#ifndef Z64AUDIO_H
#define Z64AUDIO_H

#include "audiothread_cmd.h"

#define NO_LAYER ((SequenceLayer*)(-1))

#define TATUMS_PER_BEAT 48

#define IS_SEQUENCE_CHANNEL_VALID(ptr) ((uintptr_t)(ptr) != (uintptr_t)&gAudioCtx.sequenceChannelNone)
#define SEQ_NUM_CHANNELS 16
#define SEQ_IO_VAL_NONE -1

typedef enum {
    /* 0x00 */ AUDIO_CUSTOM_FUNCTION_SEQ_0,
    /* 0x01 */ AUDIO_CUSTOM_FUNCTION_SEQ_1,
    /* 0x02 */ AUDIO_CUSTOM_FUNCTION_SEQ_2,
    /* 0x03 */ AUDIO_CUSTOM_FUNCTION_SEQ_3,
    /* 0xFE */ AUDIO_CUSTOM_FUNCTION_SYNTH = 0xFE,
    /* 0xFF */ AUDIO_CUSTOM_FUNCTION_REVERB
} AudioCustomFunctions;

typedef enum {
    /* 0 */ SEQPLAYER_STATE_0,
    /* 1 */ SEQPLAYER_STATE_FADE_IN,
    /* 2 */ SEQPLAYER_STATE_FADE_OUT
} SeqPlayerState;

#define MAX_CHANNELS_PER_BANK 3

#define MUTE_FLAGS_STOP_SAMPLES (1 << 3)           // prevent further noteSubEus from playing
#define MUTE_FLAGS_STOP_LAYER (1 << 4)           // stop something in seqLayer scripts
#define MUTE_FLAGS_SOFTEN (1 << 5)      // lower volume, by default to half
#define MUTE_FLAGS_STOP_NOTES (1 << 6)  // prevent further notes from playing
#define MUTE_FLAGS_STOP_SCRIPT (1 << 7) // stop processing sequence/channel scripts

#define AUDIO_LERPIMP(v0, v1, t) (v0 + ((v1 - v0) * t))

#define ADSR_DISABLE 0
#define ADSR_HANG -1
#define ADSR_GOTO -2
#define ADSR_RESTART -3

// size of a single sample point
#define SAMPLE_SIZE sizeof(s16)

// Samples are processed in groups of 16 called a "frame"
#define SAMPLES_PER_FRAME ADPCMFSIZE

// The length of one left/right channel is 13 frames
#define DMEM_1CH_SIZE (13 * SAMPLES_PER_FRAME * SAMPLE_SIZE)
// Both left and right channels
#define DMEM_2CH_SIZE (2 * DMEM_1CH_SIZE)

#define AIBUF_LEN (88 * SAMPLES_PER_FRAME) // number of samples
#define AIBUF_SIZE (AIBUF_LEN * SAMPLE_SIZE) // number of bytes

// Filter sizes
#define FILTER_SIZE (8 * SAMPLE_SIZE)
#define FILTER_BUF_PART1 (8 * SAMPLE_SIZE)
#define FILTER_BUF_PART2 (8 * SAMPLE_SIZE)

// Must be the same amount of samples as copied by aDuplicate() (audio microcode)
#define WAVE_SAMPLE_COUNT 64

#define AUDIO_RELOCATED_ADDRESS_START K0BASE

#define REVERB_INDEX_NONE -1

// To be used with AudioThread_CountAndReleaseNotes()
#define AUDIO_NOTE_RELEASE (1 << 0)
#define AUDIO_NOTE_SAMPLE_NOTES (1 << 1)

typedef enum {
    /* 0 */ REVERB_DATA_TYPE_SETTINGS, // Reverb Settings (Init)
    /* 1 */ REVERB_DATA_TYPE_DELAY, // Reverb Delay (numSamples)
    /* 2 */ REVERB_DATA_TYPE_DECAY, // Reverb Decay Ratio
    /* 3 */ REVERB_DATA_TYPE_SUB_VOLUME, // Reverb Sub-Volume
    /* 4 */ REVERB_DATA_TYPE_VOLUME, // Reverb Volume
    /* 5 */ REVERB_DATA_TYPE_LEAK_RIGHT, // Reverb Leak Right Channel
    /* 6 */ REVERB_DATA_TYPE_LEAK_LEFT, // Reverb Leak Left Channel
    /* 7 */ REVERB_DATA_TYPE_FILTER_LEFT, // Reverb Left Filter
    /* 8 */ REVERB_DATA_TYPE_FILTER_RIGHT, // Reverb Right Filter
    /* 9 */ REVERB_DATA_TYPE_9 // Reverb Unk
} ReverbDataType;

typedef enum {
    /*  0x1 */ AUDIO_ERROR_NO_INST = 1,
    /*  0x3 */ AUDIO_ERROR_INVALID_INST_ID = 3,
    /*  0x4 */ AUDIO_ERROR_INVALID_DRUM_SFX_ID,
    /*  0x5 */ AUDIO_ERROR_NO_DRUM_SFX,
    /* 0x10 */ AUDIO_ERROR_FONT_NOT_LOADED = 0x10
} AudioError;

#define AUDIO_ERROR(fontId, id, err) (((fontId << 8) + id) + (err << 24))

typedef enum {
    /* 0 */ SOUNDMODE_STEREO,
    /* 1 */ SOUNDMODE_HEADSET,
    /* 2 */ SOUNDMODE_SURROUND_EXTERNAL,
    /* 3 */ SOUNDMODE_MONO,
    /* 4 */ SOUNDMODE_SURROUND,
} SoundMode;

typedef enum {
    /* 0 */ ADSR_STATE_DISABLED,
    /* 1 */ ADSR_STATE_INITIAL,
    /* 2 */ ADSR_STATE_START_LOOP,
    /* 3 */ ADSR_STATE_LOOP,
    /* 4 */ ADSR_STATE_FADE,
    /* 5 */ ADSR_STATE_HANG,
    /* 6 */ ADSR_STATE_DECAY,
    /* 7 */ ADSR_STATE_RELEASE,
    /* 8 */ ADSR_STATE_SUSTAIN
} AdsrStatus;

typedef enum {
    /* 0 */ MEDIUM_RAM,
    /* 1 */ MEDIUM_UNK,
    /* 2 */ MEDIUM_CART,
    /* 3 */ MEDIUM_DISK_DRIVE,
    /* 5 */ MEDIUM_RAM_UNLOADED = 5
} SampleMedium;

typedef enum {
    /* 0 */ CODEC_ADPCM, // 16 2-byte samples (32 bytes) compressed into 4-bit samples (8 bytes) + 1 header byte
    /* 1 */ CODEC_S8, // 16 2-byte samples (32 bytes) compressed into 8-bit samples (16 bytes)
    /* 2 */ CODEC_S16_INMEMORY,
    /* 3 */ CODEC_SMALL_ADPCM, // 16 2-byte samples (32 bytes) compressed into 2-bit samples (4 bytes) + 1 header byte
    /* 4 */ CODEC_REVERB,
    /* 5 */ CODEC_S16,
    /* 6 */ CODEC_UNK6,
    /* 7 */ CODEC_UNK7 // processed as uncompressed samples
} SampleCodec;

typedef enum {
    /* 0 */ SEQUENCE_TABLE,
    /* 1 */ FONT_TABLE,
    /* 2 */ SAMPLE_TABLE
} SampleBankTableType;

typedef enum {
    /* 0 */ CACHE_LOAD_PERMANENT,
    /* 1 */ CACHE_LOAD_PERSISTENT,
    /* 2 */ CACHE_LOAD_TEMPORARY,
    /* 3 */ CACHE_LOAD_EITHER,
    /* 4 */ CACHE_LOAD_EITHER_NOSYNC
} AudioCacheLoadType;

typedef enum {
    /* 0 */ CACHE_TEMPORARY,
    /* 1 */ CACHE_PERSISTENT,
    /* 2 */ CACHE_EITHER,
    /* 3 */ CACHE_PERMANENT
} AudioCacheType;

typedef enum {
    /* 0 */ LOAD_STATUS_NOT_LOADED,
    /* 1 */ LOAD_STATUS_IN_PROGRESS,
    /* 2 */ LOAD_STATUS_COMPLETE,
    /* 3 */ LOAD_STATUS_DISCARDABLE,
    /* 4 */ LOAD_STATUS_MAYBE_DISCARDABLE,
    /* 5 */ LOAD_STATUS_PERMANENT
} AudioLoadStatus;

typedef s32 (*DmaHandler)(OSPiHandle* handle, OSIoMesg* mb, s32 direction);

struct Note;
struct NotePool;
struct SequenceChannel;
struct SequenceLayer;

