rockbox/apps/codecs/libcook/cook_fixpoint.h

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/*
* COOK compatible decoder, fixed point implementation.
* Copyright (c) 2007 Ian Braithwaite
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
*/
/**
* @file cook_fixpoint.h
*
* Cook AKA RealAudio G2 fixed point functions.
*
* Fixed point values are represented as 32 bit signed integers,
* which can be added and subtracted directly in C (without checks for
* overflow/saturation.
* Two multiplication routines are provided:
* 1) Multiplication by powers of two (2^-31 .. 2^31), implemented
* with C's bit shift operations.
* 2) Multiplication by 16 bit fractions (0 <= x < 1), implemented
* in C using two 32 bit integer multiplications.
*/
#ifdef ROCKBOX
/* get definitions of MULT31, MULT31_SHIFT15, vect_add, from codelib */
#include "asm_arm.h"
#include "asm_mcf5249.h"
#include "codeclib_misc.h"
#include "codeclib.h"
#endif
/* cplscales was moved from cookdata_fixpoint.h since only *
* cook_fixpoint.h should see/use it. */
static const FIXPU* cplscales[5] = {
cplscale2, cplscale3, cplscale4, cplscale5, cplscale6
};
/**
* Fixed point multiply by power of two.
*
* @param x fix point value
* @param i integer power-of-two, -31..+31
*/
static inline FIXP fixp_pow2(FIXP x, int i)
{
if (i < 0)
return (x >> -i);
else
return x << i; /* no check for overflow */
}
/**
* Fixed point multiply by fraction.
*
* @param a fix point value
* @param b fix point fraction, 0 <= b < 1
*/
#ifdef ROCKBOX
#define fixp_mult_su(x,y) (MULT31_SHIFT15(x,y))
#else
static inline FIXP fixp_mult_su(FIXP a, FIXPU b)
{
int32_t hb = (a >> 16) * b;
uint32_t lb = (a & 0xffff) * b;
return hb + (lb >> 16) + ((lb & 0x8000) >> 15);
}
#endif
/* Faster version of the above using 32x32=64 bit multiply */
#ifdef ROCKBOX
#define fixmul31(x,y) (MULT31(x,y))
#else
static inline int32_t fixmul31(int32_t x, int32_t y)
{
int64_t temp;
temp = x;
temp *= y;
temp >>= 31; //16+31-16 = 31 bits
return (int32_t)temp;
}
#endif
/**
* Clips a signed integer value into the amin-amax range.
* @param a value to clip
* @param amin minimum value of the clip range
* @param amax maximum value of the clip range
* @return clipped value
*/
static inline int av_clip(int a, int amin, int amax)
{
if (a < amin) return amin;
else if (a > amax) return amax;
else return a;
}
/**
* The real requantization of the mltcoefs
*
* @param q pointer to the COOKContext
* @param index index
* @param quant_index quantisation index for this band
* @param subband_coef_index array of indexes to quant_centroid_tab
* @param subband_coef_sign use random noise instead of predetermined value
* @param mlt_ptr pointer to the mlt coefficients
*/
static void scalar_dequant_math(COOKContext *q, int index,
int quant_index, int* subband_coef_index,
int* subband_coef_sign, REAL_T *mlt_p)
{
/* Num. half bits to right shift */
const int s = (33 - quant_index + av_log2(q->samples_per_channel)) >> 1;
const FIXP *table = quant_tables[s & 1][index];
FIXP f;
int i;
if(s >= 32)
memset(mlt_p, 0, sizeof(REAL_T)*SUBBAND_SIZE);
else
{
for(i=0 ; i<SUBBAND_SIZE ; i++) {
f = (table[subband_coef_index[i]])>>s;
/* noise coding if subband_coef_index[i] == 0 */
if (((subband_coef_index[i] == 0) && cook_random(q)) ||
((subband_coef_index[i] != 0) && subband_coef_sign[i]))
f = -f;
*mlt_p++ = f;
}
}
}
/**
* The modulated lapped transform, this takes transform coefficients
* and transforms them into timedomain samples.
