296 lines
8.9 KiB
C
296 lines
8.9 KiB
C
/* ***** BEGIN LICENSE BLOCK *****
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* Version: RCSL 1.0/RPSL 1.0
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*
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* Portions Copyright (c) 1995-2002 RealNetworks, Inc. All Rights Reserved.
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*
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* The contents of this file, and the files included with this file, are
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* subject to the current version of the RealNetworks Public Source License
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* Version 1.0 (the "RPSL") available at
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* http://www.helixcommunity.org/content/rpsl unless you have licensed
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* the file under the RealNetworks Community Source License Version 1.0
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* (the "RCSL") available at http://www.helixcommunity.org/content/rcsl,
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* in which case the RCSL will apply. You may also obtain the license terms
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* directly from RealNetworks. You may not use this file except in
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* compliance with the RPSL or, if you have a valid RCSL with RealNetworks
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* applicable to this file, the RCSL. Please see the applicable RPSL or
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* RCSL for the rights, obligations and limitations governing use of the
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* contents of the file.
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*
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* This file is part of the Helix DNA Technology. RealNetworks is the
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* developer of the Original Code and owns the copyrights in the portions
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* it created.
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*
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* This file, and the files included with this file, is distributed and made
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* available on an 'AS IS' basis, WITHOUT WARRANTY OF ANY KIND, EITHER
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* EXPRESS OR IMPLIED, AND REALNETWORKS HEREBY DISCLAIMS ALL SUCH WARRANTIES,
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* INCLUDING WITHOUT LIMITATION, ANY WARRANTIES OF MERCHANTABILITY, FITNESS
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* FOR A PARTICULAR PURPOSE, QUIET ENJOYMENT OR NON-INFRINGEMENT.
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*
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* Technology Compatibility Kit Test Suite(s) Location:
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* http://www.helixcommunity.org/content/tck
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*
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* Contributor(s):
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*
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* ***** END LICENSE BLOCK ***** */
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/**************************************************************************************
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* Fixed-point MP3 decoder
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* Jon Recker (jrecker@real.com), Ken Cooke (kenc@real.com)
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* June 2003
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*
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* polyphase.c - final stage of subband transform (polyphase synthesis filter)
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*
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* This is the C reference version using __int64
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* Look in the appropriate subdirectories for optimized asm implementations
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* (e.g. arm/asmpoly.s)
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**************************************************************************************/
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#include "coder.h"
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#include "assembly.h"
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/* input to Polyphase = Q(DQ_FRACBITS_OUT-2), gain 2 bits in convolution
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* we also have the implicit bias of 2^15 to add back, so net fraction bits =
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* DQ_FRACBITS_OUT - 2 - 2 - 15
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* (see comment on Dequantize() for more info)
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*/
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#define DEF_NFRACBITS (DQ_FRACBITS_OUT - 2 - 2 - 15)
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#define CSHIFT 12 /* coefficients have 12 leading sign bits for early-terminating multiplies */
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static __inline short ClipToShort(int x, int fracBits)
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{
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int sign;
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/* assumes you've already rounded (x += (1 << (fracBits-1))) */
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x >>= fracBits;
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/* Ken's trick: clips to [-32768, 32767] */
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sign = x >> 31;
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if (sign != (x >> 15))
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x = sign ^ ((1 << 15) - 1);
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return (short)x;
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}
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#define MC0M(x) { \
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c1 = *coef; coef++; c2 = *coef; coef++; \
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vLo = *(vb1+(x)); vHi = *(vb1+(23-(x))); \
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sum1L = MADD64(sum1L, vLo, c1); sum1L = MADD64(sum1L, vHi, -c2); \
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}
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#define MC1M(x) { \
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c1 = *coef; coef++; \
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vLo = *(vb1+(x)); \
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sum1L = MADD64(sum1L, vLo, c1); \
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}
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#define MC2M(x) { \
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c1 = *coef; coef++; c2 = *coef; coef++; \
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vLo = *(vb1+(x)); vHi = *(vb1+(23-(x))); \
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sum1L = MADD64(sum1L, vLo, c1); sum2L = MADD64(sum2L, vLo, c2); \
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sum1L = MADD64(sum1L, vHi, -c2); sum2L = MADD64(sum2L, vHi, c1); \
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}
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/**************************************************************************************
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* Function: PolyphaseMono
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*
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* Description: filter one subband and produce 32 output PCM samples for one channel
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*
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* Inputs: pointer to PCM output buffer
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* number of "extra shifts" (vbuf format = Q(DQ_FRACBITS_OUT-2))
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* pointer to start of vbuf (preserved from last call)
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* start of filter coefficient table (in proper, shuffled order)
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* no minimum number of guard bits is required for input vbuf
