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