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mm/include/z64audio.h

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#ifndef Z64AUDIO_H
#define Z64AUDIO_H
#include "PR/ultratypes.h"
#include "PR/os_voice.h"
#include "audiothread_cmd.h"
#include "libc/stddef.h"
#include "unk.h"
#include "audiothread_cmd.h"
#include "audio/effects.h"
#include "audio/heap.h"
#include "audio/load.h"
#include "audio/soundfont.h"
#include "sequence.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 {
/* 0 */ AUDIO_HEAP_RESET_STATE_NONE,
/* 1 */ AUDIO_HEAP_RESET_STATE_RESETTING,
/* 2 */ AUDIO_HEAP_RESET_STATE_RESETTING_ALT // Never set to
} AudioHeapResetState;
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))
// 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;
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
/**
* 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* 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 SequencePlayer {
/* 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 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
// 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 useVibrato : 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 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
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;
/* 0x4 */ void* asPtr;
};
} AudioCmd; // size = 0x8
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 {
/* 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 {
/* 0x00 */ f32 volCur;
/* 0x04 */ f32 volTarget;
/* 0x08 */ f32 volStep;
/* 0x0C */ f32 freqScaleCur;
/* 0x10 */ f32 freqScaleTarget;
/* 0x14 */ f32 freqScaleStep;
/* 0x18 */ u16 volTimer;
/* 0x1A */ u16 freqScaleTimer;
} ActiveSequenceChannelData; // size = 0x1C
typedef struct {
/* 0x000 */ ActiveSequenceChannelData channelData[SEQ_NUM_CHANNELS];
/* 0x1C0 */ f32 volCur;
/* 0x1C4 */ f32 volTarget;
/* 0x1C8 */ f32 volStep;
/* 0x1CC */ u32 tempoCmd;
/* 0x1D0 */ f32 tempoCur;
/* 0x1D4 */ f32 tempoTarget;
/* 0x1D8 */ f32 tempoStep;
/* 0x1DC */ u32 setupCmd[8]; // setup commands
/* 0x1FC */ u32 startAsyncSeqCmd; // temporarily stores the seqCmd used in SEQCMD_PLAY_SEQUENCE, to be called again once the font is reloaded in
/* 0x200 */ u16 volTimer;
/* 0x202 */ u16 tempoOriginal;
/* 0x204 */ u16 tempoTimer;
/* 0x206 */ u16 freqScaleChannelFlags;
/* 0x208 */ u16 volChannelFlags;
/* 0x20A */ u16 seqId;
/* 0x20C */ u16 prevSeqId; // last seqId played on a player
/* 0x20E */ u16 channelPortMask;
/* 0x210 */ u8 isWaitingForFonts;
/* 0x211 */ u8 fontId;
/* 0x212 */ u8 volScales[VOL_SCALE_INDEX_MAX];
/* 0x216 */ u8 volFadeTimer;
/* 0x217 */ u8 fadeVolUpdate;
/* 0x218 */ u8 setupCmdTimer;
/* 0x219 */ u8 setupCmdNum; // number of setup commands
/* 0x21A */ u8 setupFadeTimer;
/* 0x21B */ u8 isSeqPlayerInit;
} ActiveSequence; // size = 0x21C
typedef struct {
/* 0x0 */ u8 seqId;
/* 0x1 */ u8 priority;
} SeqRequest; // 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 void* (*AudioCustomReverbFunction)(Sample*, s32, s8, s32);
typedef Acmd* (*AudioCustomSynthFunction)(Acmd*, s32, s32);
#endif
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