = {21, 23, 25, 27, 30, 32, 35, + 38, 42, 45, 49, 54, 59, 64}; + INT v_stop_freq_64[14] = {20, 22, 24, 26, 29, 31, 34, + 37, 41, 45, 49, 54, 59, 64}; + INT v_stop_freq_88[14] = {15, 17, 19, 21, 23, 26, 29, + 33, 37, 41, 46, 51, 57, 64}; + INT v_stop_freq_96[14] = {13, 15, 17, 19, 21, 24, 27, + 31, 35, 39, 44, 50, 57, 64}; + INT v_stop_freq_192[14] = {7, 8, 10, 12, 14, 16, 19, + 23, 27, 32, 38, 46, 54, 64}; + + switch (fsCore) { + case 8000: + k1_min = 48; + v_stop_freq = v_stop_freq_16; + break; + case 11025: + k1_min = 35; + v_stop_freq = v_stop_freq_22; + break; + case 12000: + k1_min = 32; + v_stop_freq = v_stop_freq_24; + break; + case 16000: + k1_min = 32; + v_stop_freq = v_stop_freq_32; + break; + case 22050: + k1_min = 23; + v_stop_freq = v_stop_freq_44; + break; + case 24000: + k1_min = 21; + v_stop_freq = v_stop_freq_48; + break; + case 32000: + k1_min = 20; + v_stop_freq = v_stop_freq_64; + break; + case 44100: + k1_min = 15; + v_stop_freq = v_stop_freq_88; + break; + case 48000: + k1_min = 13; + v_stop_freq = v_stop_freq_96; + break; + case 96000: + k1_min = 7; + v_stop_freq = v_stop_freq_192; + break; + default: + k1_min = 21; /* illegal fs */ } - /* if no valid core samplingrate is used this loop produces - a segfault, because v_stop_freq is not initialized */ /* Ensure increasing bandwidth */ - for(i = 0; i <= 12; i++) { - v_dstop[i] = v_stop_freq[i+1] - v_stop_freq[i]; + for (i = 0; i <= 12; i++) { + v_dstop[i] = v_stop_freq[i + 1] - v_stop_freq[i]; } FDKsbrEnc_Shellsort_int(v_dstop, 13); /* Sort bandwidth changes */ result = k1_min; - for(i = 0; i < stop_freq; i++) { + for (i = 0; i < stop_freq; i++) { result = result + v_dstop[i]; } - return(result); - -}/* End getStopFreq */ + return (result); +} /* End getStopFreq */ /******************************************************************************* Functionname: FDKsbrEnc_FindStartAndStopBand @@ -341,31 +366,23 @@ getStopFreq(INT fsCore, const INT stop_freq) Return: Error code (0 is OK) *******************************************************************************/ -INT -FDKsbrEnc_FindStartAndStopBand( - const INT srSbr, - const INT srCore, - const INT noChannels, - const INT startFreq, - const INT stopFreq, - INT *k0, - INT *k2 - ) -{ - +INT FDKsbrEnc_FindStartAndStopBand(const INT srSbr, const INT srCore, + const INT noChannels, const INT startFreq, + const INT stopFreq, INT *k0, INT *k2) { /* Update startFreq struct */ *k0 = getStartFreq(srCore, startFreq); /* Test if start freq is outside corecoder range */ - if( srSbr*noChannels < *k0 * srCore ) { - return (1); /* raise the cross-over frequency and/or lower the number - of target bands per octave (or lower the sampling frequency) */ + if (srSbr * noChannels < *k0 * srCore) { + return ( + 1); /* raise the cross-over frequency and/or lower the number + of target bands per octave (or lower the sampling frequency) */ } /*Update stopFreq struct */ - if ( stopFreq < 14 ) { + if (stopFreq < 14) { *k2 = getStopFreq(srCore, stopFreq); - } else if( stopFreq == 14 ) { + } else if (stopFreq == 14) { *k2 = 2 * *k0; } else { *k2 = 3 * *k0; @@ -376,23 +393,21 @@ FDKsbrEnc_FindStartAndStopBand( *k2 = noChannels; } - - /* Test for invalid k0 k2 combinations */ - if ( (srCore == 22050) && ( (*k2 - *k0) > MAX_FREQ_COEFFS_FS44100 ) ) - return (1); /* Number of bands exceeds valid range of MAX_FREQ_COEFFS for fs=44.1kHz */ + if ((srCore == 22050) && ((*k2 - *k0) > MAX_FREQ_COEFFS_FS44100)) + return (1); /* Number of bands exceeds valid range of MAX_FREQ_COEFFS for + fs=44.1kHz */ - if ( (srCore >= 24000) && ( (*k2 - *k0) > MAX_FREQ_COEFFS_FS48000 ) ) - return (1); /* Number of bands exceeds valid range of MAX_FREQ_COEFFS for fs>=48kHz */ + if ((srCore >= 24000) && ((*k2 - *k0) > MAX_FREQ_COEFFS_FS48000)) + return (1); /* Number of bands exceeds valid range of MAX_FREQ_COEFFS for + fs>=48kHz */ if ((*k2 - *k0) > MAX_FREQ_COEFFS) - return (1);/*Number of bands exceeds valid range of MAX_FREQ_COEFFS */ - - if ((*k2 - *k0) < 0) - return (1);/* Number of bands is negative */ + return (1); /*Number of bands exceeds valid range of MAX_FREQ_COEFFS */ + if ((*k2 - *k0) < 0) return (1); /* Number of bands is negative */ - return(0); + return (0); } /******************************************************************************* @@ -404,207 +419,188 @@ FDKsbrEnc_FindStartAndStopBand( Return: *******************************************************************************/ -INT -FDKsbrEnc_UpdateFreqScale( - UCHAR *v_k_master, - INT *h_num_bands, - const INT k0, - const INT k2, - const INT freqScale, - const INT alterScale - ) +INT FDKsbrEnc_UpdateFreqScale(UCHAR *v_k_master, INT *h_num_bands, const INT k0, + const INT k2, const INT freqScale, + const INT alterScale) { - - INT b_p_o = 0; /* bands_per_octave */ + INT b_p_o = 0; /* bands_per_octave */ FIXP_DBL warp = FL2FXCONST_DBL(0.0f); - INT dk = 0; + INT dk = 0; /* Internal variables */ - INT k1 = 0, i; - INT num_bands0; - INT num_bands1; - INT diff_tot[MAX_OCTAVE + MAX_SECOND_REGION]; - INT *diff0 = diff_tot; - INT *diff1 = diff_tot+MAX_OCTAVE; - INT k2_achived; - INT k2_diff; - INT incr = 0; + INT k1 = 0, i; + INT num_bands0; + INT num_bands1; + INT diff_tot[MAX_OCTAVE + MAX_SECOND_REGION]; + INT *diff0 = diff_tot; + INT *diff1 = diff_tot + MAX_OCTAVE; + INT k2_achived; + INT k2_diff; + INT incr = 0; /* Init */ - if (freqScale==1) b_p_o = 12; - if (freqScale==2) b_p_o = 10; - if (freqScale==3) b_p_o = 8; - - - if(freqScale > 0) /*Bark*/ + if (freqScale == 1) b_p_o = 12; + if (freqScale == 2) b_p_o = 10; + if (freqScale == 3) b_p_o = 8; + + if (freqScale > 0) /*Bark*/ + { + if (alterScale == 0) + warp = FL2FXCONST_DBL(0.5f); /* 1.0/(1.0*2.0) */ + else + warp = FL2FXCONST_DBL(1.0f / 2.6f); /* 1.0/(1.3*2.0); */ + + if (4 * k2 >= 9 * k0) /*two or more regions (how many times the basis band + is copied)*/ { - if(alterScale==0) - warp = FL2FXCONST_DBL(0.5f); /* 1.0/(1.0*2.0) */ - else - warp = FL2FXCONST_DBL(1.0f/2.6f); /* 1.0/(1.3*2.0); */ - - - if(4*k2 >= 9*k0) /*two or more regions (how many times the basis band is copied)*/ - { - k1=2*k0; + k1 = 2 * k0; - num_bands0=numberOfBands(b_p_o, k0, k1, FL2FXCONST_DBL(0.5f)); - num_bands1=numberOfBands(b_p_o, k1, k2, warp); + num_bands0 = numberOfBands(b_p_o, k0, k1, FL2FXCONST_DBL(0.5f)); + num_bands1 = numberOfBands(b_p_o, k1, k2, warp); - CalcBands(diff0, k0, k1, num_bands0);/*CalcBands1 => diff0 */ - FDKsbrEnc_Shellsort_int( diff0, num_bands0);/*SortBands sort diff0 */ + CalcBands(diff0, k0, k1, num_bands0); /*CalcBands1 => diff0 */ + FDKsbrEnc_Shellsort_int(diff0, num_bands0); /*SortBands sort diff0 */ - if (diff0[0] == 0) /* too wide FB bands for target tuning */ - { - return (1);/* raise the cross-over frequency and/or lower the number - of target bands per octave (or lower the sampling frequency */ - } - - cumSum(k0, diff0, num_bands0, v_k_master); /* cumsum */ - - CalcBands(diff1, k1, k2, num_bands1); /* CalcBands2 => diff1 */ - FDKsbrEnc_Shellsort_int( diff1, num_bands1); /* SortBands sort diff1 */ - if(diff0[num_bands0-1] > diff1[0]) /* max(1) > min(2) */ - { - if(modifyBands(diff0[num_bands0-1],diff1, num_bands1)) - return(1); - } - - /* Add 2'nd region */ - cumSum(k1, diff1, num_bands1, &v_k_master[num_bands0]); - *h_num_bands=num_bands0+num_bands1; /* Output nr of bands */ - - } - else /* one region */ - { - k1=k2; + if (diff0[0] == 0) /* too wide FB bands for target tuning */ + { + return (1); /* raise the cross-over frequency and/or lower the number + of target bands per octave (or lower the sampling + frequency */ + } - num_bands0=numberOfBands(b_p_o, k0, k1, FL2FXCONST_DBL(0.5f)); - CalcBands(diff0, k0, k1, num_bands0);/* CalcBands1 => diff0 */ - FDKsbrEnc_Shellsort_int( diff0, num_bands0); /* SortBands sort diff0 */ + cumSum(k0, diff0, num_bands0, v_k_master); /* cumsum */ - if (diff0[0] == 0) /* too wide FB bands for target tuning */ - { - return (1); /* raise the cross-over frequency and/or lower the number - of target bands per octave (or lower the sampling frequency */ - } + CalcBands(diff1, k1, k2, num_bands1); /* CalcBands2 => diff1 */ + FDKsbrEnc_Shellsort_int(diff1, num_bands1); /* SortBands sort diff1 */ + if (diff0[num_bands0 - 1] > diff1[0]) /* max(1) > min(2) */ + { + if (modifyBands(diff0[num_bands0 - 1], diff1, num_bands1)) return (1); + } - cumSum(k0, diff0, num_bands0, v_k_master);/* cumsum */ - *h_num_bands=num_bands0; /* Output nr of bands */ + /* Add 2'nd region */ + cumSum(k1, diff1, num_bands1, &v_k_master[num_bands0]); + *h_num_bands = num_bands0 + num_bands1; /* Output nr of bands */ - } - } - else /* Linear mode */ + } else /* one region */ { - if (alterScale==0) { - dk = 1; - num_bands0 = 2 * ((k2 - k0)/2); /* FLOOR to get to few number of bands*/ - } else { - dk = 2; - num_bands0 = 2 * (((k2 - k0)/dk +1)/2); /* ROUND to get closest fit */ - } - - k2_achived = k0 + num_bands0*dk; - k2_diff = k2 - k2_achived; + k1 = k2; - for(i=0;i<num_bands0;i++) - diff_tot[i] = dk; + num_bands0 = numberOfBands(b_p_o, k0, k1, FL2FXCONST_DBL(0.5f)); + CalcBands(diff0, k0, k1, num_bands0); /* CalcBands1 => diff0 */ + FDKsbrEnc_Shellsort_int(diff0, num_bands0); /* SortBands sort diff0 */ - /* If linear scale wasn't achived */ - /* and we got wide SBR are */ - if (k2_diff < 0) { - incr = 1; - i = 0; + if (diff0[0] == 0) /* too wide FB bands for target tuning */ + { + return (1); /* raise the cross-over frequency and/or lower the number + of target bands per octave (or lower the sampling + frequency */ } - /* If linear scale wasn't achived */ - /* and we got small SBR are */ - if (k2_diff > 0) { - incr = -1; - i = num_bands0-1; - } + cumSum(k0, diff0, num_bands0, v_k_master); /* cumsum */ + *h_num_bands = num_bands0; /* Output nr of bands */ + } + } else /* Linear mode */ + { + if (alterScale == 0) { + dk = 1; + num_bands0 = 2 * ((k2 - k0) / 2); /* FLOOR to get to few number of bands*/ + } else { + dk = 2; + num_bands0 = + 2 * (((k2 - k0) / dk + 1) / 2); /* ROUND to get closest fit */ + } - /* 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; - } + k2_achived = k0 + num_bands0 * dk; + k2_diff = k2 - k2_achived; - cumSum(k0, diff_tot, num_bands0, v_k_master);/* cumsum */ - *h_num_bands=num_bands0; /* Output nr of bands */ + for (i = 0; i < num_bands0; i++) diff_tot[i] = dk; + /* If linear scale wasn't achived */ + /* and we got wide SBR are */ + if (k2_diff < 0) { + incr = 1; + i = 0; } - if (*h_num_bands < 1) - return(1); /*To small sbr area */ - - return (0); -}/* End FDKsbrEnc_UpdateFreqScale */ + /* If linear scale wasn't achived */ + /* and we got small SBR are */ + if (k2_diff > 0) { + incr = -1; + i = num_bands0 - 1; + } -static INT -numberOfBands(INT b_p_o, INT start, INT stop, FIXP_DBL warp_factor) -{ - INT result=0; - /* result = 2* (INT) ( (double)b_p_o * (double)(FDKlog((double)stop/(double)start)/FDKlog((double)2)) * (double)FX_DBL2FL(warp_factor) + 0.5); */ - result = ( ( b_p_o * fMult( (CalcLdInt(stop) - CalcLdInt(start)), warp_factor) + (FL2FX_DBL(0.5f)>>LD_DATA_SHIFT) - ) >> ((DFRACT_BITS-1)-LD_DATA_SHIFT) ) << 1; /* do not optimize anymore (rounding!!) */ + /* 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; + } - return(result); -} + cumSum(k0, diff_tot, num_bands0, v_k_master); /* cumsum */ + *h_num_bands = num_bands0; /* Output nr of bands */ + } + if (*h_num_bands < 1) return (1); /*To small sbr area */ -static void -CalcBands(INT * diff, INT start , INT stop , INT num_bands) -{ - INT i, qb, qe, qtmp; - INT previous; - INT current; - FIXP_DBL base, exp, tmp; + return (0); +} /* End FDKsbrEnc_UpdateFreqScale */ + +static INT numberOfBands(INT b_p_o, INT start, INT stop, FIXP_DBL warp_factor) { + INT result = 0; + /* result = 2* (INT) ( (double)b_p_o * + * (double)(FDKlog((double)stop/(double)start)/FDKlog((double)2)) * + * (double)FX_DBL2FL(warp_factor) + 0.5); */ + result = ((b_p_o * fMult((CalcLdInt(stop) - CalcLdInt(start)), warp_factor) + + (FL2FX_DBL(0.5f) >> LD_DATA_SHIFT)) >> + ((DFRACT_BITS - 1) - LD_DATA_SHIFT)) + << 1; /* do not optimize anymore (rounding!!) */ - previous=start; - for(i=1; i<= num_bands; i++) - { - base = fDivNorm((FIXP_DBL)stop, (FIXP_DBL)start, &qb); - exp = fDivNorm((FIXP_DBL)i, (FIXP_DBL)num_bands, &qe); - tmp = fPow(base, qb, exp, qe, &qtmp); - tmp = fMult(tmp, (FIXP_DBL)(start<<24)); - current = (INT)scaleValue(tmp, qtmp-23); - current = (current+1) >> 1; /* rounding*/ - diff[i-1] = current-previous; - previous = current; - } + return (result); +} -}/* End CalcBands */ +static void CalcBands(INT *diff, INT start, INT stop, INT num_bands) { + INT i, qb, qe, qtmp; + INT previous; + INT current; + FIXP_DBL base, exp, tmp; + + previous = start; + for (i = 1; i <= num_bands; i++) { + base = fDivNorm((FIXP_DBL)stop, (FIXP_DBL)start, &qb); + exp = fDivNorm((FIXP_DBL)i, (FIXP_DBL)num_bands, &qe); + tmp = fPow(base, qb, exp, qe, &qtmp); + tmp = fMult(tmp, (FIXP_DBL)(start << 24)); + current = (INT)scaleValue(tmp, qtmp - 23); + current = (current + 1) >> 1; /* rounding*/ + diff[i - 1] = current - previous; + previous = current; + } +} /* End CalcBands */ -static void -cumSum(INT start_value, INT* diff, INT length, UCHAR *start_adress) -{ +static void cumSum(INT start_value, INT *diff, INT 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]; + start_adress[0] = start_value; + for (i = 1; i <= length; i++) + start_adress[i] = start_adress[i - 1] + diff[i - 1]; } /* End cumSum */ +static INT modifyBands(INT max_band_previous, INT *diff, INT length) { + INT change = max_band_previous - diff[0]; -static INT -modifyBands(INT max_band_previous, INT * diff, INT 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]) / 2 ) - change = (diff[length-1] - diff[0]) / 2; + /* Limit the change so that the last band cannot get narrower than the first + * one */ + if (change > (diff[length - 1] - diff[0]) / 2) + change = (diff[length - 1] - diff[0]) / 2; diff[0] += change; - diff[length-1] -= change; + diff[length - 1] -= change; FDKsbrEnc_Shellsort_int(diff, length); - return(0); -}/* End modifyBands */ - + return (0); +} /* End modifyBands */ /******************************************************************************* Functionname: FDKsbrEnc_UpdateHiRes @@ -616,43 +612,34 @@ modifyBands(INT max_band_previous, INT * diff, INT length) Return: *******************************************************************************/ -INT -FDKsbrEnc_UpdateHiRes( - UCHAR *h_hires, - INT *num_hires, - UCHAR *v_k_master, - INT num_master, - INT *xover_band - ) -{ +INT FDKsbrEnc_UpdateHiRes(UCHAR *h_hires, INT *num_hires, UCHAR *v_k_master, + INT num_master, INT *xover_band) { INT i; - INT max1,max2; + INT max1, max2; - if( (v_k_master[*xover_band] > 32 ) || /* v_k_master[*xover_band] > noQMFChannels(dualRate)/divider */ - ( *xover_band > num_master ) ) { - /* xover_band error, too big for this startFreq. Will be clipped */ + if ((v_k_master[*xover_band] > + 32) || /* v_k_master[*xover_band] > noQMFChannels(dualRate)/divider */ + (*xover_band > num_master)) { + /* xover_band error, too big for this startFreq. Will be clipped */ /* Calculate maximum value for xover_band */ - max1=0; - max2=num_master; - while( (v_k_master[max1+1] < 32 ) && /* noQMFChannels(dualRate)/divider */ - ( (max1+1) < max2) ) - { - max1++; - } + max1 = 0; + max2 = num_master; + while ((v_k_master[max1 + 1] < 32) && /* noQMFChannels(dualRate)/divider */ + ((max1 + 1) < max2)) { + max1++; + } - *xover_band=max1; + *xover_band = max1; } *num_hires = num_master - *xover_band; - for(i = *xover_band; i <= num_master; i++) - { - h_hires[i - *xover_band] = v_k_master[i]; - } + for (i = *xover_band; i <= num_master; i++) { + h_hires[i - *xover_band] = v_k_master[i]; + } return (0); -}/* End FDKsbrEnc_UpdateHiRes */ - +} /* End FDKsbrEnc_UpdateHiRes */ /******************************************************************************* Functionname: FDKsbrEnc_UpdateLoRes @@ -663,29 +650,25 @@ FDKsbrEnc_UpdateHiRes( Return: *******************************************************************************/ -void -FDKsbrEnc_UpdateLoRes(UCHAR * h_lores, INT *num_lores, UCHAR * h_hires, INT num_hires) -{ +void FDKsbrEnc_UpdateLoRes(UCHAR *h_lores, INT *num_lores, UCHAR *h_hires, + INT num_hires) { INT i; - if(num_hires%2 == 0) /* if even number of hires bands */ - { - *num_lores=num_hires/2; - /* Use every second lores=hires[0,2,4...] */ - for(i=0;i<=*num_lores;i++) - h_lores[i]=h_hires[i*2]; + if (num_hires % 2 == 0) /* if even number of hires bands */ + { + *num_lores = num_hires / 2; + /* 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) / 2; + + /* 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]; } - else /* odd number of hires which means xover is odd */ - { - *num_lores=(num_hires+1)/2; - - /* 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]; - } - } + } -}/* End FDKsbrEnc_UpdateLoRes */ +} /* End FDKsbrEnc_UpdateLoRes */ diff --git a/libSBRenc/src/sbrenc_freq_sca.h b/libSBRenc/src/sbrenc_freq_sca.h index 6f2bb84..9b8d360 100644 --- a/libSBRenc/src/sbrenc_freq_sca.h +++ b/libSBRenc/src/sbrenc_freq_sca.h @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,59 +90,43 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ /*! \file - \brief frequency scale prototypes + \brief frequency scale prototypes $Revision: 92790 $ */ -#ifndef __FREQ_SCA2_H -#define __FREQ_SCA2_H +#ifndef SBRENC_FREQ_SCA_H +#define SBRENC_FREQ_SCA_H #include "sbr_encoder.h" #include "sbr_def.h" -#define MAX_OCTAVE 29 +#define MAX_OCTAVE 29 #define MAX_SECOND_REGION 50 +INT FDKsbrEnc_UpdateFreqScale(UCHAR *v_k_master, INT *h_num_bands, const INT k0, + const INT k2, const INT freq_scale, + const INT alter_scale); + +INT FDKsbrEnc_UpdateHiRes(UCHAR *h_hires, INT *num_hires, UCHAR *v_k_master, + INT num_master, INT *xover_band); + +void FDKsbrEnc_UpdateLoRes(UCHAR *v_lores, INT *num_lores, UCHAR *v_hires, + INT num_hires); + +INT FDKsbrEnc_FindStartAndStopBand(const INT srSbr, const INT srCore, + const INT noChannels, const INT startFreq, + const INT stop_freq, INT *k0, INT *k2); -INT -FDKsbrEnc_UpdateFreqScale( - UCHAR *v_k_master, - INT *h_num_bands, - const INT k0, - const INT k2, - const INT freq_scale, - const INT alter_scale - ); - -INT -FDKsbrEnc_UpdateHiRes( - UCHAR *h_hires, - INT *num_hires, - UCHAR *v_k_master, - INT num_master, - INT *xover_band - ); - -void FDKsbrEnc_UpdateLoRes( - UCHAR *v_lores, - INT *num_lores, - UCHAR *v_hires, - INT num_hires - ); - -INT -FDKsbrEnc_FindStartAndStopBand( - const INT srSbr, - const INT srCore, - const INT noChannels, - const INT startFreq, - const INT stop_freq, - INT *k0, - INT *k2 - ); - -INT FDKsbrEnc_getSbrStartFreqRAW (INT startFreq, INT fsCore); -INT FDKsbrEnc_getSbrStopFreqRAW (INT stopFreq, INT fsCore); +INT FDKsbrEnc_getSbrStartFreqRAW(INT startFreq, INT fsCore); +INT FDKsbrEnc_getSbrStopFreqRAW(INT stopFreq, INT fsCore); #endif diff --git a/libSBRenc/src/sbrenc_ram.cpp b/libSBRenc/src/sbrenc_ram.cpp new file mode 100644 index 0000000..fb30fa2 --- /dev/null +++ b/libSBRenc/src/sbrenc_ram.cpp @@ -0,0 +1,249 @@ +/* ----------------------------------------------------------------------------- +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 encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief Memory layout + $Revision: 92864 $ + + This module declares all static and dynamic memory spaces +*/ +#include "sbrenc_ram.h" + +#include "sbr.h" +#include "genericStds.h" + +C_AALLOC_MEM(Ram_SbrDynamic_RAM, FIXP_DBL, + ((SBR_ENC_DYN_RAM_SIZE) / sizeof(FIXP_DBL))) + +/*! + \name StaticSbrData + + Static memory areas, must not be overwritten in other sections of the encoder +*/ +/* @{ */ + +/*! static sbr encoder instance for one encoder (2 channels) + all major static and dynamic memory areas are located + in module sbr_ram and sbr rom +*/ +C_ALLOC_MEM(Ram_SbrEncoder, SBR_ENCODER, 1) +C_ALLOC_MEM2(Ram_SbrChannel, SBR_CHANNEL, 1, (8)) +C_ALLOC_MEM2(Ram_SbrElement, SBR_ELEMENT, 1, (8)) + +/*! Filter states for QMF-analysis. <br> + Dimension: #MAXNRSBRCHANNELS * #SBR_QMF_FILTER_LENGTH +*/ +C_AALLOC_MEM2_L(Ram_Sbr_QmfStatesAnalysis, FIXP_QAS, 640, (8), SECT_DATA_L1) + +/*! Matrix holding the quota values for all estimates, all channels + Dimension #MAXNRSBRCHANNELS * +#SBR_QMF_CHANNELS* #MAX_NO_OF_ESTIMATES +*/ +C_ALLOC_MEM2_L(Ram_Sbr_quotaMatrix, FIXP_DBL, (MAX_NO_OF_ESTIMATES * 64), (8), + SECT_DATA_L1) + +/*! Matrix holding the sign values for all estimates, all channels + Dimension #MAXNRSBRCHANNELS * +#SBR_QMF_CHANNELS* #MAX_NO_OF_ESTIMATES +*/ +C_ALLOC_MEM2(Ram_Sbr_signMatrix, INT, (MAX_NO_OF_ESTIMATES * 64), (8)) + +/*! Frequency band table (low res) <br> + Dimension #MAX_FREQ_COEFFS/2+1 +*/ +C_ALLOC_MEM2(Ram_Sbr_freqBandTableLO, UCHAR, (MAX_FREQ_COEFFS / 2 + 1), (8)) + +/*! Frequency band table (high res) <br> + Dimension #MAX_FREQ_COEFFS +1 +*/ +C_ALLOC_MEM2(Ram_Sbr_freqBandTableHI, UCHAR, (MAX_FREQ_COEFFS + 1), (8)) + +/*! vk matser table <br> + Dimension #MAX_FREQ_COEFFS +1 +*/ +C_ALLOC_MEM2(Ram_Sbr_v_k_master, UCHAR, (MAX_FREQ_COEFFS + 1), (8)) + +/* + Missing harmonics detection +*/ + +/*! sbr_detectionVectors <br> + Dimension #MAX_NUM_CHANNELS*#MAX_NO_OF_ESTIMATES*#MAX_FREQ_COEFFS] +*/ +C_ALLOC_MEM2(Ram_Sbr_detectionVectors, UCHAR, + (MAX_NO_OF_ESTIMATES * MAX_FREQ_COEFFS), (8)) + +/*! sbr_prevCompVec[ <br> + Dimension #MAX_NUM_CHANNELS*#MAX_FREQ_COEFFS] +*/ +C_ALLOC_MEM2(Ram_Sbr_prevEnvelopeCompensation, UCHAR, MAX_FREQ_COEFFS, (8)) +/*! sbr_guideScfb[ <br> + Dimension #MAX_NUM_CHANNELS*#MAX_FREQ_COEFFS] +*/ +C_ALLOC_MEM2(Ram_Sbr_guideScfb, UCHAR, MAX_FREQ_COEFFS, (8)) + +/*! sbr_guideVectorDetected <br> + Dimension #MAX_NUM_CHANNELS*#MAX_NO_OF_ESTIMATES*#MAX_FREQ_COEFFS] +*/ +C_ALLOC_MEM2(Ram_Sbr_guideVectorDetected, UCHAR, + (MAX_NO_OF_ESTIMATES * MAX_FREQ_COEFFS), (8)) +C_ALLOC_MEM2(Ram_Sbr_guideVectorDiff, FIXP_DBL, + (MAX_NO_OF_ESTIMATES * MAX_FREQ_COEFFS), (8)) +C_ALLOC_MEM2(Ram_Sbr_guideVectorOrig, FIXP_DBL, + (MAX_NO_OF_ESTIMATES * MAX_FREQ_COEFFS), (8)) + +/* + Static Parametric Stereo memory +*/ +C_AALLOC_MEM_L(Ram_PsQmfStatesSynthesis, FIXP_DBL, 640 / 2, SECT_DATA_L1) + +C_ALLOC_MEM_L(Ram_PsEncode, PS_ENCODE, 1, SECT_DATA_L1) +C_ALLOC_MEM(Ram_ParamStereo, PARAMETRIC_STEREO, 1) + +/* @} */ + +/*! + \name DynamicSbrData + + Dynamic memory areas, might be reused in other algorithm sections, + e.g. the core encoder. +*/ +/* @{ */ + +/*! Energy buffer for envelope extraction <br> + Dimension #MAXNRSBRCHANNELS * +#SBR_QMF_SLOTS * #SBR_QMF_CHANNELS +*/ +C_ALLOC_MEM2(Ram_Sbr_envYBuffer, FIXP_DBL, (32 / 2 * 64), (8)) + +FIXP_DBL* GetRam_Sbr_envYBuffer(int n, UCHAR* dynamic_RAM) { + FDK_ASSERT(dynamic_RAM != 0); + /* The