Mercurial > audlegacy
changeset 4305:59b573234ba2
Automated merge with ssh://hg.atheme.org//hg/audacious
| author | Eugene Zagidullin <e.asphyx@gmail.com> |
|---|---|
| date | Sun, 24 Feb 2008 18:35:50 +0300 |
| parents | eb5f10afe3a0 (current diff) fec498c745f6 (diff) |
| children | 172d49b1200b |
| files | src/audacious/iir.c src/audacious/iir.h src/audacious/iir_cfs.c src/audacious/iir_cfs.h src/audacious/iir_fpu.c src/audacious/iir_fpu.h |
| diffstat | 12 files changed, 2 insertions(+), 742 deletions(-) [+] |
line wrap: on
line diff
--- a/configure.ac Sun Feb 24 18:35:40 2008 +0300 +++ b/configure.ac Sun Feb 24 18:35:50 2008 +0300 @@ -1,6 +1,6 @@ dnl Process this file with autoconf to produce a configure script. -AC_INIT([audacious], [1.5.0], [bugs+audacious@atheme.org]) +AC_INIT([audacious], [1.9.49], [bugs+audacious@atheme.org]) AC_PREREQ([2.59]) AC_CANONICAL_HOST
--- a/src/audacious/Makefile Sun Feb 24 18:35:40 2008 +0300 +++ b/src/audacious/Makefile Sun Feb 24 18:35:50 2008 +0300 @@ -21,9 +21,6 @@ hints.c \ hook.c \ icons-stock.c \ - iir.c \ - iir_cfs.c \ - iir_fpu.c \ input.c \ logger.c \ main.c \
--- a/src/audacious/iir.c Sun Feb 24 18:35:40 2008 +0300 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,125 +0,0 @@ -/* - * PCM time-domain equalizer - * - * Copyright (C) 2002-2005 Felipe Rivera <liebremx at users sourceforge net> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 3 of the License, or - * (at your option) any later version. - * - * This program 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 General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program. If not, see <http://www.gnu.org/licenses>. - * - * $Id: iir.c,v 1.15 2005/10/17 01:57:59 liebremx Exp $ - */ - -#include <math.h> - -#include "main.h" -#include "iir.h" - -/* Coefficients */ -sIIRCoefficients *iir_cf; - -/* Volume gain - * values should be between 0.0 and 1.0 - * Use the preamp from XMMS for now - * */ -float preamp[EQ_CHANNELS]; - -#ifdef BENCHMARK -#include "benchmark.h" -double timex = 0.0; -int count = 0; -unsigned int blength = 0; -#endif - -/* - * Global vars - */ -gint rate; -int band_count; - -void set_preamp(gint chn, float val) -{ - preamp[chn] = val; -} - -/* Init the filters */ -void init_iir() -{ - calc_coeffs(); -#if 0 - band_count = cfg.band_num; -#endif - - band_count = 10; - - rate = 44100; - - iir_cf = get_coeffs(&band_count, rate, TRUE); - clean_history(); -} - -#ifdef ARCH_X86 -/* Round function provided by Frank Klemm which saves around 100K - * CPU cycles in my PIII for each call to the IIR function with 4K samples - */ -__inline__ int round_trick(float floatvalue_to_round) -{ - float floattmp ; - int rounded_value ; - - floattmp = (int) 0x00FD8000L + (floatvalue_to_round); - rounded_value = *(int*)(&floattmp) - (int)0x4B7D8000L; - - if ( rounded_value != (short) rounded_value ) - rounded_value = ( rounded_value >> 31 ) ^ 0x7FFF; - return rounded_value; -} -#endif - -static void -byteswap(size_t size, - gint16 * buf) -{ - gint16 *it; - size &= ~1; /* must be multiple of 2 */ - for (it = buf; it < buf + size / 2; ++it) - *(guint16 *) it = GUINT16_SWAP_LE_BE(*(guint16 *) it); -} - -void -iir_flow(FlowContext *context) -{ - static int