// A node in a circularly linked list. Each node is either a head or an item:
// - Items can be either detached (prev = NULL), or attached to a list.
//   'value' points to something of interest.
// - List heads are always attached; if a list is empty, its head points
//   to itself. 'count' contains the size of the list.
// If the list holds notes, 'pool' points back to the pool where it lives.
// Otherwise, that member is NULL.
typedef struct AudioListItem {
    /* 0x00 */ struct AudioListItem* prev;
    /* 0x04 */ struct AudioListItem* next;
    union {
        /* 0x08 */ void* value; // either Note* or SequenceLayer*
        /* 0x08 */ s32 count;
               } u;
    /* 0x0C */ struct NotePool* pool;
} AudioListItem; // size = 0x10

typedef struct NotePool {
    /* 0x00 */ AudioListItem disabled;
    /* 0x10 */ AudioListItem decaying;
    /* 0x20 */ AudioListItem releasing;
    /* 0x30 */ AudioListItem active;
} NotePool; // size = 0x40

// Pitch sliding by up to one octave in the positive direction. Negative
// direction is "supported" by setting extent to be negative. The code
// extrapolates exponentially in the wrong direction in that case, but that
// doesn't prevent seqplayer from doing it, AFAICT.
typedef struct {
    /* 0x0 */ u8 mode; // bit 0x80 denotes something; the rest are an index 0-5
    /* 0x2 */ u16 cur;
    /* 0x4 */ u16 speed;
    /* 0x8 */ f32 extent;
} Portamento; // size = 0xC

typedef struct {
    /* 0x0 */ s16 delay;
    /* 0x2 */ s16 arg;
} EnvelopePoint; // size = 0x4

typedef struct {
    /* 0x00 */ u32 start;
    /* 0x04 */ u32 loopEnd; // numSamples into the sample where the loop ends
    /* 0x08 */ u32 count;
    /* 0x0C */ u32 sampleEnd; // total number of s16-samples in the 
    /* 0x10 */ s16 predictorState[16]; // only exists if count != 0. 8-byte aligned
} AdpcmLoop; // size = 0x30 (or 0x10)

/**
 * The procedure used to design the codeBook is based on an adaptive clustering algorithm.
 * The size of the codeBook is (8 * order * numPredictors) and is 8-byte aligned
 */
typedef struct {
    /* 0x00 */ s32 order;
    /* 0x04 */ s32 numPredictors;
    /* 0x08 */ s16 codeBook[1]; // a table of prediction coefficients that the coder selects from to optimize sound quality.
} AdpcmBook; // size >= 0x8

typedef struct {
    /* 0x00 */ u32 unk_0 : 1;
    /* 0x00 */ u32 codec : 3; // The state of compression or decompression
    /* 0x00 */ u32 medium : 2; // Medium where sample is currently stored
    /* 0x00 */ u32 unk_bit26 : 1;
    /* 0x00 */ u32 isRelocated : 1; // Has the sample header been relocated (offsets to pointers)
    /* 0x01 */ u32 size : 24; // Size of the sample
    /* 0x04 */ u8* sampleAddr; // Raw sample data. Offset from the start of the sample bank or absolute address to either rom or ram
    /* 0x08 */ AdpcmLoop* loop; // Adpcm loop parameters used by the sample. Offset from the start of the sound font / pointer to ram
    /* 0x0C */ AdpcmBook* book; // Adpcm book parameters used by the sample. Offset from the start of the sound font / pointer to ram
} Sample; // size = 0x10

typedef struct {
    /* 0x00 */ Sample* sample;
    /* 0x04 */ f32 tuning; // frequency scale factor
} TunedSample; // size = 0x8

/**
 * Stores an entry of decompressed samples in a reverb ring buffer.
 * By storing the sample in a ring buffer, the time it takes to loop
 * around back to the same sample acts as a delay, leading to an echo effect.
 */
typedef struct {
    /* 0x00 */ s16 numSamplesAfterDownsampling; // never read
    /* 0x02 */ s16 numSamples; // never read
    /* 0x04 */ s16* toDownsampleLeft;
    /* 0x08 */ s16* toDownsampleRight; // data pointed to by left and right are adjacent in memory
    /* 0x0C */ s32 startPos; // start pos in ring buffer
    /* 0x10 */ s16 size; // first length in ring buffer (from startPos, at most until end)
    /* 0x12 */ s16 wrappedSize; // second length in ring buffer (from pos 0)
    /* 0x14 */ u16 loadResamplePitch;
    /* 0x16 */ u16 saveResamplePitch;
    /* 0x18 */ u16 saveResampleNumSamples;
} ReverbBufferEntry; // size = 0x1C

typedef struct {
    /* 0x000 */ u8 resampleFlags;
    /* 0x001 */ u8 useReverb;
    /* 0x002 */ u8 framesToIgnore;
    /* 0x003 */ u8 curFrame;
    /* 0x004 */ u8 downsampleRate;
    /* 0x005 */ s8 mixReverbIndex; // mix in reverb from this index. set to 0xFF to not mix any
    /* 0x006 */ u16 delayNumSamples; // number of samples between echos
    /* 0x008 */ s16 mixReverbStrength; // the gain/amount to mix in reverb from mixReverbIndex
    /* 0x00A */ s16 volume;
    /* 0x00C */ u16 decayRatio; // determines how fast reverb dissipate
    /* 0x00E */ u16 downsamplePitch;
    /* 0x010 */ s16 leakRtl;
    /* 0x012 */ s16 leakLtr;
    /* 0x014 */ u16 subDelay; // number of samples between sub echos
    /* 0x016 */ s16 subVolume; // strength of the sub echos
    /* 0x018 */ u8 resampleEffectOn;
    /* 0x019 */ s8 resampleEffectExtraSamples;
    /* 0x01A */ u16 resampleEffectLoadUnk;
    /* 0x01C */ u16 resampleEffectSaveUnk;
    /* 0x01E */ u8 delayNumSamplesAfterDownsampling;
    /* 0x020 */ s32 nextReverbBufPos;
    /* 0x024 */ s32 delayNumSamplesUnk; // May be bufSizePerChan
    /* 0x028 */ s16* leftReverbBuf;
    /* 0x02C */ s16* rightReverbBuf;
    /* 0x030 */ s16* leftLoadResampleBuf;
    /* 0x034 */ s16* rightLoadResampleBuf;
    /* 0x038 */ s16* leftSaveResampleBuf;
    /* 0x03C */ s16* rightSaveResampleBuf;
    /* 0x040 */ ReverbBufferEntry bufEntry[2][5];
    /* 0x158 */ ReverbBufferEntry subBufEntry[2][5];
    /* 0x270 */ s16* filterLeft;
    /* 0x274 */ s16* filterRight;
    /* 0x278 */ s16* filterLeftInit;
    /* 0x27C */ s16* filterRightInit;
    /* 0x280 */ s16* filterLeftState;
    /* 0x284 */ s16* filterRightState;
    /* 0x288 */ TunedSample tunedSample;
    /* 0x290 */ Sample sample;
    /* 0x2A0 */ AdpcmLoop loop;
} SynthesisReverb; // size = 0x2D0

typedef struct {
    /* 0x00 */ u8 isRelocated; // have the envelope and all samples been relocated (offsets to pointers)
    /* 0x01 */ u8 normalRangeLo;
    /* 0x02 */ u8 normalRangeHi;
    /* 0x03 */ u8 adsrDecayIndex; // index used to obtain adsr decay rate from adsrDecayTable
    /* 0x04 */ EnvelopePoint* envelope;
    /* 0x08 */ TunedSample lowPitchTunedSample;
    /* 0x10 */ TunedSample normalPitchTunedSample;
    /* 0x18 */ TunedSample highPitchTunedSample;
} Instrument; // size = 0x20

typedef struct {
    /* 0x00 */ u8 adsrDecayIndex; // index used to obtain adsr decay rate from adsrDecayTable
    /* 0x01 */ u8 pan;
    /* 0x02 */ u8 isRelocated; // have tunedSample.sample and envelope been relocated (offsets to pointers)
    /* 0x04 */ TunedSample tunedSample;
    /* 0x0C */ EnvelopePoint* envelope;
} Drum; // size = 0x10

typedef struct {
    /* 0x00 */ TunedSample tunedSample;
} SoundEffect; // size = 0x08

typedef struct {
    /* 0x00 */ u8 numInstruments;
    /* 0x01 */ u8 numDrums;
    /* 0x02 */ u8 sampleBankId1;
    /* 0x03 */ u8 sampleBankId2;
    /* 0x04 */ u16 numSfx;
    /* 0x08 */ Instrument** instruments;
    /* 0x0C */ Drum** drums;
    /* 0x10 */ SoundEffect* soundEffects;
} SoundFont; // size = 0x14

typedef struct {
    /* 0x00 */ u8* pc; // program counter
    /* 0x04 */ u8* stack[4];
    /* 0x14 */ u8 remLoopIters[4]; // remaining loop iterations
    /* 0x18 */ u8 depth;
    /* 0x19 */ s8 value;
} SeqScriptState; // size = 0x1C

