We are so fucking close omg

10 files changed, 122 insertions, 147 deletions

diff --git a/src/audio/audio_context.c b/src/audio/audio_context.c
index 5a34571d..9985b0c0 100644
--- a/src/audio/audio_context.c
+++ b/src/audio/audio_context.c
@@ -233,9 +233,9 @@ AudioSpec gAudioSpecs[] = {
 };
 s32 D_800C7C28 = 0x20000000; // unused?
 s16 gSeqTicksPerBeat = 0x30;
-s32 gAudioHeapSize = 0xAFE00;
-s32 gInitPoolSize = 0x26000;
-u32 gPermanentPoolSize = 0x21000;
+s32 gAudioHeapSize = 0xAFE00 * 2;
+s32 gInitPoolSize = 0x26000 * 2;
+u32 gPermanentPoolSize = 0x21000 * 2;
 u16 gSequenceMedium = 0;
 u16 gSoundFontMedium = 0;
 u16 gSampleBankMedium = 0;
diff --git a/src/audio/audio_effects.c b/src/audio/audio_effects.c
index 61d2ccaa..0a184871 100644
--- a/src/audio/audio_effects.c
+++ b/src/audio/audio_effects.c
@@ -1,5 +1,6 @@
 #include "sys.h"
 #include "sf64audio_provisional.h"
+#include "endianness.h"
 
 static const char devstr[] = "Audio:Envp: overflow  %f\n";
 
@@ -199,7 +200,7 @@ f32 func_80013B90(AdsrState* adsr) {
             adsr->state = ADSR_STATE_LOOP;
         case_ADSR_STATE_LOOP:
         case ADSR_STATE_LOOP:
-            adsr->delay = adsr->envelope[adsr->envIndex].delay;
+            adsr->delay = BSWAP16(adsr->envelope[adsr->envIndex].delay);
             switch (adsr->delay) {
                 case ADSR_DISABLE:
                     adsr->state = ADSR_STATE_DISABLED;
@@ -208,7 +209,7 @@ f32 func_80013B90(AdsrState* adsr) {
                     adsr->state = ADSR_STATE_HANG;
                     break;
                 case ADSR_GOTO:
-                    adsr->envIndex = adsr->envelope[adsr->envIndex].arg;
+                    adsr->envIndex = BSWAP16(adsr->envelope[adsr->envIndex].arg);
                     goto case_ADSR_STATE_LOOP;
                 case ADSR_RESTART:
                     adsr->state = ADSR_STATE_INITIAL;
@@ -221,7 +222,7 @@ f32 func_80013B90(AdsrState* adsr) {
                         adsr->delay = 1;
                     }
 
-                    adsr->target = adsr->envelope[adsr->envIndex].arg / 32767.0f;
+                    adsr->target = (s16)BE16SWAP(adsr->envelope[adsr->envIndex].arg) / 32767.0f;
                     adsr->target = SQ(adsr->target);
                     adsr->velocity = (adsr->target - adsr->current) / adsr->delay;
                     adsr->state = ADSR_STATE_FADE;
diff --git a/src/audio/audio_general.c b/src/audio/audio_general.c
index 70b73fc1..3f91fa5e 100644
--- a/src/audio/audio_general.c
+++ b/src/audio/audio_general.c
@@ -637,7 +637,7 @@ void Audio_ResetSfxChannelState(void) {
 }
 
 void Audio_StartSequence(u8 seqPlayId, u8 seqId, u8 seqArgs, u16 fadeInTime) {
-    u8 i;
+    u8 i = 0;
     s32 pad;
 
     if (!sStartSeqDisabled || (seqPlayId == SEQ_PLAYER_SFX)) {
@@ -2644,8 +2644,7 @@ void Audio_RestoreVolumeSettings(u8 audioType) {
     }
 }
 