reinterpret_cast is used to suppress a compiler warning. We know that + * (dynamic_RAM + OFFSET_NRG + (n*Y_2_BUF_BYTE)) is sufficiently aligned, so + * the cast is safe */ + return reinterpret_cast<FIXP_DBL*>( + reinterpret_cast<void*>(dynamic_RAM + OFFSET_NRG + (n * Y_2_BUF_BYTE))); +} + +/* + * QMF data + */ +/* The SBR encoder uses a single channel overlapping buffer set (always n=0), + * but PS does not. */ +FIXP_DBL* GetRam_Sbr_envRBuffer(int n, UCHAR* dynamic_RAM) { + FDK_ASSERT(dynamic_RAM != 0); + /* The reinterpret_cast is used to suppress a compiler warning. We know that + * (dynamic_RAM + OFFSET_QMF + (n*(ENV_R_BUFF_BYTE+ENV_I_BUFF_BYTE))) is + * sufficiently aligned, so the cast is safe */ + return reinterpret_cast<FIXP_DBL*>(reinterpret_cast<void*>( + dynamic_RAM + OFFSET_QMF + (n * (ENV_R_BUFF_BYTE + ENV_I_BUFF_BYTE)))); +} +FIXP_DBL* GetRam_Sbr_envIBuffer(int n, UCHAR* dynamic_RAM) { + FDK_ASSERT(dynamic_RAM != 0); + /* The reinterpret_cast is used to suppress a compiler warning. We know that + * (dynamic_RAM + OFFSET_QMF + (ENV_R_BUFF_BYTE) + + * (n*(ENV_R_BUFF_BYTE+ENV_I_BUFF_BYTE))) is sufficiently aligned, so the cast + * is safe */ + return reinterpret_cast<FIXP_DBL*>( + reinterpret_cast<void*>(dynamic_RAM + OFFSET_QMF + (ENV_R_BUFF_BYTE) + + (n * (ENV_R_BUFF_BYTE + ENV_I_BUFF_BYTE)))); +} + +/* @} */ diff --git a/libSBRenc/src/sbrenc_ram.h b/libSBRenc/src/sbrenc_ram.h new file mode 100644 index 0000000..cf23378 --- /dev/null +++ b/libSBRenc/src/sbrenc_ram.h @@ -0,0 +1,199 @@ +/* ----------------------------------------------------------------------------- +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 encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! +\file +\brief Memory layout +$Revision: 92790 $ +*/ +#ifndef SBRENC_RAM_H +#define SBRENC_RAM_H + +#include "sbr_def.h" +#include "env_est.h" +#include "sbr_encoder.h" +#include "sbr.h" + +#include "ps_main.h" +#include "ps_encode.h" + +#define ENV_TRANSIENTS_BYTE ((sizeof(FIXP_DBL) * (MAX_NUM_CHANNELS * 3 * 32))) + +#define ENV_R_BUFF_BYTE ((sizeof(FIXP_DBL) * ((32) * MAX_HYBRID_BANDS))) +#define ENV_I_BUFF_BYTE ((sizeof(FIXP_DBL) * ((32) * MAX_HYBRID_BANDS))) +#define Y_BUF_CH_BYTE \ + ((2 * sizeof(FIXP_DBL) * (((32) - (32 / 2)) * MAX_HYBRID_BANDS))) + +#define ENV_R_BUF_PS_BYTE ((sizeof(FIXP_DBL) * 32 * 64 / 2)) +#define ENV_I_BUF_PS_BYTE ((sizeof(FIXP_DBL) * 32 * 64 / 2)) + +#define TON_BUF_CH_BYTE \ + ((sizeof(FIXP_DBL) * (MAX_NO_OF_ESTIMATES * MAX_FREQ_COEFFS))) + +#define Y_2_BUF_BYTE (Y_BUF_CH_BYTE) + +/* Workbuffer RAM - Allocation */ +/* + ++++++++++++++++++++++++++++++++++++++++++++++++++++ + | OFFSET_QMF | OFFSET_NRG | + ++++++++++++++++++++++++++++++++++++++++++++++++++++ + ------------------------- ------------------------- + | | 0.5 * | + | sbr_envRBuffer | sbr_envYBuffer_size | + | sbr_envIBuffer | | + ------------------------- ------------------------- + +*/ +#define BUF_NRG_SIZE ((MAX_NUM_CHANNELS * Y_2_BUF_BYTE)) +#define BUF_QMF_SIZE (ENV_R_BUFF_BYTE + ENV_I_BUFF_BYTE) + +/* Size of the shareable memory region than can be reused */ +#define SBR_ENC_DYN_RAM_SIZE (BUF_QMF_SIZE + BUF_NRG_SIZE) + +#define OFFSET_QMF (0) +#define OFFSET_NRG (OFFSET_QMF + BUF_QMF_SIZE) + +/* + ***************************************************************************************************** + */ + +H_ALLOC_MEM(Ram_SbrDynamic_RAM, FIXP_DBL) + +H_ALLOC_MEM(Ram_SbrEncoder, SBR_ENCODER) +H_ALLOC_MEM(Ram_SbrChannel, SBR_CHANNEL) +H_ALLOC_MEM(Ram_SbrElement, SBR_ELEMENT) + +H_ALLOC_MEM(Ram_Sbr_quotaMatrix, FIXP_DBL) +H_ALLOC_MEM(Ram_Sbr_signMatrix, INT) + +H_ALLOC_MEM(Ram_Sbr_QmfStatesAnalysis, FIXP_QAS) + +H_ALLOC_MEM(Ram_Sbr_freqBandTableLO, UCHAR) +H_ALLOC_MEM(Ram_Sbr_freqBandTableHI, UCHAR) +H_ALLOC_MEM(Ram_Sbr_v_k_master, UCHAR) + +H_ALLOC_MEM(Ram_Sbr_detectionVectors, UCHAR) +H_ALLOC_MEM(Ram_Sbr_prevEnvelopeCompensation, UCHAR) +H_ALLOC_MEM(Ram_Sbr_guideScfb, UCHAR) +H_ALLOC_MEM(Ram_Sbr_guideVectorDetected, UCHAR) + +/* Dynamic Memory Allocation */ + +H_ALLOC_MEM(Ram_Sbr_envYBuffer, FIXP_DBL) +FIXP_DBL* GetRam_Sbr_envYBuffer(int n, UCHAR* dynamic_RAM); +FIXP_DBL* GetRam_Sbr_envRBuffer(int n, UCHAR* dynamic_RAM); +FIXP_DBL* GetRam_Sbr_envIBuffer(int n, UCHAR* dynamic_RAM); + +H_ALLOC_MEM(Ram_Sbr_guideVectorDiff, FIXP_DBL) +H_ALLOC_MEM(Ram_Sbr_guideVectorOrig, FIXP_DBL) + +H_ALLOC_MEM(Ram_PsQmfStatesSynthesis, FIXP_DBL) + +H_ALLOC_MEM(Ram_PsEncode, PS_ENCODE) + +FIXP_DBL* FDKsbrEnc_SliceRam_PsRqmf(FIXP_DBL* rQmfData, UCHAR* dynamic_RAM, + int n, int i, int qmfSlots); +FIXP_DBL* FDKsbrEnc_SliceRam_PsIqmf(FIXP_DBL* iQmfData, UCHAR* dynamic_RAM, + int n, int i, int qmfSlots); + +H_ALLOC_MEM(Ram_ParamStereo, PARAMETRIC_STEREO) +#endif diff --git a/libSBRenc/src/sbrenc_rom.cpp b/libSBRenc/src/sbrenc_rom.cpp new file mode 100644 index 0000000..737afaf --- /dev/null +++ b/libSBRenc/src/sbrenc_rom.cpp @@ -0,0 +1,910 @@ +/* ----------------------------------------------------------------------------- +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 encoder library ****************************** + + Author(s): Tobias Chalupka + + Description: Definition of constant tables + +*******************************************************************************/ + +/*! + \file + \brief Definition of constant tables + $Revision: 95404 $ + + This module contains most of the constant data that can be stored in ROM. +*/ + +#include "sbrenc_rom.h" +#include "genericStds.h" + +//@{ +/******************************************************************************* + + Table Overview: + + o envelope level, 1.5 dB: + 1a) v_Huff_envelopeLevelC10T[121] + 1b) v_Huff_envelopeLevelL10T[121] + 2a) v_Huff_envelopeLevelC10F[121] + 2b) v_Huff_envelopeLevelL10F[121] + + o envelope balance, 1.5 dB: + 3a) bookSbrEnvBalanceC10T[49] + 3b) bookSbrEnvBalanceL10T[49] + 4a) bookSbrEnvBalanceC10F[49] + 4b) bookSbrEnvBalanceL10F[49] + + o envelope level, 3.0 dB: + 5a) v_Huff_envelopeLevelC11T[63] + 5b) v_Huff_envelopeLevelL11T[63] + 6a) v_Huff_envelopeLevelC11F[63] + 6b) v_Huff_envelopeLevelC11F[63] + + o envelope balance, 3.0 dB: + 7a) bookSbrEnvBalanceC11T[25] + 7b) bookSbrEnvBalanceL11T[25] + 8a) bookSbrEnvBalanceC11F[25] + 8b) bookSbrEnvBalanceL11F[25] + + o noise level, 3.0 dB: + 9a) v_Huff_NoiseLevelC11T[63] + 9b) v_Huff_NoiseLevelL11T[63] + - ) (v_Huff_envelopeLevelC11F[63] is used for freq dir) + - ) (v_Huff_envelopeLevelL11F[63] is used for freq dir) + + o noise balance, 3.0 dB: + 10a) bookSbrNoiseBalanceC11T[25] + 10b) bookSbrNoiseBalanceL11T[25] + - ) (bookSbrEnvBalanceC11F[25] is used for freq dir) + - ) (bookSbrEnvBalanceL11F[25] is used for freq dir) + + + (1.5 dB is never used for noise) + +********************************************************************************/ + +/*******************************************************************************/ +/* table : envelope level, 1.5 dB */ +/* theor range : [-58,58], CODE_BOOK_SCF_LAV = 58 */ +/* implem range: [-60,60], CODE_BOOK_SCF_LAV10 = 60 */ +/* raw stats : envelopeLevel_00 (yes, wrong suffix in name) KK 01-03-09 */ +/*******************************************************************************/ + +/* direction: time + contents : codewords + raw table: HuffCode3C2FIX.m/envelopeLevel_00T_cF.mat/v_nChex_cF + built by : FH 01-07-05 */ + +const INT v_Huff_envelopeLevelC10T[121] = { + 0x0003FFD6, 0x0003FFD7, 0x0003FFD8, 0x0003FFD9, 0x0003FFDA, 0x0003FFDB, + 0x0007FFB8, 0x0007FFB9, 0x0007FFBA, 0x0007FFBB, 0x0007FFBC, 0x0007FFBD, + 0x0007FFBE, 0x0007FFBF, 0x0007FFC0, 0x0007FFC1, 0x0007FFC2, 0x0007FFC3, + 0x0007FFC4, 0x0007FFC5, 0x0007FFC6, 0x0007FFC7, 0x0007FFC8, 0x0007FFC9, + 0x0007FFCA, 0x0007FFCB, 0x0007FFCC, 0x0007FFCD, 0x0007FFCE, 0x0007FFCF, + 0x0007FFD0, 0x0007FFD1, 0x0007FFD2, 0x0007FFD3, 0x0001FFE6, 0x0003FFD4, + 0x0000FFF0, 0x0001FFE9, 0x0003FFD5, 0x0001FFE7, 0x0000FFF1, 0x0000FFEC, + 0x0000FFED, 0x0000FFEE, 0x00007FF4, 0x00003FF9, 0x00003FF7, 0x00001FFA, + 0x00001FF9, 0x00000FFB, 0x000007FC, 0x000003FC, 0x000001FD, 0x000000FD, + 0x0000007D, 0x0000003D, 0x0000001D, 0x0000000D, 0x00000005, 0x00000001, + 0x00000000, 0x00000004, 0x0000000C, 0x0000001C, 0x0000003C, 0x0000007C, + 0x000000FC, 0x000001FC, 0x000003FD, 0x00000FFA, 0x00001FF8, 0x00003FF6, + 0x00003FF8, 0x00007FF5, 0x0000FFEF, 0x0001FFE8, 0x0000FFF2, 0x0007FFD4, + 0x0007FFD5, 0x0007FFD6, 0x0007FFD7, 0x0007FFD8, 0x0007FFD9, 0x0007FFDA, + 0x0007FFDB, 0x0007FFDC, 0x0007FFDD, 0x0007FFDE, 0x0007FFDF, 0x0007FFE0, + 0x0007FFE1, 0x0007FFE2, 0x0007FFE3, 0x0007FFE4, 0x0007FFE5, 0x0007FFE6, + 0x0007FFE7, 0x0007FFE8, 0x0007FFE9, 0x0007FFEA, 0x0007FFEB, 0x0007FFEC, + 0x0007FFED, 0x0007FFEE, 0x0007FFEF, 0x0007FFF0, 0x0007FFF1, 0x0007FFF2, + 0x0007FFF3, 0x0007FFF4, 0x0007FFF5, 0x0007FFF6, 0x0007FFF7, 0x0007FFF8, + 0x0007FFF9, 0x0007FFFA, 0x0007FFFB, 0x0007FFFC, 0x0007FFFD, 0x0007FFFE, + 0x0007FFFF}; + +/* direction: time + contents : codeword lengths + raw table: HuffCode3C2FIX.m/envelopeLevel_00T_cF.mat/v_nLhex_cF + built by : FH 01-07-05 */ + +const UCHAR v_Huff_envelopeLevelL10T[121] = { + 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x13, 0x13, 0x13, 0x13, 0x13, + 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, + 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, + 0x13, 0x11, 0x12, 0x10, 0x11, 0x12, 0x11, 0x10, 0x10, 0x10, 0x10, + 0x0F, 0x0E, 0x0E, 0x0D, 0x0D, 0x0C, 0x0B, 0x0A, 0x09, 0x08, 0x07, + 0x06, 0x05, 0x04, 0x03, 0x02, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07, + 0x08, 0x09, 0x0A, 0x0C, 0x0D, 0x0E, 0x0E, 0x0F, 0x10, 0x11, 0x10, + 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, + 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, + 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, + 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13}; + +/* direction: freq + contents : codewords + raw table: HuffCode3C2FIX.m/envelopeLevel_00F_cF.mat/v_nChex_cF + built by : FH 01-07-05 */ + +const INT v_Huff_envelopeLevelC10F[121] = { + 0x0007FFE7, 0x0007FFE8, 0x000FFFD2, 0x000FFFD3, 0x000FFFD4, 0x000FFFD5, + 0x000FFFD6, 0x000FFFD7, 0x000FFFD8, 0x0007FFDA, 0x000FFFD9, 0x000FFFDA, + 0x000FFFDB, 0x000FFFDC, 0x0007FFDB, 0x000FFFDD, 0x0007FFDC, 0x0007FFDD, + 0x000FFFDE, 0x0003FFE4, 0x000FFFDF, 0x000FFFE0, 0x000FFFE1, 0x0007FFDE, + 0x000FFFE2, 0x000FFFE3, 0x000FFFE4, 0x0007FFDF, 0x000FFFE5, 0x0007FFE0, + 0x0003FFE8, 0x0007FFE1, 0x0003FFE0, 0x0003FFE9, 0x0001FFEF, 0x0003FFE5, + 0x0001FFEC, 0x0001FFED, 0x0001FFEE, 0x0000FFF4, 0x0000FFF3, 0x0000FFF0, + 0x00007FF7, 0x00007FF6, 0x00003FFA, 0x00001FFA, 0x00001FF9, 0x00000FFA, + 0x00000FF8, 0x000007F9, 0x000003FB, 0x000001FC, 0x000001FA, 0x000000FB, + 0x0000007C, 0x0000003C, 0x0000001C, 0x0000000C, 0x00000005, 0x00000001, + 0x00000000, 0x00000004, 0x0000000D, 0x0000001D, 0x0000003D, 0x000000FA, + 0x000000FC, 0x000001FB, 0x000003FA, 0x000007F8, 0x000007FA, 0x000007FB, + 0x00000FF9, 0x00000FFB, 0x00001FF8, 0x00001FFB, 0x00003FF8, 0x00003FF9, + 0x0000FFF1, 0x0000FFF2, 0x0001FFEA, 0x0001FFEB, 0x0003FFE1, 0x0003FFE2, + 0x0003FFEA, 0x0003FFE3, 0x0003FFE6, 0x0003FFE7, 0x0003FFEB, 0x000FFFE6, + 0x0007FFE2, 0x000FFFE7, 0x000FFFE8, 0x000FFFE9, 0x000FFFEA, 0x000FFFEB, + 0x000FFFEC, 0x0007FFE3, 0x000FFFED, 0x000FFFEE, 0x000FFFEF, 0x000FFFF0, + 0x0007FFE4, 0x000FFFF1, 0x0003FFEC, 0x000FFFF2, 0x000FFFF3, 0x0007FFE5, + 0x0007FFE6, 0x000FFFF4, 0x000FFFF5, 0x000FFFF6, 0x000FFFF7, 0x000FFFF8, + 0x000FFFF9, 0x000FFFFA, 0x000FFFFB, 0x000FFFFC, 0x000FFFFD, 0x000FFFFE, + 0x000FFFFF}; + +/* direction: freq + contents : codeword lengths + raw table: HuffCode3C2FIX.m/envelopeLevel_00F_cF.mat/v_nLhex_cF + built by : FH 01-07-05 */ + +const UCHAR v_Huff_envelopeLevelL10F[121] = { + 0x13, 0x13, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x13, 0x14, + 0x14, 0x14, 0x14, 0x13, 0x14, 0x13, 0x13, 0x14, 0x12, 0x14, 0x14, + 0x14, 0x13, 0x14, 0x14, 0x14, 0x13, 0x14, 0x13, 0x12, 0x13, 0x12, + 0x12, 0x11, 0x12, 0x11, 0x11, 0x11, 0x10, 0x10, 0x10, 0x0F, 0x0F, + 0x0E, 0x0D, 0x0D, 0x0C, 0x0C, 0x0B, 0x0A, 0x09, 0x09, 0x08, 0x07, + 0x06, 0x05, 0x04, 0x03, 0x02, 0x02, 0x03, 0x04, 0x05, 0x06, 0x08, + 0x08, 0x09, 0x0A, 0x0B, 0x0B, 0x0B, 0x0C, 0x0C, 0x0D, 0x0D, 0x0E, + 0x0E, 0x10, 0x10, 0x11, 0x11, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, + 0x12, 0x14, 0x13, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x13, 0x14, + 0x14, 0x14, 0x14, 0x13, 0x14, 0x12, 0x14, 0x14, 0x13, 0x13, 0x14, + 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14}; + +/*******************************************************************************/ +/* table : envelope balance, 1.5 dB */ +/* theor range : [-48,48], CODE_BOOK_SCF_LAV = 48 */ +/* implem range: same but mapped to [-24,24], CODE_BOOK_SCF_LAV_BALANCE10 = 24 + */ +/* raw stats : envelopePan_00 (yes, wrong suffix in name) KK 01-03-09 */ +/*******************************************************************************/ + +/* direction: time + contents : codewords + raw table: HuffCode3C.m/envelopePan_00T.mat/v_nBhex + built by : FH 01-05-15 */ + +const INT bookSbrEnvBalanceC10T[49] = { + 0x0000FFE4, 0x0000FFE5, 0x0000FFE6, 0x0000FFE7, 0x0000FFE8, 0x0000FFE9, + 0x0000FFEA, 0x0000FFEB, 0x0000FFEC, 0x0000FFED, 0x0000FFEE, 0x0000FFEF, + 0x0000FFF0, 0x0000FFF1, 0x0000FFF2, 0x0000FFF3, 0x0000FFF4, 0x0000FFE2, + 0x00000FFC, 0x000007FC, 0x000001FE, 0x0000007E, 0x0000001E, 0x00000006, + 0x00000000, 0x00000002, 0x0000000E, 0x0000003E, 0x000000FE, 0x000007FD, + 0x00000FFD, 0x00007FF0, 0x0000FFE3, 0x0000FFF5, 0x0000FFF6, 0x0000FFF7, + 0x0000FFF8, 0x0000FFF9, 0x0000FFFA, 0x0001FFF6, 0x0001FFF7, 0x0001FFF8, + 0x0001FFF9, 0x0001FFFA, 0x0001FFFB, 0x0001FFFC, 0x0001FFFD, 0x0001FFFE, + 0x0001FFFF}; + +/* direction: time + contents : codeword lengths + raw table: HuffCode3C.m/envelopePan_00T.mat/v_nLhex + built by : FH 01-05-15 */ + +const UCHAR bookSbrEnvBalanceL10T[49] = { + 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, + 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x0C, 0x0B, + 0x09, 0x07, 0x05, 0x03, 0x01, 0x02, 0x04, 0x06, 0x08, 0x0B, + 0x0C, 0x0F, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x11, + 0x11, 0x11, 0x11, 0x11, 0x11, 0x11, 0x11, 0x11, 0x11}; + +/* direction: freq + contents : codewords + raw table: HuffCode3C.m/envelopePan_00F.mat/v_nBhex + built by : FH 01-05-15 */ + +const INT bookSbrEnvBalanceC10F[49] = { + 0x0003FFE2, 0x0003FFE3, 0x0003FFE4, 0x0003FFE5, 0x0003FFE6, 0x0003FFE7, + 0x0003FFE8, 0x0003FFE9, 0x0003FFEA, 0x0003FFEB, 0x0003FFEC, 0x0003FFED, + 0x0003FFEE, 0x0003FFEF, 0x0003FFF0, 0x0000FFF7, 0x0001FFF0, 0x00003FFC, + 0x000007FE, 0x000007FC, 0x000000FE, 0x0000007E, 0x0000000E, 0x00000002, + 0x00000000, 0x00000006, 0x0000001E, 0x0000003E, 0x000001FE, 0x000007FD, + 0x00000FFE, 0x00007FFA, 0x0000FFF6, 0x0003FFF1, 0x0003FFF2, 0x0003FFF3, + 0x0003FFF4, 0x0003FFF5, 0x0003FFF6, 0x0003FFF7, 0x0003FFF8, 0x0003FFF9, + 0x0003FFFA, 0x0003FFFB, 0x0003FFFC, 0x0003FFFD, 0x0003FFFE, 0x0007FFFE, + 0x0007FFFF}; + +/* direction: freq + contents : codeword lengths + raw table: HuffCode3C.m/envelopePan_00F.mat/v_nLhex + built by : FH 01-05-15 */ + +const UCHAR bookSbrEnvBalanceL10F[49] = { + 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, + 0x12, 0x12, 0x12, 0x12, 0x12, 0x10, 0x11, 0x0E, 0x0B, 0x0B, + 0x08, 0x07, 0x04, 0x02, 0x01, 0x03, 0x05, 0x06, 0x09, 0x0B, + 0x0C, 0x0F, 0x10, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, + 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x12, 0x13, 0x13}; + +/*******************************************************************************/ +/* table : envelope level, 3.0 dB */ +/* theor range : [-29,29], CODE_BOOK_SCF_LAV = 29 */ +/* implem range: [-31,31], CODE_BOOK_SCF_LAV11 = 31 */ +/* raw stats : envelopeLevel_11 KK 00-02-03 */ +/*******************************************************************************/ + +/* direction: time + contents : codewords + raw table: HuffCode2.m + built by : FH 00-02-04 */ + +const INT v_Huff_envelopeLevelC11T[63] = { + 0x0003FFED, 0x0003FFEE, 0x0007FFDE, 0x0007FFDF, 0x0007FFE0, 0x0007FFE1, + 0x0007FFE2, 0x0007FFE3, 0x0007FFE4, 0x0007FFE5, 0x0007FFE6, 0x0007FFE7, + 0x0007FFE8, 0x0007FFE9, 0x0007FFEA, 0x0007FFEB, 0x0007FFEC, 0x0001FFF4, + 0x0000FFF7, 0x0000FFF9, 0x0000FFF8, 0x00003FFB, 0x00003FFA, 0x00003FF8, + 0x00001FFA, 0x00000FFC, 0x000007FC, 0x000000FE, 0x0000003E, 0x0000000E, + 0x00000002, 0x00000000, 0x00000006, 0x0000001E, 0x0000007E, 0x000001FE, + 0x000007FD, 0x00001FFB, 0x00003FF9, 0x00003FFC, 0x00007FFA, 0x0000FFF6, + 0x0001FFF5, 0x0003FFEC, 0x0007FFED, 0x0007FFEE, 0x0007FFEF, 0x0007FFF0, + 0x0007FFF1, 0x0007FFF2, 0x0007FFF3, 0x0007FFF4, 0x0007FFF5, 0x0007FFF6, + 0x0007FFF7, 0x0007FFF8, 0x0007FFF9, 0x0007FFFA, 0x0007FFFB, 0x0007FFFC, + 0x0007FFFD, 0x0007FFFE, 0x0007FFFF}; + +/* direction: time + contents : codeword lengths + raw table: HuffCode2.m + built by : FH 00-02-04 */ + +const UCHAR v_Huff_envelopeLevelL11T[63] = { + 0x12, 0x12, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, + 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x11, 0x10, 0x10, 0x10, 0x0E, + 0x0E, 0x0E, 0x0D, 0x0C, 0x0B, 0x08, 0x06, 0x04, 0x02, 0x01, 0x03, + 0x05, 0x07, 0x09, 0x0B, 0x0D, 0x0E, 0x0E, 0x0F, 0x10, 0x11, 0x12, + 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, + 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13}; + +/* direction: freq + contents : codewords + raw table: HuffCode2.m + built by : FH 00-02-04 */ + +const INT v_Huff_envelopeLevelC11F[63] = { + 0x000FFFF0, 0x000FFFF1, 0x000FFFF2, 0x000FFFF3, 0x000FFFF4, 0x000FFFF5, + 0x000FFFF6, 0x0003FFF3, 0x0007FFF5, 0x0007FFEE, 0x0007FFEF, 0x0007FFF6, + 0x0003FFF4, 0x0003FFF2, 0x000FFFF7, 0x0007FFF0, 0x0001FFF5, 0x0003FFF0, + 0x0001FFF4, 0x0000FFF7, 0x0000FFF6, 0x00007FF8, 0x00003FFB, 0x00000FFD, + 0x000007FD, 0x000003FD, 0x000001FD, 0x000000FD, 0x0000003E, 0x0000000E, + 0x00000002, 0x00000000, 0x00000006, 0x0000001E, 0x000000FC, 0x000001FC, + 0x000003FC, 0x000007FC, 0x00000FFC, 0x00001FFC, 0x00003FFA, 0x00007FF9, + 0x00007FFA, 0x0000FFF8, 0x0000FFF9, 0x0001FFF6, 0x0001FFF7, 0x0003FFF5, + 0x0003FFF6, 0x0003FFF1, 0x000FFFF8, 0x0007FFF1, 0x0007FFF2, 0x0007FFF3, + 0x000FFFF9, 0x0007FFF7, 0x0007FFF4, 0x000FFFFA, 0x000FFFFB, 0x000FFFFC, + 0x000FFFFD, 0x000FFFFE, 0x000FFFFF}; + +/* direction: freq + contents : codeword lengths + raw table: HuffCode2.m + built by : FH 00-02-04 */ + +const UCHAR v_Huff_envelopeLevelL11F[63] = { + 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14, 0x12, 0x13, 0x13, 0x13, + 0x13, 0x12, 0x12, 0x14, 0x13, 0x11, 0x12, 0x11, 0x10, 0x10, 0x0F, + 0x0E, 0x0C, 0x0B, 0x0A, 0x09, 0x08, 0x06, 0x04, 0x02, 0x01, 0x03, + 0x05, 0x08, 0x09, 0x0A, 0x0B, 0x0C, 0x0D, 0x0E, 0x0F, 0x0F, 0x10, + 0x10, 0x11, 0x11, 0x12, 0x12, 0x12, 0x14, 0x13, 0x13, 0x13, 0x14, + 0x13, 0x13, 0x14, 0x14, 0x14, 0x14, 0x14, 0x14}; + +/*******************************************************************************/ +/* table : envelope balance, 3.0 dB */ +/* theor range : [-24,24], CODE_BOOK_SCF_LAV = 24 */ +/* implem range: same but mapped to [-12,12], CODE_BOOK_SCF_LAV_BALANCE11 = 12 + */ +/* raw stats : envelopeBalance_11 KK 00-02-03 */ +/*******************************************************************************/ + +/* direction: time + contents : codewords + raw table: HuffCode3C.m/envelopeBalance_11T.mat/v_nBhex + built by : FH 01-05-15 */ + +const INT bookSbrEnvBalanceC11T[25] = { + 0x00001FF2, 0x00001FF3, 0x00001FF4, 0x00001FF5, 0x00001FF6, + 0x00001FF7, 0x00001FF8, 0x00000FF8, 0x000000FE, 0x0000007E, + 0x0000000E, 0x00000006, 0x00000000, 0x00000002, 0x0000001E, + 0x0000003E, 0x000001FE, 0x00001FF9, 0x00001FFA, 0x00001FFB, + 0x00001FFC, 0x00001FFD, 0x00001FFE, 0x00003FFE, 0x00003FFF}; + +/* direction: time + contents : codeword lengths + raw table: HuffCode3C.m/envelopeBalance_11T.mat/v_nLhex + built by : FH 01-05-15 */ + +const UCHAR bookSbrEnvBalanceL11T[25] = { + 0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0C, 0x08, + 0x07, 0x04, 0x03, 0x01, 0x02, 0x05, 0x06, 0x09, 0x0D, + 0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0E, 0x0E}; + +/* direction: freq + contents : codewords + raw table: HuffCode3C.m/envelopeBalance_11F.mat/v_nBhex + built by : FH 01-05-15 */ + +const INT bookSbrEnvBalanceC11F[25] = { + 0x00001FF7, 0x00001FF8, 0x00001FF9, 0x00001FFA, 0x00001FFB, + 0x00003FF8, 0x00003FF9, 0x000007FC, 0x000000FE, 0x0000007E, + 0x0000000E, 0x00000002, 0x00000000, 0x00000006, 0x0000001E, + 0x0000003E, 0x000001FE, 0x00000FFA, 0x00001FF6, 0x00003FFA, + 0x00003FFB, 0x00003FFC, 0x00003FFD, 0x00003FFE, 0x00003FFF}; + +/* direction: freq + contents : codeword lengths + raw table: HuffCode3C.m/envelopeBalance_11F.mat/v_nLhex + built by : FH 01-05-15 */ + +const UCHAR bookSbrEnvBalanceL11F[25] = { + 0x0D, 0x0D, 0x0D, 0x0D, 0x0D, 0x0E, 0x0E, 0x0B, 0x08, + 0x07, 0x04, 0x02, 0x01, 0x03, 0x05, 0x06, 0x09, 0x0C, + 0x0D, 0x0E, 0x0E, 0x0E, 0x0E, 0x0E, 0x0E}; + +/*******************************************************************************/ +/* table : noise level, 3.0 dB */ +/* theor range : [-29,29], CODE_BOOK_SCF_LAV = 29 */ +/* implem range: [-31,31], CODE_BOOK_SCF_LAV11 = 31 */ +/* raw stats : noiseLevel_11 KK 00-02-03 */ +/*******************************************************************************/ + +/* direction: time + contents : codewords + raw table: HuffCode2.m + built by : FH 00-02-04 */ + +const INT v_Huff_NoiseLevelC11T[63] = { + 0x00001FCE, 0x00001FCF, 0x00001FD0, 0x00001FD1, 0x00001FD2, 0x00001FD3, + 0x00001FD4, 0x00001FD5, 0x00001FD6, 0x00001FD7, 0x00001FD8, 0x00001FD9, + 0x00001FDA, 0x00001FDB, 0x00001FDC, 0x00001FDD, 0x00001FDE, 0x00001FDF, + 0x00001FE0, 0x00001FE1, 0x00001FE2, 0x00001FE3, 0x00001FE4, 0x00001FE5, + 0x00001FE6, 0x00001FE7, 0x000007F2, 0x000000FD, 0x0000003E, 0x0000000E, + 0x00000006, 0x00000000, 0x00000002, 0x0000001E, 0x000000FC, 0x000003F8, + 0x00001FCC, 0x00001FE8, 0x00001FE9, 0x00001FEA, 0x00001FEB, 0x00001FEC, + 0x00001FCD, 0x00001FED, 0x00001FEE, 0x00001FEF, 0x00001FF0, 0x00001FF1, + 0x00001FF2, 0x00001FF3, 0x00001FF4, 0x00001FF5, 0x00001FF6, 0x00001FF7, + 0x00001FF8, 0x00001FF9, 0x00001FFA, 0x00001FFB, 0x00001FFC, 0x00001FFD, + 0x00001FFE, 0x00003FFE, 0x00003FFF}; + +/* direction: time + contents : codeword lengths + raw table: HuffCode2.m + built by : FH 00-02-04 */ + +const UCHAR v_Huff_NoiseLevelL11T[63] = { + 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, + 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, + 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, + 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, + 0x0000000D, 0x0000000D, 0x0000000B, 0x00000008, 0x00000006, 0x00000004, + 0x00000003, 0x00000001, 0x00000002, 0x00000005, 0x00000008, 0x0000000A, + 