Mercurial > audlegacy
changeset 420:44c513ae3527 trunk
[svn] Remove equalization code. We do this same processing in the core in 0.2.
author | nenolod |
---|---|
date | Fri, 13 Jan 2006 14:50:54 -0800 |
parents | 2291caab50ca |
children | 147325df7461 |
files | Plugins/Input/wma/Makefile.am Plugins/Input/wma/iir.c Plugins/Input/wma/iir.h Plugins/Input/wma/wma.c |
diffstat | 4 files changed, 2 insertions(+), 275 deletions(-) [+] |
line wrap: on
line diff
--- a/Plugins/Input/wma/Makefile.am Thu Jan 12 18:27:22 2006 -0800 +++ b/Plugins/Input/wma/Makefile.am Fri Jan 13 14:50:54 2006 -0800 @@ -10,6 +10,6 @@ libwma_la_LDFLAGS = $(PLUGIN_LDFLAGS) ./libffwma/libffwma.la -libwma_la_SOURCES = wma.c iir.c +libwma_la_SOURCES = wma.c INCLUDES = $(GTK_CFLAGS) -I$(top_builddir)/intl -I$(top_srcdir) -I./libffwma -std=c99
--- a/Plugins/Input/wma/iir.c Thu Jan 12 18:27:22 2006 -0800 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,177 +0,0 @@ -/* - * PCM time-domain equalizer - * - * Copyright (C) 2002 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 2 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, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - * - * $Id: iir.c,v 1.5 2003/09/10 21:53:08 liebremx Exp $ - */ - -/* IIR filter coefficients */ -#include "iir.h" -#include <math.h> -#include <strings.h> - -/* History for two filters */ -static sXYData data_history[EQ_MAX_BANDS][EQ_CHANNELS] __attribute__((aligned)); -static sXYData data_history2[EQ_MAX_BANDS][EQ_CHANNELS] __attribute__((aligned)); - -/* Coefficients */ -static sIIRCoefficients *iir_cf; - -/* Config EQ */ -static int band_num = 10; /* Set num band: 10 - default in XMMS */ -static int extra_filtering = 1; /* Set extra filtering: 0 - OFF, 1 - ON */ - -/* BETA, ALPHA, GAMMA */ -static sIIRCoefficients iir_cforiginal10[] __attribute__((aligned)) = { - { (9.9421504945e-01), (2.8924752745e-03), (1.9941421835e+00) }, /* 60.0 Hz */ - { (9.8335039428e-01), (8.3248028618e-03), (1.9827686547e+00) }, /* 170.0 Hz */ - { (9.6958094144e-01), (1.5209529281e-02), (1.9676601546e+00) }, /* 310.0 Hz */ - { (9.4163923306e-01), (2.9180383468e-02), (1.9345490229e+00) }, /* 600.0 Hz */ - { (9.0450844499e-01), (4.7745777504e-02), (1.8852109613e+00) }, /* 1000.0 Hz */ - { (7.3940088234e-01), (1.3029955883e-01), (1.5829158753e+00) }, /* 3000.0 Hz */ - { (5.4697667908e-01), (2.2651166046e-01), (1.0153238114e+00) }, /* 6000.0 Hz */ - { (3.1023210589e-01), (3.4488394706e-01), (-1.8142472036e-01) }, /* 12000.0 Hz */ - { (2.6718639778e-01), (3.6640680111e-01), (-5.2117742267e-01) }, /* 14000.0 Hz */ - { (2.4201241845e-01), (3.7899379077e-01), (-8.0847117831e-01) }, /* 16000.0 Hz */ -}; - -/* Init the filter */ -void init_iir() -{ - iir_cf = iir_cforiginal10; - - /* Zero the history arrays */ - bzero(data_history, sizeof(sXYData) * EQ_MAX_BANDS * EQ_CHANNELS); - bzero(data_history2, sizeof(sXYData) * EQ_MAX_BANDS * EQ_CHANNELS); -} - -__inline__ int iir(gpointer * d, gint length) -{ - 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 = 0, j = 2, k = 1; - - gint index, band, channel; - gint tempgint, halflength; - float out[EQ_CHANNELS], pcm[EQ_CHANNELS]; - - /** - * 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+=2) - { - /* For each channel */ - for (channel = 0; channel < EQ_CHANNELS; channel++) - { - /* No need to scale when processing the PCM with the filter */ - pcm[channel] = data[index+channel]; - /* Preamp gain */ - pcm[channel] *= preamp[channel]; - - - out[channel] = 0; - /* For each band */ - for (band = 0; band < band_num; 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 (extra_filtering) - { - /* Filter the sample again */ - for (band = 0; band < band_num; 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 - */ - - out[channel] += (data[index+channel]>>2); - - /* Round and convert to integer */ - tempgint = (int)lroundf(out[channel]); - - /* 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 */ - - i++; j++; k++; - - /* Wrap around the indexes */ - if (i == 3) i = 0; - else if (j == 3) j = 0; - else k = 0; - - - }/* For each pair of samples */ - - return length; -}
