Implemented dynamic low-pass filter for SB16/AWE32 DSP playback.

src/filters.h

@@ -345,3 +345,33 @@ static inline float dac_iir(int i, float NewSample) {
     
     return y[i][0];
 }
+
+
+#define SB16_NCoef 51
+
+extern float low_fir_sb16_coef[SB16_NCoef];
+
+static inline float low_fir_sb16(int i, float NewSample)
+{
+        static float x[2][SB16_NCoef+1]; //input samples
+        static int pos = 0;
+        float out = 0.0;
+        int n;
+
+        //Calculate the new output
+        x[i][pos] = NewSample;
+
+        for (n = 0; n < ((SB16_NCoef+1)-pos); n++)
+                out += low_fir_sb16_coef[n] * x[i][n+pos];
+        for (; n < SB16_NCoef; n++)
+                out += low_fir_sb16_coef[n] * x[i][(n+pos) - (SB16_NCoef+1)];
+
+        if (i == 1)
+        {
+                pos++;
+                if (pos > SB16_NCoef)
+                        pos = 0;
+        }
+                        
+        return out;
+}

src/sound_sb.c

@@ -261,9 +261,8 @@ static void sb_get_buffer_sb16(int32_t *buffer, int len, void *p)
                 in_l = (mixer->input_selector_left&INPUT_MIDI_L) ? out_l : 0 + (mixer->input_selector_left&INPUT_MIDI_R) ? out_r : 0;
                 in_r = (mixer->input_selector_right&INPUT_MIDI_L) ? out_l : 0 + (mixer->input_selector_right&INPUT_MIDI_R) ? out_r : 0;
         
-                /*TODO: CT1745 features dynamic filtering. https://www.vogons.org/viewtopic.php?f=62&t=51514 */
+                out_l += ((int32_t)(low_fir_sb16(0, (float)sb->dsp.buffer[c])     * mixer->voice_l) / 3) >> 15;
-                out_l += ((int32_t)(sb->dsp.buffer[c]     * mixer->voice_l) / 3) >> 15;
+                out_r += ((int32_t)(low_fir_sb16(1, (float)sb->dsp.buffer[c + 1]) * mixer->voice_r) / 3) >> 15;
-                out_r += ((int32_t)(sb->dsp.buffer[c + 1] * mixer->voice_r) / 3) >> 15;
                 
                 out_l = (out_l * mixer->master_l) >> 15;
                 out_r = (out_r * mixer->master_r) >> 15;
@@ -341,10 +340,8 @@ static void sb_get_buffer_emu8k(int32_t *buffer, int len, void *p)
                 in_l = (mixer->input_selector_left&INPUT_MIDI_L) ? out_l : 0 + (mixer->input_selector_left&INPUT_MIDI_R) ? out_r : 0;
                 in_r = (mixer->input_selector_right&INPUT_MIDI_L) ? out_l : 0 + (mixer->input_selector_right&INPUT_MIDI_R) ? out_r : 0;
                 
-
+                out_l += ((int32_t)(low_fir_sb16(0, (float)sb->dsp.buffer[c])     * mixer->voice_l) / 3) >> 15;
-                /*TODO: CT1745 features dynamic filtering. https://www.vogons.org/viewtopic.php?f=62&t=51514 */
+                out_r += ((int32_t)(low_fir_sb16(1, (float)sb->dsp.buffer[c + 1]) * mixer->voice_r) / 3) >> 15;
-                out_l += ((int32_t)(sb->dsp.buffer[c]     * mixer->voice_l) / 3) >> 15;
-                out_r += ((int32_t)(sb->dsp.buffer[c + 1] * mixer->voice_r) / 3) >> 15;
 
                 out_l = (out_l * mixer->master_l) >> 15;
                 out_r = (out_r * mixer->master_r) >> 15;

src/sound_sb_dsp.c

@@ -6,10 +6,12 @@
 
 #include <stdint.h>
 #include <stdio.h>
+#include <math.h>
 #include "ibm.h"
 
 
 #include "dma.h"
+#include "filters.h"
 #include "io.h"
 #include "pic.h"
 #include "sound.h"
@@ -108,6 +110,42 @@ uint16_t sb_dsp_versions[] = {0, 0, 0x105, 0x200, 0x201, 0x300, 0x302, 0x405, 0x
 		252, 0, 252, 0
 	};
 
+float low_fir_sb16_coef[SB16_NCoef];
+
+static inline double sinc(double x)
+{
+	return sin(M_PI * x) / (M_PI * x);
+}
+
+static void recalc_sb16_filter(int playback_freq)
+{
+        /*Cutoff frequency = playback / 2*/
+        float fC = ((float)playback_freq / 2.0) / 48000.0;
+        float gain;
+        int n;
+        
+        for (n = 0; n < SB16_NCoef; n++)
+        {
+                /*Blackman window*/
+                double w = 0.42 - (0.5 * cos((2.0*n*M_PI)/(double)(SB16_NCoef-1))) + (0.08 * cos((4.0*n*M_PI)/(double)(SB16_NCoef-1)));
+                /*Sinc filter*/
+                double h = sinc(2.0 * fC * ((double)n - ((double)(SB16_NCoef-1) / 2.0)));
+                
+                /*Create windowed-sinc filter*/
+                low_fir_sb16_coef[n] = w * h;
+        }
+        
+        low_fir_sb16_coef[(SB16_NCoef - 1) / 2] = 1.0;
+
+        gain = 0.0;
+        for (n = 0; n < SB16_NCoef; n++)
+                gain += low_fir_sb16_coef[n];
+
+        /*Normalise filter, to produce unity gain*/
+        for (n = 0; n < SB16_NCoef; n++)
+                low_fir_sb16_coef[n] /= gain;
+}        
+
 
 
 void sb_irq(sb_dsp_t *dsp, int irq8)
@@ -399,6 +437,8 @@ void sb_exec_command(sb_dsp_t *dsp)
                 temp = 256 - dsp->sb_data[0];
                 temp = 1000000 / temp;
 //                pclog("Sample rate - %ihz (%i)\n",temp, dsp->sblatcho);
+                if (dsp->sb_freq != temp && dsp->sb_type >= SB16)
+                        recalc_sb16_filter(temp);
                 dsp->sb_freq = temp;
                 break;
                 case 0x41: /*Set output sampling rate*/
@@ -406,10 +446,13 @@ void sb_exec_command(sb_dsp_t *dsp)
                 if (dsp->sb_type < SB16) break;
                 dsp->sblatcho = (int)(TIMER_USEC * (1000000.0f / (float)(dsp->sb_data[1] + (dsp->sb_data[0] << 8))));
 //                pclog("Sample rate - %ihz (%i)\n",dsp->sb_data[1]+(dsp->sb_data[0]<<8), dsp->sblatcho);
+                temp = dsp->sb_freq;
                 dsp->sb_freq = dsp->sb_data[1] + (dsp->sb_data[0] << 8);
                 dsp->sb_timeo = 256 + dsp->sb_freq;
                 dsp->sblatchi = dsp->sblatcho;
                 dsp->sb_timei = dsp->sb_timeo;
+                if (dsp->sb_freq != temp && dsp->sb_type >= SB16)
+                        recalc_sb16_filter(dsp->sb_freq);
                 break;
                 case 0x48: /*Set DSP block transfer size*/
                 dsp->sb_8_autolen = dsp->sb_data[0] + (dsp->sb_data[1] << 8);