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, + 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, + 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, + 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, 0x0000000D, + 0x0000000D, 0x0000000E, 0x0000000E}; + +/*******************************************************************************/ +/* table : noise balance, 3.0 dB */ +/* theor range : [-24,24], CODE_BOOK_SCF_LAV = 24 */ +/* implem range: same but mapped to [-12,12], CODE_BOOK_SCF_LAV_BALANCE11 = 12 + */ +/* raw stats : noiseBalance_11 KK 00-02-03 */ +/*******************************************************************************/ + +/* direction: time + contents : codewords + raw table: HuffCode3C.m/noiseBalance_11.mat/v_nBhex + built by : FH 01-05-15 */ + +const INT bookSbrNoiseBalanceC11T[25] = { + 0x000000EC, 0x000000ED, 0x000000EE, 0x000000EF, 0x000000F0, + 0x000000F1, 0x000000F2, 0x000000F3, 0x000000F4, 0x000000F5, + 0x0000001C, 0x00000002, 0x00000000, 0x00000006, 0x0000003A, + 0x000000F6, 0x000000F7, 0x000000F8, 0x000000F9, 0x000000FA, + 0x000000FB, 0x000000FC, 0x000000FD, 0x000000FE, 0x000000FF}; + +/* direction: time + contents : codeword lengths + raw table: HuffCode3C.m/noiseBalance_11.mat/v_nLhex + built by : FH 01-05-15 */ + +const UCHAR bookSbrNoiseBalanceL11T[25] = { + 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, + 0x08, 0x05, 0x02, 0x01, 0x03, 0x06, 0x08, 0x08, 0x08, + 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08}; + +/* + tuningTable +*/ +const sbrTuningTable_t sbrTuningTable[] = { + /* Some of the low bitrates are commented out here, this is because the + encoder could lose frames at those bitrates and throw an error + because it has insufficient bits to encode for some test items. + */ + + /*** HE-AAC section ***/ + /* sf,sfsp,sf,sfsp,nnb,nfo,saml,SM,FS*/ + + /*** mono ***/ + + /* 8/16 kHz dual rate */ + {CODEC_AAC, 8000, 10000, 8000, 1, 7, 6, 11, 10, 1, 0, 6, SBR_MONO, 3}, + {CODEC_AAC, 10000, 12000, 8000, 1, 11, 7, 13, 12, 1, 0, 6, SBR_MONO, 3}, + {CODEC_AAC, 12000, 16001, 8000, 1, 14, 10, 13, 13, 1, 0, 6, SBR_MONO, 3}, + {CODEC_AAC, 16000, 24000, 8000, 1, 14, 10, 14, 14, 2, 0, 3, SBR_MONO, 2}, + {CODEC_AAC, 24000, 32000, 8000, 1, 14, 10, 14, 14, 2, 0, 3, SBR_MONO, 2}, + {CODEC_AAC, 32000, 48001, 8000, 1, 14, 11, 15, 15, 2, 0, 3, SBR_MONO, 2}, + + /* 11/22 kHz dual rate */ + {CODEC_AAC, 8000, 10000, 11025, 1, 5, 4, 6, 6, 1, 0, 6, SBR_MONO, 3}, + {CODEC_AAC, 10000, 12000, 11025, 1, 8, 5, 12, 9, 1, 0, 6, SBR_MONO, 3}, + {CODEC_AAC, 12000, 16000, 11025, 1, 12, 8, 13, 8, 1, 0, 6, SBR_MONO, 3}, + {CODEC_AAC, 16000, 20000, 11025, 1, 12, 8, 13, 8, 1, 0, 6, SBR_MONO, 3}, + {CODEC_AAC, 20000, 24001, 11025, 1, 13, 9, 13, 8, 1, 0, 6, SBR_MONO, 3}, + {CODEC_AAC, 24000, 32000, 11025, 1, 14, 10, 14, 9, 2, 0, 3, SBR_MONO, 2}, + {CODEC_AAC, 32000, 48000, 11025, 1, 15, 11, 15, 10, 2, 0, 3, SBR_MONO, 2}, + {CODEC_AAC, 48000, 64001, 11025, 1, 15, 11, 15, 10, 2, 0, 3, SBR_MONO, 1}, + + /* 12/24 kHz dual rate */ + {CODEC_AAC, 8000, 10000, 12000, 1, 4, 3, 6, 6, 1, 0, 6, SBR_MONO, 3}, + {CODEC_AAC, 10000, 12000, 12000, 1, 7, 4, 11, 8, 1, 0, 6, SBR_MONO, 3}, + {CODEC_AAC, 12000, 16000, 12000, 1, 11, 7, 12, 8, 1, 0, 6, SBR_MONO, 3}, + {CODEC_AAC, 16000, 20000, 12000, 1, 11, 7, 12, 8, 1, 0, 6, SBR_MONO, 3}, + {CODEC_AAC, 20000, 24001, 12000, 1, 12, 8, 12, 8, 1, 0, 6, SBR_MONO, 3}, + {CODEC_AAC, 24000, 32000, 12000, 1, 13, 9, 13, 9, 2, 0, 3, SBR_MONO, 2}, + {CODEC_AAC, 32000, 48000, 12000, 1, 14, 10, 14, 10, 2, 0, 3, SBR_MONO, 2}, + {CODEC_AAC, 48000, 64001, 12000, 1, 14, 11, 15, 11, 2, 0, 3, SBR_MONO, 1}, + + /* 16/32 kHz dual rate */ + {CODEC_AAC, 8000, 10000, 16000, 1, 1, 1, 0, 0, 1, 0, 6, SBR_MONO, 3}, + {CODEC_AAC, 10000, 12000, 16000, 1, 2, 1, 6, 0, 1, 0, 6, SBR_MONO, 3}, + {CODEC_AAC, 12000, 16000, 16000, 1, 4, 2, 6, 0, 1, 0, 6, SBR_MONO, 3}, + {CODEC_AAC, 16000, 18000, 16000, 1, 4, 2, 8, 3, 1, 0, 6, SBR_MONO, 3}, + {CODEC_AAC, 18000, 22000, 16000, 1, 6, 5, 11, 7, 2, 0, 6, SBR_MONO, 2}, + {CODEC_AAC, 22000, 28000, 16000, 1, 10, 9, 12, 8, 2, 0, 6, SBR_MONO, 2}, + {CODEC_AAC, 28000, 36000, 16000, 1, 12, 12, 13, 13, 2, 0, 3, SBR_MONO, 2}, + {CODEC_AAC, 36000, 44000, 16000, 1, 14, 14, 13, 13, 2, 0, 3, SBR_MONO, 1}, + {CODEC_AAC, 44000, 64001, 16000, 1, 14, 14, 13, 13, 2, 0, 3, SBR_MONO, 1}, + + /* 22.05/44.1 kHz dual rate */ + /* { CODEC_AAC, 8000, 11369, 22050, 1, 1, 1, 1, 1, 1, 0, 6, + SBR_MONO, 3 }, */ + {CODEC_AAC, 11369, 16000, 22050, 1, 3, 1, 4, 4, 1, 0, 6, SBR_MONO, 3}, + {CODEC_AAC, 16000, 18000, 22050, 1, 3, 1, 5, 4, 1, 0, 6, SBR_MONO, 3}, + {CODEC_AAC, 18000, 22000, 22050, 1, 4, 4, 8, 5, 2, 0, 6, SBR_MONO, 2}, + {CODEC_AAC, 22000, 28000, 22050, 1, 7, 6, 8, 6, 2, 0, 6, SBR_MONO, 2}, + {CODEC_AAC, 28000, 36000, 22050, 1, 10, 10, 9, 9, 2, 0, 3, SBR_MONO, 2}, + {CODEC_AAC, 36000, 44000, 22050, 1, 11, 11, 10, 10, 2, 0, 3, SBR_MONO, 1}, + {CODEC_AAC, 44000, 64001, 22050, 1, 13, 13, 12, 12, 2, 0, 3, SBR_MONO, 1}, + + /* 24/48 kHz dual rate */ + /* { CODEC_AAC, 8000, 12000, 24000, 1, 1, 1, 1, 1, 1, 0, 6, + SBR_MONO, 3 }, */ + {CODEC_AAC, 12000, 16000, 24000, 1, 3, 1, 4, 4, 1, 0, 6, SBR_MONO, 3}, + {CODEC_AAC, 16000, 18000, 24000, 1, 3, 1, 5, 4, 1, 0, 6, SBR_MONO, 3}, + {CODEC_AAC, 18000, 22000, 24000, 1, 4, 3, 8, 5, 2, 0, 6, SBR_MONO, 2}, + {CODEC_AAC, 22000, 28000, 24000, 1, 7, 6, 8, 6, 2, 0, 6, SBR_MONO, 2}, + {CODEC_AAC, 28000, 36000, 24000, 1, 10, 10, 9, 9, 2, 0, 3, SBR_MONO, 2}, + {CODEC_AAC, 36000, 44000, 24000, 1, 11, 11, 10, 10, 2, 0, 3, SBR_MONO, 1}, + {CODEC_AAC, 44000, 64001, 24000, 1, 13, 13, 11, 11, 2, 0, 3, SBR_MONO, 1}, + + /* 32/64 kHz dual rate */ + {CODEC_AAC, 24000, 36000, 32000, 1, 4, 4, 4, 4, 2, 0, 3, SBR_MONO, 3}, + {CODEC_AAC, 36000, 60000, 32000, 1, 7, 7, 6, 6, 2, 0, 3, SBR_MONO, 2}, + {CODEC_AAC, 60000, 72000, 32000, 1, 9, 9, 8, 8, 2, 0, 3, SBR_MONO, 1}, + {CODEC_AAC, 72000, 100000, 32000, 1, 11, 11, 10, 10, 2, 0, 3, SBR_MONO, 1}, + {CODEC_AAC, 100000, 160001, 32000, 1, 13, 13, 11, 11, 2, 0, 3, SBR_MONO, 1}, + + /* 44.1/88.2 kHz dual rate */ + {CODEC_AAC, 24000, 36000, 44100, 1, 4, 4, 4, 4, 2, 0, 3, SBR_MONO, 3}, + {CODEC_AAC, 36000, 60000, 44100, 1, 7, 7, 6, 6, 2, 0, 3, SBR_MONO, 2}, + {CODEC_AAC, 60000, 72000, 44100, 1, 9, 9, 8, 8, 2, 0, 3, SBR_MONO, 1}, + {CODEC_AAC, 72000, 100000, 44100, 1, 11, 11, 10, 10, 2, 0, 3, SBR_MONO, 1}, + {CODEC_AAC, 100000, 160001, 44100, 1, 13, 13, 11, 11, 2, 0, 3, SBR_MONO, 1}, + + /* 48/96 kHz dual rate */ + {CODEC_AAC, 32000, 36000, 48000, 1, 4, 4, 9, 9, 2, 0, 3, SBR_MONO, 3}, + {CODEC_AAC, 36000, 60000, 48000, 1, 7, 7, 10, 10, 2, 0, 3, SBR_MONO, 2}, + {CODEC_AAC, 60000, 72000, 48000, 1, 9, 9, 10, 10, 2, 0, 3, SBR_MONO, 1}, + {CODEC_AAC, 72000, 100000, 48000, 1, 11, 11, 11, 11, 2, 0, 3, SBR_MONO, 1}, + {CODEC_AAC, 100000, 160001, 48000, 1, 13, 13, 11, 11, 2, 0, 3, SBR_MONO, 1}, + + /*** stereo ***/ + /* 08/16 kHz dual rate */ + {CODEC_AAC, 16000, 24000, 8000, 2, 6, 6, 9, 7, 1, 0, -3, SBR_SWITCH_LRC, 3}, + {CODEC_AAC, 24000, 28000, 8000, 2, 9, 9, 11, 9, 1, 0, -3, SBR_SWITCH_LRC, + 3}, + {CODEC_AAC, 28000, 36000, 8000, 2, 11, 9, 11, 9, 2, 0, -3, SBR_SWITCH_LRC, + 2}, + {CODEC_AAC, 36000, 44000, 8000, 2, 13, 11, 13, 11, 2, 0, -3, SBR_SWITCH_LRC, + 2}, + {CODEC_AAC, 44000, 52000, 8000, 2, 14, 12, 13, 12, 2, 0, -3, SBR_SWITCH_LRC, + 2}, + {CODEC_AAC, 52000, 60000, 8000, 2, 14, 14, 13, 13, 3, 0, -3, SBR_SWITCH_LRC, + 1}, + {CODEC_AAC, 60000, 76000, 8000, 2, 14, 14, 13, 13, 3, 0, -3, SBR_LEFT_RIGHT, + 1}, + {CODEC_AAC, 76000, 128001, 8000, 2, 14, 14, 13, 13, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + + /* 11/22 kHz dual rate */ + {CODEC_AAC, 16000, 24000, 11025, 2, 7, 5, 9, 7, 1, 0, -3, SBR_SWITCH_LRC, + 3}, + {CODEC_AAC, 24000, 28000, 11025, 2, 10, 8, 10, 8, 1, 0, -3, SBR_SWITCH_LRC, + 3}, + {CODEC_AAC, 28000, 36000, 11025, 2, 12, 8, 12, 8, 2, 0, -3, SBR_SWITCH_LRC, + 2}, + {CODEC_AAC, 36000, 44000, 11025, 2, 13, 9, 13, 9, 2, 0, -3, SBR_SWITCH_LRC, + 2}, + {CODEC_AAC, 44000, 52000, 11025, 2, 14, 11, 13, 11, 2, 0, -3, + SBR_SWITCH_LRC, 2}, + {CODEC_AAC, 52000, 60000, 11025, 2, 15, 15, 13, 13, 3, 0, -3, + SBR_SWITCH_LRC, 1}, + {CODEC_AAC, 60000, 76000, 11025, 2, 15, 15, 13, 13, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + {CODEC_AAC, 76000, 128001, 11025, 2, 15, 15, 13, 13, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + + /* 12/24 kHz dual rate */ + {CODEC_AAC, 16000, 24000, 12000, 2, 6, 4, 9, 7, 1, 0, -3, SBR_SWITCH_LRC, + 3}, + {CODEC_AAC, 24000, 28000, 12000, 2, 9, 7, 10, 8, 1, 0, -3, SBR_SWITCH_LRC, + 3}, + {CODEC_AAC, 28000, 36000, 12000, 2, 11, 7, 12, 8, 2, 0, -3, SBR_SWITCH_LRC, + 2}, + {CODEC_AAC, 36000, 44000, 12000, 2, 12, 9, 12, 9, 2, 0, -3, SBR_SWITCH_LRC, + 2}, + {CODEC_AAC, 44000, 52000, 12000, 2, 13, 12, 13, 12, 2, 0, -3, + SBR_SWITCH_LRC, 2}, + {CODEC_AAC, 52000, 60000, 12000, 2, 14, 14, 13, 13, 3, 0, -3, + SBR_SWITCH_LRC, 1}, + {CODEC_AAC, 60000, 76000, 12000, 2, 14, 14, 13, 13, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + {CODEC_AAC, 76000, 128001, 12000, 2, 14, 14, 13, 13, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + + /* 16/32 kHz dual rate */ + {CODEC_AAC, 16000, 24000, 16000, 2, 4, 2, 1, 0, 1, 0, -3, SBR_SWITCH_LRC, + 3}, + {CODEC_AAC, 24000, 28000, 16000, 2, 8, 7, 10, 8, 1, 0, -3, SBR_SWITCH_LRC, + 3}, + {CODEC_AAC, 28000, 36000, 16000, 2, 10, 9, 12, 11, 2, 0, -3, SBR_SWITCH_LRC, + 2}, + {CODEC_AAC, 36000, 44000, 16000, 2, 13, 13, 13, 13, 2, 0, -3, + SBR_SWITCH_LRC, 2}, + {CODEC_AAC, 44000, 52000, 16000, 2, 14, 14, 13, 13, 2, 0, -3, + SBR_SWITCH_LRC, 2}, + {CODEC_AAC, 52000, 60000, 16000, 2, 14, 14, 13, 13, 3, 0, -3, + SBR_SWITCH_LRC, 1}, + {CODEC_AAC, 60000, 76000, 16000, 2, 14, 14, 13, 13, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + {CODEC_AAC, 76000, 128001, 16000, 2, 14, 14, 13, 13, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + + /* 22.05/44.1 kHz dual rate */ + {CODEC_AAC, 16000, 24000, 22050, 2, 2, 1, 1, 0, 1, 0, -3, SBR_SWITCH_LRC, + 3}, + {CODEC_AAC, 24000, 28000, 22050, 2, 5, 4, 6, 5, 1, 0, -3, SBR_SWITCH_LRC, + 3}, + {CODEC_AAC, 28000, 32000, 22050, 2, 5, 4, 8, 7, 2, 0, -3, SBR_SWITCH_LRC, + 2}, + {CODEC_AAC, 32000, 36000, 22050, 2, 7, 6, 8, 7, 2, 0, -3, SBR_SWITCH_LRC, + 2}, + {CODEC_AAC, 36000, 44000, 22050, 2, 10, 10, 9, 9, 2, 0, -3, SBR_SWITCH_LRC, + 2}, + {CODEC_AAC, 44000, 52000, 22050, 2, 12, 12, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, + 2}, + {CODEC_AAC, 52000, 60000, 22050, 2, 13, 13, 10, 10, 3, 0, -3, + SBR_SWITCH_LRC, 1}, + {CODEC_AAC, 60000, 76000, 22050, 2, 14, 14, 12, 12, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + {CODEC_AAC, 76000, 128001, 22050, 2, 14, 14, 12, 12, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + + /* 24/48 kHz dual rate */ + {CODEC_AAC, 16000, 24000, 24000, 2, 2, 1, 1, 0, 1, 0, -3, SBR_SWITCH_LRC, + 3}, + {CODEC_AAC, 24000, 28000, 24000, 2, 5, 5, 6, 6, 1, 0, -3, SBR_SWITCH_LRC, + 3}, + {CODEC_AAC, 28000, 36000, 24000, 2, 7, 6, 8, 7, 2, 0, -3, SBR_SWITCH_LRC, + 2}, + {CODEC_AAC, 36000, 44000, 24000, 2, 10, 10, 9, 9, 2, 0, -3, SBR_SWITCH_LRC, + 2}, + {CODEC_AAC, 44000, 52000, 24000, 2, 12, 12, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, + 2}, + {CODEC_AAC, 52000, 60000, 24000, 2, 13, 13, 10, 10, 3, 0, -3, + SBR_SWITCH_LRC, 1}, + {CODEC_AAC, 60000, 76000, 24000, 2, 14, 14, 12, 12, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + {CODEC_AAC, 76000, 128001, 24000, 2, 14, 14, 12, 12, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + + /* 32/64 kHz dual rate */ + {CODEC_AAC, 32000, 60000, 32000, 2, 4, 4, 4, 4, 2, 0, -3, SBR_SWITCH_LRC, + 3}, + {CODEC_AAC, 60000, 80000, 32000, 2, 7, 7, 6, 6, 3, 0, -3, SBR_SWITCH_LRC, + 2}, + {CODEC_AAC, 80000, 112000, 32000, 2, 9, 9, 8, 8, 3, 0, -3, SBR_LEFT_RIGHT, + 1}, + {CODEC_AAC, 112000, 144000, 32000, 2, 11, 11, 10, 10, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + {CODEC_AAC, 144000, 256001, 32000, 2, 13, 13, 11, 11, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + + /* 44.1/88.2 kHz dual rate */ + {CODEC_AAC, 32000, 60000, 44100, 2, 4, 4, 4, 4, 2, 0, -3, SBR_SWITCH_LRC, + 3}, + {CODEC_AAC, 60000, 80000, 44100, 2, 7, 7, 6, 6, 3, 0, -3, SBR_SWITCH_LRC, + 2}, + {CODEC_AAC, 80000, 112000, 44100, 2, 9, 9, 8, 8, 3, 0, -3, SBR_LEFT_RIGHT, + 1}, + {CODEC_AAC, 112000, 144000, 44100, 2, 11, 11, 10, 10, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + {CODEC_AAC, 144000, 256001, 44100, 2, 13, 13, 11, 11, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + + /* 48/96 kHz dual rate */ + {CODEC_AAC, 36000, 60000, 48000, 2, 4, 4, 9, 9, 2, 0, -3, SBR_SWITCH_LRC, + 3}, + {CODEC_AAC, 60000, 80000, 48000, 2, 7, 7, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, + 2}, + {CODEC_AAC, 80000, 112000, 48000, 2, 9, 9, 10, 10, 3, 0, -3, SBR_LEFT_RIGHT, + 1}, + {CODEC_AAC, 112000, 144000, 48000, 2, 11, 11, 11, 11, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + {CODEC_AAC, 144000, 256001, 48000, 2, 13, 13, 11, 11, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + + /** AAC LOW DELAY SECTION **/ + + /* 24 kHz dual rate - 12kHz singlerate is not allowed (deactivated in + FDKsbrEnc_IsSbrSettingAvail()) */ + {CODEC_AACLD, 8000, 32000, 12000, 1, 1, 1, 0, 0, 1, 0, 6, SBR_MONO, 3}, + + /*** mono ***/ + /* 16/32 kHz dual rate */ + {CODEC_AACLD, 16000, 18000, 16000, 1, 4, 5, 9, 7, 1, 0, 6, SBR_MONO, 3}, + {CODEC_AACLD, 18000, 22000, 16000, 1, 7, 7, 12, 12, 1, 6, 9, SBR_MONO, 3}, + {CODEC_AACLD, 22000, 28000, 16000, 1, 6, 6, 9, 9, 2, 3, 6, SBR_MONO, 3}, + {CODEC_AACLD, 28000, 36000, 16000, 1, 8, 8, 12, 7, 2, 9, 12, SBR_MONO, 3}, + {CODEC_AACLD, 36000, 44000, 16000, 1, 10, 14, 12, 13, 2, 0, 3, SBR_MONO, 1}, + {CODEC_AACLD, 44000, 64001, 16000, 1, 11, 14, 13, 13, 2, 0, 3, SBR_MONO, 1}, + + /* 22.05/44.1 kHz dual rate */ + {CODEC_AACLD, 18000, 22000, 22050, 1, 4, 4, 5, 5, 2, 0, 6, SBR_MONO, 3}, + {CODEC_AACLD, 22000, 28000, 22050, 1, 5, 5, 6, 6, 2, 0, 6, SBR_MONO, 2}, + {CODEC_AACLD, 28000, 36000, 22050, 1, 7, 8, 8, 8, 2, 0, 3, SBR_MONO, 2}, + {CODEC_AACLD, 36000, 44000, 22050, 1, 9, 9, 9, 9, 2, 0, 3, SBR_MONO, 1}, + {CODEC_AACLD, 44000, 52000, 22050, 1, 12, 11, 11, 11, 2, 0, 3, SBR_MONO, 1}, + {CODEC_AACLD, 52000, 64001, 22050, 1, 13, 11, 11, 10, 2, 0, 3, SBR_MONO, 1}, + + /* 24/48 kHz dual rate */ + {CODEC_AACLD, 20000, 22000, 24000, 1, 3, 4, 8, 8, 2, 0, 6, SBR_MONO, 2}, + {CODEC_AACLD, 22000, 28000, 24000, 1, 3, 8, 8, 7, 2, 0, 3, SBR_MONO, 2}, + {CODEC_AACLD, 28000, 36000, 24000, 1, 4, 8, 8, 7, 2, 0, 3, SBR_MONO, 2}, + {CODEC_AACLD, 36000, 56000, 24000, 1, 8, 9, 9, 8, 2, 0, 3, SBR_MONO, 1}, + {CODEC_AACLD, 56000, 64001, 24000, 1, 13, 11, 11, 10, 2, 0, 3, SBR_MONO, 1}, + + /* 32/64 kHz dual rate */ + {CODEC_AACLD, 24000, 36000, 32000, 1, 4, 4, 4, 4, 2, 0, 3, SBR_MONO, 3}, + {CODEC_AACLD, 36000, 60000, 32000, 1, 7, 7, 6, 6, 2, 0, 3, SBR_MONO, 2}, + {CODEC_AACLD, 60000, 72000, 32000, 1, 9, 9, 8, 8, 2, 0, 3, SBR_MONO, 1}, + {CODEC_AACLD, 72000, 100000, 32000, 1, 11, 11, 10, 10, 2, 0, 3, SBR_MONO, + 1}, + {CODEC_AACLD, 100000, 160001, 32000, 1, 13, 13, 11, 11, 2, 0, 3, SBR_MONO, + 1}, + + /* 44/88 kHz dual rate */ + {CODEC_AACLD, 36000, 60000, 44100, 1, 8, 7, 6, 9, 2, 0, 3, SBR_MONO, 2}, + {CODEC_AACLD, 60000, 72000, 44100, 1, 9, 9, 10, 10, 2, 0, 3, SBR_MONO, 1}, + {CODEC_AACLD, 72000, 100000, 44100, 1, 11, 11, 11, 11, 2, 0, 3, SBR_MONO, + 1}, + {CODEC_AACLD, 100000, 160001, 44100, 1, 13, 13, 11, 11, 2, 0, 3, SBR_MONO, + 1}, + + /* 48/96 kHz dual rate */ /* 32 and 40kbps line tuned for dual-rate SBR + */ + {CODEC_AACLD, 36000, 60000, 48000, 1, 4, 7, 4, 4, 2, 0, 3, SBR_MONO, 3}, + {CODEC_AACLD, 60000, 72000, 48000, 1, 9, 9, 10, 10, 2, 0, 3, SBR_MONO, 1}, + {CODEC_AACLD, 72000, 100000, 48000, 1, 11, 11, 11, 11, 2, 0, 3, SBR_MONO, + 1}, + {CODEC_AACLD, 100000, 160001, 48000, 1, 13, 13, 11, 11, 2, 0, 3, SBR_MONO, + 1}, + + /*** stereo ***/ + /* 16/32 kHz dual rate */ + {CODEC_AACLD, 32000, 36000, 16000, 2, 10, 9, 12, 11, 2, 0, -3, + SBR_SWITCH_LRC, 2}, + {CODEC_AACLD, 36000, 44000, 16000, 2, 13, 13, 13, 13, 2, 0, -3, + SBR_SWITCH_LRC, 2}, + {CODEC_AACLD, 44000, 52000, 16000, 2, 10, 9, 11, 9, 2, 0, -3, + SBR_SWITCH_LRC, 2}, + {CODEC_AACLD, 52000, 60000, 16000, 2, 14, 14, 13, 13, 3, 0, -3, + SBR_SWITCH_LRC, 1}, + {CODEC_AACLD, 60000, 76000, 16000, 2, 14, 14, 13, 13, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + {CODEC_AACLD, 76000, 128001, 16000, 2, 14, 14, 13, 13, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + + /* 22.05/44.1 kHz dual rate */ + {CODEC_AACLD, 32000, 36000, 22050, 2, 5, 4, 7, 6, 2, 0, -3, SBR_SWITCH_LRC, + 2}, + {CODEC_AACLD, 36000, 44000, 22050, 2, 5, 8, 8, 8, 2, 0, -3, SBR_SWITCH_LRC, + 2}, + {CODEC_AACLD, 44000, 52000, 22050, 2, 7, 10, 8, 8, 3, 0, -3, SBR_SWITCH_LRC, + 2}, + {CODEC_AACLD, 52000, 60000, 22050, 2, 9, 11, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, + 1}, + {CODEC_AACLD, 60000, 76000, 22050, 2, 10, 12, 10, 11, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + {CODEC_AACLD, 76000, 82000, 22050, 2, 12, 12, 11, 11, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + {CODEC_AACLD, 82000, 128001, 22050, 2, 13, 12, 11, 11, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + + /* 24/48 kHz dual rate */ + {CODEC_AACLD, 32000, 36000, 24000, 2, 5, 4, 7, 6, 2, 0, -3, SBR_SWITCH_LRC, + 2}, + {CODEC_AACLD, 36000, 44000, 24000, 2, 4, 8, 8, 8, 2, 0, -3, SBR_SWITCH_LRC, + 2}, + {CODEC_AACLD, 44000, 52000, 24000, 2, 6, 10, 8, 8, 3, 0, -3, SBR_SWITCH_LRC, + 2}, + {CODEC_AACLD, 52000, 60000, 24000, 2, 9, 11, 9, 9, 3, 0, -3, SBR_SWITCH_LRC, + 1}, + {CODEC_AACLD, 60000, 76000, 24000, 2, 11, 12, 10, 11, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + {CODEC_AACLD, 76000, 88000, 24000, 2, 12, 13, 11, 11, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + {CODEC_AACLD, 88000, 128001, 24000, 2, 13, 13, 11, 11, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + + /* 32/64 kHz dual rate */ + {CODEC_AACLD, 60000, 80000, 32000, 2, 7, 7, 6, 6, 3, 0, -3, SBR_SWITCH_LRC, + 2}, + {CODEC_AACLD, 80000, 112000, 32000, 2, 9, 9, 8, 8, 3, 0, -3, SBR_LEFT_RIGHT, + 1}, + {CODEC_AACLD, 112000, 144000, 32000, 2, 11, 11, 10, 10, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + {CODEC_AACLD, 144000, 256001, 32000, 2, 13, 13, 11, 11, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + + /* 44.1/88.2 kHz dual rate */ + {CODEC_AACLD, 60000, 80000, 44100, 2, 7, 7, 6, 6, 3, 0, -3, SBR_SWITCH_LRC, + 2}, + {CODEC_AACLD, 80000, 112000, 44100, 2, 10, 10, 8, 8, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + {CODEC_AACLD, 112000, 144000, 44100, 2, 12, 12, 10, 10, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + {CODEC_AACLD, 144000, 256001, 44100, 2, 13, 13, 11, 11, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + + /* 48/96 kHz dual rate */ + {CODEC_AACLD, 60000, 80000, 48000, 2, 7, 7, 10, 10, 2, 0, -3, + SBR_SWITCH_LRC, 2}, + {CODEC_AACLD, 80000, 112000, 48000, 2, 9, 9, 10, 10, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + {CODEC_AACLD, 112000, 144000, 48000, 2, 11, 11, 11, 11, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + {CODEC_AACLD, 144000, 176000, 48000, 2, 12, 12, 11, 11, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + {CODEC_AACLD, 176000, 256001, 48000, 2, 13, 13, 11, 11, 3, 0, -3, + SBR_LEFT_RIGHT, 1}, + +}; + +const int sbrTuningTableSize = + sizeof(sbrTuningTable) / sizeof(sbrTuningTable[0]); + +const psTuningTable_t psTuningTable[4] = { + {8000, 22000, PSENC_STEREO_BANDS_10, PSENC_NENV_1, + FL2FXCONST_DBL(3.0f / 4.0f)}, + {22000, 28000, PSENC_STEREO_BANDS_20, PSENC_NENV_1, + FL2FXCONST_DBL(2.0f / 4.0f)}, + {28000, 36000, PSENC_STEREO_BANDS_20, PSENC_NENV_2, + FL2FXCONST_DBL(1.5f / 4.0f)}, + {36000, 160001, PSENC_STEREO_BANDS_20, PSENC_NENV_4, + FL2FXCONST_DBL(1.1f / 4.0f)}, +}; + +//@} diff --git a/libSBRenc/src/sbrenc_rom.h b/libSBRenc/src/sbrenc_rom.h new file mode 100644 index 0000000..18c1fb9 --- /dev/null +++ b/libSBRenc/src/sbrenc_rom.h @@ -0,0 +1,145 @@ +/* ----------------------------------------------------------------------------- +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 encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! +\file +\brief Declaration of constant tables +$Revision: 92790 $ +*/ +#ifndef SBRENC_ROM_H +#define SBRENC_ROM_H + +#include "sbr_def.h" +#include "sbr_encoder.h" + +#include "ps_main.h" + +/* + huffman tables +*/ +extern const INT v_Huff_envelopeLevelC10T[121]; +extern const UCHAR v_Huff_envelopeLevelL10T[121]; +extern const INT v_Huff_envelopeLevelC10F[121]; +extern const UCHAR v_Huff_envelopeLevelL10F[121]; +extern const INT bookSbrEnvBalanceC10T[49]; +extern const UCHAR bookSbrEnvBalanceL10T[49]; +extern const INT bookSbrEnvBalanceC10F[49]; +extern const UCHAR bookSbrEnvBalanceL10F[49]; +extern const INT v_Huff_envelopeLevelC11T[63]; +extern const UCHAR v_Huff_envelopeLevelL11T[63]; +extern const INT v_Huff_envelopeLevelC11F[63]; +extern const UCHAR v_Huff_envelopeLevelL11F[63]; +extern const INT bookSbrEnvBalanceC11T[25]; +extern const UCHAR bookSbrEnvBalanceL11T[25]; +extern const INT bookSbrEnvBalanceC11F[25]; +extern const UCHAR bookSbrEnvBalanceL11F[25]; +extern const INT v_Huff_NoiseLevelC11T[63]; +extern const UCHAR v_Huff_NoiseLevelL11T[63]; +extern const INT bookSbrNoiseBalanceC11T[25]; +extern const UCHAR bookSbrNoiseBalanceL11T[25]; + +extern const sbrTuningTable_t sbrTuningTable[]; +extern const int sbrTuningTableSize; + +extern const psTuningTable_t psTuningTable[4]; + +#endif diff --git a/libSBRenc/src/ton_corr.cpp b/libSBRenc/src/ton_corr.cpp index af5afba..1c050e2 100644 --- a/libSBRenc/src/ton_corr.cpp +++ b/libSBRenc/src/ton_corr.cpp @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,22 +90,26 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ -#include "ton_corr.h" +/**************************** SBR encoder library ****************************** -#include "sbr_ram.h" -#include "sbr_misc.h" -#include "genericStds.h" -#include "autocorr2nd.h" + Author(s): + Description: +*******************************************************************************/ -/*************************************************************************** +#include "ton_corr.h" - Send autoCorrSecondOrder to mlfile +#include "sbrenc_ram.h" +#include "sbr_misc.h" +#include "genericStds.h" +#include "autocorr2nd.h" -****************************************************************************/ +#define BAND_V_SIZE 32 +#define NUM_V_COMBINE \ + 8 /* Must be a divisor of 64 and fulfill the ASSERTs below */ /**************************************************************************/ /*! @@ -107,7 +122,7 @@ amm-info@iis.fraunhofer.de (noise energy B). Hence the quota-matrix contains A/B = q/(1-q). The samples in nrgVector are scaled by 1.0/16.0 - The samples in pNrgVectorFreq are scaled by 1.0/2.0 + The samples in pNrgVectorFreq are scaled by 1.0/2.0 The samples in quotaMatrix are scaled by RELAXATION \return none. @@ -115,84 +130,83 @@ amm-info@iis.fraunhofer.de */ /**************************************************************************/ -void -FDKsbrEnc_CalculateTonalityQuotas( HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */ - FIXP_DBL **RESTRICT sourceBufferReal, /*!