// Also known as a Group, according to debug strings.
typedef struct {
    /* 0x000 */ u8 enabled : 1;
    /* 0x000 */ u8 finished : 1;
    /* 0x000 */ u8 muted : 1;
    /* 0x000 */ u8 seqDmaInProgress : 1;
    /* 0x000 */ u8 fontDmaInProgress : 1;
    /* 0x000 */ u8 recalculateVolume : 1;
    /* 0x000 */ u8 stopScript : 1;
    /* 0x000 */ u8 applyBend : 1;
    /* 0x001 */ u8 state;
    /* 0x002 */ u8 noteAllocPolicy;
    /* 0x003 */ u8 muteFlags;
    /* 0x004 */ u8 seqId;
    /* 0x005 */ u8 defaultFont;
    /* 0x006 */ u8 unk_06[1];
    /* 0x007 */ s8 playerIndex;
    /* 0x008 */ u16 tempo; // tatums per minute
    /* 0x00A */ u16 tempoAcc;
    /* 0x00C */ s16 tempoChange;
    /* 0x00E */ s16 transposition;
    /* 0x010 */ u16 delay;
    /* 0x012 */ u16 fadeTimer;
    /* 0x014 */ u16 storedFadeTimer;
    /* 0x016 */ u16 unk_16;
    /* 0x018 */ u8* seqData;
    /* 0x01C */ f32 fadeVolume;
    /* 0x020 */ f32 fadeVelocity;
    /* 0x024 */ f32 volume;
    /* 0x028 */ f32 muteVolumeScale;
    /* 0x02C */ f32 fadeVolumeScale;
    /* 0x030 */ f32 appliedFadeVolume;
    /* 0x034 */ f32 bend;
    /* 0x038 */ struct SequenceChannel* channels[16];
    /* 0x078 */ SeqScriptState scriptState;
    /* 0x094 */ u8* shortNoteVelocityTable;
    /* 0x098 */ u8* shortNoteGateTimeTable;
    /* 0x09C */ NotePool notePool;
    /* 0x0DC */ s32 skipTicks;
    /* 0x0E0 */ u32 scriptCounter;
    /* 0x0E4 */ UNK_TYPE1 unk_E4[0x74]; // unused struct members for sequence/sound font dma management, according to sm64 decomp
    /* 0x158 */ s8 seqScriptIO[8];
} SequencePlayer; // size = 0x160

typedef struct {
    /* 0x0 */ u8 decayIndex; // index used to obtain adsr decay rate from adsrDecayTable
    /* 0x1 */ u8 sustain;
    /* 0x4 */ EnvelopePoint* envelope;
} AdsrSettings; // size = 0x8

typedef struct {
    union {
        struct {
            /* 0x00 */ u8 unused : 1;
            /* 0x00 */ u8 hang : 1;
            /* 0x00 */ u8 decay : 1;
            /* 0x00 */ u8 release : 1;
            /* 0x00 */ u8 state : 4;
        } s;
        /* 0x00 */ u8 asByte;
    } action;
    /* 0x01 */ u8 envIndex;
    /* 0x02 */ s16 delay;
    /* 0x04 */ f32 sustain;
    /* 0x08 */ f32 velocity;
    /* 0x0C */ f32 fadeOutVel;
    /* 0x10 */ f32 current;
    /* 0x14 */ f32 target;
    /* 0x18 */ UNK_TYPE1 unk_18[4];
    /* 0x1C */ EnvelopePoint* envelope;
} AdsrState; // size = 0x20

typedef union {
    struct {
        /* 0x0 */ u8 unused : 2;
        /* 0x0 */ u8 type : 2;
        /* 0x0 */ u8 strongRight : 1;
        /* 0x0 */ u8 strongLeft : 1;
        /* 0x0 */ u8 strongReverbRight : 1;
        /* 0x0 */ u8 strongReverbLeft : 1;
    };
    /* 0x0 */ u8 asByte;
} StereoData; // size = 0x1

typedef struct {
    /* 0x00 */ u8 targetReverbVol;
    /* 0x01 */ u8 gain; // Increases volume by a multiplicative scaling factor. Represented as a UQ4.4 number
    /* 0x02 */ u8 pan;
    /* 0x03 */ u8 surroundEffectIndex;
    /* 0x04 */ StereoData stereoData;
    /* 0x05 */ u8 combFilterSize;
    /* 0x06 */ u16 combFilterGain;
    /* 0x08 */ f32 freqScale;
    /* 0x0C */ f32 velocity;
    /* 0x10 */ s16* filter;
    /* 0x14 */ s16* filterBuf;
} NoteAttributes; // size = 0x18

typedef struct VibratoSubStruct {
    /* 0x0 */ u16 vibratoRateStart;
    /* 0x2 */ u16 vibratoDepthStart;
    /* 0x4 */ u16 vibratoRateTarget;
    /* 0x6 */ u16 vibratoDepthTarget;
    /* 0x8 */ u16 vibratoRateChangeDelay;
    /* 0xA */ u16 vibratoDepthChangeDelay;
    /* 0xC */ u16 vibratoDelay;
} VibratoSubStruct; // size = 0xE

// Also known as a SubTrack, according to sm64 debug strings.
typedef struct SequenceChannel {
    /* 0x00 */ u8 enabled : 1;
    /* 0x00 */ u8 finished : 1;
    /* 0x00 */ u8 stopScript : 1;
    /* 0x00 */ u8 muted : 1; // sets SequenceLayer.muted
    /* 0x00 */ u8 hasInstrument : 1;
    /* 0x00 */ u8 stereoHeadsetEffects : 1;
    /* 0x00 */ u8 largeNotes : 1; // notes specify duration and velocity
    /* 0x00 */ u8 unused : 1;
    union {
        struct {
            /* 0x01 */ u8 freqScale : 1;
            /* 0x01 */ u8 volume : 1;
            /* 0x01 */ u8 pan : 1;
        } s;
        /* 0x01 */ u8 asByte;
    } changes;
    /* 0x02 */ u8 noteAllocPolicy;
    /* 0x03 */ u8 muteFlags;
    /* 0x04 */ u8 targetReverbVol; // or dry/wet mix
    /* 0x05 */ u8 notePriority; // 0-3
    /* 0x06 */ u8 someOtherPriority;
    /* 0x07 */ u8 fontId;
    /* 0x08 */ u8 reverbIndex;
    /* 0x09 */ u8 bookOffset;
    /* 0x0A */ u8 newPan;
    /* 0x0B */ u8 panChannelWeight;  // proportion of pan that comes from the channel (0..128)
    /* 0x0C */ u8 gain; // Increases volume by a multiplicative scaling factor. Represented as a UQ4.4 number
    /* 0x0D */ u8 velocityRandomVariance;
    /* 0x0E */ u8 gateTimeRandomVariance;
    /* 0x0F */ u8 combFilterSize;
    /* 0x10 */ u8 surroundEffectIndex;
    /* 0x11 */ u8 channelIndex;
    /* 0x12 */ VibratoSubStruct vibrato;
    /* 0x20 */ u16 delay;
    /* 0x22 */ u16 combFilterGain;
    /* 0x24 */ u16 unk_22; // Used for indexing data
    /* 0x26 */ s16 instOrWave; // either 0 (none), instrument index + 1, or
                             // 0x80..0x83 for sawtooth/triangle/sine/square waves.
    /* 0x28 */ s16 transposition;
    /* 0x2C */ f32 volumeScale;
    /* 0x30 */ f32 volume;
    /* 0x34 */ s32 pan;
    /* 0x38 */ f32 appliedVolume;
    /* 0x3C */ f32 freqScale;
    /* 0x40 */ u8 (*dynTable)[][2];
    /* 0x44 */ struct Note* noteUnused;
    /* 0x48 */ struct SequenceLayer* layerUnused;
    /* 0x4C */ Instrument* instrument;
    /* 0x50 */ SequencePlayer* seqPlayer;
    /* 0x54 */ struct SequenceLayer* layers[4];
    /* 0x64 */ SeqScriptState scriptState;
    /* 0x80 */ AdsrSettings adsr;
    /* 0x88 */ NotePool notePool;
    /* 0xC8 */ s8 seqScriptIO[8]; // bridge between sound script and audio lib, "io ports"
    /* 0xD0 */ u8* sfxState; // SfxChannelState
    /* 0xD4 */ s16* filter;
    /* 0xD8 */ StereoData stereoData;
    /* 0xDC */ s32 startSamplePos;
    /* 0xE0 */ s32 unk_E0;
} SequenceChannel; // size = 0xE4

