PacBox: Premultiply sound prom data on load rather than during emulation. Use 16-bit data for 'raw' output instead of int.

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@27208 a1c6a512-1295-4272-9138-f99709370657

apps/plugins/pacbox/arcade.c

@@ -640,10 +640,10 @@ void renderSprites( unsigned char * buffer )
     }
 }
 
-void playSound( int * buf, int len )
+void playSound( int16_t * buf, int len )
 {
     /* Clear the buffer */
-    memset( buf, 0, sizeof (int)*len);
+    memset( buf, 0, sizeof (int16_t)*len);
 
     /* Exit now if sound is disabled */
     if( (output_devices_ & SoundEnabled) == 0 )

apps/plugins/pacbox/arcade.h

@@ -123,7 +123,7 @@ void init_PacmanMachine(int dip);
 int run(void);
 void reset_PacmanMachine(void);
 void decodeROMs(void);
-void playSound( int * buf, int len );
+void playSound( int16_t * buf, int len );
 
 
 /**

apps/plugins/pacbox/pacbox.c

@@ -281,9 +281,9 @@ static bool pacbox_menu(void)
 static uint32_t sound_buf[NBSAMPLES];
 #if CONFIG_CPU == MCF5249
 /* Not enough to put this in IRAM */
-static int raw_buf[NBSAMPLES];
+static int16_t raw_buf[NBSAMPLES];
 #else
-static int raw_buf[NBSAMPLES] IBSS_ATTR;
+static int16_t raw_buf[NBSAMPLES] IBSS_ATTR;
 #endif
 
 /*
@@ -291,22 +291,23 @@ static int raw_buf[NBSAMPLES] IBSS_ATTR;
  */
 static void get_more(unsigned char **start, size_t *size)
 {
-    int i;
+    int32_t *out, *outend;
-    int32_t *out;
+    int16_t *raw;
-    int *raw;
 
     /* Emulate the audio for the current register settings */
     playSound(raw_buf, NBSAMPLES);
 
     out = sound_buf;
+    outend = out + NBSAMPLES;
     raw = raw_buf;
 
-    /* Normalize the audio and convert to stereo */
+    /* Convert to stereo */
-    for (i = 0; i < NBSAMPLES; i++)
+    do
     {
-        uint32_t sample = (uint16_t)*raw++ << 6;
+        uint32_t sample = (uint16_t)*raw++;
         *out++ = sample | (sample << 16);
     }
+    while (out < outend);
 
     *start = (unsigned char *)sound_buf;
     *size = NBSAMPLES*sizeof(sound_buf[0]); 

apps/plugins/pacbox/wsg3.c

@@ -65,7 +65,7 @@ static bool wsg3_get_voice(struct wsg3_voice *voice, int index)
     return true;
 }
 
-void wsg3_play_sound(int * buf, int len)
+void wsg3_play_sound(int16_t * buf, int len)
 {
     struct wsg3_voice voice;
 
@@ -73,7 +73,7 @@ void wsg3_play_sound(int * buf, int len)
     {
         unsigned offset = wsg3.wave_offset[0];
         unsigned step = voice.frequency * wsg3.resample_step;
-        int * wave_data = wsg3.sound_wave_data + 32 * voice.waveform;
+        int16_t * wave_data = wsg3.sound_wave_data + 32 * voice.waveform;
         int volume = voice.volume;
         int i;
 
@@ -81,7 +81,7 @@ void wsg3_play_sound(int * buf, int len)
         {
             /* Should be shifted right by 15, but we must also get rid
              * of the 10 bits used for decimals */
-            buf[i] += wave_data[(offset >> 25) & 0x1F] * volume;
+            buf[i] = (int)wave_data[(offset >> 25) & 0x1F] * volume;
             offset += step;
         }
 
@@ -92,7 +92,7 @@ void wsg3_play_sound(int * buf, int len)
     {
         unsigned offset = wsg3.wave_offset[1];
         unsigned step = voice.frequency * wsg3.resample_step;
-        int * wave_data = wsg3.sound_wave_data + 32 * voice.waveform;
+        int16_t * wave_data = wsg3.sound_wave_data + 32 * voice.waveform;
         int volume = voice.volume;
         int i;
 
@@ -100,7 +100,7 @@ void wsg3_play_sound(int * buf, int len)
         {
             /* Should be shifted right by 15, but we must also get rid
              * of the 10 bits used for decimals */
-            buf[i] += wave_data[(offset >> 25) & 0x1F] * volume;
+            buf[i] += (int)wave_data[(offset >> 25) & 0x1F] * volume;
             offset += step;
         }
 
@@ -111,7 +111,7 @@ void wsg3_play_sound(int * buf, int len)
     {
         unsigned offset = wsg3.wave_offset[2];
         unsigned step = voice.frequency * wsg3.resample_step;
-        int * wave_data = wsg3.sound_wave_data + 32 * voice.waveform;
+        int16_t * wave_data = wsg3.sound_wave_data + 32 * voice.waveform;
         int volume = voice.volume;
         int i;
 
@@ -119,7 +119,7 @@ void wsg3_play_sound(int * buf, int len)
         {
             /* Should be shifted right by 15, but we must also get rid
              * of the 10 bits used for decimals */
-            buf[i] += wave_data[(offset >> 25) & 0x1F] * volume;
+            buf[i] += (int)wave_data[(offset >> 25) & 0x1F] * volume;
             offset += step;
         }
 
@@ -137,8 +137,12 @@ void wsg3_set_sound_prom( const unsigned char * prom )
 {
     int i;
 
+    memcpy(wsg3.sound_prom, prom, 32*8);
+
+    /* Copy wave data and convert 4-bit unsigned -> 16-bit signed,
+     * prenormalized */
     for (i = 0; i < 32*8; i++)
-        wsg3.sound_wave_data[i] = (int)*prom++ - 8;
+        wsg3.sound_wave_data[i] = ((int16_t)wsg3.sound_prom[i] - 8) * 85;
 }
 
 void wsg3_init(unsigned master_clock)

apps/plugins/pacbox/wsg3.h

@@ -56,7 +56,7 @@ struct wsg3
     unsigned char sound_prom[32*8];
     unsigned resample_step;
     unsigned wave_offset[3];
-    int sound_wave_data[32*8];
+    int16_t sound_wave_data[32*8]; /* sign-extended 4-bit, prenormalized */
 };
 
 extern struct wsg3 wsg3;
@@ -106,7 +106,7 @@ static inline unsigned char wsg3_get_register(unsigned reg)
     @param buf pointer to sound buffer that receives the audio samples
     @param len length of the sound buffer
 */
-void wsg3_play_sound(int * buf, int len);
+void wsg3_play_sound(int16_t * buf, int len);
 
 /**
     Returns the sampling rate currently in use for rendering sound.