mirror of
https://github.com/libretro/scummvm.git
synced 2024-12-13 21:31:53 +00:00
More cleanup.
svn-id: r15681
This commit is contained in:
Torbjörn Andersson
parent
d591de9ab3
commit
22581c3c10
@ -35,25 +35,25 @@
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// **************************
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// files missing
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#define ERR_PRESET1 1
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#define ERR_PRESET2 2
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#define ERR_DRUMPAT 3
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#define ERR_PATCHLOG 4
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#define ERR_MT32ROM 5
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#define ERR_PRESET1 1
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#define ERR_PRESET2 2
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#define ERR_DRUMPAT 3
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#define ERR_PATCHLOG 4
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#define ERR_MT32ROM 5
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// HW spec
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#define PRESENT_SSE 6
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#define PRESENT_3DNOW 7
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#define USING_SSE 8
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#define USING_3DNOW 9
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#define PRESENT_SSE 6
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#define PRESENT_3DNOW 7
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#define USING_SSE 8
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#define USING_3DNOW 9
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// General info
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#define LCD_MESSAGE 10
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#define DEV_RESET 11
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#define DEV_RECONFIG 12
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#define NEW_REVERB_MODE 13
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#define NEW_REVERB_TIME 14
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#define NEW_REVERB_LEVEL 15
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#define LCD_MESSAGE 10
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#define DEV_RESET 11
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#define DEV_RECONFIG 12
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#define NEW_REVERB_MODE 13
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#define NEW_REVERB_TIME 14
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#define NEW_REVERB_LEVEL 15
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#if !defined(__GNUC__)
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#pragma START_PACK_STRUCTS
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@ -363,13 +363,13 @@ int16 Moog2(int16 wg, float *hist, float usefilt, float resonance) {
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input -= hist[4] * fb;
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input *= 0.35013 * (f*f)*(f*f);
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hist[1] = input + 0.3 * hist[5] + (invf) * hist[1]; // Pole 1
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hist[1] = input + 0.3 * hist[5] + (invf) * hist[1]; // Pole 1
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hist[5] = input;
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hist[2] = hist[1] + 0.3 * hist[6] + (invf) * hist[2]; // Pole 2
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hist[2] = hist[1] + 0.3 * hist[6] + (invf) * hist[2]; // Pole 2
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hist[6] = hist[1];
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hist[3] = hist[2] + 0.3 * hist[7] + (invf) * hist[3]; // Pole 3
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hist[3] = hist[2] + 0.3 * hist[7] + (invf) * hist[3]; // Pole 3
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hist[7] = hist[2];
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hist[4] = hist[3] + 0.3 * hist[0] + (invf) * hist[4]; // Pole 4
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hist[4] = hist[3] + 0.3 * hist[0] + (invf) * hist[4]; // Pole 4
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hist[0] = hist[3];
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return (int16)(hist[4] * 32767.0);
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@ -439,47 +439,43 @@ public:
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float allpassfilt(float input, COMB_STATE *state);
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};
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/*
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t60 = reverb time
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hlratio = ratio of low freq t60 to high freq t60
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dur = duration of event/dealloc. on last samp
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hall_fact= mult. factor for delay times
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revstate = running values for event reverb
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*/
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Reverb::Reverb(float t60, float hlratio, float dur, float hall_fact, int sampling_rate)
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{
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/**
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* @param t60 reverb time
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* @param hlratio ratio of low freq t60 to high freq t60
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* @param dur duration of event/dealloc. on last samp
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* @param hall_fact mult. factor for delay times
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* @param revstate running values for event reverb
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*/
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Reverb::Reverb(float t60, float hlratio, float dur, float hall_fact, int sampling_rate) {
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revstate = new ST_REVERB;
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SR = sampling_rate;
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int i;
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float glow[NUM_COMBS], ghi[NUM_COMBS];
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/* initialize sample counter and compute last sample */
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// initialize sample counter and compute last sample
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revstate->cursamp=0;
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revstate->lastsamp = (int)(dur*(float)SR);
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revstate->lastsamp = (int)(dur * (float)SR);
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revstate->done=0;
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/* ALLPASS INITIALIZATIONS */
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// ALLPASS INITIALIZATIONS
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revstate->allpass[0].tau = .006 * hall_fact;
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revstate->allpass[1].tau = .0065 * hall_fact;
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/* allocate allpass delay buffers and head/tail ptr. */
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for(i=0; i<2; i++){
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revstate->allpass[i].bufsiz = (int) (revstate->allpass[i].tau*SR + .5);
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revstate->allpass[i].delbuf =
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new float[revstate->allpass[i].bufsiz];
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memset(revstate->allpass[i].delbuf, 0,
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revstate->allpass[i].bufsiz*sizeof(float));
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// allocate allpass delay buffers and head/tail ptr.
