Cleaup some leftover code and comments

4 files changed, 106 insertions, 137 deletions

diff --git a/src/uxn/devices/apu.c b/src/apu.c
index 47151a6..8dd4ce5 100644
--- a/src/uxn/devices/apu.c
+++ b/src/apu.c
@@ -1,24 +1,3 @@
-// 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 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,
-    236,   250,   264,   280,   297,   315,   333,   353,
-    374,   396,   420,   445,   472,   500,   529,   561,
-    594,   630,   667,   707,   749,   793,   841,   891,
-    944,   1000,  1059,  1122,  1189,  1260,  1335,  1414,
-    1498,  1587,  1682,  1782,  1888,  2000,  2119,  2245,
-    2379,  2520,  2670,  2829,  2997,  3175,  3364,  3564,
-    3776,  4001,  4239,  4491,  4758,  5041,  5341,  5658,
-    5995,  6351,  6729,  7129,  7553,  8002,  8478,  8982,
-    9517,  10083, 10682, 11317, 11990, 12703, 13459, 14259,
-    15107, 16005, 16957, 17965, 19034, 20166, 21365, 22635,
-    23981, 25407, 26918, 28519, 30215, 32011, 33915, 35931,
-    38068, 40332, 42730, 45271, 47963, 50815, 53837, 57038,
-};
-
 //
 //       REG_TM0D                    frequency    buffer size
 //          |                            |            |
@@ -48,7 +27,7 @@ typedef struct Audio {
 
 typedef struct AudioChannel {
     // Pointer to the raw data in the ROM.
-    s8 *data;
+    u8 *data;
     // Current position in the data (20.12 fixed-point).
     u32 pos;
     // Increment (20.12 fixed-point).
@@ -63,70 +42,75 @@ typedef struct AudioChannel {
     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];
+    // The filter is built with the ADSR and has a maximum duration of
+    // 4 seconds. Each component can last up to 1 second.
+    u8 filter[240];
     // 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));
-}
-
+// 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 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,
+    236,   250,   264,   280,   297,   315,   333,   353,
+    374,   396,   420,   445,   472,   500,   529,   561,
+    594,   630,   667,   707,   749,   793,   841,   891,
+    944,   1000,  1059,  1122,  1189,  1260,  1335,  1414,
+    1498,  1587,  1682,  1782,  1888,  2000,  2119,  2245,
+    2379,  2520,  2670,  2829,  2997,  3175,  3364,  3564,
+    3776,  4001,  4239,  4491,  4758,  5041,  5341,  5658,
+    5995,  6351,  6729,  7129,  7553,  8002,  8478,  8982,
+    9517,  10083, 10682, 11317, 11990, 12703, 13459, 14259,
+    15107, 16005, 16957, 17965, 19034, 20166, 21365, 22635,
+    23981, 25407, 26918, 28519, 30215, 32011, 33915, 35931,
+    38068, 40332, 42730, 45271, 47963, 50815, 53837, 57038,
+};
 
 IWRAM_CODE
 void
 build_adsr(AudioChannel *chan, u16 adsr) {
-    // u8 a = (adsr >> 12);
+    chan->filter_pos = 0;
+    if (adsr == chan->adsr) {
+        return;
+    }
+
     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));
+    dma_fill(chan->filter, 0, sizeof(chan->filter), 3);
     u8 k = 0;
 
     // Attack.
+    u32 interval = FP_DIV(1 << 6, (4 * a) << 6, 6);
     for (u32 i = 0; i < 4 * a; ++i) {
-        chan->filter[k++] = fp_lerp(0, chan->vol, fp_div(i << 6, (4 * a) << 6));
+        chan->filter[k++] = FP_LERP(0, chan->vol, i * interval, 6);
     }
     // Decay.
+    interval = FP_DIV(1 << 6, (4 * d) << 6, 6);
     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));
+        chan->filter[k++] = FP_LERP(chan->vol, chan->vol / 2, i * interval, 6);
     }
     // Sustain.
     for (u32 i = 0; i < 4 * s; ++i) {
         chan->filter[k++] = chan->vol / 2;
     }
     // Release.
+    interval = FP_DIV(1 << 6, (4 * r) << 6, 6);
     for (u32 i = 0; i < 4 * r; ++i) {
-        chan->filter[k++] = fp_lerp(chan->vol / 2, 0, fp_div(i << 6, (4 * r) << 6));
+        chan->filter[k++] = FP_LERP(chan->vol / 2, 0, i * interval, 6);
     }
 
     // Setup the channel vars.
     chan->adsr = adsr;
-    chan->filter_pos = 0;
     chan->filter_len = k;
 }
 
