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5ed5f196d9
* [headers] audio functions to z64audio.h and z64ocarina.h * bss
1219 lines
49 KiB
C
1219 lines
49 KiB
C
#ifndef Z64_AUDIO_H
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#define Z64_AUDIO_H
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#include "ultra64.h"
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#include "sequence.h"
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#include "z64math.h"
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struct GfxPrint;
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typedef void (*AudioCustomUpdateFunction)(void);
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#define REFRESH_RATE_PAL 50
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#define REFRESH_RATE_MPAL 60
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#define REFRESH_RATE_NTSC 60
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// Small deviation parameters used in estimating the max tempo
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// It is unclear why these vary by region, and aren't all just 1
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#define REFRESH_RATE_DEVIATION_PAL 1.001521f
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#define REFRESH_RATE_DEVIATION_MPAL 0.99276f
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#define REFRESH_RATE_DEVIATION_NTSC 1.00278f
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#define AUDIO_MK_CMD(b0,b1,b2,b3) ((((b0) & 0xFF) << 0x18) | (((b1) & 0xFF) << 0x10) | (((b2) & 0xFF) << 0x8) | (((b3) & 0xFF) << 0))
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#define NO_LAYER ((SequenceLayer*)(-1))
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// Also known as "Pulses Per Quarter Note" or "Tatums Per Beat"
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#define SEQTICKS_PER_BEAT 48
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#define IS_SEQUENCE_CHANNEL_VALID(ptr) ((u32)(ptr) != (u32)&gAudioCtx.sequenceChannelNone)
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#define SEQ_NUM_CHANNELS 16
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#define SEQ_IO_VAL_NONE -1
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#define MAX_CHANNELS_PER_BANK 3
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#define MUTE_BEHAVIOR_3 (1 << 3) // prevent further noteSubEus from playing
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#define MUTE_BEHAVIOR_4 (1 << 4) // stop something in seqLayer scripts
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#define MUTE_BEHAVIOR_SOFTEN (1 << 5) // lower volume, by default to half
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#define MUTE_BEHAVIOR_STOP_NOTES (1 << 6) // prevent further notes from playing
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#define MUTE_BEHAVIOR_STOP_SCRIPT (1 << 7) // stop processing sequence/channel scripts
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#define ADSR_DISABLE 0
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#define ADSR_HANG -1
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#define ADSR_GOTO -2
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#define ADSR_RESTART -3
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// size of a single sample point
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#define SAMPLE_SIZE sizeof(s16)
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// Samples are processed in groups of 16 called a "frame"
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#define SAMPLES_PER_FRAME ADPCMFSIZE
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// The length of one left/right channel is 13 frames
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#define DMEM_1CH_SIZE (13 * SAMPLES_PER_FRAME * SAMPLE_SIZE)
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// Both left and right channels
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#define DMEM_2CH_SIZE (2 * DMEM_1CH_SIZE)
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#define AIBUF_LEN (88 * SAMPLES_PER_FRAME) // number of samples
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#define AIBUF_SIZE (AIBUF_LEN * SAMPLE_SIZE) // number of bytes
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// Filter sizes
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#define FILTER_SIZE (8 * SAMPLE_SIZE)
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#define FILTER_BUF_PART1 (8 * SAMPLE_SIZE)
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#define FILTER_BUF_PART2 (8 * SAMPLE_SIZE)
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// Must be the same amount of samples as copied by aDuplicate() (audio microcode)
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#define WAVE_SAMPLE_COUNT 64
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#define AUDIO_RELOCATED_ADDRESS_START K0BASE
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typedef enum SoundMode {
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/* 0 */ SOUNDMODE_STEREO,
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/* 1 */ SOUNDMODE_HEADSET,
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/* 2 */ SOUNDMODE_SURROUND,
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/* 3 */ SOUNDMODE_MONO
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} SoundMode;
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typedef enum AdsrStatus {
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/* 0 */ ADSR_STATE_DISABLED,
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/* 1 */ ADSR_STATE_INITIAL,
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/* 2 */ ADSR_STATE_START_LOOP,
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/* 3 */ ADSR_STATE_LOOP,
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/* 4 */ ADSR_STATE_FADE,
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/* 5 */ ADSR_STATE_HANG,
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/* 6 */ ADSR_STATE_DECAY,
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/* 7 */ ADSR_STATE_RELEASE,
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/* 8 */ ADSR_STATE_SUSTAIN
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} AdsrStatus;
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typedef enum SampleMedium {
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/* 0 */ MEDIUM_RAM,
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/* 1 */ MEDIUM_UNK,
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/* 2 */ MEDIUM_CART,
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/* 3 */ MEDIUM_DISK_DRIVE
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} SampleMedium;
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typedef enum SampleCodec {
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/* 0 */ CODEC_ADPCM, // 16 2-byte samples (32 bytes) compressed into 4-bit samples (8 bytes) + 1 header byte
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/* 1 */ CODEC_S8, // 16 2-byte samples (32 bytes) compressed into 8-bit samples (16 bytes)
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/* 2 */ CODEC_S16_INMEMORY,
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/* 3 */ CODEC_SMALL_ADPCM, // 16 2-byte samples (32 bytes) compressed into 2-bit samples (4 bytes) + 1 header byte
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/* 4 */ CODEC_REVERB,
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/* 5 */ CODEC_S16
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} SampleCodec;
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typedef enum SampleBankTableType {
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/* 0 */ SEQUENCE_TABLE,
