f40bfc9267
Change-Id: Id7f4717d51ed02d67cb9f9cb3c0ada4a81843f97 Reviewed-on: http://gerrit.rockbox.org/137 Reviewed-by: Nils Wallménius <nils@rockbox.org> Tested-by: Nils Wallménius <nils@rockbox.org>
278 lines
8.4 KiB
C
278 lines
8.4 KiB
C
/*
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* COOK compatible decoder, fixed point implementation.
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* Copyright (c) 2007 Ian Braithwaite
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*
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*/
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/**
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* @file cook_fixpoint.h
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*
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* Cook AKA RealAudio G2 fixed point functions.
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*
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* Fixed point values are represented as 32 bit signed integers,
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* which can be added and subtracted directly in C (without checks for
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* overflow/saturation.
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* Two multiplication routines are provided:
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* 1) Multiplication by powers of two (2^-31 .. 2^31), implemented
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* with C's bit shift operations.
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* 2) Multiplication by 16 bit fractions (0 <= x < 1), implemented
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* in C using two 32 bit integer multiplications.
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*/
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#ifdef ROCKBOX
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/* get definitions of MULT31, MULT31_SHIFT16, vect_add, from codelib */
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#include "codeclib_misc.h"
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#include "codeclib.h"
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#endif
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/* cplscales was moved from cookdata_fixpoint.h since only *
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* cook_fixpoint.h should see/use it. */
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static const FIXPU* cplscales[5] = {
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cplscale2, cplscale3, cplscale4, cplscale5, cplscale6
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};
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/**
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* Fixed point multiply by power of two.
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*
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* @param x fix point value
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* @param i integer power-of-two, -31..+31
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*/
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static inline FIXP fixp_pow2(FIXP x, int i)
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{
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if (i < 0)
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return (x >> -i);
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else
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return x << i; /* no check for overflow */
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}
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/**
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* Fixed point multiply by fraction.
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*
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* @param a fix point value
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* @param b fix point fraction, 0 <= b < 1
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*/
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#ifdef ROCKBOX
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#define fixp_mult_su(x,y) (MULT31_SHIFT16(x,y))
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#else
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static inline FIXP fixp_mult_su(FIXP a, FIXPU b)
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{
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int32_t hb = (a >> 16) * b;
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uint32_t lb = (a & 0xffff) * b;
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return hb + (lb >> 16) + ((lb & 0x8000) >> 15);
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}
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#endif
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/* Faster version of the above using 32x32=64 bit multiply */
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#ifdef ROCKBOX
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#define fixmul31(x,y) (MULT31(x,y))
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#else
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static inline int32_t fixmul31(int32_t x, int32_t y)
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{
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int64_t temp;
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temp = x;
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temp *= y;
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temp >>= 31; //16+31-16 = 31 bits
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return (int32_t)temp;
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}
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#endif
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/**
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* Clips a signed integer value into the amin-amax range.
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* @param a value to clip
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* @param amin minimum value of the clip range
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* @param amax maximum value of the clip range
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* @return clipped value
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*/
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static inline int av_clip(int a, int amin, int amax)
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{
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if (a < amin) return amin;
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else if (a > amax) return amax;
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else return a;
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}
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/**
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* The real requantization of the mltcoefs
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*
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* @param q pointer to the COOKContext
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* @param index index
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* @param quant_index quantisation index for this band
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* @param subband_coef_index array of indexes to quant_centroid_tab
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* @param subband_coef_sign use random noise instead of predetermined value
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* @param mlt_ptr pointer to the mlt coefficients
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*/
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static void scalar_dequant_math(COOKContext *q, int index,
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int quant_index, int* subband_coef_index,
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int* subband_coef_sign, REAL_T *mlt_p)
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ICODE_ATTR_COOK_DECODE;
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static void scalar_dequant_math(COOKContext *q, int index,
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int quant_index, int* subband_coef_index,
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int* subband_coef_sign, REAL_T *mlt_p)
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{
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/* Num. half bits to right shift */
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const int s = 33 - quant_index + av_log2(q->samples_per_channel);
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const FIXP *table = quant_tables[s & 1][index];
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FIXP f;
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int i;
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if(s >= 64)
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memset(mlt_p, 0, sizeof(REAL_T)*SUBBAND_SIZE);
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else
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{
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for(i=0 ; i<SUBBAND_SIZE ; i++) {
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f = (table[subband_coef_index[i]]) >> (s >> 1);
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/* noise coding if subband_coef_index[i] == 0 */
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if (((subband_coef_index[i] == 0) && cook_random(q)) ||
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((subband_coef_index[i] != 0) && subband_coef_sign[i]))
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f = -f;
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*mlt_p++ = f;
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}
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}
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}
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/**
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* The modulated lapped transform, this takes transform coefficients
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* and transforms them into timedomain samples.
