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Diffstat (limited to 'circuitpython/lib/mp3/src/polyphase.c')
| -rw-r--r-- | circuitpython/lib/mp3/src/polyphase.c | 304 |
1 files changed, 304 insertions, 0 deletions
diff --git a/circuitpython/lib/mp3/src/polyphase.c b/circuitpython/lib/mp3/src/polyphase.c new file mode 100644 index 0000000..a4937ee --- /dev/null +++ b/circuitpython/lib/mp3/src/polyphase.c @@ -0,0 +1,304 @@ +/* ***** 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; + Word64 sum1L, sum2L, rndVal; + + rndVal = (Word64)( 1 << (DEF_NFRACBITS - 1 + (32 - CSHIFT)) ); + + /* special case, output sample 0 */ + coef = coefBase; + vb1 = vbuf; + sum1L = rndVal; + + c1 = *coef; + coef++; + c2 = *coef; + coef++; + vLo = *(vb1+(0)); + vHi = *(vb1+(23-(0))); + sum1L = MADD64(sum1L, vLo, c1); + sum1L = MADD64(sum1L, vHi, -c2); + + //MC0M(0) // a + 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; + Word64 sum1L, sum2L, sum1R, sum2R, rndVal; + + rndVal = (Word64)( 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; + } +} |