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* (see additional scaling comments below)
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*
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* Outputs: 32 samples of one channel of decoded PCM data, (i.e. Q16.0)
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*
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* Return: none
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*
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* TODO: add 32-bit version for platforms where 64-bit mul-acc is not supported
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* (note max filter gain - see polyCoef[] comments)
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**************************************************************************************/
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void PolyphaseMono(short *pcm, int *vbuf, const int *coefBase)
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{
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int i;
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const int *coef;
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int *vb1;
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int vLo, vHi, c1, c2;
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Word64 sum1L, sum2L, rndVal;
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rndVal = (Word64)( 1 << (DEF_NFRACBITS - 1 + (32 - CSHIFT)) );
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/* special case, output sample 0 */
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coef = coefBase;
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vb1 = vbuf;
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sum1L = rndVal;
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MC0M(0)
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MC0M(1)
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MC0M(2)
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MC0M(3)
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MC0M(4)
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MC0M(5)
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MC0M(6)
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MC0M(7)
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*(pcm + 0) = ClipToShort((int)SAR64(sum1L, (32-CSHIFT)), DEF_NFRACBITS);
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/* special case, output sample 16 */
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coef = coefBase + 256;
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vb1 = vbuf + 64*16;
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sum1L = rndVal;
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MC1M(0)
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MC1M(1)
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MC1M(2)
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MC1M(3)
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MC1M(4)
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MC1M(5)
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MC1M(6)
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MC1M(7)
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*(pcm + 16) = ClipToShort((int)SAR64(sum1L, (32-CSHIFT)), DEF_NFRACBITS);
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/* main convolution loop: sum1L = samples 1, 2, 3, ... 15 sum2L = samples 31, 30, ... 17 */
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coef = coefBase + 16;
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vb1 = vbuf + 64;
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pcm++;
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/* right now, the compiler creates bad asm from this... */
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for (i = 15; i > 0; i--) {
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sum1L = sum2L = rndVal;
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MC2M(0)
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MC2M(1)
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MC2M(2)
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MC2M(3)
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MC2M(4)
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MC2M(5)
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MC2M(6)
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MC2M(7)
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vb1 += 64;
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*(pcm) = ClipToShort((int)SAR64(sum1L, (32-CSHIFT)), DEF_NFRACBITS);
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*(pcm + 2*i) = ClipToShort((int)SAR64(sum2L, (32-CSHIFT)), DEF_NFRACBITS);
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pcm++;
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}
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}
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#define MC0S(x) { \
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c1 = *coef; coef++; c2 = *coef; coef++; \
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vLo = *(vb1+(x)); vHi = *(vb1+(23-(x))); \
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sum1L = MADD64(sum1L, vLo, c1); sum1L = MADD64(sum1L, vHi, -c2); \
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vLo = *(vb1+32+(x)); vHi = *(vb1+32+(23-(x))); \
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sum1R = MADD64(sum1R, vLo, c1); sum1R = MADD64(sum1R, vHi, -c2); \
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}
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#define MC1S(x) { \
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c1 = *coef; coef++; \
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vLo = *(vb1+(x)); \
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sum1L = MADD64(sum1L, vLo, c1); \
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vLo = *(vb1+32+(x)); \
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sum1R = MADD64(sum1R, vLo, c1); \
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}
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#define MC2S(x) { \
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c1 = *coef; coef++; c2 = *coef; coef++; \
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vLo = *(vb1+(x)); vHi = *(vb1+(23-(x))); \
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sum1L = MADD64(sum1L, vLo, c1); sum2L = MADD64(sum2L, vLo, c2); \
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sum1L = MADD64(sum1L, vHi, -c2); sum2L = MADD64(sum2L, vHi, c1); \
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vLo = *(vb1+32+(x)); vHi = *(vb1+32+(23-(x))); \
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sum1R = MADD64(sum1R, vLo, c1); sum2R = MADD64(sum2R, vLo, c2); \
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sum1R = MADD64(sum1R, vHi, -c2); sum2R = MADD64(sum2R, vHi, c1); \
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}
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/**************************************************************************************
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* Function: PolyphaseStereo
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*
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* Description: filter one subband and produce 32 output PCM samples for each channel
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*
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* Inputs: pointer to PCM output buffer
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* number of "extra shifts" (vbuf format = Q(DQ_FRACBITS_OUT-2))
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* pointer to start of vbuf (preserved from last call)
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* start of filter coefficient table (in proper, shuffled order)
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* no minimum number of guard bits is required for input vbuf
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* (see additional scaling comments below)
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*
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* Outputs: 32 samples of two channels of decoded PCM data, (i.e. Q16.0)
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*
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* Return: none
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*
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* Notes: interleaves PCM samples LRLRLR...