< The real part of the QMF-matrix. */ - FIXP_DBL **RESTRICT sourceBufferImag, /*!< The imaginary part of the QMF-matrix. */ - INT usb, /*!< upper side band, highest + 1 QMF band in the SBR range. */ - INT qmfScale /*!< sclefactor of QMF subsamples */ - ) -{ - INT i, k, r, r2, timeIndex, autoCorrScaling; - - INT startIndexMatrix = hTonCorr->startIndexMatrix; - INT totNoEst = hTonCorr->numberOfEstimates; - INT noEstPerFrame = hTonCorr->numberOfEstimatesPerFrame; - INT move = hTonCorr->move; - INT noQmfChannels = hTonCorr->noQmfChannels; /* Numer of Bands */ - INT buffLen = hTonCorr->bufferLength; /* Numer of Slots */ - INT stepSize = hTonCorr->stepSize; - INT *pBlockLength = hTonCorr->lpcLength; - INT** RESTRICT signMatrix = hTonCorr->signMatrix; - FIXP_DBL* RESTRICT nrgVector = hTonCorr->nrgVector; - FIXP_DBL** RESTRICT quotaMatrix = hTonCorr->quotaMatrix; - FIXP_DBL* RESTRICT pNrgVectorFreq = hTonCorr->nrgVectorFreq; - -#define BAND_V_SIZE QMF_MAX_TIME_SLOTS -#define NUM_V_COMBINE 8 /* Must be a divisor of 64 and fulfill the ASSERTs below */ +void FDKsbrEnc_CalculateTonalityQuotas( + HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */ + FIXP_DBL **RESTRICT + sourceBufferReal, /*!< The real part of the QMF-matrix. */ + FIXP_DBL **RESTRICT + sourceBufferImag, /*!< The imaginary part of the QMF-matrix. */ + INT usb, /*!< upper side band, highest + 1 QMF band in the SBR range. */ + INT qmfScale /*!< sclefactor of QMF subsamples */ +) { + INT i, k, r, r2, timeIndex, autoCorrScaling; + + INT startIndexMatrix = hTonCorr->startIndexMatrix; + INT totNoEst = hTonCorr->numberOfEstimates; + INT noEstPerFrame = hTonCorr->numberOfEstimatesPerFrame; + INT move = hTonCorr->move; + INT noQmfChannels = hTonCorr->noQmfChannels; /* Number of Bands */ + INT buffLen = hTonCorr->bufferLength; /* Number of Slots */ + INT stepSize = hTonCorr->stepSize; + INT *pBlockLength = hTonCorr->lpcLength; + INT **RESTRICT signMatrix = hTonCorr->signMatrix; + FIXP_DBL *RESTRICT nrgVector = hTonCorr->nrgVector; + FIXP_DBL **RESTRICT quotaMatrix = hTonCorr->quotaMatrix; + FIXP_DBL *RESTRICT pNrgVectorFreq = hTonCorr->nrgVectorFreq; FIXP_DBL *realBuf; FIXP_DBL *imagBuf; - FIXP_DBL alphar[2],alphai[2],fac; - - C_ALLOC_SCRATCH_START(ac, ACORR_COEFS, 1); - C_ALLOC_SCRATCH_START(realBufRef, FIXP_DBL, 2*BAND_V_SIZE*NUM_V_COMBINE); + FIXP_DBL alphar[2], alphai[2], fac; + C_ALLOC_SCRATCH_START(ac, ACORR_COEFS, 1) + C_ALLOC_SCRATCH_START(realBufRef, FIXP_DBL, 2 * BAND_V_SIZE * NUM_V_COMBINE) realBuf = realBufRef; - imagBuf = realBuf + BAND_V_SIZE*NUM_V_COMBINE; - + imagBuf = realBuf + BAND_V_SIZE * NUM_V_COMBINE; FDK_ASSERT(buffLen <= BAND_V_SIZE); - FDK_ASSERT(sizeof(FIXP_DBL)*NUM_V_COMBINE*BAND_V_SIZE*2 < (1024*sizeof(FIXP_DBL)-sizeof(ACORR_COEFS)) ); + FDK_ASSERT(sizeof(FIXP_DBL) * NUM_V_COMBINE * BAND_V_SIZE * 2 < + (1024 * sizeof(FIXP_DBL) - sizeof(ACORR_COEFS))); /* * Buffering of the quotaMatrix and the quotaMatrixTransp. *********************************************************/ - for(i = 0 ; i < move; i++){ - FDKmemcpy(quotaMatrix[i],quotaMatrix[i + noEstPerFrame],noQmfChannels * sizeof(FIXP_DBL)); - FDKmemcpy(signMatrix[i],signMatrix[i + noEstPerFrame],noQmfChannels * sizeof(INT)); + for (i = 0; i < move; i++) { + FDKmemcpy(quotaMatrix[i], quotaMatrix[i + noEstPerFrame], + noQmfChannels * sizeof(FIXP_DBL)); + FDKmemcpy(signMatrix[i], signMatrix[i + noEstPerFrame], + noQmfChannels * sizeof(INT)); } - FDKmemmove(nrgVector,nrgVector+noEstPerFrame,move*sizeof(FIXP_DBL)); - FDKmemclear(nrgVector+startIndexMatrix,(totNoEst-startIndexMatrix)*sizeof(FIXP_DBL)); - FDKmemclear(pNrgVectorFreq,noQmfChannels * sizeof(FIXP_DBL)); + FDKmemmove(nrgVector, nrgVector + noEstPerFrame, move * sizeof(FIXP_DBL)); + FDKmemclear(nrgVector + startIndexMatrix, + (totNoEst - startIndexMatrix) * sizeof(FIXP_DBL)); + FDKmemclear(pNrgVectorFreq, noQmfChannels * sizeof(FIXP_DBL)); /* * Calculate the quotas for the current time steps. **************************************************/ - for (r = 0; r < usb; r++) - { + for (r = 0; r < usb; r++) { int blockLength; k = hTonCorr->nextSample; /* startSample */ timeIndex = startIndexMatrix; - /* Copy as many as possible Band accross all Slots at once */ + /* Copy as many as possible Band across all Slots at once */ if (realBuf != realBufRef) { realBuf -= BAND_V_SIZE; imagBuf -= BAND_V_SIZE; } else { - realBuf += BAND_V_SIZE*(NUM_V_COMBINE-1); - imagBuf += BAND_V_SIZE*(NUM_V_COMBINE-1); + realBuf += BAND_V_SIZE * (NUM_V_COMBINE - 1); + imagBuf += BAND_V_SIZE * (NUM_V_COMBINE - 1); + for (i = 0; i < buffLen; i++) { int v; FIXP_DBL *ptr; - ptr = realBuf+i; - for (v=0; v<NUM_V_COMBINE; v++) - { - ptr[0] = sourceBufferReal[i][r+v]; - ptr[0+BAND_V_SIZE*NUM_V_COMBINE] = sourceBufferImag[i][r+v]; + ptr = realBuf + i; + for (v = 0; v < NUM_V_COMBINE; v++) { + ptr[0] = sourceBufferReal[i][r + v]; + ptr[0 + BAND_V_SIZE * NUM_V_COMBINE] = sourceBufferImag[i][r + v]; ptr -= BAND_V_SIZE; } } @@ -200,54 +214,66 @@ FDKsbrEnc_CalculateTonalityQuotas( HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< H blockLength = pBlockLength[0]; - while(k <= buffLen - blockLength) - { - autoCorrScaling = fixMin(getScalefactor(&realBuf[k-LPC_ORDER], LPC_ORDER+blockLength), getScalefactor(&imagBuf[k-LPC_ORDER], LPC_ORDER+blockLength)); - autoCorrScaling = fixMax(0, autoCorrScaling-1); + while (k <= buffLen - blockLength) { + autoCorrScaling = fixMin( + getScalefactor(&realBuf[k - LPC_ORDER], LPC_ORDER + blockLength), + getScalefactor(&imagBuf[k - LPC_ORDER], LPC_ORDER + blockLength)); + autoCorrScaling = fixMax(0, autoCorrScaling - 1); - scaleValues(&realBuf[k-LPC_ORDER], LPC_ORDER+blockLength, autoCorrScaling); - scaleValues(&imagBuf[k-LPC_ORDER], LPC_ORDER+blockLength, autoCorrScaling); + scaleValues(&realBuf[k - LPC_ORDER], LPC_ORDER + blockLength, + autoCorrScaling); + scaleValues(&imagBuf[k - LPC_ORDER], LPC_ORDER + blockLength, + autoCorrScaling); autoCorrScaling <<= 1; /* consider qmf buffer scaling twice */ - autoCorrScaling += autoCorr2nd_cplx ( ac, realBuf+k, imagBuf+k, blockLength ); + autoCorrScaling += + autoCorr2nd_cplx(ac, realBuf + k, imagBuf + k, blockLength); - - if(ac->det == FL2FXCONST_DBL(0.0f)){ + if (ac->det == FL2FXCONST_DBL(0.0f)) { alphar[1] = alphai[1] = FL2FXCONST_DBL(0.0f); - alphar[0] = (ac->r01r)>>2; - alphai[0] = (ac->r01i)>>2; - - fac = fMultDiv2(ac->r00r, ac->r11r)>>1; + alphar[0] = (ac->r01r) >> 2; + alphai[0] = (ac->r01i) >> 2; + + fac = fMultDiv2(ac->r00r, ac->r11r) >> 1; + } else { + alphar[1] = (fMultDiv2(ac->r01r, ac->r12r) >> 1) - + (fMultDiv2(ac->r01i, ac->r12i) >> 1) - + (fMultDiv2(ac->r02r, ac->r11r) >> 1); + alphai[1] = (fMultDiv2(ac->r01i, ac->r12r) >> 1) + + (fMultDiv2(ac->r01r, ac->r12i) >> 1) - + (fMultDiv2(ac->r02i, ac->r11r) >> 1); + + alphar[0] = (fMultDiv2(ac->r01r, ac->det) >> (ac->det_scale + 1)) + + fMult(alphar[1], ac->r12r) + fMult(alphai[1], ac->r12i); + alphai[0] = (fMultDiv2(ac->r01i, ac->det) >> (ac->det_scale + 1)) + + fMult(alphai[1], ac->r12r) - fMult(alphar[1], ac->r12i); + + fac = fMultDiv2(ac->r00r, fMult(ac->det, ac->r11r)) >> + (ac->det_scale + 1); } - else{ - alphar[1] = (fMultDiv2(ac->r01r, ac->r12r)>>1) - (fMultDiv2(ac->r01i, ac->r12i)>>1) - (fMultDiv2(ac->r02r, ac->r11r)>>1); - alphai[1] = (fMultDiv2(ac->r01i, ac->r12r)>>1) + (fMultDiv2(ac->r01r, ac->r12i)>>1) - (fMultDiv2(ac->r02i, ac->r11r)>>1); - alphar[0] = (fMultDiv2(ac->r01r, ac->det)>>(ac->det_scale+1)) + fMult(alphar[1], ac->r12r) + fMult(alphai[1], ac->r12i); - alphai[0] = (fMultDiv2(ac->r01i, ac->det)>>(ac->det_scale+1)) + fMult(alphai[1], ac->r12r) - fMult(alphar[1], ac->r12i); - - fac = fMultDiv2(ac->r00r, fMult(ac->det, ac->r11r))>>(ac->det_scale+1); - } - - if(fac == FL2FXCONST_DBL(0.0f)){ + if (fac == FL2FXCONST_DBL(0.0f)) { quotaMatrix[timeIndex][r] = FL2FXCONST_DBL(0.0f); signMatrix[timeIndex][r] = 0; - } - else { + } else { /* quotaMatrix is scaled with the factor RELAXATION - parse RELAXATION in fractional part and shift factor: 1/(1/0.524288 * 2^RELAXATION_SHIFT) */ - FIXP_DBL tmp,num,denom; - INT numShift,denomShift,commonShift; + parse RELAXATION in fractional part and shift factor: 1/(1/0.524288 * + 2^RELAXATION_SHIFT) */ + FIXP_DBL tmp, num, denom; + INT numShift, denomShift, commonShift; INT sign; - num = fMultDiv2(alphar[0], ac->r01r) + fMultDiv2(alphai[0], ac->r01i) - fMultDiv2(alphar[1], fMult(ac->r02r, ac->r11r)) - fMultDiv2(alphai[1], fMult(ac->r02i, ac->r11r)); + num = fMultDiv2(alphar[0], ac->r01r) + fMultDiv2(alphai[0], ac->r01i) - + fMultDiv2(alphar[1], fMult(ac->r02r, ac->r11r)) - + fMultDiv2(alphai[1], fMult(ac->r02i, ac->r11r)); num = fixp_abs(num); - denom = (fac>>1) + (fMultDiv2(fac,RELAXATION_FRACT)>>RELAXATION_SHIFT) - num; + denom = (fac >> 1) + + (fMultDiv2(fac, RELAXATION_FRACT) >> RELAXATION_SHIFT) - num; denom = fixp_abs(denom); - num = fMult(num,RELAXATION_FRACT); + num = fMult(num, RELAXATION_FRACT); numShift = CountLeadingBits(num) - 2; num = scaleValue(num, numShift); @@ -256,46 +282,53 @@ FDKsbrEnc_CalculateTonalityQuotas( HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< H denom = (FIXP_DBL)denom << denomShift; if ((num > FL2FXCONST_DBL(0.0f)) && (denom != FL2FXCONST_DBL(0.0f))) { - commonShift = fixMin(numShift - denomShift + RELAXATION_SHIFT, DFRACT_BITS-1); + commonShift = + fixMin(numShift - denomShift + RELAXATION_SHIFT, DFRACT_BITS - 1); if (commonShift < 0) { commonShift = -commonShift; - tmp = schur_div(num,denom,16); - commonShift = fixMin(commonShift,CountLeadingBits(tmp)); + tmp = schur_div(num, denom, 16); + commonShift = fixMin(commonShift, CountLeadingBits(tmp)); quotaMatrix[timeIndex][r] = tmp << commonShift; + } else { + quotaMatrix[timeIndex][r] = + schur_div(num, denom, 16) >> commonShift; } - else { - quotaMatrix[timeIndex][r] = schur_div(num,denom,16) >> commonShift; - } - } - else { + } else { quotaMatrix[timeIndex][r] = FL2FXCONST_DBL(0.0f); } if (ac->r11r != FL2FXCONST_DBL(0.0f)) { - if ( ( (ac->r01r >= FL2FXCONST_DBL(0.0f) ) && ( ac->r11r >= FL2FXCONST_DBL(0.0f) ) ) - ||( (ac->r01r < FL2FXCONST_DBL(0.0f) ) && ( ac->r11r < FL2FXCONST_DBL(0.0f) ) ) ) { + if (((ac->r01r >= FL2FXCONST_DBL(0.0f)) && + (ac->r11r >= FL2FXCONST_DBL(0.0f))) || + ((ac->r01r < FL2FXCONST_DBL(0.0f)) && + (ac->r11r < FL2FXCONST_DBL(0.0f)))) { sign = 1; - } - else { + } else { sign = -1; } - } - else { + } else { sign = 1; } - if(sign < 0) { - r2 = r; /* (INT) pow(-1, band); */ - } - else { - r2 = r + 1; /* (INT) pow(-1, band+1); */ + if (sign < 0) { + r2 = r; /* (INT) pow(-1, band); */ + } else { + r2 = r + 1; /* (INT) pow(-1, band+1); */ } - signMatrix[timeIndex][r] = 1 - 2*(r2 & 0x1); + signMatrix[timeIndex][r] = 1 - 2 * (r2 & 0x1); } - nrgVector[timeIndex] += ((ac->r00r) >> fixMin(DFRACT_BITS-1,(2*qmfScale+autoCorrScaling + SCALE_NRGVEC))); - /* pNrgVectorFreq[r] finally has to be divided by noEstPerFrame, replaced division by shifting with one */ - pNrgVectorFreq[r] = pNrgVectorFreq[r] + ((ac->r00r) >> fixMin(DFRACT_BITS-1,(2*qmfScale+autoCorrScaling + SCALE_NRGVEC))); + nrgVector[timeIndex] += + ((ac->r00r) >> + fixMin(DFRACT_BITS - 1, + (2 * qmfScale + autoCorrScaling + SCALE_NRGVEC))); + /* pNrgVectorFreq[r] finally has to be divided by noEstPerFrame, replaced + * division by shifting with one */ + pNrgVectorFreq[r] = + pNrgVectorFreq[r] + + ((ac->r00r) >> + fixMin(DFRACT_BITS - 1, + (2 * qmfScale + autoCorrScaling + SCALE_NRGVEC))); blockLength = pBlockLength[1]; k += stepSize; @@ -303,9 +336,8 @@ FDKsbrEnc_CalculateTonalityQuotas( HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< H } } - - C_ALLOC_SCRATCH_END(realBuf, FIXP_DBL, 2*BAND_V_SIZE*NUM_V_COMBINE); - C_ALLOC_SCRATCH_END(ac, ACORR_COEFS, 1); + C_ALLOC_SCRATCH_END(realBufRef, FIXP_DBL, 2 * BAND_V_SIZE * NUM_V_COMBINE) + C_ALLOC_SCRATCH_END(ac, ACORR_COEFS, 1) } /**************************************************************************/ @@ -324,117 +356,101 @@ FDKsbrEnc_CalculateTonalityQuotas( HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< H */ /**************************************************************************/ -void -FDKsbrEnc_TonCorrParamExtr(HANDLE_SBR_TON_CORR_EST hTonCorr,/*!< Handle to SBR_TON_CORR struct. */ - INVF_MODE* infVec, /*!< Vector where the inverse filtering levels will be stored. */ - FIXP_DBL * noiseLevels, /*!< Vector where the noise levels will be stored. */ - INT* missingHarmonicFlag, /*!< Flag set to one or zero, dependent on if any strong sines are missing.*/ - UCHAR * missingHarmonicsIndex, /*!< Vector indicating where sines are missing. */ - UCHAR * envelopeCompensation, /*!< Vector to store compensation values for the energies in. */ - const SBR_FRAME_INFO *frameInfo, /*!< Frame info struct, contains the time and frequency grid of the current frame.*/ - UCHAR* transientInfo, /*!< Transient info.*/ - UCHAR* freqBandTable, /*!< Frequency band tables for high-res.*/ - INT nSfb, /*!< Number of scalefactor bands for high-res. */ - XPOS_MODE xposType, /*!< Type of transposer used in the decoder.*/ - UINT sbrSyntaxFlags - ) -{ +void FDKsbrEnc_TonCorrParamExtr( + HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */ + INVF_MODE *infVec, /*!< Vector where the inverse filtering levels will be + stored. */ + FIXP_DBL *noiseLevels, /*!< Vector where the noise levels will be stored. */ + INT *missingHarmonicFlag, /*!< Flag set to one or zero, dependent on if any + strong sines are missing.*/ + UCHAR *missingHarmonicsIndex, /*!< Vector indicating where sines are + missing. */ + UCHAR *envelopeCompensation, /*!< Vector to store compensation values for + the energies in. */ + const SBR_FRAME_INFO *frameInfo, /*!< Frame info struct, contains the time + and frequency grid of the current + frame.*/ + UCHAR *transientInfo, /*!< Transient info.*/ + UCHAR *freqBandTable, /*!< Frequency band tables for high-res.*/ + INT nSfb, /*!< Number of scalefactor bands for high-res. */ + XPOS_MODE xposType, /*!< Type of transposer used in the decoder.*/ + UINT sbrSyntaxFlags) { INT band; - INT transientFlag = transientInfo[1] ; /*!< Flag indicating if a transient is present in the current frame. */ - INT transientPos = transientInfo[0]; /*!< Position of the transient.*/ + INT transientFlag = transientInfo[1]; /*!< Flag indicating if a transient is + present in the current frame. */ + INT transientPos = transientInfo[0]; /*!< Position of the transient.*/ INT transientFrame, transientFrameInvfEst; - INVF_MODE* infVecPtr; - + INVF_MODE *infVecPtr; /* Determine if this is a frame where a transient starts... - The detection of noise-floor, missing harmonics and invf_est, is not in sync for the - non-buf-opt decoder such as AAC. Hence we need to keep track on the transient in the - present frame as well as in the next. + The detection of noise-floor, missing harmonics and invf_est, is not in sync + for the non-buf-opt decoder such as AAC. Hence we need to keep track on the + transient in the present frame as well as in the next. */ transientFrame = 0; - if(hTonCorr->transientNextFrame){ /* The transient was detected in the previous frame, but is actually */ + if (hTonCorr->transientNextFrame) { /* The transient was detected in the + previous frame, but is actually */ transientFrame = 1; hTonCorr->transientNextFrame = 0; - if(transientFlag){ - if(transientPos + hTonCorr->transientPosOffset >= frameInfo->borders[frameInfo->nEnvelopes]){ + if (transientFlag) { + if (transientPos + hTonCorr->transientPosOffset >= + frameInfo->borders[frameInfo->nEnvelopes]) { hTonCorr->transientNextFrame = 1; } } - } - else{ - if(transientFlag){ - if(transientPos + hTonCorr->transientPosOffset < frameInfo->borders[frameInfo->nEnvelopes]){ + } else { + if (transientFlag) { + if (transientPos + hTonCorr->transientPosOffset < + frameInfo->borders[frameInfo->nEnvelopes]) { transientFrame = 1; hTonCorr->transientNextFrame = 0; - } - else{ + } else { hTonCorr->transientNextFrame = 1; } } } transientFrameInvfEst = transientFrame; - /* Estimate the required invese filtereing level. */ if (hTonCorr->switchInverseFilt) - FDKsbrEnc_qmfInverseFilteringDetector(&hTonCorr->sbrInvFilt, - hTonCorr->quotaMatrix, - hTonCorr->nrgVector, - hTonCorr->indexVector, - hTonCorr->frameStartIndexInvfEst, - hTonCorr->numberOfEstimatesPerFrame + hTonCorr->frameStartIndexInvfEst, - transientFrameInvfEst, - infVec); + FDKsbrEnc_qmfInverseFilteringDetector( + &hTonCorr->sbrInvFilt, hTonCorr->quotaMatrix, hTonCorr->nrgVector, + hTonCorr->indexVector, hTonCorr->frameStartIndexInvfEst, + hTonCorr->numberOfEstimatesPerFrame + hTonCorr->frameStartIndexInvfEst, + transientFrameInvfEst, infVec); /* Detect what tones will be missing. */ - if (xposType == XPOS_LC ){ - FDKsbrEnc_SbrMissingHarmonicsDetectorQmf(&hTonCorr->sbrMissingHarmonicsDetector, - hTonCorr->quotaMatrix, - hTonCorr->signMatrix, - hTonCorr->indexVector, - frameInfo, - transientInfo, - missingHarmonicFlag, - missingHarmonicsIndex, - freqBandTable, - nSfb, - envelopeCompensation, - hTonCorr->nrgVectorFreq); - } - else{ + if (xposType == XPOS_LC) { + FDKsbrEnc_SbrMissingHarmonicsDetectorQmf( + &hTonCorr->sbrMissingHarmonicsDetector, hTonCorr->quotaMatrix, + hTonCorr->signMatrix, hTonCorr->indexVector, frameInfo, transientInfo, + missingHarmonicFlag, missingHarmonicsIndex, freqBandTable, nSfb, + envelopeCompensation, hTonCorr->nrgVectorFreq); + } else { *missingHarmonicFlag = 0; - FDKmemclear(missingHarmonicsIndex,nSfb*sizeof(UCHAR)); + FDKmemclear(missingHarmonicsIndex, nSfb * sizeof(UCHAR)); } - - /* Noise floor estimation */ infVecPtr = hTonCorr->sbrInvFilt.prevInvfMode; - FDKsbrEnc_sbrNoiseFloorEstimateQmf(&hTonCorr->sbrNoiseFloorEstimate, - frameInfo, - noiseLevels, - hTonCorr->quotaMatrix, - hTonCorr->indexVector, - *missingHarmonicFlag, - hTonCorr->frameStartIndex, - hTonCorr->numberOfEstimatesPerFrame, - transientFrame, - infVecPtr, - sbrSyntaxFlags); - + FDKsbrEnc_sbrNoiseFloorEstimateQmf( + &hTonCorr->sbrNoiseFloorEstimate, frameInfo, noiseLevels, + hTonCorr->quotaMatrix, hTonCorr->indexVector, *missingHarmonicFlag, + hTonCorr->frameStartIndex, hTonCorr->numberOfEstimatesPerFrame, + transientFrame, infVecPtr, sbrSyntaxFlags); /* Store the invfVec data for the next frame...*/ - for(band = 0 ; band < hTonCorr->sbrInvFilt.noDetectorBands; band++){ + for (band = 0; band < hTonCorr->sbrInvFilt.noDetectorBands; band++) { hTonCorr->sbrInvFilt.prevInvfMode[band] = infVec[band]; } } @@ -449,28 +465,22 @@ FDKsbrEnc_TonCorrParamExtr(HANDLE_SBR_TON_CORR_EST hTonCorr,/*!< Handle to SBR_T */ /**************************************************************************/ -static INT -findClosestEntry(INT goalSb, - UCHAR *v_k_master, - INT numMaster, - INT direction) -{ +static INT findClosestEntry(INT goalSb, UCHAR *v_k_master, INT numMaster, + INT direction) { INT index; - if( goalSb <= v_k_master[0] ) - return v_k_master[0]; + if (goalSb <= v_k_master[0]) return v_k_master[0]; - if( goalSb >= v_k_master[numMaster] ) - return v_k_master[numMaster]; + if (goalSb >= v_k_master[numMaster]) return v_k_master[numMaster]; - if(direction) { + if (direction) { index = 0; - while( v_k_master[index] < goalSb ) { + while (v_k_master[index] < goalSb) { index++; } } else { index = numMaster; - while( v_k_master[index] > goalSb ) { + while (v_k_master[index] > goalSb) { index--; } } @@ -478,7 +488,6 @@ findClosestEntry(INT goalSb, return v_k_master[index]; } - /**************************************************************************/ /*! \brief resets the patch @@ -489,32 +498,36 @@ findClosestEntry(INT goalSb, */ /**************************************************************************/ -static INT -resetPatch(HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */ - INT xposctrl, /*!< Different patch modes. */ - INT highBandStartSb, /*!< Start band of the SBR range. */ - UCHAR *v_k_master, /*!< Master frequency table from which all other table are derived.*/ - INT numMaster, /*!< Number of elements in the master table. */ - INT fs, /*!< Sampling frequency. */ - INT noChannels) /*!< Number of QMF-channels. */ +static INT resetPatch( + HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */ + INT xposctrl, /*!< Different patch modes. */ + INT highBandStartSb, /*!< Start band of the SBR range. */ + UCHAR *v_k_master, /*!< Master frequency table from which all other table + are derived.*/ + INT numMaster, /*!< Number of elements in the master table. */ + INT fs, /*!< Sampling frequency. */ + INT noChannels) /*!