// Might also be known as a Track, according to sm64 debug strings (?).
typedef struct SequenceLayer {
    /* 0x00 */ u8 enabled : 1;
    /* 0x00 */ u8 finished : 1;
    /* 0x00 */ u8 muted : 1;
    /* 0x00 */ u8 continuousNotes : 1; // keep the same note for consecutive notes with the same sound
    /* 0x00 */ u8 bit3 : 1; // "loaded"?
    /* 0x00 */ u8 ignoreDrumPan : 1;
    /* 0x00 */ u8 bit1 : 1; // "has initialized continuous notes"?
    /* 0x00 */ u8 notePropertiesNeedInit : 1;
    /* 0x01 */ StereoData stereoData;
    /* 0x02 */ u8 instOrWave;
    /* 0x03 */ u8 gateTime;
    /* 0x04 */ u8 semitone;
    /* 0x05 */ u8 portamentoTargetNote;
    /* 0x06 */ u8 pan; // 0..128
    /* 0x07 */ u8 notePan;
    /* 0x08 */ u8 surroundEffectIndex;
    /* 0x09 */ u8 targetReverbVol;
    union {
        struct {
            /* 0x0A */ u16 bit_0 : 1;
            /* 0x0A */ u16 bit_1 : 1;
            /* 0x0A */ u16 bit_2 : 1;
            /* 0x0A */ u16 bit_3 : 1;
            /* 0x0A */ u16 bit_4 : 1;
            /* 0x0A */ u16 bit_5 : 1;
            /* 0x0A */ u16 bit_6 : 1;
            /* 0x0A */ u16 bit_7 : 1;
            /* 0x0A */ u16 bit_8 : 1;
            /* 0x0A */ u16 bit_9 : 1;
            /* 0x0A */ u16 bit_A : 1;
            /* 0x0A */ u16 bit_B : 1;
            /* 0x0A */ u16 bit_C : 1;
            /* 0x0A */ u16 bit_D : 1;
            /* 0x0A */ u16 bit_E : 1;
            /* 0x0A */ u16 bit_F : 1;
        } s;
        /* 0x0A */ u16 asByte;
    } unk_0A;
    /* 0x0C */ VibratoSubStruct vibrato;
    /* 0x1A */ s16 delay;
    /* 0x1C */ s16 gateDelay;
    /* 0x1E */ s16 delay2;
    /* 0x20 */ u16 portamentoTime;
    /* 0x22 */ s16 transposition; // #semitones added to play commands
                                  // (seq instruction encoding only allows referring to the limited range
                                  // 0..0x3F; this makes 0x40..0x7F accessible as well)
    /* 0x24 */ s16 shortNoteDefaultDelay;
    /* 0x26 */ s16 lastDelay;
    /* 0x28 */ AdsrSettings adsr;
    /* 0x30 */ Portamento portamento;
    /* 0x3C */ struct Note* note;
    /* 0x40 */ f32 freqScale;
    /* 0x44 */ f32 bend;
    /* 0x48 */ f32 velocitySquare2;
    /* 0x4C */ f32 velocitySquare; // not sure which one of those corresponds to the sm64 original
    /* 0x50 */ f32 noteVelocity;
    /* 0x54 */ f32 noteFreqScale;
    /* 0x58 */ Instrument* instrument;
    /* 0x5C */ TunedSample* tunedSample;
    /* 0x60 */ SequenceChannel* channel; // Not SequenceChannel?
    /* 0x64 */ SeqScriptState scriptState;
    /* 0x80 */ AudioListItem listItem;
} SequenceLayer; // size = 0x90

typedef struct {
    /* 0x000 */ s16 adpcmState[16];
    /* 0x020 */ s16 finalResampleState[16];
    /* 0x040 */ s16 filterState[32];
    /* 0x080 */ s16 unusedState[16];
    /* 0x0A0 */ s16 haasEffectDelayState[32];
    /* 0x0E0 */ s16 combFilterState[128];
    /* 0x1E0 */ s16 surroundEffectState[128];
} NoteSynthesisBuffers; // size = 0x2E0

typedef struct {
    /* 0x00 */ u8 atLoopPoint : 1;
    /* 0x00 */ u8 stopLoop : 1;
    /* 0x01 */ u8 sampleDmaIndex;
    /* 0x02 */ u8 prevHaasEffectLeftDelaySize;
    /* 0x03 */ u8 prevHaasEffectRightDelaySize;
    /* 0x04 */ u8 curReverbVol;
    /* 0x05 */ u8 numParts;
    /* 0x06 */ u16 samplePosFrac; // Fractional part of the sample position
    /* 0x08 */ u16 surroundEffectGain;
    /* 0x0C */ s32 samplePosInt; // Integer part of the sample position
    /* 0x10 */ NoteSynthesisBuffers* synthesisBuffers;
    /* 0x14 */ s16 curVolLeft;
    /* 0x16 */ s16 curVolRight;
    /* 0x18 */ UNK_TYPE1 unk_14[0x6];
    /* 0x1E */ u8 combFilterNeedsInit;
    /* 0x1F */ u8 unk_1F;
    /* 0x20 */ UNK_TYPE1 unk_20[0x4];
} NoteSynthesisState; // size = 0x24

typedef struct {
    /* 0x00 */ struct VibratoSubStruct* vibSubStruct; // Something else?
    /* 0x04 */ u32 time;
    /* 0x08 */ s16* curve;
    /* 0x0C */ f32 depth;
    /* 0x10 */ f32 rate;
    /* 0x14 */ u8 active;
    /* 0x16 */ u16 rateChangeTimer;
    /* 0x18 */ u16 depthChangeTimer;
    /* 0x1A */ u16 delay;
} VibratoState; // size = 0x1C

typedef enum {
    /* 0 */ PLAYBACK_STATUS_0,
    /* 1 */ PLAYBACK_STATUS_1,
    /* 2 */ PLAYBACK_STATUS_2
} NotePlaybackStatus;

typedef struct {
    /* 0x00 */ u8 priority;
    /* 0x01 */ u8 waveId;
    /* 0x02 */ u8 harmonicIndex; // the harmonic index for the synthetic wave contained in gWaveSamples (also matches the base 2 logarithm of the harmonic order)
    /* 0x03 */ u8 fontId;
    /* 0x04 */ u8 status;
    /* 0x05 */ u8 stereoHeadsetEffects;
    /* 0x06 */ s16 adsrVolScaleUnused;
    /* 0x08 */ f32 portamentoFreqScale;
    /* 0x0C */ f32 vibratoFreqScale;
    /* 0x18 */ SequenceLayer* wantedParentLayer;
    /* 0x14 */ SequenceLayer* parentLayer;
    /* 0x10 */ SequenceLayer* prevParentLayer;
    /* 0x1C */ NoteAttributes attributes;
    /* 0x34 */ AdsrState adsr;
    /* 0x54 */ Portamento portamento;
    /* 0x60 */ VibratoState vibratoState;
    /* 0x7C */ UNK_TYPE1 pad7C[0x4];
    /* 0x80 */ u8 unk_80;
    /* 0x84 */ u32 startSamplePos;
    /* 0x88 */ UNK_TYPE1 unk_BC[0x1C]; 
} NotePlaybackState; // size = 0xA4