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/* 1 */ FONT_TABLE,
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/* 2 */ SAMPLE_TABLE
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} SampleBankTableType;
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typedef enum AudioCacheType {
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/* 0 */ CACHE_TEMPORARY,
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/* 1 */ CACHE_PERSISTENT,
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/* 2 */ CACHE_EITHER,
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/* 3 */ CACHE_PERMANENT
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} AudioCacheType;
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typedef enum AudioCacheLoadType {
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/* 0 */ CACHE_LOAD_PERMANENT,
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/* 1 */ CACHE_LOAD_PERSISTENT,
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/* 2 */ CACHE_LOAD_TEMPORARY,
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/* 3 */ CACHE_LOAD_EITHER,
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/* 4 */ CACHE_LOAD_EITHER_NOSYNC
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} AudioCacheLoadType;
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typedef enum AudioLoadStatus {
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/* 0 */ LOAD_STATUS_NOT_LOADED, // the entry data is not loaded
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/* 1 */ LOAD_STATUS_IN_PROGRESS, // the entry data is being loaded asynchronously
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/* 2 */ LOAD_STATUS_COMPLETE, // the entry data is loaded, it may be discarded if not stored persistently, and either no longer in use, or the memory is needed for something else
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/* 3 */ LOAD_STATUS_DISCARDABLE, // the entry data is loaded, and can be discarded
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/* 4 */ LOAD_STATUS_MAYBE_DISCARDABLE, // only for font table entries, like COMPLETE but prefer discarding it over a COMPLETE entry
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/* 5 */ LOAD_STATUS_PERMANENTLY_LOADED // the entry data is loaded in the permanent pool, it won't be discarded
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} AudioLoadStatus;
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typedef s32 (*DmaHandler)(OSPiHandle* handle, OSIoMesg* mb, s32 direction);
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struct Note;
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struct NotePool;
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struct SequenceChannel;
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struct SequenceLayer;
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typedef struct AudioListItem {
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// A node in a circularly linked list. Each node is either a head or an item:
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// - Items can be either detached (prev = NULL), or attached to a list.
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// 'value' points to something of interest.
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// - List heads are always attached; if a list is empty, its head points
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// to itself. 'count' contains the size of the list.
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// If the list holds notes, 'pool' points back to the pool where it lives.
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// Otherwise, that member is NULL.
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/* 0x00 */ struct AudioListItem* prev;
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/* 0x04 */ struct AudioListItem* next;
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/* 0x08 */ union {
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void* value; // either Note* or SequenceLayer*
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s32 count;
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} u;
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/* 0x0C */ struct NotePool* pool;
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} AudioListItem; // size = 0x10
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typedef struct NotePool {
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/* 0x00 */ AudioListItem disabled;
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/* 0x10 */ AudioListItem decaying;
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/* 0x20 */ AudioListItem releasing;
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/* 0x30 */ AudioListItem active;
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} NotePool; // size = 0x40
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// Pitch sliding by up to one octave in the positive direction. Negative
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// direction is "supported" by setting extent to be negative. The code
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// exterpolates exponentially in the wrong direction in that case, but that
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// doesn't prevent seqplayer from doing it, AFAICT.
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typedef struct Portamento {
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/* 0x00 */ u8 mode; // bit 0x80 denotes something; the rest are an index 0-5
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/* 0x02 */ u16 cur;
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/* 0x04 */ u16 speed;
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/* 0x08 */ f32 extent;
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} Portamento; // size = 0xC
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typedef struct EnvelopePoint {
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/* 0x0 */ s16 delay;
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/* 0x2 */ s16 arg;
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} EnvelopePoint; // size = 0x4
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typedef struct AdpcmLoopHeader {
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/* 0x00 */ u32 start;
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/* 0x04 */ u32 end; // s16 sample position where the loop ends
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/* 0x08 */ u32 count; // The number of times the loop is played before the sound completes. Setting count to -1 indicates that the loop should play indefinitely.
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/* 0x0C */ char unk_0C[0x4];
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} AdpcmLoopHeader; // size = 0x10
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typedef struct AdpcmLoop {
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/* 0x00 */ AdpcmLoopHeader header;
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/* 0x10 */ s16 predictorState[16]; // only exists if count != 0. 8-byte aligned
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} AdpcmLoop; // size = 0x30 (or 0x10)
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typedef struct AdpcmBookHeader {
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/* 0x00 */ s32 order;
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/* 0x04 */ s32 numPredictors;
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} AdpcmBookHeader; // size = 0x8
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/**
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* The procedure used to design the codeBook is based on an adaptive clustering algorithm.
