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| author | Matthias P. Braendli <matthias.braendli@mpb.li> | 2019-11-11 11:38:02 +0100 |
|---|---|---|
| committer | Matthias P. Braendli <matthias.braendli@mpb.li> | 2019-11-11 11:38:02 +0100 |
| commit | 0e5af65c467b2423a0b857ae3ad98c91acc1e190 (patch) | |
| tree | d07f69550d8886271e44fe79c4dcfb299cafbd38 /fdk-aac/libSBRdec/src/env_calc.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/libSBRdec/src/env_calc.cpp')
| -rw-r--r-- | fdk-aac/libSBRdec/src/env_calc.cpp | 3158 |
1 files changed, 3158 insertions, 0 deletions
diff --git a/fdk-aac/libSBRdec/src/env_calc.cpp b/fdk-aac/libSBRdec/src/env_calc.cpp new file mode 100644 index 0000000..cb1474f --- /dev/null +++ b/fdk-aac/libSBRdec/src/env_calc.cpp @@ -0,0 +1,3158 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief Envelope calculation + + The envelope adjustor compares the energies present in the transposed + highband to the reference energies conveyed with the bitstream. + The highband is amplified (sometimes) or attenuated (mostly) to the + desired level. + + The spectral shape of the reference energies can be changed several times per + frame if necessary. Each set of energy values corresponding to a certain range + in time will be called an <em>envelope</em> here. + The bitstream supports several frequency scales and two resolutions. Normally, + one or more QMF-subbands are grouped to one SBR-band. An envelope contains + reference energies for each SBR-band. + In addition to the energy envelopes, noise envelopes are transmitted that + define the ratio of energy which is generated by adding noise instead of + transposing the lowband. The noise envelopes are given in a coarser time + and frequency resolution. + If a signal contains strong tonal components, synthetic sines can be + generated in individual SBR bands. + + An overlap buffer of 6 QMF-timeslots is used to allow a more + flexible alignment of the envelopes in time that is not restricted to the + core codec's frame borders. + Therefore the envelope adjustor has access to the spectral data of the + current frame as well as the last 6 QMF-timeslots of the previous frame. + However, in average only the data of 1 frame is being processed as + the adjustor is called once per frame. + + Depending on the frequency range set in the bitstream, only QMF-subbands + between <em>lowSubband</em> and <em>highSubband</em> are adjusted. + + Scaling of spectral data to maximize SNR (see #QMF_SCALE_FACTOR) as well as a + special Mantissa-Exponent format ( see calculateSbrEnvelope() ) are being + used. The main entry point for this modules is calculateSbrEnvelope(). + + \sa sbr_scale.h, #QMF_SCALE_FACTOR, calculateSbrEnvelope(), \ref + documentationOverview +*/ + +#include "env_calc.h" + +#include "sbrdec_freq_sca.h" +#include "env_extr.h" +#include "transcendent.h" +#include "sbr_ram.h" +#include "sbr_rom.h" + +#include "genericStds.h" /* need FDKpow() for debug outputs */ + +typedef struct { + FIXP_DBL nrgRef[MAX_FREQ_COEFFS]; + FIXP_DBL nrgEst[MAX_FREQ_COEFFS]; + FIXP_DBL nrgGain[MAX_FREQ_COEFFS]; + FIXP_DBL noiseLevel[MAX_FREQ_COEFFS]; + FIXP_DBL nrgSine[MAX_FREQ_COEFFS]; + + SCHAR nrgRef_e[MAX_FREQ_COEFFS]; + SCHAR nrgEst_e[MAX_FREQ_COEFFS]; + SCHAR nrgGain_e[MAX_FREQ_COEFFS]; + SCHAR noiseLevel_e[MAX_FREQ_COEFFS]; + SCHAR nrgSine_e[MAX_FREQ_COEFFS]; + /* yet another exponent [0]: for ts < no_cols; [1]: for ts >= no_cols */ + SCHAR exponent[2]; +} ENV_CALC_NRGS; + +static void equalizeFiltBufferExp(FIXP_DBL *filtBuffer, SCHAR *filtBuffer_e, + FIXP_DBL *NrgGain, SCHAR *NrgGain_e, + int subbands); + +static void calcNrgPerSubband(FIXP_DBL **analysBufferReal, + FIXP_DBL **analysBufferImag, int lowSubband, + int highSubband, int start_pos, int next_pos, + SCHAR frameExp, FIXP_DBL *nrgEst, + SCHAR *nrgEst_e); + +static void calcNrgPerSfb(FIXP_DBL **analysBufferReal, + FIXP_DBL **analysBufferImag, int nSfb, + UCHAR *freqBandTable, int start_pos, int next_pos, + SCHAR input_e, FIXP_DBL *nrg_est, SCHAR *nrg_est_e); + +static void calcSubbandGain(FIXP_DBL nrgRef, SCHAR nrgRef_e, + ENV_CALC_NRGS *nrgs, int c, FIXP_DBL tmpNoise, + SCHAR tmpNoise_e, UCHAR sinePresentFlag, + UCHAR sineMapped, int noNoiseFlag); + +static void calcAvgGain(ENV_CALC_NRGS *nrgs, int lowSubband, int highSubband, + FIXP_DBL *sumRef_m, SCHAR *sumRef_e, + FIXP_DBL *ptrAvgGain_m, SCHAR *ptrAvgGain_e); + +static void adjustTimeSlot_EldGrid(FIXP_DBL *ptrReal, ENV_CALC_NRGS *nrgs, + UCHAR *ptrHarmIndex, int lowSubbands, + int noSubbands, int scale_change, + int noNoiseFlag, int *ptrPhaseIndex, + int scale_diff_low); + +static void adjustTimeSlotLC(FIXP_DBL *ptrReal, ENV_CALC_NRGS *nrgs, + UCHAR *ptrHarmIndex, int lowSubbands, + int noSubbands, int scale_change, int noNoiseFlag, + int *ptrPhaseIndex); + +/** + * \brief Variant of adjustTimeSlotHQ() which only regards gain and noise but no + * additional harmonics + */ +static void adjustTimeSlotHQ_GainAndNoise( + FIXP_DBL *ptrReal, FIXP_DBL *ptrImag, + HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, ENV_CALC_NRGS *nrgs, + int lowSubbands, int noSubbands, int scale_change, FIXP_SGL smooth_ratio, + int noNoiseFlag, int filtBufferNoiseShift); +/** + * \brief Variant of adjustTimeSlotHQ() which only adds the additional harmonics + */ +static void adjustTimeSlotHQ_AddHarmonics( + FIXP_DBL *ptrReal, FIXP_DBL *ptrImag, + HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, ENV_CALC_NRGS *nrgs, + int lowSubbands, int noSubbands, int scale_change); + +static void adjustTimeSlotHQ(FIXP_DBL *ptrReal, FIXP_DBL *ptrImag, + HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, + ENV_CALC_NRGS *nrgs, int lowSubbands, + int noSubbands, int scale_change, + FIXP_SGL smooth_ratio, int noNoiseFlag, + int filtBufferNoiseShift); + +/*! + \brief Map sine flags from bitstream to QMF bands + + The bitstream carries only 1 sine flag per band (Sfb) and frame. + This function maps every sine flag from the bitstream to a specific QMF + subband and to a specific envelope where the sine shall start. The result is + stored in the vector sineMapped which contains one entry per QMF subband. The + value of an entry specifies the envelope where a sine shall start. A value of + 32 indicates that no sine is present in the subband. The missing harmonics + flags from the previous frame (harmFlagsPrev) determine if a sine starts at + the beginning of the frame or at the transient position. Additionally, the + flags in harmFlagsPrev are being updated by this function for the next frame. +*/ +static void mapSineFlags( + UCHAR *freqBandTable, /*!< Band borders (there's only 1 flag per band) */ + int nSfb, /*!< Number of bands in the table */ + ULONG *addHarmonics, /*!< Packed addHarmonics of current frame (aligned to + the MSB) */ + ULONG *harmFlagsPrev, /*!< Packed addHarmonics of previous frame (aligned to + the LSB) */ + ULONG *harmFlagsPrevActive, /*!< Packed sineMapped of previous frame + (aligned to the LSB) */ + int tranEnv, /*!< Transient position */ + SCHAR *sineMapped) /*!< Resulting vector of sine start positions for each + QMF band */ + +{ + int i; + int bitcount = 31; + ULONG harmFlagsQmfBands[ADD_HARMONICS_FLAGS_SIZE] = {0}; + ULONG *curFlags = addHarmonics; + + /* + Format of addHarmonics (aligned to MSB): + + Up to MAX_FREQ_COEFFS sfb bands can be flagged for a sign. + first word = flags for lowest 32 sfb bands in use + second word = flags for higest 32 sfb bands (if present) + + Format of harmFlagsPrev (aligned to LSB): + + Index is absolute (not relative to lsb) so it is correct even if lsb + changes first word = flags for lowest 32 qmf bands (0...31) second word = + flags for next higher 32 qmf bands (32...63) + + */ + + /* Reset the output vector first */ + FDKmemset(sineMapped, 32, + MAX_FREQ_COEFFS * sizeof(SCHAR)); /* 32 means 'no sine' */ + FDKmemclear(harmFlagsPrevActive, ADD_HARMONICS_FLAGS_SIZE * sizeof(ULONG)); + for (i = 0; i < nSfb; i++) { + ULONG maskSfb = + 1 << bitcount; /* mask to extract addHarmonics flag of current Sfb */ + + if (*curFlags & maskSfb) { /* There is a sine in this band */ + const int lsb = freqBandTable[0]; /* start of sbr range */ + /* qmf band to which sine should be added */ + const int qmfBand = (freqBandTable[i] + freqBandTable[i + 1]) >> 1; + const int qmfBandDiv32 = qmfBand >> 5; + const int maskQmfBand = + 1 << (qmfBand & + 31); /* mask to extract harmonic flag from prevFlags */ + + /* mapping of sfb with sine to a certain qmf band -> for harmFlagsPrev */ + harmFlagsQmfBands[qmfBandDiv32] |= maskQmfBand; + + /* + If there was a sine in the last frame, let it continue from the first + envelope on else start at the transient position. Indexing of sineMapped + starts relative to lsb. + */ + sineMapped[qmfBand - lsb] = + (harmFlagsPrev[qmfBandDiv32] & maskQmfBand) ? 0 : tranEnv; + if (sineMapped[qmfBand - lsb] < PVC_NTIMESLOT) { + harmFlagsPrevActive[qmfBandDiv32] |= maskQmfBand; + } + } + + if (bitcount-- == 0) { + bitcount = 31; + curFlags++; + } + } + FDKmemcpy(harmFlagsPrev, harmFlagsQmfBands, + sizeof(ULONG) * ADD_HARMONICS_FLAGS_SIZE); +} + +/*! + \brief Restore sineMapped of previous frame + + For PVC it might happen that the PVC framing (always 0) is out of sync with + the SBR framing. The adding of additional harmonics is done based on the SBR + framing. If the SBR framing is trailing the PVC framing the sine mapping of + the previous SBR frame needs to be used for the overlapping time slots. +*/ +/*static*/ void mapSineFlagsPvc( + UCHAR *freqBandTable, /*!< Band borders (there's only 1 flag per + band) */ + int nSfb, /*!< Number of bands in the table */ + ULONG *harmFlagsPrev, /*!< Packed addHarmonics of previous frame + (aligned to the MSB) */ + ULONG *harmFlagsPrevActive, /*!< Packed sineMapped of previous + frame (aligned to the LSB) */ + SCHAR *sineMapped, /*!< Resulting vector of sine start positions + for each QMF band */ + int sinusoidalPos, /*!< sinusoidal position */ + SCHAR *sinusoidalPosPrev, /*!< sinusoidal position of previous + frame */ + int trailingSbrFrame) /*!< indication if the SBR framing is + trailing the PVC framing */ +{ + /* Reset the output vector first */ + FDKmemset(sineMapped, 32, MAX_FREQ_COEFFS); /* 32 means 'no sine' */ + + if (trailingSbrFrame) { + /* restore sineMapped[] of previous frame */ + int i; + const int lsb = freqBandTable[0]; + const int usb = freqBandTable[nSfb]; + for (i = lsb; i < usb; i++) { + const int qmfBandDiv32 = i >> 5; + const int maskQmfBand = + 1 << (i & 31); /* mask to extract harmonic flag from prevFlags */ + + /* Two cases need to be distinguished ... */ + if (harmFlagsPrevActive[qmfBandDiv32] & maskQmfBand) { + /* the sine mapping already started last PVC frame -> seamlessly + * continue */ + sineMapped[i - lsb] = 0; + } else if (harmFlagsPrev[qmfBandDiv32] & maskQmfBand) { + /* sinusoidalPos of prev PVC frame was >= PVC_NTIMESLOT -> sine starts + * in this frame */ + sineMapped[i - lsb] = + *sinusoidalPosPrev - PVC_NTIMESLOT; /* we are 16 sbr time slots + ahead of last frame now */ + } + } + } + *sinusoidalPosPrev = sinusoidalPos; +} + +/*! + \brief Reduce gain-adjustment induced aliasing for real valued filterbank. +*/ +/*static*/ void aliasingReduction( + FIXP_DBL *degreeAlias, /*!< estimated aliasing for each QMF + channel */ + ENV_CALC_NRGS *nrgs, + UCHAR *useAliasReduction, /*!< synthetic sine energy for each + subband, used as flag */ + int noSubbands) /*!< number of QMF channels to process */ +{ + FIXP_DBL *nrgGain = nrgs->nrgGain; /*!< subband gains to be modified */ + SCHAR *nrgGain_e = + nrgs->nrgGain_e; /*!< subband gains to be modified (exponents) */ + FIXP_DBL *nrgEst = nrgs->nrgEst; /*!< subband energy before amplification */ + SCHAR *nrgEst_e = + nrgs->nrgEst_e; /*!