@@ -153,35 +137,57 @@ init_sound(void) {
         | SOUND_DSOUND_RIGHT_A
         | SOUND_DSOUND_RESET_A;
 
+    // The timer depends on the buffer length.
     TIMER_DATA_0 = AUDIO_TIMER;
     TIMER_CTRL_0 = TIMER_CTRL_ENABLE;
 }
 
-void sound_vsync() {
-    // buffer 1 just got over
+void
+sound_vsync() {
     if(audio.active_buffer == 1) {
-        // Start playing buffer 0
-        dma_transfer_copy(SOUND_FIFO_A, audio.mix_buffer, 1, 1,
-                DMA_DST_FIXED | DMA_CHUNK_32 | DMA_REFRESH | DMA_REPEAT | DMA_ENABLE);
-
-        // Set the current buffer pointer to the start of buffer 1
+        // Start playing and set the backbuffer.
+        dma_transfer_copy(SOUND_FIFO_A, audio.mix_buffer, 1, 1, DMA_DST_FIXED
+                | DMA_CHUNK_32 | DMA_REFRESH | DMA_REPEAT | DMA_ENABLE);
         audio.current_buffer = audio.mix_buffer + AUDIO_BUF_LEN;
         audio.active_buffer = 0;
     } else {
-        // buffer 0 just got over
-        // DMA points to buffer 1 already, so don't bother stopping and resetting it
-        // Set the current buffer pointer to the start of buffer 0
+        // Flip front/backbuffer.
         audio.current_buffer = audio.mix_buffer;
         audio.active_buffer = 1;
     }
 }
 
 IWRAM_CODE
-void sound_mix() {
-    // Initialize and clear mix_buffer.
-    s16 mix_buffer[AUDIO_BUF_LEN];
-    u32 fill = 0;
-    dma_fill(mix_buffer, fill, sizeof(mix_buffer), 3);
+void
+update_channel(AudioChannel *c, u8 *data, u16 length, u8 pitch, u16 adsr,
+        u8 vol, bool loop) {
+    c->pos = 0;
+    c->length = length << 12;
+    c->data = data;
+    c->vol = vol;
+    c->pitch = pitch;
+
+    if (loop) {
+        c->loop_length = c->length;
+    } else {
+        c->loop_length = 0;
+    }
+
+    u32 sampling_rate = length;
+    if (length > 256) {
+        sampling_rate = 44100;
+    }
+    c->inc = (pitch_table[c->pitch] * sampling_rate) >> 5;
+
+    build_adsr(c, adsr);
+}
+
+IWRAM_CODE
+void
+sound_mix(void) {
+    // Clear mix_buffer.
+    u16 mix_buffer[AUDIO_BUF_LEN];
+    dma_fill(mix_buffer, 0, sizeof(mix_buffer), 3);
 
     // Mix channels into the temporary buffer.
     for (size_t j = 0; j < POLYPHONY; ++j) {
@@ -198,33 +204,23 @@ void sound_mix() {
                 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 ^ ch->data[ch->pos >> 12]) * vol;
-                ch->pos += ch->inc;
-
-                if (ch->pos >= ch->length) {
-                    // If looping is not active disable the channel.
-                    if (ch->loop_length == 0) {
-                        ch->data = NULL;
-                        break;
-                    }
-
-                    // Loop the sample.
-                    while (ch->pos >= ch->length) {
-                        ch->pos -= ch->loop_length;
-                    }
+
+        for(size_t i = 0; i < AUDIO_BUF_LEN; i++) {
+            // Remember we are using fixed point values.
+            mix_buffer[i] += (0x80 ^ (s8)ch->data[ch->pos >> 12]) * vol;
+            ch->pos += ch->inc;
+
+            if (ch->pos >= ch->length) {
+                // If looping is not active disable the channel.
+                if (ch->loop_length == 0) {
+                    ch->data = NULL;
+                    break;
+                }
+
+                // Loop the sample.
+                while (ch->pos >= ch->length) {
+                    ch->pos -= ch->loop_length;
                 }
-            }
-        } else {
-            // 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] +=  (0x80 ^ ch->data[ch->pos>>12]) * vol;
-                ch->pos += ch->inc;
             }
         }
     }
@@ -234,9 +230,9 @@ void sound_mix() {
     u32 *buf_ptr = audio.current_buffer;
     for (size_t i = 0; i < AUDIO_BUF_LEN / 4; i++) {
         u64 mix = mix_ptr[i];
-        buf_ptr[i] = (mix >> 8) & 0xFF
-            | (mix >> 16) & 0xFF00
-            | (mix >> 24) & 0xFF0000
-            | (mix >> 32) & 0xFF000000;
+        buf_ptr[i] = ((mix >> 8) & 0xFF)
+            | ((mix >> 16) & 0xFF00)
+            | ((mix >> 24) & 0xFF0000)
+            | ((mix >> 32) & 0xFF000000);
     }
 }
diff --git a/src/common.h b/src/common.h
index 048708b..0fe2752 100644
--- a/src/common.h
+++ b/src/common.h
@@ -722,6 +722,11 @@ wait_vsync(void) {
 #define CLAMP(X, MIN, MAX) ((X) <= (MIN) ? (MIN) : (X) > (MAX) ? (MAX): (X))
 #define LEN(ARR) (sizeof(ARR) / sizeof((ARR)[0]))
 