init = 0; - int swapped = 0; - guint myorder = G_BYTE_ORDER == G_LITTLE_ENDIAN ? FMT_S16_LE : FMT_S16_BE; - int caneq = (context->fmt == FMT_S16_NE || context->fmt == myorder); - - if (!caneq && cfg.equalizer_active) { /* wrong byte order */ - byteswap(context->len, context->data); /* so convert */ - ++swapped; - ++caneq; - } /* can eq now, mark swapd */ - else if (caneq && !cfg.equalizer_active) /* right order but no eq */ - caneq = 0; /* so don't eq */ - - if (caneq) { /* if eq enab */ - if (!init) { /* if first run */ - init_iir(); /* then init eq */ - ++init; - } - - iir(&context->data, context->len, context->channels); - - if (swapped) /* if was swapped */ - byteswap(context->len, context->data); /* swap back for output */ - } -}
--- a/src/audacious/iir.h Sun Feb 24 18:35:40 2008 +0300 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,86 +0,0 @@ -/* - * PCM time-domain equalizer - * - * Copyright (C) 2002-2005 Felipe Rivera <liebremx at users.sourceforge.net> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 3 of the License, or - * (at your option) any later version. - * - * This program 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 General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program. If not, see <http://www.gnu.org/licenses>. - * - * $Id: iir.h,v 1.12 2005/10/17 01:57:59 liebremx Exp $ - */ -#ifndef IIR_H -#define IIR_H - -#include <glib.h> -#include "main.h" -#include "flow.h" -#include "iir_cfs.h" - -/* - * Flush-to-zero to avoid flooding the CPU with underflow exceptions - */ -#ifdef SSE_MATH -#define FTZ 0x8000 -#define FTZ_ON { \ - unsigned int mxcsr; \ - __asm__ __volatile__ ("stmxcsr %0" : "=m" (*&mxcsr)); \ - mxcsr |= FTZ; \ - __asm__ __volatile__ ("ldmxcsr %0" : : "m" (*&mxcsr)); \ -} -#define FTZ_OFF { \ - unsigned int mxcsr; \ - __asm__ __volatile__ ("stmxcsr %0" : "=m" (*&mxcsr)); \ - mxcsr &= ~FTZ; \ - __asm__ __volatile__ ("ldmxcsr %0" : : "m" (*&mxcsr)); \ -} -#else -#define FTZ_ON -#define FTZ_OFF -#endif - -/* - * Function prototypes - */ -extern void init_iir(); -extern void clean_history(); -extern void set_gain(gint index, gint chn, float val); -extern void set_preamp(gint chn, float val); - - -extern int iir(gpointer * d, gint length, gint nch); - -#ifdef ARCH_X86 -extern int round_trick(float floatvalue_to_round); -#endif -#ifdef ARCH_PPC -extern int round_ppc(float x); -#endif - -#define EQ_CHANNELS 2 -#define EQ_MAX_BANDS 10 - -extern float preamp[EQ_CHANNELS]; -extern sIIRCoefficients *iir_cf; -extern gint rate; -extern gint band_count; - -#ifdef BENCHMARK -extern double timex; -extern int count; -extern unsigned int blength; -#endif - -void iir_flow(FlowContext *context); - -#endif /* #define IIR_H */ -
--- a/src/audacious/iir_cfs.c Sun Feb 24 18:35:40 2008 +0300 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,234 +0,0 @@ -/* - * Copyright (C) 2002-2005 Felipe Rivera <liebremx at users.sourceforge.net> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 3 of the License, or - * (at your option) any later version. - * - * This program 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 General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program. If not, see <http://www.gnu.org/licenses>. - * - * - * Coefficient stuff - * - * $Id: iir_cfs.c,v 1.1 2005/10/17 01:57:59 liebremx Exp $ - */ - -#include "iir_cfs.h" -#include <stdio.h> -#include <math.h> - -/*************************** - * IIR filter coefficients * - ***************************/ -static sIIRCoefficients iir_cf10_11k_11025[10] __attribute__((aligned)); -static sIIRCoefficients iir_cf10_22k_22050[10] __attribute__((aligned)); -static sIIRCoefficients iir_cforiginal10_44100[10] __attribute__((aligned)); -static sIIRCoefficients iir_cforiginal10_48000[10] __attribute__((aligned)); -static sIIRCoefficients iir_cf10_44100[10] __attribute__((aligned)); -static sIIRCoefficients iir_cf10_48000[10] __attribute__((aligned)); -static sIIRCoefficients iir_cf15_44100[15] __attribute__((aligned)); -static sIIRCoefficients iir_cf15_48000[15] __attribute__((aligned)); -static sIIRCoefficients iir_cf25_44100[25] __attribute__((aligned)); -static sIIRCoefficients iir_cf25_48000[25] __attribute__((aligned)); -static sIIRCoefficients iir_cf31_44100[31] __attribute__((aligned)); -static sIIRCoefficients iir_cf31_48000[31] __attribute__((aligned)); - -/****************************************************************** - * Definitions and data structures to calculate the coefficients - ******************************************************************/ -static const double band_f011k[] = -{ 31, 62, 125, 250, 500, 1000, 2000, 3000, 4000, 5500 -}; -static const double band_f022k[] = -{ 31, 62, 125, 250, 500, 1000, 2000, 4000, 8000, 11000 -}; -static const double band_f010[] = -{ 31, 62, 125, 250, 500, 1000, 2000, 4000, 8000, 16000 -}; -static const double band_original_f010[] = -{ 60, 170, 310, 600, 1000, 3000, 6000, 12000, 14000, 16000 -}; -static const double band_f015[] = -{ 25,40,63,100,160,250,400,630,1000,1600,2500,4000,6300,10000,16000 -}; -static const double band_f025[] = -{ 20,31.5,40,50,80,100,125,160,250,315,400,500,800, - 1000,1250,1600,2500,3150,4000,5000,8000,10000,12500,16000,20000 -}; -static const double band_f031[] = -{ 20,25,31.5,40,50,63,80,100,125,160,200,250,315,400,500,630,800, - 1000,1250,1600,2000,2500,3150,4000,5000,6300,8000,10000,12500,16000,20000 -}; - -#define GAIN_F0 1.0 -#define GAIN_F1 GAIN_F0 / M_SQRT2 - -#define SAMPLING_FREQ 44100.0 -#define TETA(f) (2*M_PI*(double)f/bands[n].sfreq) -#define TWOPOWER(value) (value * value) - -#define BETA2(tf0, tf) \ -(TWOPOWER(GAIN_F1)*TWOPOWER(cos(tf0)) \ - - 2.0 * TWOPOWER(GAIN_F1) * cos(tf) * cos(tf0) \ - + TWOPOWER(GAIN_F1) \ - - TWOPOWER(GAIN_F0) * TWOPOWER(sin(tf))) -#define BETA1(tf0, tf) \ - (2.0 * TWOPOWER(GAIN_F1) * TWOPOWER(cos(tf)) \ - + TWOPOWER(GAIN_F1) * TWOPOWER(cos(tf0)) \ - - 2.0 * TWOPOWER(GAIN_F1) * cos(tf) * cos(tf0) \ - - TWOPOWER(GAIN_F1) + TWOPOWER(GAIN_F0) * TWOPOWER(sin(tf))) -#define BETA0(tf0, tf) \ - (0.25 * TWOPOWER(GAIN_F1) * TWOPOWER(cos(tf0)) \ - - 0.5 * TWOPOWER(GAIN_F1) * cos(tf) * cos(tf0) \ - + 0.25 * TWOPOWER(GAIN_F1) \ - - 0.25 * TWOPOWER(GAIN_F0) * TWOPOWER(sin(tf))) - -#define GAMMA(beta, tf0) ((0.5 + beta) * cos(tf0)) -#define ALPHA(beta) ((0.5 - beta)/2.0) - -struct { - sIIRCoefficients *coeffs; - const double *cfs; - double octave; - int band_count; - double