< subband energy before amplification (exponents) */ + int grouping = 0, index = 0, noGroups, k; + int groupVector[MAX_FREQ_COEFFS]; + + /* Calculate grouping*/ + for (k = 0; k < noSubbands - 1; k++) { + if ((degreeAlias[k + 1] != FL2FXCONST_DBL(0.0f)) && useAliasReduction[k]) { + if (grouping == 0) { + groupVector[index++] = k; + grouping = 1; + } else { + if (groupVector[index - 1] + 3 == k) { + groupVector[index++] = k + 1; + grouping = 0; + } + } + } else { + if (grouping) { + if (useAliasReduction[k]) + groupVector[index++] = k + 1; + else + groupVector[index++] = k; + grouping = 0; + } + } + } + + if (grouping) { + groupVector[index++] = noSubbands; + } + noGroups = index >> 1; + + /*Calculate new gain*/ + for (int group = 0; group < noGroups; group++) { + FIXP_DBL nrgOrig = FL2FXCONST_DBL( + 0.0f); /* Original signal energy in current group of bands */ + SCHAR nrgOrig_e = 0; + FIXP_DBL nrgAmp = FL2FXCONST_DBL( + 0.0f); /* Amplified signal energy in group (using current gains) */ + SCHAR nrgAmp_e = 0; + FIXP_DBL nrgMod = FL2FXCONST_DBL( + 0.0f); /* Signal energy in group when applying modified gains */ + SCHAR nrgMod_e = 0; + FIXP_DBL groupGain; /* Total energy gain in group */ + SCHAR groupGain_e; + FIXP_DBL compensation; /* Compensation factor for the energy change when + applying modified gains */ + SCHAR compensation_e; + + int startGroup = groupVector[2 * group]; + int stopGroup = groupVector[2 * group + 1]; + + /* Calculate total energy in group before and after amplification with + * current gains: */ + for (k = startGroup; k < stopGroup; k++) { + /* Get original band energy */ + FIXP_DBL tmp = nrgEst[k]; + SCHAR tmp_e = nrgEst_e[k]; + + FDK_add_MantExp(tmp, tmp_e, nrgOrig, nrgOrig_e, &nrgOrig, &nrgOrig_e); + + /* Multiply band energy with current gain */ + tmp = fMult(tmp, nrgGain[k]); + tmp_e = tmp_e + nrgGain_e[k]; + + FDK_add_MantExp(tmp, tmp_e, nrgAmp, nrgAmp_e, &nrgAmp, &nrgAmp_e); + } + + /* Calculate total energy gain in group */ + FDK_divide_MantExp(nrgAmp, nrgAmp_e, nrgOrig, nrgOrig_e, &groupGain, + &groupGain_e); + + for (k = startGroup; k < stopGroup; k++) { + FIXP_DBL tmp; + SCHAR tmp_e; + + FIXP_DBL alpha = degreeAlias[k]; + if (k < noSubbands - 1) { + if (degreeAlias[k + 1] > alpha) alpha = degreeAlias[k + 1]; + } + + /* Modify gain depending on the degree of aliasing */ + FDK_add_MantExp( + fMult(alpha, groupGain), groupGain_e, + fMult(/*FL2FXCONST_DBL(1.0f)*/ (FIXP_DBL)MAXVAL_DBL - alpha, + nrgGain[k]), + nrgGain_e[k], &nrgGain[k], &nrgGain_e[k]); + + /* Apply modified gain to original energy */ + tmp = fMult(nrgGain[k], nrgEst[k]); + tmp_e = nrgGain_e[k] + nrgEst_e[k]; + + /* Accumulate energy with modified gains applied */ + FDK_add_MantExp(tmp, tmp_e, nrgMod, nrgMod_e, &nrgMod, &nrgMod_e); + } + + /* Calculate compensation factor to retain the energy of the amplified + * signal */ + FDK_divide_MantExp(nrgAmp, nrgAmp_e, nrgMod, nrgMod_e, &compensation, + &compensation_e); + + /* Apply compensation factor to all gains of the group */ + for (k = startGroup; k < stopGroup; k++) { + nrgGain[k] = fMult(nrgGain[k], compensation); + nrgGain_e[k] = nrgGain_e[k] + compensation_e; + } + } +} + +#define INTER_TES_SF_CHANGE 3 + +typedef struct { + FIXP_DBL subsample_power_low[(((1024) / (32) * (4) / 2) + (3 * (4)))]; + FIXP_DBL subsample_power_high[(((1024) / (32) * (4) / 2) + (3 * (4)))]; + FIXP_DBL gain[(((1024) / (32) * (4) / 2) + (3 * (4)))]; + SCHAR subsample_power_low_sf[(((1024) / (32) * (4) / 2) + (3 * (4)))]; + SCHAR subsample_power_high_sf[(((1024) / (32) * (4) / 2) + (3 * (4)))]; +} ITES_TEMP; + +static void apply_inter_tes(FIXP_DBL **qmfReal, FIXP_DBL **qmfImag, + const QMF_SCALE_FACTOR *sbrScaleFactor, + const SCHAR exp[2], const int RATE, + const int startPos, const int stopPos, + const int lowSubband, const int nbSubband, + const UCHAR gamma_idx) { + int highSubband = lowSubband + nbSubband; + FIXP_DBL *subsample_power_high, *subsample_power_low; + SCHAR *subsample_power_high_sf, *subsample_power_low_sf; + FIXP_DBL total_power_high = (FIXP_DBL)0; + FIXP_DBL total_power_low = (FIXP_DBL)0; + FIXP_DBL *gain; + int gain_sf[(((1024) / (32) * (4) / 2) + (3 * (4)))]; + + /* gamma[gamma_idx] = {0.0f, 1.0f, 2.0f, 4.0f} */ + int gamma_sf = + (int)gamma_idx - 1; /* perhaps +1 to save one bit? (0.99999f vs 1.f) */ + + int nbSubsample = stopPos - startPos; + int i, j; + + C_ALLOC_SCRATCH_START(pTmp, ITES_TEMP, 1); + subsample_power_high = pTmp->subsample_power_high; + subsample_power_low = pTmp->subsample_power_low; + subsample_power_high_sf = pTmp->subsample_power_high_sf; + subsample_power_low_sf = pTmp->subsample_power_low_sf; + gain = pTmp->gain; + + if (gamma_idx > 0) { + int preShift2 = 32 - fNormz((FIXP_DBL)nbSubsample); + int total_power_low_sf = 1 - DFRACT_BITS; + int total_power_high_sf = 1 - DFRACT_BITS; + + for (i = 0; i < nbSubsample; ++i) { + FIXP_DBL bufferReal[(((1024) / (32) * (4) / 2) + (3 * (4)))]; + FIXP_DBL bufferImag[(((1024) / (32) * (4) / 2) + (3 * (4)))]; + FIXP_DBL maxVal = (FIXP_DBL)0; + + int ts = startPos + i; + + int low_sf = (ts < 3 * RATE) ? sbrScaleFactor->ov_lb_scale + : sbrScaleFactor->lb_scale; + low_sf = 15 - low_sf; + + for (j = 0; j < lowSubband; ++j) { + bufferImag[j] = qmfImag[startPos + i][j]; + maxVal |= (FIXP_DBL)((LONG)(bufferImag[j]) ^ + ((LONG)bufferImag[j] >> (DFRACT_BITS - 1))); + bufferReal[j] = qmfReal[startPos + i][j]; + maxVal |= (FIXP_DBL)((LONG)(bufferReal[j]) ^ + ((LONG)bufferReal[j] >> (DFRACT_BITS - 1))); + } + + subsample_power_low[i] = (FIXP_DBL)0; + subsample_power_low_sf[i] = 0; + + if (maxVal != FL2FXCONST_DBL(0.f)) { + /* multiply first, then shift for safe summation */ + int preShift = 1 - CntLeadingZeros(maxVal); + int postShift = 32 - fNormz((FIXP_DBL)lowSubband); + + /* reduce preShift because otherwise we risk to square -1.f */ + if (preShift != 0) preShift++; + + subsample_power_low_sf[i] += (low_sf + preShift) * 2 + postShift + 1; + + scaleValues(bufferReal, lowSubband, -preShift); + scaleValues(bufferImag, lowSubband, -preShift); + for (j = 0; j < lowSubband; ++j) { + FIXP_DBL addme; + addme = fPow2Div2(bufferReal[j]); + subsample_power_low[i] += addme >> postShift; + addme = fPow2Div2(bufferImag[j]); + subsample_power_low[i] += addme >> postShift; + } + } + + /* now get high */ + + maxVal = (FIXP_DBL)0; + + int high_sf = exp[(ts < 16 * RATE) ? 0 : 1]; + + for (j = lowSubband; j < highSubband; ++j) { + bufferImag[j] = qmfImag[startPos + i][j]; + maxVal |= (FIXP_DBL)((LONG)(bufferImag[j]) ^ + ((LONG)bufferImag[j] >> (DFRACT_BITS - 1))); + bufferReal[j] = qmfReal[startPos + i][j]; + maxVal |= (FIXP_DBL)((LONG)(bufferReal[j]) ^ + ((LONG)bufferReal[j] >> (DFRACT_BITS - 1))); + } + + subsample_power_high[i] = (FIXP_DBL)0; + subsample_power_high_sf[i] = 0; + + if (maxVal != FL2FXCONST_DBL(0.f)) { + int preShift = 1 - CntLeadingZeros(maxVal); + /* reduce preShift because otherwise we risk to square -1.f */ + if (preShift != 0) preShift++; + + int postShift = 32 - fNormz((FIXP_DBL)(highSubband - lowSubband)); + subsample_power_high_sf[i] += (high_sf + preShift) * 2 + postShift + 1; + + scaleValues(&bufferReal[lowSubband], highSubband - lowSubband, + -preShift); + scaleValues(&bufferImag[lowSubband], highSubband - lowSubband, + -preShift); + for (j = lowSubband; j < highSubband; j++) { + subsample_power_high[i] += fPow2Div2(bufferReal[j]) >> postShift; + subsample_power_high[i] += fPow2Div2(bufferImag[j]) >> postShift; + } + } + + /* sum all together */ + FIXP_DBL new_summand = subsample_power_low[i]; + int new_summand_sf = subsample_power_low_sf[i]; + + /* make sure the current sum, and the new summand have the same SF */ + if (new_summand_sf > total_power_low_sf) { + int diff = fMin(DFRACT_BITS - 1, new_summand_sf - total_power_low_sf); + total_power_low >>= diff; + total_power_low_sf = new_summand_sf; + } else if (new_summand_sf < total_power_low_sf) { + new_summand >>= + fMin(DFRACT_BITS - 1, total_power_low_sf - new_summand_sf); + } + + total_power_low += (new_summand >> preShift2); + + new_summand = subsample_power_high[i]; + new_summand_sf = subsample_power_high_sf[i]; + if (new_summand_sf > total_power_high_sf) { + total_power_high >>= + fMin(DFRACT_BITS - 1, new_summand_sf - total_power_high_sf); + total_power_high_sf = new_summand_sf; + } else if (new_summand_sf < total_power_high_sf) { + new_summand >>= + fMin(DFRACT_BITS - 1, total_power_high_sf - new_summand_sf); + } + + total_power_high += (new_summand >> preShift2); + } + + total_power_low_sf += preShift2; + total_power_high_sf += preShift2; + + /* gain[i] = e_LOW[i] */ + for (i = 0; i < nbSubsample; ++i) { + int sf2; + FIXP_DBL mult = + fMultNorm(subsample_power_low[i], (FIXP_DBL)nbSubsample, &sf2); + int mult_sf = subsample_power_low_sf[i] + DFRACT_BITS - 1 + sf2; + + if (total_power_low != FIXP_DBL(0)) { + gain[i] = fDivNorm(mult, total_power_low, &sf2); + gain_sf[i] = mult_sf - total_power_low_sf + sf2; + gain[i] = sqrtFixp_lookup(gain[i], &gain_sf[i]); + if (gain_sf[i] < 0) { + gain[i] >>= -gain_sf[i]; + gain_sf[i] = 0; + } + } else { + if (mult == FIXP_DBL(0)) { + gain[i] = FIXP_DBL(0); + gain_sf[i] = 0; + } else { + gain[i] = (FIXP_DBL)MAXVAL_DBL; + gain_sf[i] = 0; + } + } + } + + FIXP_DBL total_power_high_after = (FIXP_DBL)0; + int total_power_high_after_sf = 1 - DFRACT_BITS; + + /* gain[i] = g_inter[i] */ + for (i = 0; i < nbSubsample; ++i) { + if (gain_sf[i] < 0) { + gain[i] >>= -gain_sf[i]; + gain_sf[i] = 0; + } + + /* calculate: gain[i] = 1.0f + gamma * (gain[i] - 1.0f); */ + FIXP_DBL one = (FIXP_DBL)MAXVAL_DBL >> + gain_sf[i]; /* to substract this from gain[i] */ + + /* gamma is actually always 1 according to the table, so skip the + * fMultDiv2 */ + FIXP_DBL mult = (gain[i] - one) >> 1; + int mult_sf = gain_sf[i] + gamma_sf; + + one = FL2FXCONST_DBL(0.5f) >> mult_sf; + gain[i] = one + mult; + gain_sf[i] += gamma_sf + 1; /* +1 because of fMultDiv2() */ + + /* set gain to at least 0.2f */ + FIXP_DBL point_two = FL2FXCONST_DBL(0.8f); /* scaled up by 2 */ + int point_two_sf = -2; + + FIXP_DBL tmp = gain[i]; + if (point_two_sf < gain_sf[i]) { + point_two >>= gain_sf[i] - point_two_sf; + } else { + tmp >>= point_two_sf - gain_sf[i]; + } + + /* limit and calculate gain[i]^2 too */ + FIXP_DBL gain_pow2; + int gain_pow2_sf; + if (tmp < point_two) { + gain[i] = FL2FXCONST_DBL(0.8f); + gain_sf[i] = -2; + gain_pow2 = FL2FXCONST_DBL(0.64f); + gain_pow2_sf = -4; + } else { + /* this upscaling seems quite important */ + int r = CountLeadingBits(gain[i]); + gain[i] <<= r; + gain_sf[i] -= r; + + gain_pow2 = fPow2(gain[i]); + gain_pow2_sf = gain_sf[i] << 1; + } + + int room; + subsample_power_high[i] = + fMultNorm(subsample_power_high[i], gain_pow2, &room); + subsample_power_high_sf[i] = + subsample_power_high_sf[i] + gain_pow2_sf + room; + + int new_summand_sf = subsample_power_high_sf[i]; /* + gain_pow2_sf; */ + if (new_summand_sf > total_power_high_after_sf) { + total_power_high_after >>= + fMin(DFRACT_BITS - 1, new_summand_sf - total_power_high_after_sf); + total_power_high_after_sf = new_summand_sf; + } else if (new_summand_sf < total_power_high_after_sf) { + subsample_power_high[i] >>= total_power_high_after_sf - new_summand_sf; + } + total_power_high_after += subsample_power_high[i] >> preShift2; + } + + total_power_high_after_sf += preShift2; + + int sf2 = 0; + FIXP_DBL gain_adj_2 = FL2FX_DBL(0.5f); + int gain_adj_2_sf = 1; + + if ((total_power_high != (FIXP_DBL)0) && + (total_power_high_after != (FIXP_DBL)0)) { + gain_adj_2 = fDivNorm(total_power_high, total_power_high_after, &sf2); + gain_adj_2_sf = total_power_high_sf - total_power_high_after_sf + sf2; + } + + FIXP_DBL gain_adj = sqrtFixp_lookup(gain_adj_2, &gain_adj_2_sf); + int gain_adj_sf = gain_adj_2_sf; + + for (i = 0; i < nbSubsample; ++i) { + gain[i] = fMult(gain[i], gain_adj); + gain_sf[i] += gain_adj_sf; + + /* limit gain */ + if (gain_sf[i] > INTER_TES_SF_CHANGE) { + gain[i] = (FIXP_DBL)MAXVAL_DBL; + gain_sf[i] = INTER_TES_SF_CHANGE; + } + } + + for (i = 0; i < nbSubsample; ++i) { + /* equalize gain[]'s scale factors */ + gain[i] >>= INTER_TES_SF_CHANGE - gain_sf[i]; + + for (j = lowSubband; j < highSubband; j++) { + qmfReal[startPos + i][j] = fMult(qmfReal[startPos + i][j], gain[i]); + qmfImag[startPos + i][j] = fMult(qmfImag[startPos + i][j], gain[i]); + } + } + } else { /* gamma_idx == 0 */ + /* Inter-TES is not active. Still perform the scale change to have a + * consistent scaling for all envelopes of this frame. */ + for (i = 0; i < nbSubsample; ++i) { + for (j = lowSubband; j < highSubband; j++) { + qmfReal[startPos + i][j] >>= INTER_TES_SF_CHANGE; + qmfImag[startPos + i][j] >>= INTER_TES_SF_CHANGE; + } + } + } + C_ALLOC_SCRATCH_END(pTmp, ITES_TEMP, 1); +} + +/*! + \brief Apply spectral envelope to subband samples + + This function is called from sbr_dec.cpp in each frame. + + To enhance accuracy and due to the usage of tables for squareroots and + inverse, some calculations are performed with the operands being split + into mantissa and exponent. The variable names in the source code carry + the suffixes <em>_m</em> and <em>_e</em> respectively. The control data + in #hFrameData containts envelope data which is represented by this format but + stored in single words. (See requantizeEnvelopeData() for details). This data + is unpacked within calculateSbrEnvelope() to follow the described suffix + convention. + + The actual value (comparable to the corresponding