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-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright 1995 - 2013 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) */
-