typedef struct {
    struct {
        /* 0x00 */ volatile u8 enabled : 1;
        /* 0x00 */ u8 needsInit : 1;
        /* 0x00 */ u8 finished : 1;
        /* 0x00 */ u8 unused : 1;
        /* 0x00 */ u8 strongRight : 1;
        /* 0x00 */ u8 strongLeft : 1;
        /* 0x00 */ u8 strongReverbRight : 1;
        /* 0x00 */ u8 strongReverbLeft : 1;
    } bitField0;
    struct {
        /* 0x01 */ u8 reverbIndex : 3;
        /* 0x01 */ u8 bookOffset : 2;
        /* 0x01 */ u8 isSyntheticWave : 1;
        /* 0x01 */ u8 hasTwoParts : 1;
        /* 0x01 */ u8 useHaasEffect : 1;
    } bitField1;
    /* 0x02 */ u8 gain; // Increases volume by a multiplicative scaling factor. Represented as a UQ4.4 number
    /* 0x03 */ u8 haasEffectLeftDelaySize;
    /* 0x04 */ u8 haasEffectRightDelaySize;
    /* 0x05 */ u8 targetReverbVol;
    /* 0x06 */ u8 harmonicIndexCurAndPrev; // bits 3..2 store curHarmonicIndex, bits 1..0 store prevHarmonicIndex
    /* 0x07 */ u8 combFilterSize;
    /* 0x08 */ u16 targetVolLeft;
    /* 0x0A */ u16 targetVolRight;
    /* 0x0C */ u16 frequencyFixedPoint;
    /* 0x0E */ u16 combFilterGain;
        union {
    /* 0x10 */ TunedSample* tunedSample;
    /* 0x10 */ s16* waveSampleAddr; // used for synthetic waves
        };
    /* 0x14 */ s16* filter;
    /* 0x18 */ UNK_TYPE1 unk_18;
    /* 0x19 */ u8 surroundEffectIndex;
    /* 0x1A */ UNK_TYPE1 unk_1A[0x6];
} NoteSampleState; // size = 0x20

typedef struct Note {
    /* 0x00 */ AudioListItem listItem;
    /* 0x10 */ NoteSynthesisState synthesisState;
    /* 0x34 */ NotePlaybackState playbackState;
    /* 0xD8 */ NoteSampleState sampleState;
} Note; // size = 0xF8

typedef struct {
    /* 0x00 */ u8 downsampleRate;
    /* 0x02 */ u16 delayNumSamples;
    /* 0x04 */ u16 decayRatio; // determines how fast reverb dissipates
    /* 0x06 */ u16 subDelay;
    /* 0x08 */ u16 subVolume;
    /* 0x0A */ u16 volume;
    /* 0x0C */ u16 leakRtl;
    /* 0x0E */ u16 leakLtr;
    /* 0x10 */ s8 mixReverbIndex;
    /* 0x12 */ u16 mixReverbStrength;
    /* 0x14 */ s16 lowPassFilterCutoffLeft;
    /* 0x16 */ s16 lowPassFilterCutoffRight;
} ReverbSettings; // size = 0x18

/**
 * The high-level audio specifications requested when initializing or resetting the audio pool.
 * Most often resets during scene transitions, but will highly depend on game play.
 */ 
typedef struct {
    /* 0x00 */ u32 samplingFreq; // Target sampling rate in Hz
    /* 0x04 */ u8 unk_04;
    /* 0x05 */ u8 numNotes;
    /* 0x06 */ u8 numSequencePlayers;
    /* 0x07 */ u8 unk_07; // unused, set to zero
    /* 0x08 */ u8 unk_08; // unused, set to zero
    /* 0x09 */ u8 numReverbs;
    /* 0x0C */ ReverbSettings* reverbSettings;
    /* 0x10 */ u16 sampleDmaBufSize1;
    /* 0x12 */ u16 sampleDmaBufSize2;
    /* 0x14 */ u16 unk_14;
    /* 0x18 */ size_t persistentSeqCacheSize;        // size of cache on audio pool to store sequences persistently
    /* 0x1C */ size_t persistentFontCacheSize;       // size of cache on audio pool to store soundFonts persistently
    /* 0x20 */ size_t persistentSampleBankCacheSize; // size of cache on audio pool to store entire sample banks persistently
    /* 0x24 */ size_t temporarySeqCacheSize;         // size of cache on audio pool to store sequences temporarily
    /* 0x28 */ size_t temporaryFontCacheSize;        // size of cache on audio pool to store soundFonts temporarily
    /* 0x2C */ size_t temporarySampleBankCacheSize;  // size of cache on audio pool to store entire sample banks temporarily
    /* 0x30 */ size_t persistentSampleCacheSize;     // size of cache on audio pool to store individual samples persistently
    /* 0x34 */ size_t temporarySampleCacheSize;      // size of cache on audio pool to store individual samples temporarily
} AudioSpec; // size = 0x38

/**
 * The audio buffer stores the fully processed digital audio before it is sent to the audio interface (AI), then to the
 * digital-analog converter (DAC), then to play on the speakers. The audio buffer is written to by the rsp after
 * processing audio commands, and the audio buffer is read by AI which sends the data to the DAC.
 * This struct parameterizes that buffer.
 */
typedef struct {
    /* 0x00 */ s16 specUnk4;
    /* 0x02 */ u16 samplingFreq; // Target sampling rate in Hz
    /* 0x04 */ u16 aiSamplingFreq; // True sampling rate set to the audio interface (AI) for the audio digital-analog converter (DAC)
    /* 0x06 */ s16 numSamplesPerFrameTarget;
    /* 0x08 */ s16 numSamplesPerFrameMax;
    /* 0x0A */ s16 numSamplesPerFrameMin;
    /* 0x0C */ s16 updatesPerFrame; // for each frame of the audio thread (default 60 fps), number of updates to process audio
    /* 0x0E */ s16 numSamplesPerUpdate;
    /* 0x10 */ s16 numSamplesPerUpdateMax;
    /* 0x12 */ s16 numSamplesPerUpdateMin;
    /* 0x14 */ s16 numSequencePlayers;
    /* 0x18 */ f32 resampleRate;
    /* 0x1C */ f32 updatesPerFrameInv; // inverse (reciprocal) of updatesPerFrame
    /* 0x20 */ f32 updatesPerFrameInvScaled; // updatesPerFrameInv scaled down by a factor of 256
    /* 0x24 */ f32 updatesPerFrameScaled; // updatesPerFrame scaled down by a factor of 4
} AudioBufferParameters; // size = 0x28

/**
 * Meta-data associated with a pool (contain withing the Audio Heap)
 */
typedef struct {
    /* 0x0 */ u8* startAddr; // start addr of the pool
    /* 0x4 */ u8* curAddr; // address of the next available memory for allocation
    /* 0x8 */ size_t size; // size of the pool
    /* 0xC */ s32 count; // number of entries allocated to the pool
} AudioAllocPool; // size = 0x10

/**
 * Audio cache entry data to store a single entry containing either a sequence, soundfont, or entire sample banks
 */
typedef struct {
    /* 0x0 */ u8* addr;
    /* 0x4 */ size_t size;
    /* 0x8 */ s16 tableType;
    /* 0xA */ s16 id;
} AudioCacheEntry; // size = 0xC

/**
 * Audio cache entry data to store a single entry containing an individual sample
 */
typedef struct {
    /* 0x00 */ s8 inUse;
    /* 0x01 */ s8 origMedium;
    /* 0x02 */ u8 sampleBankId;
    /* 0x03 */ char unk_03[0x5];
    /* 0x08 */ u8* allocatedAddr;
    /* 0x0C */ void* sampleAddr;
    /* 0x10 */ size_t size;
} SampleCacheEntry; // size = 0x14

