/* * 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>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 ; isamples_per_channel ; i++) { q->mono_mdct_output[i] = (q->mono_mdct_output[i]<< gain) + buffer[i]; } } else { for(i=0 ; isamples_per_channel ; i++) { q->mono_mdct_output[i] = (q->mono_mdct_output[i]>>-gain) + buffer[i]; } } #else for(i=0 ; isamples_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]); }