| |
| /* ----------------------------------------------------------------------------------------------------------- |
| Software License for The Fraunhofer FDK AAC Codec Library for Android |
| |
| © Copyright 1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. |
| All rights reserved. |
| |
| 1. INTRODUCTION |
| The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements |
| the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. |
| This FDK AAC Codec software is intended to be used on a wide variety of Android devices. |
| |
| AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual |
| audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by |
| independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part |
| of the MPEG specifications. |
| |
| Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) |
| may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners |
| individually for the purpose of encoding or decoding bit streams in products that are compliant with |
| the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license |
| these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec |
| software may already be covered under those patent licenses when it is used for those licensed purposes only. |
| |
| Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, |
| are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional |
| applications information and documentation. |
| |
| 2. COPYRIGHT LICENSE |
| |
| Redistribution and use in source and binary forms, with or without modification, are permitted without |
| payment of copyright license fees provided that you satisfy the following conditions: |
| |
| You must retain the complete text of this software license in redistributions of the FDK AAC Codec or |
| your modifications thereto in source code form. |
| |
| You must retain the complete text of this software license in the documentation and/or other materials |
| provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. |
| You must make available free of charge copies of the complete source code of the FDK AAC Codec and your |
| modifications thereto to recipients of copies in binary form. |
| |
| The name of Fraunhofer may not be used to endorse or promote products derived from this library without |
| prior written permission. |
| |
| You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec |
| software or your modifications thereto. |
| |
| Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software |
| and the date of any change. For modified versions of the FDK AAC Codec, the term |
| "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term |
| "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." |
| |
| 3. NO PATENT LICENSE |
| |
| NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, |
| ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with |
| respect to this software. |
| |
| You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized |
| by appropriate patent licenses. |
| |
| 4. DISCLAIMER |
| |
| This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors |
| "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties |
| of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR |
| CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, |
| including but not limited to procurement of substitute goods or services; loss of use, data, or profits, |
| or business interruption, however caused and on any theory of liability, whether in contract, strict |
| liability, or tort (including negligence), arising in any way out of the use of this software, even if |
| advised of the possibility of such damage. |
| |
| 5. CONTACT INFORMATION |
| |
| Fraunhofer Institute for Integrated Circuits IIS |
| Attention: Audio and Multimedia Departments - FDK AAC LL |
| Am Wolfsmantel 33 |
| 91058 Erlangen, Germany |
| |
| www.iis.fraunhofer.de/amm |
| amm-info@iis.fraunhofer.de |
| ----------------------------------------------------------------------------------------------------------- */ |
| |
| /*! |
| \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 */ |
| |
| #if defined(__arm__) |
| #include "arm/env_calc_arm.cpp" |
| #endif |
| |
| 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]; |
| } |
| 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 adjustTimeSlotLC(FIXP_DBL *ptrReal, |
| ENV_CALC_NRGS* nrgs, |
| UCHAR *ptrHarmIndex, |
| int lowSubbands, |
| int noSubbands, |
| int scale_change, |
| int noNoiseFlag, |
| int *ptrPhaseIndex, |
| int fCldfb); |
| /*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 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 #MAX_ENVELOPES 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 */ |
| UCHAR *addHarmonics, /*!< vector with 1 flag per sfb */ |
