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Diffstat (limited to 'libFDK/src/arm/qmf_arm.cpp')
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diff --git a/libFDK/src/arm/qmf_arm.cpp b/libFDK/src/arm/qmf_arm.cpp new file mode 100644 index 0000000..39795d7 --- /dev/null +++ b/libFDK/src/arm/qmf_arm.cpp @@ -0,0 +1,766 @@ + +/* ----------------------------------------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------------------------------------- */ + +#if (QMF_NO_POLY==5) + +#define FUNCTION_qmfForwardModulationLP_odd + +#ifdef FUNCTION_qmfForwardModulationLP_odd +static void +qmfForwardModulationLP_odd( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */ + const FIXP_QMF *timeIn, /*!< Time Signal */ + FIXP_QMF *rSubband ) /*!< Real Output */ +{ + int i; + int L = anaQmf->no_channels; + int M = L>>1; + int shift = (anaQmf->no_channels>>6) + 1; + int rSubband_e = 0; + + FIXP_QMF *rSubbandPtr0 = &rSubband[M+0]; /* runs with increment */ + FIXP_QMF *rSubbandPtr1 = &rSubband[M-1]; /* runs with decrement */ + FIXP_QMF *timeIn0 = (FIXP_DBL *) &timeIn[0]; /* runs with increment */ + FIXP_QMF *timeIn1 = (FIXP_DBL *) &timeIn[L]; /* runs with increment */ + FIXP_QMF *timeIn2 = (FIXP_DBL *) &timeIn[L-1]; /* runs with decrement */ + FIXP_QMF *timeIn3 = (FIXP_DBL *) &timeIn[2*L-1]; /* runs with decrement */ + + for (i = 0; i < M; i++) + { + *rSubbandPtr0++ = (*timeIn2-- >> 1) - (*timeIn0++ >> shift); + *rSubbandPtr1-- = (*timeIn1++ >> 1) + (*timeIn3-- >> shift); + } + + dct_IV(rSubband,L, &rSubband_e); +} +#endif /* FUNCTION_qmfForwardModulationLP_odd */ + + +/* NEON optimized QMF currently builts only with RVCT toolchain */ + +#if defined(__ARM_ARCH_6__) || defined(__ARM_ARCH_5TE__) + +#if (SAMPLE_BITS == 16) +#define FUNCTION_qmfAnaPrototypeFirSlot +#endif + +#ifdef FUNCTION_qmfAnaPrototypeFirSlot + +#if defined(__GNUC__) /* cppp replaced: elif */ + +inline INT SMULBB (const SHORT a, const LONG b) +{ + INT result ; + __asm__ ("smulbb %0, %1, %2" + : "=r" (result) + : "r" (a), "r" (b)) ; + return result ; +} +inline INT SMULBT (const SHORT a, const LONG b) +{ + INT result ; + __asm__ ("smulbt %0, %1, %2" + : "=r" (result) + : "r" (a), "r" (b)) ; + return result ; +} + +inline INT SMLABB(const LONG accu, const SHORT a, const LONG b) +{ + INT result ; + __asm__ ("smlabb %0, %1, %2,%3" + : "=r" (result) + : "r" (a), "r" (b), "r" (accu)) ; + return result; +} +inline INT SMLABT(const LONG accu, const SHORT a, const LONG b) +{ + INT result ; + __asm__ ("smlabt %0, %1, %2,%3" + : "=r" (result) + : "r" (a), "r" (b), "r" (accu)) ; + return result; +} +#endif /* compiler selection */ + + +void qmfAnaPrototypeFirSlot( FIXP_QMF *analysisBuffer, + int no_channels, /*!< Number channels of analysis filter */ + const FIXP_PFT *p_filter, + int p_stride, /*!