float-variable in the + research-implementation) of a mantissa/exponent-pair can be calculated as + + \f$ value = value\_m * 2^{value\_e} \f$ + + All energies and noise levels decoded from the bitstream suit for an + original signal magnitude of \f$\pm 32768 \f$ rather than \f$ \pm 1\f$. + Therefore, the scale factor <em>hb_scale</em> passed into this function will + be converted to an 'input exponent' (#input_e), which fits the internal + representation. + + Before the actual processing, an exponent #adj_e for resulting adjusted + samples is derived from the maximum reference energy. + + Then, for each envelope, the following steps are performed: + + \li Calculate energy in the signal to be adjusted. Depending on the the value + of #interpolFreq (interpolation mode), this is either done seperately for each + QMF-subband or for each SBR-band. The resulting energies are stored in + #nrgEst_m[#MAX_FREQ_COEFFS] (mantissas) and #nrgEst_e[#MAX_FREQ_COEFFS] + (exponents). \li Calculate gain and noise level for each subband:<br> \f$ gain + = \sqrt{ \frac{nrgRef}{nrgEst} \cdot (1 - noiseRatio) } \hspace{2cm} noise = + \sqrt{ nrgRef \cdot noiseRatio } \f$<br> where <em>noiseRatio</em> and + <em>nrgRef</em> are extracted from the bitstream and <em>nrgEst</em> is the + subband energy before adjustment. The resulting gains are stored in + #nrgGain_m[#MAX_FREQ_COEFFS] (mantissas) and #nrgGain_e[#MAX_FREQ_COEFFS] + (exponents), the noise levels are stored in #noiseLevel_m[#MAX_FREQ_COEFFS] + and #noiseLevel_e[#MAX_FREQ_COEFFS] (exponents). The sine levels are stored in + #nrgSine_m[#MAX_FREQ_COEFFS] and #nrgSine_e[#MAX_FREQ_COEFFS]. \li Noise + limiting: The gain for each subband is limited both absolutely and relatively + compared to the total gain over all subbands. \li Boost gain: Calculate and + apply boost factor for each limiter band in order to compensate for the energy + loss imposed by the limiting. \li Apply gains and add noise: The gains and + noise levels are applied to all timeslots of the current envelope. A short + FIR-filter (length 4 QMF-timeslots) can be used to smooth the sudden change at + the envelope borders. Each complex subband sample of the current timeslot is + multiplied by the smoothed gain, then random noise with the calculated level + is added. + + \note + To reduce the stack size, some of the local arrays could be located within + the time output buffer. Of the 512 samples temporarily available there, + about half the size is already used by #SBR_FRAME_DATA. A pointer to the + remaining free memory could be supplied by an additional argument to + calculateSbrEnvelope() in sbr_dec: + + \par + \code + calculateSbrEnvelope (&hSbrDec->sbrScaleFactor, + &hSbrDec->SbrCalculateEnvelope, + hHeaderData, + hFrameData, + QmfBufferReal, + QmfBufferImag, + timeOutPtr + sizeof(SBR_FRAME_DATA)/sizeof(Float) + + 1); \endcode + + \par + Within calculateSbrEnvelope(), some pointers could be defined instead of the + arrays #nrgRef_m, #nrgRef_e, #nrgEst_m, #nrgEst_e, #noiseLevel_m: + + \par + \code + fract* nrgRef_m = timeOutPtr; + SCHAR* nrgRef_e = nrgRef_m + MAX_FREQ_COEFFS; + fract* nrgEst_m = nrgRef_e + MAX_FREQ_COEFFS; + SCHAR* nrgEst_e = nrgEst_m + MAX_FREQ_COEFFS; + fract* noiseLevel_m = nrgEst_e + MAX_FREQ_COEFFS; + \endcode + + <br> +*/ +void calculateSbrEnvelope( + QMF_SCALE_FACTOR *sbrScaleFactor, /*!< Scaling factors */ + HANDLE_SBR_CALCULATE_ENVELOPE + h_sbr_cal_env, /*!< Handle to struct filled by the create-function */ + HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */ + HANDLE_SBR_FRAME_DATA hFrameData, /*!< Control data of current frame */ + PVC_DYNAMIC_DATA *pPvcDynamicData, + FIXP_DBL * + *analysBufferReal, /*!< Real part of subband samples to be processed */ + FIXP_DBL * + *analysBufferImag, /*!< Imag part of subband samples to be processed */ + const int useLP, + FIXP_DBL *degreeAlias, /*!< Estimated aliasing for each QMF channel */ + const UINT flags, const int frameErrorFlag) { + int c, i, i_stop, j, envNoise = 0; + UCHAR *borders = hFrameData->frameInfo.borders; + UCHAR *bordersPvc = hFrameData->frameInfo.pvcBorders; + int pvc_mode = pPvcDynamicData->pvc_mode; + int first_start = + ((pvc_mode > 0) ? bordersPvc[0] : borders[0]) * hHeaderData->timeStep; + FIXP_SGL *noiseLevels = hFrameData->sbrNoiseFloorLevel; + HANDLE_FREQ_BAND_DATA hFreq = &hHeaderData->freqBandData; + UCHAR **pFreqBandTable = hFreq->freqBandTable; + UCHAR *pFreqBandTableNoise = hFreq->freqBandTableNoise; + + int lowSubband = hFreq->lowSubband; + int highSubband = hFreq->highSubband; + int noSubbands = highSubband - lowSubband; + + /* old high subband before headerchange + we asume no headerchange here */ + int ov_highSubband = hFreq->highSubband; + + int noNoiseBands = hFreq->nNfb; + UCHAR *noSubFrameBands = hFreq->nSfb; + int no_cols = hHeaderData->numberTimeSlots * hHeaderData->timeStep; + + SCHAR sineMapped[MAX_FREQ_COEFFS]; + SCHAR ov_adj_e = SCALE2EXP(sbrScaleFactor->ov_hb_scale); + SCHAR adj_e = 0; + SCHAR output_e; + SCHAR final_e = 0; + /* inter-TES is active in one or more envelopes of the current SBR frame */ + const int iTES_enable = hFrameData->iTESactive; + const int iTES_scale_change = (iTES_enable) ? INTER_TES_SF_CHANGE : 0; + SCHAR maxGainLimit_e = (frameErrorFlag) ? MAX_GAIN_CONCEAL_EXP : MAX_GAIN_EXP; + + UCHAR smooth_length = 0; + + FIXP_SGL *pIenv = hFrameData->iEnvelope; + + C_ALLOC_SCRATCH_START(useAliasReduction, UCHAR, 64) + + /* if values differ we had a headerchange; if old highband is bigger then new + one we need to patch overlap-highband-scaling for this frame (see use of + ov_highSubband) as overlap contains higher frequency components which would + get lost */ + if (hFreq->highSubband < hFreq->ov_highSubband) { + ov_highSubband = hFreq->ov_highSubband; + } + + if (pvc_mode > 0) { + if (hFrameData->frameInfo.bordersNoise[0] > bordersPvc[0]) { + /* noise envelope of previous frame is trailing into current PVC frame */ + envNoise = -1; + noiseLevels = h_sbr_cal_env->prevSbrNoiseFloorLevel; + noNoiseBands = h_sbr_cal_env->prevNNfb; + noSubFrameBands = h_sbr_cal_env->prevNSfb; + lowSubband = h_sbr_cal_env->prevLoSubband; + highSubband = h_sbr_cal_env->prevHiSubband; + + noSubbands = highSubband - lowSubband; + ov_highSubband = highSubband; + if (highSubband < h_sbr_cal_env->prev_ov_highSubband) { + ov_highSubband = h_sbr_cal_env->prev_ov_highSubband; + } + + pFreqBandTable[0] = h_sbr_cal_env->prevFreqBandTableLo; + pFreqBandTable[1] = h_sbr_cal_env->prevFreqBandTableHi; + pFreqBandTableNoise = h_sbr_cal_env->prevFreqBandTableNoise; + } + + mapSineFlagsPvc(pFreqBandTable[1], noSubFrameBands[1], + h_sbr_cal_env->harmFlagsPrev, + h_sbr_cal_env->harmFlagsPrevActive, sineMapped, + hFrameData->sinusoidal_position, + &h_sbr_cal_env->sinusoidal_positionPrev, + (borders[0] > bordersPvc[0]) ? 1 : 0); + } else { + /* + Extract sine flags for all QMF bands + */ + mapSineFlags(pFreqBandTable[1], noSubFrameBands[1], + hFrameData->addHarmonics, h_sbr_cal_env->harmFlagsPrev, + h_sbr_cal_env->harmFlagsPrevActive, + hFrameData->frameInfo.tranEnv, sineMapped); + } + + /* + Scan for maximum in bufferd noise levels. + This is needed in case that we had strong noise in the previous frame + which is smoothed into the current frame. + The resulting exponent is used as start value for the maximum search + in reference energies + */ + if (!useLP) + adj_e = h_sbr_cal_env->filtBufferNoise_e - + getScalefactor(h_sbr_cal_env->filtBufferNoise, noSubbands); + + /* + Scan for maximum reference energy to be able + to select appropriate values for adj_e and final_e. + */ + if (pvc_mode > 0) { + INT maxSfbNrg_e = pPvcDynamicData->predEsg_expMax; + + /* Energy -> magnitude (sqrt halfens exponent) */ + maxSfbNrg_e = + (maxSfbNrg_e + 1) >> 1; /* +1 to go safe (round to next higher int) */ + + /* Some safety margin is needed for 2 reasons: + - The signal energy is not equally spread over all subband samples in + a specific sfb of an envelope (Nrg could be too high by a factor of + envWidth * sfbWidth) + - Smoothing can smear high gains of the previous envelope into the + current + */ + maxSfbNrg_e += 6; + + adj_e = maxSfbNrg_e; + // final_e should not exist for PVC fixfix framing + } else { + for (i = 0; i < hFrameData->frameInfo.nEnvelopes; i++) { + INT maxSfbNrg_e = + -FRACT_BITS + NRG_EXP_OFFSET; /* start value for maximum search */ + + /* Fetch frequency resolution for current envelope: */ + for (j = noSubFrameBands[hFrameData->frameInfo.freqRes[i]]; j != 0; j--) { + maxSfbNrg_e = fixMax(maxSfbNrg_e, (INT)((LONG)(*pIenv++) & MASK_E)); + } + maxSfbNrg_e -= NRG_EXP_OFFSET; + + /* Energy -> magnitude (sqrt halfens exponent) */ + maxSfbNrg_e = + (maxSfbNrg_e + 1) >> 1; /* +1 to go safe (round to next higher int) */ + + /* Some safety margin is needed for 2 reasons: + - The signal energy is not equally spread over all subband samples in + a specific sfb of an envelope (Nrg could be too high by a factor of + envWidth * sfbWidth) + - Smoothing can smear high gains of the previous envelope into the + current + */ + maxSfbNrg_e += 6; + + if (borders[i] < hHeaderData->numberTimeSlots) + /* This envelope affects timeslots that belong to the output frame */ + adj_e = fMax(maxSfbNrg_e, adj_e); + + if (borders[i + 1] > hHeaderData->numberTimeSlots) + /* This envelope affects timeslots after the output frame */ + final_e = fMax(maxSfbNrg_e, final_e); + } + } + /* + Calculate adjustment factors and apply them for every envelope. + */ + pIenv = hFrameData->iEnvelope; + + if (pvc_mode > 0) { + /* iterate over SBR time slots starting with bordersPvc[i] */ + i = bordersPvc[0]; /* usually 0; can be >0 if switching from legacy SBR to + PVC */ + i_stop = PVC_NTIMESLOT; + FDK_ASSERT(bordersPvc[hFrameData->frameInfo.nEnvelopes] == PVC_NTIMESLOT); + } else { + /* iterate over SBR envelopes starting with 0 */ + i = 0; + i_stop = hFrameData->frameInfo.nEnvelopes; + } + for (; i < i_stop; i++) { + int k, noNoiseFlag; + SCHAR noise_e, input_e = SCALE2EXP(sbrScaleFactor->hb_scale); + C_ALLOC_SCRATCH_START(pNrgs, ENV_CALC_NRGS, 1); + + /* + Helper variables. + */ + int start_pos, stop_pos, freq_res; + if (pvc_mode > 0) { + start_pos = + hHeaderData->timeStep * + i; /* Start-position in time (subband sample) for current envelope. */ + stop_pos = hHeaderData->timeStep * (i + 1); /* Stop-position in time + (subband sample) for + current envelope. */ + freq_res = + hFrameData->frameInfo + .freqRes[0]; /* Frequency resolution for current envelope. */ + FDK_ASSERT( + freq_res == + hFrameData->frameInfo.freqRes[hFrameData->frameInfo.nEnvelopes - 1]); + } else { + start_pos = hHeaderData->timeStep * + borders[i]; /* Start-position in time (subband sample) for + current envelope. */ + stop_pos = hHeaderData->timeStep * + borders[i + 1]; /* Stop-position in time (subband sample) for + current envelope. */ + freq_res = + hFrameData->frameInfo + .freqRes[i]; /* Frequency resolution for current envelope. */ + } + + /* Always fully initialize the temporary energy table. This prevents + negative energies and extreme gain factors in cases where the number of + limiter bands exceeds the number of subbands. The latter can be caused by + undetected bit errors and is tested by some streams from the + certification set. */ + FDKmemclear(pNrgs, sizeof(ENV_CALC_NRGS)); + + if (pvc_mode > 0) { + /* get predicted energy values from PVC module */ + expandPredEsg(pPvcDynamicData, i, (int)MAX_FREQ_COEFFS, pNrgs->nrgRef, + pNrgs->nrgRef_e); + + if (i == borders[0]) { + mapSineFlags(pFreqBandTable[1], noSubFrameBands[1], + hFrameData->addHarmonics, h_sbr_cal_env->harmFlagsPrev, + h_sbr_cal_env->harmFlagsPrevActive, + hFrameData->sinusoidal_position, sineMapped); + } + + if (i >= hFrameData->frameInfo.bordersNoise[envNoise + 1]) { + if (envNoise >= 0) { + noiseLevels += noNoiseBands; /* The noise floor data is stored in a + row [noiseFloor1 noiseFloor2...].*/ + } else { + /* leave trailing noise envelope of past frame */ + noNoiseBands = hFreq->nNfb; + noSubFrameBands = hFreq->nSfb; + noiseLevels = hFrameData->sbrNoiseFloorLevel; + + lowSubband = hFreq->lowSubband; + highSubband = hFreq->highSubband; + + noSubbands = highSubband - lowSubband; + ov_highSubband = highSubband; + if (highSubband < hFreq->ov_highSubband) { + ov_highSubband = hFreq->ov_highSubband; + } + + pFreqBandTable[0] = hFreq->freqBandTableLo; + pFreqBandTable[1] = hFreq->freqBandTableHi; + pFreqBandTableNoise = hFreq->freqBandTableNoise; + } + envNoise++; + } + } else { + /* If the start-pos of the current envelope equals the stop pos of the + current noise envelope, increase the pointer (i.e. choose the next + noise-floor).