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* The size of the codeBook is (8 * order * numPredictors) and is 8-byte aligned
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*/
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typedef s16 AdpcmBookData[];
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typedef struct AdpcmBook {
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/* 0x00 */ AdpcmBookHeader header;
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/* 0x08 */ AdpcmBookData book; // size 8 * order * numPredictors. 8-byte aligned
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} AdpcmBook; // size >= 0x8
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typedef struct Sample {
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/* 0x00 */ u32 codec : 4; // The state of compression or decompression
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/* 0x00 */ u32 medium : 2; // Medium where sample is currently stored
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/* 0x00 */ u32 unk_bit26 : 1;
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/* 0x00 */ u32 isRelocated : 1; // Has the sample header been relocated (offsets to pointers)
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/* 0x01 */ u32 size : 24; // Size of the sample
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/* 0x04 */ u8* sampleAddr; // Raw sample data. Offset from the start of the sample bank or absolute address to either rom or ram
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/* 0x08 */ AdpcmLoop* loop; // Adpcm loop parameters used by the sample. Offset from the start of the sound font / pointer to ram
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/* 0x0C */ AdpcmBook* book; // Adpcm book parameters used by the sample. Offset from the start of the sound font / pointer to ram
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} Sample; // size = 0x10
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typedef struct TunedSample {
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/* 0x00 */ Sample* sample;
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/* 0x04 */ f32 tuning; // frequency scale factor
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} TunedSample; // size = 0x8
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typedef struct Instrument {
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/* 0x00 */ u8 isRelocated; // have the envelope and all samples been relocated (offsets to pointers)
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/* 0x01 */ u8 normalRangeLo;
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/* 0x02 */ u8 normalRangeHi;
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/* 0x03 */ u8 adsrDecayIndex; // index used to obtain adsr decay rate from adsrDecayTable
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/* 0x04 */ EnvelopePoint* envelope;
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/* 0x08 */ TunedSample lowPitchTunedSample;
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/* 0x10 */ TunedSample normalPitchTunedSample;
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/* 0x18 */ TunedSample highPitchTunedSample;
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} Instrument; // size = 0x20
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typedef struct Drum {
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/* 0x00 */ u8 adsrDecayIndex; // index used to obtain adsr decay rate from adsrDecayTable
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/* 0x01 */ u8 pan;
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/* 0x02 */ u8 isRelocated; // have tunedSample.sample and envelope been relocated (offsets to pointers)
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/* 0x04 */ TunedSample tunedSample;
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/* 0x0C */ EnvelopePoint* envelope;
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} Drum; // size = 0x10
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typedef struct SoundEffect {
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/* 0x00 */ TunedSample tunedSample;
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} SoundEffect; // size = 0x08
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/**
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* Stores parsed information from soundfont data
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*/
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typedef struct SoundFont {
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/* 0x00 */ u8 numInstruments;
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/* 0x01 */ u8 numDrums;
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/* 0x02 */ u8 sampleBankId1;
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/* 0x03 */ u8 sampleBankId2;
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/* 0x04 */ u16 numSfx;
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/* 0x08 */ Instrument** instruments;
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/* 0x0C */ Drum** drums;
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/* 0x10 */ SoundEffect* soundEffects;
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} SoundFont; // size = 0x14
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typedef struct ReverbRingBufferItem {
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/* 0x00 */ s16 numSamplesAfterDownsampling; // never read
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/* 0x02 */ s16 chunkLen; // never read
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/* 0x04 */ s16* toDownsampleLeft;
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/* 0x08 */ s16* toDownsampleRight; // data pointed to by left and right are adjacent in memory
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/* 0x0C */ s32 startPos; // start pos in ring buffer
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/* 0x10 */ s16 lengthA; // first length in ring buffer (from startPos, at most until end)
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/* 0x12 */ s16 lengthB; // second length in ring buffer (from pos 0)
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/* 0x14 */ u16 unk_14;
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/* 0x16 */ u16 unk_16;
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/* 0x18 */ u16 unk_18;
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} ReverbRingBufferItem; // size = 0x1C
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typedef struct SynthesisReverb {
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/* 0x000 */ u8 resampleFlags;
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/* 0x001 */ u8 useReverb;
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/* 0x002 */ u8 framesToIgnore;
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/* 0x003 */ u8 curFrame;
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/* 0x004 */ u8 downsampleRate;
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/* 0x005 */ s8 unk_05;
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/* 0x006 */ u16 windowSize;
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/* 0x008 */ s16 unk_08;
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/* 0x00A */ s16 volume;
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/* 0x00C */ u16 decayRatio; // determines how much reverb persists
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/* 0x00E */ u16 unk_0E;
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/* 0x010 */ s16 leakRtl;
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/* 0x012 */ s16 leakLtr;
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/* 0x014 */ u16 unk_14;
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/* 0x016 */ s16 unk_16;
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/* 0x018 */ u8 unk_18;
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/* 0x019 */ u8 unk_19;
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/* 0x01A */ u8 unk_1A;
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/* 0x01B */ u8 unk_1B;
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/* 0x01C */ s32 nextRingBufPos;
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/* 0x020 */ s32 unk_20;
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/* 0x024 */ s32 bufSizePerChan;
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/* 0x028 */ s16* leftRingBuf;
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/* 0x02C */ s16* rightRingBuf;
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/* 0x030 */ void* unk_30;
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/* 0x034 */ void* unk_34;
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/* 0x038 */ void* unk_38;
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/* 0x03C */ void* unk_3C;
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/* 0x040 */ ReverbRingBufferItem items[2][5];
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/* 0x158 */ ReverbRingBufferItem items2[2][5];
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/* 0x270 */ s16* filterLeft;
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/* 0x274 */ s16* filterRight;
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/* 0x278 */ s16* filterLeftState;
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/* 0x27C */ s16* filterRightState;
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/* 0x280 */ TunedSample tunedSample;
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/* 0x288 */ Sample sample;
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/* 0x298 */ AdpcmLoop loop;
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} SynthesisReverb; // size = 0x2C8
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typedef struct SeqScriptState {
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/* 0x00 */ u8* pc; // program counter
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/* 0x04 */ u8* stack[4];
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/* 0x14 */ u8 remLoopIters[4]; // remaining loop iterations
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/* 0x18 */ u8 depth;
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/* 0x19 */ s8 value;
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} SeqScriptState; // size = 0x1C
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// Also known as a Group, according to debug strings.