/**
 * Audio cache entry data to store individual samples
 */
typedef struct {
    /* 0x000 */ AudioAllocPool pool;
    /* 0x010 */ SampleCacheEntry entries[128];
    /* 0xA10 */ s32 numEntries;
} AudioSampleCache; // size = 0xA14

typedef struct {
    /* 0x00 */ u32 numEntries;
    /* 0x04 */ AudioAllocPool pool;
    /* 0x14 */ AudioCacheEntry entries[16];
} AudioPersistentCache; // size = 0xD4

typedef struct {
    /* 0x00 */ u32 nextSide;
    /* 0x04 */ AudioAllocPool pool;
    /* 0x14 */ AudioCacheEntry entries[2];
} AudioTemporaryCache; // size = 0x3C

typedef struct {
    /* 0x000 */ AudioPersistentCache persistent;
    /* 0x0D4 */ AudioTemporaryCache temporary;
    /* 0x100 */ u8 unk_100[0x10];
} AudioCache; // size = 0x110

typedef struct {
    /* 0x0 */ size_t persistentCommonPoolSize;
    /* 0x4 */ size_t temporaryCommonPoolSize;
} AudioCachePoolSplit; // size = 0x8

typedef struct {
    /* 0x0 */ size_t seqCacheSize;
    /* 0x4 */ size_t fontCacheSize;
    /* 0x8 */ size_t sampleBankCacheSize;
} AudioCommonPoolSplit; // size = 0xC

typedef struct {
    /* 0x0 */ size_t miscPoolSize;
    /* 0x4 */ u32 unkSizes[2];
    /* 0xC */ size_t cachePoolSize; 
} AudioSessionPoolSplit; // size = 0x10

typedef struct {
    /* 0x00 */ u32 endAndMediumKey;
    /* 0x04 */ Sample* sample;
    /* 0x08 */ u8* ramAddr;
    /* 0x0C */ u32 encodedInfo;
    /* 0x10 */ s32 isFree;
} AudioPreloadReq; // size = 0x14

typedef struct {
    union {
    /* 0x0 */ u32 opArgs;
        struct {
            /* 0x0 */ u8 op;
            /* 0x1 */ u8 arg0;
            /* 0x2 */ u8 arg1;
            /* 0x3 */ u8 arg2;
        };
    };
    union {
        /* 0x4 */ void* data;
        /* 0x4 */ f32 asFloat;
        /* 0x4 */ s32 asInt;
        /* 0x4 */ u16 asUShort;
        /* 0x4 */ s8 asSbyte;
        /* 0x4 */ u8 asUbyte;
        /* 0x4 */ u32 asUInt;
    };
} AudioCmd; // size = 0x8

typedef struct {
    /* 0x00 */ s8 status;
    /* 0x01 */ s8 delay;
    /* 0x02 */ s8 medium;
    /* 0x04 */ u8* ramAddr;
    /* 0x08 */ uintptr_t curDevAddr;
    /* 0x0C */ u8* curRamAddr;
    /* 0x10 */ size_t bytesRemaining;
    /* 0x14 */ size_t chunkSize;
    /* 0x18 */ s32 unkMediumParam;
    /* 0x1C */ u32 retMsg;
    /* 0x20 */ OSMesgQueue* retQueue;
    /* 0x24 */ OSMesgQueue msgQueue;
    /* 0x3C */ OSMesg msg;
    /* 0x40 */ OSIoMesg ioMesg;
} AudioAsyncLoad; // size = 0x58

typedef struct {
    /* 0x00 */ u8 medium;
    /* 0x01 */ u8 seqOrFontId;
    /* 0x02 */ u16 instId;
    /* 0x04 */ s32 unkMediumParam;
    /* 0x08 */ uintptr_t curDevAddr;
    /* 0x0C */ u8* curRamAddr;
    /* 0x10 */ u8* ramAddr;
    /* 0x14 */ s32 status;
    /* 0x18 */ size_t bytesRemaining;
    /* 0x1C */ s8* isDone; // TODO: rename in OoT and sync up here. This is an external status while (s32 status) is an internal status
    /* 0x20 */ Sample sample;
    /* 0x30 */ OSMesgQueue msgqueue;
    /* 0x48 */ OSMesg msg;
    /* 0x4C */ OSIoMesg ioMesg;
} AudioSlowLoad; // size = 0x64

typedef struct {
    /* 0x00 */ uintptr_t romAddr;
    /* 0x04 */ size_t size;
    /* 0x08 */ s8 medium;
    /* 0x09 */ s8 cachePolicy;
    /* 0x0A */ s16 shortData1;
    /* 0x0C */ s16 shortData2;
    /* 0x0E */ s16 shortData3;
} AudioTableEntry; // size = 0x10

typedef struct {
    /* 0x00 */ s16 numEntries;
    /* 0x02 */ s16 unkMediumParam;
    /* 0x04 */ uintptr_t romAddr;
    /* 0x08 */ char pad[0x8];
    /* 0x10 */ AudioTableEntry entries[1]; // (dynamic size)
} AudioTable; // size >= 0x20

typedef struct {
    /* 0x00 */ OSTask task;
    /* 0x40 */ OSMesgQueue* taskQueue;
    /* 0x44 */ void* unk_44; // probably a message that gets unused.
    /* 0x48 */ char unk_48[0x8];
} AudioTask; // size = 0x50

typedef struct {
    /* 0x00 */ u8* ramAddr;
    /* 0x04 */ uintptr_t devAddr;
    /* 0x08 */ u16 sizeUnused;
    /* 0x0A */ u16 size;
    /* 0x0C */ u8 unused;
    /* 0x0D */ u8 reuseIndex; // position in sSampleDmaReuseQueue1/2, if ttl == 0
    /* 0x0E */ u8 ttl;        // Time To Live: duration after which the DMA can be discarded
} SampleDma; // size = 0x10