< Stide of analysis filter */ + FIXP_QAS *RESTRICT pFilterStates + ) +{ + LONG *p_flt = (LONG *) p_filter; + LONG flt; + FIXP_QMF *RESTRICT pData_0 = analysisBuffer + 2*no_channels - 1; + FIXP_QMF *RESTRICT pData_1 = analysisBuffer; + + FIXP_QAS *RESTRICT sta_0 = (FIXP_QAS *)pFilterStates; + FIXP_QAS *RESTRICT sta_1 = (FIXP_QAS *)pFilterStates + (2*QMF_NO_POLY*no_channels) - 1; + + FIXP_DBL accu0, accu1; + FIXP_QAS sta0, sta1; + + int staStep1 = no_channels<<1; + int staStep2 = (no_channels<<3) - 1; /* Rewind one less */ + + if (p_stride == 1) + { + /* FIR filter 0 */ + flt = *p_flt++; + sta1 = *sta_1; sta_1 -= staStep1; + accu1 = SMULBB( sta1, flt); + sta1 = *sta_1; sta_1 -= staStep1; + accu1 = SMLABT( accu1, sta1, flt); + + flt = *p_flt++; + sta1 = *sta_1; sta_1 -= staStep1; + accu1 = SMLABB( accu1, sta1, flt); + sta1 = *sta_1; sta_1 -= staStep1; + accu1 = SMLABT( accu1, sta1, flt); + + flt = *p_flt++; + sta1 = *sta_1; sta_1 += staStep2; + accu1 = SMLABB( accu1, sta1, flt); + *pData_1++ = FX_DBL2FX_QMF(accu1<<1); + + /* FIR filters 1..63 127..65 or 1..31 63..33 */ + no_channels >>= 1; + for (; --no_channels; ) + { + sta0 = *sta_0; sta_0 += staStep1; /* 1,3,5, ... 29/61 */ + sta1 = *sta_1; sta_1 -= staStep1; + accu0 = SMULBT( sta0, flt); + accu1 = SMULBT( sta1, flt); + + flt = *p_flt++; + sta0 = *sta_0; sta_0 += staStep1; + sta1 = *sta_1; sta_1 -= staStep1; + accu0 = SMLABB( accu0, sta0, flt); + accu1 = SMLABB( accu1, sta1, flt); + + sta0 = *sta_0; sta_0 += staStep1; + sta1 = *sta_1; sta_1 -= staStep1; + accu0 = SMLABT( accu0, sta0, flt); + accu1 = SMLABT( accu1, sta1, flt); + + flt = *p_flt++; + sta0 = *sta_0; sta_0 += staStep1; + sta1 = *sta_1; sta_1 -= staStep1; + accu0 = SMLABB( accu0, sta0, flt); + accu1 = SMLABB( accu1, sta1, flt); + + sta0 = *sta_0; sta_0 -= staStep2; + sta1 = *sta_1; sta_1 += staStep2; + accu0 = SMLABT( accu0, sta0, flt); + accu1 = SMLABT( accu1, sta1, flt); + + *pData_0-- = FX_DBL2FX_QMF(accu0<<1); + *pData_1++ = FX_DBL2FX_QMF(accu1<<1); + + /* Same sequence as above, but mix B=bottom with T=Top */ + + flt = *p_flt++; + sta0 = *sta_0; sta_0 += staStep1; /* 2,4,6, ... 30/62 */ + sta1 = *sta_1; sta_1 -= staStep1; + accu0 = SMULBB( sta0, flt); + accu1 = SMULBB( sta1, flt); + + sta0 = *sta_0; sta_0 += staStep1; + sta1 = *sta_1; sta_1 -= staStep1; + accu0 = SMLABT( accu0, sta0, flt); + accu1 = SMLABT( accu1, sta1, flt); + + flt = *p_flt++; + sta0 = *sta_0; sta_0 += staStep1; + sta1 = *sta_1; sta_1 -= staStep1; + accu0 = SMLABB( accu0, sta0, flt); + accu1 = SMLABB( accu1, sta1, flt); + + sta0 = *sta_0; sta_0 += staStep1; + sta1 = *sta_1; sta_1 -= staStep1; + accu0 = SMLABT( accu0, sta0, flt); + accu1 = SMLABT( accu1, sta1, flt); + + flt = *p_flt++; + sta0 = *sta_0; sta_0 -= staStep2; + sta1 = *sta_1; sta_1 += staStep2; + accu0 = SMLABB( accu0, sta0, flt); + accu1 = SMLABB( accu1, sta1, flt); + + *pData_0-- = FX_DBL2FX_QMF(accu0<<1); + *pData_1++ = FX_DBL2FX_QMF(accu1<<1); + } + + /* FIR filter 31/63 and 33/65 */ + sta0 = *sta_0; sta_0 += staStep1; + sta1 = *sta_1; sta_1 -= staStep1; + accu0 = SMULBT( sta0, flt); + accu1 = SMULBT( sta1, flt); + + flt = *p_flt++; + sta0 = *sta_0; sta_0 += staStep1; + sta1 = *sta_1; sta_1 -= staStep1; + accu0 = SMLABB( accu0, sta0, flt); + accu1 = SMLABB( accu1, sta1, flt); + + sta0 = *sta_0; sta_0 += staStep1; + sta1 = *sta_1; sta_1 -= staStep1; + accu0 = SMLABT( accu0, sta0, flt); + accu1 = SMLABT( accu1, sta1, flt); + + flt = *p_flt++; + sta0 = *sta_0; sta_0 += staStep1; + sta1 = *sta_1; sta_1 -= staStep1; + accu0 = SMLABB( accu0, sta0, flt); + accu1 = SMLABB( accu1, sta1, flt); + + sta0 = *sta_0; sta_0 -= staStep2; + sta1 = *sta_1; sta_1 += staStep2; + accu0 = SMLABT( accu0, sta0, flt); + accu1 = SMLABT( accu1, sta1, flt); + + *pData_0-- = FX_DBL2FX_QMF(accu0<<1); + *pData_1++ = FX_DBL2FX_QMF(accu1<<1); + + /* FIR filter 32/64 */ + flt = *p_flt++; + sta0 = *sta_0; sta_0 += staStep1; + sta1 = *sta_1; sta_1 -= staStep1; + accu0 = SMULBB( sta0, flt); + accu1 = SMULBB( sta1, flt); + + sta0 = *sta_0; sta_0 += staStep1; + sta1 = *sta_1; sta_1 -= staStep1; + accu0 = SMLABT( accu0, sta0, flt); + accu1 = SMLABT( accu1, sta1, flt); + + flt = *p_flt++; + sta0 = *sta_0; sta_0 += staStep1; + sta1 = *sta_1; sta_1 -= staStep1; + accu0 = SMLABB( accu0, sta0, flt); + accu1 = SMLABB( accu1, sta1, flt); + + sta0 = *sta_0; sta_0 += staStep1; + sta1 = *sta_1; sta_1 -= staStep1; + accu0 = SMLABT( accu0, sta0, flt); + accu1 = SMLABT( accu1, sta1, flt); + + flt = *p_flt; + sta0 = *sta_0; + sta1 = *sta_1; + accu0 = SMLABB( accu0, sta0, flt); + accu1 = SMLABB( accu1, sta1, flt); + + *pData_0-- = FX_DBL2FX_QMF(accu0<<1); + *pData_1++ = FX_DBL2FX_QMF(accu1<<1); + } + else + { + int pfltStep = QMF_NO_POLY * (p_stride-1); + + flt = p_flt[0]; + sta1 = *sta_1; sta_1 -= staStep1; + accu1 = SMULBB( sta1, flt); + sta1 = *sta_1; sta_1 -= staStep1; + accu1 = SMLABT( accu1, sta1, flt); + + flt = p_flt[1]; + sta1 = *sta_1; sta_1 -= staStep1; + accu1 = SMLABB( accu1, sta1, flt); + sta1 = *sta_1; sta_1 -= staStep1; + accu1 = SMLABT( accu1, sta1, flt); + + flt = p_flt[2]; p_flt += pfltStep; + sta1 = *sta_1; sta_1 += staStep2; + accu1 = SMLABB( accu1, sta1, flt); + *pData_1++ = FX_DBL2FX_QMF(accu1<<1); + + /* FIR filters 1..63 127..65 or 1..31 63..33 */ + for (; --no_channels; ) + { + flt = p_flt[0]; + sta0 = *sta_0; sta_0 += staStep1; + sta1 = *sta_1; sta_1 -= staStep1; + accu0 = SMULBB( sta0, flt); + accu1 = SMULBB( sta1, flt); + + sta0 = *sta_0; sta_0 += staStep1; + sta1 = *sta_1; sta_1 -= staStep1; + accu0 = SMLABT( accu0, sta0, flt); + accu1 = SMLABT( accu1, sta1, flt); + + flt = p_flt[1]; + sta0 = *sta_0; sta_0 += staStep1; + sta1 = *sta_1; sta_1 -= staStep1; + accu0 = SMLABB( accu0, sta0, flt); + accu1 = SMLABB( accu1, sta1, flt); + + sta0 = *sta_0; sta_0 += staStep1; + sta1 = *sta_1; sta_1 -= staStep1; + accu0 = SMLABT( accu0, sta0, flt); + accu1 = SMLABT( accu1, sta1, flt); + + flt = p_flt[2]; p_flt += pfltStep; + sta0 = *sta_0; sta_0 -= staStep2; + sta1 = *sta_1; sta_1 += staStep2; + accu0 = SMLABB( accu0, sta0, flt); + accu1 = SMLABB( accu1, sta1, flt); + + *pData_0-- = FX_DBL2FX_QMF(accu0<<1); + *pData_1++ = FX_DBL2FX_QMF(accu1<<1); + } + + /* FIR filter 32/64 */ + flt = p_flt[0]; + sta0 = *sta_0; sta_0 += staStep1; + accu0 = SMULBB( sta0, flt); + sta0 = *sta_0; sta_0 += staStep1; + accu0 = SMLABT( accu0, sta0, flt); + + flt = p_flt[1]; + sta0 = *sta_0; sta_0 += staStep1; + accu0 = SMLABB( accu0, sta0, flt); + sta0 = *sta_0; sta_0 += staStep1; + accu0 = SMLABT( accu0, sta0, flt); + + flt = p_flt[2]; + sta0 = *sta_0; + accu0 = SMLABB( accu0, sta0, flt); + *pData_0-- = FX_DBL2FX_QMF(accu0<<1); + } +} +#endif /* FUNCTION_qmfAnaPrototypeFirSlot */ +#endif /* #if defined(__CC_ARM) && defined(__ARM_ARCH_6__) */ + +#if ( defined(__ARM_ARCH_5TE__) && (SAMPLE_BITS == 16) ) && !defined(QMF_TABLE_FULL) + +#define FUNCTION_qmfSynPrototypeFirSlot + +#if defined(FUNCTION_qmfSynPrototypeFirSlot) + +#if defined(__GNUC__) /* cppp replaced: elif */ + +inline INT SMULWB (const LONG a, const LONG b) +{ + INT result ; + __asm__ ("smulwb %0, %1, %2" + : "=r" (result) + : "r" (a), "r" (b)) ; + + return result ; +} +inline INT SMULWT (const LONG a, const LONG b) +{ + INT result ; + __asm__ ("smulwt %0, %1, %2" + : "=r" (result) + : "r" (a), "r" (b)) ; + + return result ; +} + +inline INT SMLAWB(const LONG accu, const LONG a, const LONG b) +{ + INT result; + asm("smlawb %0, %1, %2, %3 " + : "=r" (result) + : "r" (a), "r" (b), "r" (accu) ); + return result ; +} + +inline INT SMLAWT(const LONG accu, const LONG a, const LONG b) +{ + INT result; + asm("smlawt %0, %1, %2, %3 " + : "=r" (result) + : "r" (a), "r" (b), "r" (accu) ); + return result ; +} + +#endif /* ARM compiler selector */ + + +static void qmfSynPrototypeFirSlot1_filter(FIXP_QMF *RESTRICT realSlot, + FIXP_QMF *RESTRICT imagSlot, + const FIXP_DBL *RESTRICT p_flt, + FIXP_QSS *RESTRICT sta, + FIXP_DBL *pMyTimeOut, + int no_channels) +{ + /* This code was the base for the above listed assembler sequence */ + /* It can be used for debugging purpose or further optimizations */ + const FIXP_DBL *RESTRICT p_fltm = p_flt + 155; + + do + { + FIXP_DBL result; + FIXP_DBL A, B, real, imag, sta0; + + real = *--realSlot; + imag = *--imagSlot; + B = p_flt[4]; /* Bottom=[8] Top=[9] */ + A = p_fltm[3]; /* Bottom=[316] Top=[317] */ + sta0 = sta[0]; /* save state[0] */ + *sta++ = SMLAWT( sta[1], imag, B ); /* index=9...........319 */ + *sta++ = SMLAWB( sta[1], real, A ); /* index=316...........6 */ + *sta++ = SMLAWB( sta[1], imag, B ); /* index=8,18, ...318 */ + B = p_flt[3]; /* Bottom=[6] Top=[7] */ + *sta++ = SMLAWT( sta[1], real, A ); /* index=317...........7 */ + A = p_fltm[4]; /* Bottom=[318] Top=[319] */ + *sta++ = SMLAWT( sta[1], imag, B ); /* index=7...........317 */ + *sta++ = SMLAWB( sta[1], real, A ); /* index=318...........8 */ + *sta++ = SMLAWB( sta[1], imag, B ); /* index=6...........316 */ + B = p_flt[2]; /* Bottom=[X] Top=[5] */ + *sta++ = SMLAWT( sta[1], real, A ); /* index=9...........319 */ + A = p_fltm[2]; /* Bottom=[X] Top=[315] */ + *sta++ = SMULWT( imag, B ); /* index=5,15, ... 315 */ + result = SMLAWT( sta0, real, A ); /* index=315...........5 */ + + *pMyTimeOut++ = result; + + real = *--realSlot; + imag = *--imagSlot; + A = p_fltm[0]; /* Bottom=[310] Top=[311] */ + B = p_flt[7]; /* Bottom=[14] Top=[15] */ + result = SMLAWB( sta[0], real, A ); /* index=310...........0 */ + *sta++ = SMLAWB( sta[1], imag, B ); /* index=14..........324 */ + *pMyTimeOut++ = result; + B = p_flt[6]; /* Bottom=[12] Top=[13] */ + *sta++ = SMLAWT( sta[1], real, A ); /* index=311...........1 */ + A = p_fltm[1]; /* Bottom=[312] Top=[313] */ + *sta++ = SMLAWT( sta[1], imag, B ); /* index=13..........323 */ + *sta++ = SMLAWB( sta[1], real, A ); /* index=312...........2 */ + *sta++ = SMLAWB( sta[1], imag, B ); /* index=12..........322 */ + *sta++ = SMLAWT( sta[1], real, A ); /* index=313...........3 */ + A = p_fltm[2]; /* Bottom=[314] Top=[315] */ + B = p_flt[5]; /* Bottom=[10] Top=[11] */ + *sta++ = SMLAWT( sta[1], imag, B ); /* index=11..........321 */ + *sta++ = SMLAWB( sta[1], real, A ); /* index=314...........4 */ + *sta++ = SMULWB( imag, B ); /* index=10..........320 */ + + + p_flt += 5; + p_fltm -= 5; + } + while ((--no_channels) != 0); + +} + + + +INT qmfSynPrototypeFirSlot2( + HANDLE_QMF_FILTER_BANK qmf, + FIXP_QMF *RESTRICT realSlot, /*!< Input: Pointer to real Slot */ + FIXP_QMF *RESTRICT imagSlot, /*!< Input: Pointer to imag Slot */ + INT_PCM *RESTRICT timeOut, /*!< Time domain data */ + INT stride /*!< Time output buffer stride factor*/ + ) +{ + FIXP_QSS *RESTRICT sta = (FIXP_QSS*)qmf->FilterStates; + int no_channels = qmf->no_channels; + int scale = ((DFRACT_BITS-SAMPLE_BITS)-1-qmf->outScalefactor); + + /* We map an arry of 16-bit values upon an array of 2*16-bit values to read 2 values in one shot */ + const FIXP_DBL *RESTRICT p_flt = (FIXP_DBL *) qmf->p_filter; /* low=[0], high=[1] */ + const FIXP_DBL *RESTRICT p_fltm = (FIXP_DBL *) qmf->p_filter + 155; /* low=[310], high=[311] */ + + FDK_ASSERT(SAMPLE_BITS-1-qmf->outScalefactor >= 0); // (DFRACT_BITS-SAMPLE_BITS)-1-qmf->outScalefactor >= 0); + FDK_ASSERT(qmf->p_stride==2 && qmf->no_channels == 32); + + FDK_ASSERT((no_channels&3) == 0); /* should be a multiple of 4 */ + + realSlot += no_channels-1; // ~~"~~ + imagSlot += no_channels-1; // no_channels-1 .. 