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author | Matthias P. Braendli <matthias.braendli@mpb.li> | 2019-11-11 11:38:02 +0100 |
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committer | Matthias P. Braendli <matthias.braendli@mpb.li> | 2019-11-11 11:38:02 +0100 |
commit | 0e5af65c467b2423a0b857ae3ad98c91acc1e190 (patch) | |
tree | d07f69550d8886271e44fe79c4dcfb299cafbd38 /fdk-aac/libAACenc/src/metadata_compressor.cpp | |
parent | efe406d9724f959c8bc2a31802559ca6d41fd897 (diff) | |
download | ODR-AudioEnc-0e5af65c467b2423a0b857ae3ad98c91acc1e190.tar.gz ODR-AudioEnc-0e5af65c467b2423a0b857ae3ad98c91acc1e190.tar.bz2 ODR-AudioEnc-0e5af65c467b2423a0b857ae3ad98c91acc1e190.zip |
Include patched FDK-AAC in the repository
The initial idea was to get the DAB+ patch into upstream, but since
that follows the android source releases, there is no place for a custom
DAB+ patch there.
So instead of having to maintain a patched fdk-aac that has to have the
same .so version as the distribution package on which it is installed,
we prefer having a separate fdk-aac-dab library to avoid collision.
At that point, there's no reason to keep fdk-aac in a separate
repository, as odr-audioenc is the only tool that needs DAB+ encoding
support. Including it here simplifies installation, and makes it
consistent with toolame-dab, also shipped in this repository.
DAB+ decoding support (needed by ODR-SourceCompanion, dablin, etisnoop,
welle.io and others) can be done using upstream FDK-AAC.
Diffstat (limited to 'fdk-aac/libAACenc/src/metadata_compressor.cpp')
-rw-r--r-- | fdk-aac/libAACenc/src/metadata_compressor.cpp | 1579 |
1 files changed, 1579 insertions, 0 deletions
diff --git a/fdk-aac/libAACenc/src/metadata_compressor.cpp b/fdk-aac/libAACenc/src/metadata_compressor.cpp new file mode 100644 index 0000000..bdac80a --- /dev/null +++ b/fdk-aac/libAACenc/src/metadata_compressor.cpp @@ -0,0 +1,1579 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Neusinger + + Description: Compressor for AAC Metadata Generator + +*******************************************************************************/ + +#include "metadata_compressor.h" +#include "channel_map.h" + +#define LOG2 0.69314718056f /* natural logarithm of 2 */ +#define ILOG2 1.442695041f /* 1/LOG2 */ +#define FIXP_ILOG2_DIV2 (FL2FXCONST_DBL(ILOG2 / 2)) + +/*----------------- defines ----------------------*/ + +#define MAX_DRC_CHANNELS (8) /*!< Max number of audio input channels. */ +#define DOWNMIX_SHIFT (3) /*!< Max 8 channel. */ +#define WEIGHTING_FILTER_SHIFT (2) /*!< Scaling used in weighting filter. */ + +#define METADATA_INT_BITS 10 +#define METADATA_LINT_BITS 20 +#define METADATA_INT_SCALE (INT64(1) << (METADATA_INT_BITS)) +#define METADATA_FRACT_BITS (DFRACT_BITS - 1 - METADATA_INT_BITS) +#define METADATA_FRACT_SCALE (INT64(1) << (METADATA_FRACT_BITS)) + +/** + * Enum for channel assignment. + */ +enum { L = 0, R = 1, C = 2, LFE = 3, LS = 4, RS = 5, S = 6, LS2 = 7, RS2 = 8 }; + +/*--------------- structure definitions --------------------*/ + +/** + * Structure holds weighting filter filter states. + */ +struct WEIGHTING_STATES { + FIXP_DBL x1; + FIXP_DBL x2; + FIXP_DBL y1; + FIXP_DBL y2; +}; + +/** + * Dynamic Range Control compressor structure. + */ +struct DRC_COMP { + FIXP_DBL maxBoostThr[2]; /*!< Max boost threshold. */ + FIXP_DBL boostThr[2]; /*!< Boost threshold. */ + FIXP_DBL earlyCutThr[2]; /*!< Early cut threshold. */ + FIXP_DBL cutThr[2]; /*!< Cut threshold. */ + FIXP_DBL maxCutThr[2]; /*!< Max cut threshold. */ + + FIXP_DBL boostFac[2]; /*!< Precalculated factor for boost compression. */ + FIXP_DBL + earlyCutFac[2]; /*!< Precalculated factor for early cut compression. */ + FIXP_DBL cutFac[2]; /*!< Precalculated factor for cut compression. */ + + FIXP_DBL maxBoost[2]; /*!< Maximum boost. */ + FIXP_DBL maxCut[2]; /*!< Maximum cut. */ + FIXP_DBL maxEarlyCut[2]; /*!< Maximum early cut. */ + + FIXP_DBL fastAttack[2]; /*!< Fast attack coefficient. */ + FIXP_DBL fastDecay[2]; /*!< Fast release coefficient. */ + FIXP_DBL slowAttack[2]; /*!< Slow attack coefficient. */ + FIXP_DBL slowDecay[2]; /*!< Slow release coefficient. */ + UINT holdOff[2]; /*!< Hold time in blocks. */ + + FIXP_DBL attackThr[2]; /*!< Slow/fast attack threshold. */ + FIXP_DBL decayThr[2]; /*!< Slow/fast release threshold. */ + + DRC_PROFILE profile[2]; /*!< DRC profile. */ + INT blockLength; /*!< Block length in samples. */ + UINT sampleRate; /*!< Sample rate. */ + CHANNEL_MODE chanConfig; /*!< Channel configuration. */ + + UCHAR useWeighting; /*!< Use weighting filter. */ + + UINT channels; /*!< Number of channels. */ + UINT fullChannels; /*!< Number of full range channels. */ + INT channelIdx[9]; /*!< Offsets of interleaved channel samples (L, R, C, LFE, + Ls, Rs, S, Ls2, Rs2). */ + + FIXP_DBL smoothLevel[2]; /*!< level smoothing states */ + FIXP_DBL smoothGain[2]; /*!< gain smoothing states */ + UINT holdCnt[2]; /*!< hold counter */ + + FIXP_DBL limGain[2]; /*!< limiter gain */ + FIXP_DBL limDecay; /*!< limiter decay (linear) */ + FIXP_DBL prevPeak[2]; /*!< max peak of previous block (stereo/mono)*/ + + WEIGHTING_STATES + filter[MAX_DRC_CHANNELS]; /*!< array holds weighting filter states */ +}; + +/*---------------- constants -----------------------*/ + +/** + * Profile tables. + */ +static const FIXP_DBL tabMaxBoostThr[] = { + (FIXP_DBL)(-(43 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(53 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(55 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(65 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(50 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(40 << METADATA_FRACT_BITS))}; +static const FIXP_DBL tabBoostThr[] = { + (FIXP_DBL)(-(31 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(41 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(31 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(41 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(31 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(31 << METADATA_FRACT_BITS))}; +static const FIXP_DBL tabEarlyCutThr[] = { + (FIXP_DBL)(-(26 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(21 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(26 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(21 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(26 