typedef struct {
    /* 0x0000 */ char unk_0000;
    /* 0x0001 */ s8 numSynthesisReverbs;
    /* 0x0002 */ u16 unk_2; // reads from audio spec unk_14, never used, always set to 0x7FFF
    /* 0x0004 */ u16 unk_4;
    /* 0x0006 */ char unk_0006[0xA];
    /* 0x0010 */ s16* adpcmCodeBook;
    /* 0x0014 */ NoteSampleState* sampleStateList;
    /* 0x0018 */ SynthesisReverb synthesisReverbs[4];
    /* 0x0B58 */ char unk_0B58[0x30];
    /* 0x0B88 */ Sample* usedSamples[128];
    /* 0x0D88 */ AudioPreloadReq preloadSampleStack[128];
    /* 0x1788 */ s32 numUsedSamples;
    /* 0x178C */ s32 preloadSampleStackTop;
    /* 0x1790 */ AudioAsyncLoad asyncLoads[0x10];
    /* 0x1D10 */ OSMesgQueue asyncLoadUnkMediumQueue;
    /* 0x1D28 */ char unk_1D08[0x40];
    /* 0x1D68 */ AudioAsyncLoad* curUnkMediumLoad;
    /* 0x1D6C */ u32 slowLoadPos;
    /* 0x1D70 */ AudioSlowLoad slowLoads[2];
    /* 0x1E38 */ OSPiHandle* cartHandle;
    /* 0x1E2C */ OSPiHandle* driveHandle;
    /* 0x1E40 */ OSMesgQueue externalLoadQueue;
    /* 0x1E58 */ OSMesg externalLoadMesgBuf[0x10];
    /* 0x1E98 */ OSMesgQueue preloadSampleQueue;
    /* 0x1EB0 */ OSMesg preloadSampleMesgBuf[0x10];
    /* 0x1EF0 */ OSMesgQueue curAudioFrameDmaQueue;
    /* 0x1F08 */ OSMesg currAudioFrameDmaMesgBuf[0x40];
    /* 0x2008 */ OSIoMesg currAudioFrameDmaIoMesgBuf[0x40];
    /* 0x2608 */ OSMesgQueue syncDmaQueue;
    /* 0x2620 */ OSMesg syncDmaMesg;
    /* 0x2624 */ OSIoMesg syncDmaIoMesg;
    /* 0x263C */ SampleDma* sampleDmas;
    /* 0x2640 */ u32 sampleDmaCount;
    /* 0x2644 */ u32 sampleDmaListSize1;
    /* 0x2648 */ s32 unused2648;
    /* 0x264C */ u8 sampleDmaReuseQueue1[0x100]; // read pos <= write pos, wrapping mod 256
    /* 0x274C */ u8 sampleDmaReuseQueue2[0x100];
    /* 0x284C */ u8 sampleDmaReuseQueue1RdPos; // Read position for dma 1
    /* 0x284D */ u8 sampleDmaReuseQueue2RdPos; // Read position for dma 2
    /* 0x284E */ u8 sampleDmaReuseQueue1WrPos; // Write position for dma 1
    /* 0x284F */ u8 sampleDmaReuseQueue2WrPos; // Write position for dma 2
    /* 0x2850 */ AudioTable* sequenceTable;
    /* 0x2854 */ AudioTable* soundFontTable;
    /* 0x2858 */ AudioTable* sampleBankTable;
    /* 0x285C */ char unk_285C[0x4];
    /* 0x2860 */ u8* sequenceFontTable;
    /* 0x2864 */ u16 numSequences;
    /* 0x2868 */ SoundFont* soundFontList;
    /* 0x286C */ AudioBufferParameters audioBufferParameters;
    /* 0x2994 */ f32 unk_2870;
    /* 0x2898 */ s32 sampleDmaBufSize1;
    /* 0x289C */ s32 sampleDmaBufSize2;
    /* 0x28A0 */ char unk_287C[0x10];
    /* 0x28B0 */ s32 sampleDmaBufSize;
    /* 0x28B4 */ s32 maxAudioCmds;
    /* 0x28B8 */ s32 numNotes;
    /* 0x2898 */ s16 maxTempo;
    /* 0x28BE */ s8 soundMode;
    /* 0x28C0 */ s32 totalTaskCount; // The total number of times the top-level function on the audio thread is run since the last audio reset
    /* 0x28C4 */ s32 curAudioFrameDmaCount;
    /* 0x28C8 */ s32 rspTaskIndex;
    /* 0x28CC */ s32 curAiBufferIndex;
    /* 0x28AC */ Acmd* abiCmdBufs[2]; // Pointer to audio heap where the audio binary interface command lists are stored. Two lists that alternative every frame
    /* 0x28B4 */ Acmd* curAbiCmdBuf;
    /* 0x28DC */ AudioTask* curTask;
    /* 0x28C0 */ AudioTask rspTask[2];
    /* 0x2980 */ f32 unk_2960;
    /* 0x2984*/ s32 refreshRate;
    /* 0x2988 */ s16* aiBuffers[3]; // Pointers to the audio buffer allocated on the initPool contained in the audio heap. Stores fully processed digital audio before transferring to the audio interface (AI)
    /* 0x2994 */ s16 numSamplesPerFrame[3]; // Number of samples to transfer to the audio interface buffer
    /* 0x299C */ u32 audioRandom;
    /* 0x29A0 */ s32 audioErrorFlags;
    /* 0x29A4 */ volatile u32 resetTimer;
    /* 0x29A8 */ u32 (*customSeqFunctions[4])(s8 value, SequenceChannel* channel);
    /* 0x29B8 */ s8 unk_29B8;
    /* 0x29BC */ s32 numAbiCmdsMax; // sMaxAbiCmdCnt
    /* 0x29C0 */ AudioAllocPool sessionPool; // A sub-pool to main pool, contains all sub-pools and data that changes every audio reset
    /* 0x29D0 */ AudioAllocPool externalPool; // pool allocated on an external device. Never used in game
    /* 0x29E0 */ AudioAllocPool initPool; // A sub-pool to the main pool, contains all sub-pools and data that persists every audio reset
    /* 0x29F0 */ AudioAllocPool miscPool; // A sub-pool to the session pool, 
    /* 0x2A00 */ char unk_29D0[0x20]; // probably two unused pools
    /* 0x2A20 */ AudioAllocPool cachePool; // The common pool for all cache entries
    /* 0x2A30 */ AudioAllocPool persistentCommonPool; // A sub-pool to the cache pool, contains all caches for data stored persistently
    /* 0x2A40 */ AudioAllocPool temporaryCommonPool; // A sub-pool to the cache pool, contains all caches for data stored temporarily
    /* 0x2A50 */ AudioCache seqCache; // Cache to store sequences
    /* 0x2B60 */ AudioCache fontCache; // Cache to store soundFonts
    /* 0x2C70 */ AudioCache sampleBankCache; // Cache for loading entire sample banks
    /* 0x2D80 */ AudioAllocPool permanentPool; // Pool to stores audio data that is always loaded in. Primarily used for sfxs
    /* 0x2D90 */ AudioCacheEntry permanentEntries[32]; // indificual entries to the permanent pool
    /* 0x3690 */ AudioSampleCache persistentSampleCache; // Stores individual samples persistently
    /* 0x40A4 */ AudioSampleCache temporarySampleCache; // Stores individual samples temporarily
    /* 0x4338 */ AudioSessionPoolSplit sessionPoolSplit; // splits session pool into the cache pool and misc pool
    /* 0x4348 */ AudioCachePoolSplit cachePoolSplit; // splits cache pool into the persistent & temporary common pools
    /* 0x4350 */ AudioCommonPoolSplit persistentCommonPoolSplit; // splits persistent common pool into caches for sequences, soundFonts, sample banks
    /* 0x435C */ AudioCommonPoolSplit temporaryCommonPoolSplit; // splits temporary common pool into caches for sequences, soundFonts, sample banks
    /* 0x4368 */ u8 sampleFontLoadStatus[0x30];
    /* 0x4398 */ u8 fontLoadStatus[0x30];
    /* 0x43C8 */ u8 seqLoadStatus[0x80];
    /* 0x4448 */ volatile u8 resetStatus;
    /* 0x4449 */ u8 specId;
    /* 0x444C */ s32 audioResetFadeOutFramesLeft;
    /* 0x4450 */ f32* adsrDecayTable; // A table on the audio heap that stores decay rates used for ADSR
    /* 0x4454 */ u8* audioHeap;
    /* 0x4458 */ size_t audioHeapSize;
    /* 0x445C */ Note* notes;
    /* 0x4460 */ SequencePlayer seqPlayers[5];
    /* 0x4B40 */ SequenceLayer sequenceLayers[80];
    /* 0x7840 */ SequenceChannel sequenceChannelNone;
    /* 0x7924 */ s32 sampleStateOffset; // Start of the list of sample states for this update. Resets after each audio frame.
    /* 0x7928 */ AudioListItem layerFreeList;
    /* 0x7938 */ NotePool noteFreeLists;
    /* 0x7978 */ u8 threadCmdWritePos;
    /* 0x7979 */ u8 threadCmdReadPos;
    /* 0x797A */ u8 threadCmdQueueFinished;
    /* 0x797C */ u16 threadCmdChannelMask[5]; // bit-packed for 16 channels. When processing an audio thread channel command on all channels, only process channels with their bit set.
    /* 0x7988 */ OSMesgQueue* audioResetQueueP;
    /* 0x798C */ OSMesgQueue* taskStartQueueP;
    /* 0x7990 */ OSMesgQueue* threadCmdProcQueueP;
    /* 0x7994 */ OSMesgQueue taskStartQueue;
    /* 0x79AC */ OSMesgQueue threadCmdProcQueue;
    /* 0x79C4 */ OSMesgQueue audioResetQueue;
    /* 0x79DC */ OSMesg taskStartMsgs[1];
    /* 0x79E0 */ OSMesg audioResetMesgs[1];
    /* 0x79E4 */ OSMesg threadCmdProcMsgBuf[4];
    /* 0x79F4 */ AudioCmd threadCmdBuf[0x100]; // Audio commands used to transfer audio requests from the graph thread to the audio thread
    /* 0x81F4 */ UNK_TYPE1 unk_81F4[4];
} AudioContext; // size = 0x81F8

typedef struct {
    /* 0x00 */ u8 targetReverbVol;
    /* 0x01 */ u8 gain; // Increases volume by a multiplicative scaling factor. Represented as a UQ4.4 number
    /* 0x02 */ u8 pan;
    /* 0x03 */ u8 surroundEffectIndex;
    /* 0x04 */ StereoData stereoData;
    /* 0x08 */ f32 frequency;
    /* 0x0C */ f32 velocity;
    /* 0x10 */ char unk_0C[0x4];
    /* 0x14 */ s16* filter;
    /* 0x18 */ u8 combFilterSize;
    /* 0x1A */ u16 combFilterGain;
} NoteSubAttributes; // size = 0x1A

typedef struct {
    /* 0x0 */ size_t heapSize; // total number of bytes allocated to the audio heap. Must be <= the size of `gAudioHeap` (ideally about the same size)
    /* 0x4 */ size_t initPoolSize; // The entire audio heap is split into two pools. 
    /* 0x8 */ size_t permanentPoolSize;
} AudioHeapInitSizes; // size = 0xC

