diff options
author | The Android Open Source Project <initial-contribution@android.com> | 2012-07-11 10:15:24 -0700 |
---|---|---|
committer | The Android Open Source Project <initial-contribution@android.com> | 2012-07-11 10:15:24 -0700 |
commit | 2228e360595641dd906bf1773307f43d304f5b2e (patch) | |
tree | 57f3d390ebb0782cc0de0fb984c8ea7e45b4f386 /libAACenc/src/psy_configuration.cpp | |
download | fdk-aac-dabplus-2228e360595641dd906bf1773307f43d304f5b2e.tar.gz fdk-aac-dabplus-2228e360595641dd906bf1773307f43d304f5b2e.tar.bz2 fdk-aac-dabplus-2228e360595641dd906bf1773307f43d304f5b2e.zip |
Snapshot 2bda038c163298531d47394bc2c09e1409c5d0db
Change-Id: If584e579464f28b97d50e51fc76ba654a5536c54
Diffstat (limited to 'libAACenc/src/psy_configuration.cpp')
-rw-r--r-- | libAACenc/src/psy_configuration.cpp | 656 |
1 files changed, 656 insertions, 0 deletions
diff --git a/libAACenc/src/psy_configuration.cpp b/libAACenc/src/psy_configuration.cpp new file mode 100644 index 0000000..96f6a71 --- /dev/null +++ b/libAACenc/src/psy_configuration.cpp @@ -0,0 +1,656 @@ + +/* ----------------------------------------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2012 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 +----------------------------------------------------------------------------------------------------------- */ + +/******************************** MPEG Audio Encoder ************************** + + Initial author: M.Werner + contents/description: Psychoaccoustic configuration + +******************************************************************************/ + +#include "psy_configuration.h" +#include "adj_thr.h" +#include "aacEnc_rom.h" + +#include "genericStds.h" + +#include "FDK_trigFcts.h" + +typedef struct{ + LONG sampleRate; + const SFB_PARAM_LONG *paramLong; + const SFB_PARAM_SHORT *paramShort; +}SFB_INFO_TAB; + + +static const SFB_INFO_TAB sfbInfoTab[] = { + {8000, &p_FDKaacEnc_8000_long_1024, &p_FDKaacEnc_8000_short_128}, + {11025, &p_FDKaacEnc_11025_long_1024, &p_FDKaacEnc_11025_short_128}, + {12000, &p_FDKaacEnc_12000_long_1024, &p_FDKaacEnc_12000_short_128}, + {16000, &p_FDKaacEnc_16000_long_1024, &p_FDKaacEnc_16000_short_128}, + {22050, &p_FDKaacEnc_22050_long_1024, &p_FDKaacEnc_22050_short_128}, + {24000, &p_FDKaacEnc_24000_long_1024, &p_FDKaacEnc_24000_short_128}, + {32000, &p_FDKaacEnc_32000_long_1024, &p_FDKaacEnc_32000_short_128}, + {44100, &p_FDKaacEnc_44100_long_1024, &p_FDKaacEnc_44100_short_128}, + {48000, &p_FDKaacEnc_48000_long_1024, &p_FDKaacEnc_48000_short_128}, + {64000, &p_FDKaacEnc_64000_long_1024, &p_FDKaacEnc_64000_short_128}, + {88200, &p_FDKaacEnc_88200_long_1024, &p_FDKaacEnc_88200_short_128}, + {96000, &p_FDKaacEnc_96000_long_1024, &p_FDKaacEnc_96000_short_128} + +}; + +/* 22050 and 24000 Hz */ +static const SFB_PARAM_LONG p_22050_long_512 = { + 31, + { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, + 4, 8, 8, 8, 12, 12, 12, 16, 20, 24, + 28, 32, 32, 32, 32, 32, 