Add ADSR filter

2 files changed, 88 insertions, 7 deletions

diff --git a/src/main.c b/src/main.c
index cbf781d..aee68f8 100644
--- a/src/main.c
+++ b/src/main.c
@@ -155,10 +155,13 @@ audio_talk(Device *d, u8 b0, u8 w) {
         }
         c->inc = (pitch_table[c->pitch] * sampling_rate) >> 5;
 
+        u16 adsr = mempeek16(d->dat, 0x8);
+        build_adsr(c, adsr);
             txt_position(0, 0);
         txt_printf("note:   %d     \n", c->pitch);
-        txt_printf("inc:    %ld    \n", c->inc);
-        txt_printf("length: %ld    \n", c->length >> 12);
+        txt_printf("inc:    %lu    \n", c->inc);
+        txt_printf("length: %lu    \n", c->length >> 12);
+        txt_printf("chan:   %lu    \n", d - devaudio);
         }
 }
 
@@ -423,21 +426,24 @@ int main(void) {
     int frame_counter = 0;
     evaluxn(&u, 0x0100);
     u32 flip_cycles = 0;
+    u32 eval_cycles = 0;
+    u32 input_cycles = 0;
+    u32 mix_cycles = 0;
     while(true) {
         bios_vblank_wait();
         profile_start();
         handle_input(&u);
-        u32 input_cycles = profile_stop();
+        input_cycles = MAX(profile_stop(), input_cycles);
         profile_start();
         evaluxn(&u, mempeek16(devscreen->dat, 0));
-        u32 eval_cycles = profile_stop();
+        eval_cycles = MAX(profile_stop(), eval_cycles);
         txt_position(0, 8);
         profile_start();
         flip_cycles = profile_stop();
         frame_counter++;
         profile_start();
         sound_mix();
-        u32 mix_cycles = profile_stop();
+        mix_cycles = MAX(profile_stop(), mix_cycles);
 
         txt_position(0, 15);
         txt_printf("INPUT: %lu    \n", input_cycles);
diff --git a/src/uxn/devices/apu.c b/src/uxn/devices/apu.c
index 4447e12..d968840 100644
--- a/src/uxn/devices/apu.c
+++ b/src/uxn/devices/apu.c
@@ -1,7 +1,7 @@
 // Calculated as ((261.6 / 18157) << 17) for C4. If multiplied by sampling rate
 // we will have a u32 (15.17) fixed-point number. This should be enough to
 // accurately portray samples up to 75300 Hz.
-static u32 pitch_table[120] = {
+static u16 pitch_table[120] = {
     59,    62,    66,    70,    74,    78,    83,    88,
     93,    99,    105,   111,   118,   125,   132,   140,
     148,   157,   166,   176,   187,   198,   210,   222,
@@ -61,8 +61,75 @@ typedef struct AudioChannel {
     u32 loop_length;
     // Pitch encoded as a MIDI note.
     u8 pitch;
+    // Keeping track of the original adsr values.
+    u16 adsr;
+    // The filter is built with the ADSR and has a duration of 60 ticks. Since
+    // weare synced at 60FPS it should be pretty straightforward.
+    u8 filter[241];
+    // Current position in the filter (0-60).
+    u8 filter_pos;
+    // Duration of the filter.
+    u8 filter_len;
 } AudioChannel;
 
+// NOTE: fixed-point (1.6) precision for volume adjustments.
+typedef u32 fp32;
+
+inline
+fp32
+fp_mul(fp32 a, fp32 b) {
+    return (a * b) >> 6;
+}
+
+inline
+fp32
+fp_div(fp32 a, fp32 b) {
+    return (a << 6) / b;
+}
+
+inline
+fp32
+fp_lerp(fp32 y0, fp32 y1, fp32 x) {
+    return y0 + fp_mul(x, (y1 - y0));
+}
+
+
+IWRAM_CODE
+void
+build_adsr(AudioChannel *chan, u16 adsr) {
+    // u8 a = (adsr >> 12);
+    u8 a = (adsr >> 12);
+    u8 d = (adsr >> 8) & 0xF;
+    u8 s = (adsr >> 4) & 0xF;
+    u8 r = (adsr >> 0) & 0xF;
+
+    // Initialize the filter array.
+    memset(chan->filter, 0, sizeof(chan->filter));
+    u8 k = 0;
+
+    // Attack.
+    for (u32 i = 0; i < 4 * a; ++i) {
+        chan->filter[k++] = fp_lerp(0, chan->vol, fp_div(i << 6, (4 * a) << 6));
+    }
+    // Decay.
+    for (u32 i = 0; i < 4 * d; ++i) {
+        chan->filter[k++] = fp_lerp(chan->vol, chan->vol / 2, fp_div(i << 6, (4 * d) << 6));
+    }
+    // Sustain.
+    for (u32 i = 0; i < 4 * s; ++i) {
+        chan->filter[k++] = chan->vol / 2;
+    }
+    // Release.
+    for (u32 i = 0; i < 4 * r; ++i) {
+        chan->filter[k++] = fp_lerp(chan->vol / 2, 0, fp_div(i << 6, (4 * r) << 6));
+    }
+
+    // Setup the channel vars.
+    chan->adsr = adsr;
+    chan->filter_pos = 0;
+    chan->filter_len = k;
+}
+
 static Audio audio;
 
 #define POLYPHONY 4
@@ -123,12 +190,20 @@ void sound_mix() {
         if (ch->data == NULL || ch->pitch >= 108) {
             continue;
         }
+
+        u32 vol = ch->vol;
+        if (ch->adsr != 0) {
+            vol = ch->filter[ch->filter_pos++];
+            if (ch->filter_pos == ch->filter_len) {
+                continue;
+            }
+        }
         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] += (0x80 + (u8)ch->data[ch->pos >> 12]) * ch->vol;
+                mix_buffer[i] += (0x80 + (u8)ch->data[ch->pos >> 12]) * vol;
                 ch->pos += ch->inc;
 
                 if (ch->pos >= ch->length) {