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typedef struct SequencePlayer {
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/* 0x000 */ u8 enabled : 1;
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/* 0x000 */ u8 finished : 1;
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/* 0x000 */ u8 muted : 1;
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/* 0x000 */ u8 seqDmaInProgress : 1;
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/* 0x000 */ u8 fontDmaInProgress : 1;
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/* 0x000 */ u8 recalculateVolume : 1;
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/* 0x000 */ u8 stopScript : 1;
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/* 0x000 */ u8 applyBend : 1;
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/* 0x001 */ u8 state;
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/* 0x002 */ u8 noteAllocPolicy;
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/* 0x003 */ u8 muteBehavior;
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/* 0x004 */ u8 seqId;
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/* 0x005 */ u8 defaultFont;
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/* 0x006 */ u8 unk_06[1];
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/* 0x007 */ s8 playerIdx;
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/* 0x008 */ u16 tempo; // seqTicks per minute
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/* 0x00A */ u16 tempoAcc; // tempo accumulation, used in a discretized algorithm to apply tempo.
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/* 0x00C */ u16 tempoChange; // Used to adjust the tempo without altering the base tempo.
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/* 0x00E */ s16 transposition;
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/* 0x010 */ u16 delay;
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/* 0x012 */ u16 fadeTimer; // in ticks
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/* 0x014 */ u16 fadeTimerUnkEu;
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/* 0x018 */ u8* seqData;
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/* 0x01C */ f32 fadeVolume;
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/* 0x020 */ f32 fadeVelocity;
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/* 0x024 */ f32 volume;
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/* 0x028 */ f32 muteVolumeScale;
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/* 0x02C */ f32 fadeVolumeScale;
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/* 0x030 */ f32 appliedFadeVolume;
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/* 0x034 */ f32 bend;
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/* 0x038 */ struct SequenceChannel* channels[16];
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/* 0x078 */ SeqScriptState scriptState;
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/* 0x094 */ u8* shortNoteVelocityTable;
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/* 0x098 */ u8* shortNoteGateTimeTable;
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/* 0x09C */ NotePool notePool;
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/* 0x0DC */ s32 skipTicks;
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/* 0x0E0 */ u32 scriptCounter;
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/* 0x0E4 */ char unk_E4[0x74]; // unused struct members for sequence/sound font dma management, according to sm64 decomp
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/* 0x158 */ s8 seqScriptIO[8];
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} SequencePlayer; // size = 0x160
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typedef struct AdsrSettings {
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/* 0x0 */ u8 decayIndex; // index used to obtain adsr decay rate from adsrDecayTable
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/* 0x1 */ u8 sustain;
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/* 0x4 */ EnvelopePoint* envelope;
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} AdsrSettings; // size = 0x8
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typedef struct AdsrState {
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/* 0x00 */ union {
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struct A {
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/* 0x00 */ u8 unused : 1;
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/* 0x00 */ u8 hang : 1;
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/* 0x00 */ u8 decay : 1;
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/* 0x00 */ u8 release : 1;
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/* 0x00 */ u8 state : 4;
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} s;
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/* 0x00 */ u8 asByte;
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} action;
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/* 0x01 */ u8 envIndex;
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/* 0x02 */ s16 delay;
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/* 0x04 */ f32 sustain;
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/* 0x08 */ f32 velocity;
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/* 0x0C */ f32 fadeOutVel;
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/* 0x10 */ f32 current;
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/* 0x14 */ f32 target;
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/* 0x18 */ char unk_18[4];
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/* 0x1C */ EnvelopePoint* envelope;
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} AdsrState; // size = 0x20
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typedef struct StereoData {
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/* 0x00 */ u8 unused : 2;
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/* 0x00 */ u8 bit2 : 2;
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/* 0x00 */ u8 strongRight : 1;
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/* 0x00 */ u8 strongLeft : 1;
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/* 0x00 */ u8 stereoHeadsetEffects : 1;
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/* 0x00 */ u8 usesHeadsetPanEffects : 1;
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} StereoData; // size = 0x1
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typedef union Stereo {
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/* 0x00 */ StereoData s;
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/* 0x00 */ u8 asByte;
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} Stereo; // size = 0x1
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typedef struct NoteAttributes {
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/* 0x00 */ u8 reverb;
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/* 0x01 */ u8 gain; // Increases volume by a multiplicative scaling factor. Represented as a UQ4.4 number
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/* 0x02 */ u8 pan;
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/* 0x03 */ Stereo stereo;
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/* 0x04 */ u8 combFilterSize;
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/* 0x06 */ u16 combFilterGain;
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/* 0x08 */ f32 freqScale;
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/* 0x0C */ f32 velocity;
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/* 0x10 */ s16* filter;
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/* 0x14 */ s16 filterBuf[8];
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} NoteAttributes; // size = 0x24
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// Also known as a SubTrack, according to sm64 debug strings.
