/* * 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. */ /* 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 }; /** * Initialise fixed point implementation. * Nothing to do for fixed point. * * @param q pointer to the COOKContext */ static inline int init_cook_math(COOKContext *q) { return 0; } /** * Free resources used by floating point implementation. * Nothing to do for fixed point. * * @param q pointer to the COOKContext */ static inline void free_cook_math(COOKContext *q) { return; } /** * 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 */ } /** * Fixed point multiply by fraction. * * @param a fix point value * @param b fix point fraction, 0 <= b < 1 */ 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); } /* 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); const FIXP *table = quant_tables[s & 1][index]; FIXP f; int i; for(i=0 ; i= 64) ? 0 : fixp_pow2(f, -(s/2)); } } /** * 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); } /** * 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; for(i=0 ; isamples_per_channel ; i++) { q->mono_mdct_output[i] = fixp_pow2(q->mono_mdct_output[i], gain) + 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, 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; for(i = 0; i < gain_size_factor; i++) { buffer[i] = fixp_mult_su(buffer[i], pow128_tab[x]); buffer[i] = fixp_pow2(buffer[i], gain_index+1); x += step; gain_index += (x + 128) / 128 - 1; x = (x + 128) % 128; } } } /** * 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, int16_t *out, int chan) { int j; for (j = 0; j < q->samples_per_channel; j++) { out[chan + q->nb_channels * j] = av_clip(fixp_pow2(q->mono_mdct_output[j], -11), -32768, 32767); } }