32, 32, 32, 32, + 32} +}; + +/* 32000 Hz */ +static const SFB_PARAM_LONG p_32000_long_512 = { + 37, + { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, + 4, 4, 4, 4, 8, 8, 8, 8, 8, 12, + 12, 12, 12, 16, 16, 16, 20, 24, 24, 28, + 32, 32, 32, 32, 32, 32, 32} +}; + +/* 44100 Hz */ +static const SFB_PARAM_LONG p_44100_long_512 = { + 36, + {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, + 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, + 12, 12, 12, 12, 16, 20, 24, 28, 32, 32, + 32, 32, 32, 32, 32, 52} +}; + +static const SFB_INFO_TAB sfbInfoTabLD512[] = { + { 8000, &p_22050_long_512, NULL}, + {11025, &p_22050_long_512, NULL}, + {12000, &p_22050_long_512, NULL}, + {16000, &p_22050_long_512, NULL}, + {22050, &p_22050_long_512, NULL}, + {24000, &p_22050_long_512, NULL}, + {32000, &p_32000_long_512, NULL}, + {44100, &p_44100_long_512, NULL}, + {48000, &p_44100_long_512, NULL}, + {64000, &p_44100_long_512, NULL}, + {88200, &p_44100_long_512, NULL}, + {96000, &p_44100_long_512, NULL}, + +}; + + +/* 22050 and 24000 Hz */ +static const SFB_PARAM_LONG p_22050_long_480 = { + 30, + { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, + 4, 8, 8, 8, 12, 12, 12, 16, 20, 24, + 28, 32, 32, 32, 32, 32, 32, 32, 32, 32} +}; + +/* 32000 Hz */ +static const SFB_PARAM_LONG p_32000_long_480 = { + 37, + { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, + 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, + 8, 8, 12, 12, 12, 16, 16, 20, 24, 32, + 32, 32, 32, 32, 32, 32, 32} +}; + +/* 44100 Hz */ +static const SFB_PARAM_LONG p_44100_long_480 = { + 35, + { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, + 4, 4, 4, 4, 8, 8, 8, 8, 8, 12, + 12, 12, 12, 12, 16, 16, 24, 28, 32, 32, + 32, 32, 32, 32, 48} +}; + +static const SFB_INFO_TAB sfbInfoTabLD480[] = { + { 8000, &p_22050_long_480, NULL}, + {11025, &p_22050_long_480, NULL}, + {12000, &p_22050_long_480, NULL}, + {16000, &p_22050_long_480, NULL}, + {22050, &p_22050_long_480, NULL}, + {24000, &p_22050_long_480, NULL}, + {32000, &p_32000_long_480, NULL}, + {44100, &p_44100_long_480, NULL}, + {48000, &p_44100_long_480, NULL}, + {64000, &p_44100_long_480, NULL}, + {88200, &p_44100_long_480, NULL}, + {96000, &p_44100_long_480, NULL}, + +}; + +/* Fixed point precision definitions */ +#define Q_BARCVAL (25) + +static AAC_ENCODER_ERROR FDKaacEnc_initSfbTable(LONG sampleRate, INT blockType, INT granuleLength, INT *sfbOffset, INT *sfbCnt) +{ + INT i, specStartOffset = 0; + const UCHAR* sfbWidth = NULL; + const SFB_INFO_TAB *sfbInfo = NULL; + int size; + + /* + select table + */ + switch(granuleLength) { + case 1024: + case 960: + sfbInfo = sfbInfoTab; + size = (INT)(sizeof(sfbInfoTab)/sizeof(SFB_INFO_TAB)); + break; + case 512: + sfbInfo = sfbInfoTabLD512; + size = sizeof(sfbInfoTabLD512); + break; + case 480: + sfbInfo = sfbInfoTabLD480; + size = sizeof(sfbInfoTabLD480); + break; + default: + return AAC_ENC_INVALID_FRAME_LENGTH; + } + + for(i = 0; i < size; i++){ + if(sfbInfo[i].sampleRate == sampleRate){ + switch(blockType){ + case LONG_WINDOW: + case START_WINDOW: + case STOP_WINDOW: + sfbWidth = sfbInfo[i].paramLong->sfbWidth; + *sfbCnt = sfbInfo[i].paramLong->sfbCnt; + break; + case SHORT_WINDOW: + sfbWidth = sfbInfo[i].paramShort->sfbWidth; + *sfbCnt = sfbInfo[i].paramShort->sfbCnt; + granuleLength /= TRANS_FAC; + break; + } + break; + } + } + if (i == size) { + return AAC_ENC_UNSUPPORTED_SAMPLINGRATE; + } + + /* + calc sfb offsets + */ + for(i = 0; i < *sfbCnt; i++){ + sfbOffset[i] = specStartOffset; + specStartOffset += sfbWidth[i]; + if (specStartOffset >= granuleLength) { + i++; + break; + } + } + *sfbCnt = fixMin(i,*sfbCnt); + sfbOffset[*sfbCnt] = fixMin(specStartOffset,granuleLength); + + return AAC_ENC_OK; +} + + +/***************************************************************************** + + functionname: FDKaacEnc_BarcLineValue + description: Calculates barc value for one frequency line + returns: barc value of line + input: number of lines in transform, index of line to check, Fs + output: + +*****************************************************************************/ +static FIXP_DBL FDKaacEnc_BarcLineValue(INT noOfLines, INT fftLine, LONG samplingFreq) +{ + + FIXP_DBL FOURBY3EM4 = (FIXP_DBL)0x45e7b273; /* 4.0/3 * 0.0001 in q43 */ + FIXP_DBL PZZZ76 = (FIXP_DBL)0x639d5e4a; /* 0.00076 in q41 */ + FIXP_DBL ONE3P3 = (FIXP_DBL)0x35333333; /* 13.3 in q26 */ + FIXP_DBL THREEP5 = (FIXP_DBL)0x1c000000; /* 3.5 in q27 */ + FIXP_DBL INV480 = (FIXP_DBL)0x44444444; // 1/480 in q39 + + FIXP_DBL center_freq, x1, x2; + FIXP_DBL bvalFFTLine, atan1, atan2; + + /* Theoritical maximum of center_freq (samp_freq*0.5) is 96khz * 0.5 = 48000 */ + /* Theoritical maximum of x1 is 1.3333333e-4f * center_freq = 6.4, can keep in q28 */ + /* Theoritical maximum of x2 is 0.00076f * center_freq = 36.48, can keep in q25 */ + + center_freq = fftLine * samplingFreq; /* q11 or q8 */ + + switch (noOfLines) { + case 1024: + center_freq = center_freq << 2; /* q13 */ + break; + case 128: + center_freq = center_freq << 5; /* q13 */ + break; + case 512: + center_freq = (fftLine * samplingFreq) << 3; // q13 + break; + case 480: + center_freq = fMult(center_freq, INV480) << 4; // q13 + break; + default: + center_freq = (FIXP_DBL)0; + } + + x1 = fMult(center_freq, FOURBY3EM4); /* q13 * q43 - (DFRACT_BITS-1) = q25 */ + x2 = fMult(center_freq, PZZZ76) << 2; /* q13 * q41 - (DFRACT_BITS-1) + 2 = q25 */ + + atan1 = fixp_atan(x1); + atan2 = fixp_atan(x2); + + /* q25 (q26 * q30 - (DFRACT_BITS-1)) + q25 (q27 * q30 * q30) */ + bvalFFTLine = fMult(ONE3P3, atan2) + fMult(THREEP5, fMult(atan1, atan1)); + return(bvalFFTLine); + +} + +/* + do not consider energies below a certain input signal level, + i.e. of -96dB or 1 bit at 16 bit PCM resolution, + might need to be configurable to e.g. 24 bit PCM Input or a lower + resolution for low bit rates +*/ +static void FDKaacEnc_InitMinPCMResolution(int