1 files changed, 25 insertions, 117 deletions

diff --git a/src/uxn/devices/apu.c b/src/uxn/devices/apu.c
index c3168c2..2b1c8ff 100644
--- a/src/uxn/devices/apu.c
+++ b/src/uxn/devices/apu.c
@@ -1,36 +1,7 @@
-#include "samples.c"
-
 #define SAMPLE_FREQUENCY 44100
 #define NOTE_PERIOD (SAMPLE_FREQUENCY * 0x4000 / 11025)
 #define ADSR_STEP (SAMPLE_FREQUENCY / 0xf)
 
-// FIXME: We have two structs for now, needs fixing.
-typedef struct AudioChannel {
-    u8 *samples;
-    u16 n_samples;
-        u16 position;
-    u16 sampling_freq;
-    u8 pitch;
-    bool loop;
-    // TODO: u16 adsr; // attack, decay, sustain, release
-    // TODO: u8 pitch;  // Bit 8 is the "loop" bit
-    // TODO: u8 volume; // VOL_LEFT | (VOL_RIGHT << 4)
-        // u8 *addr;
-        // u32 count, advance, period, age, a, d, s, r;
-        // s8 volume[2];
-        // u8 pitch, repeat;
-} AudioChannel;
-
-typedef struct APU {
-    AudioChannel chan_0;
-    // u32 *samples_1;
-    // u32 *samples_2;
-    // u32 *samples_3;
-} APU;
-
-static APU apu;
-
-
 static u16 pitch_table[] = {
     12173, 11490, 10845, 10237, 9662, 9120, 8608, 8125,
     7669, 7238, 6832, 6448, 6086, 5745, 5422, 5118,
@@ -50,48 +21,12 @@ static u16 pitch_table[] = {
     2,
 };
 
-void
-reset_sound(AudioChannel *chan) {
-    TIMER_CTRL_0 = 0;
-    TIMER_CTRL_1 = 0;
-    DMA_CTRL(1) = 0;
-    if (chan->pitch >= 108 || chan->n_samples == 0) {
-        return;
-    }
-
-    // Set max volume, left-right sound, fifo reset and use timer 0 for
-    // DirectSound A.
-    SOUND_DSOUND_MASTER = SOUND_DSOUND_RATIO_A
-        | SOUND_DSOUND_LEFT_A
-        | SOUND_DSOUND_RIGHT_A
-        | SOUND_DSOUND_RESET_A;
-
-    // Prepare DMA copy.
-    dma_transfer_copy(SOUND_FIFO_A, chan->samples, 1, 1,
-            DMA_DST_FIXED | DMA_CHUNK_32 | DMA_REFRESH | DMA_REPEAT | DMA_ENABLE);
-
-    // Timer 1 used to stop playing samples.
-    u32 sample_duration = chan->n_samples;
-    TIMER_DATA_1 = 0xFFFF - sample_duration;
-    TIMER_CTRL_1 = TIMER_CTRL_IRQ
-        | TIMER_CTRL_ENABLE
-        | TIMER_CTRL_CASCADE;
-
-    // Timer 0 used to stop sample playing.
-    // TIMER_DATA_0 = 0xFFFF - sound_freq[chan->pitch];
-    // TIMER_DATA_0 = 0xFFFF - CPU_FREQUENCY / chan->sampling_freq / chan->pitch;
-    TIMER_DATA_0 = 0xFFFF - pitch_table[chan->pitch];
-    TIMER_CTRL_0 = TIMER_CTRL_ENABLE;
-}
-
 s8 square_wave[] =  {
     (s8)0x00 + (s8)0x80, (s8)0xFF + (s8)0x80
 };
 
 #include "text.h"
 
-
-
 //
 //       REG_TM0D                    frequency    buffer size
 //          |                            |            |
@@ -119,7 +54,7 @@ typedef struct Audio {
     u8 active_buffer;
 } Audio;
 
-typedef struct Channel {
+typedef struct AudioChannel {
     // Pointer to the raw data in the ROM.
     s8 *data;
     // Current position in the data (20.12 fixed-point).
@@ -132,28 +67,31 @@ typedef struct Channel {
     u32 length;
     // Length of looped portion (20.12 fixed-point, 0 to disable looping).
     u32 loop_length;
-} Channel;
+    // TODO: this should be different?
+    u8 pitch;
+} AudioChannel;
 
 static Audio audio;
 