*/ + if (borders[i] == hFrameData->frameInfo.bordersNoise[envNoise + 1]) { + noiseLevels += noNoiseBands; /* The noise floor data is stored in a row + [noiseFloor1 noiseFloor2...].*/ + envNoise++; + } + } + if (i == hFrameData->frameInfo.tranEnv || + i == h_sbr_cal_env->prevTranEnv) /* attack */ + { + noNoiseFlag = 1; + if (!useLP) smooth_length = 0; /* No smoothing on attacks! */ + } else { + noNoiseFlag = 0; + if (!useLP) + smooth_length = (1 - hHeaderData->bs_data.smoothingLength) + << 2; /* can become either 0 or 4 */ + } + + /* + Energy estimation in transposed highband. + */ + if (hHeaderData->bs_data.interpolFreq) + calcNrgPerSubband(analysBufferReal, (useLP) ? NULL : analysBufferImag, + lowSubband, highSubband, start_pos, stop_pos, input_e, + pNrgs->nrgEst, pNrgs->nrgEst_e); + else + calcNrgPerSfb(analysBufferReal, (useLP) ? NULL : analysBufferImag, + noSubFrameBands[freq_res], pFreqBandTable[freq_res], + start_pos, stop_pos, input_e, pNrgs->nrgEst, + pNrgs->nrgEst_e); + + /* + Calculate subband gains + */ + { + UCHAR *table = pFreqBandTable[freq_res]; + UCHAR *pUiNoise = + &pFreqBandTableNoise[1]; /*! Upper limit of the current noise floor + band. */ + + FIXP_SGL *pNoiseLevels = noiseLevels; + + FIXP_DBL tmpNoise = + FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pNoiseLevels) & MASK_M)); + SCHAR tmpNoise_e = + (UCHAR)((LONG)(*pNoiseLevels++) & MASK_E) - NOISE_EXP_OFFSET; + + int cc = 0; + c = 0; + if (pvc_mode > 0) { + for (j = 0; j < noSubFrameBands[freq_res]; j++) { + UCHAR sinePresentFlag = 0; + int li = table[j]; + int ui = table[j + 1]; + + for (k = li; k < ui; k++) { + sinePresentFlag |= (i >= sineMapped[cc]); + cc++; + } + + for (k = li; k < ui; k++) { + FIXP_DBL refNrg = pNrgs->nrgRef[k - lowSubband]; + SCHAR refNrg_e = pNrgs->nrgRef_e[k - lowSubband]; + + if (k >= *pUiNoise) { + tmpNoise = + FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pNoiseLevels) & MASK_M)); + tmpNoise_e = + (SCHAR)((LONG)(*pNoiseLevels++) & MASK_E) - NOISE_EXP_OFFSET; + + pUiNoise++; + } + + FDK_ASSERT(k >= lowSubband); + + if (useLP) useAliasReduction[k - lowSubband] = !sinePresentFlag; + + pNrgs->nrgSine[c] = FL2FXCONST_DBL(0.0f); + pNrgs->nrgSine_e[c] = 0; + + calcSubbandGain(refNrg, refNrg_e, pNrgs, c, tmpNoise, tmpNoise_e, + sinePresentFlag, i >= sineMapped[c], noNoiseFlag); + + c++; + } + } + } else { + for (j = 0; j < noSubFrameBands[freq_res]; j++) { + FIXP_DBL refNrg = FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pIenv) & MASK_M)); + SCHAR refNrg_e = (SCHAR)((LONG)(*pIenv) & MASK_E) - NRG_EXP_OFFSET; + + UCHAR sinePresentFlag = 0; + int li = table[j]; + int ui = table[j + 1]; + + for (k = li; k < ui; k++) { + sinePresentFlag |= (i >= sineMapped[cc]); + cc++; + } + + for (k = li; k < ui; k++) { + if (k >= *pUiNoise) { + tmpNoise = + FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pNoiseLevels) & MASK_M)); + tmpNoise_e = + (SCHAR)((LONG)(*pNoiseLevels++) & MASK_E) - NOISE_EXP_OFFSET; + + pUiNoise++; + } + + FDK_ASSERT(k >= lowSubband); + + if (useLP) useAliasReduction[k - lowSubband] = !sinePresentFlag; + + pNrgs->nrgSine[c] = FL2FXCONST_DBL(0.0f); + pNrgs->nrgSine_e[c] = 0; + + calcSubbandGain(refNrg, refNrg_e, pNrgs, c, tmpNoise, tmpNoise_e, + sinePresentFlag, i >= sineMapped[c], noNoiseFlag); + + pNrgs->nrgRef[c] = refNrg; + pNrgs->nrgRef_e[c] = refNrg_e; + + c++; + } + pIenv++; + } + } + } + + /* + Noise limiting + */ + + for (c = 0; c < hFreq->noLimiterBands; c++) { + FIXP_DBL sumRef, boostGain, maxGain; + FIXP_DBL accu = FL2FXCONST_DBL(0.0f); + SCHAR sumRef_e, boostGain_e, maxGain_e, accu_e = 0; + int maxGainLimGainSum_e = 0; + + calcAvgGain(pNrgs, hFreq->limiterBandTable[c], + hFreq->limiterBandTable[c + 1], &sumRef, &sumRef_e, &maxGain, + &maxGain_e); + + /* Multiply maxGain with limiterGain: */ + maxGain = fMult( + maxGain, + FDK_sbrDecoder_sbr_limGains_m[hHeaderData->bs_data.limiterGains]); + /* maxGain_e += + * FDK_sbrDecoder_sbr_limGains_e[hHeaderData->bs_data.limiterGains]; */ + /* The addition of maxGain_e and FDK_sbrDecoder_sbr_limGains_e[3] might + yield values greater than 127 which doesn't fit into an SCHAR! In these + rare situations limit maxGain_e to 127. + */ + maxGainLimGainSum_e = + maxGain_e + + FDK_sbrDecoder_sbr_limGains_e[hHeaderData->bs_data.limiterGains]; + maxGain_e = + (maxGainLimGainSum_e > 127) ? (SCHAR)127 : (SCHAR)maxGainLimGainSum_e; + + /* Scale mantissa of MaxGain into range between 0.5 and 1: */ + if (maxGain == FL2FXCONST_DBL(0.0f)) + maxGain_e = -FRACT_BITS; + else { + SCHAR charTemp = CountLeadingBits(maxGain); + maxGain_e -= charTemp; + maxGain <<= (int)charTemp; + } + + if (maxGain_e >= maxGainLimit_e) { /* upper limit (e.g. 96 dB) */ + maxGain = FL2FXCONST_DBL(0.5f); + maxGain_e = maxGainLimit_e; + } + + /* Every subband gain is compared to the scaled "average gain" + and limited if necessary: */ + for (k = hFreq->limiterBandTable[c]; k < hFreq->limiterBandTable[c + 1]; + k++) { + if ((pNrgs->nrgGain_e[k] > maxGain_e) || + (pNrgs->nrgGain_e[k] == maxGain_e && pNrgs->nrgGain[k] > maxGain)) { + FIXP_DBL noiseAmp; + SCHAR noiseAmp_e; + + FDK_divide_MantExp(maxGain, maxGain_e, pNrgs->nrgGain[k], + pNrgs->nrgGain_e[k], &noiseAmp, &noiseAmp_e); + pNrgs->noiseLevel[k] = fMult(pNrgs->noiseLevel[k], noiseAmp); + pNrgs->noiseLevel_e[k] += noiseAmp_e; + pNrgs->nrgGain[k] = maxGain; + pNrgs->nrgGain_e[k] = maxGain_e; + } + } + + /* -- Boost gain + Calculate and apply boost factor for each limiter band: + 1. Check how much energy would be present when using the limited gain + 2. Calculate boost factor by comparison with reference energy + 3. Apply boost factor to compensate for the energy loss due to limiting + */ + for (k = hFreq->limiterBandTable[c]; k < hFreq->limiterBandTable[c + 1]; + k++) { + /* 1.a Add energy of adjusted signal (using preliminary gain) */ + FIXP_DBL tmp = fMult(pNrgs->nrgGain[k], pNrgs->nrgEst[k]); + SCHAR tmp_e = pNrgs->nrgGain_e[k] + pNrgs->nrgEst_e[k]; + FDK_add_MantExp(tmp, tmp_e, accu, accu_e, &accu, &accu_e); + + /* 1.b Add sine energy (if present) */ + if (pNrgs->nrgSine[k] != FL2FXCONST_DBL(0.0f)) { + FDK_add_MantExp(pNrgs->nrgSine[k], pNrgs->nrgSine_e[k], accu, accu_e, + &accu, &accu_e); + } else { + /* 1.c Add noise energy (if present) */ + if (noNoiseFlag == 0) { + FDK_add_MantExp(pNrgs->noiseLevel[k], pNrgs->noiseLevel_e[k], accu, + accu_e, &accu, &accu_e); + } + } + } + + /* 2.a Calculate ratio of wanted energy and accumulated energy */ + if (accu == (FIXP_DBL)0) { /* If divisor is 0, limit quotient to +4 dB */ + boostGain = FL2FXCONST_DBL(0.6279716f); + boostGain_e = 2; + } else { + INT div_e; + boostGain = fDivNorm(sumRef, accu, &div_e); + boostGain_e = sumRef_e - accu_e + div_e; + } + + /* 2.b Result too high? --> Limit the boost factor to +4 dB */ + if ((boostGain_e > 3) || + (boostGain_e == 2 && boostGain > FL2FXCONST_DBL(0.6279716f)) || + (boostGain_e == 3 && boostGain > FL2FXCONST_DBL(0.3139858f))) { + boostGain = FL2FXCONST_DBL(0.6279716f); + boostGain_e = 2; + } + /* 3. Multiply all signal components with the boost factor */ + for (k = hFreq->limiterBandTable[c]; k < hFreq->limiterBandTable[c + 1]; + k++) { + pNrgs->nrgGain[k] = fMultDiv2(pNrgs->nrgGain[k], boostGain); + pNrgs->nrgGain_e[k] = pNrgs->nrgGain_e[k] + boostGain_e + 1; + + pNrgs->nrgSine[k] = fMultDiv2(pNrgs->nrgSine[k], boostGain); + pNrgs->nrgSine_e[k] = pNrgs->nrgSine_e[k] + boostGain_e + 1; + + pNrgs->noiseLevel[k] = fMultDiv2(pNrgs->noiseLevel[k], boostGain); + pNrgs->noiseLevel_e[k] = pNrgs->noiseLevel_e[k] + boostGain_e + 1; + } + } + /* End of noise limiting */ + + if (useLP) + aliasingReduction(degreeAlias + lowSubband, pNrgs, useAliasReduction, + noSubbands); + + /* For the timeslots within the range for the output frame, + use the same scale for the noise levels. + Drawback: If the envelope exceeds the frame border, the noise levels + will have to be rescaled later to fit final_e of + the gain-values. + */ + noise_e = (start_pos < no_cols) ? adj_e : final_e; + + /* + Convert energies to amplitude levels + */ + for (k = 0; k < noSubbands; k++) { + FDK_sqrt_MantExp(&pNrgs->nrgSine[k], &pNrgs->nrgSine_e[k], &noise_e); + FDK_sqrt_MantExp(&pNrgs->nrgGain[k], &pNrgs->nrgGain_e[k], + &pNrgs->nrgGain_e[k]); + FDK_sqrt_MantExp(&pNrgs->noiseLevel[k], &pNrgs->noiseLevel_e[k], + &noise_e); + } + + /* + Apply calculated gains and adaptive noise + */ + + /* assembleHfSignals() */ + { + int scale_change, sc_change; + FIXP_SGL smooth_ratio; + int filtBufferNoiseShift = 0; + + /* Initialize smoothing buffers with the first valid values */ + if (h_sbr_cal_env->startUp) { + if (!useLP) { + h_sbr_cal_env->filtBufferNoise_e = noise_e; + + FDKmemcpy(h_sbr_cal_env->filtBuffer_e, pNrgs->nrgGain_e, + noSubbands * sizeof(SCHAR)); + FDKmemcpy(h_sbr_cal_env->filtBufferNoise, pNrgs->noiseLevel, + noSubbands * sizeof(FIXP_DBL)); + FDKmemcpy(h_sbr_cal_env->filtBuffer, pNrgs->nrgGain, + noSubbands * sizeof(FIXP_DBL)); + } + h_sbr_cal_env->startUp = 0; + } + + if (!useLP) { + equalizeFiltBufferExp(h_sbr_cal_env->filtBuffer, /* buffered */ + h_sbr_cal_env->filtBuffer_e, /* buffered */ + pNrgs->nrgGain, /* current */ + pNrgs->nrgGain_e, /* current */ + noSubbands); + + /* Adapt exponent of buffered noise levels to the current exponent + so they can easily be smoothed */ + if ((h_sbr_cal_env->filtBufferNoise_e - noise_e) >= 0) { + int shift = fixMin(DFRACT_BITS - 1, + (int)(h_sbr_cal_env->filtBufferNoise_e - noise_e)); + for (k = 0; k < noSubbands; k++) + h_sbr_cal_env->filtBufferNoise[k] <<= shift; + } else { + int shift = + fixMin(DFRACT_BITS - 1, + -(int)(h_sbr_cal_env->filtBufferNoise_e - noise_e)); + for (k = 0; k < noSubbands; k++) + h_sbr_cal_env->filtBufferNoise[k] >>= shift; + } + + h_sbr_cal_env->filtBufferNoise_e = noise_e; + } + + /* find best scaling! */ + scale_change = -(DFRACT_BITS - 1); + for (k = 0; k < noSubbands; k++) { + scale_change = fixMax(scale_change, (int)pNrgs->nrgGain_e[k]); + } + sc_change = (start_pos < no_cols) ? adj_e - input_e : final_e - input_e; + + if ((scale_change - sc_change + 1) < 0) + scale_change -= (scale_change - sc_change + 1); + + scale_change = (scale_change - sc_change) + 1; + + for (k = 0; k < noSubbands; k++) { + int sc = scale_change - pNrgs->nrgGain_e[k] + (sc_change - 1); + pNrgs->nrgGain[k] >>= sc; + pNrgs->nrgGain_e[k] += sc; + } + + if (!useLP) { + for (k = 0; k < noSubbands; k++) { + int sc = + scale_change - h_sbr_cal_env->filtBuffer_e[k] + (sc_change - 1); + h_sbr_cal_env->filtBuffer[k] >>= sc; + } + } + + for (j = start_pos; j < stop_pos; j++) { + /* This timeslot is located within the first part of the processing + buffer and will be fed into the QMF-synthesis for the current frame. + adj_e - input_e + This timeslot will not yet be fed into the QMF so we do not care + about the adj_e. + sc_change = final_e - input_e + */ + if ((j == no_cols) && (start_pos < no_cols)) { + int shift = (int)(noise_e - final_e); + if (!useLP) + filtBufferNoiseShift = shift; /* shifting of + h_sbr_cal_env->filtBufferNoise[k] + will be applied in function + adjustTimeSlotHQ() */ + if (shift >= 0) { + shift = fixMin(DFRACT_BITS - 1, shift); + for (k = 0; k < noSubbands; k++) { + pNrgs->nrgSine[k] <<= shift; + pNrgs->noiseLevel[k] <<= shift; + /* + if (!useLP) + h_sbr_cal_env->filtBufferNoise[k] <<= shift; + */ + } + } else { + shift = fixMin(DFRACT_BITS - 1, -shift); + for (k = 0; k < noSubbands; k++) { + pNrgs->nrgSine[k] >>= shift; + pNrgs->noiseLevel[k] >>= shift; + /* + if (!useLP) + h_sbr_cal_env->filtBufferNoise[k] >>= shift; + */ + } + } + + /* update noise scaling */ + noise_e = final_e; + if (!useLP) + h_sbr_cal_env->filtBufferNoise_e = + noise_e; /* scaling value unused! */ + + /* update gain buffer*/ + sc_change -= (final_e - input_e); + + if (sc_change < 0) { + for (k = 0; k < noSubbands; k++) { + pNrgs->nrgGain[k] >>= -sc_change; + pNrgs->nrgGain_e[k] += -sc_change; + } + if (!useLP) { + for (k = 0; k < noSubbands; k++) { + h_sbr_cal_env->filtBuffer[k] >>= -sc_change; + } + } + } else { + scale_change += sc_change; + } + + } /* if */ + + if (!useLP) { + /* Prevent the smoothing filter from running on constant levels */ + if (j - start_pos < smooth_length) + smooth_ratio = FDK_sbrDecoder_sbr_smoothFilter[j - start_pos]; + else + smooth_ratio = FL2FXCONST_SGL(0.0f); + + if (iTES_enable) { + /* adjustTimeSlotHQ() without adding of additional harmonics */ + adjustTimeSlotHQ_GainAndNoise( + &analysBufferReal[j][lowSubband], + &analysBufferImag[j][lowSubband], h_sbr_cal_env, pNrgs, + lowSubband, noSubbands, fMin(scale_change, DFRACT_BITS - 1), + smooth_ratio, noNoiseFlag, filtBufferNoiseShift); + } else { + adjustTimeSlotHQ(&analysBufferReal[j][lowSubband], + &analysBufferImag[j][lowSubband], h_sbr_cal_env, + pNrgs, lowSubband, noSubbands, + fMin(scale_change, DFRACT_BITS - 1), smooth_ratio, + noNoiseFlag, filtBufferNoiseShift); + } + } else { + FDK_ASSERT(!iTES_enable); /* not supported */ + if (flags & SBRDEC_ELD_GRID) { + /* FDKmemset(analysBufferReal[j], 0, 64 * sizeof(FIXP_DBL)); */ + adjustTimeSlot_EldGrid(&analysBufferReal[j][lowSubband], pNrgs, + &h_sbr_cal_env->harmIndex, lowSubband, + noSubbands, + fMin(scale_change, DFRACT_BITS - 1), + noNoiseFlag, &h_sbr_cal_env->phaseIndex, + EXP2SCALE(adj_e) - sbrScaleFactor->lb_scale); + } else { + adjustTimeSlotLC(&analysBufferReal[j][lowSubband], pNrgs, + &h_sbr_cal_env->harmIndex, lowSubband, noSubbands, + fMin(scale_change, DFRACT_BITS - 1), noNoiseFlag, + &h_sbr_cal_env->phaseIndex); + } + } + /* In case the envelope spans accross the no_cols border both exponents + * are needed. */ + /* nrgGain_e[0...