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typedef struct SequenceChannel {
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/* 0x00 */ u8 enabled : 1;
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/* 0x00 */ u8 finished : 1;
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/* 0x00 */ u8 stopScript : 1;
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/* 0x00 */ u8 muted : 1; // sets SequenceLayer.muted
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/* 0x00 */ u8 hasInstrument : 1;
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/* 0x00 */ u8 stereoHeadsetEffects : 1;
|
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/* 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 muteBehavior;
|
|
/* 0x04 */ u8 targetReverbVol;
|
|
/* 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 */ u16 vibratoRateStart;
|
|
/* 0x12 */ u16 vibratoDepthStart;
|
|
/* 0x14 */ u16 vibratoRateTarget;
|
|
/* 0x16 */ u16 vibratoDepthTarget;
|
|
/* 0x18 */ u16 vibratoRateChangeDelay;
|
|
/* 0x1A */ u16 vibratoDepthChangeDelay;
|
|
/* 0x1C */ u16 vibratoDelay;
|
|
/* 0x1E */ u16 delay;
|
|
/* 0x20 */ u16 combFilterGain;
|
|
/* 0x22 */ u16 unk_22;
|
|
/* 0x24 */ s16 instOrWave; // either 0 (none), instrument index + 1, or
|
|
// 0x80..0x83 for sawtooth/triangle/sine/square waves.
|
|
/* 0x26 */ s16 transposition;
|
|
/* 0x28 */ f32 volumeScale;
|
|
/* 0x2C */ f32 volume;
|
|
/* 0x30 */ s32 pan;
|
|
/* 0x34 */ f32 appliedVolume;
|
|
/* 0x38 */ f32 freqScale;
|
|
/* 0x3C */ u8 (*dynTable)[][2];
|
|
/* 0x40 */ struct Note* noteUnused;
|
|
/* 0x44 */ struct SequenceLayer* layerUnused;
|
|
/* 0x48 */ Instrument* instrument;
|
|
/* 0x4C */ SequencePlayer* seqPlayer;
|
|
/* 0x50 */ struct SequenceLayer* layers[4];
|
|
/* 0x60 */ SeqScriptState scriptState;
|
|
/* 0x7C */ AdsrSettings adsr;
|
|
/* 0x84 */ NotePool notePool;
|
|
/* 0xC4 */ s8 seqScriptIO[8]; // bridge between .seq script and audio lib, "io ports"
|
|
/* 0xCC */ s16* filter;
|
|
/* 0xD0 */ Stereo stereo;
|
|
} SequenceChannel; // size = 0xD4
|
|
|
|
// 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 */ Stereo stereo;
|
|
/* 0x02 */ u8 instOrWave;
|
|
/* 0x03 */ u8 gateTime;
|
|
/* 0x04 */ u8 semitone;
|
|
/* 0x05 */ u8 portamentoTargetNote;
|
|
/* 0x06 */ u8 pan; // 0..128
|
|
/* 0x07 */ u8 notePan;
|
|
/* 0x08 */ s16 delay;
|
|
/* 0x0A */ s16 gateDelay;
|
|
/* 0x0C */ s16 delay2;
|
|
/* 0x0E */ u16 portamentoTime;
|
|
/* 0x10 */ 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)
|
|
/* 0x12 */ s16 shortNoteDefaultDelay;
|
|
/* 0x14 */ s16 lastDelay;
|
|
/* 0x18 */ AdsrSettings adsr;