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for (i = 0; i < 2; i++) {
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revstate->allpass[i].bufsiz = (int) (revstate->allpass[i].tau * SR + .5);
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revstate->allpass[i].delbuf = new float[revstate->allpass[i].bufsiz];
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memset(revstate->allpass[i].delbuf, 0, revstate->allpass[i].bufsiz * sizeof(float));
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revstate->allpass[i].bufpos = -1;
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}
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revstate->allpass[0].g = .71f;
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revstate->allpass[1].g = .7f;
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revstate->allpass[0].gsqu =
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revstate->allpass[0].g * revstate->allpass[0].g;
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revstate->allpass[1].gsqu =
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revstate->allpass[1].g * revstate->allpass[1].g;
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revstate->allpass[0].gsqu = revstate->allpass[0].g * revstate->allpass[0].g;
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revstate->allpass[1].gsqu = revstate->allpass[1].g * revstate->allpass[1].g;
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/* COMB AND IIR LOWPASS FILTER INITIALIZATIONS */
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// COMB AND IIR LOWPASS FILTER INITIALIZATIONS
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revstate->comb[0].tau = .0050 * hall_fact;
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revstate->comb[1].tau = .0068 * hall_fact;
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@ -488,74 +484,70 @@ Reverb::Reverb(float t60, float hlratio, float dur, float hall_fact, int samplin
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revstate->comb[4].tau = .0061 * hall_fact;
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revstate->comb[5].tau = .0078 * hall_fact;
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/* allocate comb delay buffers and head/tail ptr. */
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for(i=0; i<NUM_COMBS; i++) {
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// allocate comb delay buffers and head/tail ptr.
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for (i = 0; i < NUM_COMBS; i++) {
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revstate->comb[i].bufsiz = (int)(revstate->comb[i].tau * SR + .5);
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revstate->comb[i].delbuf =
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new float[revstate->comb[i].bufsiz];
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memset(revstate->comb[i].delbuf, 0,
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revstate->comb[i].bufsiz*sizeof(float));
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revstate->comb[i].delbuf = new float[revstate->comb[i].bufsiz];
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memset(revstate->comb[i].delbuf, 0, revstate->comb[i].bufsiz * sizeof(float));
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revstate->comb[i].bufpos = -1;
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revstate->lowpass[i].lastval = 0.;
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}
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/* if hlratio set by user, set various values */
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if (hlratio != 0.) {
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for(i=0; i<NUM_COMBS; i++) {
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// if hlratio set by user, set various values
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// else, use default g's and coef's
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if (hlratio != 0.) {
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for (i = 0; i < NUM_COMBS; i++) {
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/* compute reverb attenuation factor for hi and low */
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/* frequency reverberation times */
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glow[i] =
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pow(10.,(-3. * revstate->comb[i].tau) / t60);
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ghi[i] =
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pow(10.,(-3. * revstate->comb[i].tau)/( t60 * hlratio));
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// compute reverb attenuation factor for hi and low
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// frequency reverberation times
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glow[i] = pow(10.,(-3. * revstate->comb[i].tau) / t60);
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ghi[i] = pow(10.,(-3. * revstate->comb[i].tau) / ( t60 * hlratio));
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/* compute recursive lowpass factor and comb attenuation */
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/* factor to produce the correct reverberation time for */
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/* both hi and low frequencies */
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revstate->lowpass[i].coef = (glow[i] - ghi[i])/(glow[i] + ghi[i]);
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revstate->comb[i].g = glow[i] * (1. - revstate->lowpass[i].coef);
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// compute recursive lowpass factor and comb
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// attenuation factor to produce the correct
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// reverberation time for both hi and low frequencies
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revstate->lowpass[i].coef = (glow[i] - ghi[i]) / (glow[i] + ghi[i]);
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revstate->comb[i].g = glow[i] * (1. - revstate->lowpass[i].coef);
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}
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}
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/* else, use default g's and coef's */
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else {
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revstate->lowpass[0].coef = .24f; revstate->lowpass[1].coef = .26f;
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revstate->lowpass[2].coef = .28f; revstate->lowpass[3].coef = .29f;
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revstate->lowpass[4].coef = .30f; revstate->lowpass[5].coef = .32f;
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} else {
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revstate->lowpass[0].coef = .24f;
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revstate->lowpass[1].coef = .26f;
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revstate->lowpass[2].coef = .28f;
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revstate->lowpass[3].coef = .29f;
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revstate->lowpass[4].coef = .30f;
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revstate->lowpass[5].coef = .32f;
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for(i=0; i<6; i++) {
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/* compute reverb attenuation factor and comb */
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/* attenuation factor based on default coef */
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glow[i] =
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pow(10., (-3. * revstate->comb[i].tau) / t60);
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revstate->comb[i].g = glow[i] *