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(20 << METADATA_FRACT_BITS))}; +static const FIXP_DBL tabCutThr[] = {(FIXP_DBL)(-(16 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(11 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(16 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(21 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(16 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(10 << METADATA_FRACT_BITS))}; +static const FIXP_DBL tabMaxCutThr[] = { + (FIXP_DBL)(4 << METADATA_FRACT_BITS), (FIXP_DBL)(9 << METADATA_FRACT_BITS), + (FIXP_DBL)(4 << METADATA_FRACT_BITS), (FIXP_DBL)(9 << METADATA_FRACT_BITS), + (FIXP_DBL)(4 << METADATA_FRACT_BITS), (FIXP_DBL)(4 << METADATA_FRACT_BITS)}; +static const FIXP_DBL tabBoostRatio[] = { + FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), + FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), + FL2FXCONST_DBL(((1.f / 5.f) - 1.f)), FL2FXCONST_DBL(((1.f / 5.f) - 1.f))}; +static const FIXP_DBL tabEarlyCutRatio[] = { + FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), + FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), FL2FXCONST_DBL(((1.f / 1.f) - 1.f)), + FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), FL2FXCONST_DBL(((1.f / 2.f) - 1.f))}; +static const FIXP_DBL tabCutRatio[] = { + FL2FXCONST_DBL(((1.f / 20.f) - 1.f)), FL2FXCONST_DBL(((1.f / 20.f) - 1.f)), + FL2FXCONST_DBL(((1.f / 20.f) - 1.f)), FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), + FL2FXCONST_DBL(((1.f / 20.f) - 1.f)), FL2FXCONST_DBL(((1.f / 20.f) - 1.f))}; +static const FIXP_DBL tabMaxBoost[] = {(FIXP_DBL)(6 << METADATA_FRACT_BITS), + (FIXP_DBL)(6 << METADATA_FRACT_BITS), + (FIXP_DBL)(12 << METADATA_FRACT_BITS), + (FIXP_DBL)(12 << METADATA_FRACT_BITS), + (FIXP_DBL)(15 << METADATA_FRACT_BITS), + (FIXP_DBL)(15 << METADATA_FRACT_BITS)}; +static const FIXP_DBL tabMaxCut[] = {(FIXP_DBL)(24 << METADATA_FRACT_BITS), + (FIXP_DBL)(24 << METADATA_FRACT_BITS), + (FIXP_DBL)(24 << METADATA_FRACT_BITS), + (FIXP_DBL)(15 << METADATA_FRACT_BITS), + (FIXP_DBL)(24 << METADATA_FRACT_BITS), + (FIXP_DBL)(24 << METADATA_FRACT_BITS)}; +static const FIXP_DBL tabFastAttack[] = { + FL2FXCONST_DBL((10.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((10.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((10.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((10.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((10.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((0.f / 1000.f) / METADATA_INT_SCALE)}; +static const FIXP_DBL tabFastDecay[] = { + FL2FXCONST_DBL((1000.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((1000.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((1000.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((1000.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((200.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((0.f / 1000.f) / METADATA_INT_SCALE)}; +static const FIXP_DBL tabSlowAttack[] = { + FL2FXCONST_DBL((100.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((100.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((100.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((100.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((100.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((0.f / 1000.f) / METADATA_INT_SCALE)}; +static const FIXP_DBL tabSlowDecay[] = { + FL2FXCONST_DBL((3000.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((3000.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((10000.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((3000.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((1000.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((0.f / 1000.f) / METADATA_INT_SCALE)}; + +static const INT tabHoldOff[] = {10, 10, 10, 10, 10, 0}; + +static const FIXP_DBL tabAttackThr[] = {(FIXP_DBL)(15 << METADATA_FRACT_BITS), + (FIXP_DBL)(15 << METADATA_FRACT_BITS), + (FIXP_DBL)(15 << METADATA_FRACT_BITS), + (FIXP_DBL)(15 << METADATA_FRACT_BITS), + (FIXP_DBL)(10 << METADATA_FRACT_BITS), + (FIXP_DBL)(0 << METADATA_FRACT_BITS)}; +static const FIXP_DBL tabDecayThr[] = {(FIXP_DBL)(20 << METADATA_FRACT_BITS), + (FIXP_DBL)(20 << METADATA_FRACT_BITS), + (FIXP_DBL)(20 << METADATA_FRACT_BITS), + (FIXP_DBL)(20 << METADATA_FRACT_BITS), + (FIXP_DBL)(10 << METADATA_FRACT_BITS), + (FIXP_DBL)(0 << METADATA_FRACT_BITS)}; + +/** + * Weighting filter coefficients (biquad bandpass). + */ +static const FIXP_DBL b0 = FL2FXCONST_DBL(0.53050662f); /* b1 = 0, b2 = -b0 */ +static const FIXP_DBL a1 = FL2FXCONST_DBL(-0.95237983f), + a2 = FL2FXCONST_DBL(-0.02248836f); /* a0 = 1 */ + +/*------------- function definitions ----------------*/ + +/** + * \brief Calculate scaling factor for denoted processing block. + * + * \param blockLength Length of processing block. + * + * \return shiftFactor + */ +static UINT getShiftFactor(const UINT length) { + UINT ldN; + for (ldN = 1; (((UINT)1) << ldN) < length; ldN++) + ; + + return ldN; +} + +/** + * \brief Sum up fixpoint values with best possible accuracy. + * + * \param value1 First input value. + * \param q1 Scaling factor of first input value. + * \param pValue2 Pointer to second input value, will be modified on + * return. + * \param pQ2 Pointer to second scaling factor, will be modified on + * return. + * + * \return void + */ +static void fixpAdd(const FIXP_DBL value1, const int q1, + FIXP_DBL* const pValue2, int* const pQ2) { + const int headroom1 = fNormz(fixp_abs(value1)) - 1; + const int headroom2 = fNormz(fixp_abs(*pValue2)) - 1; + int resultScale = fixMax(q1 - headroom1, (*pQ2) - headroom2); + + if ((value1 != FL2FXCONST_DBL(0.f)) && (*pValue2 != FL2FXCONST_DBL(0.f))) { + resultScale++; + } + + *pValue2 = scaleValue(value1, q1 - resultScale) + + scaleValue(*pValue2, (*pQ2) - resultScale); + *pQ2 = (*pValue2 != (FIXP_DBL)0) ? resultScale : DFRACT_BITS - 1; +} + +/** + * \brief Function for converting time constant to filter coefficient. + * + * \param t Time constant. + * \param sampleRate Sampling rate in Hz. + * \param blockLength Length of processing block in samples per channel. + * + * \return result = 1.0 - exp(-1.0/((t) * (f))) + */ +static FIXP_DBL tc2Coeff(const FIXP_DBL t, const INT sampleRate, + const INT blockLength) { + FIXP_DBL sampleRateFract; + FIXP_DBL blockLengthFract; + FIXP_DBL f, product; + FIXP_DBL exponent, result; + INT e_res; + + /* f = sampleRate/blockLength */ + sampleRateFract = + (FIXP_DBL)(sampleRate << (DFRACT_BITS - 1 - METADATA_LINT_BITS)); + blockLengthFract = + (FIXP_DBL)(blockLength << (DFRACT_BITS - 1 - METADATA_LINT_BITS)); + f = fDivNorm(sampleRateFract, blockLengthFract, &e_res); + f = scaleValue(f, e_res - METADATA_INT_BITS); /* convert to METADATA_FRACT */ + + /* product = t*f */ + product = fMultNorm(t, f, &e_res); + product = scaleValue( + product, e_res + METADATA_INT_BITS); /* convert to METADATA_FRACT */ + + /* exponent = (-1.0/((t) * (f))) */ + exponent = fDivNorm(METADATA_FRACT_SCALE, product, &e_res); + exponent = scaleValue( + exponent, e_res - METADATA_INT_BITS); /* convert to METADATA_FRACT */ + + /* exponent * ld(e) */ + exponent = fMult(exponent, FIXP_ILOG2_DIV2) << 1; /* e^(x) = 2^(x*ld(e)) */ + + /* exp(-1.0/((t) * (f))) */ + result = f2Pow(-exponent, DFRACT_BITS - 1 - METADATA_FRACT_BITS, &e_res); + + /* result = 1.0 - exp(-1.0/((t) * (f))) */ + result = (FIXP_DBL)MAXVAL_DBL - scaleValue(result, e_res); + + return result; +} + +static void findPeakLevels(HDRC_COMP drcComp, const INT_PCM* const inSamples, + const FIXP_DBL clev, const FIXP_DBL slev, + const FIXP_DBL ext_leva, const FIXP_DBL ext_levb, + const FIXP_DBL lfe_lev, const FIXP_DBL dmxGain5, + const FIXP_DBL dmxGain2, FIXP_DBL peak[2]) { + int i, c; + FIXP_DBL tmp = FL2FXCONST_DBL(0.f); + INT_PCM maxSample = 0; + + /* find peak level */ + peak[0] = peak[1] = FL2FXCONST_DBL(0.f); + for (i = 0; i < drcComp->blockLength; i++) { + const INT_PCM* pSamples = &inSamples[i * drcComp->channels]; + + /* single channels */ + for (c = 0; c < (int)drcComp->channels; c++) { + maxSample = fMax(maxSample, (INT_PCM)fAbs(pSamples[c])); + } + } + peak[0] = fixMax(peak[0], FX_PCM2FX_DBL(maxSample) >> DOWNMIX_SHIFT); + + /* 7.1/6.1 to 5.1 downmixes */ + if (drcComp->fullChannels > 5) { + for (i = 0; i < drcComp->blockLength; i++) { + const INT_PCM* pSamples = &inSamples[i * drcComp->channels]; + + /* channel 1 (L, Ls,...) */ + tmp = FL2FXCONST_DBL(0.f); + switch (drcComp->chanConfig) { + case MODE_6_1: + tmp += + fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[6]]) >> + (DOWNMIX_SHIFT - 1); /* Cs */ + break; + case MODE_7_1_BACK: + case MODE_7_1_REAR_SURROUND: + tmp += + fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> + (DOWNMIX_SHIFT - 1); /* Lrs / Lss */ + break; + case MODE_1_2_2_2_1: + case MODE_7_1_FRONT_CENTER: + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> + DOWNMIX_SHIFT); /* L */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> + (DOWNMIX_SHIFT - 1); /* Lc */ + break; + case MODE_7_1_TOP_FRONT: + tmp += + fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> + (DOWNMIX_SHIFT - 1); /* L */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> + (DOWNMIX_SHIFT - 1); /* Lvh */ + break; + default: + break; + } + peak[0] = fixMax(peak[0], fixp_abs(tmp)); + + /* channel 2 (R, Rs,...) */ + tmp = FL2FXCONST_DBL(0.f); + switch (drcComp->chanConfig) { + case MODE_6_1: + tmp += + fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[6]]) >> + (DOWNMIX_SHIFT - 1); /* Cs */ + break; + case MODE_7_1_BACK: + case MODE_7_1_REAR_SURROUND: + tmp += + fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> + (DOWNMIX_SHIFT - 1); /* Rrs / Rss */ + break; + case MODE_1_2_2_2_1: + case MODE_7_1_FRONT_CENTER: + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> + DOWNMIX_SHIFT); /* R */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> + (DOWNMIX_SHIFT - 1); /* Rc */ + break; + case MODE_7_1_TOP_FRONT: + tmp += + fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> + (DOWNMIX_SHIFT - 1); /* R */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> + (DOWNMIX_SHIFT - 1); /* Rvh */ + break; + default: + break; + } + peak[0] = fixMax(peak[0], fixp_abs(tmp)); + + /* channel 3 (C) */ + tmp = FL2FXCONST_DBL(0.f); + switch (drcComp->chanConfig) { + case MODE_1_2_2_2_1: + case MODE_7_1_FRONT_CENTER: + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> + DOWNMIX_SHIFT); /* C */ + tmp += + fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> + (DOWNMIX_SHIFT - 1); /* Lc */ + tmp += + fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> + (DOWNMIX_SHIFT - 1); /* Rc */ + break; + default: + break; + } + peak[0] = fixMax(peak[0], fixp_abs(tmp)); + + } /* for (blocklength) */ + + /* take downmix gain into accout */ + peak[0] = fMult(dmxGain5, peak[0]) + << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); + } + + /* 7.1 / 5.1 to stereo downmixes */ + if (drcComp->fullChannels > 2) { + /* Lt/Rt downmix */ + for (i = 0; i < drcComp->blockLength; i++) { + const INT_PCM* pSamples = &inSamples[i * drcComp->channels]; + + /* Lt */ + tmp = FL2FXCONST_DBL(0.f); + if (drcComp->channelIdx[LS] >= 0) + tmp -= fMultDiv2(FL2FXCONST_DBL(0.707f), + (FIXP_PCM)pSamples[drcComp->channelIdx[LS]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + if (drcComp->channelIdx[LS2] >= 0) + tmp -= fMultDiv2(FL2FXCONST_DBL(0.707f), + (FIXP_PCM)pSamples[drcComp->channelIdx[LS2]]) >> + (DOWNMIX_SHIFT - 1); /* Ls2 */ + if (drcComp->channelIdx[RS] >= 0) + tmp -= fMultDiv2(FL2FXCONST_DBL(0.707f), + (FIXP_PCM)pSamples[drcComp->channelIdx[RS]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + if (drcComp->channelIdx[RS2] >= 0) + tmp -= fMultDiv2(FL2FXCONST_DBL(0.707f), + (FIXP_PCM)pSamples[drcComp->channelIdx[RS2]]) >> + (DOWNMIX_SHIFT - 1); /* Rs2 */ + if ((drcComp->channelIdx[LS] >= 0) && (drcComp->channelIdx[LS2] >= 0)) + tmp = fMult(FL2FXCONST_DBL(0.707f), tmp); /* 7.1ch */ + if (drcComp->channelIdx[S] >= 0) + tmp -= fMultDiv2(FL2FXCONST_DBL(0.707f), + (FIXP_PCM)pSamples[drcComp->channelIdx[S]]) >> + (DOWNMIX_SHIFT - 1); /* S */ + if (drcComp->channelIdx[C] >= 0) + tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), + (FIXP_PCM)pSamples[drcComp->channelIdx[C]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[L]]) >> + DOWNMIX_SHIFT); /* L */ + + /* apply scaling of downmix gains */ + /* only for positive values only, as legacy decoders might not know this + * parameter */ + if (dmxGain2 > FL2FXCONST_DBL(0.f)) { + if (drcComp->fullChannels > 5) { + tmp = fMult(dmxGain5, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); + } + tmp = fMult(dmxGain2, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); + } + peak[0] = fixMax(peak[0], fixp_abs(tmp)); + + /* Rt */ + tmp = FL2FXCONST_DBL(0.f); + if (drcComp->channelIdx[LS] >= 0) + tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), + (FIXP_PCM)pSamples[drcComp->channelIdx[LS]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + if (drcComp->channelIdx[LS2] >= 0) + tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), + (FIXP_PCM)pSamples[drcComp->channelIdx[LS2]]) >> + (DOWNMIX_SHIFT - 1); /* Ls2 */ + if (drcComp->channelIdx[RS] >= 0) + tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), + (FIXP_PCM)pSamples[drcComp->channelIdx[RS]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + if (drcComp->channelIdx[RS2] >= 0) + tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), + (FIXP_PCM)pSamples[drcComp->channelIdx[RS2]]) >> + (DOWNMIX_SHIFT - 1); /* Rs2 */ + if ((drcComp->channelIdx[RS] >= 0) && (drcComp->channelIdx[RS2] >= 0)) + tmp = fMult(FL2FXCONST_DBL(0.707f), tmp); /* 7.1ch */ + if (drcComp->channelIdx[S] >= 0) + tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), + (FIXP_PCM)pSamples[drcComp->channelIdx[S]]) >> + (DOWNMIX_SHIFT - 1); /* S */ + if (drcComp->channelIdx[C] >= 0) + tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), + (FIXP_PCM)pSamples[drcComp->channelIdx[C]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[R]]) >> + DOWNMIX_SHIFT); /* R */ + + /* apply scaling of downmix gains */ + /* only for positive values only, as legacy decoders might not know this + * parameter */ + if (dmxGain2 > FL2FXCONST_DBL(0.f)) { + if (drcComp->fullChannels > 5) { + tmp = fMult(dmxGain5, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); + } + tmp = fMult(dmxGain2, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); + } + peak[0] = fixMax(peak[0], fixp_abs(tmp)); + } + + /* Lo/Ro downmix */ + for (i = 0; i < drcComp->blockLength; i++) { + const INT_PCM* pSamples = &inSamples[i * drcComp->channels]; + + /* Lo */ + tmp = FL2FXCONST_DBL(0.f); + switch (drcComp->chanConfig) { + case MODE_6_1: + tmp += fMultDiv2(fMult(slev, ext_leva), + (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + tmp += fMultDiv2(fMult(slev, ext_levb), + (FIXP_PCM)pSamples[drcComp->channelIdx[6]]) >> + (DOWNMIX_SHIFT - 1); /* Cs */ + tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> + DOWNMIX_SHIFT); /* L */ + tmp += + fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + break; + case MODE_7_1_BACK: + case MODE_7_1_REAR_SURROUND: + tmp += fMultDiv2(fMult(slev, ext_leva), + (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + tmp += fMultDiv2(fMult(slev, ext_levb), + (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> + (DOWNMIX_SHIFT - 1); /* Lrs / Lss*/ + tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> + DOWNMIX_SHIFT); /* L */ + tmp += + fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + break; + case MODE_1_2_2_2_1: + case MODE_7_1_FRONT_CENTER: + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> + DOWNMIX_SHIFT); /* L */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> + (DOWNMIX_SHIFT - 1); /* Lc */ + tmp += fMultDiv2(fMult(ext_leva, clev), + (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> + (DOWNMIX_SHIFT - 1); /* Lc - second path*/ + tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + tmp += + fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + break; + case MODE_7_1_TOP_FRONT: + tmp += + fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> + (DOWNMIX_SHIFT - 1); /* L */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> + (DOWNMIX_SHIFT - 1); /* Lvh */ + tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + tmp += + fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + break; + default: + if (drcComp->channelIdx[LS] >= 0) + tmp += + fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[LS]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + if (drcComp->channelIdx[LS2] >= 0) + tmp += + fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[LS2]]) >> + (DOWNMIX_SHIFT - 1); /* Ls2 */ + if ((drcComp->channelIdx[LS] >= 0) && (drcComp->channelIdx[LS2] >= 0)) + tmp = fMult(FL2FXCONST_DBL(0.707f), tmp); /* 7.1ch */ + if (drcComp->channelIdx[S] >= 0) + tmp += + fMultDiv2(slev, + fMult(FL2FXCONST_DBL(0.7f), + (FIXP_PCM)pSamples[drcComp->channelIdx[S]])) >> + (DOWNMIX_SHIFT - 1); /* S */ + if (drcComp->channelIdx[C] >= 0) + tmp += + fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[C]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + if (drcComp->channelIdx[3] >= 0) + tmp += fMultDiv2(lfe_lev, + (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[L]]) >> + DOWNMIX_SHIFT); /* L */ + break; + } + + /* apply scaling of downmix gains */ + /* only for positive values only, as legacy decoders might not know this + * parameter */ + if (dmxGain2 > FL2FXCONST_DBL(0.f)) { + if (drcComp->fullChannels > 5) { + tmp = fMult(dmxGain5, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); + } + tmp = fMult(dmxGain2, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); + } + peak[0] = fixMax(peak[0], fixp_abs(tmp)); + + /* Ro */ + tmp = FL2FXCONST_DBL(0.f); + switch (drcComp->chanConfig) { + case MODE_6_1: + tmp += fMultDiv2(fMult(slev, ext_leva), + (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + tmp += fMultDiv2(fMult(slev, ext_levb), + (FIXP_PCM)pSamples[drcComp->channelIdx[6]]) >> + (DOWNMIX_SHIFT - 1); /* Cs */ + tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> + DOWNMIX_SHIFT); /* R */ + tmp += + fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + break; + case MODE_7_1_BACK: + case MODE_7_1_REAR_SURROUND: + tmp += fMultDiv2(fMult(slev, ext_leva), + (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + tmp += fMultDiv2(fMult(slev, ext_levb), + (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> + (DOWNMIX_SHIFT - 1); /* Rrs / Rss*/ + tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> + DOWNMIX_SHIFT); /* R */ + tmp += + fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + break; + case MODE_1_2_2_2_1: + case MODE_7_1_FRONT_CENTER: + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> + DOWNMIX_SHIFT); /* R */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> + (DOWNMIX_SHIFT - 1); /* Rc */ + tmp += fMultDiv2(fMult(ext_leva, clev), + (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> + (DOWNMIX_SHIFT - 1); /* Rc - second path*/ + tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + tmp += + fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + break; + case MODE_7_1_TOP_FRONT: + tmp += + fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> + (DOWNMIX_SHIFT - 1); /* R */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> + (DOWNMIX_SHIFT - 1); /* Rvh */ + tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + tmp += + fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + break; + default: + if (drcComp->channelIdx[RS] >= 0) + tmp += + fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[RS]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + if (drcComp->channelIdx[RS2] >= 0) + tmp += + fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[RS2]]) >> + (DOWNMIX_SHIFT - 1); /* Rs2 */ + if ((drcComp->channelIdx[RS] >= 0) && (drcComp->channelIdx[RS2] >= 0)) + tmp = fMult(FL2FXCONST_DBL(0.707f), tmp); /* 7.1ch */ + if (drcComp->channelIdx[S] >= 0) + tmp += + fMultDiv2(slev, + fMult(FL2FXCONST_DBL(0.7f), + (FIXP_PCM)pSamples[drcComp->channelIdx[S]])) >> + (DOWNMIX_SHIFT - 1); /* S */ + if (drcComp->channelIdx[C] >= 0) + tmp += + fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[C]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + if (drcComp->channelIdx[3] >= 0) + tmp += fMultDiv2(lfe_lev, + (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[R]]) >> + DOWNMIX_SHIFT); /* R */ + } + + /* apply scaling of downmix gains */ + /* only for positive values only, as legacy decoders might not know this + * parameter */ + if (dmxGain2 > FL2FXCONST_DBL(0.f)) { + if (drcComp->fullChannels > 5) { + tmp = fMult(dmxGain5, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); + } + tmp = fMult(dmxGain2, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); + } + peak[0] = fixMax(peak[0], fixp_abs(tmp)); + } + } + + peak[1] = fixMax(peak[0], peak[1]); + + /* Mono Downmix - for comp_val only */ + if (drcComp->fullChannels > 1) { + for (i = 0; i < drcComp->blockLength; i++) { + const INT_PCM* pSamples = &inSamples[i * drcComp->channels]; + + tmp = FL2FXCONST_DBL(0.f); + switch (drcComp->chanConfig) { + case MODE_6_1: + tmp += fMultDiv2(fMult(slev, ext_leva), + (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + tmp += fMultDiv2(fMult(slev, ext_leva), + (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + tmp += fMult(fMult(slev, ext_levb), + (FIXP_PCM)pSamples[drcComp->channelIdx[6]]) >> + (DOWNMIX_SHIFT - 1); /* Cs */ + tmp += fMult(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> + DOWNMIX_SHIFT); /* L */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> + DOWNMIX_SHIFT); /* R */ + tmp += fMult(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + break; + case MODE_7_1_BACK: + case MODE_7_1_REAR_SURROUND: + tmp += fMultDiv2(fMult(slev, ext_leva), + (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + tmp += fMultDiv2(fMult(slev, ext_leva), + (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + tmp += fMultDiv2(fMult(slev, ext_levb), + (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> + (DOWNMIX_SHIFT - 1); /* Lrs / Lss*/ + tmp += fMultDiv2(fMult(slev, ext_levb), + (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> + (DOWNMIX_SHIFT - 1); /* Rrs / Rss*/ + tmp += fMult(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> + DOWNMIX_SHIFT); /* L */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> + DOWNMIX_SHIFT); /* R */ + tmp += fMult(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + break; + case MODE_1_2_2_2_1: + case MODE_7_1_FRONT_CENTER: + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> + DOWNMIX_SHIFT); /* L */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> + DOWNMIX_SHIFT); /* R */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> + (DOWNMIX_SHIFT - 1); /* Lc */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> + (DOWNMIX_SHIFT - 1); /* Rc */ + tmp += fMultDiv2(fMult(ext_leva, clev), + (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> + (DOWNMIX_SHIFT - 1); /* Lc - second path*/ + tmp += fMultDiv2(fMult(ext_leva, clev), + (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> + (DOWNMIX_SHIFT - 1); /* Rc - second path*/ + tmp += fMult(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + tmp += fMult(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + break; + case MODE_7_1_TOP_FRONT: + tmp += + fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> + (DOWNMIX_SHIFT - 1); /* L */ + tmp += + fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> + (DOWNMIX_SHIFT - 1); /* R */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> + (DOWNMIX_SHIFT - 1); /* Lvh */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> + (DOWNMIX_SHIFT - 1); /* Rvh */ + tmp += fMult(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + tmp += fMult(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + break; + default: + if (drcComp->channelIdx[LS] >= 0) + tmp += + fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[LS]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + if (drcComp->channelIdx[LS2] >= 0) + tmp += + fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[LS2]]) >> + (DOWNMIX_SHIFT - 1); /* Ls2 */ + if (drcComp->channelIdx[RS] >= 0) + tmp += + fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[RS]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + if (drcComp->channelIdx[RS2] >= 0) + tmp += + fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[RS2]]) >> + (DOWNMIX_SHIFT - 1); /* Rs2 */ + if ((drcComp->channelIdx[LS] >= 0) && (drcComp->channelIdx[LS2] >= 0)) + tmp = fMult(FL2FXCONST_DBL(0.707f), tmp); /* 7.1ch */ + /*if ((drcComp->channelIdx[RS] >= 0) && (drcComp->channelIdx[RS2] >= 0)) tmp *=0.707f;*/ /* 7.1ch */ + if (drcComp->channelIdx[S] >= 0) + tmp += + fMultDiv2(slev, + fMult(FL2FXCONST_DBL(0.7f), + (FIXP_PCM)pSamples[drcComp->channelIdx[S]])) >> + (DOWNMIX_SHIFT - 1); /* S */ + if (drcComp->channelIdx[C] >= 0) + tmp += fMult(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[C]]) >> + (DOWNMIX_SHIFT - 1); /* C (2*clev) */ + if (drcComp->channelIdx[3] >= 0) + tmp += fMult(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[L]]) >> + DOWNMIX_SHIFT); /* L */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[R]]) >> + DOWNMIX_SHIFT); /* R */ + } + + /* apply scaling of downmix gains */ + /* only for positive values only, as legacy decoders might not know this + * parameter */ + if (dmxGain2 > FL2FXCONST_DBL(0.f)) { + if (drcComp->fullChannels > 5) { + tmp = fMult(dmxGain5, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); + } + tmp = fMult(dmxGain2, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); + } + peak[1] = fixMax(peak[1], fixp_abs(tmp)); + } + } +} + +INT FDK_DRC_Generator_Open(HDRC_COMP* phDrcComp) { + INT err = 0; + HDRC_COMP hDcComp = NULL; + + if (phDrcComp == NULL) { + err = -1; + goto bail; + } + + /* allocate memory */ + hDcComp = (HDRC_COMP)FDKcalloc(1, sizeof(DRC_COMP)); + + if (hDcComp == NULL) { + err = -1; + goto bail; + } + + FDKmemclear(hDcComp, sizeof(DRC_COMP)); + + /* Return drc compressor instance */ + *phDrcComp = hDcComp; + return err; +bail: + FDK_DRC_Generator_Close(&hDcComp); + return err; +} + +INT FDK_DRC_Generator_Close(HDRC_COMP* phDrcComp) { + if (phDrcComp == NULL) { + return -1; + } + if (*phDrcComp != NULL) { + FDKfree(*phDrcComp); + *phDrcComp = NULL; + } + return 0; +} + +INT FDK_DRC_Generator_Initialize(HDRC_COMP drcComp, + const DRC_PROFILE profileLine, + const DRC_PROFILE profileRF, + const INT blockLength, const UINT sampleRate, + const CHANNEL_MODE channelMode, + const CHANNEL_ORDER channelOrder, + const UCHAR useWeighting) { + int i; + CHANNEL_MAPPING channelMapping; + + drcComp->limDecay = + FL2FXCONST_DBL(((0.006f / 256) * blockLength) / METADATA_INT_SCALE); + + /* Save parameters. */ + drcComp->blockLength = blockLength; + drcComp->sampleRate = sampleRate; + drcComp->chanConfig = channelMode; + drcComp->useWeighting = useWeighting; + + if (FDK_DRC_Generator_setDrcProfile(drcComp, profileLine, profileRF) != + 0) { /* expects initialized blockLength and sampleRate */ + return (-1); + } + + /* Set number of channels and channel offsets. */ + if (FDKaacEnc_InitChannelMapping(channelMode, channelOrder, + &channelMapping) != AAC_ENC_OK) { + return (-2); + } + + for (i = 0; i < 9; i++) drcComp->channelIdx[i] = -1; + + switch (channelMode) { + case MODE_1: /* mono */ + drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; + break; + case MODE_2: /* stereo */ + drcComp->channelIdx[L] = channelMapping.elInfo[0].ChannelIndex[0]; + drcComp->channelIdx[R] = channelMapping.elInfo[0].ChannelIndex[1]; + break; + case MODE_1_2: /* 3ch */ + drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; + drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; + drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; + break; + case MODE_1_2_1: /* 4ch */ + drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; + drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; + drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; + drcComp->channelIdx[S] = channelMapping.elInfo[2].ChannelIndex[0]; + break; + case MODE_1_2_2: /* 5ch */ + drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; + drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; + drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; + drcComp->channelIdx[LS] = channelMapping.elInfo[2].ChannelIndex[0]; + drcComp->channelIdx[RS] = channelMapping.elInfo[2].ChannelIndex[1]; + break; + case MODE_1_2_2_1: /* 5.1 ch */ + drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; + drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; + drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; + drcComp->channelIdx[LFE] = channelMapping.elInfo[3].ChannelIndex[0]; + drcComp->channelIdx[LS] = channelMapping.elInfo[2].ChannelIndex[0]; + drcComp->channelIdx[RS] = channelMapping.elInfo[2].ChannelIndex[1]; + break; + case MODE_1_2_2_2_1: /* 7.1 ch */ + case MODE_7_1_FRONT_CENTER: + drcComp->channelIdx[L] = channelMapping.elInfo[2].ChannelIndex[0]; /* l */ + drcComp->channelIdx[R] = channelMapping.elInfo[2].ChannelIndex[1]; /* r */ + drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; /* c */ + drcComp->channelIdx[LFE] = + channelMapping.elInfo[4].ChannelIndex[0]; /* lfe */ + drcComp->channelIdx[LS] = + channelMapping.elInfo[3].ChannelIndex[0]; /* ls */ + drcComp->channelIdx[RS] = + channelMapping.elInfo[3].ChannelIndex[1]; /* rs */ + drcComp->channelIdx[LS2] = + channelMapping.elInfo[1].ChannelIndex[0]; /* lc */ + drcComp->channelIdx[RS2] = + channelMapping.elInfo[1].ChannelIndex[1]; /* rc */ + break; + case MODE_7_1_BACK: + case MODE_7_1_REAR_SURROUND: + drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; /* l */ + drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; /* r */ + drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; /* c */ + drcComp->channelIdx[LFE] = + channelMapping.elInfo[4].ChannelIndex[0]; /* lfe */ + drcComp->channelIdx[LS] = + channelMapping.elInfo[3].ChannelIndex[0]; /* lrear */ + drcComp->channelIdx[RS] = + channelMapping.elInfo[3].ChannelIndex[1]; /* rrear */ + drcComp->channelIdx[LS2] = + channelMapping.elInfo[2].ChannelIndex[0]; /* ls */ + drcComp->channelIdx[RS2] = + channelMapping.elInfo[2].ChannelIndex[1]; /* rs */ + break; + case MODE_6_1: + drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; /* l */ + drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; /* r */ + drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; /* c */ + drcComp->channelIdx[LFE] = + channelMapping.elInfo[4].ChannelIndex[0]; /* lfe */ + drcComp->channelIdx[LS] = + channelMapping.elInfo[2].ChannelIndex[0]; /* ls */ + drcComp->channelIdx[RS] = + channelMapping.elInfo[2].ChannelIndex[1]; /* rs */ + drcComp->channelIdx[S] = channelMapping.elInfo[3].ChannelIndex[0]; /* s */ + break; + case MODE_7_1_TOP_FRONT: + drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; /* l */ + drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; /* r */ + drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; /* c */ + drcComp->channelIdx[LFE] = + channelMapping.elInfo[3].ChannelIndex[0]; /* lfe */ + drcComp->channelIdx[LS] = + channelMapping.elInfo[2].ChannelIndex[0]; /* ls */ + drcComp->channelIdx[RS] = + channelMapping.elInfo[2].ChannelIndex[1]; /* rs */ + drcComp->channelIdx[LS2] = + channelMapping.elInfo[4].ChannelIndex[0]; /* lvh2 */ + drcComp->channelIdx[RS2] = + channelMapping.elInfo[4].ChannelIndex[1]; /* rvh2 */ + break; + default: + return (-1); + } + + drcComp->fullChannels = channelMapping.nChannelsEff; + drcComp->channels = channelMapping.nChannels; + + /* Init states. */ + drcComp->smoothLevel[0] = drcComp->smoothLevel[1] = + (FIXP_DBL)(-(135 << METADATA_FRACT_BITS)); + + FDKmemclear(drcComp->smoothGain, sizeof(drcComp->smoothGain)); + FDKmemclear(drcComp->holdCnt, sizeof(drcComp->holdCnt)); + FDKmemclear(drcComp->limGain, sizeof(drcComp->limGain)); + FDKmemclear(drcComp->prevPeak, sizeof(drcComp->prevPeak)); + FDKmemclear(drcComp->filter, sizeof(drcComp->filter)); + + return (0); +} + +INT FDK_DRC_Generator_setDrcProfile(HDRC_COMP drcComp, + const DRC_PROFILE profileLine, + const DRC_PROFILE profileRF) { + int profileIdx, i; + + drcComp->profile[0] = profileLine; + drcComp->profile[1] = profileRF; + + for (i = 0; i < 2; i++) { + /* get profile index */ + switch (drcComp->profile[i]) { + case DRC_NONE: + case DRC_NOT_PRESENT: + case DRC_FILMSTANDARD: + profileIdx = 0; + break; + case DRC_FILMLIGHT: + profileIdx = 1; + break; + case DRC_MUSICSTANDARD: + profileIdx = 2; + break; + case DRC_MUSICLIGHT: + profileIdx = 3; + break; + case DRC_SPEECH: + profileIdx = 4; + break; + case DRC_DELAY_TEST: + profileIdx = 5; + break; + default: + return (-1); + } + + /* get parameters for selected profile */ + if (profileIdx >= 0) { + drcComp->maxBoostThr[i] = tabMaxBoostThr[profileIdx]; + drcComp->boostThr[i] = tabBoostThr[profileIdx]; + drcComp->earlyCutThr[i] = tabEarlyCutThr[profileIdx]; + drcComp->cutThr[i] = tabCutThr[profileIdx]; + drcComp->maxCutThr[i] = tabMaxCutThr[profileIdx]; + + drcComp->boostFac[i] = tabBoostRatio[profileIdx]; + drcComp->earlyCutFac[i] = tabEarlyCutRatio[profileIdx]; + drcComp->cutFac[i] = tabCutRatio[profileIdx]; + + drcComp->maxBoost[i] = tabMaxBoost[profileIdx]; + drcComp->maxCut[i] = tabMaxCut[profileIdx]; + drcComp->maxEarlyCut[i] = + -fMult((drcComp->cutThr[i] - drcComp->earlyCutThr[i]), + drcComp->earlyCutFac[i]); /* no scaling after mult needed, + earlyCutFac is in FIXP_DBL */ + + drcComp->fastAttack[i] = tc2Coeff( + tabFastAttack[profileIdx], drcComp->sampleRate, drcComp->blockLength); + drcComp->fastDecay[i] = tc2Coeff( + tabFastDecay[profileIdx], drcComp->sampleRate, drcComp->blockLength); + drcComp->slowAttack[i] = tc2Coeff( + tabSlowAttack[profileIdx], drcComp->sampleRate, drcComp->blockLength); + drcComp->slowDecay[i] = tc2Coeff( + tabSlowDecay[profileIdx], drcComp->sampleRate, drcComp->blockLength); + drcComp->holdOff[i] = tabHoldOff[profileIdx] * 256 / drcComp->blockLength; + + drcComp->attackThr[i] = tabAttackThr[profileIdx]; + drcComp->decayThr[i] = tabDecayThr[profileIdx]; + } + + drcComp->smoothGain[i] = FL2FXCONST_DBL(0.f); + } + return (0); +} + +INT FDK_DRC_Generator_Calc(HDRC_COMP drcComp, const INT_PCM* const inSamples, + const UINT inSamplesBufSize, const INT dialnorm, + const INT drc_TargetRefLevel, + const INT comp_TargetRefLevel, const FIXP_DBL clev, + const FIXP_DBL slev, const FIXP_DBL ext_leva, + const FIXP_DBL ext_levb, const FIXP_DBL lfe_lev, + const INT dmxGain5, const INT dmxGain2, + INT* const pDynrng, INT* const pCompr) { + int i, c; + FIXP_DBL peak[2]; + + /************************************************************************** + * compressor + **************************************************************************/ + if ((drcComp->profile[0] != DRC_NONE) || (drcComp->profile[1] != DRC_NONE)) { + /* Calc loudness level */ + FIXP_DBL level_b = FL2FXCONST_DBL(0.f); + int level_e = DFRACT_BITS - 1; + + /* Increase energy time resolution with shorter processing blocks. 16 is an + * empiric value. */ + const int granuleLength = fixMin(16, drcComp->blockLength); + + if (drcComp->useWeighting) { + FIXP_DBL x1, x2, y, y1, y2; + /* sum of filter coefficients about 2.5 -> squared value is 6.25 + WEIGHTING_FILTER_SHIFT is 2 -> scaling about 16, therefore reduce + granuleShift by 1. + */ + const int granuleShift = getShiftFactor(granuleLength) - 1; + + for (c = 0; c < (int)drcComp->channels; c++) { + const INT_PCM* pSamples = inSamples + c * inSamplesBufSize; + + if (c == drcComp->channelIdx[LFE]) { + continue; /* skip LFE */ + } + + /* get filter states */ + x1 = drcComp->filter[c].x1; + x2 = drcComp->filter[c].x2; + y1 = drcComp->filter[c].y1; + y2 = drcComp->filter[c].y2; + + i = 0; + + do { + int offset = i; + FIXP_DBL accu = FL2FXCONST_DBL(0.f); + + for (i = offset; + i < fixMin(offset + granuleLength, drcComp->blockLength); i++) { + /* apply weighting filter */ + FIXP_DBL x = + FX_PCM2FX_DBL((FIXP_PCM)pSamples[i]) >> WEIGHTING_FILTER_SHIFT; + + /* y = b0 * (x - x2) - a1 * y1 - a2 * y2; */ + y = fMult(b0, x - x2) - fMult(a1, y1) - fMult(a2, y2); + + x2 = x1; + x1 = x; + y2 = y1; + y1 = y; + + accu += fPow2Div2(y) >> (granuleShift - 1); /* partial energy */ + } /* i */ + + fixpAdd(accu, granuleShift + 2 * WEIGHTING_FILTER_SHIFT, &level_b, + &level_e); /* sup up partial energies */ + + } while (i < drcComp->blockLength); + + /* save filter states */ + drcComp->filter[c].x1 = x1; + drcComp->filter[c].x2 = x2; + drcComp->filter[c].y1 = y1; + drcComp->filter[c].y2 = y2; + } /* c */ + } /* weighting */ + else { + const int granuleShift = getShiftFactor(granuleLength); + + for (c = 0; c < (int)drcComp->channels; c++) { + const INT_PCM* pSamples = inSamples + c * inSamplesBufSize; + + if ((int)c == drcComp->channelIdx[LFE]) { + continue; /* skip LFE */ + } + + i = 0; + + do { + int offset = i; + FIXP_DBL accu = FL2FXCONST_DBL(0.f); + + for (i = offset; + i < fixMin(offset + granuleLength, drcComp->blockLength); i++) { + /* partial energy */ + accu += fPow2Div2((FIXP_PCM)pSamples[i]) >> (granuleShift - 1); + } /* i */ + + fixpAdd(accu, granuleShift, &level_b, + &level_e); /* sup up partial energies */ + + } while (i < drcComp->blockLength); + } + } /* weighting */ + + /* + * Convert to dBFS, apply dialnorm + */ + /* level scaling */ + + /* descaled level in ld64 representation */ + FIXP_DBL ldLevel = + CalcLdData(level_b) + + (FIXP_DBL)((level_e - 12) << (DFRACT_BITS - 1 - LD_DATA_SHIFT)) - + CalcLdData((FIXP_DBL)(drcComp->blockLength << (DFRACT_BITS - 1 - 12))); + + /* if (level < 1e-10) level = 1e-10f; */ + ldLevel = + fMax(ldLevel, FL2FXCONST_DBL(-0.51905126482615036685473741085772f)); + + /* level = 10 * log(level)/log(10) + 3; + * = 10*log(2)/log(10) * ld(level) + 3; + * = 10 * 0.30102999566398119521373889472449 * ld(level) + 3 + * = 10 * (0.30102999566398119521373889472449 * ld(level) + 0.3) + * = 10 * (0.30102999566398119521373889472449 * ld64(level) + 0.3/64) + * * 64 + * + * additional scaling with METADATA_FRACT_BITS: + * = 10 * (0.30102999566398119521373889472449 * ld64(level) + 0.3/64) + * * 64 * 2^(METADATA_FRACT_BITS) = 10 * (0.30102999566398119521373889472449 + * * ld64(level) + 0.3/64) * 2^(METADATA_FRACT_BITS+LD_DATA_SHIFT) = + * 