|
|
/* 0x20 */ Portamento portamento;
|
|
/* 0x2C */ struct Note* note;
|
|
/* 0x30 */ f32 freqScale;
|
|
/* 0x34 */ f32 bend;
|
|
/* 0x38 */ f32 velocitySquare2;
|
|
/* 0x3C */ f32 velocitySquare; // not sure which one of those corresponds to the sm64 original
|
|
/* 0x40 */ f32 noteVelocity;
|
|
/* 0x44 */ f32 noteFreqScale;
|
|
/* 0x48 */ Instrument* instrument;
|
|
/* 0x4C */ TunedSample* tunedSample;
|
|
/* 0x50 */ SequenceChannel* channel;
|
|
/* 0x54 */ SeqScriptState scriptState;
|
|
/* 0x70 */ AudioListItem listItem;
|
|
} SequenceLayer; // size = 0x80
|
|
|
|
typedef struct NoteSynthesisBuffers {
|
|
/* 0x000 */ s16 adpcmdecState[16];
|
|
/* 0x020 */ s16 finalResampleState[16];
|
|
/* 0x040 */ s16 mixEnvelopeState[32];
|
|
/* 0x080 */ s16 unusedState[16];
|
|
/* 0x0A0 */ s16 haasEffectDelayState[32];
|
|
/* 0x0E0 */ s16 combFilterState[128];
|
|
} NoteSynthesisBuffers; // size = 0x1E0
|
|
|
|
typedef struct NoteSynthesisState {
|
|
/* 0x00 */ u8 restart;
|
|
/* 0x01 */ u8 sampleDmaIndex;
|
|
/* 0x02 */ u8 prevHaasEffectLeftDelaySize;
|
|
/* 0x03 */ u8 prevHaasEffectRightDelaySize;
|
|
/* 0x04 */ u8 reverbVol;
|
|
/* 0x05 */ u8 numParts;
|
|
/* 0x06 */ u16 samplePosFrac;
|
|
/* 0x08 */ s32 samplePosInt;
|
|
/* 0x0C */ NoteSynthesisBuffers* synthesisBuffers;
|
|
/* 0x10 */ s16 curVolLeft;
|
|
/* 0x12 */ s16 curVolRight;
|
|
/* 0x14 */ char unk_14[0x6];
|
|
/* 0x1A */ u8 combFilterNeedsInit;
|
|
/* 0x1C */ char unk_1C[0x4];
|
|
} NoteSynthesisState; // size = 0x20
|
|
|
|
typedef struct VibratoState {
|
|
/* 0x00 */ struct SequenceChannel* channel;
|
|
/* 0x04 */ u32 time;
|
|
/* 0x08 */ s16* curve; // sineWave
|
|
/* 0x0C */ f32 depth;
|
|
/* 0x10 */ f32 rate;
|
|
/* 0x14 */ u8 active;
|
|
/* 0x16 */ u16 rateChangeTimer;
|
|
/* 0x18 */ u16 depthChangeTimer;
|
|
/* 0x1A */ u16 delay;
|
|
} VibratoState; // size = 0x1C
|
|
|
|
typedef struct NotePlaybackState {
|
|
/* 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 unk_04;
|
|
/* 0x05 */ u8 stereoHeadsetEffects;
|
|
/* 0x06 */ s16 adsrVolScaleUnused;
|
|
/* 0x08 */ f32 portamentoFreqScale;
|
|
/* 0x0C */ f32 vibratoFreqScale;
|
|
/* 0x10 */ SequenceLayer* prevParentLayer;
|
|
/* 0x14 */ SequenceLayer* parentLayer;
|
|
/* 0x18 */ SequenceLayer* wantedParentLayer;
|
|
/* 0x1C */ NoteAttributes attributes;
|
|
/* 0x40 */ AdsrState adsr;
|
|
/* 0x60 */ Portamento portamento;
|
|
/* 0x6C */ VibratoState vibratoState;
|
|
} NotePlaybackState; // size = 0x88
|
|
|
|
typedef struct NoteSubEu {
|
|
struct {
|
|
/* 0x00 */ volatile u8 enabled : 1;
|
|
/* 0x00 */ u8 needsInit : 1;
|
|
/* 0x00 */ u8 finished : 1; // ?