* A window step is also included.
*
* @param q pointer to the COOKContext
* @param inbuffer pointer to the mltcoefficients
* @param outbuffer pointer to the timedomain buffer
* @param mlt_tmp pointer to temporary storage space
*/
#include "../lib/mdct_lookup.h"
void imlt_math(COOKContext *q, FIXP *in) ICODE_ATTR;
void imlt_math(COOKContext *q, FIXP *in)
{
const int n = q->samples_per_channel;
const int step = 2 << (10 - av_log2(n));
REAL_T *mdct_out = q->mono_mdct_output;
REAL_T tmp;
int i = 0, j = 0;
ff_imdct_calc(q->mdct_nbits, q->mono_mdct_output, in);
do {
tmp = mdct_out[i];
mdct_out[i ] = fixmul31(-mdct_out[n+i], (sincos_lookup0[j ]));
mdct_out[n+i] = fixmul31(tmp , (sincos_lookup0[j+1]));
j += step;
} while (++i < n/2);
do {
j -= step;
tmp = mdct_out[i];
mdct_out[i ] = fixmul31(-mdct_out[n+i], (sincos_lookup0[j+1]));
mdct_out[n+i] = fixmul31(tmp , (sincos_lookup0[j ]));
} while (++i < n);
}
/**
* Perform buffer overlapping.
*
* @param q pointer to the COOKContext
* @param gain gain correction to apply first to output buffer
* @param buffer data to overlap
*/
void overlap_math(COOKContext *q, int gain, FIXP buffer[]) ICODE_ATTR;
void overlap_math(COOKContext *q, int gain, FIXP buffer[])
{
int i;
#ifdef ROCKBOX
if(LIKELY(gain == 0))
{
vect_add(q->mono_mdct_output, buffer, q->samples_per_channel);
} else if (gain > 0){
for(i=0 ; i<q->samples_per_channel ; i++) {
q->mono_mdct_output[i] = (q->mono_mdct_output[i]<< gain) + buffer[i]; }
} else {
for(i=0 ; i<q->samples_per_channel ; i++) {
q->mono_mdct_output[i] = (q->mono_mdct_output[i]>>-gain) + buffer[i];
}
}
#else
for(i=0 ; i<q->samples_per_channel ; i++) {
q->mono_mdct_output[i] =
fixp_pow2(q->mono_mdct_output[i], gain) + buffer[i];
}
#endif
}
/**
* the actual requantization of the timedomain samples
*
* @param q pointer to the COOKContext
* @param buffer pointer to the timedomain buffer
* @param gain_index index for the block multiplier
* @param gain_index_next index for the next block multiplier
*/
static inline void
interpolate_math(COOKContext *q, register FIXP* buffer,
int gain_index, int gain_index_next)
{
int i;
int gain_size_factor = q->samples_per_channel / 8;
if(gain_index == gain_index_next){ //static gain
for(i = 0; i < gain_size_factor; i++) {
buffer[i] = fixp_pow2(buffer[i], gain_index);
}
} else { //smooth gain
int step = (gain_index_next - gain_index)
<< (7 - av_log2(gain_size_factor));
int x = 0;
register FIXP* bufferend = buffer+gain_size_factor;
while(buffer < bufferend )
{
*buffer = fixp_pow2(
fixp_mult_su(*buffer, pow128_tab[x]),
gain_index+1);
buffer++;
x += step;
gain_index += ( (x + 128) >> 7 ) - 1;
x = ( (x + 128) & 127 );
}
}
}
/**
* Decoupling calculation for joint stereo coefficients.
*
* @param x mono coefficient
* @param table number of decoupling table
* @param i table index
*/
static inline FIXP cplscale_math(FIXP x, int table, int i)
{
return fixp_mult_su(x, cplscales[table-2][i]);
}