< Number of QMF-channels. */ { - INT patch,k,i; + INT patch, k, i; INT targetStopBand; - PATCH_PARAM *patchParam = hTonCorr->patchParam; + PATCH_PARAM *patchParam = hTonCorr->patchParam; INT sbGuard = hTonCorr->guard; INT sourceStartBand; INT patchDistance; INT numBandsInPatch; - INT lsb = v_k_master[0]; /* Lowest subband related to the synthesis filterbank */ - INT usb = v_k_master[numMaster]; /* Stop subband related to the synthesis filterbank */ - INT xoverOffset = highBandStartSb - v_k_master[0]; /* Calculate distance in subbands between k0 and kx */ + INT lsb = + v_k_master[0]; /* Lowest subband related to the synthesis filterbank */ + INT usb = v_k_master[numMaster]; /* Stop subband related to the synthesis + filterbank */ + INT xoverOffset = + highBandStartSb - + v_k_master[0]; /* Calculate distance in subbands between k0 and kx */ INT goalSb; - /* * Initialize the patching parameter */ @@ -524,47 +537,55 @@ resetPatch(HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct xoverOffset = 0; } - goalSb = (INT)( (2 * noChannels * 16000 + (fs>>1)) / fs ); /* 16 kHz band */ - goalSb = findClosestEntry(goalSb, v_k_master, numMaster, 1); /* Adapt region to master-table */ + goalSb = (INT)((2 * noChannels * 16000 + (fs >> 1)) / fs); /* 16 kHz band */ + goalSb = findClosestEntry(goalSb, v_k_master, numMaster, + 1); /* Adapt region to master-table */ /* First patch */ sourceStartBand = hTonCorr->shiftStartSb + xoverOffset; targetStopBand = lsb + xoverOffset; - /* even (odd) numbered channel must be patched to even (odd) numbered channel */ + /* even (odd) numbered channel must be patched to even (odd) numbered channel + */ patch = 0; - while(targetStopBand < usb) { - + while (targetStopBand < usb) { /* To many patches */ - if (patch >= MAX_NUM_PATCHES) - return(1); /*Number of patches to high */ + if (patch >= MAX_NUM_PATCHES) return (1); /*Number of patches to high */ patchParam[patch].guardStartBand = targetStopBand; targetStopBand += sbGuard; patchParam[patch].targetStartBand = targetStopBand; - numBandsInPatch = goalSb - targetStopBand; /* get the desired range of the patch */ + numBandsInPatch = + goalSb - targetStopBand; /* get the desired range of the patch */ - if ( numBandsInPatch >= lsb - sourceStartBand ) { + 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 */ + patchDistance = + targetStopBand - sourceStartBand; /* get the targetOffset */ + patchDistance = + patchDistance & ~1; /* rounding off odd numbers and make all even */ numBandsInPatch = lsb - (targetStopBand - patchDistance); - numBandsInPatch = findClosestEntry(targetStopBand + numBandsInPatch, v_k_master, numMaster, 0) - - targetStopBand; /* Adapt region to master-table */ + numBandsInPatch = findClosestEntry(targetStopBand + numBandsInPatch, + v_k_master, numMaster, 0) - + targetStopBand; /* Adapt region to master-table */ } - /* 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 */ + /* 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) { patch--; } else { patchParam[patch].sourceStartBand = targetStopBand - patchDistance; - patchParam[patch].targetBandOffs = patchDistance; + patchParam[patch].targetBandOffs = patchDistance; patchParam[patch].numBandsInPatch = numBandsInPatch; - patchParam[patch].sourceStopBand = patchParam[patch].sourceStartBand + numBandsInPatch; + patchParam[patch].sourceStopBand = + patchParam[patch].sourceStartBand + numBandsInPatch; targetStopBand += patchParam[patch].numBandsInPatch; } @@ -573,42 +594,38 @@ resetPatch(HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct sourceStartBand = hTonCorr->shiftStartSb; /* Check if we are close to goalSb */ - if( fixp_abs(targetStopBand - goalSb) < 3) { + if (fixp_abs(targetStopBand - goalSb) < 3) { goalSb = usb; } patch++; - } patch--; /* if highest patch contains less than three subband: skip it */ - if ( patchParam[patch].numBandsInPatch < 3 && patch > 0 ) { + if (patchParam[patch].numBandsInPatch < 3 && patch > 0) { patch--; - targetStopBand = patchParam[patch].targetStartBand + patchParam[patch].numBandsInPatch; } hTonCorr->noOfPatches = patch + 1; - /* Assign the index-vector, so we know where to look for the high-band. -1 represents a guard-band. */ - for(k = 0; k < hTonCorr->patchParam[0].guardStartBand; k++) + for (k = 0; k < hTonCorr->patchParam[0].guardStartBand; k++) hTonCorr->indexVector[k] = k; - for(i = 0; i < hTonCorr->noOfPatches; i++) - { - INT sourceStart = hTonCorr->patchParam[i].sourceStartBand; - INT targetStart = hTonCorr->patchParam[i].targetStartBand; - INT numberOfBands = hTonCorr->patchParam[i].numBandsInPatch; + for (i = 0; i < hTonCorr->noOfPatches; i++) { + INT sourceStart = hTonCorr->patchParam[i].sourceStartBand; + INT targetStart = hTonCorr->patchParam[i].targetStartBand; + INT numberOfBands = hTonCorr->patchParam[i].numBandsInPatch; INT startGuardBand = hTonCorr->patchParam[i].guardStartBand; - for(k = 0; k < (targetStart- startGuardBand); k++) - hTonCorr->indexVector[startGuardBand+k] = -1; + for (k = 0; k < (targetStart - startGuardBand); k++) + hTonCorr->indexVector[startGuardBand + k] = -1; - for(k = 0; k < numberOfBands; k++) - hTonCorr->indexVector[targetStart+k] = sourceStart+k; + for (k = 0; k < numberOfBands; k++) + hTonCorr->indexVector[targetStart + k] = sourceStart + k; } return (0); @@ -624,27 +641,41 @@ resetPatch(HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct \return errorCode, noError if successful. */ /**************************************************************************/ -INT -FDKsbrEnc_CreateTonCorrParamExtr(HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Pointer to handle to SBR_TON_CORR struct. */ - INT chan) /*!< Channel index, needed for mem allocation */ +INT FDKsbrEnc_CreateTonCorrParamExtr( + HANDLE_SBR_TON_CORR_EST + hTonCorr, /*!< Pointer to handle to SBR_TON_CORR struct. */ + INT chan) /*!< Channel index, needed for mem allocation */ { INT i; - FIXP_DBL* quotaMatrix = GetRam_Sbr_quotaMatrix(chan); - INT* signMatrix = GetRam_Sbr_signMatrix(chan); + FIXP_DBL *quotaMatrix = GetRam_Sbr_quotaMatrix(chan); + INT *signMatrix = GetRam_Sbr_signMatrix(chan); + + if ((NULL == quotaMatrix) || (NULL == signMatrix)) { + goto bail; + } FDKmemclear(hTonCorr, sizeof(SBR_TON_CORR_EST)); - for (i=0; i<MAX_NO_OF_ESTIMATES; i++) { - hTonCorr->quotaMatrix[i] = quotaMatrix + (i*QMF_CHANNELS); - hTonCorr->signMatrix[i] = signMatrix + (i*QMF_CHANNELS); + for (i = 0; i < MAX_NO_OF_ESTIMATES; i++) { + hTonCorr->quotaMatrix[i] = quotaMatrix + (i * 64); + hTonCorr->signMatrix[i] = signMatrix + (i * 64); } - FDKsbrEnc_CreateSbrMissingHarmonicsDetector (&hTonCorr->sbrMissingHarmonicsDetector, chan); + if (0 != FDKsbrEnc_CreateSbrMissingHarmonicsDetector( + &hTonCorr->sbrMissingHarmonicsDetector, chan)) { + goto bail; + } return 0; -} +bail: + hTonCorr->quotaMatrix[0] = quotaMatrix; + hTonCorr->signMatrix[0] = signMatrix; + + FDKsbrEnc_DeleteTonCorrParamExtr(hTonCorr); + return -1; +} /**************************************************************************/ /*! @@ -656,27 +687,29 @@ FDKsbrEnc_CreateTonCorrParamExtr(HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Pointer \return errorCode, noError if successful. */ /**************************************************************************/ -INT -FDKsbrEnc_InitTonCorrParamExtr (INT frameSize, /*!< Current SBR frame size. */ - HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Pointer to handle to SBR_TON_CORR struct. */ - HANDLE_SBR_CONFIG_DATA sbrCfg, /*!< Pointer to SBR configuration parameters. */ - INT timeSlots, /*!< Number of time-slots per frame */ - INT xposCtrl, /*!< Different patch modes. */ - INT ana_max_level, /*!< Maximum level of the adaptive noise. */ - INT noiseBands, /*!< Number of noise bands per octave. */ - INT noiseFloorOffset, /*!< Noise floor offset. */ - UINT useSpeechConfig) /*!< Speech or music tuning. */ +INT FDKsbrEnc_InitTonCorrParamExtr( + INT frameSize, /*!< Current SBR frame size. */ + HANDLE_SBR_TON_CORR_EST + hTonCorr, /*!< Pointer to handle to SBR_TON_CORR struct. */ + HANDLE_SBR_CONFIG_DATA + sbrCfg, /*!< Pointer to SBR configuration parameters. */ + INT timeSlots, /*!< Number of time-slots per frame */ + INT xposCtrl, /*!< Different patch modes. */ + INT ana_max_level, /*!< Maximum level of the adaptive noise. */ + INT noiseBands, /*!< Number of noise bands per octave. */ + INT noiseFloorOffset, /*!< Noise floor offset. */ + UINT useSpeechConfig) /*!< Speech or music tuning. */ { INT nCols = sbrCfg->noQmfSlots; - INT fs = sbrCfg->sampleFreq; + INT fs = sbrCfg->sampleFreq; INT noQmfChannels = sbrCfg->noQmfBands; INT highBandStartSb = sbrCfg->freqBandTable[LOW_RES][0]; - UCHAR *v_k_master = sbrCfg->v_k_master; - INT numMaster = sbrCfg->num_Master; + UCHAR *v_k_master = sbrCfg->v_k_master; + INT numMaster = sbrCfg->num_Master; - UCHAR **freqBandTable = sbrCfg->freqBandTable; - INT *nSfb = sbrCfg->nSfb; + UCHAR **freqBandTable = sbrCfg->freqBandTable; + INT *nSfb = sbrCfg->nSfb; INT i; @@ -686,113 +719,102 @@ FDKsbrEnc_InitTonCorrParamExtr (INT frameSize, /*!< Current */ if (sbrCfg->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { switch (timeSlots) { - case NUMBER_TIME_SLOTS_1920: - hTonCorr->lpcLength[0] = 8 - LPC_ORDER; - hTonCorr->lpcLength[1] = 7 - LPC_ORDER; - hTonCorr->numberOfEstimates = NO_OF_ESTIMATES_LD; - hTonCorr->numberOfEstimatesPerFrame = 2; /* sbrCfg->noQmfSlots / 7 */ - hTonCorr->frameStartIndexInvfEst = 0; - hTonCorr->transientPosOffset = FRAME_MIDDLE_SLOT_512LD; - break; - case NUMBER_TIME_SLOTS_2048: - hTonCorr->lpcLength[0] = 8 - LPC_ORDER; - hTonCorr->lpcLength[1] = 8 - LPC_ORDER; - hTonCorr->numberOfEstimates = NO_OF_ESTIMATES_LD; - hTonCorr->numberOfEstimatesPerFrame = 2; /* sbrCfg->noQmfSlots / 8 */ - hTonCorr->frameStartIndexInvfEst = 0; - hTonCorr->transientPosOffset = FRAME_MIDDLE_SLOT_512LD; - break; + case NUMBER_TIME_SLOTS_1920: + hTonCorr->lpcLength[0] = 8 - LPC_ORDER; + hTonCorr->lpcLength[1] = 7 - LPC_ORDER; + hTonCorr->numberOfEstimates = NO_OF_ESTIMATES_LD; + hTonCorr->numberOfEstimatesPerFrame = 2; /* sbrCfg->noQmfSlots / 7 */ + hTonCorr->frameStartIndexInvfEst = 0; + hTonCorr->transientPosOffset = FRAME_MIDDLE_SLOT_512LD; + break; + case NUMBER_TIME_SLOTS_2048: + hTonCorr->lpcLength[0] = 8 - LPC_ORDER; + hTonCorr->lpcLength[1] = 8 - LPC_ORDER; + hTonCorr->numberOfEstimates = NO_OF_ESTIMATES_LD; + hTonCorr->numberOfEstimatesPerFrame = 2; /* sbrCfg->noQmfSlots / 8 */ + hTonCorr->frameStartIndexInvfEst = 0; + hTonCorr->transientPosOffset = FRAME_MIDDLE_SLOT_512LD; + break; } } else - switch (timeSlots) { - case NUMBER_TIME_SLOTS_2048: - hTonCorr->lpcLength[0] = 16 - LPC_ORDER; /* blockLength[0] */ - hTonCorr->lpcLength[1] = 16 - LPC_ORDER; /* blockLength[0] */ - hTonCorr->numberOfEstimates = NO_OF_ESTIMATES_LC; - hTonCorr->numberOfEstimatesPerFrame = sbrCfg->noQmfSlots / 16; - hTonCorr->frameStartIndexInvfEst = 0; - hTonCorr->transientPosOffset = FRAME_MIDDLE_SLOT_2048; - break; - case NUMBER_TIME_SLOTS_1920: - hTonCorr->lpcLength[0] = 15 - LPC_ORDER; /* blockLength[0] */ - hTonCorr->lpcLength[1] = 15 - LPC_ORDER; /* blockLength[0] */ - hTonCorr->numberOfEstimates = NO_OF_ESTIMATES_LC; - hTonCorr->numberOfEstimatesPerFrame = sbrCfg->noQmfSlots / 15; - hTonCorr->frameStartIndexInvfEst = 0; - hTonCorr->transientPosOffset = FRAME_MIDDLE_SLOT_1920; - break; - default: - return -1; - } + switch (timeSlots) { + case NUMBER_TIME_SLOTS_2048: + hTonCorr->lpcLength[0] = 16 - LPC_ORDER; /* blockLength[0] */ + hTonCorr->lpcLength[1] = 16 - LPC_ORDER; /* blockLength[0] */ + hTonCorr->numberOfEstimates = NO_OF_ESTIMATES_LC; + hTonCorr->numberOfEstimatesPerFrame = sbrCfg->noQmfSlots / 16; + hTonCorr->frameStartIndexInvfEst = 0; + hTonCorr->transientPosOffset = FRAME_MIDDLE_SLOT_2048; + break; + case NUMBER_TIME_SLOTS_1920: + hTonCorr->lpcLength[0] = 15 - LPC_ORDER; /* blockLength[0] */ + hTonCorr->lpcLength[1] = 15 - LPC_ORDER; /* blockLength[0] */ + hTonCorr->numberOfEstimates = NO_OF_ESTIMATES_LC; + hTonCorr->numberOfEstimatesPerFrame = sbrCfg->noQmfSlots / 15; + hTonCorr->frameStartIndexInvfEst = 0; + hTonCorr->transientPosOffset = FRAME_MIDDLE_SLOT_1920; + break; + default: + return -1; + } - hTonCorr->bufferLength = nCols; - hTonCorr->stepSize = hTonCorr->lpcLength[0] + LPC_ORDER; /* stepSize[0] implicitly 0. */ + hTonCorr->bufferLength = nCols; + hTonCorr->stepSize = + hTonCorr->lpcLength[0] + LPC_ORDER; /* stepSize[0] implicitly 0. */ - hTonCorr->nextSample = LPC_ORDER; /* firstSample */ - hTonCorr->move = hTonCorr->numberOfEstimates - hTonCorr->numberOfEstimatesPerFrame; /* Number of estimates to move when buffering.*/ - hTonCorr->startIndexMatrix = hTonCorr->numberOfEstimates - hTonCorr->numberOfEstimatesPerFrame; /* Where to store the latest estimations in the tonality Matrix.*/ - hTonCorr->frameStartIndex = 0; /* Where in the tonality matrix the current frame (to be sent to the decoder) starts. */ + hTonCorr->nextSample = LPC_ORDER; /* firstSample */ + hTonCorr->move = hTonCorr->numberOfEstimates - + hTonCorr->numberOfEstimatesPerFrame; /* Number of estimates + to move when + buffering.*/ + if (hTonCorr->move < 0) { + return -1; + } + hTonCorr->startIndexMatrix = + hTonCorr->numberOfEstimates - + hTonCorr->numberOfEstimatesPerFrame; /* Where to store the latest + estimations in the tonality + Matrix.*/ + hTonCorr->frameStartIndex = 0; /* Where in the tonality matrix the current + frame (to be sent to the decoder) starts. */ hTonCorr->prevTransientFlag = 0; hTonCorr->transientNextFrame = 0; hTonCorr->noQmfChannels = noQmfChannels; - for (i=0; i<hTonCorr->numberOfEstimates; i++) { - FDKmemclear (hTonCorr->quotaMatrix[i] , sizeof(FIXP_DBL)*noQmfChannels); - FDKmemclear (hTonCorr->signMatrix[i] , sizeof(INT)*noQmfChannels); + for (i = 0; i < hTonCorr->numberOfEstimates; i++) { + FDKmemclear(hTonCorr->quotaMatrix[i], sizeof(FIXP_DBL) * noQmfChannels); + FDKmemclear(hTonCorr->signMatrix[i], sizeof(INT) * noQmfChannels); } - /* Reset the patch.*/ + /* Reset the patch.*/ hTonCorr->guard = 0; hTonCorr->shiftStartSb = 1; - if(resetPatch(hTonCorr, - xposCtrl, - highBandStartSb, - v_k_master, - numMaster, - fs, - noQmfChannels)) - return(1); - - if(FDKsbrEnc_InitSbrNoiseFloorEstimate (&hTonCorr->sbrNoiseFloorEstimate, - ana_max_level, - freqBandTable[LO], - nSfb[LO], - noiseBands, - noiseFloorOffset, - timeSlots, - useSpeechConfig)) - return(1); - - - if(FDKsbrEnc_initInvFiltDetector(&hTonCorr->sbrInvFilt, - hTonCorr->sbrNoiseFloorEstimate.freqBandTableQmf, - hTonCorr->sbrNoiseFloorEstimate.noNoiseBands, - useSpeechConfig)) - return(1); - + if (resetPatch(hTonCorr, xposCtrl, highBandStartSb, v_k_master, numMaster, fs, + noQmfChannels)) + return (1); + if (FDKsbrEnc_InitSbrNoiseFloorEstimate( + &hTonCorr->sbrNoiseFloorEstimate, ana_max_level, freqBandTable[LO], + nSfb[LO], noiseBands, noiseFloorOffset, timeSlots, useSpeechConfig)) + return (1); - if(FDKsbrEnc_InitSbrMissingHarmonicsDetector( - &hTonCorr->sbrMissingHarmonicsDetector, - fs, - frameSize, - nSfb[HI], - noQmfChannels, - hTonCorr->numberOfEstimates, - hTonCorr->move, - hTonCorr->numberOfEstimatesPerFrame, - sbrCfg->sbrSyntaxFlags)) - return(1); - + if (FDKsbrEnc_initInvFiltDetector( + &hTonCorr->sbrInvFilt, + hTonCorr->sbrNoiseFloorEstimate.freqBandTableQmf, + hTonCorr->sbrNoiseFloorEstimate.noNoiseBands, useSpeechConfig)) + return (1); + if (FDKsbrEnc_InitSbrMissingHarmonicsDetector( + &hTonCorr->sbrMissingHarmonicsDetector, fs, frameSize, nSfb[HI], + noQmfChannels, hTonCorr->numberOfEstimates, hTonCorr->move, + hTonCorr->numberOfEstimatesPerFrame, sbrCfg->sbrSyntaxFlags)) + return (1); return (0); } - - /**************************************************************************/ /*! \brief resets tonality correction parameter module. @@ -803,59 +825,48 @@ FDKsbrEnc_InitTonCorrParamExtr (INT frameSize, /*!< Current */ /**************************************************************************/ -INT -FDKsbrEnc_ResetTonCorrParamExtr(HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */ - INT xposctrl, /*!< Different patch modes. */ - INT highBandStartSb, /*!< Start band of the SBR range. */ - UCHAR *v_k_master, /*!< Master frequency table from which all other table are derived.*/ - INT numMaster, /*!< Number of elements in the master table. */ - INT fs, /*!< Sampling frequency (of the SBR part). */ - UCHAR ** freqBandTable, /*!< Frequency band table for low-res and high-res. */ - INT* nSfb, /*!< Number of frequency bands (hig-res and low-res). */ - INT noQmfChannels /*!< Number of QMF channels. */ - ) -{ - +INT FDKsbrEnc_ResetTonCorrParamExtr( + HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */ + INT xposctrl, /*!< Different patch modes. */ + INT highBandStartSb, /*!< Start band of the SBR range. */ + UCHAR *v_k_master, /*!< Master frequency table from which all other table + are derived.*/ + INT numMaster, /*!< Number of elements in the master table. */ + INT fs, /*!< Sampling frequency (of the SBR part). */ + UCHAR * + *freqBandTable, /*!< Frequency band table for low-res and high-res. */ + INT *nSfb, /*!< Number of frequency bands (hig-res and low-res). */ + INT noQmfChannels /*!< Number of QMF channels. */ +) { /* Reset the patch.*/ hTonCorr->guard = 0; hTonCorr->shiftStartSb = 1; - if(resetPatch(hTonCorr, - xposctrl, - highBandStartSb, - v_k_master, - numMaster, - fs, - noQmfChannels)) - return(1); - - + if (resetPatch(hTonCorr, xposctrl, highBandStartSb, v_k_master, numMaster, fs, + noQmfChannels)) + return (1); /* Reset the noise floor estimate.*/ - if(FDKsbrEnc_resetSbrNoiseFloorEstimate (&hTonCorr->sbrNoiseFloorEstimate, - freqBandTable[LO], - nSfb[LO])) - return(1); + if (FDKsbrEnc_resetSbrNoiseFloorEstimate(&hTonCorr->sbrNoiseFloorEstimate, + freqBandTable[LO], nSfb[LO])) + return (1); /* Reset the inveerse filtereing detector. */ - if(FDKsbrEnc_resetInvFiltDetector(&hTonCorr->sbrInvFilt, - hTonCorr->sbrNoiseFloorEstimate.freqBandTableQmf, - hTonCorr->sbrNoiseFloorEstimate.noNoiseBands)) - return(1); -/* Reset the missing harmonics detector. */ - if(FDKsbrEnc_ResetSbrMissingHarmonicsDetector (&hTonCorr->sbrMissingHarmonicsDetector, - nSfb[HI])) - return(1); + if (FDKsbrEnc_resetInvFiltDetector( + &hTonCorr->sbrInvFilt, + hTonCorr->sbrNoiseFloorEstimate.freqBandTableQmf, + hTonCorr->sbrNoiseFloorEstimate.noNoiseBands)) + return (1); + /* Reset the missing harmonics detector. */ + if (FDKsbrEnc_ResetSbrMissingHarmonicsDetector( + &hTonCorr->sbrMissingHarmonicsDetector, nSfb[HI])) + return (1); return (0); } - - - - /**************************************************************************/ /*! \brief Deletes the tonality correction paramtere module. @@ -866,16 +877,15 @@ FDKsbrEnc_ResetTonCorrParamExtr(HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to */ /**************************************************************************/ -void -FDKsbrEnc_DeleteTonCorrParamExtr (HANDLE_SBR_TON_CORR_EST hTonCorr) /*!< Handle to SBR_TON_CORR struct. */ +void FDKsbrEnc_DeleteTonCorrParamExtr( + HANDLE_SBR_TON_CORR_EST hTonCorr) /*!< Handle to SBR_TON_CORR struct. */ { - if (hTonCorr) { + FreeRam_Sbr_quotaMatrix(hTonCorr->quotaMatrix); - FreeRam_Sbr_quotaMatrix(hTonCorr->quotaMatrix); - - FreeRam_Sbr_signMatrix(hTonCorr->signMatrix); + FreeRam_Sbr_signMatrix(hTonCorr->signMatrix); - FDKsbrEnc_DeleteSbrMissingHarmonicsDetector (&hTonCorr->sbrMissingHarmonicsDetector); + FDKsbrEnc_DeleteSbrMissingHarmonicsDetector( + &hTonCorr->sbrMissingHarmonicsDetector); } } diff --git a/libSBRenc/src/ton_corr.h b/libSBRenc/src/ton_corr.h index 504ab03..91aa278 100644 --- a/libSBRenc/src/ton_corr.h +++ b/libSBRenc/src/ton_corr.h @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,134 +90,169 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ /*! \file \brief General tonality correction detector module. */ -#ifndef _TON_CORR_EST_H -#define _TON_CORR_EST_H +#ifndef TON_CORR_H +#define TON_CORR_H #include "sbr_encoder.h" #include "mh_det.h" #include "nf_est.h" #include "invf_est.h" - #define MAX_NUM_PATCHES 6 #define SCALE_NRGVEC 4 /** parameter set for one single patch */ typedef struct { - INT sourceStartBand; /*!< first band in lowbands where to take the samples from */ - INT sourceStopBand; /*!< first band in lowbands which is not included in the patch anymore */ - INT guardStartBand; /*!< first band in highbands to be filled with zeros in order to - reduce interferences between patches */ - INT targetStartBand; /*!< first band in highbands to be filled with whitened lowband signal */ - INT targetBandOffs; /*!< difference between 'startTargetBand' and 'startSourceBand' */ - INT numBandsInPatch; /*!< number of consecutive bands in this one patch */ + INT sourceStartBand; /*!< first band in lowbands where to take the samples + from */ + INT sourceStopBand; /*!< first band in lowbands which is not included in the + patch anymore */ + INT guardStartBand; /*!< first band in highbands to be filled with zeros in + order to reduce interferences between patches */ + INT targetStartBand; /*!< first band in highbands to be filled with whitened + lowband signal */ + INT targetBandOffs; /*!< difference between 'startTargetBand' and + 'startSourceBand' */ + INT numBandsInPatch; /*!< number of consecutive bands in this one patch */ } PATCH_PARAM; - - - -typedef struct -{ - INT switchInverseFilt; /*!< Flag to enable dynamic adaption of invf. detection */ +typedef struct { + INT switchInverseFilt; /*!< Flag to enable dynamic adaption of invf. detection + */ INT noQmfChannels; - INT bufferLength; /*!< Length of the r and i buffers. */ - INT stepSize; /*!< Stride for the lpc estimate. */ - INT numberOfEstimates; /*!< The total number of estiamtes, available in the quotaMatrix.*/ - UINT numberOfEstimatesPerFrame; /*!< The number of estimates per frame available in the quotaMatrix.*/ - INT lpcLength[2]; /*!< Segment length used for second order LPC analysis.*/ - INT nextSample; /*!< Where to start the LPC analysis of the current frame.*/ - INT move; /*!< How many estimates to move in the quotaMatrix, when buffering. */ - INT frameStartIndex; /*!< The start index for the current frame in the r and i buffers. */ - INT startIndexMatrix; /*!< The start index for the current frame in the quotaMatrix. */ - INT frameStartIndexInvfEst; /*!< The start index of the inverse filtering, not the same as the others, - dependent on what decoder is used (buffer opt, or no buffer opt). */ - INT prevTransientFlag; /*!< The transisent flag (from the transient detector) for the previous frame. */ - INT transientNextFrame; /*!< Flag to indicate that the transient will show up in the next frame. */ - INT transientPosOffset; /*!< An offset value to match the transient pos as calculated by the transient detector - with the actual position in the frame.*/ - - INT *signMatrix[MAX_NO_OF_ESTIMATES]; /*!< Matrix holding the sign of each channe, i.e. indicating in what - part of a QMF channel a possible sine is. */ - - FIXP_DBL *quotaMatrix[MAX_NO_OF_ESTIMATES];/*!< Matrix holding the quota values for all estimates, all channels. */ - - FIXP_DBL nrgVector[MAX_NO_OF_ESTIMATES]; /*!< Vector holding the averaged energies for every QMF band. */ - FIXP_DBL nrgVectorFreq[QMF_CHANNELS]; /*!< Vector holding the averaged energies for every QMF channel */ - - SCHAR indexVector[QMF_CHANNELS]; /*!< Index vector poINTing to the correct lowband channel, - when indexing a highband channel, -1 represents a guard band */ - PATCH_PARAM patchParam[MAX_NUM_PATCHES]; /*!< new parameter set for patching */ - INT guard; /*!< number of guardbands between every patch */ - INT shiftStartSb; /*!< lowest subband of source range to be included in the patches */ - INT noOfPatches; /*!< number of patches */ - - SBR_MISSING_HARMONICS_DETECTOR sbrMissingHarmonicsDetector; /*!< SBR_MISSING_HARMONICS_DETECTOR struct. */ - SBR_NOISE_FLOOR_ESTIMATE sbrNoiseFloorEstimate; /*!< SBR_NOISE_FLOOR_ESTIMATE struct. */ - SBR_INV_FILT_EST sbrInvFilt; /*!< SBR_INV_FILT_EST struct. */ -} -SBR_TON_CORR_EST; - -typedef SBR_TON_CORR_EST *HANDLE_SBR_TON_CORR_EST; - -void -FDKsbrEnc_TonCorrParamExtr(HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */ - INVF_MODE* infVec, /*!< Vector where the inverse filtering levels will be stored. */ - FIXP_DBL * noiseLevels, /*!< Vector where the noise levels will be stored. */ - INT* missingHarmonicFlag, /*!< Flag set to one or zero, dependent on if any strong sines are missing.