(noSubbands-1)] are equalized by + * equalizeFiltBufferExp() */ + pNrgs->exponent[(j < no_cols) ? 0 : 1] = + (SCHAR)((15 - sbrScaleFactor->hb_scale) + pNrgs->nrgGain_e[0] + 1 - + scale_change); + } /* for */ + + if (iTES_enable) { + apply_inter_tes( + analysBufferReal, /* pABufR, */ + analysBufferImag, /* pABufI, */ + sbrScaleFactor, pNrgs->exponent, hHeaderData->timeStep, start_pos, + stop_pos, lowSubband, noSubbands, + hFrameData + ->interTempShapeMode[i] /* frameData->interTempShapeMode[env] */ + ); + + /* add additional harmonics */ + for (j = start_pos; j < stop_pos; j++) { + /* match exponent of additional harmonics to scale change of QMF data + * caused by apply_inter_tes() */ + scale_change = 0; + + if ((start_pos <= no_cols) && (stop_pos > no_cols)) { + /* Scaling of analysBuffers was potentially changed within this + envelope. The pNrgs->nrgSine_e match the second part of the + envelope. For (j<=no_cols) the exponent of the sine energies has + to be adapted. */ + scale_change = pNrgs->exponent[1] - pNrgs->exponent[0]; + } + + adjustTimeSlotHQ_AddHarmonics( + &analysBufferReal[j][lowSubband], + &analysBufferImag[j][lowSubband], h_sbr_cal_env, pNrgs, + lowSubband, noSubbands, + -iTES_scale_change + ((j < no_cols) ? scale_change : 0)); + } + } + + if (!useLP) { + /* Update time-smoothing-buffers for gains and noise levels + The gains and the noise values of the current envelope are copied + into the buffer. This has to be done at the end of each envelope as + the values are required for a smooth transition to the next envelope. + */ + FDKmemcpy(h_sbr_cal_env->filtBuffer, pNrgs->nrgGain, + noSubbands * sizeof(FIXP_DBL)); + FDKmemcpy(h_sbr_cal_env->filtBuffer_e, pNrgs->nrgGain_e, + noSubbands * sizeof(SCHAR)); + FDKmemcpy(h_sbr_cal_env->filtBufferNoise, pNrgs->noiseLevel, + noSubbands * sizeof(FIXP_DBL)); + } + } + C_ALLOC_SCRATCH_END(pNrgs, ENV_CALC_NRGS, 1); + } + + /* adapt adj_e to the scale change caused by apply_inter_tes() */ + adj_e += iTES_scale_change; + + /* Rescale output samples */ + { + FIXP_DBL maxVal; + int ov_reserve, reserve; + + /* Determine headroom in old adjusted samples */ + maxVal = + maxSubbandSample(analysBufferReal, (useLP) ? NULL : analysBufferImag, + lowSubband, ov_highSubband, 0, first_start); + + ov_reserve = fNorm(maxVal); + + /* Determine headroom in new adjusted samples */ + maxVal = + maxSubbandSample(analysBufferReal, (useLP) ? NULL : analysBufferImag, + lowSubband, highSubband, first_start, no_cols); + + reserve = fNorm(maxVal); + + /* Determine common output exponent */ + output_e = fMax(ov_adj_e - ov_reserve, adj_e - reserve); + + /* Rescale old samples */ + rescaleSubbandSamples(analysBufferReal, (useLP) ? NULL : analysBufferImag, + lowSubband, ov_highSubband, 0, first_start, + ov_adj_e - output_e); + + /* Rescale new samples */ + rescaleSubbandSamples(analysBufferReal, (useLP) ? NULL : analysBufferImag, + lowSubband, highSubband, first_start, no_cols, + adj_e - output_e); + } + + /* Update hb_scale */ + sbrScaleFactor->hb_scale = EXP2SCALE(output_e); + + /* Save the current final exponent for the next frame: */ + /* adapt final_e to the scale change caused by apply_inter_tes() */ + sbrScaleFactor->ov_hb_scale = EXP2SCALE(final_e + iTES_scale_change); + + /* We need to remember to the next frame that the transient + will occur in the first envelope (if tranEnv == nEnvelopes). */ + if (hFrameData->frameInfo.tranEnv == hFrameData->frameInfo.nEnvelopes) + h_sbr_cal_env->prevTranEnv = 0; + else + h_sbr_cal_env->prevTranEnv = -1; + + if (pvc_mode > 0) { + /* Not more than just the last noise envelope reaches into the next PVC + frame! This should be true because bs_noise_position is <= 15 */ + FDK_ASSERT(hFrameData->frameInfo + .bordersNoise[hFrameData->frameInfo.nNoiseEnvelopes - 1] < + PVC_NTIMESLOT); + if (hFrameData->frameInfo + .bordersNoise[hFrameData->frameInfo.nNoiseEnvelopes] > + PVC_NTIMESLOT) { + FDK_ASSERT(noiseLevels == + (hFrameData->sbrNoiseFloorLevel + + (hFrameData->frameInfo.nNoiseEnvelopes - 1) * noNoiseBands)); + h_sbr_cal_env->prevNNfb = noNoiseBands; + + h_sbr_cal_env->prevNSfb[0] = noSubFrameBands[0]; + h_sbr_cal_env->prevNSfb[1] = noSubFrameBands[1]; + + h_sbr_cal_env->prevLoSubband = lowSubband; + h_sbr_cal_env->prevHiSubband = highSubband; + h_sbr_cal_env->prev_ov_highSubband = ov_highSubband; + + FDKmemcpy(h_sbr_cal_env->prevFreqBandTableLo, pFreqBandTable[0], + noSubFrameBands[0] + 1); + FDKmemcpy(h_sbr_cal_env->prevFreqBandTableHi, pFreqBandTable[1], + noSubFrameBands[1] + 1); + FDKmemcpy(h_sbr_cal_env->prevFreqBandTableNoise, + hFreq->freqBandTableNoise, sizeof(hFreq->freqBandTableNoise)); + + FDKmemcpy(h_sbr_cal_env->prevSbrNoiseFloorLevel, noiseLevels, + MAX_NOISE_COEFFS * sizeof(FIXP_SGL)); + } + } + + C_ALLOC_SCRATCH_END(useAliasReduction, UCHAR, 64) +} + +/*! + \brief Create envelope instance + + Must be called once for each channel before calculateSbrEnvelope() can be + used. + + \return errorCode, 0 if successful +*/ +SBR_ERROR +createSbrEnvelopeCalc( + HANDLE_SBR_CALCULATE_ENVELOPE hs, /*!< pointer to envelope instance */ + HANDLE_SBR_HEADER_DATA + hHeaderData, /*!< static SBR control data, initialized with defaults */ + const int chan, /*!< Channel for which to assign buffers */ + const UINT flags) { + SBR_ERROR err = SBRDEC_OK; + int i; + + /* Clear previous missing harmonics flags */ + for (i = 0; i < ADD_HARMONICS_FLAGS_SIZE; i++) { + hs->harmFlagsPrev[i] = 0; + hs->harmFlagsPrevActive[i] = 0; + } + hs->harmIndex = 0; + + FDKmemclear(hs->prevSbrNoiseFloorLevel, sizeof(hs->prevSbrNoiseFloorLevel)); + hs->prevNNfb = 0; + FDKmemclear(hs->prevFreqBandTableNoise, sizeof(hs->prevFreqBandTableNoise)); + hs->sinusoidal_positionPrev = 0; + + /* + Setup pointers for time smoothing. + The buffer itself will be initialized later triggered by the startUp-flag. + */ + hs->prevTranEnv = -1; + + /* initialization */ + resetSbrEnvelopeCalc(hs); + + if (chan == 0) { /* do this only once */ + err = resetFreqBandTables(hHeaderData, flags); + } + + return err; +} + +/*! + \brief Create envelope instance + + Must be called once for each channel before calculateSbrEnvelope() can be + used. + + \return errorCode, 0 if successful +*/ +int deleteSbrEnvelopeCalc(HANDLE_SBR_CALCULATE_ENVELOPE hs) { return 0; } + +/*! + \brief Reset envelope instance + + This function must be called for each channel on a change of configuration. + Note that resetFreqBandTables should also be called in this case. + + \return errorCode, 0 if successful +*/ +void resetSbrEnvelopeCalc( + HANDLE_SBR_CALCULATE_ENVELOPE hCalEnv) /*!< pointer to envelope instance */ +{ + hCalEnv->phaseIndex = 0; + + /* Noise exponent needs to be reset because the output exponent for the next + * frame depends on it */ + hCalEnv->filtBufferNoise_e = 0; + + hCalEnv->startUp = 1; +} + +/*! + \brief Equalize exponents of the buffered gain values and the new ones + + After equalization of exponents, the FIR-filter addition for smoothing + can be performed. + This function is called once for each envelope before adjusting. +*/ +static void equalizeFiltBufferExp( + FIXP_DBL *filtBuffer, /*!< bufferd gains */ + SCHAR *filtBuffer_e, /*!< exponents of bufferd gains */ + FIXP_DBL *nrgGain, /*!< gains for current envelope */ + SCHAR *nrgGain_e, /*!< exponents of gains for current envelope */ + int subbands) /*!< Number of QMF subbands */ +{ + int band; + int diff; + + for (band = 0; band < subbands; band++) { + diff = (int)(nrgGain_e[band] - filtBuffer_e[band]); + if (diff > 0) { + filtBuffer[band] >>= + diff; /* Compensate for the scale change by shifting the mantissa. */ + filtBuffer_e[band] += diff; /* New gain is bigger, use its exponent */ + } else if (diff < 0) { + /* The buffered gains seem to be larger, but maybe there + are some unused bits left in the mantissa */ + + int reserve = CntLeadingZeros(fixp_abs(filtBuffer[band])) - 1; + + if ((-diff) <= reserve) { + /* There is enough space in the buffered mantissa so + that we can take the new exponent as common. + */ + filtBuffer[band] <<= (-diff); + filtBuffer_e[band] += diff; /* becomes equal to *ptrNewExp */ + } else { + filtBuffer[band] <<= + reserve; /* Shift the mantissa as far as possible: */ + filtBuffer_e[band] -= reserve; /* Compensate in the exponent: */ + + /* For the remaining difference, change the new gain value */ + diff = fixMin(-(reserve + diff), DFRACT_BITS - 1); + nrgGain[band] >>= diff; + nrgGain_e[band] += diff; + } + } + } +} + +/*! + \brief Shift left the mantissas of all subband samples + in the giventime and frequency range by the specified number of bits. + + This function is used to rescale the audio data in the overlap buffer + which has already been envelope adjusted with the last frame. +*/ +void rescaleSubbandSamples( + FIXP_DBL **re, /*!< Real part of input and output subband samples */ + FIXP_DBL **im, /*!< Imaginary part of input and output subband samples */ + int lowSubband, /*!< Begin of frequency range to process */ + int highSubband, /*!< End of frequency range to process */ + int start_pos, /*!< Begin of time rage (QMF-timeslot) */ + int next_pos, /*!< End of time rage (QMF-timeslot) */ + int shift) /*!< number of bits to shift */ +{ + int width = highSubband - lowSubband; + + if ((width > 0) && (shift != 0)) { + if (im != NULL) { + for (int l = start_pos; l < next_pos; l++) { + scaleValues(&re[l][lowSubband], width, shift); + scaleValues(&im[l][lowSubband], width, shift); + } + } else { + for (int l = start_pos; l < next_pos; l++) { + scaleValues(&re[l][lowSubband], width, shift); + } + } + } +} + +static inline FIXP_DBL FDK_get_maxval_real(FIXP_DBL maxVal, FIXP_DBL *reTmp, + INT width) { + maxVal = (FIXP_DBL)0; + while (width-- != 0) { + FIXP_DBL tmp = *(reTmp++); + maxVal |= (FIXP_DBL)((LONG)(tmp) ^ ((LONG)tmp >> (DFRACT_BITS - 1))); + } + + return maxVal; +} + +/*! + \brief Determine headroom for shifting + + Determine by how much the spectrum can be shifted left + for better accuracy in later processing. + + \return Number of free bits in the biggest spectral value +*/ + +FIXP_DBL maxSubbandSample( + FIXP_DBL **re, /*!< Real part of input and output subband samples */ + FIXP_DBL **im, /*!< Real part of input and output subband samples */ + int lowSubband, /*!< Begin of frequency range to process */ + int highSubband, /*!< Number of QMF bands to process */ + int start_pos, /*!< Begin of time rage (QMF-timeslot) */ + int next_pos /*!< End of time rage (QMF-timeslot) */ +) { + FIXP_DBL maxVal = FL2FX_DBL(0.0f); + unsigned int width = highSubband - lowSubband; + + FDK_ASSERT(width <= (64)); + + if (width > 0) { + if (im != NULL) { + for (int l = start_pos; l < next_pos; l++) { + int k = width; + FIXP_DBL *reTmp = &re[l][lowSubband]; + FIXP_DBL *imTmp = &im[l][lowSubband]; + do { + FIXP_DBL tmp1 = *(reTmp++); + FIXP_DBL tmp2 = *(imTmp++); + maxVal |= + (FIXP_DBL)((LONG)(tmp1) ^ ((LONG)tmp1 >> (DFRACT_BITS - 1))); + maxVal |= + (FIXP_DBL)((LONG)(tmp2) ^ ((LONG)tmp2 >> (DFRACT_BITS - 1))); + } while (--k != 0); + } + } else { + for (int l = start_pos; l < next_pos; l++) { + maxVal |= FDK_get_maxval_real(maxVal, &re[l][lowSubband], width); + } + } + } + + if (maxVal > (FIXP_DBL)0) { + /* For negative input values, maxVal is too small by 1. Add 1 only when + * necessary: if maxVal is a power of 2 */ + FIXP_DBL lowerPow2 = + (FIXP_DBL)(1 << (DFRACT_BITS - 1 - CntLeadingZeros(maxVal))); + if (maxVal == lowerPow2) maxVal += (FIXP_DBL)1; + } + + return (maxVal); +} + +/* #define SHIFT_BEFORE_SQUARE (3) */ /* (7/2) */ +/* Avoid assertion failures triggerd by overflows which occured in robustness + tests. Setting the SHIFT_BEFORE_SQUARE to 4 has negligible effect on (USAC) + conformance results. */ +#define SHIFT_BEFORE_SQUARE (4) /* ((8 - 0) / 2) */ + +/*!