sfreq; -} bands[] = { - { iir_cf10_11k_11025, band_f011k, 1.0, 10, 11025.0 }, - { iir_cf10_22k_22050, band_f022k, 1.0, 10, 22050.0 }, - { iir_cforiginal10_44100, band_original_f010, 1.0, 10, 44100.0 }, - { iir_cforiginal10_48000, band_original_f010, 1.0, 10, 48000.0 }, - { iir_cf10_44100, band_f010, 1.0, 10, 44100.0 }, - { iir_cf10_48000, band_f010, 1.0, 10, 48000.0 }, - { iir_cf15_44100, band_f015, 2.0/3.0, 15, 44100.0 }, - { iir_cf15_48000, band_f015, 2.0/3.0, 15, 48000.0 }, - { iir_cf25_44100, band_f025, 1.0/3.0, 25, 44100.0 }, - { iir_cf25_48000, band_f025, 1.0/3.0, 25, 48000.0 }, - { iir_cf31_44100, band_f031, 1.0/3.0, 31, 44100.0 }, - { iir_cf31_48000, band_f031, 1.0/3.0, 31, 48000.0 }, - { 0, 0, 0, 0, 0 } -}; - -/************* - * Functions * - *************/ - -/* Get the coeffs for a given number of bands and sampling frequency */ -sIIRCoefficients* get_coeffs(gint *bands, gint sfreq, gboolean use_xmms_original_freqs) -{ - sIIRCoefficients *iir_cf = 0; - switch(sfreq) - { - case 11025: iir_cf = iir_cf10_11k_11025; - *bands = 10; - break; - case 22050: iir_cf = iir_cf10_22k_22050; - *bands = 10; - break; - case 48000: - switch(*bands) - { - case 31: iir_cf = iir_cf31_48000; break; - case 25: iir_cf = iir_cf25_48000; break; - case 15: iir_cf = iir_cf15_48000; break; - default: - iir_cf = use_xmms_original_freqs ? - iir_cforiginal10_48000 : - iir_cf10_48000; - break; - } - break; - default: - switch(*bands) - { - case 31: iir_cf = iir_cf31_44100; break; - case 25: iir_cf = iir_cf25_44100; break; - case 15: iir_cf = iir_cf15_44100; break; - default: - iir_cf = use_xmms_original_freqs ? - iir_cforiginal10_44100 : - iir_cf10_44100; - break; - } - break; - } - return iir_cf; -} - -/* Get the freqs at both sides of F0. These will be cut at -3dB */ -static void find_f1_and_f2(double f0, double octave_percent, double *f1, double *f2) -{ - double octave_factor = pow(2.0, octave_percent/2.0); - *f1 = f0/octave_factor; - *f2 = f0*octave_factor; -} - -/* Find the quadratic root - * Always return the smallest root */ -static int find_root(double a, double b, double c, double *x0) { - double k = c-((b*b)/(4.*a)); - double h = -(b/(2.*a)); - double x1 = 0.; - if (-(k/a) < 0.) - return -1; - *x0 = h - sqrt(-(k/a)); - x1 = h + sqrt(-(k/a)); - if (x1 < *x0) - *x0 = x1; - return 0; -} - -/* Calculate all the coefficients as specified in the bands[] array */ -void calc_coeffs() -{ - int i, n; - double f1, f2; - double x0; - - n = 0; - for (; bands[n].cfs; n++) { - double *freqs = (double *)bands[n].cfs; - for (i=0; i<bands[n].band_count; i++) - { - - /* Find -3dB frequencies for the center freq */ - find_f1_and_f2(freqs[i], bands[n].octave, &f1, &f2); - /* Find Beta */ - if ( find_root( - BETA2(TETA(freqs[i]), TETA(f1)), - BETA1(TETA(freqs[i]), TETA(f1)), - BETA0(TETA(freqs[i]), TETA(f1)), - &x0) == 0) - { - /* Got a solution, now calculate the rest of the factors */ - /* Take the smallest root always (find_root returns the smallest one) - * - * NOTE: The IIR equation is - * y[n] = 2 * (alpha*(x[n]-x[n-2]) + gamma*y[n-1] - beta*y[n-2]) - * Now the 2 factor has been distributed in the coefficients - */ - /* Now store the coefficients */ - bands[n].coeffs[i].beta = 2.0 * x0; - bands[n].coeffs[i].alpha = 2.0 * ALPHA(x0); - bands[n].coeffs[i].gamma = 2.0 * GAMMA(x0, TETA(freqs[i])); -#ifdef DEBUG - printf("Freq[%d]: %f. Beta: %.10e Alpha: %.10e Gamma %.10e\n", - i, freqs[i], bands[n].coeffs[i].beta, - bands[n].coeffs[i].alpha, bands[n].coeffs[i].gamma); -#endif - } else { - /* Shouldn't happen */ - bands[n].coeffs[i].beta = 0.; - bands[n].coeffs[i].alpha = 0.; - bands[n].coeffs[i].gamma = 0.; - printf(" **** Where are the roots?\n"); - } - }// for i - }//for n -}