-void Audio_SetVolume(u8 audioType, u8 volume) {
-    return;
+void Audio_SetVolume(u8 audioType, u8 volume) {    
     if (volume > 99) {
         volume = 99;
     }
diff --git a/src/audio/audio_heap.c b/src/audio/audio_heap.c
index 663f5fd7..ea35a463 100644
--- a/src/audio/audio_heap.c
+++ b/src/audio/audio_heap.c
@@ -148,6 +148,7 @@ void* AudioHeap_Alloc(AudioAllocPool* pool, u32 size) {
 
 void AudioHeap_InitPool(AudioAllocPool* pool, void* ramAddr, u32 size) {
     pool->curRamAddr = pool->startRamAddr = (u8*) ramAddr;
+    size *= 4;
     pool->size = size - ((uintptr_t) ramAddr & 0xF);
     pool->numEntries = 0;
 }
diff --git a/src/audio/audio_load.c b/src/audio/audio_load.c
index 3cbc7d6d..eedb62f2 100644
--- a/src/audio/audio_load.c
+++ b/src/audio/audio_load.c
@@ -1,8 +1,12 @@
+#include <stdlib.h>
 #include "sys.h"
 #include "sf64dma.h"
 #include "sf64audio_provisional.h"
 #include "assets/ast_audio.h"
 #include "port/Engine.h"
+#include "endianness.h"
+#include "port/Engine.h"
+#include "port/resource/loaders/AudioLoader.h"
 
 s32 D_80146D80;
 s32 PAD_80146D88[2];
@@ -33,8 +37,8 @@ void AudioLoad_ProcessAsyncLoads(s32 resetStatus);
 void AudioLoad_ProcessAsyncLoad(AudioAsyncLoad* asyncLoad, s32 resetStatus);
 void AudioLoad_AsyncDma(AudioAsyncLoad* asyncLoad, u32 size);
 void AudioLoad_AsyncDmaUnkMedium(uintptr_t devAddr, u8* ramAddr, u32 size, s32 unkMediumParam);
-void AudioLoad_RelocateSample(TunedSample* tSample, u32 fontDataAddr, SampleBankRelocInfo* relocInfo);
-s32 AudioLoad_RelocateFontAndPreloadSamples(s32 fontId, u32 fontDataAddr, SampleBankRelocInfo* relocData, s32 isAsync);
+void AudioLoad_RelocateSample(TunedSample* sample, TunedSample32Bit* tSample, uintptr_t fontDataAddr, SampleBankRelocInfo* relocInfo);
+s32 AudioLoad_RelocateFontAndPreloadSamples(s32 fontId, uintptr_t fontDataAddr, SampleBankRelocInfo* relocData, s32 isAsync);
 s32 AudioLoad_ProcessSamplePreloads(s32 resetStatus);
 
 void AudioLoad_DecreaseSampleDmaTtls(void) {
@@ -386,13 +390,12 @@ void AudioLoad_SyncInitSeqPlayerInternal(s32 playerIdx, s32 seqId, s32 arg2) {
 
     func_800144E4(&gSeqPlayers[playerIdx]);
 
-    index = *((u16*) gSeqFontTable + seqId);
-    numFonts = gSeqFontTable[index++];
+    index = BSWAP16(*((u16*) gSeqFontTable + seqId));
     fontId = 0xFF;
 
-    for (numFonts; numFonts > 0; numFonts--) {
+    for (numFonts = gSeqFontTable[index++]; numFonts > 0; numFonts--) {
         fontId = gSeqFontTable[index++];
-        AudioLoad_SyncLoadFont(fontId);
+        // AudioLoad_SyncLoadFont(fontId);
     }
 
     seqData = AudioLoad_SyncLoadSeq(seqId);
@@ -401,23 +404,17 @@ void AudioLoad_SyncInitSeqPlayerInternal(s32 playerIdx, s32 seqId, s32 arg2) {
 