< + If the accumulator does not provide enough overflow bits or + does not provide a high dynamic range, the below energy calculation + requires an additional shift operation for each sample. + On the other hand, doing the shift allows using a single-precision + multiplication for the square (at least 16bit x 16bit). + For even values of OVRFLW_BITS (0, 2, 4, 6), saturated arithmetic + is required for the energy accumulation. + Theoretically, the sample-squares can sum up to a value of 76, + requiring 7 overflow bits. However since such situations are *very* + rare, accu can be limited to 64. + In case native saturated arithmetic is not available, overflows + can be prevented by replacing the above #define by + #define SHIFT_BEFORE_SQUARE ((8 - OVRFLW_BITS) / 2) + which will result in slightly reduced accuracy. +*/ + +/*! + \brief Estimates the mean energy of each filter-bank channel for the + duration of the current envelope + + This function is used when interpolFreq is true. +*/ +static void calcNrgPerSubband( + FIXP_DBL **analysBufferReal, /*!< Real part of subband samples */ + FIXP_DBL **analysBufferImag, /*!< Imaginary part of subband samples */ + int lowSubband, /*!< Begin of the SBR frequency range */ + int highSubband, /*!< High end of the SBR frequency range */ + int start_pos, /*!< First QMF-slot of current envelope */ + int next_pos, /*!< Last QMF-slot of current envelope + 1 */ + SCHAR frameExp, /*!< Common exponent for all input samples */ + FIXP_DBL *nrgEst, /*!< resulting Energy (0..1) */ + SCHAR *nrgEst_e) /*!< Exponent of resulting Energy */ +{ + FIXP_SGL invWidth; + SCHAR preShift; + SCHAR shift; + FIXP_DBL sum; + int k; + + /* Divide by width of envelope later: */ + invWidth = FX_DBL2FX_SGL(GetInvInt(next_pos - start_pos)); + /* The common exponent needs to be doubled because all mantissas are squared: + */ + frameExp = frameExp << 1; + + for (k = lowSubband; k < highSubband; k++) { + FIXP_DBL bufferReal[(((1024) / (32) * (4) / 2) + (3 * (4)))]; + FIXP_DBL bufferImag[(((1024) / (32) * (4) / 2) + (3 * (4)))]; + FIXP_DBL maxVal; + + if (analysBufferImag != NULL) { + int l; + maxVal = FL2FX_DBL(0.0f); + for (l = start_pos; l < next_pos; l++) { + bufferImag[l] = analysBufferImag[l][k]; + maxVal |= (FIXP_DBL)((LONG)(bufferImag[l]) ^ + ((LONG)bufferImag[l] >> (DFRACT_BITS - 1))); + bufferReal[l] = analysBufferReal[l][k]; + maxVal |= (FIXP_DBL)((LONG)(bufferReal[l]) ^ + ((LONG)bufferReal[l] >> (DFRACT_BITS - 1))); + } + } else { + int l; + maxVal = FL2FX_DBL(0.0f); + for (l = start_pos; l < next_pos; l++) { + bufferReal[l] = analysBufferReal[l][k]; + maxVal |= (FIXP_DBL)((LONG)(bufferReal[l]) ^ + ((LONG)bufferReal[l] >> (DFRACT_BITS - 1))); + } + } + + if (maxVal != FL2FXCONST_DBL(0.f)) { + /* If the accu does not provide enough overflow bits, we cannot + shift the samples up to the limit. + Instead, keep up to 3 free bits in each sample, i.e. up to + 6 bits after calculation of square. + Please note the comment on saturated arithmetic above! + */ + FIXP_DBL accu; + preShift = CntLeadingZeros(maxVal) - 1; + preShift -= SHIFT_BEFORE_SQUARE; + + /* Limit preShift to a maximum value to prevent accumulator overflow in + exceptional situations where the signal in the analysis-buffer is very + small (small maxVal). + */ + preShift = fMin(preShift, (SCHAR)25); + + accu = FL2FXCONST_DBL(0.0f); + if (preShift >= 0) { + int l; + if (analysBufferImag != NULL) { + for (l = start_pos; l < next_pos; l++) { + FIXP_DBL temp1 = bufferReal[l] << (int)preShift; + FIXP_DBL temp2 = bufferImag[l] << (int)preShift; + accu = fPow2AddDiv2(accu, temp1); + accu = fPow2AddDiv2(accu, temp2); + } + } else { + for (l = start_pos; l < next_pos; l++) { + FIXP_DBL temp = bufferReal[l] << (int)preShift; + accu = fPow2AddDiv2(accu, temp); + } + } + } else { /* if negative shift value */ + int l; + int negpreShift = -preShift; + if (analysBufferImag != NULL) { + for (l = start_pos; l < next_pos; l++) { + FIXP_DBL temp1 = bufferReal[l] >> (int)negpreShift; + FIXP_DBL temp2 = bufferImag[l] >> (int)negpreShift; + accu = fPow2AddDiv2(accu, temp1); + accu = fPow2AddDiv2(accu, temp2); + } + } else { + for (l = start_pos; l < next_pos; l++) { + FIXP_DBL temp = bufferReal[l] >> (int)negpreShift; + accu = fPow2AddDiv2(accu, temp); + } + } + } + accu <<= 1; + + /* Convert double precision to Mantissa/Exponent: */ + shift = fNorm(accu); + sum = accu << (int)shift; + + /* Divide by width of envelope and apply frame scale: */ + *nrgEst++ = fMult(sum, invWidth); + shift += 2 * preShift; + if (analysBufferImag != NULL) + *nrgEst_e++ = frameExp - shift; + else + *nrgEst_e++ = frameExp - shift + 1; /* +1 due to missing imag. part */ + } /* maxVal!=0 */ + else { + /* Prevent a zero-mantissa-number from being misinterpreted + due to its exponent. */ + *nrgEst++ = FL2FXCONST_DBL(0.0f); + *nrgEst_e++ = 0; + } + } +} + +/*! + \brief Estimates the mean energy of each Scale factor band for the + duration of the current envelope. + + This function is used when interpolFreq is false. +*/ +static void calcNrgPerSfb( + FIXP_DBL **analysBufferReal, /*!< Real part of subband samples */ + FIXP_DBL **analysBufferImag, /*!< Imaginary part of subband samples */ + int nSfb, /*!< Number of scale factor bands */ + UCHAR *freqBandTable, /*!< First Subband for each Sfb */ + int start_pos, /*!< First QMF-slot of current envelope */ + int next_pos, /*!< Last QMF-slot of current envelope + 1 */ + SCHAR input_e, /*!< Common exponent for all input samples */ + FIXP_DBL *nrgEst, /*!< resulting Energy (0..1) */ + SCHAR *nrgEst_e) /*!< Exponent of resulting Energy */ +{ + FIXP_SGL invWidth; + FIXP_DBL temp; + SCHAR preShift; + SCHAR shift, sum_e; + FIXP_DBL sum; + + int j, k, l, li, ui; + FIXP_DBL sumAll, sumLine; /* Single precision would be sufficient, + but overflow bits are required for accumulation */ + + /* Divide by width of envelope later: */ + invWidth = FX_DBL2FX_SGL(GetInvInt(next_pos - start_pos)); + /* The common exponent needs to be doubled because all mantissas are squared: + */ + input_e = input_e << 1; + + for (j = 0; j < nSfb; j++) { + li = freqBandTable[j]; + ui = freqBandTable[j + 1]; + + FIXP_DBL maxVal = maxSubbandSample(analysBufferReal, analysBufferImag, li, + ui, start_pos, next_pos); + + if (maxVal != FL2FXCONST_DBL(0.f)) { + preShift = CntLeadingZeros(maxVal) - 1; + + /* If the accu does not provide enough overflow bits, we cannot + shift the samples up to the limit. + Instead, keep up to 3 free bits in each sample, i.e. up to + 6 bits after calculation of square. + Please note the comment on saturated arithmetic above! + */ + preShift -= SHIFT_BEFORE_SQUARE; + + sumAll = FL2FXCONST_DBL(0.0f); + + for (k = li; k < ui; k++) { + sumLine = FL2FXCONST_DBL(0.0f); + + if (analysBufferImag != NULL) { + if (preShift >= 0) { + for (l = start_pos; l < next_pos; l++) { + temp = analysBufferReal[l][k] << (int)preShift; + sumLine += fPow2Div2(temp); + temp = analysBufferImag[l][k] << (int)preShift; + sumLine += fPow2Div2(temp); + } + } else { + for (l = start_pos; l < next_pos; l++) { + temp = analysBufferReal[l][k] >> -(int)preShift; + sumLine += fPow2Div2(temp); + temp = analysBufferImag[l][k] >> -(int)preShift; + sumLine += fPow2Div2(temp); + } + } + } else { + if (preShift >= 0) { + for (l = start_pos; l < next_pos; l++) { + temp = analysBufferReal[l][k] << (int)preShift; + sumLine += fPow2Div2(temp); + } + } else { + for (l = start_pos; l < next_pos; l++) { + temp = analysBufferReal[l][k] >> -(int)preShift; + sumLine += fPow2Div2(temp); + } + } + } + + /* The number of QMF-channels per SBR bands may be up to 15. + Shift right to avoid overflows in sum over all channels. */ + sumLine = sumLine >> (4 - 1); + sumAll += sumLine; + } + + /* Convert double precision to Mantissa/Exponent: */ + shift = fNorm(sumAll); + sum = sumAll << (int)shift; + + /* Divide by width of envelope: */ + sum = fMult(sum, invWidth); + + /* Divide by width of Sfb: */ + sum = fMult(sum, FX_DBL2FX_SGL(GetInvInt(ui - li))); + + /* Set all Subband energies in the Sfb to the average energy: */ + if (analysBufferImag != NULL) + sum_e = input_e + 4 - shift; /* -4 to compensate right-shift */ + else + sum_e = input_e + 4 + 1 - + shift; /* -4 to compensate right-shift; +1 due to missing + imag. part */ + + sum_e -= 2 * preShift; + } /* maxVal!=0 */ + else { + /* Prevent a zero-mantissa-number from being misinterpreted + due to its exponent. */ + sum = FL2FXCONST_DBL(0.0f); + sum_e = 0; + } + + for (k = li; k < ui; k++) { + *nrgEst++ = sum; + *nrgEst_e++ = sum_e; + } + } +} + +/*! + \brief Calculate gain, noise, and additional sine level for one subband. + + The resulting energy gain is given by mantissa and exponent. +*/ +static void calcSubbandGain( + FIXP_DBL nrgRef, /*!< Reference Energy according to envelope data */ + SCHAR + nrgRef_e, /*!< Reference Energy according to envelope data (exponent) */ + ENV_CALC_NRGS *nrgs, int i, FIXP_DBL tmpNoise, /*!< Relative noise level */ + SCHAR tmpNoise_e, /*!< Relative noise level (exponent) */ + UCHAR sinePresentFlag, /*!< Indicates if sine is present on band */ + UCHAR sineMapped, /*!< Indicates if sine must be added */ + int noNoiseFlag) /*!< Flag to suppress noise addition */ +{ + FIXP_DBL nrgEst = nrgs->nrgEst[i]; /*!< Energy in transposed signal */ + SCHAR nrgEst_e = + nrgs->nrgEst_e[i]; /*!< Energy in transposed signal (exponent) */ + FIXP_DBL *ptrNrgGain = &nrgs->nrgGain[i]; /*!< Resulting energy gain */ + SCHAR *ptrNrgGain_e = + &nrgs->nrgGain_e[i]; /*!< Resulting energy gain (exponent) */ + FIXP_DBL *ptrNoiseLevel = + &nrgs->noiseLevel[i]; /*!< Resulting absolute noise energy */ + SCHAR *ptrNoiseLevel_e = + &nrgs->noiseLevel_e[i]; /*!< Resulting absolute noise energy (exponent) */ + FIXP_DBL *ptrNrgSine = &nrgs->nrgSine[i]; /*!< Additional sine energy */ + SCHAR *ptrNrgSine_e = + &nrgs->nrgSine_e[i]; /*!< Additional sine energy (exponent) */ + + FIXP_DBL a, b, c; + SCHAR a_e, b_e, c_e; + + /* + This addition of 1 prevents divisions by zero in the reference code. + For very small energies in nrgEst, it prevents the gains from becoming + very high which could cause some trouble due to the smoothing. + */ + b_e = (int)(nrgEst_e - 1); + if (b_e >= 0) { + nrgEst = (FL2FXCONST_DBL(0.5f) >> (INT)fixMin(b_e + 1, DFRACT_BITS - 1)) + + (nrgEst >> 1); + nrgEst_e += 1; /* shift by 1 bit to avoid overflow */ + + } else { + nrgEst = (nrgEst >> (INT)(fixMin(-b_e + 1, DFRACT_BITS - 1))) + + (FL2FXCONST_DBL(0.5f) >> 1); + nrgEst_e = 2; /* shift by 1 bit to avoid overflow */ + } + + /* A = NrgRef * TmpNoise */ + a = fMult(nrgRef, tmpNoise); + a_e = nrgRef_e + tmpNoise_e; + + /* B = 1 + TmpNoise */ + b_e = (int)(tmpNoise_e - 1); + if (b_e >= 0) { + b = (FL2FXCONST_DBL(0.5f) >> (INT)fixMin(b_e + 1, DFRACT_BITS - 1)) + + (tmpNoise >> 1); + b_e = tmpNoise_e + 1; /* shift by 1 bit to avoid overflow */ + } else { + b = (tmpNoise >> (INT)(fixMin(-b_e + 1, DFRACT_BITS - 1))) + + (FL2FXCONST_DBL(0.5f) >> 1); + b_e = 2; /* shift by 1 bit to avoid overflow */ + } + + /* noiseLevel = A / B = (NrgRef * TmpNoise) / (1 + TmpNoise) */ + FDK_divide_MantExp(a, a_e, b, b_e, ptrNoiseLevel, ptrNoiseLevel_e); + + if (sinePresentFlag) { + /* C = (1 + TmpNoise) * NrgEst */ + c = fMult(b, nrgEst); + c_e = b_e + nrgEst_e; + + /* gain = A / C = (NrgRef * TmpNoise) / (1 + TmpNoise) * NrgEst */ + FDK_divide_MantExp(a, a_e, c, c_e, ptrNrgGain, ptrNrgGain_e); + + if (sineMapped) { + /* sineLevel = nrgRef/ (1 + TmpNoise) */ + FDK_divide_MantExp(nrgRef, nrgRef_e, b, b_e, ptrNrgSine, ptrNrgSine_e); + } + } else { + if (noNoiseFlag) { + /* B = NrgEst */ + b = nrgEst; + b_e = nrgEst_e; + } else { + /* B = NrgEst * (1 + TmpNoise) */ + b = fMult(b, nrgEst); + b_e = b_e + nrgEst_e; + } + + /* gain = nrgRef / B */ + FDK_divide_MantExp(nrgRef, nrgRef_e, b, b_e, ptrNrgGain, ptrNrgGain_e); + } +} + +/*! + \brief Calculate "average gain" for the specified subband range. + + This is rather a gain of the average magnitude than the average + of gains! + The result is used as a relative limit for all gains within the + current "limiter band" (a certain frequency range). +*/ +static void calcAvgGain( + ENV_CALC_NRGS *nrgs, int lowSubband, /*!< Begin of the limiter band */ + int highSubband, /*!< High end of the limiter band */ + FIXP_DBL *ptrSumRef, SCHAR *ptrSumRef_e, + FIXP_DBL *ptrAvgGain, /*!< Resulting overall gain (mantissa) */ + SCHAR *ptrAvgGain_e) /*!