numPb, + int *pbOffset, + FIXP_DBL *sfbPCMquantThreshold) +{ + /* PCM_QUANT_NOISE = FDKpow(10.0f, - 20.f / 10.0f) * ABS_LOW * NORM_PCM_ENERGY * FDKpow(2,PCM_QUANT_THR_SCALE) */ + #define PCM_QUANT_NOISE ((FIXP_DBL)0x00547062) + + for( int i = 0; i < numPb; i++ ) { + sfbPCMquantThreshold[i] = (pbOffset[i+1] - pbOffset[i]) * PCM_QUANT_NOISE; + } +} + +static FIXP_DBL getMaskFactor( + const FIXP_DBL dbVal_fix, + const INT dbVal_e, + const FIXP_DBL ten_fix, + const INT ten_e + ) +{ + INT q_msk; + FIXP_DBL mask_factor; + + mask_factor = fPow(ten_fix, DFRACT_BITS-1-ten_e, -dbVal_fix, DFRACT_BITS-1-dbVal_e, &q_msk); + q_msk = fixMin(DFRACT_BITS-1,fixMax(-(DFRACT_BITS-1),q_msk)); + + if ( (q_msk>0) && (mask_factor>(FIXP_DBL)MAXVAL_DBL>>q_msk) ) { + mask_factor = (FIXP_DBL)MAXVAL_DBL; + } + else { + mask_factor = scaleValue(mask_factor, q_msk); + } + + return (mask_factor); +} + +static void FDKaacEnc_initSpreading(INT numPb, + FIXP_DBL *pbBarcValue, + FIXP_DBL *pbMaskLoFactor, + FIXP_DBL *pbMaskHiFactor, + FIXP_DBL *pbMaskLoFactorSprEn, + FIXP_DBL *pbMaskHiFactorSprEn, + const LONG bitrate, + const INT blockType) + +{ + INT i; + FIXP_DBL MASKLOWSPREN, MASKHIGHSPREN; + + FIXP_DBL MASKHIGH = (FIXP_DBL)0x30000000; /* 1.5 in q29 */ + FIXP_DBL MASKLOW = (FIXP_DBL)0x60000000; /* 3.0 in q29 */ + FIXP_DBL MASKLOWSPRENLONG = (FIXP_DBL)0x60000000; /* 3.0 in q29 */ + FIXP_DBL MASKHIGHSPRENLONG = (FIXP_DBL)0x40000000; /* 2.0 in q29 */ + FIXP_DBL MASKHIGHSPRENLONGLOWBR = (FIXP_DBL)0x30000000; /* 1.5 in q29 */ + FIXP_DBL MASKLOWSPRENSHORT = (FIXP_DBL)0x40000000; /* 2.0 in q29 */ + FIXP_DBL MASKHIGHSPRENSHORT = (FIXP_DBL)0x30000000; /* 1.5 in q29 */ + FIXP_DBL TEN = (FIXP_DBL)0x50000000; /* 10.0 in q27 */ + + if (blockType != SHORT_WINDOW) + { + MASKLOWSPREN = MASKLOWSPRENLONG; + MASKHIGHSPREN = (bitrate>20000)?MASKHIGHSPRENLONG:MASKHIGHSPRENLONGLOWBR; + } + else + { + MASKLOWSPREN = MASKLOWSPRENSHORT; + MASKHIGHSPREN = MASKHIGHSPRENSHORT; + } + + for(i=0; i<numPb; i++) + { + if (i > 0) + { + pbMaskHiFactor[i] = getMaskFactor( + fMult(MASKHIGH, (pbBarcValue[i] - pbBarcValue[i-1])), 23, + TEN, 27); + + pbMaskLoFactor[i-1] = getMaskFactor( + fMult(MASKLOW, (pbBarcValue[i] - pbBarcValue[i-1])), 23, + TEN, 27); + + pbMaskHiFactorSprEn[i] = getMaskFactor( + fMult(MASKHIGHSPREN, (pbBarcValue[i] - pbBarcValue[i-1])), 23, + TEN, 27); + + pbMaskLoFactorSprEn[i-1] = getMaskFactor( + fMult(MASKLOWSPREN, (pbBarcValue[i] - pbBarcValue[i-1])), 23, + TEN, 27); + } + else + { + pbMaskHiFactor[i] = (FIXP_DBL)0; + pbMaskLoFactor[numPb-1] = (FIXP_DBL)0; + pbMaskHiFactorSprEn[i] = (FIXP_DBL)0; + pbMaskLoFactorSprEn[numPb-1] = (FIXP_DBL)0; + } + } +} + +static void FDKaacEnc_initBarcValues(INT numPb, + INT *pbOffset, + INT numLines, + INT samplingFrequency, + FIXP_DBL *pbBval) +{ + INT i; + FIXP_DBL MAX_BARC = (FIXP_DBL)0x30000000; /* 24.0 in q25 */ + + for(i=0; i<numPb; i++) + { + FIXP_DBL v1, v2, cur_bark; + v1 = FDKaacEnc_BarcLineValue(numLines, pbOffset[i], samplingFrequency); + v2 = FDKaacEnc_BarcLineValue(numLines, pbOffset[i+1], samplingFrequency); + cur_bark = (v1 >> 1) + (v2 >> 1); + pbBval[i] = fixMin(cur_bark, MAX_BARC); + } +} + +static void FDKaacEnc_initMinSnr(const LONG bitrate, + const LONG samplerate, + const INT numLines, + const INT *sfbOffset, + const INT sfbActive, + const INT blockType, + FIXP_DBL *sfbMinSnrLdData) +{ + INT sfb; + + /* Fix conversion variables */ + INT qbfac, qperwin, qdiv, qpeprt_const, qpeprt; + INT qtmp, qsnr, sfbWidth; + + FIXP_DBL MAX_BARC = (FIXP_DBL)0x30000000; /* 24.0 in q25 */ + FIXP_DBL MAX_BARCP1 = (FIXP_DBL)0x32000000; /* 25.0 in q25 */ + FIXP_DBL BITS2PEFAC = (FIXP_DBL)0x4b851eb8; /* 1.18 in q30 */ + FIXP_DBL PERS2P4 = (FIXP_DBL)0x624dd2f2; /* 0.024 in q36 */ + FIXP_DBL ONEP5 = (FIXP_DBL)0x60000000; /* 1.5 in q30 */ + FIXP_DBL MAX_SNR = (FIXP_DBL)0x33333333; /* 0.8 in q30 */ + FIXP_DBL MIN_SNR = (FIXP_DBL)0x003126e9; /* 0.003 in q30 */ + + FIXP_DBL barcFactor, pePerWindow, pePart, barcWidth; + FIXP_DBL pePart_const, tmp, snr, one_qsnr, one_point5; + + /* relative number of active barks */ + barcFactor = fDivNorm(fixMin(FDKaacEnc_BarcLineValue(numLines, sfbOffset[sfbActive], samplerate), MAX_BARC), + MAX_BARCP1, &qbfac); + + qbfac = DFRACT_BITS-1-qbfac; + + pePerWindow = fDivNorm(bitrate, samplerate, &qperwin); + qperwin = DFRACT_BITS-1-qperwin; + pePerWindow = fMult(pePerWindow, BITS2PEFAC); qperwin = qperwin + 30 - (DFRACT_BITS-1); + pePerWindow = fMult(pePerWindow, PERS2P4); qperwin = qperwin + 36 - (DFRACT_BITS-1); + + switch (numLines) { + case 1024: + qperwin = qperwin - 10; + break; + case 128: + qperwin = qperwin - 7; + break; + case 512: + qperwin = qperwin - 9; + break; + case 480: + qperwin = qperwin - 9; + pePerWindow = fMult(pePerWindow, FL2FXCONST_DBL(480.f/512.f)); + break; + } + + /* for short blocks it is assumed that more bits are available */ + if (blockType == SHORT_WINDOW) + { + pePerWindow = fMult(pePerWindow, ONEP5); + qperwin = qperwin + 30 - (DFRACT_BITS-1); + } + pePart_const = fDivNorm(pePerWindow, barcFactor, &qdiv); qpeprt_const = qperwin - qbfac + DFRACT_BITS-1-qdiv; + + for (sfb = 0; sfb < sfbActive; sfb++) + { + barcWidth = FDKaacEnc_BarcLineValue(numLines, sfbOffset[sfb+1], samplerate) - + FDKaacEnc_BarcLineValue(numLines, sfbOffset[sfb], samplerate); + + /* adapt to sfb bands */ + pePart = fMult(pePart_const, barcWidth); qpeprt = qpeprt_const + 25 - (DFRACT_BITS-1); + + /* pe -> snr calculation */ + sfbWidth = (sfbOffset[sfb+1] - sfbOffset[sfb]); + pePart = fDivNorm(pePart, sfbWidth, &qdiv); qpeprt += DFRACT_BITS-1-qdiv; + + tmp = f2Pow(pePart, DFRACT_BITS-1-qpeprt, &qtmp); + qtmp = DFRACT_BITS-1-qtmp; + + /* Subtract 1.5 */ + qsnr = fixMin(qtmp, 30); + tmp = tmp >> (qtmp - qsnr); + + if((30+1-qsnr) > (DFRACT_BITS-1)) + one_point5 = (FIXP_DBL)0; + else + one_point5 = (FIXP_DBL)(ONEP5 >> (30+1-qsnr)); + + snr = (tmp>>1) - (one_point5); qsnr -= 1; + + /* max(snr, 