Cleaup some leftover code and comments

1 files changed, 0 insertions, 242 deletions

diff --git a/src/uxn/devices/apu.c b/src/uxn/devices/apu.c
deleted file mode 100644
index 47151a6..0000000
--- a/src/uxn/devices/apu.c
+++ /dev/null
@@ -1,242 +0,0 @@
-// 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
-//          |                            |            |
-//          V                            V            V
-//
-// Timer = 62610 = 65536 - (16777216 /  5734), buf = 96
-// Timer = 63940 = 65536 - (16777216 / 10512), buf = 176
-// Timer = 64282 = 65536 - (16777216 / 13379), buf = 224
-// Timer = 64612 = 65536 - (16777216 / 18157), buf = 304
-// Timer = 64738 = 65536 - (16777216 / 21024), buf = 352
-// Timer = 64909 = 65536 - (16777216 / 26758), buf = 448
-// Timer = 65004 = 65536 - (16777216 / 31536), buf = 528
-// Timer = 65073 = 65536 - (16777216 / 36314), buf = 608
-// Timer = 65118 = 65536 - (16777216 / 40137), buf = 672
-// Timer = 65137 = 65536 - (16777216 / 42048), buf = 704
-//
-// Source: https://deku.gbadev.org/program/sound1.html
-#define AUDIO_FREQ    18157
-#define AUDIO_BUF_LEN 304
-#define AUDIO_TIMER   64612
-
-typedef struct Audio {
-    s8 mix_buffer[AUDIO_BUF_LEN * 2];
-    s8 *current_buffer;
-    u8 active_buffer;
-} Audio;
-
-typedef struct AudioChannel {
-    // Pointer to the raw data in the ROM.
-    s8 *data;
-    // Current position in the data (20.12 fixed-point).
-    u32 pos;
-    // Increment (20.12 fixed-point).
-    u32 inc;
-    // Volume (0-64, 1.6 fixed-point).
-    u32 vol;
-    // Sound length (20.12 fixed-point).
-    u32 length;
-    // Length of looped portion (20.12 fixed-point, 0 to disable looping).
-    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
-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] = (AudioChannel){0};
-    }
-
-    // Enable the sound chip.
-    SOUND_STATUS = SOUND_ENABLE;
-
-    // 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;
-
-    TIMER_DATA_0 = AUDIO_TIMER;
-    TIMER_CTRL_0 = TIMER_CTRL_ENABLE;
-}
-
-void sound_vsync() {
-    // buffer 1 just got over
-    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
-        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
-        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);
-
-    // Mix channels into the temporary buffer.
-    for (size_t j = 0; j < POLYPHONY; ++j) {
-        AudioChannel *ch = &channels[j];
-        // Check if channel is active.
-        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 ^ 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;
-            }
-        }
-    }
-
-    // Downsample and copy to the playing buffer (Vectorized).
-    u64 *mix_ptr = mix_buffer;
-    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;
-    }
-}