 #define POLYPHONY 4
-static Channel channels[POLYPHONY];
+static AudioChannel channels[POLYPHONY];
 
 void
 init_sound(void) {
     // Initialize audio buffers/channels.
     audio = (Audio){0};
     for (size_t i = 0; i < POLYPHONY; ++i) {
-        channels[i] = (Channel){0};
+        channels[i] = (AudioChannel){0};
     }
 
     // DEBUG: testing channel 0 with square wave
-    channels[0].data = samples;
-    channels[0].inc = (8363 << 12) / AUDIO_FREQ;
-    channels[0].length = (LEN(samples) - 1)  << 12;
-    channels[0].pos = 0;
-    channels[0].vol = 64;
-    channels[0].loop_length = channels[0].length;
+    // channels[0].data = samples;
+    // channels[0].length = (LEN(samples) - 1)  << 12;
+    // channels[0].pos = 0;
+    // channels[0].inc = (44100 << 12) / AUDIO_FREQ;
+    // channels[0].vol = 32;
+    // channels[0].loop_length = channels[0].length;
+    // channels[0].loop_length = 0;
 
     // Enable the sound chip.
     SOUND_STATUS = SOUND_ENABLE;
@@ -197,29 +135,29 @@ void sound_mix() {
 
     //  Mix channels into the temporary buffer.
     for (size_t j = 0; j < POLYPHONY; ++j) {
-        Channel *chan = &channels[j];
+        AudioChannel *ch = &channels[j];
         // Check if channel is active.
-        if (chan->data == NULL) {
+        if (ch->data == NULL || ch->pitch >= 108) {
             continue;
         }
-        if (chan->pos + chan->inc * AUDIO_BUF_LEN >= chan->length) {
+        if (ch->pos + ch->inc * AUDIO_BUF_LEN >= ch->length) {
             // Sample is going to finish, need to consider this for looping or
             // stopping.
             for(size_t i = 0; i < AUDIO_BUF_LEN; i++) {
                 // Remember we are using fixed point values.
-                mix_buffer[i] += chan->data[chan->pos >> 12] * chan->vol;
-                chan->pos += chan->inc;
+                mix_buffer[i] += (0x80 + (u8)ch->data[ch->pos >> 12]) * ch->vol;
+                ch->pos += ch->inc;
 
-                if (chan->pos >= chan->length) {
+                if (ch->pos >= ch->length) {
                     // If looping is not active disable the channel.
-                    if (chan->loop_length == 0) {
-                        chan->data = NULL;
+                    if (ch->loop_length == 0) {
+                        ch->data = NULL;
                         break;
                     }
 
                     // Loop the sample.
-                    while (chan->pos >= chan->length) {
-                        chan->pos -= chan->loop_length;
+                    while (ch->pos >= ch->length) {
+                        ch->pos -= ch->loop_length;
                     }
                 }
             }
@@ -227,8 +165,8 @@ void sound_mix() {
             // Sample still have room to go, no need to check for looping or
             // end of sample.
             for(size_t i = 0; i < AUDIO_BUF_LEN; i++) {
-                mix_buffer[i] += chan->data[chan->pos>>12] * chan->vol;
-                chan->pos += chan->inc;
+                mix_buffer[i] +=  (0x80 + (u8)ch->data[ch->pos>>12]) * ch->vol;
+                ch->pos += ch->inc;
             }
         }
     }
@@ -240,33 +178,3 @@ void sound_mix() {
         audio.current_buffer[i] = mix_buffer[i] >> 8;
     }
 }
-
-void
-irs_stop_sample(void) {
-    if (apu.chan_0.loop) {
-        reset_sound(&apu.chan_0);
-    } else {
-        TIMER_CTRL_0 = 0;
-        DMA_CTRL(1) = 0;
-    }
-}
-
-// void
-// apu_start(Apu *c, u16 adsr, u8 pitch) {
-//      // if(pitch < 108 && c->len)
-//      //      c->advance = advances[pitch % 12] >> (8 - pitch / 12);
-//      // else {
-//      //      c->advance = 0;
-//      //      return;
-//      // }
-//      // c->a = ADSR_STEP * (adsr >> 12);
-//      // c->d = ADSR_STEP * (adsr >> 8 & 0xf) + c->a;
-//      // c->s = ADSR_STEP * (adsr >> 4 & 0xf) + c->d;
-//      // c->r = ADSR_STEP * (adsr >> 0 & 0xf) + c->s;
-//      // c->age = 0;
-//      // c->i = 0;
-//      // if(c->len <= 0x100) /* single cycle mode */
-//      //      c->period = NOTE_PERIOD * 337 / 2 / c->len;
-//      // else /* sample repeat mode */
-//      //      c->period = NOTE_PERIOD;
-// }