< Resulting overall gain (exponent) */ +{ + FIXP_DBL *nrgRef = + nrgs->nrgRef; /*!< Reference Energy according to envelope data */ + SCHAR *nrgRef_e = + nrgs->nrgRef_e; /*!< Reference Energy according to envelope data + (exponent) */ + FIXP_DBL *nrgEst = nrgs->nrgEst; /*!< Energy in transposed signal */ + SCHAR *nrgEst_e = + nrgs->nrgEst_e; /*!< Energy in transposed signal (exponent) */ + + FIXP_DBL sumRef = 1; + FIXP_DBL sumEst = 1; + SCHAR sumRef_e = -FRACT_BITS; + SCHAR sumEst_e = -FRACT_BITS; + int k; + + for (k = lowSubband; k < highSubband; k++) { + /* Add nrgRef[k] to sumRef: */ + FDK_add_MantExp(sumRef, sumRef_e, nrgRef[k], nrgRef_e[k], &sumRef, + &sumRef_e); + + /* Add nrgEst[k] to sumEst: */ + FDK_add_MantExp(sumEst, sumEst_e, nrgEst[k], nrgEst_e[k], &sumEst, + &sumEst_e); + } + + FDK_divide_MantExp(sumRef, sumRef_e, sumEst, sumEst_e, ptrAvgGain, + ptrAvgGain_e); + + *ptrSumRef = sumRef; + *ptrSumRef_e = sumRef_e; +} + +static void adjustTimeSlot_EldGrid( + FIXP_DBL *RESTRICT + ptrReal, /*!< Subband samples to be adjusted, real part */ + ENV_CALC_NRGS *nrgs, UCHAR *ptrHarmIndex, /*!< Harmonic index */ + int lowSubband, /*!< Lowest QMF-channel in the currently used SBR range. */ + int noSubbands, /*!< Number of QMF subbands */ + int scale_change, /*!< Number of bits to shift adjusted samples */ + int noNoiseFlag, /*!< Flag to suppress noise addition */ + int *ptrPhaseIndex, /*!< Start index to random number array */ + int scale_diff_low) /*!< */ + +{ + int k; + FIXP_DBL signalReal, sbNoise; + int tone_count = 0; + + FIXP_DBL *pGain = nrgs->nrgGain; /*!< Gains of current envelope */ + FIXP_DBL *RESTRICT pNoiseLevel = + nrgs->noiseLevel; /*!< Noise levels of current envelope */ + FIXP_DBL *RESTRICT pSineLevel = nrgs->nrgSine; /*!< Sine levels */ + + int phaseIndex = *ptrPhaseIndex; + UCHAR harmIndex = *ptrHarmIndex; + + static const INT harmonicPhase[4][2] = {{1, 0}, {0, 1}, {-1, 0}, {0, -1}}; + + static const FIXP_DBL harmonicPhaseX[4][2] = { + {FL2FXCONST_DBL(2.0 * 1.245183154539139e-001), + FL2FXCONST_DBL(2.0 * 1.245183154539139e-001)}, + {FL2FXCONST_DBL(2.0 * -1.123767859325028e-001), + FL2FXCONST_DBL(2.0 * 1.123767859325028e-001)}, + {FL2FXCONST_DBL(2.0 * -1.245183154539139e-001), + FL2FXCONST_DBL(2.0 * -1.245183154539139e-001)}, + {FL2FXCONST_DBL(2.0 * 1.123767859325028e-001), + FL2FXCONST_DBL(2.0 * -1.123767859325028e-001)}}; + + const FIXP_DBL *p_harmonicPhaseX = &harmonicPhaseX[harmIndex][0]; + const INT *p_harmonicPhase = &harmonicPhase[harmIndex][0]; + + *(ptrReal - 1) = fAddSaturate( + *(ptrReal - 1), + SATURATE_SHIFT(fMultDiv2(p_harmonicPhaseX[lowSubband & 1], pSineLevel[0]), + scale_diff_low, DFRACT_BITS)); + FIXP_DBL pSineLevel_prev = (FIXP_DBL)0; + + int idx_k = lowSubband & 1; + + for (k = 0; k < noSubbands; k++) { + FIXP_DBL sineLevel_curr = *pSineLevel++; + phaseIndex = (phaseIndex + 1) & (SBR_NF_NO_RANDOM_VAL - 1); + + signalReal = fMultDiv2(*ptrReal, *pGain++) << ((int)scale_change); + sbNoise = *pNoiseLevel++; + if (((INT)sineLevel_curr | noNoiseFlag) == 0) { + signalReal += + (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[phaseIndex][0], sbNoise) + << 4); + } + signalReal += sineLevel_curr * p_harmonicPhase[0]; + signalReal = + fMultAddDiv2(signalReal, pSineLevel_prev, p_harmonicPhaseX[idx_k]); + pSineLevel_prev = sineLevel_curr; + idx_k = !idx_k; + if (k < noSubbands - 1) { + signalReal = + fMultAddDiv2(signalReal, pSineLevel[0], p_harmonicPhaseX[idx_k]); + } else /* (k == noSubbands - 1) */ + { + if (k + lowSubband + 1 < 63) { + *(ptrReal + 1) += fMultDiv2(pSineLevel_prev, p_harmonicPhaseX[idx_k]); + } + } + *ptrReal++ = signalReal; + + if (pSineLevel_prev != FL2FXCONST_DBL(0.0f)) { + if (++tone_count == 16) { + k++; + break; + } + } + } + /* Run again, if previous loop got breaked with tone_count = 16 */ + for (; k < noSubbands; k++) { + FIXP_DBL sineLevel_curr = *pSineLevel++; + phaseIndex = (phaseIndex + 1) & (SBR_NF_NO_RANDOM_VAL - 1); + + signalReal = fMultDiv2(*ptrReal, *pGain++) << ((int)scale_change); + sbNoise = *pNoiseLevel++; + if (((INT)sineLevel_curr | noNoiseFlag) == 0) { + signalReal += + (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[phaseIndex][0], sbNoise) + << 4); + } + signalReal += sineLevel_curr * p_harmonicPhase[0]; + *ptrReal++ = signalReal; + } + + *ptrHarmIndex = (harmIndex + 1) & 3; + *ptrPhaseIndex = phaseIndex & (SBR_NF_NO_RANDOM_VAL - 1); +} + +/*! + \brief Amplify one timeslot of the signal with the calculated gains + and add the noisefloor. +*/ + +static void adjustTimeSlotLC( + FIXP_DBL *ptrReal, /*!< Subband samples to be adjusted, real part */ + ENV_CALC_NRGS *nrgs, UCHAR *ptrHarmIndex, /*!< Harmonic index */ + int lowSubband, /*!< Lowest QMF-channel in the currently used SBR range. */ + int noSubbands, /*!< Number of QMF subbands */ + int scale_change, /*!< Number of bits to shift adjusted samples */ + int noNoiseFlag, /*!< Flag to suppress noise addition */ + int *ptrPhaseIndex) /*!< Start index to random number array */ +{ + FIXP_DBL *pGain = nrgs->nrgGain; /*!< Gains of current envelope */ + FIXP_DBL *pNoiseLevel = + nrgs->noiseLevel; /*!< Noise levels of current envelope */ + FIXP_DBL *pSineLevel = nrgs->nrgSine; /*!< Sine levels */ + + int k; + int index = *ptrPhaseIndex; + UCHAR harmIndex = *ptrHarmIndex; + UCHAR freqInvFlag = (lowSubband & 1); + FIXP_DBL signalReal, sineLevel, sineLevelNext, sineLevelPrev; + int tone_count = 0; + int sineSign = 1; + +#define C1 ((FIXP_SGL)FL2FXCONST_SGL(2.f * 0.00815f)) +#define C1_CLDFB ((FIXP_SGL)FL2FXCONST_SGL(2.f * 0.16773f)) + + /* + First pass for k=0 pulled out of the loop: + */ + + index = (index + 1) & (SBR_NF_NO_RANDOM_VAL - 1); + + /* + The next multiplication constitutes the actual envelope adjustment + of the signal and should be carried out with full accuracy + (supplying #FRACT_BITS valid bits). + */ + signalReal = fMultDiv2(*ptrReal, *pGain++) << ((int)scale_change); + sineLevel = *pSineLevel++; + sineLevelNext = (noSubbands > 1) ? pSineLevel[0] : FL2FXCONST_DBL(0.0f); + + if (sineLevel != FL2FXCONST_DBL(0.0f)) + tone_count++; + else if (!noNoiseFlag) + /* Add noisefloor to the amplified signal */ + signalReal += + (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0]) + << 4); + + { + if (!(harmIndex & 0x1)) { + /* harmIndex 0,2 */ + signalReal += (harmIndex & 0x2) ? -sineLevel : sineLevel; + *ptrReal++ = signalReal; + } else { + /* harmIndex 1,3 in combination with freqInvFlag */ + int shift = (int)(scale_change + 1); + shift = (shift >= 0) ? fixMin(DFRACT_BITS - 1, shift) + : fixMax(-(DFRACT_BITS - 1), shift); + + FIXP_DBL tmp1 = (shift >= 0) ? (fMultDiv2(C1, sineLevel) >> shift) + : (fMultDiv2(C1, sineLevel) << (-shift)); + FIXP_DBL tmp2 = fMultDiv2(C1, sineLevelNext); + + /* save switch and compare operations and reduce to XOR statement */ + if (((harmIndex >> 1) & 0x1) ^ freqInvFlag) { + *(ptrReal - 1) += tmp1; + signalReal -= tmp2; + } else { + *(ptrReal - 1) -= tmp1; + signalReal += tmp2; + } + *ptrReal++ = signalReal; + freqInvFlag = !freqInvFlag; + } + } + + pNoiseLevel++; + + if (noSubbands > 2) { + if (!(harmIndex & 0x1)) { + /* harmIndex 0,2 */ + if (!harmIndex) { + sineSign = 0; + } + + for (k = noSubbands - 2; k != 0; k--) { + FIXP_DBL sinelevel = *pSineLevel++; + index++; + if (((signalReal = (sineSign ? -sinelevel : sinelevel)) == + FL2FXCONST_DBL(0.0f)) && + !noNoiseFlag) { + /* Add noisefloor to the amplified signal */ + index &= (SBR_NF_NO_RANDOM_VAL - 1); + signalReal += (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], + pNoiseLevel[0]) + << 4); + } + + /* The next multiplication constitutes the actual envelope adjustment of + * the signal. */ + signalReal += fMultDiv2(*ptrReal, *pGain++) << ((int)scale_change); + + pNoiseLevel++; + *ptrReal++ = signalReal; + } /* for ... */ + } else { + /* harmIndex 1,3 in combination with freqInvFlag */ + if (harmIndex == 1) freqInvFlag = !freqInvFlag; + + for (k = noSubbands - 2; k != 0; k--) { + index++; + /* The next multiplication constitutes the actual envelope adjustment of + * the signal. */ + signalReal = fMultDiv2(*ptrReal, *pGain++) << ((int)scale_change); + + if (*pSineLevel++ != FL2FXCONST_DBL(0.0f)) + tone_count++; + else if (!noNoiseFlag) { + /* Add noisefloor to the amplified signal */ + index &= (SBR_NF_NO_RANDOM_VAL - 1); + signalReal += (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], + pNoiseLevel[0]) + << 4); + } + + pNoiseLevel++; + + if (tone_count <= 16) { + FIXP_DBL addSine = fMultDiv2((pSineLevel[-2] - pSineLevel[0]), C1); + signalReal += (freqInvFlag) ? (-addSine) : (addSine); + } + + *ptrReal++ = signalReal; + freqInvFlag = !freqInvFlag; + } /* for ... */ + } + } + + if (noSubbands > -1) { + index++; + /* The next multiplication constitutes the actual envelope adjustment of the + * signal. */ + signalReal = fMultDiv2(*ptrReal, *pGain) << ((int)scale_change); + sineLevelPrev = fMultDiv2(pSineLevel[-1], FL2FX_SGL(0.0163f)); + sineLevel = pSineLevel[0]; + + if (pSineLevel[0] != FL2FXCONST_DBL(0.0f)) + tone_count++; + else if (!noNoiseFlag) { + /* Add noisefloor to the amplified signal */ + index &= (SBR_NF_NO_RANDOM_VAL - 1); + signalReal = + signalReal + + (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0]) + << 4); + } + + if (!(harmIndex & 0x1)) { + /* harmIndex 0,2 */ + *ptrReal = signalReal + ((sineSign) ? -sineLevel : sineLevel); + } else { + /* harmIndex 1,3 in combination with freqInvFlag */ + if (tone_count <= 16) { + if (freqInvFlag) { + *ptrReal++ = signalReal - sineLevelPrev; + if (noSubbands + lowSubband < 63) + *ptrReal = *ptrReal + fMultDiv2(C1, sineLevel); + } else { + *ptrReal++ = signalReal + sineLevelPrev; + if (noSubbands + lowSubband < 63) + *ptrReal = *ptrReal - fMultDiv2(C1, sineLevel); + } + } else + *ptrReal = signalReal; + } + } + *ptrHarmIndex = (harmIndex + 1) & 3; + *ptrPhaseIndex = index & (SBR_NF_NO_RANDOM_VAL - 1); +} + +static void adjustTimeSlotHQ_GainAndNoise( + FIXP_DBL *RESTRICT + ptrReal, /*!< Subband samples to be adjusted, real part */ + FIXP_DBL *RESTRICT + ptrImag, /*!< Subband samples to be adjusted, imag part */ + HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, ENV_CALC_NRGS *nrgs, + int lowSubband, /*!< Lowest QMF-channel in the currently used SBR range. */ + int noSubbands, /*!< Number of QMF subbands */ + int scale_change, /*!< Number of bits to shift adjusted samples */ + FIXP_SGL smooth_ratio, /*!< Impact of last envelope */ + int noNoiseFlag, /*!< Start index to random number array */ + int filtBufferNoiseShift) /*!< Shift factor of filtBufferNoise */ +{ + FIXP_DBL *RESTRICT gain = nrgs->nrgGain; /*!< Gains of current envelope */ + FIXP_DBL *RESTRICT noiseLevel = + nrgs->noiseLevel; /*!< Noise levels of current envelope */ + FIXP_DBL *RESTRICT pSineLevel = nrgs->nrgSine; /*!< Sine levels */ + + FIXP_DBL *RESTRICT filtBuffer = + h_sbr_cal_env->filtBuffer; /*!< Gains of last envelope */ + FIXP_DBL *RESTRICT filtBufferNoise = + h_sbr_cal_env->filtBufferNoise; /*!< Noise levels of last envelope */ + int *RESTRICT ptrPhaseIndex = + &h_sbr_cal_env->phaseIndex; /*!< Start index to random number array */ + + int k; + FIXP_DBL signalReal, signalImag; + FIXP_DBL noiseReal, noiseImag; + FIXP_DBL smoothedGain, smoothedNoise; + FIXP_SGL direct_ratio = + /*FL2FXCONST_SGL(1.0f) */ (FIXP_SGL)MAXVAL_SGL - smooth_ratio; + int index = *ptrPhaseIndex; + int shift; + + *ptrPhaseIndex = (index + noSubbands) & (SBR_NF_NO_RANDOM_VAL - 1); + + filtBufferNoiseShift += + 1; /* due to later use of fMultDiv2 instead of fMult */ + if (filtBufferNoiseShift < 0) { + shift = fixMin(DFRACT_BITS - 1, -filtBufferNoiseShift); + } else { + shift = fixMin(DFRACT_BITS - 1, filtBufferNoiseShift); + } + + if (smooth_ratio > FL2FXCONST_SGL(0.0f)) { + for (k = 0; k < noSubbands; k++) { + /* + Smoothing: The old envelope has been bufferd and a certain ratio + of the old gains and noise levels is used. + */ + smoothedGain = + fMult(smooth_ratio, filtBuffer[k]) + fMult(direct_ratio, gain[k]); + + if (filtBufferNoiseShift < 0) { + smoothedNoise = (fMultDiv2(smooth_ratio, filtBufferNoise[k]) >> shift) + + fMult(direct_ratio, noiseLevel[k]); + } else { + smoothedNoise = (fMultDiv2(smooth_ratio, filtBufferNoise[k]) << shift) + + fMult(direct_ratio, noiseLevel[k]); + } + + /* + The next 2 multiplications constitute the actual envelope adjustment + of the signal and should be carried out with full accuracy + (supplying #DFRACT_BITS valid bits). + */ + signalReal = fMultDiv2(*ptrReal, smoothedGain) << ((int)scale_change); + signalImag = fMultDiv2(*ptrImag, smoothedGain) << ((int)scale_change); + + index++; + + if ((pSineLevel[k] != FL2FXCONST_DBL(0.0f)) || noNoiseFlag) { + /* Just the amplified signal is saved */ + *ptrReal++ = signalReal; + *ptrImag++ = signalImag; + } else { + /* Add noisefloor to the amplified signal */ + index &= (SBR_NF_NO_RANDOM_VAL - 1); + noiseReal = + fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], smoothedNoise) + << 4; + noiseImag = + fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][1], smoothedNoise) + << 4; + *ptrReal++ = (signalReal + noiseReal); + *ptrImag++ = (signalImag + noiseImag); + } + } + } else { + for (k = 0; k < noSubbands; k++) { + smoothedGain = gain[k]; + signalReal = fMultDiv2(*ptrReal, smoothedGain) << scale_change; + signalImag = fMultDiv2(*ptrImag, smoothedGain) << scale_change; + + index++; + + if ((pSineLevel[k] == FL2FXCONST_DBL(0.0f)) && (noNoiseFlag == 0)) { + /* Add noisefloor to the amplified signal */ + smoothedNoise = noiseLevel[k]; + index &= (SBR_NF_NO_RANDOM_VAL - 1); + noiseReal = + fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], smoothedNoise); + noiseImag = + fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][1], smoothedNoise); + + /* FDK_sbrDecoder_sbr_randomPhase is downscaled by 2^3 */ + signalReal += noiseReal << 4; + signalImag += noiseImag << 4; + } + *ptrReal++ = signalReal; + *ptrImag++ = signalImag; + } + } +} + +static void adjustTimeSlotHQ_AddHarmonics( + FIXP_DBL *RESTRICT + ptrReal, /*!