Applied all mainline PCem commits;

Removed SCSI thread waiting (except after executing the READ SUBCHANNEL command, to alleviate the effects of READ SUBCHANNEL spam done by CD players).

src/sound/filters.h

@@ -1,9 +1,7 @@
-/* Copyright holders: Sarah Walker
-   see COPYING for more details
-*/
 #define NCoef 2
 
-static __inline float low_iir(int i, float NewSample) {
+/* fc=350Hz */
+static inline float low_iir(int i, float NewSample) {
     float ACoef[NCoef+1] = {
         0.00049713569693400649,
         0.00099427139386801299,
@@ -35,7 +33,8 @@ static __inline float low_iir(int i, float NewSample) {
     return y[i][0];
 }
 
-static __inline float low_cut_iir(int i, float NewSample) {
+/* fc=350Hz */
+static inline float low_cut_iir(int i, float NewSample) {
     float ACoef[NCoef+1] = {
         0.96839970114733542000,
         -1.93679940229467080000,
@@ -67,7 +66,8 @@ static __inline float low_cut_iir(int i, float NewSample) {
     return y[i][0];
 }
 
-static __inline float high_iir(int i, float NewSample) {
+/* fc=3.5kHz */
+static inline float high_iir(int i, float NewSample) {
     float ACoef[NCoef+1] = {
         0.72248704753064896000,
         -1.44497409506129790000,
@@ -98,7 +98,8 @@ static __inline float high_iir(int i, float NewSample) {
     return y[i][0];
 }
 
-static __inline float high_cut_iir(int i, float NewSample) {
+/* fc=3.5kHz */
+static inline float high_cut_iir(int i, float NewSample) {
     float ACoef[NCoef+1] = {
         0.03927726802250377400,
         0.07855453604500754700,
@@ -133,7 +134,8 @@ static __inline float high_cut_iir(int i, float NewSample) {
 #undef NCoef
 #define NCoef 2
 
-static __inline float sb_iir(int i, float NewSample) {
+/* fc=3.2kHz */
+static inline float sb_iir(int i, float NewSample) {
     float ACoef[NCoef+1] = {
         0.03356837051492005100,
         0.06713674102984010200,
@@ -155,7 +157,6 @@ static __inline float sb_iir(int i, float NewSample) {
         1.00000000000000000000,
         -0.64940759319751051000
     };*/
-
     static float y[2][NCoef+1]; /* output samples */
     static float x[2][NCoef+1]; /* input samples */
     int n;
@@ -180,7 +181,8 @@ static __inline float sb_iir(int i, float NewSample) {
 #undef NCoef
 #define NCoef 2
 
-static __inline float adgold_highpass_iir(int i, float NewSample) {
+/* fc=150Hz */
+static inline float adgold_highpass_iir(int i, float NewSample) {
     float ACoef[NCoef+1] = {
         0.98657437157334349000,
         -1.97314874314668700000,
@@ -212,7 +214,8 @@ static __inline float adgold_highpass_iir(int i, float NewSample) {
     return y[i][0];
 }
 
-static __inline float adgold_lowpass_iir(int i, float NewSample) {
+/* fc=150Hz */
+static inline float adgold_lowpass_iir(int i, float NewSample) {
     float ACoef[NCoef+1] = {
         0.00009159473951071446,
         0.00018318947902142891,
@@ -244,7 +247,8 @@ static __inline float adgold_lowpass_iir(int i, float NewSample) {
     return y[i][0];
 }
 
-static __inline float adgold_pseudo_stereo_iir(float NewSample) {
+/* fc=56Hz */
+static inline float adgold_pseudo_stereo_iir(float NewSample) {
     float ACoef[NCoef+1] = {
         0.00001409030866231767,
         0.00002818061732463533,
@@ -275,3 +279,69 @@ static __inline float adgold_pseudo_stereo_iir(float NewSample) {
     