0 + + FIXP_DBL MyTimeOut[32]; + FIXP_DBL *pMyTimeOut = &MyTimeOut[0]; + + for (no_channels = no_channels; no_channels--;) + { + FIXP_DBL result; + FIXP_DBL A, B, real, imag; + + real = *realSlot--; + imag = *imagSlot--; + A = p_fltm[0]; /* Bottom=[310] Top=[311] */ + B = p_flt[7]; /* Bottom=[14] Top=[15] */ + result = SMLAWB( sta[0], real, A ); /* index=310...........0 */ + *sta++ = SMLAWB( sta[1], imag, B ); /* index=14..........324 */ + B = p_flt[6]; /* Bottom=[12] Top=[13] */ + *sta++ = SMLAWT( sta[1], real, A ); /* index=311...........1 */ + A = p_fltm[1]; /* Bottom=[312] Top=[313] */ + *sta++ = SMLAWT( sta[1], imag, B ); /* index=13..........323 */ + *sta++ = SMLAWB( sta[1], real, A ); /* index=312...........2 */ + *sta++ = SMLAWB( sta[1], imag, B ); /* index=12..........322 */ + *sta++ = SMLAWT( sta[1], real, A ); /* index=313...........3 */ + A = p_fltm[2]; /* Bottom=[314] Top=[315] */ + B = p_flt[5]; /* Bottom=[10] Top=[11] */ + *sta++ = SMLAWT( sta[1], imag, B ); /* index=11..........321 */ + *sta++ = SMLAWB( sta[1], real, A ); /* index=314...........4 */ + *sta++ = SMULWB( imag, B ); /* index=10..........320 */ + + *pMyTimeOut++ = result; + + p_fltm -= 5; + p_flt += 5; + } + + pMyTimeOut = &MyTimeOut[0]; +#if (SAMPLE_BITS == 16) + const FIXP_DBL max_pos = (FIXP_DBL) 0x00007FFF << scale; + const FIXP_DBL max_neg = (FIXP_DBL) 0xFFFF8001 << scale; +#else + scale = -scale; + const FIXP_DBL max_pos = (FIXP_DBL) 0x7FFFFFFF >> scale; + const FIXP_DBL max_neg = (FIXP_DBL) 0x80000001 >> scale; +#endif + const FIXP_DBL add_neg = (1 << scale) - 1; + + no_channels = qmf->no_channels; + + timeOut += no_channels*stride; + + FDK_ASSERT(scale >= 0); + + if (qmf->outGain != 0x80000000) + { + FIXP_DBL gain = qmf->outGain; + for (no_channels>>=2; no_channels--;) + { + FIXP_DBL result1, result2; + + result1 = *pMyTimeOut++; + result2 = *pMyTimeOut++; + + result1 = fMult(result1,gain); + timeOut -= stride; + if (result1 < 0) result1 += add_neg; + if (result1 < max_neg) result1 = max_neg; + if (result1 > max_pos) result1 = max_pos; +#if (SAMPLE_BITS == 16) + timeOut[0] = result1 >> scale; +#else + timeOut[0] = result1 << scale; +#endif + + result2 = fMult(result2,gain); + timeOut -= stride; + if (result2 < 0) result2 += add_neg; + if (result2 < max_neg) result2 = max_neg; + if (result2 > max_pos) result2 = max_pos; +#if (SAMPLE_BITS == 16) + timeOut[0] = result2 >> scale; +#else + timeOut[0] = result2 << scale; +#endif + + result1 = *pMyTimeOut++; + result2 = *pMyTimeOut++; + + result1 = fMult(result1,gain); + timeOut -= stride; + if (result1 < 0) result1 += add_neg; + if (result1 < max_neg) result1 = max_neg; + if (result1 > max_pos) result1 = max_pos; +#if (SAMPLE_BITS == 16) + timeOut[0] = result1 >> scale; +#else + timeOut[0] = result1 << scale; +#endif + + result2 = fMult(result2,gain); + timeOut -= stride; + if (result2 < 0) result2 += add_neg; + if (result2 < max_neg) result2 = max_neg; + if (result2 > max_pos) result2 = max_pos; +#if (SAMPLE_BITS == 16) + timeOut[0] = result2 >> scale; +#else + timeOut[0] = result2 << scale; +#endif + } + } + else + { + for (no_channels>>=2; no_channels--;) + { + FIXP_DBL result1, result2; + result1 = *pMyTimeOut++; + result2 = *pMyTimeOut++; + timeOut -= stride; + if (result1 < 0) result1 += add_neg; + if (result1 < max_neg) result1 = max_neg; + if (result1 > max_pos) result1 = max_pos; +#if (SAMPLE_BITS == 16) + timeOut[0] = result1 >> scale; +#else + timeOut[0] = result1 << scale; +#endif + + timeOut -= stride; + if (result2 < 0) result2 += add_neg; + if (result2 < max_neg) result2 = max_neg; + if (result2 > max_pos) result2 = max_pos; +#if (SAMPLE_BITS == 16) + timeOut[0] = result2 >> scale; +#else + timeOut[0] = result2 << scale; +#endif + + result1 = *pMyTimeOut++; + result2 = *pMyTimeOut++; + timeOut -= stride; + if (result1 < 0) result1 += add_neg; + if (result1 < max_neg) result1 = max_neg; + if (result1 > max_pos) result1 = max_pos; +#if (SAMPLE_BITS == 16) + timeOut[0] = result1 >> scale; +#else + timeOut[0] = result1 << scale; +#endif + + timeOut -= stride; + if (result2 < 0) result2 += add_neg; + if (result2 < max_neg) result2 = max_neg; + if (result2 > max_pos) result2 = max_pos; +#if (SAMPLE_BITS == 16) + timeOut[0] = result2 >> scale; +#else + timeOut[0] = result2 << scale; +#endif + } + } + return 0; +} + +static +void qmfSynPrototypeFirSlot_fallback( HANDLE_QMF_FILTER_BANK qmf, + FIXP_DBL *realSlot, /*!< Input: Pointer to real Slot */ + FIXP_DBL *imagSlot, /*!< Input: Pointer to imag Slot */ + INT_PCM *timeOut, /*!< Time domain data */ + const int stride + ); + +/*! + \brief Perform Synthesis Prototype Filtering on a single slot of input data. + + The filter takes 2 * #MAX_SYNTHESIS_CHANNELS of input data and + generates #MAX_SYNTHESIS_CHANNELS time domain output samples. +*/ + +static +void qmfSynPrototypeFirSlot( HANDLE_QMF_FILTER_BANK qmf, + FIXP_DBL *realSlot, /*!< Input: Pointer to real Slot */ + FIXP_DBL *imagSlot, /*!< Input: Pointer to imag Slot */ + INT_PCM *timeOut, /*!< Time domain data */ + const int stride + ) +{ + INT err = -1; + + switch (qmf->p_stride) { + case 2: + err = qmfSynPrototypeFirSlot2(qmf, realSlot, imagSlot, timeOut, stride); + break; + default: + err = -1; + } + + /* fallback if configuration not available or failed */ + if(err!=0) { + qmfSynPrototypeFirSlot_fallback(qmf, realSlot, imagSlot, timeOut, stride); + } +} +#endif /* FUNCTION_qmfSynPrototypeFirSlot */ + +#endif /* ( defined(__CC_ARM) && defined(__ARM_ARCH_5TE__) && (SAMPLE_BITS == 16) ) && !defined(QMF_TABLE_FULL) */ + + + +/* #####################################################################################*/ + + + +#endif /* (QMF_NO_POLY==5) */ + |