*/ - UCHAR* missingHarmonicsIndex, /*!< Vector indicating where sines are missing. */ - UCHAR* envelopeCompensation, /*!< Vector to store compensation values for the energies in. */ - const SBR_FRAME_INFO *frameInfo, /*!< Frame info struct, contains the time and frequency grid of the current frame.*/ - UCHAR* transientInfo, /*!< Transient info.*/ - UCHAR * freqBandTable, /*!< Frequency band tables for high-res.*/ - INT nSfb, /*!< Number of scalefactor bands for high-res. */ - XPOS_MODE xposType, /*!< Type of transposer used in the decoder.*/ - UINT sbrSyntaxFlags - ); - -INT -FDKsbrEnc_CreateTonCorrParamExtr(HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Pointer to handle to SBR_TON_CORR struct. */ - INT chan); /*!< Channel index, needed for mem allocation */ - -INT -FDKsbrEnc_InitTonCorrParamExtr(INT frameSize, /*!< Current SBR frame size. */ - HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Pointer to handle to SBR_TON_CORR struct. */ - HANDLE_SBR_CONFIG_DATA sbrCfg, /*!< Pointer to SBR configuration parameters. */ - INT timeSlots, /*!< Number of time-slots per frame */ - INT xposCtrl, /*!< Different patch modes. */ - INT ana_max_level, /*!< Maximum level of the adaptive noise. */ - INT noiseBands, /*!< Number of noise bands per octave. */ - INT noiseFloorOffset, /*!< Noise floor offset. */ - UINT useSpeechConfig /*!< Speech or music tuning. */ - ); - -void -FDKsbrEnc_DeleteTonCorrParamExtr(HANDLE_SBR_TON_CORR_EST hTonCorr); /*!< Handle to SBR_TON_CORR struct. */ - - -void -FDKsbrEnc_CalculateTonalityQuotas(HANDLE_SBR_TON_CORR_EST hTonCorr, - FIXP_DBL **sourceBufferReal, - FIXP_DBL **sourceBufferImag, - INT usb, - INT qmfScale /*!< sclefactor of QMF subsamples */ - ); - -INT -FDKsbrEnc_ResetTonCorrParamExtr(HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */ - INT xposctrl, /*!< Different patch modes. */ - INT highBandStartSb, /*!< Start band of the SBR range. */ - UCHAR *v_k_master, /*!< Master frequency table from which all other table are derived.*/ - INT numMaster, /*!< Number of elements in the master table. */ - INT fs, /*!< Sampling frequency (of the SBR part). */ - UCHAR** freqBandTable, /*!< Frequency band table for low-res and high-res. */ - INT* nSfb, /*!< Number of frequency bands (hig-res and low-res). */ - INT noQmfChannels /*!< Number of QMF channels. */ - ); + INT bufferLength; /*!< Length of the r and i buffers. */ + INT stepSize; /*!< Stride for the lpc estimate. */ + INT numberOfEstimates; /*!< The total number of estiamtes, available in the + quotaMatrix.*/ + UINT numberOfEstimatesPerFrame; /*!< The number of estimates per frame + available in the quotaMatrix.*/ + INT lpcLength[2]; /*!< Segment length used for second order LPC analysis.*/ + INT nextSample; /*!< Where to start the LPC analysis of the current frame.*/ + INT move; /*!< How many estimates to move in the quotaMatrix, when buffering. + */ + INT frameStartIndex; /*!< The start index for the current frame in the r and i + buffers. */ + INT startIndexMatrix; /*!< The start index for the current frame in the + quotaMatrix. */ + INT frameStartIndexInvfEst; /*!< The start index of the inverse filtering, not + the same as the others, dependent on what + decoder is used (buffer opt, or no buffer opt). + */ + INT prevTransientFlag; /*!< The transisent flag (from the transient detector) + for the previous frame. */ + INT transientNextFrame; /*!< Flag to indicate that the transient will show up + in the next frame. */ + INT transientPosOffset; /*!< An offset value to match the transient pos as + calculated by the transient detector with the + actual position in the frame.*/ + + INT* signMatrix[MAX_NO_OF_ESTIMATES]; /*!< Matrix holding the sign of each + channe, i.e. indicating in what part + of a QMF channel a possible sine is. + */ + + FIXP_DBL* quotaMatrix[MAX_NO_OF_ESTIMATES]; /*!< Matrix holding the quota + values for all estimates, all + channels. */ + + FIXP_DBL nrgVector[MAX_NO_OF_ESTIMATES]; /*!< Vector holding the averaged + energies for every QMF band. */ + FIXP_DBL nrgVectorFreq[64]; /*!< Vector holding the averaged energies for + every QMF channel */ + + SCHAR indexVector[64]; /*!< Index vector poINTing to the correct lowband + channel, when indexing a highband channel, -1 + represents a guard band */ + PATCH_PARAM + patchParam[MAX_NUM_PATCHES]; /*!< new parameter set for patching */ + INT guard; /*!< number of guardbands between every patch */ + INT shiftStartSb; /*!< lowest subband of source range to be included in the + patches */ + INT noOfPatches; /*!< number of patches */ + + SBR_MISSING_HARMONICS_DETECTOR + sbrMissingHarmonicsDetector; /*!< SBR_MISSING_HARMONICS_DETECTOR struct. + */ + SBR_NOISE_FLOOR_ESTIMATE + sbrNoiseFloorEstimate; /*!< SBR_NOISE_FLOOR_ESTIMATE struct. */ + SBR_INV_FILT_EST sbrInvFilt; /*!< SBR_INV_FILT_EST struct. */ +} SBR_TON_CORR_EST; + +typedef SBR_TON_CORR_EST* HANDLE_SBR_TON_CORR_EST; + +void FDKsbrEnc_TonCorrParamExtr( + HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */ + INVF_MODE* infVec, /*!< Vector where the inverse filtering levels will be + stored. */ + FIXP_DBL* noiseLevels, /*!< Vector where the noise levels will be stored. */ + INT* missingHarmonicFlag, /*!< Flag set to one or zero, dependent on if any + strong sines are missing.*/ + UCHAR* missingHarmonicsIndex, /*!< Vector indicating where sines are + missing. */ + UCHAR* envelopeCompensation, /*!< Vector to store compensation values for + the energies in. */ + const SBR_FRAME_INFO* frameInfo, /*!< Frame info struct, contains the time + and frequency grid of the current + frame.*/ + UCHAR* transientInfo, /*!< Transient info.*/ + UCHAR* freqBandTable, /*!< Frequency band tables for high-res.*/ + INT nSfb, /*!< Number of scalefactor bands for high-res. */ + XPOS_MODE xposType, /*!< Type of transposer used in the decoder.*/ + UINT sbrSyntaxFlags); + +INT FDKsbrEnc_CreateTonCorrParamExtr( + HANDLE_SBR_TON_CORR_EST + hTonCorr, /*!< Pointer to handle to SBR_TON_CORR struct. */ + INT chan); /*!< Channel index, needed for mem allocation */ + +INT FDKsbrEnc_InitTonCorrParamExtr( + INT frameSize, /*!< Current SBR frame size. */ + HANDLE_SBR_TON_CORR_EST + hTonCorr, /*!< Pointer to handle to SBR_TON_CORR struct. */ + HANDLE_SBR_CONFIG_DATA + sbrCfg, /*!< Pointer to SBR configuration parameters. */ + INT timeSlots, /*!< Number of time-slots per frame */ + INT xposCtrl, /*!< Different patch modes. */ + INT ana_max_level, /*!< Maximum level of the adaptive noise. */ + INT noiseBands, /*!< Number of noise bands per octave. */ + INT noiseFloorOffset, /*!< Noise floor offset. */ + UINT useSpeechConfig /*!< Speech or music tuning. */ +); + +void FDKsbrEnc_DeleteTonCorrParamExtr( + HANDLE_SBR_TON_CORR_EST hTonCorr); /*!< Handle to SBR_TON_CORR struct. */ + +void FDKsbrEnc_CalculateTonalityQuotas( + HANDLE_SBR_TON_CORR_EST hTonCorr, FIXP_DBL** sourceBufferReal, + FIXP_DBL** sourceBufferImag, INT usb, + INT qmfScale /*!< sclefactor of QMF subsamples */ +); + +INT FDKsbrEnc_ResetTonCorrParamExtr( + HANDLE_SBR_TON_CORR_EST hTonCorr, /*!< Handle to SBR_TON_CORR struct. */ + INT xposctrl, /*!< Different patch modes. */ + INT highBandStartSb, /*!< Start band of the SBR range. */ + UCHAR* v_k_master, /*!< Master frequency table from which all other table + are derived.*/ + INT numMaster, /*!< Number of elements in the master table. */ + INT fs, /*!< Sampling frequency (of the SBR part). */ + UCHAR** + freqBandTable, /*!< Frequency band table for low-res and high-res. */ + INT* nSfb, /*!< Number of frequency bands (hig-res and low-res). */ + INT noQmfChannels /*!< Number of QMF channels. */ +); #endif - diff --git a/libSBRenc/src/tran_det.cpp b/libSBRenc/src/tran_det.cpp index 33ea60e..ba9ae68 100644 --- a/libSBRenc/src/tran_det.cpp +++ b/libSBRenc/src/tran_det.cpp @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,19 +90,28 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): Tobias Chalupka + + Description: SBR encoder transient detector + +*******************************************************************************/ #include "tran_det.h" #include "fram_gen.h" -#include "sbr_ram.h" +#include "sbrenc_ram.h" #include "sbr_misc.h" #include "genericStds.h" #define NORM_QMF_ENERGY 9.31322574615479E-10 /* 2^-30 */ -/* static FIXP_DBL ABS_THRES = fixMax( FL2FXCONST_DBL(1.28e5 * NORM_QMF_ENERGY), (FIXP_DBL)1) Minimum threshold for detecting changes */ +/* static FIXP_DBL ABS_THRES = fixMax( FL2FXCONST_DBL(1.28e5 * + * NORM_QMF_ENERGY), (FIXP_DBL)1) Minimum threshold for detecting changes */ #define ABS_THRES ((FIXP_DBL)16) /******************************************************************************* @@ -106,126 +126,128 @@ amm-info@iis.fraunhofer.de \return calculated value *******************************************************************************/ -#define NRG_SHIFT 3 /* for energy summation */ - -static FIXP_DBL spectralChange(FIXP_DBL Energies[NUMBER_TIME_SLOTS_2304][MAX_FREQ_COEFFS], - INT *scaleEnergies, - FIXP_DBL EnergyTotal, - INT nSfb, - INT start, - INT border, - INT YBufferWriteOffset, - INT stop, - INT *result_e) -{ - INT i,j; - INT len1,len2; - SCHAR energies_e_diff[NUMBER_TIME_SLOTS_2304], energies_e, energyTotal_e=19, energies_e_add; +#define NRG_SHIFT 3 /* for energy summation */ + +static FIXP_DBL spectralChange( + FIXP_DBL Energies[NUMBER_TIME_SLOTS_2304][MAX_FREQ_COEFFS], + INT *scaleEnergies, FIXP_DBL EnergyTotal, INT nSfb, INT start, INT border, + INT YBufferWriteOffset, INT stop, INT *result_e) { + INT i, j; + INT len1, len2; + SCHAR energies_e_diff[NUMBER_TIME_SLOTS_2304], energies_e, energyTotal_e = 19, + energies_e_add; SCHAR prevEnergies_e_diff, newEnergies_e_diff; - FIXP_DBL tmp0,tmp1; - FIXP_DBL accu1,accu2,accu1_init,accu2_init; + FIXP_DBL tmp0, tmp1; FIXP_DBL delta, delta_sum; INT accu_e, tmp_e; delta_sum = FL2FXCONST_DBL(0.0f); *result_e = 0; - len1 = border-start; - len2 = stop-border; + len1 = border - start; + len2 = stop - border; /* prefer borders near the middle of the frame */ - FIXP_DBL pos_weight; - pos_weight = FL2FXCONST_DBL(0.5f) - (len1*GetInvInt(len1+len2)); - pos_weight = /*FL2FXCONST_DBL(1.0)*/ (FIXP_DBL)MAXVAL_DBL - (fMult(pos_weight, pos_weight)<<2); + FIXP_DBL pos_weight; + pos_weight = FL2FXCONST_DBL(0.5f) - (len1 * GetInvInt(len1 + len2)); + pos_weight = /*FL2FXCONST_DBL(1.0)*/ (FIXP_DBL)MAXVAL_DBL - + (fMult(pos_weight, pos_weight) << 2); /*** Calc scaling for energies ***/ FDK_ASSERT(scaleEnergies[0] >= 0); FDK_ASSERT(scaleEnergies[1] >= 0); - energies_e = 19 - FDKmin(scaleEnergies[0], scaleEnergies[1]); + energies_e = 19 - fMin(scaleEnergies[0], scaleEnergies[1]); /* limit shift for energy accumulation, energies_e can be -10 min. */ if (energies_e < -10) { - energies_e_add = -10 - energies_e; - energies_e = -10; + energies_e_add = -10 - energies_e; + energies_e = -10; } else if (energies_e > 17) { - energies_e_add = energies_e - 17; - energies_e = 17; + energies_e_add = energies_e - 17; + energies_e = 17; } else { - energies_e_add = 0; + energies_e_add = 0; } - /* compensate scaling differences between scaleEnergies[0] and scaleEnergies[1] */ - prevEnergies_e_diff = scaleEnergies[0] - FDKmin(scaleEnergies[0], scaleEnergies[1]) + energies_e_add + NRG_SHIFT; - newEnergies_e_diff = scaleEnergies[1] - FDKmin(scaleEnergies[0], scaleEnergies[1]) + energies_e_add + NRG_SHIFT; + /* compensate scaling differences between scaleEnergies[0] and + * scaleEnergies[1] */ + prevEnergies_e_diff = scaleEnergies[0] - + fMin(scaleEnergies[0], scaleEnergies[1]) + + energies_e_add + NRG_SHIFT; + newEnergies_e_diff = scaleEnergies[1] - + fMin(scaleEnergies[0], scaleEnergies[1]) + + energies_e_add + NRG_SHIFT; - prevEnergies_e_diff = fMin(prevEnergies_e_diff, DFRACT_BITS-1); - newEnergies_e_diff = fMin(newEnergies_e_diff, DFRACT_BITS-1); + prevEnergies_e_diff = fMin(prevEnergies_e_diff, DFRACT_BITS - 1); + newEnergies_e_diff = fMin(newEnergies_e_diff, DFRACT_BITS - 1); - for (i=start; i<YBufferWriteOffset; i++) { + for (i = start; i < YBufferWriteOffset; i++) { energies_e_diff[i] = prevEnergies_e_diff; } - for (i=YBufferWriteOffset; i<stop; i++) { + for (i = YBufferWriteOffset; i < stop; i++) { energies_e_diff[i] = newEnergies_e_diff; } /* Sum up energies of all QMF-timeslots for both halfs */ - FDK_ASSERT(len1<=8); /* otherwise an overflow is possible */ - FDK_ASSERT(len2<=8); /* otherwise an overflow is possible */ - /* init with some energy to prevent division by zero - and to prevent splitting for very low levels */ - accu1_init = scaleValue((FL2FXCONST_DBL((1.0e6*NORM_QMF_ENERGY))),-energies_e); - accu2_init = scaleValue((FL2FXCONST_DBL((1.0e6*NORM_QMF_ENERGY))),-energies_e); - accu1_init = fMult(accu1_init, (FIXP_DBL)len1<<((DFRACT_BITS-1)-NRG_SHIFT-1))<<1; - accu2_init = fMult(accu2_init, (FIXP_DBL)len2<<((DFRACT_BITS-1)-NRG_SHIFT-1))<<1; + FDK_ASSERT(len1 <= 8); /* otherwise an overflow is possible */ + FDK_ASSERT(len2 <= 8); /* otherwise an overflow is possible */ - for (j=0; j<nSfb; j++) { - - accu1 = accu1_init; - accu2 = accu2_init; - accu_e = energies_e+3; + for (j = 0; j < nSfb; j++) { + FIXP_DBL accu1 = FL2FXCONST_DBL(0.f); + FIXP_DBL accu2 = FL2FXCONST_DBL(0.f); + accu_e = energies_e + 3; /* Sum up energies in first half */ - for (i=start; i<border; i++) { + for (i = start; i < border; i++) { accu1 += scaleValue(Energies[i][j], -energies_e_diff[i]); } /* Sum up energies in second half */ - for (i=border; i<stop; i++) { + for (i = border; i < stop; i++) { accu2 += scaleValue(Energies[i][j], -energies_e_diff[i]); } - /* Energy change in current band */ - #define LN2 FL2FXCONST_DBL(0.6931471806f) /* ln(2) */ + /* Ensure certain energy to prevent division by zero and to prevent + * splitting for very low levels */ + accu1 = fMax(accu1, (FIXP_DBL)len1); + accu2 = fMax(accu2, (FIXP_DBL)len2); + +/* Energy change in current band */ +#define LN2 FL2FXCONST_DBL(0.6931471806f) /* ln(2) */ tmp0 = fLog2(accu2, accu_e) - fLog2(accu1, accu_e); tmp1 = fLog2((FIXP_DBL)len1, 31) - fLog2((FIXP_DBL)len2, 31); delta = fMult(LN2, (tmp0 + tmp1)); - delta = (FIXP_DBL)FDKabs( delta ); + delta = (FIXP_DBL)fAbs(delta); /* Weighting with amplitude ratio of this band */ - accu_e++; - accu1>>=1; - accu2>>=1; + accu_e++; /* scale at least one bit due to (accu1+accu2) */ + accu1 >>= 1; + accu2 >>= 1; + if (accu_e & 1) { - accu_e++; - accu1>>=1; - accu2>>=1; + accu_e++; /* for a defined square result exponent, the exponent has to be + even */ + accu1 >>= 1; + accu2 >>= 1; } - delta_sum += fMult(sqrtFixp(accu1+accu2), delta); - *result_e = ((accu_e>>1) + LD_DATA_SHIFT); + delta_sum += fMult(sqrtFixp(accu1 + accu2), delta); + *result_e = ((accu_e >> 1) + LD_DATA_SHIFT); + } + + if (energyTotal_e & 1) { + energyTotal_e += 1; /* for a defined square result exponent, the exponent + has to be even */ + EnergyTotal >>= 1; } - energyTotal_e+=1; /* for a defined square result exponent, the exponent has to be even */ - EnergyTotal<<=1; delta_sum = fMult(delta_sum, invSqrtNorm2(EnergyTotal, &tmp_e)); - *result_e = *result_e + (tmp_e-(energyTotal_e>>1)); + *result_e = *result_e + (tmp_e - (energyTotal_e >> 1)); return fMult(delta_sum, pos_weight); - } - /******************************************************************************* Functionname: addLowbandEnergies ******************************************************************************* @@ -238,40 +260,37 @@ static FIXP_DBL spectralChange(FIXP_DBL Energies[NUMBER_TIME_SLOTS_2304][MAX_FRE \return total energy in the lowband, scaled by the factor 2^19 *******************************************************************************/ -static FIXP_DBL addLowbandEnergies(FIXP_DBL **Energies, - int *scaleEnergies, - int YBufferWriteOffset, - int nrgSzShift, - int tran_off, - UCHAR *freqBandTable, - int slots) -{ - FIXP_DBL nrgTotal; +static FIXP_DBL addLowbandEnergies(FIXP_DBL **Energies, int *scaleEnergies, + int YBufferWriteOffset, int nrgSzShift, + int tran_off, UCHAR *freqBandTable, + int slots) { + INT nrgTotal_e; + FIXP_DBL nrgTotal_m; FIXP_DBL accu1 = FL2FXCONST_DBL(0.0f); FIXP_DBL accu2 = FL2FXCONST_DBL(0.0f); - int tran_offdiv2 = tran_off>>nrgSzShift; - int ts,k; + int tran_offdiv2 = tran_off >> nrgSzShift; + int ts, k; /* Sum up lowband energy from one frame at offset tran_off */ /* freqBandTable[LORES] has MAX_FREQ_COEFFS/2 +1 coeefs max. */ - for (ts=tran_offdiv2; ts<YBufferWriteOffset; ts++) { + for (ts = tran_offdiv2; ts < YBufferWriteOffset; ts++) { for (k = 0; k < freqBandTable[0]; k++) { accu1 += Energies[ts][k] >> 6; } } - for (; ts<tran_offdiv2+(slots>>nrgSzShift); ts++) { + for (; ts < tran_offdiv2 + (slots >> nrgSzShift); ts++) { for (k = 0; k < freqBandTable[0]; k++) { accu2 += Energies[ts][k] >> 9; } } - nrgTotal = ( scaleValueSaturate(accu1, 1-scaleEnergies[0]) ) - + ( scaleValueSaturate(accu2, 4-scaleEnergies[1]) ); + nrgTotal_m = fAddNorm(accu1, 1 - scaleEnergies[0], accu2, + 4 - scaleEnergies[1], &nrgTotal_e); + nrgTotal_m = scaleValueSaturate(nrgTotal_m, nrgTotal_e); - return(nrgTotal); + return (nrgTotal_m); } - /******************************************************************************* Functionname: addHighbandEnergies ******************************************************************************* @@ -289,35 +308,35 @@ static FIXP_DBL addLowbandEnergies(FIXP_DBL **Energies, \return total energy in the highband, scaled by factor 2^19 *******************************************************************************/ -static FIXP_DBL addHighbandEnergies(FIXP_DBL **RESTRICT Energies, /*!< input */ - INT *scaleEnergies, - INT YBufferWriteOffset, - FIXP_DBL EnergiesM[NUMBER_TIME_SLOTS_2304][MAX_FREQ_COEFFS], /*!< Combined output */ - UCHAR *RESTRICT freqBandTable, - INT nSfb, - INT sbrSlots, - INT timeStep) -{ - INT i,j,k,slotIn,slotOut,scale[2]; - INT li,ui; +static FIXP_DBL addHighbandEnergies( + FIXP_DBL **RESTRICT Energies, /*!< input */ + INT *scaleEnergies, INT YBufferWriteOffset, + FIXP_DBL EnergiesM[NUMBER_TIME_SLOTS_2304] + [MAX_FREQ_COEFFS], /*!< Combined output */ + UCHAR *RESTRICT freqBandTable, INT nSfb, INT sbrSlots, INT timeStep) { + INT i, j, k, slotIn, slotOut, scale[2]; + INT li, ui; FIXP_DBL nrgTotal; FIXP_DBL accu = FL2FXCONST_DBL(0.0f); /* Combine QMF-timeslots to SBR-timeslots, combine QMF-bands to SBR-bands, combine Left and Right channel */ - for (slotOut=0; slotOut<sbrSlots; slotOut++) { - slotIn = timeStep*slotOut; + for (slotOut = 0; slotOut < sbrSlots; slotOut++) { + /* Note: Below slotIn = slotOut and not slotIn = timeStep*slotOut + because the Energies[] time resolution is always the SBR slot resolution + regardless of the timeStep. */ + slotIn = slotOut; - for (j=0; j<nSfb; j++) { + for (j = 0; j < nSfb; j++) { accu = FL2FXCONST_DBL(0.0f); li = freqBandTable[j]; ui = freqBandTable[j + 1]; - for (k=li; k<ui; k++) { - for (i=0; i<timeStep; i++) { - accu += (Energies[(slotIn+i)>>1][k] >> 5); + for (k = li; k < ui; k++) { + for (i = 0; i < timeStep; i++) { + accu += Energies[slotIn][k] >> 5; } } EnergiesM[slotOut][j] = accu; @@ -325,34 +344,34 @@ static FIXP_DBL addHighbandEnergies(FIXP_DBL **RESTRICT Energies, /*!< input */ } /* scale energies down before add up */ - scale[0] = fixMin(8,scaleEnergies[0]); - scale[1] = fixMin(8,scaleEnergies[1]); + scale[0] = fixMin(8, scaleEnergies[0]); + scale[1] = fixMin(8, scaleEnergies[1]); - if ((scaleEnergies[0]-scale[0]) > (DFRACT_BITS-1) || (scaleEnergies[1]-scale[0]) > (DFRACT_BITS-1)) + if ((scaleEnergies[0] - scale[0]) > (DFRACT_BITS - 1) || + (scaleEnergies[1] - scale[1]) > (DFRACT_BITS - 1)) nrgTotal = FL2FXCONST_DBL(0.0f); else { /* Now add all energies */ accu = FL2FXCONST_DBL(0.0f); - for (slotOut=0; slotOut<YBufferWriteOffset; slotOut++) { - for (j=0; j<nSfb; j++) { + for (slotOut = 0; slotOut < YBufferWriteOffset; slotOut++) { + for (j = 0; j < nSfb; j++) { accu += (EnergiesM[slotOut][j] >> scale[0]); } } - nrgTotal = accu >> (scaleEnergies[0]-scale[0]); + nrgTotal = accu >> (scaleEnergies[0] - scale[0]); - for (slotOut=YBufferWriteOffset; slotOut<sbrSlots; slotOut++) { - for (j=0; j<nSfb; j++) { + for (slotOut = YBufferWriteOffset; slotOut < sbrSlots; slotOut++) { + for (j = 0; j < nSfb; j++) { accu += (EnergiesM[slotOut][j] >> scale[0]); } } - nrgTotal = accu >> (scaleEnergies[1]-scale[1]); + nrgTotal = accu >> (scaleEnergies[1] - scale[1]); } - return(nrgTotal); + return (nrgTotal); } - /******************************************************************************* Functionname: FDKsbrEnc_frameSplitter ******************************************************************************* @@ -361,73 +380,55 @@ static FIXP_DBL addHighbandEnergies(FIXP_DBL **RESTRICT Energies, /*!