10*2^(METADATA_FRACT_BITS+LD_DATA_SHIFT) * ( + * 0.30102999566398119521373889472449 * ld64(level) + 0.3/64 ) + * */ + FIXP_DBL level = fMult( + (FIXP_DBL)(10 << (METADATA_FRACT_BITS + LD_DATA_SHIFT)), + fMult(FL2FXCONST_DBL(0.30102999566398119521373889472449f), ldLevel) + + (FIXP_DBL)(FL2FXCONST_DBL(0.3f) >> LD_DATA_SHIFT)); + + /* level -= dialnorm + 31 */ /* this is fixed to Dolby-ReferenceLevel as + compressor profiles are defined relative to + this */ + level -= ((FIXP_DBL)(dialnorm << (METADATA_FRACT_BITS - 16)) + + (FIXP_DBL)(31 << METADATA_FRACT_BITS)); + + for (i = 0; i < 2; i++) { + if (drcComp->profile[i] == DRC_NONE) { + /* no compression */ + drcComp->smoothGain[i] = FL2FXCONST_DBL(0.f); + } else { + FIXP_DBL gain, alpha, lvl2smthlvl; + + /* calc static gain */ + if (level <= drcComp->maxBoostThr[i]) { + /* max boost */ + gain = drcComp->maxBoost[i]; + } else if (level < drcComp->boostThr[i]) { + /* boost range */ + gain = fMult((level - drcComp->boostThr[i]), drcComp->boostFac[i]); + } else if (level <= drcComp->earlyCutThr[i]) { + /* null band */ + gain = FL2FXCONST_DBL(0.f); + } else if (level <= drcComp->cutThr[i]) { + /* early cut range */ + gain = + fMult((level - drcComp->earlyCutThr[i]), drcComp->earlyCutFac[i]); + } else if (level < drcComp->maxCutThr[i]) { + /* cut range */ + gain = fMult((level - drcComp->cutThr[i]), drcComp->cutFac[i]) - + drcComp->maxEarlyCut[i]; + } else { + /* max cut */ + gain = -drcComp->maxCut[i]; + } + + /* choose time constant */ + lvl2smthlvl = level - drcComp->smoothLevel[i]; + if (gain < drcComp->smoothGain[i]) { + /* attack */ + if (lvl2smthlvl > drcComp->attackThr[i]) { + /* fast attack */ + alpha = drcComp->fastAttack[i]; + } else { + /* slow attack */ + alpha = drcComp->slowAttack[i]; + } + } else { + /* release */ + if (lvl2smthlvl < -drcComp->decayThr[i]) { + /* fast release */ + alpha = drcComp->fastDecay[i]; + } else { + /* slow release */ + alpha = drcComp->slowDecay[i]; + } + } + + /* smooth gain & level */ + if ((gain < drcComp->smoothGain[i]) || + (drcComp->holdCnt[i] == + 0)) { /* hold gain unless we have an attack or hold + period is over */ + FIXP_DBL accu; + + /* drcComp->smoothLevel[i] = (1-alpha) * drcComp->smoothLevel[i] + + * alpha * level; */ + accu = fMult(((FIXP_DBL)MAXVAL_DBL - alpha), drcComp->smoothLevel[i]); + accu += fMult(alpha, level); + drcComp->smoothLevel[i] = accu; + + /* drcComp->smoothGain[i] = (1-alpha) * drcComp->smoothGain[i] + + * alpha * gain; */ + accu = fMult(((FIXP_DBL)MAXVAL_DBL - alpha), drcComp->smoothGain[i]); + accu += fMult(alpha, gain); + drcComp->smoothGain[i] = accu; + } + + /* hold counter */ + if (drcComp->holdCnt[i]) { + drcComp->holdCnt[i]--; + } + if (gain < drcComp->smoothGain[i]) { + drcComp->holdCnt[i] = drcComp->holdOff[i]; + } + } /* profile != DRC_NONE */ + } /* for i=1..2 */ + } else { + /* no compression */ + drcComp->smoothGain[0] = FL2FXCONST_DBL(0.f); + drcComp->smoothGain[1] = FL2FXCONST_DBL(0.f); + } + + /************************************************************************** + * limiter + **************************************************************************/ + + findPeakLevels(drcComp, inSamples, clev, slev, ext_leva, ext_levb, lfe_lev, + (FIXP_DBL)((LONG)(dmxGain5) << (METADATA_FRACT_BITS - 16)), + (FIXP_DBL)((LONG)(dmxGain2) << (METADATA_FRACT_BITS - 16)), + peak); + + for (i = 0; i < 2; i++) { + FIXP_DBL tmp = drcComp->prevPeak[i]; + drcComp->prevPeak[i] = peak[i]; + peak[i] = fixMax(peak[i], tmp); + + /* + * Convert to dBFS, apply dialnorm + */ + /* descaled peak in ld64 representation */ + FIXP_DBL ld_peak = + CalcLdData(peak[i]) + + (FIXP_DBL)((LONG)DOWNMIX_SHIFT << (DFRACT_BITS - 1 - LD_DATA_SHIFT)); + + /* if (peak < 1e-6) level = 1e-6f; */ + ld_peak = + fMax(ld_peak, FL2FXCONST_DBL(-0.31143075889569022011284244651463f)); + + /* peak[i] = 20 * log(peak[i])/log(10) + 0.2f + + * (drcComp->smoothGain[i]*2^METADATA_FRACT_BITS) peak[i] = 20 * + * log(2)/log(10) * ld(peak[i]) + 0.2f + + * (drcComp->smoothGain[i]*2^METADATA_FRACT_BITS) peak[i] = 10 * + * 2*0.30102999566398119521373889472449 * ld(peak[i]) + 0.2f + + * (drcComp->smoothGain[i]*2^METADATA_FRACT_BITS) + * + * additional scaling with METADATA_FRACT_BITS: + * peak[i] = (10 * 2*0.30102999566398119521373889472449 * ld64(peak[i]) * 64 + * + 0.2f + + * (drcComp->smoothGain[i]*2^METADATA_FRACT_BITS))*2^(-METADATA_FRACT_BITS) + * peak[i] = 10*2^(METADATA_FRACT_BITS+LD_DATA_SHIFT) * + * 2*0.30102999566398119521373889472449 * ld64(peak[i]) + * + 0.2f*2^(-METADATA_FRACT_BITS) + drcComp->smoothGain[i] + */ + peak[i] = fMult( + (FIXP_DBL)(10 << (METADATA_FRACT_BITS + LD_DATA_SHIFT)), + fMult(FL2FX_DBL(2 * 0.30102999566398119521373889472449f), ld_peak)); + peak[i] += + (FL2FX_DBL(0.5f) >> METADATA_INT_BITS); /* add a little bit headroom */ + peak[i] += drcComp->smoothGain[i]; + } + + /* peak -= dialnorm + 31; */ /* this is Dolby style only */ + peak[0] -= (FIXP_DBL)((dialnorm - drc_TargetRefLevel) + << (METADATA_FRACT_BITS - + 16)); /* peak[0] -= dialnorm - drc_TargetRefLevel */ + + /* peak += 11; */ + /* this is Dolby style only */ /* RF mode output is 11dB higher */ + /*peak += comp_TargetRefLevel - drc_TargetRefLevel;*/ + peak[1] -= + (FIXP_DBL)((dialnorm - comp_TargetRefLevel) + << (METADATA_FRACT_BITS - + 16)); /* peak[1] -= dialnorm - comp_TargetRefLevel */ + + /* limiter gain */ + drcComp->limGain[0] += drcComp->limDecay; /* linear limiter release */ + drcComp->limGain[0] = fixMin(drcComp->limGain[0], -peak[0]); + + drcComp->limGain[1] += 2 * drcComp->limDecay; /* linear limiter release */ + drcComp->limGain[1] = fixMin(drcComp->limGain[1], -peak[1]); + + /*************************************************************************/ + + /* apply limiting, return DRC gains*/ + { + FIXP_DBL tmp; + + tmp = drcComp->smoothGain[0]; + if (drcComp->limGain[0] < FL2FXCONST_DBL(0.f)) { + tmp += drcComp->limGain[0]; + } + *pDynrng = (LONG)scaleValue(tmp, -(METADATA_FRACT_BITS - 16)); + + tmp = drcComp->smoothGain[1]; + if (drcComp->limGain[1] < FL2FXCONST_DBL(0.f)) { + tmp += drcComp->limGain[1]; + } + *pCompr = (LONG)scaleValue(tmp, -(METADATA_FRACT_BITS - 16)); + } + + return 0; +} + +DRC_PROFILE FDK_DRC_Generator_getDrcProfile(const HDRC_COMP drcComp) { + return drcComp->profile[0]; +} + +DRC_PROFILE FDK_DRC_Generator_getCompProfile(const HDRC_COMP drcComp) { + return drcComp->profile[1]; +} |