     gSeqPlayers[playerIdx].seqId = seqId;
     gSeqPlayers[playerIdx].defaultFont = fontId;
-    gSeqPlayers[playerIdx].enabled = true;
+    gSeqPlayers[playerIdx].enabled = 1;
     gSeqPlayers[playerIdx].seqData = seqData;
     gSeqPlayers[playerIdx].scriptState.pc = seqData;
     gSeqPlayers[playerIdx].scriptState.depth = 0;
     gSeqPlayers[playerIdx].delay = 0;
-    gSeqPlayers[playerIdx].finished = false;
-
-    for (i = 0; i < 16; i++) {
-        ;
-    }
+    gSeqPlayers[playerIdx].finished = 0;
 }
 
 void* AudioLoad_SyncLoadSeq(s32 seqId) {
-    s32 seqIdx = AudioLoad_GetLoadTableIndex(SEQUENCE_TABLE, seqId);
-    s32 didAllocate;
-
-    return AudioLoad_SyncLoad(0, seqIdx, &didAllocate);
+    AudioTable* table = AudioLoad_GetLoadTable(SEQUENCE_TABLE);
+    return Audio_LoadBlob(gAudioTable, table->entries[seqId].romAddr);
 }
 
 void* AudioLoad_SyncLoadSampleBank(u32 sampleBankId, s32* outMedium) {
@@ -498,7 +495,7 @@ void* AudioLoad_SyncLoad(u32 tableType, u32 id, s32* didAllocate) {
     u8* ramAddr;
     u32 medium;
     s32 loadStatus;
-    u32 romAddr;
+    uintptr_t romAddr;
     s32 pad;
     s32 cachePolicy;
 
@@ -621,12 +618,10 @@ AudioTable* AudioLoad_GetLoadTable(s32 tableType) {
     return table;
 }
 
-void AudioLoad_RelocateFont(s32 fontId, u32 fontBaseAddr, void* relocData) {
-    u32* fontDataPtrs = fontBaseAddr;
-    u32** drumDataPtrs = fontBaseAddr;
+void AudioLoad_RelocateFont(s32 fontId, uintptr_t fontBaseAddr, SampleBankRelocInfo* relocData) {
+    uint32_t* fontDataPtrs = fontBaseAddr;
+    uint32_t** drumDataPtrs = fontBaseAddr;
     s32 numDrums;
-    Drum* drum;
-    Instrument* instrument;
     u32 offset;
     s32 i;
     s32 numInstruments;
@@ -634,53 +629,41 @@ void AudioLoad_RelocateFont(s32 fontId, u32 fontBaseAddr, void* relocData) {
     numDrums = gSoundFontList[fontId].numDrums;
     numInstruments = gSoundFontList[fontId].numInstruments;
 
-    if ((fontDataPtrs[0] != 0) && (numDrums != 0)) {
-        fontDataPtrs[0] += fontBaseAddr;
+    Drum** relocatedDrums = memalloc(numDrums * sizeof(Drum*));
+    Instrument** relocatedInstruments = memalloc((numInstruments + 1) * sizeof(Instrument*));
 
+    if ((fontDataPtrs[0] != 0) && (numDrums != 0)) {
         for (i = 0; i < numDrums; i++) {
             offset = *(*drumDataPtrs + i);
             if (offset != 0) {
-                drum = offset += fontBaseAddr;
-                *(*drumDataPtrs + i) = drum;
-
-                if (!drum->isRelocated) {
-                    AudioLoad_RelocateSample(&drum->tunedSample, fontBaseAddr, relocData);
-                    offset = (u32) drum->envelope;
-                    drum->envelope = offset + fontBaseAddr;
-                    drum->isRelocated = true;
-                }
+                relocatedDrums[i] = Audio_LoadDrum(BSWAP32(offset) + fontBaseAddr, 0, 0);
             }
         }
     }
 
     for (i = 1; i <= numInstruments; i++) {
         if (fontDataPtrs[i] != 0) {
+            Instrument* instrument = Audio_LoadInstrument(BSWAP32(fontDataPtrs[i]) + fontBaseAddr, 0);
+            Instrument32Bit* base = fontBaseAddr + BSWAP32(fontDataPtrs[i]);
 