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*
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* TODO: add 32-bit version for platforms where 64-bit mul-acc is not supported
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**************************************************************************************/
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void PolyphaseStereo(short *pcm, int *vbuf, const int *coefBase)
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{
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int i;
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const int *coef;
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int *vb1;
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int vLo, vHi, c1, c2;
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Word64 sum1L, sum2L, sum1R, sum2R, rndVal;
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rndVal = (Word64)( 1 << (DEF_NFRACBITS - 1 + (32 - CSHIFT)) );
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/* special case, output sample 0 */
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coef = coefBase;
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vb1 = vbuf;
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sum1L = sum1R = rndVal;
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MC0S(0)
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MC0S(1)
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MC0S(2)
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MC0S(3)
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MC0S(4)
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MC0S(5)
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MC0S(6)
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MC0S(7)
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*(pcm + 0) = ClipToShort((int)SAR64(sum1L, (32-CSHIFT)), DEF_NFRACBITS);
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*(pcm + 1) = ClipToShort((int)SAR64(sum1R, (32-CSHIFT)), DEF_NFRACBITS);
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/* special case, output sample 16 */
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coef = coefBase + 256;
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vb1 = vbuf + 64*16;
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sum1L = sum1R = rndVal;
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MC1S(0)
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MC1S(1)
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MC1S(2)
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MC1S(3)
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MC1S(4)
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MC1S(5)
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MC1S(6)
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MC1S(7)
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*(pcm + 2*16 + 0) = ClipToShort((int)SAR64(sum1L, (32-CSHIFT)), DEF_NFRACBITS);
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*(pcm + 2*16 + 1) = ClipToShort((int)SAR64(sum1R, (32-CSHIFT)), DEF_NFRACBITS);
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/* main convolution loop: sum1L = samples 1, 2, 3, ... 15 sum2L = samples 31, 30, ... 17 */
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coef = coefBase + 16;
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vb1 = vbuf + 64;
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pcm += 2;
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/* right now, the compiler creates bad asm from this... */
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for (i = 15; i > 0; i--) {
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sum1L = sum2L = rndVal;
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sum1R = sum2R = rndVal;
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MC2S(0)
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MC2S(1)
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MC2S(2)
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MC2S(3)
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MC2S(4)
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MC2S(5)
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MC2S(6)
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MC2S(7)
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vb1 += 64;
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*(pcm + 0) = ClipToShort((int)SAR64(sum1L, (32-CSHIFT)), DEF_NFRACBITS);
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*(pcm + 1) = ClipToShort((int)SAR64(sum1R, (32-CSHIFT)), DEF_NFRACBITS);
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*(pcm + 2*2*i + 0) = ClipToShort((int)SAR64(sum2L, (32-CSHIFT)), DEF_NFRACBITS);
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*(pcm + 2*2*i + 1) = ClipToShort((int)SAR64(sum2R, (32-CSHIFT)), DEF_NFRACBITS);
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pcm += 2;
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}
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}
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