+// Fixed-point arithmetic for (i.P) numbers.
+#define FP_MUL(A,B,P)      (((A) * (B)) >> (P))
+#define FP_DIV(A,B,P)      (((A) << (P)) / (B))
+#define FP_LERP(Y0,Y1,X,P) ((Y0) + FP_MUL((X), ((Y1) - (Y0)), P))
+
 //
 // Memory section macros for devkitARM.
 //
diff --git a/src/filesystem.c b/src/filesystem.c
index b60dbde..8aaa51c 100644
--- a/src/filesystem.c
+++ b/src/filesystem.c
@@ -114,7 +114,7 @@ fs_init(void) {
         }
 
         // Initialize filesystem.
-        memset(&filesystem, 0, sizeof(FileSystem));
+        dma_fill(&filesystem, 0, sizeof(FileSystem), 3);
         filesystem.initialized = FS_INIT_PATTERN;
         for (size_t i = 0; i < FILE_INDEX_NUM; ++i) {
             filesystem.files[i].first_block = FS_NULL;
diff --git a/src/main.c b/src/main.c
index aee68f8..35bc0cf 100644
--- a/src/main.c
+++ b/src/main.c
@@ -20,7 +20,7 @@ WITH REGARD TO THIS SOFTWARE.
 #include "uxn/uxn.c"
 #include "uxn/devices/ppu.h"
 #include "uxn/devices/ppu.c"
-#include "uxn/devices/apu.c"
+#include "apu.c"
 
 #include "text.h"
 
@@ -124,44 +124,12 @@ audio_talk(Device *d, u8 b0, u8 w) {
         }
         } else if(b0 == 0xf) {
         u16 length = mempeek16(d->dat, 0xa);
-
-            // Disable the channel before updating.
-                c->data = NULL;
-
-        // Data.
-        c->pos = 0;
-                c->length = length;
-                c->length <<= 12; // fixed point.
-                c->data = &d->mem[mempeek16(d->dat, 0xc)];
-
-        // Volume (Mono only for now).
-        c->vol = MAX(d->dat[0xe] >> 4, d->dat[0xe] & 0xf); // 0-64
-        c->vol *= 4 / 3;
-
-                // Looping.
-                if (!(d->dat[0xf] & 0x80)) {
-            c->loop_length = c->length;
-        } else {
-            c->loop_length = 0;
-        }
-
-        // Pitch
-                c->pitch = d->dat[0xf] & 0x7f;
-
-                // Initialization.
-                u32 sampling_rate = length;
-                if (length > 256) {
-                    sampling_rate = 44100;
-        }
-        c->inc = (pitch_table[c->pitch] * sampling_rate) >> 5;
-
+        u8 *data = &d->mem[mempeek16(d->dat, 0xc)];
+        u8 pitch = d->dat[0xf] & 0x7f;
         u16 adsr = mempeek16(d->dat, 0x8);
-        build_adsr(c, adsr);
-            txt_position(0, 0);
-        txt_printf("note:   %d     \n", c->pitch);
-        txt_printf("inc:    %lu    \n", c->inc);
-        txt_printf("length: %lu    \n", c->length >> 12);
-        txt_printf("chan:   %lu    \n", d - devaudio);
+        u32 vol = MAX(d->dat[0xe] >> 4, d->dat[0xe] & 0xf) * 4 / 3;
+        bool loop = !(d->dat[0xf] & 0x80);
+        update_channel(c, data, length, pitch, adsr, vol, loop);
         }
 }
 
@@ -412,7 +380,7 @@ int main(void) {
     // irs_set(IRQ_TIMER_1, irs_stop_sample);
 
     // Initialize VM.
-    memset(&u, 0, sizeof(u));
+    dma_fill(&u, 0, sizeof(u), 3);
     u.ram.dat = umem;
     init_uxn(&u);