-            fontDataPtrs[i] += fontBaseAddr;
-
-            instrument = fontDataPtrs[i];
-            if (!instrument->isRelocated) {
-                if (instrument->normalRangeLo != 0) {
-                    AudioLoad_RelocateSample(&instrument->lowPitchTunedSample, fontBaseAddr, relocData);
-                }
-                AudioLoad_RelocateSample(&instrument->normalPitchTunedSample, fontBaseAddr, relocData);
-                if (instrument->normalRangeHi != 0x7F) {
-                    AudioLoad_RelocateSample(&instrument->highPitchTunedSample, fontBaseAddr, relocData);
-                }
-                offset = (u32) instrument->envelope;
-                instrument->envelope = offset + fontBaseAddr;
-                instrument->isRelocated = true;
+            if (instrument->normalRangeLo != 0) {
+                AudioLoad_RelocateSample(&instrument->lowPitchTunedSample, &base->lowPitchTunedSample, fontBaseAddr, relocData);
             }
+            AudioLoad_RelocateSample(&instrument->normalPitchTunedSample, &base->normalPitchTunedSample, fontBaseAddr, relocData);
+            if (instrument->normalRangeHi != 0x7F) {
+                AudioLoad_RelocateSample(&instrument->highPitchTunedSample, &base->highPitchTunedSample, fontBaseAddr, relocData);
+            }
+
+            relocatedInstruments[i] = instrument;
         }
     }
 
-    gSoundFontList[fontId].drums = fontDataPtrs[0];
-    gSoundFontList[fontId].instruments = (u32) &fontDataPtrs[1];
+    gSoundFontList[fontId].drums = relocatedDrums;
+    gSoundFontList[fontId].instruments = relocatedInstruments;
 }
 
 void AudioLoad_SyncDma(uintptr_t devAddr, u8* ramAddr, u32 size, s32 medium) {
     size = ALIGN16(size);
-
     osInvalDCache(ramAddr, size);
 
     while (true) {
@@ -738,7 +721,7 @@ s32 AudioLoad_Dma(OSIoMesg* mesg, u32 priority, s32 direction, uintptr_t devAddr
     mesg->devAddr = devAddr;
     mesg->size = size;
 
-    handle->transferInfo.cmdType = 2;
+    // handle->transferInfo.cmdType = 2;
     // osEPiStartDma(handle, mesg, direction);
     memcpy(ramAddr, (void*) devAddr, size);
 
@@ -941,19 +924,17 @@ void AudioLoad_Init(void) {
     gSeqFontTable = gSeqFontTableInit;
     gNumSequences = gSequenceTable->base.numEntries;
 
-    AudioLoad_InitTable(gSequenceTable, LOAD_ASSET(gAudioSeq), gSequenceMedium);
-    AudioLoad_InitTable(gSoundFontTable, LOAD_ASSET(gAudioBank), gSoundFontMedium);
-    AudioLoad_InitTable(gSampleBankTable, LOAD_ASSET(gAudioTable), gSampleBankMedium);
+//    AudioLoad_InitTable(gSequenceTable, LOAD_ASSET(gAudioSeq), gSequenceMedium);
+//    AudioLoad_InitTable(gSoundFontTable, LOAD_ASSET(gAudioBank), gSoundFontMedium);
+//    AudioLoad_InitTable(gSampleBankTable, LOAD_ASSET(gAudioTable), gSampleBankMedium);
 
     numFonts = gSoundFontTable->base.numEntries;
 
     gSoundFontList = AudioHeap_Alloc(&gInitPool, numFonts * sizeof(SoundFont));
 
     for (i = 0; i < numFonts; i++) {
-        gSoundFontList[i].sampleBankId1 = (gSoundFontTable->entries[i].shortData1 >> 8) & 0xFF;
-        gSoundFontList[i].sampleBankId2 = gSoundFontTable->entries[i].shortData1 & 0xFF;
-        gSoundFontList[i].numInstruments = (gSoundFontTable->entries[i].shortData2 >> 8) & 0xFF;
-        gSoundFontList[i].numDrums = gSoundFontTable->entries[i].shortData2 & 0xFF;
+        gSoundFontList[i] = Audio_LoadFont(gSoundFontTable->entries[i]);
+        gFontLoadStatus[i] = 2;
     }
 