| int *harmFlagsPrev, /*!< Packed 'addHarmonics' */ |
| int tranEnv, /*!< Transient position */ |
| SCHAR *sineMapped) /*!< Resulting vector of sine start positions for each QMF band */ |
| |
| { |
| int i; |
| int lowSubband2 = freqBandTable[0]<<1; |
| int bitcount = 0; |
| int oldflags = *harmFlagsPrev; |
| int newflags = 0; |
| |
| /* |
| Format of harmFlagsPrev: |
| |
| first word = flags for highest 16 sfb bands in use |
| second word = flags for next lower 16 sfb bands (if present) |
| third word = flags for lowest 16 sfb bands (if present) |
| |
| Up to MAX_FREQ_COEFFS sfb bands can be flagged for a sign. |
| The lowest bit of the first word corresponds to the _highest_ sfb band in use. |
| This is ensures that each flag is mapped to the same QMF band even after a |
| change of the crossover-frequency. |
| */ |
| |
| |
| /* Reset the output vector first */ |
| FDKmemset(sineMapped, MAX_ENVELOPES,MAX_FREQ_COEFFS); /* MAX_ENVELOPES means 'no sine' */ |
| |
| freqBandTable += nSfb; |
| addHarmonics += nSfb-1; |
| |
| for (i=nSfb; i!=0; i--) { |
| int ui = *freqBandTable--; /* Upper limit of the current scale factor band. */ |
| int li = *freqBandTable; /* Lower limit of the current scale factor band. */ |
| |
| if ( *addHarmonics-- ) { /* There is a sine in this band */ |
| |
| unsigned int mask = 1 << bitcount; |
| newflags |= mask; /* Set flag */ |
| |
| /* |
| If there was a sine in the last frame, let it continue from the first envelope on |
| else start at the transient position. |
| */ |
| sineMapped[(ui+li-lowSubband2) >> 1] = ( oldflags & mask ) ? 0 : tranEnv; |
| } |
| |
| if ((++bitcount == 16) || i==1) { |
| bitcount = 0; |
| *harmFlagsPrev++ = newflags; |
| oldflags = *harmFlagsPrev; /* Fetch 16 of the old flags */ |
| newflags = 0; |
| } |
| } |
| } |
| |
| |
| /*! |
| \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, |
| int* useAliasReduction, /*!< synthetic sine engergy 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; |
| } |
| } |
| } |
| |
| |
| /* Convert headroom bits to exponent */ |
| #define SCALE2EXP(s) (15-(s)) |
| #define EXP2SCALE(e) (15-(e)) |
| |
| /*! |
| \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 */ |
| 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, j, envNoise = 0; |
| UCHAR* borders = hFrameData->frameInfo.borders; |
| |
| FIXP_SGL *noiseLevels = hFrameData->sbrNoiseFloorLevel; |
| HANDLE_FREQ_BAND_DATA hFreq = &hHeaderData->freqBandData; |
| |
| int lowSubband = hFreq->lowSubband; |
| int highSubband = hFreq->highSubband; |
| int noSubbands = highSubband - lowSubband; |
| |
| int noNoiseBands = hFreq->nNfb; |
| int no_cols = hHeaderData->numberTimeSlots * hHeaderData->timeStep; |
| UCHAR first_start = borders[0] * 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; |
| |
| SCHAR maxGainLimit_e = (frameErrorFlag) ? MAX_GAIN_CONCEAL_EXP : MAX_GAIN_EXP; |
| |
| int useAliasReduction[64]; |
| UCHAR smooth_length = 0; |
| |
| FIXP_SGL * pIenv = hFrameData->iEnvelope; |
| |
| /* |
| Extract sine flags for all QMF bands |
| */ |
| mapSineFlags(hFreq->freqBandTable[1], |
| hFreq->nSfb[1], |
| hFrameData->addHarmonics, |
| h_sbr_cal_env->harmFlagsPrev, |
| 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. |
| */ |
| |
| 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=hFreq->nSfb[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 = (maxSfbNrg_e > adj_e) ? maxSfbNrg_e : adj_e; |
| |
| if (borders[i+1] > hHeaderData->numberTimeSlots) |
| /* This envelope affects timeslots after the output frame */ |
| final_e = (maxSfbNrg_e > final_e) ? maxSfbNrg_e : final_e; |
| |
| } |
| |
| /* |
| Calculate adjustment factors and apply them for every envelope. |
| */ |
| pIenv = hFrameData->iEnvelope; |
| |
| for (i = 0; i < hFrameData->frameInfo.nEnvelopes; i++) { |
| |
| int k, noNoiseFlag; |
| SCHAR noise_e, input_e = SCALE2EXP(sbrScaleFactor->hb_scale); |
| C_ALLOC_SCRATCH_START(pNrgs, ENV_CALC_NRGS, 1); |
| |
| /* |
| Helper variables. |
| */ |
| UCHAR start_pos = hHeaderData->timeStep * borders[i]; /* Start-position in time (subband sample) for current envelope. */ |
| UCHAR stop_pos = hHeaderData->timeStep * borders[i+1]; /* Stop-position in time (subband sample) for current envelope. */ |
| UCHAR freq_res = hFrameData->frameInfo.freqRes[i]; /* Frequency resolution for current envelope. */ |
| |
| |
| /* Always do 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 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, |
| hFreq->nSfb[freq_res], |
| hFreq->freqBandTable[freq_res], |
| start_pos, stop_pos, |
| input_e, |
| pNrgs->nrgEst, |
| pNrgs->nrgEst_e); |
| |
| /* |
| Calculate subband gains |
| */ |
| { |
| UCHAR * table = hFreq->freqBandTable[freq_res]; |