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* A window step is also included.
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*
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* @param q pointer to the COOKContext
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* @param inbuffer pointer to the mltcoefficients
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* @param outbuffer pointer to the timedomain buffer
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* @param mlt_tmp pointer to temporary storage space
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*/
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#include "../lib/mdct_lookup.h"
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void imlt_math(COOKContext *q, FIXP *in) ICODE_ATTR;
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void imlt_math(COOKContext *q, FIXP *in)
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{
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const int n = q->samples_per_channel;
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const int step = 2 << (10 - av_log2(n));
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REAL_T *mdct_out = q->mono_mdct_output;
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REAL_T tmp;
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int i = 0, j = 0;
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ff_imdct_calc(q->mdct_nbits, q->mono_mdct_output, in);
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do {
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tmp = mdct_out[i];
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mdct_out[i ] = fixmul31(-mdct_out[n+i], (sincos_lookup0[j ]));
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mdct_out[n+i] = fixmul31(tmp , (sincos_lookup0[j+1]));
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j += step;
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} while (++i < n/2);
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do {
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j -= step;
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tmp = mdct_out[i];
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mdct_out[i ] = fixmul31(-mdct_out[n+i], (sincos_lookup0[j+1]));
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mdct_out[n+i] = fixmul31(tmp , (sincos_lookup0[j ]));
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} while (++i < n);
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}
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/**
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* Perform buffer overlapping.
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*
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* @param q pointer to the COOKContext
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* @param gain gain correction to apply first to output buffer
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* @param buffer data to overlap
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*/
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void overlap_math(COOKContext *q, int gain, FIXP buffer[]) ICODE_ATTR;
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void overlap_math(COOKContext *q, int gain, FIXP buffer[])
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{
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int i;
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#ifdef ROCKBOX
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if(LIKELY(gain == 0))
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{
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vect_add(q->mono_mdct_output, buffer, q->samples_per_channel);
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} else if (gain > 0){
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for(i=0 ; i<q->samples_per_channel ; i++) {
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q->mono_mdct_output[i] = (q->mono_mdct_output[i]<< gain) + buffer[i]; }
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} else {
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for(i=0 ; i<q->samples_per_channel ; i++) {
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q->mono_mdct_output[i] = (q->mono_mdct_output[i]>>-gain) + buffer[i];
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}
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}
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#else
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for(i=0 ; i<q->samples_per_channel ; i++) {
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q->mono_mdct_output[i] =
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fixp_pow2(q->mono_mdct_output[i], gain) + buffer[i];
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}
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#endif
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}
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/**
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* the actual requantization of the timedomain samples
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*
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* @param q pointer to the COOKContext
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* @param buffer pointer to the timedomain buffer
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* @param gain_index index for the block multiplier
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* @param gain_index_next index for the next block multiplier
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*/
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static inline void
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interpolate_math(COOKContext *q, register FIXP* buffer,
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int gain_index, int gain_index_next)
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{
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int i;
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int gain_size_factor = q->samples_per_channel / 8;
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if(gain_index == gain_index_next){ //static gain
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for(i = 0; i < gain_size_factor; i++) {
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buffer[i] = fixp_pow2(buffer[i], gain_index);
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}
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} else { //smooth gain
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int step = (gain_index_next - gain_index)
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<< (7 - av_log2(gain_size_factor));
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int x = 0;
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register FIXP* bufferend = buffer+gain_size_factor;
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while(buffer < bufferend )
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{
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*buffer = fixp_pow2(
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fixp_mult_su(*buffer, pow128_tab[x]),
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gain_index+1);
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buffer++;
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x += step;
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gain_index += ( (x + 128) >> 7 ) - 1;
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x = ( (x + 128) & 127 );
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}
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}
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}
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/**
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* Decoupling calculation for joint stereo coefficients.
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*
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* @param x mono coefficient
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* @param table number of decoupling table
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* @param i table index
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*/
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static inline FIXP cplscale_math(FIXP x, int table, int i)
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{
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return fixp_mult_su(x, cplscales[table-2][i]);
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}
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