typedef struct {
    /* 0x00 */ f32 unk_00;
    /* 0x04 */ f32 unk_04;
    /* 0x08 */ f32 unk_08;
    /* 0x0C */ f32 unk_10;
    /* 0x10 */ f32 unk_14;
    /* 0x14 */ f32 unk_18;
    /* 0x18 */ u16 unk_0C;
    /* 0x1A */ u16 unk_1C;
} unk_50_s; // size = 0x1C

typedef struct {
    /* 0x000 */ unk_50_s unk_50[0x10];
    /* 0x1C0 */ f32 volCur;
    /* 0x1C4 */ f32 volTarget;
    /* 0x1C8 */ f32 unk_08;
    /* 0x1CC */ u32 unk_14;
    /* 0x1D0 */ f32 unk_1C;
    /* 0x1D4 */ f32 unk_20;
    /* 0x1D8 */ f32 unk_24;
    /* 0x1DC */ u32 unk_2C[8];
    /* 0x1FC */ u32 unk_25C;
    /* 0x200 */ u16 unk_0C;
    /* 0x202 */ u16 unk_18;
    /* 0x204 */ u16 unk_28;
    /* 0x206 */ u16 unk_250;
    /* 0x208 */ u16 unk_252;
    /* 0x20A */ u16 seqId;
    /* 0x20C */ u16 unk_256;
    /* 0x20E */ u16 unk_258;
    /* 0x210 */ u8 unk_260;
    /* 0x211 */ u8 unk_261[0x1];
    /* 0x212 */ u8 volScales[0x4];
    /* 0x216 */ u8 volFadeTimer;
    /* 0x217 */ u8 fadeVolUpdate;
    /* 0x218 */ u8 unk_4C;
    /* 0x219 */ u8 unk_4D;
    /* 0x21A */ u8 unk_4E;
    /* 0x21B */ u8 unk_21B;
} ActiveSequence; // size = 0x21C

typedef struct {
    /* 0x0 */ u8 unk_0;
    /* 0x1 */ u8 unk_1; // importance?
} Struct_8016E320; // size = 0x02

typedef enum {
    /* 0 */ BANK_PLAYER,
    /* 1 */ BANK_ITEM,
    /* 2 */ BANK_ENV,
    /* 3 */ BANK_ENEMY,
    /* 4 */ BANK_SYSTEM,
    /* 5 */ BANK_OCARINA,
    /* 6 */ BANK_VOICE
} SfxBankType;

typedef enum {
    /* 0 */ SFX_STATE_EMPTY,
    /* 1 */ SFX_STATE_QUEUED,
    /* 2 */ SFX_STATE_READY,
    /* 3 */ SFX_STATE_PLAYING_REFRESH,
    /* 4 */ SFX_STATE_PLAYING,
    /* 5 */ SFX_STATE_PLAYING_ONE_FRAME
} SfxState;

typedef struct {
    /* 0x00 */ f32* posX;
    /* 0x04 */ f32* posY;
    /* 0x08 */ f32* posZ;
    /* 0x0C */ f32* freqScale;
    /* 0x10 */ f32* volume;
    /* 0x14 */ s8* reverbAdd;
    /* 0x18 */ f32 dist;
    /* 0x1C */ u32 priority; // lower is more prioritized
    /* 0x20 */ u16 sfxParams;
    /* 0x22 */ u16 sfxId;
    /* 0x25 */ u8 sfxFlags;
    /* 0x24 */ u8 sfxImportance;
    /* 0x26 */ u8 state; // uses SfxState enum
    /* 0x27 */ u8 freshness;
    /* 0x28 */ u8 prev;
    /* 0x29 */ u8 next;
    /* 0x2A */ u8 channelIndex;
    /* 0x2B */ u8 randFreq;
    /* 0x2C */ u8 token;
} SfxBankEntry; // size = 0x30

/*
 * SfxId:
 *
 * & 03FF    0000000111111111    index
 * & 0400    0000010000000000    unused flag
 * & 0800    0000100000000000    SFX_FLAG
 * & 0C00    0000110000000000    Flag Mask
 * & F000    1111000000000000    observed in audio code
 */

#define SFX_BANK_SHIFT(sfxId)   (((sfxId) >> 12) & 0xFF)

#define SFX_BANK_MASK(sfxId)    ((sfxId) & 0xF000)

#define SFX_INDEX(sfxId)    ((sfxId) & 0x3FF)
#define SFX_BANK(sfxId)     SFX_BANK_SHIFT(SFX_BANK_MASK(sfxId))

typedef struct {
    /* 0x0 */ u32 priority; // lower is more prioritized
    /* 0x4 */ u8 entryIndex;
} ActiveSfx; // size = 0x08

// SfxParams bit-packing

// Slows the decay of volume with distance (a 3-bit number ranging from 0-7)
#define SFX_PARAM_DIST_RANGE_SHIFT 0
#define SFX_PARAM_DIST_RANGE_MASK_UPPER (4 << SFX_PARAM_DIST_RANGE_SHIFT)
#define SFX_PARAM_DIST_RANGE_MASK (7 << SFX_PARAM_DIST_RANGE_SHIFT)

// Lower SEQ_PLAYER_BGM_MAIN and SEQ_PLAYER_BGM_SUB while the sfx is playing
#define SFX_FLAG_LOWER_VOLUME_BGM (1 << 3)

// Sfx priority is not raised with distance (making it more likely to be ejected)
#define SFX_FLAG_PRIORITY_NO_DIST (1 << 4)

// If a new sfx is requested at both the same position with the same importance,
// Block that new sfx from replacing the current sfx
// Note: Only 1 sfx can be played at a specific position at once
#define SFX_FLAG_BLOCK_EQUAL_IMPORTANCE (1 << 5)

// Applies increasingly random offsets to frequency (a 2-bit number ranging from 0-3)
#define SFX_PARAM_RAND_FREQ_RAISE_SHIFT 6
#define SFX_PARAM_RAND_FREQ_RAISE_MASK (3 << SFX_PARAM_RAND_FREQ_RAISE_SHIFT)

// Sets a flag to ioPort 5
#define SFX_FLAG_8 (1 << 8)

// Use lowpass filter on surround sound
#define SFX_FLAG_SURROUND_LOWPASS_FILTER (1 << 9)

// Unused remnant of OoT
#define SFX_FLAG_BEHIND_SCREEN_Z_INDEX_SHIFT 10
#define SFX_FLAG_BEHIND_SCREEN_Z_INDEX (1 << SFX_FLAG_BEHIND_SCREEN_Z_INDEX_SHIFT)

// Randomly scale base frequency each frame through mutiplicative offset
#define SFX_PARAM_RAND_FREQ_SCALE (1 << 11)

// Sfx reverb is not raised with distance
#define SFX_FLAG_REVERB_NO_DIST (1 << 12)

// Sfx volume is not lowered with distance
#define SFX_FLAG_VOLUME_NO_DIST (1 << 13)

// SFX_FLAG_VIBRATO 
// Randomly lower base frequency each frame through additive offset
#define SFX_PARAM_RAND_FREQ_LOWER (1 << 14)

// Sfx frequency is not raised with distance
#define SFX_FLAG_FREQ_NO_DIST (1 << 15)

// Force the sfx to reset from the beginning when requested again
#define SFX_FLAG2_FORCE_RESET (1 << 0)

// Unused
#define SFX_FLAG2_UNUSED2 (1 << 2)
#define SFX_FLAG2_UNUSED4 (1 << 4)

// Do not use highpass filter on surround sound
#define SFX_FLAG2_SURROUND_NO_HIGHPASS_FILTER (1 << 5)

// Unused
#define SFX_FLAG2_UNUSED6 (1 << 6)

// Apply a low-pass filter with a lowPassCutoff of 4
#define SFX_FLAG2_APPLY_LOWPASS_FILTER (1 << 7)

typedef struct {
    /* 0x0 */ u8 importance;
    /* 0x1 */ u8 flags;
    /* 0x2 */ u16 params;
} SfxParams; // size = 0x4

typedef void (*AudioCustomUpdateFunction)(void);
typedef u32 (*AudioCustomSeqFunction)(s8 value, SequenceChannel* channel);
typedef s32 (*AudioCustomReverbFunction)(Sample*, s32, s8, s32);
typedef Acmd* (*AudioCustomSynthFunction)(Acmd*, s32, s32);

#endif
