rockbox/apps/codecs/libcook/cook_fixpoint.h

/*
 * 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_float.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.
 */

/* get definitions of MULT31, MULT31_SHIFT15, CLIP_TO_15, vect_add, from codelib */
#include "asm_arm.h"
#include "asm_mcf5249.h"
#include "codeclib_misc.h"

/* The following table is taken from libavutil/mathematics.c */
const uint8_t ff_log2_tab[256] ={
        0,0,1,1,2,2,2,2,3,3,3,3,3,3,3,3,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,
        5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,
        6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
        6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
        7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
        7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
        7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
        7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7
};

/* 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) + ((x >> (-i-1)) & 1);
  else
    return x << i;              /* no check for overflow */
}

static inline FIXP fixp_pow2_neg(FIXP x, int i)
{
  return (x >> i) + ((x >> (i-1)) & 1);
}

/**
 * Fixed point multiply by fraction.
 *
 * @param a                     fix point value
 * @param b                     fix point fraction, 0 <= b < 1
 */

#define fixp_mult_su(x,y) (MULT31_SHIFT15(x,y))

/* Faster version of the above using 32x32=64 bit multiply */
#define fixmul31(x,y) (MULT31(x,y))

/* math functions taken from libavutil/common.h */

static inline int av_log2(unsigned int v)
{
    int n = 0;
    if (v & 0xffff0000) {
        v >>= 16;
        n += 16;
    }
    if (v & 0xff00) {
        v >>= 8;
        n += 8;
    }
    n += ff_log2_tab[v];

    return n;
}

/**
 * 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]];
            /* 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++ = fixp_pow2_neg(f, s);
        }
    }
}

#ifdef TEST
/**
 * 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 "cook_fixp_mdct.h"

static inline void imlt_math(COOKContext *q, FIXP *in)
{
    const int n = q->samples_per_channel;
    const int step = 4 << (10 - av_log2(n));
    int i = 0, j = step>>1;
    
    cook_mdct_backward(2 * n, in, q->mono_mdct_output);

    do {
        FIXP tmp = q->mono_mdct_output[i];
        
        q->mono_mdct_output[i] =
          fixp_mult_su(-q->mono_mdct_output[n + i], sincos_lookup[j]);
        q->mono_mdct_output[n + i] = fixp_mult_su(tmp, sincos_lookup[j+1]);
        j += step;
    } while (++i < n/2);
    do {
        FIXP tmp = q->mono_mdct_output[i];
        
        j -= step;
        q->mono_mdct_output[i] =
          fixp_mult_su(-q->mono_mdct_output[n + i], sincos_lookup[j+1]);
        q->mono_mdct_output[n + i] = fixp_mult_su(tmp, sincos_lookup[j]);
    } while (++i < n);
}
#else
#include <codecs/lib/codeclib.h>
#include <codecs/lib/mdct_lookup.h>

static inline void imlt_math(COOKContext *q, FIXP *in)
{
    const int n = q->samples_per_channel;
    const int step = 2 << (10 - av_log2(n));
    int i = 0, j = 0;
    
    mdct_backward(2 * n, in, q->mono_mdct_output);

    do {
        FIXP tmp = q->mono_mdct_output[i];
        
        q->mono_mdct_output[i] =
          fixmul31(-q->mono_mdct_output[n + i], (sincos_lookup0[j]));
          
        q->mono_mdct_output[n + i] = fixmul31(tmp, (sincos_lookup0[j+1]) );
            
        j += step;
        
    } while (++i < n/2);

    do {
        FIXP tmp = q->mono_mdct_output[i];
        
        j -= step;
        q->mono_mdct_output[i] =
          fixmul31(-q->mono_mdct_output[n + i], (sincos_lookup0[j+1]) );
        q->mono_mdct_output[n + i] = fixmul31(tmp, (sincos_lookup0[j]) );
    } while (++i < n);
}
#endif

/**
 * Perform buffer overlapping.
 *
 * @param q                 pointer to the COOKContext
 * @param gain              gain correction to apply first to output buffer
 * @param buffer            data to overlap
 */
static inline void overlap_math(COOKContext *q, int gain, FIXP buffer[])
{
    int i;
    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) + ((q->mono_mdct_output[i] >> (-gain-1)) & 1)+ buffer[i];
        }
    }
}


/**
 * 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]);
}


/**
 * Final converion from floating point values to
 * signed, 16 bit sound samples. Round and clip.
 *
 * @param q                 pointer to the COOKContext
 * @param out               pointer to the output buffer
 * @param chan              0: left or single channel, 1: right channel
 */
static inline void output_math(COOKContext *q, register int16_t *out, int chan)
{
    register REAL_T * mono_output_ptr = q->mono_mdct_output;
    register REAL_T * mono_output_end = mono_output_ptr + q->samples_per_channel;
    out += chan;
    const int STEP = q->nb_channels;
    while( mono_output_ptr < mono_output_end )
    {
      *out = CLIP_TO_15(fixp_pow2_neg(*mono_output_ptr++, 11));
      out += STEP;
    }
}