|
|
/* 0x00 */ u8 unused : 1;
|
|
/* 0x00 */ u8 stereoStrongRight : 1;
|
|
/* 0x00 */ u8 stereoStrongLeft : 1;
|
|
/* 0x00 */ u8 stereoHeadsetEffects : 1;
|
|
/* 0x00 */ u8 usesHeadsetPanEffects : 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 reverbVol;
|
|
/* 0x06 */ u8 harmonicIndexCurAndPrev; // bits 3..2 store curHarmonicIndex, bits 1..0 store prevHarmonicIndex
|
|
/* 0x07 */ u8 combFilterSize;
|
|
/* 0x08 */ u16 targetVolLeft;
|
|
/* 0x0A */ u16 targetVolRight;
|
|
/* 0x0C */ u16 resamplingRateFixedPoint;
|
|
/* 0x0E */ u16 combFilterGain;
|
|
/* 0x10 */ union {
|
|
TunedSample* tunedSample;
|
|
s16* waveSampleAddr; // used for synthetic waves
|
|
};
|
|
/* 0x14 */ s16* filter;
|
|
/* 0x18 */ char pad_18[0x8];
|
|
} NoteSubEu; // size = 0x20
|
|
|
|
typedef struct Note {
|
|
/* 0x00 */ AudioListItem listItem;
|
|
/* 0x10 */ NoteSynthesisState synthesisState;
|
|
/* 0x30 */ NotePlaybackState playbackState;
|
|
/* 0xB8 */ char unk_B8[0x4];
|
|
/* 0xBC */ u32 startSamplePos; // initial position/index to start processing s16 samples
|
|
/* 0xC0 */ NoteSubEu noteSubEu;
|
|
} Note; // size = 0xE0
|
|
|
|
typedef struct ReverbSettings {
|
|
/* 0x00 */ u8 downsampleRate;
|
|
/* 0x02 */ u16 windowSize;
|
|
/* 0x04 */ u16 decayRatio; // determines how much reverb persists
|
|
/* 0x06 */ u16 unk_6;
|
|
/* 0x08 */ u16 unk_8;
|
|
/* 0x0A */ u16 volume;
|
|
/* 0x0C */ u16 leakRtl;
|
|
/* 0x0E */ u16 leakLtr;
|
|
/* 0x10 */ s8 unk_10;
|
|
/* 0x12 */ u16 unk_12;
|
|
/* 0x14 */ s16 lowPassFilterCutoffLeft;
|
|
/* 0x16 */ s16 lowPassFilterCutoffRight;
|
|
} ReverbSettings; // size = 0x18
|
|
|
|
/**
|
|
* The high-level audio specifications requested when initializing or resetting the audio heap.
|
|
* The audio heap can be reset on various occasions, including on most scene transitions.
|
|
*/
|
|
typedef struct AudioSpec {
|
|
/* 0x00 */ u32 samplingFrequency; // 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; // size of buffers in the audio misc pool to store small snippets of individual samples. Stored short-lived.
|
|
/* 0x12 */ u16 sampleDmaBufSize2; // size of buffers in the audio misc pool to store small snippets of individual samples. Stored long-lived.
|
|
/* 0x14 */ u16 unk_14;
|
|
/* 0x18 */ u32 persistentSeqCacheSize; // size of cache on audio pool to store sequences persistently
|
|
/* 0x1C */ u32 persistentFontCacheSize; // size of cache on audio pool to store soundFonts persistently
|
|
/* 0x20 */ u32 persistentSampleBankCacheSize; // size of cache on audio pool to store entire sample banks persistently
|
|
/* 0x24 */ u32 temporarySeqCacheSize; // size of cache on audio pool to store sequences temporarily
|
|
/* 0x28 */ u32 temporaryFontCacheSize; // size of cache on audio pool to store soundFonts temporarily
|
|
/* 0x2C */ u32 temporarySampleBankCacheSize; // size of cache on audio pool to store entire sample banks temporarily
|
|
/* 0x30 */ s32 persistentSampleCacheSize; // size of cache in the audio misc pool to store individual samples persistently
|
|
/* 0x34 */ s32 temporarySampleCacheSize; // size of cache in the audio misc 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. This struct parameterizes that buffer.
|
|
*/
|
|
typedef struct AudioBufferParameters {
|
|
/* 0x00 */ s16 specUnk4;
|
|
/* 0x02 */ u16 samplingFrequency; // Target sampling rate in Hz
|
|
/* 0x04 */ u16 aiSamplingFrequency; // True sampling rate of the audio interface (AI), see `osAiSetFrequency`
|
|
/* 0x06 */ s16 samplesPerFrameTarget;
|
|
/* 0x08 */ s16 maxAiBufferLength;
|
|
/* 0x0A */ s16 minAiBufferLength;
|
|
/* 0x0C */ s16 ticksPerUpdate; // for each audio thread update, number of ticks to process audio
|
|
/* 0x0E */ s16 samplesPerTick;
|
|
/* 0x10 */ s16 samplesPerTickMax;
|
|
/* 0x12 */ s16 samplesPerTickMin;
|
|
/* 0x14 */ s16 numSequencePlayers;
|
|
/* 0x18 */ f32 resampleRate;