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(1. - revstate->lowpass[i].coef);
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for (i = 0; i < 6; i++) {
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// compute reverb attenuation factor and comb
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// attenuation factor based on default coef
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glow[i] = pow(10., (-3. * revstate->comb[i].tau) / t60);
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revstate->comb[i].g = glow[i] * (1. - revstate->lowpass[i].coef);
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}
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}
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}
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Reverb:: ~Reverb()
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{
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Reverb:: ~Reverb() {
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int i;
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for(i=0; i<NUM_COMBS; i++) delete[] revstate->comb[i].delbuf;
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for(i=0; i<2; i++) delete[] revstate->allpass[i].delbuf;
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for (i = 0; i < NUM_COMBS; i++)
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delete[] revstate->comb[i].delbuf;
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for (i = 0; i < 2; i++)
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delete[] revstate->allpass[i].delbuf;
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delete revstate;
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}
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/**
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* @param lchan non-reverberated input sample
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* @param rchan non-reverberated input sample
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* @param revfrac percent of output to be reverberated
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*/
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INLINE void Reverb::run(float *lchan, float *rchan, float revfrac)
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/* lchan,rchan non-reverberated input samples */
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/* revfrac percent of output to be reverberated */
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{
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INLINE void Reverb::run(float *lchan, float *rchan, float revfrac) {
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int i;
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float lchanrev, rchanrev, tot=0;
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//cout << " in run \n";
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float lchanrev, rchanrev, tot = 0;
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if (revstate->done) {
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*lchan = 0.0;
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@ -563,141 +555,113 @@ INLINE void Reverb::run(float *lchan, float *rchan, float revfrac)
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return;
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}
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for (i=0; i<NUM_COMBS; i++)
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tot = tot + lpcomb( (*lchan) + (*rchan),
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&(revstate->lowpass[i]),
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&(revstate->comb[i])
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);
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tot = tot/(float)NUM_COMBS;
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for (i = 0; i < NUM_COMBS; i++)
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tot = tot + lpcomb((*lchan) + (*rchan), &(revstate->lowpass[i]), &(revstate->comb[i]));
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tot = tot / (float)NUM_COMBS;
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lchanrev = allpassfilt(tot * .7, &(revstate->allpass[0]));
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// rchanrev = lchanrev ;
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rchanrev = allpassfilt(tot * .7, &(revstate->allpass[1]));
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if (revstate->cursamp == revstate->lastsamp) {
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for(i=0; i<NUM_COMBS; i++) delete[] revstate->comb[i].delbuf;
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for(i=0; i<2; i++) delete[] revstate->allpass[i].delbuf;
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if (revstate->cursamp == revstate->lastsamp) {
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for (i = 0; i < NUM_COMBS; i++)
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delete[] revstate->comb[i].delbuf;
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for (i = 0; i < 2; i++)
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delete[] revstate->allpass[i].delbuf;
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revstate->done = 1;
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}
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(revstate->cursamp)++;
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*lchan = lchanrev*revfrac + (*lchan)*(1. - revfrac) ;
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*rchan = rchanrev*revfrac + (*rchan)*(1. - revfrac) ;
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//*lchan = lchanrev*revfrac + (*lchan) ;
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//*rchan = rchanrev*revfrac + (*rchan) ;
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// cout << "lchan = \t" << *lchan <<endl;
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//cout << "rchan = \t" << *rchan <<endl;
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revstate->cursamp++;
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*lchan = lchanrev * revfrac + (*lchan) * (1. - revfrac);
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*rchan = rchanrev * revfrac + (*rchan) * (1. - revfrac);
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}
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INLINE float Reverb::lowpass(float input, LOWPASS_STATE *state)
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{
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/* simple IIR lowpass filter algorithm */
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/* y(n) = x(n) + coef * y(n-1) */
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INLINE float Reverb::lowpass(float input, LOWPASS_STATE *state) {
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// simple IIR lowpass filter algorithm
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// y(n) = x(n) + coef * y(n - 1)
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state->lastval = (input + state->coef * state->lastval);
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return(state->lastval);
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return state->lastval;
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}
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INLINE float Reverb::lpcomb(float input, LOWPASS_STATE *lpstate, COMB_STATE *cstate)
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{
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INLINE float Reverb::lpcomb(float input, LOWPASS_STATE *lpstate, COMB_STATE *cstate) {
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float temp;
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/* move head-tail pointer in circular queue */