< Subband samples to be adjusted, real part */ + FIXP_DBL *RESTRICT + ptrImag, /*!< Subband samples to be adjusted, imag part */ + HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, ENV_CALC_NRGS *nrgs, + int lowSubband, /*!< Lowest QMF-channel in the currently used SBR range. */ + int noSubbands, /*!< Number of QMF subbands */ + int scale_change /*!< Scale mismatch between QMF input and sineLevel + exponent. */ +) { + FIXP_DBL *RESTRICT pSineLevel = nrgs->nrgSine; /*!< Sine levels */ + UCHAR *RESTRICT ptrHarmIndex = + &h_sbr_cal_env->harmIndex; /*!< Harmonic index */ + + int k; + FIXP_DBL signalReal, signalImag; + UCHAR harmIndex = *ptrHarmIndex; + int freqInvFlag = (lowSubband & 1); + FIXP_DBL sineLevel; + + *ptrHarmIndex = (harmIndex + 1) & 3; + + for (k = 0; k < noSubbands; k++) { + sineLevel = pSineLevel[k]; + freqInvFlag ^= 1; + if (sineLevel != FL2FXCONST_DBL(0.f)) { + signalReal = ptrReal[k]; + signalImag = ptrImag[k]; + sineLevel = scaleValue(sineLevel, scale_change); + if (harmIndex & 2) { + /* case 2,3 */ + sineLevel = -sineLevel; + } + if (!(harmIndex & 1)) { + /* case 0,2: */ + ptrReal[k] = signalReal + sineLevel; + } else { + /* case 1,3 */ + if (!freqInvFlag) sineLevel = -sineLevel; + ptrImag[k] = signalImag + sineLevel; + } + } + } +} + +static void adjustTimeSlotHQ( + FIXP_DBL *RESTRICT + ptrReal, /*!< Subband samples to be adjusted, real part */ + FIXP_DBL *RESTRICT + ptrImag, /*!< Subband samples to be adjusted, imag part */ + HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, ENV_CALC_NRGS *nrgs, + int lowSubband, /*!< Lowest QMF-channel in the currently used SBR range. */ + int noSubbands, /*!< Number of QMF subbands */ + int scale_change, /*!< Number of bits to shift adjusted samples */ + FIXP_SGL smooth_ratio, /*!< Impact of last envelope */ + int noNoiseFlag, /*!< Start index to random number array */ + int filtBufferNoiseShift) /*!< Shift factor of filtBufferNoise */ +{ + FIXP_DBL *RESTRICT gain = nrgs->nrgGain; /*!< Gains of current envelope */ + FIXP_DBL *RESTRICT noiseLevel = + nrgs->noiseLevel; /*!< Noise levels of current envelope */ + FIXP_DBL *RESTRICT pSineLevel = nrgs->nrgSine; /*!< Sine levels */ + + FIXP_DBL *RESTRICT filtBuffer = + h_sbr_cal_env->filtBuffer; /*!< Gains of last envelope */ + FIXP_DBL *RESTRICT filtBufferNoise = + h_sbr_cal_env->filtBufferNoise; /*!< Noise levels of last envelope */ + UCHAR *RESTRICT ptrHarmIndex = + &h_sbr_cal_env->harmIndex; /*!< Harmonic index */ + int *RESTRICT ptrPhaseIndex = + &h_sbr_cal_env->phaseIndex; /*!< Start index to random number array */ + + int k; + FIXP_DBL signalReal, signalImag; + FIXP_DBL noiseReal, noiseImag; + FIXP_DBL smoothedGain, smoothedNoise; + FIXP_SGL direct_ratio = + /*FL2FXCONST_SGL(1.0f) */ (FIXP_SGL)MAXVAL_SGL - smooth_ratio; + int index = *ptrPhaseIndex; + UCHAR harmIndex = *ptrHarmIndex; + int freqInvFlag = (lowSubband & 1); + FIXP_DBL sineLevel; + int shift; + + *ptrPhaseIndex = (index + noSubbands) & (SBR_NF_NO_RANDOM_VAL - 1); + *ptrHarmIndex = (harmIndex + 1) & 3; + + /* + Possible optimization: + smooth_ratio and harmIndex stay constant during the loop. + It might be faster to include a separate loop in each path. + + the check for smooth_ratio is now outside the loop and the workload + of the whole function decreased by about 20 % + */ + + filtBufferNoiseShift += + 1; /* due to later use of fMultDiv2 instead of fMult */ + if (filtBufferNoiseShift < 0) + shift = fixMin(DFRACT_BITS - 1, -filtBufferNoiseShift); + else + shift = fixMin(DFRACT_BITS - 1, filtBufferNoiseShift); + + if (smooth_ratio > FL2FXCONST_SGL(0.0f)) { + for (k = 0; k < noSubbands; k++) { + /* + Smoothing: The old envelope has been bufferd and a certain ratio + of the old gains and noise levels is used. + */ + + smoothedGain = + fMult(smooth_ratio, filtBuffer[k]) + fMult(direct_ratio, gain[k]); + + if (filtBufferNoiseShift < 0) { + smoothedNoise = (fMultDiv2(smooth_ratio, filtBufferNoise[k]) >> shift) + + fMult(direct_ratio, noiseLevel[k]); + } else { + smoothedNoise = (fMultDiv2(smooth_ratio, filtBufferNoise[k]) << shift) + + fMult(direct_ratio, noiseLevel[k]); + } + + /* + The next 2 multiplications constitute the actual envelope adjustment + of the signal and should be carried out with full accuracy + (supplying #DFRACT_BITS valid bits). + */ + signalReal = fMultDiv2(*ptrReal, smoothedGain) << ((int)scale_change); + signalImag = fMultDiv2(*ptrImag, smoothedGain) << ((int)scale_change); + + index++; + + if (pSineLevel[k] != FL2FXCONST_DBL(0.0f)) { + sineLevel = pSineLevel[k]; + + switch (harmIndex) { + case 0: + *ptrReal++ = (signalReal + sineLevel); + *ptrImag++ = (signalImag); + break; + case 2: + *ptrReal++ = (signalReal - sineLevel); + *ptrImag++ = (signalImag); + break; + case 1: + *ptrReal++ = (signalReal); + if (freqInvFlag) + *ptrImag++ = (signalImag - sineLevel); + else + *ptrImag++ = (signalImag + sineLevel); + break; + case 3: + *ptrReal++ = signalReal; + if (freqInvFlag) + *ptrImag++ = (signalImag + sineLevel); + else + *ptrImag++ = (signalImag - sineLevel); + break; + } + } else { + if (noNoiseFlag) { + /* Just the amplified signal is saved */ + *ptrReal++ = (signalReal); + *ptrImag++ = (signalImag); + } else { + /* Add noisefloor to the amplified signal */ + index &= (SBR_NF_NO_RANDOM_VAL - 1); + /* FDK_sbrDecoder_sbr_randomPhase is downscaled by 2^3 */ + noiseReal = + fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], smoothedNoise) + << 4; + noiseImag = + fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][1], smoothedNoise) + << 4; + *ptrReal++ = (signalReal + noiseReal); + *ptrImag++ = (signalImag + noiseImag); + } + } + freqInvFlag ^= 1; + } + + } else { + for (k = 0; k < noSubbands; k++) { + smoothedGain = gain[k]; + signalReal = fMultDiv2(*ptrReal, smoothedGain) << scale_change; + signalImag = fMultDiv2(*ptrImag, smoothedGain) << scale_change; + + index++; + + if ((sineLevel = pSineLevel[k]) != FL2FXCONST_DBL(0.0f)) { + switch (harmIndex) { + case 0: + signalReal += sineLevel; + break; + case 1: + if (freqInvFlag) + signalImag -= sineLevel; + else + signalImag += sineLevel; + break; + case 2: + signalReal -= sineLevel; + break; + case 3: + if (freqInvFlag) + signalImag += sineLevel; + else + signalImag -= sineLevel; + break; + } + } else { + if (noNoiseFlag == 0) { + /* Add noisefloor to the amplified signal */ + smoothedNoise = noiseLevel[k]; + index &= (SBR_NF_NO_RANDOM_VAL - 1); + noiseReal = fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], + smoothedNoise); + noiseImag = fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][1], + smoothedNoise); + + /* FDK_sbrDecoder_sbr_randomPhase is downscaled by 2^3 */ + signalReal += noiseReal << 4; + signalImag += noiseImag << 4; + } + } + *ptrReal++ = signalReal; + *ptrImag++ = signalImag; + + freqInvFlag ^= 1; + } + } +} + +/*! + \brief Reset limiter bands. + + Build frequency band table for the gain limiter dependent on + the previously generated transposer patch areas. + + \return SBRDEC_OK if ok, SBRDEC_UNSUPPORTED_CONFIG on error +*/ +SBR_ERROR +ResetLimiterBands( + UCHAR *limiterBandTable, /*!< Resulting band borders in QMF channels */ + UCHAR *noLimiterBands, /*!< Resulting number of limiter band */ + UCHAR *freqBandTable, /*!< Table with possible band borders */ + int noFreqBands, /*!< Number of bands in freqBandTable */ + const PATCH_PARAM *patchParam, /*!< Transposer patch parameters */ + int noPatches, /*!< Number of transposer patches */ + int limiterBands, /*!< Selected 'band density' from bitstream */ + UCHAR sbrPatchingMode, int xOverQmf[MAX_NUM_PATCHES], int b41Sbr) { + int i, k, isPatchBorder[2], loLimIndex, hiLimIndex, tempNoLim, nBands; + UCHAR workLimiterBandTable[MAX_FREQ_COEFFS / 2 + MAX_NUM_PATCHES + 1]; + int patchBorders[MAX_NUM_PATCHES + 1]; + int kx, k2; + + int lowSubband = freqBandTable[0]; + int highSubband = freqBandTable[noFreqBands]; + + /* 1 limiter band. */ + if (limiterBands == 0) { + limiterBandTable[0] = 0; + limiterBandTable[1] = highSubband - lowSubband; + nBands = 1; + } else { + if (!sbrPatchingMode && xOverQmf != NULL) { + noPatches = 0; + + if (b41Sbr == 1) { + for (i = 1; i < MAX_NUM_PATCHES_HBE; i++) + if (xOverQmf[i] != 0) noPatches++; + } else { + for (i = 1; i < MAX_STRETCH_HBE; i++) + if (xOverQmf[i] != 0) noPatches++; + } + for (i = 0; i < noPatches; i++) { + patchBorders[i] = xOverQmf[i] - lowSubband; + } + } else { + for (i = 0; i < noPatches; i++) { + patchBorders[i] = patchParam[i].guardStartBand - lowSubband; + } + } + patchBorders[i] = highSubband - lowSubband; + + /* 1.2, 2, or 3 limiter bands/octave plus bandborders at patchborders. */ + for (k = 0; k <= noFreqBands; k++) { + workLimiterBandTable[k] = freqBandTable[k] - lowSubband; + } + for (k = 1; k < noPatches; k++) { + workLimiterBandTable[noFreqBands + k] = patchBorders[k]; + } + + tempNoLim = nBands = noFreqBands + noPatches - 1; + shellsort(workLimiterBandTable, tempNoLim + 1); + + loLimIndex = 0; + hiLimIndex = 1; + + while (hiLimIndex <= tempNoLim) { + FIXP_DBL div_m, oct_m, temp; + INT div_e = 0, oct_e = 0, temp_e = 0; + + k2 = workLimiterBandTable[hiLimIndex] + lowSubband; + kx = workLimiterBandTable[loLimIndex] + lowSubband; + + div_m = fDivNorm(k2, kx, &div_e); + + /* calculate number of octaves */ + oct_m = fLog2(div_m, div_e, &oct_e); + + /* multiply with limiterbands per octave */ + /* values 1, 1.2, 2, 3 -> scale factor of 2 */ + temp = fMultNorm( + oct_m, FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4_DBL[limiterBands], + &temp_e); + + /* overall scale factor of temp ist addition of scalefactors from log2 + calculation, limiter bands scalefactor (2) and limiter bands + multiplication */ + temp_e += oct_e + 2; + + /* div can be a maximum of 64 (k2 = 64 and kx = 1) + -> oct can be a maximum of 6 + -> temp can be a maximum of 18 (as limiterBandsPerOctoave is a maximum + factor of 3) + -> we need a scale factor of 5 for comparisson + */ + if (temp >> (5 - temp_e) < FL2FXCONST_DBL(0.49f) >> 5) { + if (workLimiterBandTable[hiLimIndex] == + workLimiterBandTable[loLimIndex]) { + workLimiterBandTable[hiLimIndex] = highSubband; + nBands--; + hiLimIndex++; + continue; + } + isPatchBorder[0] = isPatchBorder[1] = 0; + for (k = 0; k <= noPatches; k++) { + if (workLimiterBandTable[hiLimIndex] == patchBorders[k]) { + isPatchBorder[1] = 1; + break; + } + } + if (!isPatchBorder[1]) { + workLimiterBandTable[hiLimIndex] = highSubband; + nBands--; + hiLimIndex++; + continue; + } + for (k = 0; k <= noPatches; k++) { + if (workLimiterBandTable[loLimIndex] == patchBorders[k]) { + isPatchBorder[0] = 1; + break; + } + } + if (!isPatchBorder[0]) { + workLimiterBandTable[loLimIndex] = highSubband; + nBands--; + } + } + loLimIndex = hiLimIndex; + hiLimIndex++; + } + shellsort(workLimiterBandTable, tempNoLim + 1); + + /* Test if algorithm exceeded maximum allowed limiterbands */ + if (nBands > MAX_NUM_LIMITERS || nBands <= 0) { + return SBRDEC_UNSUPPORTED_CONFIG; + } + + /* Copy limiterbands from working buffer into final destination */ + for (k = 0; k <= nBands; k++) { + limiterBandTable[k] = workLimiterBandTable[k]; + } + } + *noLimiterBands = nBands; + + return SBRDEC_OK; +} |