     ramAddr = AudioHeap_Alloc(&gInitPool, gPermanentPoolSize);
@@ -1277,35 +1258,40 @@ void AudioLoad_AsyncDmaUnkMedium(uintptr_t devAddr, u8* ramAddr, u32 size, s32 u
 static const char devstr50[] = "Error: Already wavetable is touched %x.\n";
 static const char devstr51[] = "Touch Warning: Length zero %x\n";
 
-void AudioLoad_RelocateSample(TunedSample* tSample, u32 fontDataAddr, SampleBankRelocInfo* relocInfo) {
-    void* reloc;
-    Sample* sample;
-
-    // "Error: Already wavetable is touched %x.\n";
-    if ((u32) tSample->sample <= AUDIO_RELOCATED_ADDRESS_START) {
-        sample = tSample->sample = reloc = (u32) tSample->sample + fontDataAddr;
-        // "Touch Warning: Length zero %x\n";
-        if ((sample->size != 0) && (sample->isRelocated != 1)) {
-            sample->loop = reloc = (u32) sample->loop + fontDataAddr;
-            sample->book = reloc = (u32) sample->book + fontDataAddr;
-            switch (sample->medium) {
-                case MEDIUM_RAM:
-                    sample->sampleAddr = reloc = sample->sampleAddr + relocInfo->baseAddr1;
-                    sample->medium = relocInfo->medium1;
-                    break;
-                case MEDIUM_UNK:
-                    sample->sampleAddr = reloc = sample->sampleAddr + relocInfo->baseAddr2;
-                    sample->medium = relocInfo->medium2;
-                    break;
-                case MEDIUM_CART:
-                case MEDIUM_DISK_DRIVE:
-                    break;
-            }
+void AudioLoad_RelocateSample(TunedSample* tunedSample, TunedSample32Bit* tSample, uintptr_t fontDataAddr, SampleBankRelocInfo* relocInfo) {
+    Sample32Bit* baseSample = fontDataAddr + BSWAP32(tSample->sample);
+
+    // "Touch Warning: Length zero %x\n";
+    if ((baseSample->size != 0) && (baseSample->isRelocated != 1)) {
+        Sample* output = memalloc(sizeof(Sample));
+
+        output->loop = Audio_LoadLoop(fontDataAddr + BSWAP32(baseSample->loop));
+        output->book = Audio_LoadBook(fontDataAddr + BSWAP32(baseSample->book));
+        output->codec       = baseSample->codec;
+        output->medium      = baseSample->medium;
+        output->unk_bit26   = baseSample->unk_bit26;
+        output->isRelocated = baseSample->isRelocated;
+        output->size        = baseSample->size;
+
+        output->isRelocated = 1;
+        switch (baseSample->medium) {
+            case MEDIUM_RAM:
+//                baseSample->sampleAddr = Audio_LoadCopy(relocInfo->baseAddr1 + BSWAP32(baseSample->sampleAddr), BSWAP32(baseSample->size));
+                baseSample->medium = relocInfo->medium1;
+                break;
+            case MEDIUM_UNK:
+//                baseSample->sampleAddr = Audio_LoadCopy(relocInfo->baseAddr2 + BSWAP32(baseSample->sampleAddr), BSWAP32(baseSample->size));
+                baseSample->medium = relocInfo->medium2;
+                break;
+            case MEDIUM_CART:
+            case MEDIUM_DISK_DRIVE:
+                output->sampleAddr = fontDataAddr + BSWAP32(baseSample->sampleAddr);
+                break;
+        }
 