< input */ If no transient has been detected before, the frame can still be splitted into 2 envelopes. *******************************************************************************/ -void -FDKsbrEnc_frameSplitter(FIXP_DBL **Energies, - INT *scaleEnergies, - HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector, - UCHAR *freqBandTable, - UCHAR *tran_vector, - int YBufferWriteOffset, - int YBufferSzShift, - int nSfb, - int timeStep, - int no_cols, - FIXP_DBL* tonality) -{ - if (tran_vector[1]==0) /* no transient was detected */ +void FDKsbrEnc_frameSplitter( + FIXP_DBL **Energies, INT *scaleEnergies, + HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector, UCHAR *freqBandTable, + UCHAR *tran_vector, int YBufferWriteOffset, int YBufferSzShift, int nSfb, + int timeStep, int no_cols, FIXP_DBL *tonality) { + if (tran_vector[1] == 0) /* no transient was detected */ { FIXP_DBL delta; INT delta_e; - FIXP_DBL (*EnergiesM)[MAX_FREQ_COEFFS]; - FIXP_DBL EnergyTotal,newLowbandEnergy,newHighbandEnergy; + FIXP_DBL(*EnergiesM)[MAX_FREQ_COEFFS]; + FIXP_DBL EnergyTotal, newLowbandEnergy, newHighbandEnergy; INT border; - INT sbrSlots = fMultI(GetInvInt(timeStep),no_cols); - C_ALLOC_SCRATCH_START(_EnergiesM, FIXP_DBL, NUMBER_TIME_SLOTS_2304*MAX_FREQ_COEFFS) + INT sbrSlots = fMultI(GetInvInt(timeStep), no_cols); + C_ALLOC_SCRATCH_START(_EnergiesM, FIXP_DBL, + NUMBER_TIME_SLOTS_2304 * MAX_FREQ_COEFFS) - FDK_ASSERT( sbrSlots * timeStep == no_cols ); + FDK_ASSERT(sbrSlots * timeStep == no_cols); EnergiesM = (FIXP_DBL(*)[MAX_FREQ_COEFFS])_EnergiesM; /* - Get Lowband-energy over a range of 2 frames (Look half a frame back and ahead). + Get Lowband-energy over a range of 2 frames (Look half a frame back and + ahead). */ - newLowbandEnergy = addLowbandEnergies(Energies, - scaleEnergies, - YBufferWriteOffset, - YBufferSzShift, - h_sbrTransientDetector->tran_off, - freqBandTable, - no_cols); - - newHighbandEnergy = addHighbandEnergies(Energies, - scaleEnergies, - YBufferWriteOffset, - EnergiesM, - freqBandTable, - nSfb, - sbrSlots, - timeStep); + newLowbandEnergy = addLowbandEnergies( + Energies, scaleEnergies, YBufferWriteOffset, YBufferSzShift, + h_sbrTransientDetector->tran_off, freqBandTable, no_cols); + + newHighbandEnergy = + addHighbandEnergies(Energies, scaleEnergies, YBufferWriteOffset, + EnergiesM, freqBandTable, nSfb, sbrSlots, timeStep); { - /* prevLowBandEnergy: Corresponds to 1 frame, starting with half a frame look-behind - newLowbandEnergy: Corresponds to 1 frame, starting in the middle of the current frame */ - EnergyTotal = (newLowbandEnergy + h_sbrTransientDetector->prevLowBandEnergy) >> 1; - EnergyTotal += newHighbandEnergy; + /* prevLowBandEnergy: Corresponds to 1 frame, starting with half a frame + look-behind newLowbandEnergy: Corresponds to 1 frame, starting in the + middle of the current frame */ + EnergyTotal = (newLowbandEnergy >> 1) + + (h_sbrTransientDetector->prevLowBandEnergy >> + 1); /* mean of new and prev LB NRG */ + EnergyTotal = + fAddSaturate(EnergyTotal, newHighbandEnergy); /* Add HB NRG */ /* The below border should specify the same position as the middle border of a FIXFIX-frame with 2 envelopes. */ - border = (sbrSlots+1) >> 1; - - if ( (INT)EnergyTotal&0xffffffe0 && (scaleEnergies[0]<32 || scaleEnergies[1]<32) ) /* i.e. > 31 */ { - delta = spectralChange(EnergiesM, - scaleEnergies, - EnergyTotal, - nSfb, - 0, - border, - YBufferWriteOffset, - sbrSlots, - &delta_e - ); + border = (sbrSlots + 1) >> 1; + + if ((INT)EnergyTotal & 0xffffffe0 && + (scaleEnergies[0] < 32 || scaleEnergies[1] < 32)) /* i.e. > 31 */ { + delta = spectralChange(EnergiesM, scaleEnergies, EnergyTotal, nSfb, 0, + border, YBufferWriteOffset, sbrSlots, &delta_e); } else { delta = FL2FXCONST_DBL(0.0f); delta_e = 0; @@ -437,100 +438,98 @@ FDKsbrEnc_frameSplitter(FIXP_DBL **Energies, *tonality = FL2FXCONST_DBL(0.0f); } - - if ( fIsLessThan(h_sbrTransientDetector->split_thr_m, h_sbrTransientDetector->split_thr_e, delta, delta_e) ) { + if (fIsLessThan(h_sbrTransientDetector->split_thr_m, + h_sbrTransientDetector->split_thr_e, delta, delta_e)) { tran_vector[0] = 1; /* Set flag for splitting */ } else { tran_vector[0] = 0; } - } /* Update prevLowBandEnergy */ h_sbrTransientDetector->prevLowBandEnergy = newLowbandEnergy; h_sbrTransientDetector->prevHighBandEnergy = newHighbandEnergy; - C_ALLOC_SCRATCH_END(_EnergiesM, FIXP_DBL, NUMBER_TIME_SLOTS_2304*MAX_FREQ_COEFFS) + C_ALLOC_SCRATCH_END(_EnergiesM, FIXP_DBL, + NUMBER_TIME_SLOTS_2304 * MAX_FREQ_COEFFS) } } /* * Calculate transient energy threshold for each QMF band */ -static void -calculateThresholds(FIXP_DBL **RESTRICT Energies, - INT *RESTRICT scaleEnergies, - FIXP_DBL *RESTRICT thresholds, - int YBufferWriteOffset, - int YBufferSzShift, - int noCols, - int noRows, - int tran_off) -{ - FIXP_DBL mean_val,std_val,temp; +static void calculateThresholds(FIXP_DBL **RESTRICT Energies, + INT *RESTRICT scaleEnergies, + FIXP_DBL *RESTRICT thresholds, + int YBufferWriteOffset, int YBufferSzShift, + int noCols, int noRows, int tran_off) { + FIXP_DBL mean_val, std_val, temp; FIXP_DBL i_noCols; FIXP_DBL i_noCols1; - FIXP_DBL accu,accu0,accu1; - int scaleFactor0,scaleFactor1,commonScale; - int i,j; + FIXP_DBL accu, accu0, accu1; + int scaleFactor0, scaleFactor1, commonScale; + int i, j; - i_noCols = GetInvInt(noCols + tran_off ) << YBufferSzShift; + i_noCols = GetInvInt(noCols + tran_off) << YBufferSzShift; i_noCols1 = GetInvInt(noCols + tran_off - 1) << YBufferSzShift; /* calc minimum scale of energies of previous and current frame */ - commonScale = fixMin(scaleEnergies[0],scaleEnergies[1]); + commonScale = fixMin(scaleEnergies[0], scaleEnergies[1]); /* calc scalefactors to adapt energies to common scale */ - scaleFactor0 = fixMin((scaleEnergies[0]-commonScale), (DFRACT_BITS-1)); - scaleFactor1 = fixMin((scaleEnergies[1]-commonScale), (DFRACT_BITS-1)); + scaleFactor0 = fixMin((scaleEnergies[0] - commonScale), (DFRACT_BITS - 1)); + scaleFactor1 = fixMin((scaleEnergies[1] - commonScale), (DFRACT_BITS - 1)); FDK_ASSERT((scaleFactor0 >= 0) && (scaleFactor1 >= 0)); /* calculate standard deviation in every subband */ - for (i=0; i<noRows; i++) - { - int startEnergy = (tran_off>>YBufferSzShift); - int endEnergy = ((noCols>>YBufferSzShift)+tran_off); + for (i = 0; i < noRows; i++) { + int startEnergy = (tran_off >> YBufferSzShift); + int endEnergy = ((noCols >> YBufferSzShift) + tran_off); int shift; /* calculate mean value over decimated energy values (downsampled by 2). */ accu0 = accu1 = FL2FXCONST_DBL(0.0f); - for (j=startEnergy; j<YBufferWriteOffset; j++) - accu0 += fMult(Energies[j][i], i_noCols); - for (; j<endEnergy; j++) - accu1 += fMult(Energies[j][i], i_noCols); + for (j = startEnergy; j < YBufferWriteOffset; j++) + accu0 = fMultAddDiv2(accu0, Energies[j][i], i_noCols); + for (; j < endEnergy; j++) + accu1 = fMultAddDiv2(accu1, Energies[j][i], i_noCols); - mean_val = (accu0 >> scaleFactor0) + (accu1 >> scaleFactor1); /* average */ - shift = fixMax(0,CountLeadingBits(mean_val)-6); /* -6 to keep room for accumulating upto N = 24 values */ + mean_val = ((accu0 << 1) >> scaleFactor0) + + ((accu1 << 1) >> scaleFactor1); /* average */ + shift = fixMax( + 0, CountLeadingBits(mean_val) - + 6); /* -6 to keep room for accumulating upto N = 24 values */ /* calculate standard deviation */ accu = FL2FXCONST_DBL(0.0f); /* summe { ((mean_val-nrg)^2) * i_noCols1 } */ - for (j=startEnergy; j<YBufferWriteOffset; j++) { - temp = ((FIXP_DBL)mean_val - ((FIXP_DBL)Energies[j][i] >> scaleFactor0))<<shift; - temp = fPow2(temp); - temp = fMult(temp, i_noCols1); - accu += temp; + for (j = startEnergy; j < YBufferWriteOffset; j++) { + temp = ((FIXP_DBL)mean_val - ((FIXP_DBL)Energies[j][i] >> scaleFactor0)) + << shift; + temp = fPow2Div2(temp); + accu = fMultAddDiv2(accu, temp, i_noCols1); } - for (; j<endEnergy; j++) { - temp = ((FIXP_DBL)mean_val - ((FIXP_DBL)Energies[j][i] >> scaleFactor1))<<shift; - temp = fPow2(temp); - temp = fMult(temp, i_noCols1); - accu += temp; + for (; j < endEnergy; j++) { + temp = ((FIXP_DBL)mean_val - ((FIXP_DBL)Energies[j][i] >> scaleFactor1)) + << shift; + temp = fPow2Div2(temp); + accu = fMultAddDiv2(accu, temp, i_noCols1); } - - std_val = sqrtFixp(accu)>>shift; /* standard deviation */ + accu <<= 2; + std_val = sqrtFixp(accu) >> shift; /* standard deviation */ /* Take new threshold as average of calculated standard deviation ratio and old threshold if greater than absolute threshold */ - temp = ( commonScale<=(DFRACT_BITS-1) ) - ? fMult(FL2FXCONST_DBL(0.66f), thresholds[i]) + (fMult(FL2FXCONST_DBL(0.34f), std_val) >> commonScale) - : (FIXP_DBL) 0; + temp = (commonScale <= (DFRACT_BITS - 1)) + ? fMult(FL2FXCONST_DBL(0.66f), thresholds[i]) + + (fMult(FL2FXCONST_DBL(0.34f), std_val) >> commonScale) + : (FIXP_DBL)0; - thresholds[i] = fixMax(ABS_THRES,temp); + thresholds[i] = fixMax(ABS_THRES, temp); FDK_ASSERT(commonScale >= 0); } @@ -539,26 +538,17 @@ calculateThresholds(FIXP_DBL **RESTRICT Energies, /* * Calculate transient levels for each QMF time slot. */ -static void -extractTransientCandidates(FIXP_DBL **RESTRICT Energies, - INT *RESTRICT scaleEnergies, - FIXP_DBL *RESTRICT thresholds, - FIXP_DBL *RESTRICT transients, - int YBufferWriteOffset, - int YBufferSzShift, - int noCols, - int start_band, - int stop_band, - int tran_off, - int addPrevSamples) -{ +static void extractTransientCandidates( + FIXP_DBL **RESTRICT Energies, INT *RESTRICT scaleEnergies, + FIXP_DBL *RESTRICT thresholds, FIXP_DBL *RESTRICT transients, + int YBufferWriteOffset, int YBufferSzShift, int noCols, int start_band, + int stop_band, int tran_off, int addPrevSamples) { FIXP_DBL i_thres; - C_ALLOC_SCRATCH_START(EnergiesTemp, FIXP_DBL, 2*QMF_MAX_TIME_SLOTS); - FIXP_DBL *RESTRICT pEnergiesTemp = EnergiesTemp; + C_ALLOC_SCRATCH_START(EnergiesTemp, FIXP_DBL, 2 * 32) int tmpScaleEnergies0, tmpScaleEnergies1; int endCond; - int startEnerg,endEnerg; - int i,j,jIndex,jpBM; + int startEnerg, endEnerg; + int i, j, jIndex, jpBM; tmpScaleEnergies0 = scaleEnergies[0]; tmpScaleEnergies1 = scaleEnergies[1]; @@ -571,237 +561,227 @@ extractTransientCandidates(FIXP_DBL **RESTRICT Energies, FDK_ASSERT((tmpScaleEnergies0 >= 0) && (tmpScaleEnergies1 >= 0)); /* Keep addPrevSamples extra previous transient candidates. */ - FDKmemmove(transients, transients + noCols - addPrevSamples, (tran_off+addPrevSamples) * sizeof (FIXP_DBL)); - FDKmemclear(transients + tran_off + addPrevSamples, noCols * sizeof (FIXP_DBL)); + FDKmemmove(transients, transients + noCols - addPrevSamples, + (tran_off + addPrevSamples) * sizeof(FIXP_DBL)); + FDKmemclear(transients + tran_off + addPrevSamples, + noCols * sizeof(FIXP_DBL)); endCond = noCols; /* Amount of new transient values to be calculated. */ - startEnerg = (tran_off-3)>>YBufferSzShift; /* >>YBufferSzShift because of amount of energy values. -3 because of neighbors being watched. */ - endEnerg = ((noCols+ (YBufferWriteOffset<<YBufferSzShift))-1)>>YBufferSzShift; /* YBufferSzShift shifts because of half energy values. */ - - /* Compute differential values with two different weightings in every subband */ - for (i=start_band; i<stop_band; i++) - { + startEnerg = (tran_off - 3) >> YBufferSzShift; /* >>YBufferSzShift because of + amount of energy values. -3 + because of neighbors being + watched. */ + endEnerg = + ((noCols + (YBufferWriteOffset << YBufferSzShift)) - 1) >> + YBufferSzShift; /* YBufferSzShift shifts because of half energy values. */ + + /* Compute differential values with two different weightings in every subband + */ + for (i = start_band; i < stop_band; i++) { FIXP_DBL thres = thresholds[i]; - if((LONG)thresholds[i]>=256) - i_thres = (LONG)( (LONG)MAXVAL_DBL / ((((LONG)thresholds[i]))+1) )<<(32-24); + if ((LONG)thresholds[i] >= 256) + i_thres = (LONG)((LONG)MAXVAL_DBL / ((((LONG)thresholds[i])) + 1)) + << (32 - 24); else i_thres = (LONG)MAXVAL_DBL; /* Copy one timeslot and de-scale and de-squish */ if (YBufferSzShift == 1) { - for(j=startEnerg; j<YBufferWriteOffset; j++) { + for (j = startEnerg; j < YBufferWriteOffset; j++) { FIXP_DBL tmp = Energies[j][i]; - EnergiesTemp[(j<<1)+1] = EnergiesTemp[j<<1] = tmp>>tmpScaleEnergies0; + EnergiesTemp[(j << 1) + 1] = EnergiesTemp[j << 1] = + tmp >> tmpScaleEnergies0; } - for(; j<=endEnerg; j++) { + for (; j <= endEnerg; j++) { FIXP_DBL tmp = Energies[j][i]; - EnergiesTemp[(j<<1)+1] = EnergiesTemp[j<<1] = tmp>>tmpScaleEnergies1; + EnergiesTemp[(j << 1) + 1] = EnergiesTemp[j << 1] = + tmp >> tmpScaleEnergies1; } } else { - for(j=startEnerg; j<YBufferWriteOffset; j++) { + for (j = startEnerg; j < YBufferWriteOffset; j++) { FIXP_DBL tmp = Energies[j][i]; - EnergiesTemp[j] = tmp>>tmpScaleEnergies0; + EnergiesTemp[j] = tmp >> tmpScaleEnergies0; } - for(; j<=endEnerg; j++) { + for (; j <= endEnerg; j++) { FIXP_DBL tmp = Energies[j][i]; - EnergiesTemp[j] = tmp>>tmpScaleEnergies1; + EnergiesTemp[j] = tmp >> tmpScaleEnergies1; } } /* Detect peaks in energy values. */ jIndex = tran_off; - jpBM = jIndex+addPrevSamples; - - for (j=endCond; j--; jIndex++, jpBM++) - { + jpBM = jIndex + addPrevSamples; + for (j = endCond; j--; jIndex++, jpBM++) { FIXP_DBL delta, tran; int d; delta = (FIXP_DBL)0; - tran = (FIXP_DBL)0; + tran = (FIXP_DBL)0; - for (d=1; d<4; d++) { - delta += pEnergiesTemp[jIndex+d]; /* R */ - delta -= pEnergiesTemp[jIndex-d]; /* L */ + for (d = 1; d < 4; d++) { + delta += EnergiesTemp[jIndex + d]; /* R */ + delta -= EnergiesTemp[jIndex - d]; /* L */ delta -= thres; - if ( delta > (FIXP_DBL)0 ) { - tran += fMult(i_thres, delta); + if (delta > (FIXP_DBL)0) { + tran = fMultAddDiv2(tran, i_thres, delta); } } - transients[jpBM] += tran; + transients[jpBM] += (tran << 1); } } - C_ALLOC_SCRATCH_END(EnergiesTemp, FIXP_DBL, 2*QMF_MAX_TIME_SLOTS); + C_ALLOC_SCRATCH_END(EnergiesTemp, FIXP_DBL, 2 * 32) } -void -FDKsbrEnc_transientDetect(HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTran, - FIXP_DBL **Energies, - INT *scaleEnergies, - UCHAR *transient_info, - int YBufferWriteOffset, - int YBufferSzShift, - int timeStep, - int frameMiddleBorder) -{ +void FDKsbrEnc_transientDetect(HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTran, + FIXP_DBL **Energies, INT *scaleEnergies, + UCHAR *transient_info, int YBufferWriteOffset, + int YBufferSzShift, int timeStep, + int frameMiddleBorder) { int no_cols = h_sbrTran->no_cols; int qmfStartSample; int addPrevSamples; - int timeStepShift=0; + int timeStepShift = 0; int i, cond; /* Where to start looking for transients in the transient candidate buffer */ qmfStartSample = timeStep * frameMiddleBorder; - /* We need to look one value backwards in the transients, so we might need one more previous value. */ - addPrevSamples = (qmfStartSample > 0) ? 0: 1; + /* We need to look one value backwards in the transients, so we might need one + * more previous value. */ + addPrevSamples = (qmfStartSample > 0) ? 0 : 1; switch (timeStep) { - case 1: timeStepShift = 0; break; - case 2: timeStepShift = 1; break; - case 4: timeStepShift = 2; break; + case 1: + timeStepShift = 0; + break; + case 2: + timeStepShift = 1; + break; + case 4: + timeStepShift = 2; + break; } - calculateThresholds(Energies, - scaleEnergies, - h_sbrTran->thresholds, - YBufferWriteOffset, - YBufferSzShift, - h_sbrTran->no_cols, - h_sbrTran->no_rows, - h_sbrTran->tran_off); - - extractTransientCandidates(Energies, - scaleEnergies, - h_sbrTran->thresholds, - h_sbrTran->transients, - YBufferWriteOffset, - YBufferSzShift, - h_sbrTran->no_cols, - 0, - h_sbrTran->no_rows, - h_sbrTran->tran_off, - addPrevSamples ); + calculateThresholds(Energies, scaleEnergies, h_sbrTran->thresholds, + YBufferWriteOffset, YBufferSzShift, h_sbrTran->no_cols, + h_sbrTran->no_rows, h_sbrTran->tran_off); + + extractTransientCandidates( + Energies, scaleEnergies, h_sbrTran->thresholds, h_sbrTran->transients, + YBufferWriteOffset, YBufferSzShift, h_sbrTran->no_cols, 0, + h_sbrTran->no_rows, h_sbrTran->tran_off, addPrevSamples); transient_info[0] = 0; transient_info[1] = 0; transient_info[2] = 0; - /* Offset by the amount of additional previous transient candidates being kept. */ + /* Offset by the amount of additional previous transient candidates being + * kept. */ qmfStartSample += addPrevSamples; - /* Check for transients in second granule (pick the last value of subsequent values) */ - for (i=qmfStartSample; i<qmfStartSample + no_cols; i++) { - cond = (h_sbrTran->transients[i] < fMult(FL2FXCONST_DBL(0.9f), h_sbrTran->transients[i - 1]) ) - && (h_sbrTran->transients[i - 1] > h_sbrTran->tran_thr); + /* Check for transients in second granule (pick the last value of subsequent + * values) */ + for (i = qmfStartSample; i < qmfStartSample + no_cols; i++) { + cond = (h_sbrTran->transients[i] < + fMult(FL2FXCONST_DBL(0.9f), h_sbrTran->transients[i - 1])) && + (h_sbrTran->transients[i - 1] > h_sbrTran->tran_thr); if (cond) { - transient_info[0] = (i - qmfStartSample)>>timeStepShift; + transient_info[0] = (i - qmfStartSample) >> timeStepShift; transient_info[1] = 1; break; } } - if ( h_sbrTran->frameShift != 0) { - /* transient prediction for LDSBR */ - /* Check for transients in first <frameShift> qmf-slots of second frame */ - for (i=qmfStartSample+no_cols; i<qmfStartSample + no_cols+h_sbrTran->frameShift; i++) { - - cond = (h_sbrTran->transients[i] < fMult(FL2FXCONST_DBL(0.9f), h_sbrTran->transients[i - 1]) ) - && (h_sbrTran->transients[i - 1] > h_sbrTran->tran_thr); - - if (cond) { - int pos = (int) ( (i - qmfStartSample-no_cols) >> timeStepShift ); - if ((pos < 3) && (transient_info[1]==0)) { - transient_info[2] = 1; - } - break; + if (h_sbrTran->frameShift != 0) { + /* transient prediction for LDSBR */ + /* Check for transients in first <frameShift> qmf-slots of second frame */ + for (i = qmfStartSample + no_cols; + i < qmfStartSample + no_cols + h_sbrTran->frameShift; i++) { + cond = (h_sbrTran->transients[i] < + fMult(FL2FXCONST_DBL(0.9f), h_sbrTran->transients[i - 1])) && + (h_sbrTran->transients[i - 1] > h_sbrTran->tran_thr); + + if (cond) { + int pos = (int)((i - qmfStartSample - no_cols) >> timeStepShift); + if ((pos < 3) && (transient_info[1] == 0)) { + transient_info[2] = 1; } + break; } + } } } -int -FDKsbrEnc_InitSbrTransientDetector(HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector, - UINT sbrSyntaxFlags, /* SBR syntax flags derived from AOT. */ - INT frameSize, - INT sampleFreq, - sbrConfigurationPtr params, - int tran_fc, - int no_cols, - int no_rows, - int YBufferWriteOffset, - int YBufferSzShift, - int frameShift, - int tran_off) -{ - INT totalBitrate = params->codecSettings.standardBitrate * params->codecSettings.nChannels; - INT codecBitrate = params->codecSettings.bitRate; - FIXP_DBL bitrateFactor_m, framedur_fix; - INT bitrateFactor_e, tmp_e; - - FDKmemclear(h_sbrTransientDetector,sizeof(SBR_TRANSIENT_DETECTOR)); - - h_sbrTransientDetector->frameShift = frameShift; - h_sbrTransientDetector->tran_off = tran_off; - - if(codecBitrate) { - bitrateFactor_m = fDivNorm((FIXP_DBL)totalBitrate, (FIXP_DBL)(codecBitrate<<2),&bitrateFactor_e); - bitrateFactor_e += 2; - } - else { - bitrateFactor_m = FL2FXCONST_DBL(1.0/4.0); - bitrateFactor_e = 2; - } +int FDKsbrEnc_InitSbrTransientDetector( + HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector, + UINT sbrSyntaxFlags, /* SBR syntax flags derived from AOT. */ + INT frameSize, INT sampleFreq, sbrConfigurationPtr params, int tran_fc, + int no_cols, int no_rows, int YBufferWriteOffset, int YBufferSzShift, + int frameShift, int tran_off) { + INT totalBitrate = + params->codecSettings.standardBitrate * params->codecSettings.nChannels; + INT codecBitrate = params->codecSettings.bitRate; + FIXP_DBL bitrateFactor_m, framedur_fix; + INT bitrateFactor_e, tmp_e; + + FDKmemclear(h_sbrTransientDetector, sizeof(SBR_TRANSIENT_DETECTOR)); + + h_sbrTransientDetector->frameShift = frameShift; + h_sbrTransientDetector->tran_off = tran_off; + + if (codecBitrate) { + bitrateFactor_m = fDivNorm((FIXP_DBL)totalBitrate, + (FIXP_DBL)(codecBitrate << 2), &bitrateFactor_e); + bitrateFactor_e += 2; + } else { + bitrateFactor_m = FL2FXCONST_DBL(1.0 / 4.0); + bitrateFactor_e = 2; + } - framedur_fix = fDivNorm(frameSize, sampleFreq); + framedur_fix = fDivNorm(frameSize, sampleFreq); - /* The longer the frames, the more often should the FIXFIX- - case transmit 2 envelopes instead of 1. - Frame durations below 10 ms produce the highest threshold - so that practically always only 1 env is transmitted. */ - FIXP_DBL tmp = framedur_fix - FL2FXCONST_DBL(0.010); + /* The longer the frames, the more often should the FIXFIX- + case transmit 2 envelopes instead of 1. + Frame durations below 10 ms produce the highest threshold + so that practically always only 1 env is transmitted. */ + FIXP_DBL tmp = framedur_fix - FL2FXCONST_DBL(0.010); - tmp = fixMax(tmp, FL2FXCONST_DBL(0.0001)); - tmp = fDivNorm(FL2FXCONST_DBL(0.000075), fPow2(tmp), &tmp_e); + tmp = fixMax(tmp, FL2FXCONST_DBL(0.0001)); + tmp = fDivNorm(FL2FXCONST_DBL(0.000075), fPow2(tmp), &tmp_e); - bitrateFactor_e = (tmp_e + bitrateFactor_e); + bitrateFactor_e = (tmp_e + bitrateFactor_e); - if(sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { + if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) { bitrateFactor_e--; /* divide by 2 */ } - FDK_ASSERT(no_cols <= QMF_MAX_TIME_SLOTS); - FDK_ASSERT(no_rows <= QMF_CHANNELS); + FDK_ASSERT(no_cols <= 32); + FDK_ASSERT(no_rows <= 64); - h_sbrTransientDetector->no_cols = no_cols; - h_sbrTransientDetector->tran_thr = (FIXP_DBL)((params->tran_thr << (32-24-1)) / no_rows); - h_sbrTransientDetector->tran_fc = tran_fc; - h_sbrTransientDetector->split_thr_m = fMult(tmp, bitrateFactor_m); - h_sbrTransientDetector->split_thr_e = bitrateFactor_e; - h_sbrTransientDetector->no_rows = no_rows; - h_sbrTransientDetector->mode = params->tran_det_mode; - h_sbrTransientDetector->prevLowBandEnergy = FL2FXCONST_DBL(0.0f); + h_sbrTransientDetector->no_cols = no_cols; + h_sbrTransientDetector->tran_thr = + (FIXP_DBL)((params->tran_thr << (32 - 24 - 1)) / no_rows); + h_sbrTransientDetector->tran_fc = tran_fc; + h_sbrTransientDetector->split_thr_m = fMult(tmp, bitrateFactor_m); + h_sbrTransientDetector->split_thr_e = bitrateFactor_e; + h_sbrTransientDetector->no_rows = no_rows; + h_sbrTransientDetector->mode = params->tran_det_mode; + h_sbrTransientDetector->prevLowBandEnergy = FL2FXCONST_DBL(0.0f); - return (0); + return (0); } - #define ENERGY_SCALING_SIZE 32 INT FDKsbrEnc_InitSbrFastTransientDetector( - HANDLE_FAST_TRAN_DET h_sbrFastTransientDetector, - const INT time_slots_per_frame, - const INT bandwidth_qmf_slot, - const INT no_qmf_channels, - const INT sbr_qmf_1st_band - ) -{ - - int i, e; + HANDLE_FAST_TRAN_DET h_sbrFastTransientDetector, + const INT time_slots_per_frame, const INT bandwidth_qmf_slot, + const INT no_qmf_channels, const INT sbr_qmf_1st_band) { + int i; int buff_size; FIXP_DBL myExp; FIXP_DBL myExpSlot; @@ -809,9 +789,10 @@ INT FDKsbrEnc_InitSbrFastTransientDetector( h_sbrFastTransientDetector->lookahead = TRAN_DET_LOOKAHEAD; h_sbrFastTransientDetector->nTimeSlots = time_slots_per_frame; - buff_size = h_sbrFastTransientDetector->nTimeSlots + h_sbrFastTransientDetector->lookahead; + buff_size = h_sbrFastTransientDetector->nTimeSlots + + h_sbrFastTransientDetector->lookahead; - for(i=0; i< buff_size; i++) { + for (i = 0; i < buff_size; i++) { h_sbrFastTransientDetector->delta_energy[i] = FL2FXCONST_DBL(0.0f); h_sbrFastTransientDetector->energy_timeSlots[i] = FL2FXCONST_DBL(0.0f); h_sbrFastTransientDetector->lowpass_energy[i] = FL2FXCONST_DBL(0.0f); @@ -819,77 +800,92 @@ INT FDKsbrEnc_InitSbrFastTransientDetector( } FDK_ASSERT(bandwidth_qmf_slot > 0.f); - h_sbrFastTransientDetector->stopBand = fMin(TRAN_DET_STOP_FREQ/bandwidth_qmf_slot, no_qmf_channels); - h_sbrFastTransientDetector->startBand = fMin(sbr_qmf_1st_band, h_sbrFastTransientDetector->stopBand - TRAN_DET_MIN_QMFBANDS); + h_sbrFastTransientDetector->stopBand = + fMin(TRAN_DET_STOP_FREQ / bandwidth_qmf_slot, no_qmf_channels); + h_sbrFastTransientDetector->startBand = + fMin(sbr_qmf_1st_band, + h_sbrFastTransientDetector->stopBand - TRAN_DET_MIN_QMFBANDS); FDK_ASSERT(h_sbrFastTransientDetector->startBand < no_qmf_channels); - FDK_ASSERT(h_sbrFastTransientDetector->startBand < h_sbrFastTransientDetector->stopBand); + FDK_ASSERT(h_sbrFastTransientDetector->startBand < + h_sbrFastTransientDetector->stopBand); FDK_ASSERT(h_sbrFastTransientDetector->startBand > 1); FDK_ASSERT(h_sbrFastTransientDetector->stopBand > 1); /* the energy weighting