1.0) */ + if(qsnr > 0) + one_qsnr = (FIXP_DBL)(1 << qsnr); + else + one_qsnr = (FIXP_DBL)0; + + snr = fixMax(one_qsnr, snr); + + /* 1/snr */ + snr = fDivNorm(one_qsnr, snr, &qsnr); + qsnr = DFRACT_BITS-1-qsnr; + snr = (qsnr > 30)? (snr>>(qsnr-30)):snr; + + /* upper limit is -1 dB */ + snr = (snr > MAX_SNR) ? MAX_SNR : snr; + + /* lower limit is -25 dB */ + snr = (snr < MIN_SNR) ? MIN_SNR : snr; + snr = snr << 1; + + sfbMinSnrLdData[sfb] = CalcLdData(snr); + } +} + +AAC_ENCODER_ERROR FDKaacEnc_InitPsyConfiguration(INT bitrate, + INT samplerate, + INT bandwidth, + INT blocktype, + INT granuleLength, + INT useIS, + PSY_CONFIGURATION *psyConf, + FB_TYPE filterbank) +{ + AAC_ENCODER_ERROR ErrorStatus; + INT sfb; + FIXP_DBL sfbBarcVal[MAX_SFB]; + const INT frameLengthLong = granuleLength; + const INT frameLengthShort = granuleLength/TRANS_FAC; + + FDKmemclear(psyConf, sizeof(PSY_CONFIGURATION)); + psyConf->granuleLength = granuleLength; + psyConf->filterbank = filterbank; + + psyConf->allowIS = (useIS) && ( (bitrate/bandwidth) < 5 ); + + /* init sfb table */ + ErrorStatus = FDKaacEnc_initSfbTable(samplerate,blocktype,granuleLength,psyConf->sfbOffset,&psyConf->sfbCnt); + if (ErrorStatus != AAC_ENC_OK) + return ErrorStatus; + + /* calculate barc values for each pb */ + FDKaacEnc_initBarcValues(psyConf->sfbCnt, + psyConf->sfbOffset, + psyConf->sfbOffset[psyConf->sfbCnt], + samplerate, + sfbBarcVal); + + FDKaacEnc_InitMinPCMResolution(psyConf->sfbCnt, + psyConf->sfbOffset, + psyConf->sfbPcmQuantThreshold); + + /* calculate spreading function */ + FDKaacEnc_initSpreading(psyConf->sfbCnt, + sfbBarcVal, + psyConf->sfbMaskLowFactor, + psyConf->sfbMaskHighFactor, + psyConf->sfbMaskLowFactorSprEn, + psyConf->sfbMaskHighFactorSprEn, + bitrate, + blocktype); + + /* init ratio */ + + psyConf->maxAllowedIncreaseFactor = 2; /* integer */ + psyConf->minRemainingThresholdFactor = (FIXP_SGL)0x0148; /* FL2FXCONST_SGL(0.01f); */ /* fract */ + + psyConf->clipEnergy = (FIXP_DBL)0x773593ff; /* FL2FXCONST_DBL(1.0e9*NORM_PCM_ENERGY); */ + + if (blocktype!=SHORT_WINDOW) { + psyConf->lowpassLine = (INT)((2*bandwidth*frameLengthLong)/samplerate); + psyConf->lowpassLineLFE = LFE_LOWPASS_LINE; + } + else { + psyConf->lowpassLine = (INT)((2*bandwidth*frameLengthShort)/samplerate); + psyConf->lowpassLineLFE = 0; /* LFE only in lonf blocks */ + /* psyConf->clipEnergy /= (TRANS_FAC * TRANS_FAC); */ + psyConf->clipEnergy >>= 6; + } + + for (sfb = 0; sfb < psyConf->sfbCnt; sfb++){ + if (psyConf->sfbOffset[sfb] >= psyConf->lowpassLine) + break; + } + psyConf->sfbActive = sfb; + + for (sfb = 0; sfb < psyConf->sfbCnt; sfb++){ + if (psyConf->sfbOffset[sfb] >= psyConf->lowpassLineLFE) + break; + } + psyConf->sfbActiveLFE = sfb; + + /* calculate minSnr */ + FDKaacEnc_initMinSnr(bitrate, + samplerate, + psyConf->sfbOffset[psyConf->sfbCnt], + psyConf->sfbOffset, + psyConf->sfbActive, + blocktype, + psyConf->sfbMinSnrLdData); + + return AAC_ENC_OK; +} + |