| UCHAR * pUiNoise = &hFreq->freqBandTableNoise[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; |
| for (j = 0; j < hFreq->nSfb[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; |
| |
| 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]; |
| |
| /* 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); |
| |
| adjustTimeSlotHQ(&analysBufferReal[j][lowSubband], |
| &analysBufferImag[j][lowSubband], |
| h_sbr_cal_env, |
| pNrgs, |
| lowSubband, |
| noSubbands, |
| scale_change, |
| smooth_ratio, |
| noNoiseFlag, |
| filtBufferNoiseShift); |
| } |
| else |
| { |
| adjustTimeSlotLC(&analysBufferReal[j][lowSubband], |
| pNrgs, |
| &h_sbr_cal_env->harmIndex, |
| lowSubband, |
| noSubbands, |
| scale_change, |
| noNoiseFlag, |
| &h_sbr_cal_env->phaseIndex, |
| (flags & SBRDEC_ELD_GRID)); |
| } |
| } // for |
| |
| 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); |
| } |
| |
| /* Rescale output samples */ |
| { |
| FIXP_DBL maxVal; |
| int ov_reserve, reserve; |
| |
| /* Determine headroom in old adjusted samples */ |
| maxVal = maxSubbandSample( analysBufferReal, |
| (useLP) ? NULL : analysBufferImag, |
| lowSubband, |
| 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 */ |
| if (ov_adj_e - ov_reserve > adj_e - reserve ) /* set output_e to the maximum */ |
| output_e = ov_adj_e - ov_reserve; |
| else |
| output_e = adj_e - reserve; |
| |
| /* Rescale old samples */ |
| rescaleSubbandSamples( analysBufferReal, |
| (useLP) ? NULL : analysBufferImag, |
| lowSubband, 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: */ |
| sbrScaleFactor->ov_hb_scale = EXP2SCALE(final_e); |
| |
| |
| /* We need to remeber 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; |
| |
| } |
| |
| |
| /*! |
| \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<(MAX_FREQ_COEFFS+15)>>4; i++) { |
| hs->harmFlagsPrev[i] = 0; |
| } |
| hs->harmIndex = 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); |
| } |
| } |
| } |
| } |
| |
| |
| /*! |
| \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++) |
| { |
| #ifdef FUNCTION_FDK_get_maxval |
| maxVal = FDK_get_maxval(maxVal, &re[l][lowSubband], &im[l][lowSubband], width); |
| #else |
| 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); |
| #endif |
| } |
| } else |
| { |
| for (int l=start_pos; l<next_pos; l++) { |
| int k=width; |
| FIXP_DBL *reTmp = &re[l][lowSubband]; |
| do{ |
| FIXP_DBL tmp = *(reTmp++); |
| maxVal |= (FIXP_DBL)((LONG)(tmp)^((LONG)tmp>>(DFRACT_BITS-1))); |
| }while(--k!=0); |
| } |
| } |
| } |
| |
| return(maxVal); |
| } |
| |
| #define SHIFT_BEFORE_SQUARE (3) /* (7/2) */ |
| /*!< |
| 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,l; |
| |
| /* 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))+(6))]; |
| FIXP_DBL bufferImag[(((1024)/(32))+(6))]; |
| FIXP_DBL maxVal = FL2FX_DBL(0.0f); |
| |
| if (analysBufferImag!=NULL) |
| { |
| 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 |
| { |
| 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 = FL2FXCONST_DBL(0.0f); |
| preShift = CntLeadingZeros(maxVal)-1; |
| preShift -= SHIFT_BEFORE_SQUARE; |
| |
| if (preShift>=0) { |
| 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 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; |
| } |
| |
| |
| /*! |
| \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 */ |
| int fCldfb) /*!< CLDFB 80 flag */ |
| { |
| 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 (fCldfb) { |
| |
| 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 = scaleValue( fMultDiv2(C1_CLDFB, sineLevel), -shift ); |
| |
| FIXP_DBL tmp2 = fMultDiv2(C1_CLDFB, 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; |
| } |
| |
| } else |
| { |
| 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); |
| } |
| 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; |
| register 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); |
| 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); |
| 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 */ |
| { |
| 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; |
| FIXP_DBL temp; |
| |
| int lowSubband = freqBandTable[0]; |
| int highSubband = freqBandTable[noFreqBands]; |
| |
| /* 1 limiter band. */ |
| if(limiterBands == 0) { |
| limiterBandTable[0] = 0; |
| limiterBandTable[1] = highSubband - lowSubband; |
| nBands = 1; |
| } 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) { |
| k2 = workLimiterBandTable[hiLimIndex] + lowSubband; |
| kx = workLimiterBandTable[loLimIndex] + lowSubband; |
| |
| temp = FX_SGL2FX_DBL(FDK_getNumOctavesDiv8(kx,k2)); /* Number of octaves */ |
| temp = fMult(temp, FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4[limiterBands]); |
| |
| if (temp < 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; |
| } |
| |