--- a/Plugins/Input/wma/iir.h Thu Jan 12 18:27:22 2006 -0800 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,69 +0,0 @@ -/* - * PCM time-domain equalizer - * - * Copyright (C) 2002 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 2 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, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - * - * $Id: iir.h,v 1.1 2003/05/21 13:59:59 liebremx Exp $ - */ -#ifndef IIR_H -#define IIR_H - -#include <stdio.h> -#include <string.h> - -#include <audacious/plugin.h> -#include <libaudacious/util.h> - -#define EQ_MAX_BANDS 10 -/* Number of channels (Stereo) */ -#define EQ_CHANNELS 2 - - -// Fixed Point Fractional bits -#define FP_FRBITS 28 - -// Conversions -#define EQ_REAL(x) ((gint)((x) * (1 << FP_FRBITS))) - -/* Floating point */ -typedef struct -{ - float beta; - float alpha; - float gamma; -}sIIRCoefficients; - -/* Coefficient history for the IIR filter */ -typedef struct -{ - float x[3]; /* x[n], x[n-1], x[n-2] */ - float y[3]; /* y[n], y[n-1], y[n-2] */ -}sXYData; - -/* Gain for each band - * values should be between -0.2 and 1.0 */ -float gain[EQ_MAX_BANDS][EQ_CHANNELS] __attribute__((aligned)); -/* Volume gain - * values should be between 0.0 and 1.0 - * Use the preamp from XMMS for now - * */ -float preamp[EQ_CHANNELS] __attribute__((aligned)); - -__inline__ int iir(gpointer * d, gint length); -void init_iir(); - -#endif /* #define IIR_H */
--- a/Plugins/Input/wma/wma.c Thu Jan 12 18:27:22 2006 -0800 +++ b/Plugins/Input/wma/wma.c Fri Jan 13 14:50:54 2006 -0800 @@ -37,7 +37,6 @@ #include "avcodec.h" #include "avformat.h" -#include "iir.h" #define ABOUT_TXT "Adapted for use in audacious by Tony Vroon (chainsaw@gentoo.org) from\n \ the BEEP-WMA plugin which is Copyright (C) 2004,2005 Mokrushin I.V. aka McMCC (mcmcc@mail.ru).\n \ @@ -99,7 +98,7 @@ wma_stop, // Stop wma_do_pause, // Pause wma_seek, // Seek - wma_set_eq, // Set the equalizer, most plugins won't be able to do this + NULL, // We don't use this in audacious. wma_get_time, // Get the time, usually returns the output plugins output time NULL, // Get volume NULL, // Set volume @@ -181,7 +180,6 @@ avcodec_init(); avcodec_register_all(); av_register_all(); - init_iir(); } static int wma_is_our_file(char *filename) @@ -215,27 +213,6 @@ return -1; } -static void wma_set_eq(int q_on, float q_preamp, float *q_bands) -{ - int chn; - int index; - float value; - - wma_eq_on = q_on; - if(wma_eq_on) - { - q_preamp = q_preamp/1.6; - for(chn = 0; chn < c->channels; chn++) - preamp[chn] = 1.0 + 0.0932471 * q_preamp + 0.00279033 * q_preamp * q_preamp; - for(index = 0; index < 10; index++) - { - value = q_bands[index]/1.2; - for(chn = 0; chn < c->channels; chn++) - gain[index][chn] = 0.03 * value + 0.000999999 * value * value; - } - } -} - static gchar *extname(const char *filename) { gchar *ext = strrchr(filename, '.'); @@ -322,10 +299,6 @@ fifo_write(&f, wma_outbuf, out_size, &f.wptr); while(!fifo_read(&f, wma_s_outbuf, wma_st_buff, &f.rptr) && wma_decode) { - if(wma_eq_on) - sst_buff = iir((gpointer)&wma_s_outbuf, wma_st_buff); - else - sst_buff = wma_st_buff; if(wma_pause) memset(wma_s_outbuf, 0, sst_buff); while(wma_ip.output->buffer_free() < wma_st_buff) xmms_usleep(20000); produce_audio(wma_ip.output->written_time(), FMT_S16_NE,