--- a/src/audacious/iir_cfs.h Sun Feb 24 18:35:40 2008 +0300 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,39 +0,0 @@ -/* - * PCM time-domain equalizer - * - * Copyright (C) 2002-2005 Felipe Rivera <liebremx at users.sourceforge.net> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 3 of the License, or - * (at your option) any later version. - * - * This program 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 General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program. If not, see <http://www.gnu.org/licenses>. - * - * $Id: iir_cfs.h,v 1.1 2005/10/17 01:57:59 liebremx Exp $ - */ -#ifndef IIR_CFS_H -#define IIR_CFS_H - -#include <glib.h> - -/* Coefficients entry */ -typedef struct -{ - float beta; - float alpha; - float gamma; - float dummy; /* Word alignment */ -}sIIRCoefficients; - -sIIRCoefficients* get_coeffs(gint *bands, gint sfreq, - gboolean use_xmms_original_freqs); -void calc_coeffs(); - -#endif
--- a/src/audacious/iir_fpu.c Sun Feb 24 18:35:40 2008 +0300 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,209 +0,0 @@ -/* - * PCM time-domain equalizer - * - * Copyright (C) 2002-2005 Felipe Rivera <liebremx at users sourceforge net> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 3 of the License, or - * (at your option) any later version. - * - * This program 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 General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program. If not, see <http://www.gnu.org/licenses>. - * - * $Id: iir_fpu.c,v 1.3 2005/11/13 20:02:58 lisanet Exp $ - */ - -#include <strings.h> -#include <stdlib.h> -#include <glib.h> -#include "iir.h" -#include "iir_fpu.h" - -static sXYData data_history[EQ_MAX_BANDS][EQ_CHANNELS] __attribute__((aligned)); -static sXYData data_history2[EQ_MAX_BANDS][EQ_CHANNELS] __attribute__((aligned)); -float gain[EQ_MAX_BANDS][EQ_CHANNELS] __attribute__((aligned)); -/* random noise */ -sample_t dither[256]; -gint di; - -void set_gain(gint index, gint chn, float val) -{ - gain[index][chn] = val; -} - -void clean_history() -{ - gint n; - /* Zero the history arrays */ - bzero(data_history, sizeof(sXYData) * EQ_MAX_BANDS * EQ_CHANNELS); - bzero(data_history2, sizeof(sXYData) * EQ_MAX_BANDS * EQ_CHANNELS); - /* this is only needed if we use fpu code and there's no other place for - the moment to init the dither array*/ - for (n = 0; n < 256; n++) { - dither[n] = (rand() % 4) - 2; - } - di = 0; -} - -__inline__ int iir(gpointer * d, gint length, gint nch) -{ -/* FTZ_ON; */ - gint16 *data = (gint16 *) * d; - /* Indexes for the history arrays - * These have to be kept between calls to this function - * hence they are static */ - static gint i = 2, j = 1, k = 0; - - gint index, band, channel; - gint