     return y[0];
 }
+
+/* fc=3.2kHz - probably incorrect */
+static inline float dss_iir(float NewSample) {
+    float ACoef[NCoef+1] = {
+        0.03356837051492005100,
+        0.06713674102984010200,
+        0.03356837051492005100
+    };
+
+    float BCoef[NCoef+1] = {
+        1.00000000000000000000,
+        -1.41898265221812010000,
+        0.55326988968868285000
+    };
+    
+    static float y[NCoef+1]; /* output samples */
+    static float x[NCoef+1]; /* input samples */
+    int n;
+
+    /* shift the old samples */
+    for(n=NCoef; n>0; n--) {
+       x[n] = x[n-1];
+       y[n] = y[n-1];
+    }
+
+    /* Calculate the new output */
+    x[0] = NewSample;
+    y[0] = ACoef[0] * x[0];
+    for(n=1; n<=NCoef; n++)
+        y[0] += ACoef[n] * x[n] - BCoef[n] * y[n];
+
+    return y[0];
+}
+
+#undef NCoef
+#define NCoef 1
+/*Basic high pass to remove DC bias. fc=10Hz*/
+static inline float dac_iir(int i, float NewSample) {
+    float ACoef[NCoef+1] = {
+        0.99901119820285345000,
+        -0.99901119820285345000
+    };
+
+    float BCoef[NCoef+1] = {
+        1.00000000000000000000,
+        -0.99869185905052738000
+    };
+
+    static float y[2][NCoef+1]; /* output samples */
+    static float x[2][NCoef+1]; /* input samples */
+    int n;
+
+    /* shift the old samples */
+    for(n=NCoef; n>0; n--) {
+       x[i][n] = x[i][n-1];
+       y[i][n] = y[i][n-1];
+    }
+
+    /* Calculate the new output */
+    x[i][0] = NewSample;
+    y[i][0] = ACoef[0] * x[i][0];
+    for(n=1; n<=NCoef; n++)
+        y[i][0] += ACoef[n] * x[i][n] - BCoef[n] * y[i][n];
+    
+    return y[i][0];
+}

src/sound/snd_lpt_dac.c

@@ -0,0 +1,119 @@
+#include <stdlib.h>
+#include "../ibm.h"
+#include "../cpu/cpu.h"
+#include "filters.h"
+#include "../lpt.h"
+#include "snd_lpt_dac.h"
+#include "sound.h"
+#include "../timer.h"
+
+typedef struct lpt_dac_t
+{
+        uint8_t dac_val_l, dac_val_r;
+        
+        int is_stereo;
+        int channel;
+        
+        int16_t buffer[2][SOUNDBUFLEN];
+        int pos;
+} lpt_dac_t;
+
+static void dac_update(lpt_dac_t *lpt_dac)
+{
+        for (; lpt_dac->pos < sound_pos_global; lpt_dac->pos++)
+        {
+                lpt_dac->buffer[0][lpt_dac->pos] = (int8_t)(lpt_dac->dac_val_l ^ 0x80) * 0x40;
+                lpt_dac->buffer[1][lpt_dac->pos] = (int8_t)(lpt_dac->dac_val_r ^ 0x80) * 0x40;
+        }
+}
+
+
+static void dac_write_data(uint8_t val, void *p)
+{
+        lpt_dac_t *lpt_dac = (lpt_dac_t *)p;
+        
+        timer_clock();
+
+        if (lpt_dac->is_stereo)
+        {
+                if (lpt_dac->channel)
+                        lpt_dac->dac_val_r = val;
+                else
+                        lpt_dac->dac_val_l = val;
+        }
+        else        
+                lpt_dac->dac_val_l = lpt_dac->dac_val_r = val;
+        dac_update(lpt_dac);
+}
+
+static void dac_write_ctrl(uint8_t val, void *p)
+{
+        lpt_dac_t *lpt_dac = (lpt_dac_t *)p;
+
+        if (lpt_dac->is_stereo)
+                lpt_dac->channel = val & 0x01;
+}
+
+static uint8_t dac_read_status(void *p)
+{
+        return 0;
+}
+
+
+static void dac_get_buffer(int32_t *buffer, int len, void *p)
+{
+        lpt_dac_t *lpt_dac = (lpt_dac_t *)p;
+        int c;
+        
+        dac_update(lpt_dac);
+        
+        for (c = 0; c < len; c++)
+        {
+                buffer[c*2]     += dac_iir(0, lpt_dac->buffer[0][c]);
+                buffer[c*2 + 1] += dac_iir(1, lpt_dac->buffer[1][c]);
+        }
+        lpt_dac->pos = 0;
+}
+
+static void *dac_init()
+{
+        lpt_dac_t *lpt_dac = malloc(sizeof(lpt_dac_t));
+        memset(lpt_dac, 0, sizeof(lpt_dac_t));
+
+        sound_add_handler(dac_get_buffer, lpt_dac);
+                
+        return lpt_dac;
+}
+static void *dac_stereo_init()
+{
+        lpt_dac_t *lpt_dac = dac_init();
+        
+        lpt_dac->is_stereo = 1;
+                
+        return lpt_dac;
+}
+static void dac_close(void *p)
+{
+        lpt_dac_t *lpt_dac = (lpt_dac_t *)p;
+        
+        free(lpt_dac);
+}
+
+lpt_device_t lpt_dac_device =
+{
+        "LPT DAC / Covox Speech Thing",
+        dac_init,
+        dac_close,
+        dac_write_data,
+        dac_write_ctrl,
+        dac_read_status
+};
+lpt_device_t lpt_dac_stereo_device =
+{
+        "Stereo LPT DAC",
+        dac_stereo_init,
+        dac_close,
+        dac_write_data,
+        dac_write_ctrl,
+        dac_read_status
+};