-            sample->isRelocated = true;
-            if (sample->unk_bit26 && (sample->medium != 0)) {
-                gUsedSamples[gNumUsedSamples++] = sample;
-            }
+        baseSample->isRelocated = true;
+        if (baseSample->unk_bit26 && (baseSample->medium != 0)) {
+            gUsedSamples[gNumUsedSamples++] = output;
         }
     }
 }
@@ -1316,7 +1302,7 @@ static const char devstr54[] = "Warning: Length zero %x\n";
 static const char devstr55[] = "Wave Load %d \n";
 static const char devstr56[] = "Total Bg Wave Load %d \n";
 
-s32 AudioLoad_RelocateFontAndPreloadSamples(s32 fontId, u32 fontDataAddr, SampleBankRelocInfo* relocData, s32 isAsync) {
+s32 AudioLoad_RelocateFontAndPreloadSamples(s32 fontId, uintptr_t fontDataAddr, SampleBankRelocInfo* relocData, s32 isAsync) {
     s32 i;
     Sample* sample;
     u8* sampleRamAddr;
@@ -1534,4 +1520,4 @@ s32 AudioLoad_GetSamplesForFont(s32 fontId, Sample** sampleSet) {
     }
 
     return numLoaded;
-}
+}
\ No newline at end of file
diff --git a/src/audio/audio_playback.c b/src/audio/audio_playback.c
index a4b2ae46..05cd56ef 100644
--- a/src/audio/audio_playback.c
+++ b/src/audio/audio_playback.c
@@ -248,10 +248,6 @@ void func_80011FA8(void) {
 
         playbackState = &note->playbackState;
         if ((playbackState->parentLayer != NO_LAYER)) {
-            if ((u32) playbackState->parentLayer < 0x7FFFFFFF) {
-                continue;
-            }
-
             if ((note != playbackState->parentLayer->note) && (playbackState->unk_04 == 0)) {
                 playbackState->adsr.action.asByte |= 0x10;
                 playbackState->adsr.fadeOutVel = gAudioBufferParams.ticksPerUpdateInv;
@@ -284,7 +280,6 @@ void func_80011FA8(void) {
     block_21:
 
         if (playbackState->priority != 0) {
-            if (1) {}
             noteSub = &note->noteSubEu;
             if ((playbackState->unk_04 > 0) || noteSub->bitField0.finished) {
                 if ((playbackState->adsr.state == 0) || noteSub->bitField0.finished) {
diff --git a/src/audio/audio_seqplayer.c b/src/audio/audio_seqplayer.c
index dabcb992..62e6ffbe 100644
--- a/src/audio/audio_seqplayer.c
+++ b/src/audio/audio_seqplayer.c
@@ -268,7 +268,7 @@ void* func_800145FC(AudioListItem* list) {
     list->prev = item->prev;
     item->prev = NULL;
     list->u.count--;
-    return (void*) item->u.count;
+    return (void*) item->u.value;
 }
 
 void func_8001463C(void) {
diff --git a/src/audio/audio_synthesis.c b/src/audio/audio_synthesis.c
index 5ec7c962..7de5dd3f 100644
--- a/src/audio/audio_synthesis.c
+++ b/src/audio/audio_synthesis.c
@@ -1,6 +1,7 @@
 #include "sys.h"
 #include "sf64audio_provisional.h"
 #include "audio/mixer.h"
+#include "endianness.h"
 
 s32 D_80145D40; // unused
 
@@ -745,8 +746,8 @@ Acmd* func_8000A25C(s16* aiBuf, s32 aiBufLen, Acmd* aList, s32 updateIndex) {
     u8 sp84[0x3C];
     NoteSubEu* temp_v0;
     s16 var_s2;
-    s16 i;
-    s32 j;
+    s16 i = 0;
+    s32 j = 0;
 
