#ifndef Z64_AUDIO_H #define Z64_AUDIO_H #define AUDIO_MK_CMD(b0,b1,b2,b3) ((((b0) & 0xFF) << 0x18) | (((b1) & 0xFF) << 0x10) | (((b2) & 0xFF) << 0x8) | (((b3) & 0xFF) << 0)) #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 */ SEQPLAYER_STATE_0, /* 1 */ SEQPLAYER_STATE_1, // Fading in /* 2 */ SEQPLAYER_STATE_2 // Fading out } SeqPlayerState; #define MAX_CHANNELS_PER_BANK 3 #define MUTE_FLAGS_3 (1 << 3) // prevent further noteSubEus from playing #define MUTE_FLAGS_4 (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 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 unk_0C; /* 0x00E */ s16 transposition; /* 0x010 */ u16 delay; /* 0x012 */ u16 fadeTimer; /* 0x014 */ u16 fadeTimerUnkEu; /* 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 */ char unk_E4[0x74]; // unused struct members for sequence/sound font dma management, according to sm64 decomp /* 0x158 */ s8 soundScriptIO[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 unk_0b80 : 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 */ char 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 vibratoExtentStart; /* 0x4 */ u16 vibratoRateTarget; /* 0x6 */ u16 vibratoExtentTarget; /* 0x8 */ u16 vibratoRateChangeDelay; /* 0xA */ u16 vibratoExtentChangeDelay; /* 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 stopSomething2 : 1; // sets SequenceLayer.stopSomething /* 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 unk_11; /* 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 soundScriptIO[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 stopSomething : 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 extent; /* 0x10 */ f32 rate; /* 0x14 */ u8 active; /* 0x16 */ u16 rateChangeTimer; /* 0x18 */ u16 extentChangeTimer; /* 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 tempoInternalToExternal; /* 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 aiBufNumSamples[3]; // Number of samples to transfer to the audio interface buffer /* 0x299C */ u32 audioRandom; /* 0x29A0 */ s32 audioErrorFlags; /* 0x29A4 */ volatile u32 resetTimer; /* 0x29A8 */ u32 (*unk_29A8[4])(s8 value, SequenceChannel* channel); /* 0x29B8 */ s8 unk_29B8; /* 0x29BC */ s32 unk_29BC; // 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 audioResetSpecIdToLoad; /* 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 cmdWritePos; /* 0x7979 */ u8 cmdReadPos; /* 0x797A */ u8 cmdQueueFinished; /* 0x797C */ u16 activeChannelsFlags[5]; // bit-packed for 16 channels. Only channels with bit turned on will be processed /* 0x7988 */ OSMesgQueue* audioResetQueueP; /* 0x798C */ OSMesgQueue* taskStartQueueP; /* 0x7990 */ OSMesgQueue* cmdProcQueueP; /* 0x7994 */ OSMesgQueue taskStartQueue; /* 0x79AC */ OSMesgQueue cmdProcQueue; /* 0x79C4 */ OSMesgQueue audioResetQueue; /* 0x79DC */ OSMesg taskStartMsgs[1]; /* 0x79E0 */ OSMesg audioResetMesgs[1]; /* 0x79E4 */ OSMesg cmdProcMsgs[4]; /* 0x79F4 */ AudioCmd cmdBuf[0x100]; // Audio commands used to transfer audio requests from the graph thread to the audio thread /* 0x81F4 */ char 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 #endif