tempgint, halflength; - sample_t out[EQ_CHANNELS], pcm[EQ_CHANNELS]; - -#if 0 - /* Load the correct filter table according to the sampling rate if needed */ - if (srate != rate) - { - band_count = eqcfg.band_num; - rate = srate; - iir_cf = get_coeffs(&band_count, rate, eqcfg.use_xmms_original_freqs); - clean_history(); - } -#endif - -#ifdef BENCHMARK - start_counter(); -#endif /* BENCHMARK */ - - /** - * IIR filter equation is - * y[n] = 2 * (alpha*(x[n]-x[n-2]) + gamma*y[n-1] - beta*y[n-2]) - * - * NOTE: The 2 factor was introduced in the coefficients to save - * a multiplication - * - * This algorithm cascades two filters to get nice filtering - * at the expense of extra CPU cycles - */ - /* 16bit, 2 bytes per sample, so divide by two the length of - * the buffer (length is in bytes) - */ - halflength = (length >> 1); - for (index = 0; index < halflength; index+=nch) - { - /* For each channel */ - for (channel = 0; channel < nch; channel++) - { - pcm[channel] = data[index+channel] * 4; - /* Preamp gain */ - pcm[channel] *= (preamp[channel] / 2); - - /* add random noise */ - pcm[channel] += dither[di]; - - out[channel] = 0.0; - /* For each band */ - for (band = 0; band < band_count; band++) - { - /* Store Xi(n) */ - data_history[band][channel].x[i] = pcm[channel]; - /* Calculate and store Yi(n) */ - data_history[band][channel].y[i] = - ( - /* = alpha * [x(n)-x(n-2)] */ - iir_cf[band].alpha * ( data_history[band][channel].x[i] - - data_history[band][channel].x[k]) - /* + gamma * y(n-1) */ - + iir_cf[band].gamma * data_history[band][channel].y[j] - /* - beta * y(n-2) */ - - iir_cf[band].beta * data_history[band][channel].y[k] - ); - /* - * The multiplication by 2.0 was 'moved' into the coefficients to save - * CPU cycles here */ - /* Apply the gain */ - out[channel] += data_history[band][channel].y[i]*gain[band][channel]; /* * 2.0; */ - } /* For each band */ - - if (cfg.eq_extra_filtering) - { - /* Filter the sample again */ - for (band = 0; band < band_count; band++) - { - /* Store Xi(n) */ - data_history2[band][channel].x[i] = out[channel]; - /* Calculate and store Yi(n) */ - data_history2[band][channel].y[i] = - ( - /* y(n) = alpha * [x(n)-x(n-2)] */ - iir_cf[band].alpha * (data_history2[band][channel].x[i] - - data_history2[band][channel].x[k]) - /* + gamma * y(n-1) */ - + iir_cf[band].gamma * data_history2[band][channel].y[j] - /* - beta * y(n-2) */ - - iir_cf[band].beta * data_history2[band][channel].y[k] - ); - /* Apply the gain */ - out[channel] += data_history2[band][channel].y[i]*gain[band][channel]; - } /* For each band */ - } - - /* Volume stuff - Scale down original PCM sample and add it to the filters - output. This substitutes the multiplication by 0.25 - Go back to use the floating point multiplication before the - conversion to give more dynamic range - */ - out[channel] += pcm[channel]*0.25; - - /* remove random noise */ - out[channel] -= dither[di]*0.25; - - /* Round and convert to integer */ -#ifdef ARCH_PPC - tempgint = round_ppc(out[channel]); -#else -#ifdef ARCH_X86 - tempgint = round_trick(out[channel]); -#else - tempgint = (int)out[channel]; -#endif -#endif - - /* Limit the output */ - if (tempgint < -32768) - data[index+channel] = -32768; - else if (tempgint > 32767) - data[index+channel] = 32767; - else - data[index+channel] = tempgint; - } /* For each channel */ - - /* Wrap around the indexes */ - i = (i+1)%3; - j = (j+1)%3; - k = (k+1)%3; - /* random noise index */ - di = (di + 1) % 256; - - }/* For each pair of samples */ - -#ifdef BENCHMARK - timex += get_counter(); - blength += length; - if (count++ == 1024) - { - printf("FLOATING POINT: %f %d\n",timex/1024.0, blength/1024); - blength = 0; - timex = 0.; - count = 0; - } -#endif /* BENCHMARK */ - -/* FTZ_OFF; */ - return length; -}
--- a/src/audacious/iir_fpu.h Sun Feb 24 18:35:40 2008 +0300 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,38 +0,0 @@ -/* - * PCM time-domain equalizer - * - * Copyright (C) 2002-2005 Felipe Rivera <liebremx at users.sourceforge.net> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 3 of the License, or - * (at your option) any later version. - * - * This program 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 General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program. If not, see <http://www.gnu.org/licenses>. - * - * $Id: iir_fpu.h,v 1.2 2005/11/01 15:59:20 lisanet Exp $$ - */ -#ifndef IIR_FPU_H -#define IIR_FPU_H - -#define sample_t double - -/* - * Normal FPU implementation data structures - */ -/* Coefficient history for the IIR filter */ -typedef struct -{ - sample_t x[3]; /* x[n], x[n-1], x[n-2] */ - sample_t y[3]; /* y[n], y[n-1], y[n-2] */ - sample_t dummy1; // Word alignment - sample_t dummy2; -}sXYData; - -#endif
--- a/src/audacious/output.c Sun Feb 24 18:35:40 2008 +0300 +++ b/src/audacious/output.c Sun Feb 24 18:35:50 2008 +0300 @@ -30,7 +30,6 @@ #endif #include "output.h" -#include "iir.h" #include "main.h" #include "input.h" #include "playback.h" @@ -540,7 +539,6 @@ if (legacy_flow == NULL) { legacy_flow = flow_new(); - /*flow_link_element(legacy_flow, iir_flow);*/ flow_link_element(legacy_flow, effect_flow); }
--- a/src/audacious/plugin.h Sun Feb 24 18:35:40 2008 +0300 +++ b/src/audacious/plugin.h Sun Feb 24 18:35:50 2008 +0300 @@ -619,7 +619,7 @@ void (*flow_link_element)(Flow *flow, FlowFunction func); void (*flow_unlink_element)(Flow *flow, FlowFunction func); void (*effect_flow)(FlowContext *context); - void (*iir_flow)(FlowContext *context); + gpointer iir_flow; /* old iir_flow */ void (*volumecontrol_flow)(FlowContext *context); /* Dock */ @@ -961,7 +961,6 @@ #define aud_flow_link_element _audvt->flow_link_element #define aud_flow_unlink_element _audvt->flow_unlink_element #define aud_effect_flow _audvt->effect_flow -#define aud_iir_flow _audvt->iir_flow #define aud_volumecontrol_flow _audvt->volumecontrol_flow #define aud_flow_destroy(flow) mowgli_object_unref(flow)
--- a/src/audacious/pluginenum.c Sun Feb 24 18:35:40 2008 +0300 +++ b/src/audacious/pluginenum.c Sun Feb 24 18:35:50 2008 +0300 @@ -60,7 +60,6 @@ #include "ui_fileinfopopup.h" #include "effect.h" -#include "iir.h" #include "volumecontrol.h" const gchar *plugin_dir_list[] = { @@ -368,7 +367,6 @@ .flow_link_element = flow_link_element, .flow_unlink_element = flow_unlink_element, .effect_flow = effect_flow, - .iir_flow = iir_flow, .volumecontrol_flow = volumecontrol_flow, .util_menu_main_show = util_menu_main_show,