     var_s2 = 0;
     if (gNumSynthReverbs == 0) {
@@ -832,7 +833,7 @@ Acmd* func_8000A700(s32 noteIndex, NoteSubEu* noteSub, NoteSynthesisState* synth
     s32 padD8;
     s32 padD4;
     s32 padD0;
-    u32 sampleAddr; // spCC
+    uintptr_t sampleAddr; // spCC
     s32 padC8;
     s32 samplesLenAdjusted;     // spC4
     s32 nAdpcmSamplesProcessed; // spC0
@@ -1011,7 +1012,7 @@ Acmd* func_8000A700(s32 noteIndex, NoteSubEu* noteSub, NoteSynthesisState* synth
                                                              flags, &synthState->sampleDmaIndex, bookSample->medium);
                     }
                     // if (1){}
-                    sampleDataStartPad = (u32) sampleData & 0xF;
+                    sampleDataStartPad = (uintptr_t) sampleData & 0xF;
                     aLoadBuffer(aList++, OS_K0_TO_PHYSICAL(sampleData - sampleDataStartPad), addr, aligned);
                 } else {
                     nSamplesToDecode = 0;
diff --git a/src/audio/audio_thread.c b/src/audio/audio_thread.c
index fb4144cf..0b42ed33 100644
--- a/src/audio/audio_thread.c
+++ b/src/audio/audio_thread.c
@@ -42,68 +42,58 @@ static const char devstr14[] = "Error : Queue is not empty ( %x ) \n";
 void AudioThread_CreateNextAudioBuffer(s16 *samples, u32 num_samples) {
     static s32 gMaxAbiCmdCnt = 128;
     static SPTask* gWaitingAudioTask = NULL;
-    u32 sp54;
-    s32 sp50;
-    s32 sp4C;
-    s32 pad48;
-    OSTask_t* task;
-    u16* sp40;
-    s32 pad3C;
-    OSMesg sp38;
-    OSMesg sp34;
-    s32 pad30;
+    s32 abiCmdCount;
+    u32 specId;
+    OSMesg msg;
+
     gAudioTaskCountQ++;
     if ((gAudioTaskCountQ % gAudioBufferParams.count) != 0) {
-        return;
+        return gWaitingAudioTask;
     }
-    osSendMesg(gAudioTaskStartQueue, OS_MESG_32(gAudioTaskCountQ), 0);
+    osSendMesg(gAudioTaskStartQueue, OS_MESG_32(gAudioTaskCountQ), OS_MESG_NOBLOCK);
     gAudioTaskIndexQ ^= 1;
-    gCurAiBuffIndex++;
-    gCurAiBuffIndex %= 3;
-    sp4C = (gCurAiBuffIndex + 1) % 3;
-    sp54 = osAiGetLength() >> 2;
-    if ((gAudioResetTimer < 16) && (gAiBuffLengths[sp4C] != 0)) {
-        osAiSetNextBuffer(gAiBuffers[sp4C], gAiBuffLengths[sp4C] * 4);
-    }
-    if (gCurAudioFrameDmaCount && gCurAudioFrameDmaCount) {}
+
     gCurAudioFrameDmaCount = 0;
     AudioLoad_DecreaseSampleDmaTtls();
-    AudioLoad_ProcessLoads(gAudioResetStep);
-    if (osRecvMesg(&gAudioTaskMesgQueue, &sp38, 0) != -1) {
+    // AudioLoad_ProcessLoads(gAudioResetStep);
+
+    if (MQ_GET_MESG(gAudioSpecQueue, &specId)) {
         if (gAudioResetStep == 0) {
             gAudioResetStep = 5;
         }
-        gAudioSpecId = sp38.data32;
+        gAudioSpecId = specId;
     }
+
     if ((gAudioResetStep != 0) && (AudioHeap_ResetStep() == 0)) {
         if (gAudioResetStep == 0) {
-            osSendMesg(gAudioResetQueue, OS_MESG_32((s32) gAudioSpecId), 0);
+            osSendMesg(gAudioResetQueue, OS_MESG_32((s32) gAudioSpecId), OS_MESG_NOBLOCK);
         }
         gWaitingAudioTask = NULL;
-        return;
+        return NULL;
     }
+
     if (gAudioResetTimer > 16) {
-        return;
+        return NULL;
     }
     if (gAudioResetTimer != 0) {
         gAudioResetTimer++;
     }
+
     gAudioCurTask = &gAudioRspTasks[gAudioTaskIndexQ];
     gCurAbiCmdBuffer = gAbiCmdBuffs[gAudioTaskIndexQ];
-    sp4C = gCurAiBuffIndex;
-    sp40 = gAiBuffers[sp4C];
-    gAiBuffLengths[sp4C] = num_samples;
-    if (gAiBuffLengths[sp4C] < gAudioBufferParams.minAiBufferLength) {
-        gAiBuffLengths[sp4C] = gAudioBufferParams.minAiBufferLength;
-    }
-    if (gAiBuffLengths[sp4C] > gAudioBufferParams.maxAiBufferLength) {
-        gAiBuffLengths[sp4C] = gAudioBufferParams.maxAiBufferLength;
-    }
-    while (osRecvMesg(gThreadCmdProcQueue, &sp34, 0) != -1) {
-        AudioThread_ProcessCmds(sp34.data32);
+
+    while (MQ_GET_MESG(gThreadCmdProcQueue, &msg)) {
+        AudioThread_ProcessCmds(msg.data32);
     }
-    gCurAbiCmdBuffer = func_80009B64(gCurAbiCmdBuffer, &sp50, samples, num_samples);
+
+    gCurAbiCmdBuffer = func_80009B64(gCurAbiCmdBuffer, &abiCmdCount, samples, num_samples);
     gAudioRandom = osGetCount() * (gAudioRandom + gAudioTaskCountQ);
+
+    gAudioCurTask->msg = OS_MESG_PTR(NULL);
+
+    if (gMaxAbiCmdCnt < abiCmdCount) {
+        gMaxAbiCmdCnt = abiCmdCount;
+    }
 }
 