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// move head-tail pointer in circular queue
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cstate->bufpos = (cstate->bufpos + 1) % cstate->bufsiz;
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/* pop circular queue */
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// pop circular queue
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temp = cstate->delbuf[cstate->bufpos];
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/* add new value to end of queue */
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// add new value to end of queue
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lpstate->lastval = (cstate->delbuf[cstate->bufpos] + lpstate->coef * lpstate->lastval);
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cstate->delbuf[cstate->bufpos] =
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input + cstate->g *
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//lowpass(cstate->delbuf[cstate->bufpos], lpstate);
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lpstate->lastval;
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/* return popped value */
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return(temp);
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cstate->delbuf[cstate->bufpos] = input + cstate->g * lpstate->lastval;
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// return popped value
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return temp;
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}
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INLINE float Reverb::allpassfilt(float input, COMB_STATE* state)
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{
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INLINE float Reverb::allpassfilt(float input, COMB_STATE* state) {
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float temp;
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/* move head-tail pointer in circular queue */
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// move head-tail pointer in circular queue
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state->bufpos = (state->bufpos + 1) % state->bufsiz;
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/* pop circular queue */
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// pop circular queue
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temp = state->delbuf[state->bufpos];
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/* add new value to end of queue */
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state->delbuf[state->bufpos] = input +
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state->g * state->delbuf[state->bufpos];
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/* return a sum of the current in with the delay out */
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return(-1.* state->g * input + (1. - state->gsqu) * temp);
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// add new value to end of queue
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state->delbuf[state->bufpos] = input + state->g * state->delbuf[state->bufpos];
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// return a sum of the current in with the delay out
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return -1. * state->g * input + (1. - state->gsqu) * temp;
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}
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/* End reverb stuff */
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// End reverb stuff
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/* Begin filter stuff */
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// Begin filter stuff
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void InitFilter(float fs, float fc, float *icoeff, float Q, float resfac) {
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float *coef;
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unsigned nInd;
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double a0, a1, a2, b0, b1, b2;
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double k; /* overall gain factor */
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double a0, a1, a2, b0, b1, b2;
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double k; // overall gain factor
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/* Section 1 */
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ProtoCoef[0].a0 = 1.0;
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ProtoCoef[0].a1 = 0;
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ProtoCoef[0].a2 = 0;
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ProtoCoef[0].b0 = 1.0;
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ProtoCoef[0].b1 = 0.765367;
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ProtoCoef[0].b2 = 1.0;
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// Section 1
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ProtoCoef[0].a0 = 1.0;
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ProtoCoef[0].a1 = 0;
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ProtoCoef[0].a2 = 0;
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ProtoCoef[0].b0 = 1.0;
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ProtoCoef[0].b1 = 0.765367;
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ProtoCoef[0].b2 = 1.0;
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// Section 2
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ProtoCoef[1].a0 = 1.0;
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ProtoCoef[1].a1 = 0;
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ProtoCoef[1].a2 = 0;
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ProtoCoef[1].b0 = 1.0;
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ProtoCoef[1].b1 = 1.847759;
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ProtoCoef[1].b2 = 1.0;
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/* Section 2 */
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ProtoCoef[1].a0 = 1.0;
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ProtoCoef[1].a1 = 0;
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ProtoCoef[1].a2 = 0;
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ProtoCoef[1].b0 = 1.0;
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ProtoCoef[1].b1 = 1.847759;
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ProtoCoef[1].b2 = 1.0;
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k = 1.5; // Set overall filter gain
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coef = icoeff + 1; // Skip k, or gain
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k = 1.5; /* Set overall filter gain */
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coef = icoeff+1; /* Skip k, or gain */
|
||||
for (nInd = 0; nInd < 2; nInd++) {
|
||||
a0 = ProtoCoef[nInd].a0;
|
||||
a1 = ProtoCoef[nInd].a1;
|
||||
a2 = ProtoCoef[nInd].a2;
|
||||
|
||||
for (nInd = 0; nInd < 2; nInd++)
|
||||
{
|
||||
a0 = ProtoCoef[nInd].a0;
|
||||
a1 = ProtoCoef[nInd].a1;
|
||||
a2 = ProtoCoef[nInd].a2;
|
||||
b0 = ProtoCoef[nInd].b0;
|
||||
b1 = ProtoCoef[nInd].b1 / Q; // Divide by resonance or Q
|
||||
b2 = ProtoCoef[nInd].b2;
|
||||
|
||||
b0 = ProtoCoef[nInd].b0;
|
||||
b1 = ProtoCoef[nInd].b1 / Q; /* Divide by resonance or Q
|
||||
*/
|
||||
b2 = ProtoCoef[nInd].b2;
|
||||
szxform(&a0, &a1, &a2, &b0, &b1, &b2, fc, fs, &k, coef);
|
||||
coef += 4; /* Point to next filter
|
||||
section */
|
||||
szxform(&a0, &a1, &a2, &b0, &b1, &b2, fc, fs, &k, coef);
|
||||
coef += 4; // Point to next filter section
|
||||
}
|
||||
icoeff[0] = k;
|
||||
}
|
||||
|
||||
|
||||
#if FILTER_FLOAT == 1
|
||||
|
||||
iir_filter_type usefilter;
|
||||
|
||||
Loading…
Reference in New Issue
Block a user