and adding up has a headroom of 6 Bits, so up to 64 bands can be added without potential overflow. */ - FDK_ASSERT(h_sbrFastTransientDetector->stopBand - h_sbrFastTransientDetector->startBand <= 64); - - /* QMF_HP_dB_SLOPE_FIX says that we want a 20 dB per 16 kHz HP filter. - The following lines map this to the QMF bandwidth. */ - #define EXP_E 7 /* QMF_CHANNELS (=64) multiplications max, max. allowed sum is 0.5 */ - myExp = fMultNorm(QMF_HP_dBd_SLOPE_FIX, (FIXP_DBL)bandwidth_qmf_slot, &e); - myExp = scaleValueSaturate(myExp, e+0+DFRACT_BITS-1-EXP_E); + FDK_ASSERT(h_sbrFastTransientDetector->stopBand - + h_sbrFastTransientDetector->startBand <= + 64); + +/* QMF_HP_dB_SLOPE_FIX says that we want a 20 dB per 16 kHz HP filter. + The following lines map this to the QMF bandwidth. */ +#define EXP_E 7 /* 64 (=64) multiplications max, max. allowed sum is 0.5 */ + myExp = fMultNorm(QMF_HP_dBd_SLOPE_FIX, 0, (FIXP_DBL)bandwidth_qmf_slot, + DFRACT_BITS - 1, EXP_E); myExpSlot = myExp; - for(i=0; i<QMF_CHANNELS; i++){ + for (i = 0; i < 64; i++) { /* Calculate dBf over all qmf bands: dBf = (10^(0.002266f/10*bw(slot)))^(band) = = 2^(log2(10)*0.002266f/10*bw(slot)*band) = = 2^(0.00075275f*bw(slot)*band) */ - FIXP_DBL dBf_m; /* dBf mantissa */ - INT dBf_e; /* dBf exponent */ + FIXP_DBL dBf_m; /* dBf mantissa */ + INT dBf_e; /* dBf exponent */ INT tmp; - INT dBf_int; /* dBf integer part */ - FIXP_DBL dBf_fract; /* dBf fractional part */ + INT dBf_int; /* dBf integer part */ + FIXP_DBL dBf_fract; /* dBf fractional part */ /* myExp*(i+1) = myExp_int - myExp_fract myExp*(i+1) is split up here for better accuracy of CalcInvLdData(), for its result can be split up into an integer and a fractional part */ /* Round up to next integer */ - FIXP_DBL myExp_int = (myExpSlot & (FIXP_DBL)0xfe000000) + (FIXP_DBL)0x02000000; + FIXP_DBL myExp_int = + (myExpSlot & (FIXP_DBL)0xfe000000) + (FIXP_DBL)0x02000000; /* This is the fractional part that needs to be substracted */ FIXP_DBL myExp_fract = myExp_int - myExpSlot; /* Calc integer part */ - dBf_int = CalcInvLdData(myExp_int); - /* The result needs to be re-scaled. The ld(myExp_int) had been scaled by EXP_E, - the CalcInvLdData expects the operand to be scaled by LD_DATA_SHIFT. - Therefore, the correctly scaled result is dBf_int^(2^(EXP_E-LD_DATA_SHIFT)), - which is dBf_int^2 */ - dBf_int *= dBf_int; - - /* Calc fractional part */ - dBf_fract = CalcInvLdData(-myExp_fract); - /* The result needs to be re-scaled. The ld(myExp_fract) had been scaled by EXP_E, - the CalcInvLdData expects the operand to be scaled by LD_DATA_SHIFT. - Therefore, the correctly scaled result is dBf_fract^(2^(EXP_E-LD_DATA_SHIFT)), - which is dBf_fract^2 */ - dBf_fract = fMultNorm(dBf_fract, dBf_fract, &tmp); - - /* Get worst case scaling of multiplication result */ - dBf_e = (DFRACT_BITS-1 - tmp) - CountLeadingBits(dBf_int); - - /* Now multiply integer with fractional part of the result, thus resulting - in the overall accurate fractional result */ - dBf_m = fMultNorm(dBf_int, dBf_fract, &e); - dBf_m = scaleValueSaturate(dBf_m, e+DFRACT_BITS-1+tmp-dBf_e); - myExpSlot += myExp; + dBf_int = CalcInvLdData(myExp_int); + /* The result needs to be re-scaled. The ld(myExp_int) had been scaled by + EXP_E, the CalcInvLdData expects the operand to be scaled by + LD_DATA_SHIFT. Therefore, the correctly scaled result is + dBf_int^(2^(EXP_E-LD_DATA_SHIFT)), which is dBf_int^2 */ + + if (dBf_int <= + 46340) { /* compare with maximum allowed value for signed integer + multiplication, 46340 = + (INT)floor(sqrt((double)(((UINT)1<<(DFRACT_BITS-1))-1))) */ + dBf_int *= dBf_int; + + /* Calc fractional part */ + dBf_fract = CalcInvLdData(-myExp_fract); + /* The result needs to be re-scaled. The ld(myExp_fract) had been scaled + by EXP_E, the CalcInvLdData expects the operand to be scaled by + LD_DATA_SHIFT. Therefore, the correctly scaled result is + dBf_fract^(2^(EXP_E-LD_DATA_SHIFT)), which is dBf_fract^2 */ + dBf_fract = fMultNorm(dBf_fract, dBf_fract, &tmp); + + /* Get worst case scaling of multiplication result */ + dBf_e = (DFRACT_BITS - 1 - tmp) - CountLeadingBits(dBf_int); + + /* Now multiply integer with fractional part of the result, thus resulting + in the overall accurate fractional result */ + dBf_m = fMultNorm(dBf_int, DFRACT_BITS - 1, dBf_fract, tmp, dBf_e); + + myExpSlot += myExp; + } else { + dBf_m = (FIXP_DBL)0; + dBf_e = 0; + } /* Keep the results */ h_sbrFastTransientDetector->dBf_m[i] = dBf_m; h_sbrFastTransientDetector->dBf_e[i] = dBf_e; - } /* Make sure that dBf is greater than 1.0 (because it should be a highpass) */ @@ -899,84 +895,91 @@ INT FDKsbrEnc_InitSbrFastTransientDetector( } void FDKsbrEnc_fastTransientDetect( - const HANDLE_FAST_TRAN_DET h_sbrFastTransientDetector, - const FIXP_DBL *const *Energies, - const int *const scaleEnergies, - const INT YBufferWriteOffset, - UCHAR *const tran_vector - ) -{ + const HANDLE_FAST_TRAN_DET h_sbrFastTransientDetector, + const FIXP_DBL *const *Energies, const int *const scaleEnergies, + const INT YBufferWriteOffset, UCHAR *const tran_vector) { int timeSlot, band; - FIXP_DBL max_delta_energy; /* helper to store maximum energy ratio */ - int max_delta_energy_scale; /* helper to store scale of maximum energy ratio */ - int ind_max = 0; /* helper to store index of maximum energy ratio */ - int isTransientInFrame = 0; + FIXP_DBL max_delta_energy; /* helper to store maximum energy ratio */ + int max_delta_energy_scale; /* helper to store scale of maximum energy ratio + */ + int ind_max = 0; /* helper to store index of maximum energy ratio */ + int isTransientInFrame = 0; - const int nTimeSlots = h_sbrFastTransientDetector->nTimeSlots; - const int lookahead = h_sbrFastTransientDetector->lookahead; - const int startBand = h_sbrFastTransientDetector->startBand; - const int stopBand = h_sbrFastTransientDetector->stopBand; + const int nTimeSlots = h_sbrFastTransientDetector->nTimeSlots; + const int lookahead = h_sbrFastTransientDetector->lookahead; + const int startBand = h_sbrFastTransientDetector->startBand; + const int stopBand = h_sbrFastTransientDetector->stopBand; - int * transientCandidates = h_sbrFastTransientDetector->transientCandidates; + int *transientCandidates = h_sbrFastTransientDetector->transientCandidates; - FIXP_DBL * energy_timeSlots = h_sbrFastTransientDetector->energy_timeSlots; - int * energy_timeSlots_scale = h_sbrFastTransientDetector->energy_timeSlots_scale; + FIXP_DBL *energy_timeSlots = h_sbrFastTransientDetector->energy_timeSlots; + int *energy_timeSlots_scale = + h_sbrFastTransientDetector->energy_timeSlots_scale; - FIXP_DBL * delta_energy = h_sbrFastTransientDetector->delta_energy; - int * delta_energy_scale = h_sbrFastTransientDetector->delta_energy_scale; + FIXP_DBL *delta_energy = h_sbrFastTransientDetector->delta_energy; + int *delta_energy_scale = h_sbrFastTransientDetector->delta_energy_scale; - const FIXP_DBL thr = TRAN_DET_THRSHLD; - const INT thr_scale = TRAN_DET_THRSHLD_SCALE; + const FIXP_DBL thr = TRAN_DET_THRSHLD; + const INT thr_scale = TRAN_DET_THRSHLD_SCALE; /*reset transient info*/ tran_vector[2] = 0; /* reset transient candidates */ - FDKmemclear(transientCandidates+lookahead, nTimeSlots*sizeof(int)); + FDKmemclear(transientCandidates + lookahead, nTimeSlots * sizeof(int)); - for(timeSlot = lookahead; timeSlot < nTimeSlots + lookahead; timeSlot++) { + for (timeSlot = lookahead; timeSlot < nTimeSlots + lookahead; timeSlot++) { int i, norm; - FIXP_DBL tmpE = FL2FXCONST_DBL(0.0f); - int headroomEnSlot = DFRACT_BITS-1; + FIXP_DBL tmpE = FL2FXCONST_DBL(0.0f); + int headroomEnSlot = DFRACT_BITS - 1; FIXP_DBL smallNRG = FL2FXCONST_DBL(1e-2f); FIXP_DBL denominator; INT denominator_scale; /* determine minimum headroom of energy values for this timeslot */ - for(band = startBand; band < stopBand; band++) { - int tmp_headroom = fNormz(Energies[timeSlot][band])-1; - if(tmp_headroom < headroomEnSlot){ + for (band = startBand; band < stopBand; band++) { + int tmp_headroom = fNormz(Energies[timeSlot][band]) - 1; + if (tmp_headroom < headroomEnSlot) { headroomEnSlot = tmp_headroom; } } - for(i = 0, band = startBand; band < stopBand; band++, i++) { + for (i = 0, band = startBand; band < stopBand; band++, i++) { /* energy is weighted by weightingfactor stored in dBf_m array */ /* dBf_m index runs from 0 to stopBand-startband */ /* energy shifted by calculated headroom for maximum precision */ - FIXP_DBL weightedEnergy = fMult(Energies[timeSlot][band]<<headroomEnSlot, h_sbrFastTransientDetector->dBf_m[i]); + FIXP_DBL weightedEnergy = + fMult(Energies[timeSlot][band] << headroomEnSlot, + h_sbrFastTransientDetector->dBf_m[i]); /* energy is added up */ /* shift by 6 to have a headroom for maximum 64 additions */ /* shift by dBf_e to handle weighting factor dependent scale factors */ - tmpE += weightedEnergy >> (6 + (10 - h_sbrFastTransientDetector->dBf_e[i])); + tmpE += + weightedEnergy >> (6 + (10 - h_sbrFastTransientDetector->dBf_e[i])); } /* store calculated energy for timeslot */ energy_timeSlots[timeSlot] = tmpE; - /* calculate overall scale factor for energy of this timeslot */ - /* = original scale factor of energies (-scaleEnergies[0]+2*QMF_SCALE_OFFSET or -scaleEnergies[1]+2*QMF_SCALE_OFFSET */ - /* depending on YBufferWriteOffset) */ - /* + weighting factor scale (10) */ - /* + adding up scale factor ( 6) */ - /* - headroom of energy value (headroomEnSlot) */ - if(timeSlot < YBufferWriteOffset){ - energy_timeSlots_scale[timeSlot] = (-scaleEnergies[0]+2*QMF_SCALE_OFFSET) + (10+6) - headroomEnSlot; + /* calculate overall scale factor for energy of this timeslot */ + /* = original scale factor of energies + * (-scaleEnergies[0]+2*QMF_SCALE_OFFSET or + * -scaleEnergies[1]+2*QMF_SCALE_OFFSET */ + /* depending on YBufferWriteOffset) */ + /* + weighting factor scale (10) */ + /* + adding up scale factor ( 6) */ + /* - headroom of energy value (headroomEnSlot) */ + if (timeSlot < YBufferWriteOffset) { + energy_timeSlots_scale[timeSlot] = + (-scaleEnergies[0] + 2 * QMF_SCALE_OFFSET) + (10 + 6) - + headroomEnSlot; } else { - energy_timeSlots_scale[timeSlot] = (-scaleEnergies[1]+2*QMF_SCALE_OFFSET) + (10+6) - headroomEnSlot; + energy_timeSlots_scale[timeSlot] = + (-scaleEnergies[1] + 2 * QMF_SCALE_OFFSET) + (10 + 6) - + headroomEnSlot; } /* Add a small energy to the denominator, thus making the transient @@ -984,19 +987,21 @@ void FDKsbrEnc_fastTransientDetect( silent ones not. */ /* make sure that smallNRG does not overflow */ - if ( -energy_timeSlots_scale[timeSlot-1] + 1 > 5 ) - { + if (-energy_timeSlots_scale[timeSlot - 1] + 1 > 5) { denominator = smallNRG; denominator_scale = 0; } else { /* Leave an additional headroom of 1 bit for this addition. */ - smallNRG = scaleValue(smallNRG, -(energy_timeSlots_scale[timeSlot-1] + 1)); - denominator = (energy_timeSlots[timeSlot-1]>>1) + smallNRG; - denominator_scale = energy_timeSlots_scale[timeSlot-1]+1; + smallNRG = + scaleValue(smallNRG, -(energy_timeSlots_scale[timeSlot - 1] + 1)); + denominator = (energy_timeSlots[timeSlot - 1] >> 1) + smallNRG; + denominator_scale = energy_timeSlots_scale[timeSlot - 1] + 1; } - delta_energy[timeSlot] = fDivNorm(energy_timeSlots[timeSlot], denominator, &norm); - delta_energy_scale[timeSlot] = energy_timeSlots_scale[timeSlot] - denominator_scale + norm; + delta_energy[timeSlot] = + fDivNorm(energy_timeSlots[timeSlot], denominator, &norm); + delta_energy_scale[timeSlot] = + energy_timeSlots_scale[timeSlot] - denominator_scale + norm; } /*get transient candidates*/ @@ -1008,15 +1013,21 @@ void FDKsbrEnc_fastTransientDetect( last or the one before the last slot, it is marked as a transient.*/ FDK_ASSERT(lookahead >= 2); - for(timeSlot = lookahead; timeSlot < nTimeSlots + lookahead; timeSlot++) { - FIXP_DBL energy_cur_slot_weighted = fMult(energy_timeSlots[timeSlot],FL2FXCONST_DBL(1.0f/1.4f)); - if( !fIsLessThan(delta_energy[timeSlot], delta_energy_scale[timeSlot], thr, thr_scale) && - ( ((transientCandidates[timeSlot-2]==0) && (transientCandidates[timeSlot-1]==0)) || - !fIsLessThan(energy_cur_slot_weighted, energy_timeSlots_scale[timeSlot], energy_timeSlots[timeSlot-1], energy_timeSlots_scale[timeSlot-1] ) || - !fIsLessThan(energy_cur_slot_weighted, energy_timeSlots_scale[timeSlot], energy_timeSlots[timeSlot-2], energy_timeSlots_scale[timeSlot-2] ) - ) - ) -{ + for (timeSlot = lookahead; timeSlot < nTimeSlots + lookahead; timeSlot++) { + FIXP_DBL energy_cur_slot_weighted = + fMult(energy_timeSlots[timeSlot], FL2FXCONST_DBL(1.0f / 1.4f)); + if (!fIsLessThan(delta_energy[timeSlot], delta_energy_scale[timeSlot], thr, + thr_scale) && + (((transientCandidates[timeSlot - 2] == 0) && + (transientCandidates[timeSlot - 1] == 0)) || + !fIsLessThan(energy_cur_slot_weighted, + energy_timeSlots_scale[timeSlot], + energy_timeSlots[timeSlot - 1], + energy_timeSlots_scale[timeSlot - 1]) || + !fIsLessThan(energy_cur_slot_weighted, + energy_timeSlots_scale[timeSlot], + energy_timeSlots[timeSlot - 2], + energy_timeSlots_scale[timeSlot - 2]))) { /* in case of strong transients, subsequent * qmf slots might be recognized as transients. */ transientCandidates[timeSlot] = 1; @@ -1024,22 +1035,24 @@ void FDKsbrEnc_fastTransientDetect( } /*get transient with max energy*/ - max_delta_energy = FL2FXCONST_DBL(0.0f); + max_delta_energy = FL2FXCONST_DBL(0.0f); max_delta_energy_scale = 0; ind_max = 0; isTransientInFrame = 0; - for(timeSlot = 0; timeSlot < nTimeSlots; timeSlot++) { + for (timeSlot = 0; timeSlot < nTimeSlots; timeSlot++) { int scale = fMax(delta_energy_scale[timeSlot], max_delta_energy_scale); - if(transientCandidates[timeSlot] && ( (delta_energy[timeSlot] >> (scale - delta_energy_scale[timeSlot])) > (max_delta_energy >> (scale - max_delta_energy_scale)) ) ) { - max_delta_energy = delta_energy[timeSlot]; + if (transientCandidates[timeSlot] && + ((delta_energy[timeSlot] >> (scale - delta_energy_scale[timeSlot])) > + (max_delta_energy >> (scale - max_delta_energy_scale)))) { + max_delta_energy = delta_energy[timeSlot]; max_delta_energy_scale = scale; - ind_max = timeSlot; + ind_max = timeSlot; isTransientInFrame = 1; } } /*from all transient candidates take the one with the biggest energy*/ - if(isTransientInFrame) { + if (isTransientInFrame) { tran_vector[0] = ind_max; tran_vector[1] = 1; } else { @@ -1048,22 +1061,22 @@ void FDKsbrEnc_fastTransientDetect( } /*check for transients in lookahead*/ - for(timeSlot = nTimeSlots; timeSlot < nTimeSlots + lookahead; timeSlot++) { - if(transientCandidates[timeSlot]) { + for (timeSlot = nTimeSlots; timeSlot < nTimeSlots + lookahead; timeSlot++) { + if (transientCandidates[timeSlot]) { tran_vector[2] = 1; } } /*update buffers*/ - for(timeSlot = 0; timeSlot < lookahead; timeSlot++) { + for (timeSlot = 0; timeSlot < lookahead; timeSlot++) { transientCandidates[timeSlot] = transientCandidates[nTimeSlots + timeSlot]; /* fixpoint stuff */ - energy_timeSlots[timeSlot] = energy_timeSlots[nTimeSlots + timeSlot]; - energy_timeSlots_scale[timeSlot] = energy_timeSlots_scale[nTimeSlots + timeSlot]; + energy_timeSlots[timeSlot] = energy_timeSlots[nTimeSlots + timeSlot]; + energy_timeSlots_scale[timeSlot] = + energy_timeSlots_scale[nTimeSlots + timeSlot]; - delta_energy[timeSlot] = delta_energy[nTimeSlots + timeSlot]; - delta_energy_scale[timeSlot] = delta_energy_scale[nTimeSlots + timeSlot]; + delta_energy[timeSlot] = delta_energy[nTimeSlots + timeSlot]; + delta_energy_scale[timeSlot] = delta_energy_scale[nTimeSlots + timeSlot]; } } - diff --git a/libSBRenc/src/tran_det.h b/libSBRenc/src/tran_det.h index 6fe1023..d10a7db 100644 --- a/libSBRenc/src/tran_det.h +++ b/libSBRenc/src/tran_det.h @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© 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. +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: +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 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 +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. +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. +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." +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. +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. +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. +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 @@ -79,125 +90,102 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ /*! \file - \brief Transient detector prototypes + \brief Transient detector prototypes $Revision: 95111 $ */ -#ifndef __TRAN_DET_H -#define __TRAN_DET_H +#ifndef TRAN_DET_H +#define TRAN_DET_H #include "sbr_encoder.h" #include "sbr_def.h" -typedef struct -{ - FIXP_DBL transients[QMF_MAX_TIME_SLOTS+(QMF_MAX_TIME_SLOTS/2)]; - FIXP_DBL thresholds[QMF_CHANNELS]; - FIXP_DBL tran_thr; /* Master threshold for transient signals */ - FIXP_DBL split_thr_m; /* Threshold for splitting FIXFIX-frames into 2 env */ - INT split_thr_e; /* Scale for splitting threshold */ - FIXP_DBL prevLowBandEnergy; /* Energy of low band */ - FIXP_DBL prevHighBandEnergy; /* Energy of high band */ - INT tran_fc; /* Number of lowband subbands to discard */ - INT no_cols; - INT no_rows; - INT mode; - - int frameShift; - int tran_off; /* Offset for reading energy values. */ -} -SBR_TRANSIENT_DETECTOR; - +typedef struct { + FIXP_DBL transients[32 + (32 / 2)]; + FIXP_DBL thresholds[64]; + FIXP_DBL tran_thr; /* Master threshold for transient signals */ + FIXP_DBL split_thr_m; /* Threshold for splitting FIXFIX-frames into 2 env */ + INT split_thr_e; /* Scale for splitting threshold */ + FIXP_DBL prevLowBandEnergy; /* Energy of low band */ + FIXP_DBL prevHighBandEnergy; /* Energy of high band */ + INT tran_fc; /* Number of lowband subbands to discard */ + INT no_cols; + INT no_rows; + INT mode; + + int frameShift; + int tran_off; /* Offset for reading energy values. */ +} SBR_TRANSIENT_DETECTOR; typedef SBR_TRANSIENT_DETECTOR *HANDLE_SBR_TRANSIENT_DETECTOR; #define TRAN_DET_LOOKAHEAD 2 -#define TRAN_DET_START_FREQ 4500 /*start frequency for transient detection*/ -#define TRAN_DET_STOP_FREQ 13500 /*stop frequency for transient detection*/ -#define TRAN_DET_MIN_QMFBANDS 4 /* minimum qmf bands for transient detection */ -#define QMF_HP_dBd_SLOPE_FIX FL2FXCONST_DBL(0.00075275f) /* 0.002266f/10 * log2(10) */ -#define TRAN_DET_THRSHLD FL2FXCONST_DBL(3.2f/4.f) -#define TRAN_DET_THRSHLD_SCALE (2) - -typedef struct -{ - INT transientCandidates[QMF_MAX_TIME_SLOTS + TRAN_DET_LOOKAHEAD]; +#define TRAN_DET_START_FREQ 4500 /*start frequency for transient detection*/ +#define TRAN_DET_STOP_FREQ 13500 /*stop frequency for transient detection*/ +#define TRAN_DET_MIN_QMFBANDS \ + 4 /* minimum qmf bands for transient detection \ + */ +#define QMF_HP_dBd_SLOPE_FIX \ + FL2FXCONST_DBL(0.00075275f) /* 0.002266f/10 * log2(10) */ +#define TRAN_DET_THRSHLD FL2FXCONST_DBL(5.0f / 8.0f) +#define TRAN_DET_THRSHLD_SCALE (3) + +typedef struct { + INT transientCandidates[32 + TRAN_DET_LOOKAHEAD]; INT nTimeSlots; INT lookahead; INT startBand; INT stopBand; - FIXP_DBL dBf_m[QMF_CHANNELS]; - INT dBf_e[QMF_CHANNELS]; + FIXP_DBL dBf_m[64]; + INT dBf_e[64]; - FIXP_DBL energy_timeSlots[QMF_MAX_TIME_SLOTS + TRAN_DET_LOOKAHEAD]; - INT energy_timeSlots_scale[QMF_MAX_TIME_SLOTS + TRAN_DET_LOOKAHEAD]; + FIXP_DBL energy_timeSlots[32 + TRAN_DET_LOOKAHEAD]; + INT energy_timeSlots_scale[32 + TRAN_DET_LOOKAHEAD]; - FIXP_DBL delta_energy[QMF_MAX_TIME_SLOTS + TRAN_DET_LOOKAHEAD]; - INT delta_energy_scale[QMF_MAX_TIME_SLOTS + TRAN_DET_LOOKAHEAD]; + FIXP_DBL delta_energy[32 + TRAN_DET_LOOKAHEAD]; + INT delta_energy_scale[32 + TRAN_DET_LOOKAHEAD]; - FIXP_DBL lowpass_energy[QMF_MAX_TIME_SLOTS + TRAN_DET_LOOKAHEAD]; - INT lowpass_energy_scale[QMF_MAX_TIME_SLOTS + TRAN_DET_LOOKAHEAD]; -#if defined (FTD_LOG) - FDKFILE *ftd_log; -#endif -} -FAST_TRAN_DETECTOR; + FIXP_DBL lowpass_energy[32 + TRAN_DET_LOOKAHEAD]; + INT lowpass_energy_scale[32 + TRAN_DET_LOOKAHEAD]; +} FAST_TRAN_DETECTOR; typedef FAST_TRAN_DETECTOR *HANDLE_FAST_TRAN_DET; - INT FDKsbrEnc_InitSbrFastTransientDetector( - HANDLE_FAST_TRAN_DET h_sbrFastTransientDetector, - const INT time_slots_per_frame, - const INT bandwidth_qmf_slot, - const INT no_qmf_channels, - const INT sbr_qmf_1st_band - ); + HANDLE_FAST_TRAN_DET h_sbrFastTransientDetector, + const INT time_slots_per_frame, const INT bandwidth_qmf_slot, + const INT no_qmf_channels, const INT sbr_qmf_1st_band); void FDKsbrEnc_fastTransientDetect( - const HANDLE_FAST_TRAN_DET h_sbrFastTransientDetector, - const FIXP_DBL *const *Energies, - const int *const scaleEnergies, - const INT YBufferWriteOffset, - UCHAR *const tran_vector - ); - -void -FDKsbrEnc_transientDetect(HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector, - FIXP_DBL **Energies, - INT *scaleEnergies, - UCHAR *tran_vector, - int YBufferWriteOffset, - int YBufferSzShift, - int timeStep, - int frameMiddleBorder); - -int -FDKsbrEnc_InitSbrTransientDetector (HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector, - UINT sbrSyntaxFlags, /* SBR syntax flags derived from AOT. */ - INT frameSize, - INT sampleFreq, - sbrConfigurationPtr params, - int tran_fc, - int no_cols, - int no_rows, - int YBufferWriteOffset, - int YBufferSzShift, - int frameShift, - int tran_off); - -void -FDKsbrEnc_frameSplitter(FIXP_DBL **Energies, - INT *scaleEnergies, - HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector, - UCHAR *freqBandTable, - UCHAR *tran_vector, - int YBufferWriteOffset, - int YBufferSzShift, - int nSfb, - int timeStep, - int no_cols, - FIXP_DBL* tonality); + const HANDLE_FAST_TRAN_DET h_sbrFastTransientDetector, + const FIXP_DBL *const *Energies, const int *const scaleEnergies, + const INT YBufferWriteOffset, UCHAR *const tran_vector); + +void FDKsbrEnc_transientDetect( + HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector, FIXP_DBL **Energies, + INT *scaleEnergies, UCHAR *tran_vector, int YBufferWriteOffset, + int YBufferSzShift, int timeStep, int frameMiddleBorder); + +int FDKsbrEnc_InitSbrTransientDetector( + HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector, + UINT sbrSyntaxFlags, /* SBR syntax flags derived from AOT. */ + INT frameSize, INT sampleFreq, sbrConfigurationPtr params, int tran_fc, + int no_cols, int no_rows, int YBufferWriteOffset, int YBufferSzShift, + int frameShift, int tran_off); + +void FDKsbrEnc_frameSplitter( + FIXP_DBL **Energies, INT *scaleEnergies, + HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientDetector, UCHAR *freqBandTable, + UCHAR *tran_vector, int YBufferWriteOffset, int YBufferSzShift, int nSfb, + int timeStep, int no_cols, FIXP_DBL *tonality); #endif |