src/sound/snd_lpt_dac.h

@@ -0,0 +1,2 @@
+extern lpt_device_t lpt_dac_device;
+extern lpt_device_t lpt_dac_stereo_device;

src/sound/snd_lpt_dss.c

@@ -0,0 +1,115 @@
+#include <stdlib.h>
+#include "../ibm.h"
+#include "../cpu/cpu.h"
+#include "filters.h"
+#include "../lpt.h"
+#include "snd_lpt_dss.h"
+#include "sound.h"
+#include "../timer.h"
+
+typedef struct dss_t
+{
+        uint8_t fifo[16];
+        int read_idx, write_idx;
+        
+        uint8_t dac_val;
+        
+        int time;
+        
+        int16_t buffer[SOUNDBUFLEN];
+        int pos;
+} dss_t;
+
+static void dss_update(dss_t *dss)
+{
+        for (; dss->pos < sound_pos_global; dss->pos++)
+                dss->buffer[dss->pos] = (int8_t)(dss->dac_val ^ 0x80) * 0x40;
+}
+
+
+static void dss_write_data(uint8_t val, void *p)
+{
+        dss_t *dss = (dss_t *)p;
+
+        timer_clock();
+
+        if ((dss->write_idx - dss->read_idx) < 16)
+        {
+                dss->fifo[dss->write_idx & 15] = val;
+                dss->write_idx++;
+        }
+}
+
+static void dss_write_ctrl(uint8_t val, void *p)
+{
+}
+
+static uint8_t dss_read_status(void *p)
+{
+        dss_t *dss = (dss_t *)p;
+
+        if ((dss->write_idx - dss->read_idx) >= 16)
+                return 0x40;
+        return 0;
+}
+
+
+static void dss_get_buffer(int32_t *buffer, int len, void *p)
+{
+        dss_t *dss = (dss_t *)p;
+        int c;
+        
+        dss_update(dss);
+        
+        for (c = 0; c < len*2; c += 2)
+        {
+                int16_t val = (int16_t)dss_iir((float)dss->buffer[c >> 1]);
+                
+                buffer[c] += val;
+                buffer[c+1] += val;
+        }
+
+        dss->pos = 0;
+}
+
+static void dss_callback(void *p)
+{
+        dss_t *dss = (dss_t *)p;
+
+        dss_update(dss);
+
+        if ((dss->write_idx - dss->read_idx) > 0)
+        {
+                dss->dac_val = dss->fifo[dss->read_idx & 15];
+                dss->read_idx++;
+        }
+        
+        dss->time += (TIMER_USEC * (1000000.0 / 7000.0));
+}
+
+static void *dss_init()
+{
+        dss_t *dss = malloc(sizeof(dss_t));
+        memset(dss, 0, sizeof(dss_t));
+
+        sound_add_handler(dss_get_buffer, dss);
+        timer_add(dss_callback, &dss->time, TIMER_ALWAYS_ENABLED, dss);
+                
+        return dss;
+}
+static void dss_close(void *p)
+{
+        dss_t *dss = (dss_t *)p;
+        
+        free(dss);
+}
+
+lpt_device_t dss_device =
+{
+        "Disney Sound Source",
+        dss_init,
+        dss_close,
+        dss_write_data,
+        dss_write_ctrl,
+        dss_read_status
+};

src/sound/snd_lpt_dss.h

@@ -0,0 +1 @@
+extern lpt_device_t dss_device;