 SPTask* AudioThread_CreateTask() {
@@ -233,7 +223,7 @@ void AudioThread_ProcessGlobalCmd(AudioCmd* cmd) {
         case AUDIOCMD_OP_GLOBAL_INIT_SEQPLAYER:
         case AUDIOCMD_OP_GLOBAL_INIT_SEQPLAYER_ALT:
             AudioLoad_SyncInitSeqPlayer(cmd->arg0, cmd->arg1, cmd->arg2);
-            AudioThread_SetFadeInTimer(cmd->arg0, (s32) cmd->data);
+            AudioThread_SetFadeInTimer(cmd->arg0, (uintptr_t) cmd->data);
             break;
         case AUDIOCMD_OP_GLOBAL_DISABLE_SEQPLAYER:
             if (gSeqPlayers[cmd->arg0].enabled) {
@@ -355,7 +345,7 @@ void AudioThread_ScheduleProcessCmds(void) {
         D_800C7C70 = (u8) (gThreadCmdWritePos - gThreadCmdReadPos + 0x100);
     }
     msg = (((gThreadCmdReadPos & 0xFF) << 8) | (gThreadCmdWritePos & 0xFF));
-    osSendMesg(gThreadCmdProcQueue, OS_MESG_32(msg), OS_MESG_NOBLOCK);
+    osSendMesg32(gThreadCmdProcQueue, msg, OS_MESG_NOBLOCK);
     gThreadCmdReadPos = gThreadCmdWritePos;
 }
 
diff --git a/src/audio/note_data.c b/src/audio/note_data.c
index c7e2574b..32256fc2 100644
--- a/src/audio/note_data.c
+++ b/src/audio/note_data.c
@@ -1,4 +1,5 @@
 #include "sys.h"
+#include "endianness.h"
 #include "sf64audio_provisional.h"
 
 f32 gBendPitchOneOctaveFrequencies[] = {
@@ -197,6 +198,7 @@ u8 gDefaultShortNoteVelocityTable[] = {
 u8 gDefaultShortNoteGateTimeTable[] = {
     229, 203, 177, 151, 139, 126, 113, 100, 87, 74, 61, 48, 36, 23, 10, 0,
 };
+
 EnvelopePoint gDefaultEnvelope[] = {
     { 4, 32000 },
     { 1000, 32000 },