diff options
Diffstat (limited to 'apps/plugins/sdl/SDL_mixer/timidity/filter.c')
| -rw-r--r-- | apps/plugins/sdl/SDL_mixer/timidity/filter.c | 187 |
1 files changed, 187 insertions, 0 deletions
diff --git a/apps/plugins/sdl/SDL_mixer/timidity/filter.c b/apps/plugins/sdl/SDL_mixer/timidity/filter.c new file mode 100644 index 0000000000..e93cc6b23c --- /dev/null +++ b/apps/plugins/sdl/SDL_mixer/timidity/filter.c @@ -0,0 +1,187 @@ +/* + TiMidity -- Experimental MIDI to WAVE converter + Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi> + + This program is free software; you can redistribute it and/or modify + it under the terms of the Perl Artistic License, available in COPYING. + + filter.c: written by Vincent Pagel ( pagel@loria.fr ) + + implements fir antialiasing filter : should help when setting sample + rates as low as 8Khz. + + April 95 + - first draft + + 22/5/95 + - modify "filter" so that it simulate leading and trailing 0 in the buffer + */ + +#include "config.h" +#include "common.h" +#include "ctrlmode.h" +#include "instrum.h" +#include "filter.h" + +/* bessel function */ +static float ino(float x) +{ + float y, de, e, sde; + int i; + + y = x / 2; + e = 1.0; + de = 1.0; + i = 1; + do { + de = de * y / (float) i; + sde = de * de; + e += sde; + } while (!( (e * 1.0e-08 - sde > 0) || (i++ > 25) )); + return(e); +} + +/* Kaiser Window (symetric) */ +static void kaiser(float *w,int n,float beta) +{ + float xind, xi; + int i; + + xind = (float)((2*n - 1) * (2*n - 1)); + for (i =0; i<n ; i++) + { + xi = (float)(i + 0.5); + w[i] = ino((float)(beta * sqrt((double)(1. - 4 * xi * xi / xind)))) + / ino((float)beta); + } +} + +/* + * fir coef in g, cuttoff frequency in fc + */ +static void designfir(float *g , float fc) +{ + int i; + float xi, omega, att, beta ; + float w[ORDER2]; + + for (i =0; i < ORDER2 ;i++) + { + xi = (float) (i + 0.5); + omega = (float)(PI * xi); + g[i] = (float)(sin( (double) omega * fc) / omega); + } + + att = 40.; /* attenuation in db */ + beta = (float) (exp(log((double)0.58417 * (att - 20.96)) * 0.4) + 0.07886 + * (att - 20.96)); + kaiser( w, ORDER2, beta); + + /* Matrix product */ + for (i =0; i < ORDER2 ; i++) + g[i] = g[i] * w[i]; +} + +/* + * FIR filtering -> apply the filter given by coef[] to the data buffer + * Note that we simulate leading and trailing 0 at the border of the + * data buffer + */ +static void filter(sample_t *result,sample_t *data, int32 length,float coef[]) +{ + int32 sample,i,sample_window; + int16 peak = 0; + float sum; + + /* Simulate leading 0 at the begining of the buffer */ + for (sample = 0; sample < ORDER2 ; sample++ ) + { + sum = 0.0; + sample_window= sample - ORDER2; + + for (i = 0; i < ORDER ;i++) + sum += (float)(coef[i] * + ((sample_window<0)? 0.0 : data[sample_window++])) ; + + /* Saturation ??? */ + if (sum> 32767.) { sum=32767.; peak++; } + if (sum< -32768.) { sum=-32768; peak++; } + result[sample] = (sample_t) sum; + } + + /* The core of the buffer */ + for (sample = ORDER2; sample < length - ORDER + ORDER2 ; sample++ ) + { + sum = 0.0; + sample_window= sample - ORDER2; + + for (i = 0; i < ORDER ;i++) + sum += data[sample_window++] * coef[i]; + + /* Saturation ??? */ + if (sum> 32767.) { sum=32767.; peak++; } + if (sum< -32768.) { sum=-32768; peak++; } + result[sample] = (sample_t) sum; + } + + /* Simulate 0 at the end of the buffer */ + for (sample = length - ORDER + ORDER2; sample < length ; sample++ ) + { + sum = 0.0; + sample_window= sample - ORDER2; + + for (i = 0; i < ORDER ;i++) + sum += (float)(coef[i] * + ((sample_window>=length)? 0.0 : data[sample_window++])) ; + + /* Saturation ??? */ + if (sum> 32767.) { sum=32767.; peak++; } + if (sum< -32768.) { sum=-32768; peak++; } + result[sample] = (sample_t) sum; + } + + if (peak) + ctl->cmsg(CMSG_ERROR, VERB_NORMAL, + "Saturation %2.3f %%.", 100.0*peak/ (float) length); +} + +/***********************************************************************/ +/* Prevent aliasing by filtering any freq above the output_rate */ +/* */ +/* I don't worry about looping point -> they will remain soft if they */ +/* were already */ +/***********************************************************************/ +void antialiasing(Sample *sp, int32 output_rate ) +{ + sample_t *temp; + int i; + float fir_symetric[ORDER]; + float fir_coef[ORDER2]; + float freq_cut; /* cutoff frequency [0..1.0] FREQ_CUT/SAMP_FREQ*/ + + + ctl->cmsg(CMSG_INFO, VERB_NOISY, "Antialiasing: Fsample=%iKHz", + sp->sample_rate); + + /* No oversampling */ + if (output_rate>=sp->sample_rate) + return; + + freq_cut= (float) output_rate / (float) sp->sample_rate; + ctl->cmsg(CMSG_INFO, VERB_NOISY, "Antialiasing: cutoff=%f%%", + freq_cut*100.); + + designfir(fir_coef,freq_cut); + + /* Make the filter symetric */ + for (i = 0 ; i<ORDER2 ;i++) + fir_symetric[ORDER-1 - i] = fir_symetric[i] = fir_coef[ORDER2-1 - i]; + + /* We apply the filter we have designed on a copy of the patch */ + temp = safe_malloc(sp->data_length); + memcpy(temp,sp->data,sp->data_length); + + filter(sp->data,temp,sp->data_length/sizeof(sample_t),fir_symetric); + + free(temp); +} |