|
|
/* 0x1C */ f32 ticksPerUpdateInv; // inverse (reciprocal) of ticksPerUpdate
|
|
/* 0x20 */ f32 ticksPerUpdateInvScaled; // ticksPerUpdateInv scaled down by a factor of 256
|
|
/* 0x24 */ f32 ticksPerUpdateScaled; // ticksPerUpdate scaled down by a factor of 4
|
|
} AudioBufferParameters; // size = 0x28
|
|
|
|
/**
|
|
* Meta-data associated with a pool (contained within the Audio Heap)
|
|
*/
|
|
typedef struct AudioAllocPool {
|
|
/* 0x0 */ u8* startRamAddr; // start addr of the pool
|
|
/* 0x4 */ u8* curRamAddr; // address of the next available memory for allocation
|
|
/* 0x8 */ s32 size; // size of the pool
|
|
/* 0xC */ s32 numEntries; // 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 AudioCacheEntry {
|
|
/* 0x0 */ u8* ramAddr;
|
|
/* 0x4 */ u32 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 SampleCacheEntry {
|
|
/* 0x00 */ s8 inUse;
|
|
/* 0x01 */ s8 origMedium;
|
|
/* 0x02 */ s8 sampleBankId;
|
|
/* 0x03 */ char unk_03[0x5];
|
|
/* 0x08 */ u8* allocatedAddr;
|
|
/* 0x0C */ void* sampleAddr;
|
|
/* 0x10 */ u32 size;
|
|
} SampleCacheEntry; // size = 0x14
|
|
|
|
/**
|
|
* Audio cache entry data to store individual samples
|
|
*/
|
|
typedef struct AudioSampleCache {
|
|
/* 0x000 */ AudioAllocPool pool;
|
|
/* 0x010 */ SampleCacheEntry entries[32];
|
|
/* 0x290 */ s32 numEntries;
|
|
} AudioSampleCache; // size = 0x294
|
|
|
|
typedef struct AudioPersistentCache {
|
|
/* 0x00*/ u32 numEntries;
|
|
/* 0x04*/ AudioAllocPool pool;
|
|
/* 0x14*/ AudioCacheEntry entries[16];
|
|
} AudioPersistentCache; // size = 0xD4
|
|
|
|
typedef struct AudioTemporaryCache {
|
|
/* 0x00*/ u32 nextSide;
|
|
/* 0x04*/ AudioAllocPool pool;
|
|
/* 0x14*/ AudioCacheEntry entries[2];
|
|
} AudioTemporaryCache; // size = 0x3C
|
|
|
|
typedef struct AudioCache {
|
|
/* 0x000*/ AudioPersistentCache persistent;
|
|
/* 0x0D4*/ AudioTemporaryCache temporary;
|
|
/* 0x100*/ u8 unk_100[0x10];
|
|
} AudioCache; // size = 0x110
|
|
|
|
typedef struct AudioCachePoolSplit {
|
|
/* 0x0 */ u32 persistentCommonPoolSize;
|
|
/* 0x4 */ u32 temporaryCommonPoolSize;
|
|
} AudioCachePoolSplit; // size = 0x8
|
|
|
|
typedef struct AudioCommonPoolSplit {
|
|
/* 0x0 */ u32 seqCacheSize;
|
|
/* 0x4 */ u32 fontCacheSize;
|
|
/* 0x8 */ u32 sampleBankCacheSize;
|
|
} AudioCommonPoolSplit; // size = 0xC
|
|
|
|
typedef struct AudioSessionPoolSplit {
|
|
/* 0x0 */ u32 miscPoolSize;
|
|
/* 0x4 */ u32 unkSizes[2];
|
|
/* 0xC */ u32 cachePoolSize;
|
|
} AudioSessionPoolSplit; // size = 0x10
|
|
|
|
typedef struct AudioPreloadReq {
|
|
/* 0x00 */ u32 endAndMediumKey;
|
|
/* 0x04 */ Sample* sample;
|
|
/* 0x08 */ u8* ramAddr;
|
|
/* 0x0C */ u32 encodedInfo;
|
|
/* 0x10 */ s32 isFree;
|
|
} AudioPreloadReq; // size = 0x14
|
|
|
|
/**
|
|
* Audio commands used to transfer audio requests from the graph thread to the audio thread
|
|
*/
|
|
typedef struct AudioCmd {
|
|
/* 0x0 */ union{
|
|
u32 opArgs;
|
|
struct {
|
|
u8 op;
|
|
u8 arg0;
|
|
u8 arg1;
|
|
u8 arg2;
|
|
};
|
|
};
|
|
/* 0x4 */ union {
|
|
void* data;
|
|
f32 asFloat;
|
|
s32 asInt;
|
|
u16 asUShort;
|
|
s8 asSbyte;
|
|
u8 asUbyte;
|
|
u32 asUInt;
|
|
};
|
|
} AudioCmd; // size = 0x8
|
|
|
|
typedef struct AudioAsyncLoad {
|
|
/* 0x00 */ s8 status;
|
|
/* 0x01 */ s8 delay;
|
|
/* 0x02 */ s8 medium;
|
|
/* 0x04 */ u8* ramAddr;
|
|
/* 0x08 */ u32 curDevAddr;
|
|
/* 0x0C */ u8* curRamAddr;
|
|
/* 0x10 */ u32 bytesRemaining;
|
|
/* 0x14 */ u32 chunkSize;
|
|
/* 0x18 */ s32 unkMediumParam;
|
|
/* 0x1C */ u32 retMsg;
|
|
/* 0x20 */ OSMesgQueue* retQueue;
|
|
/* 0x24 */ OSMesgQueue msgQueue;
|
|
/* 0x3C */ OSMesg msg;
|
|
/* 0x40 */ OSIoMesg ioMesg;
|
|
} AudioAsyncLoad; // size = 0x58
|
|
|
|
typedef struct AudioSlowLoad {
|
|
/* 0x00 */ u8 medium;
|
|
/* 0x01 */ u8 seqOrFontId;
|
|
/* 0x02 */ u16 instId;
|
|
/* 0x04 */ s32 unkMediumParam;
|
|
/* 0x08 */ u32 curDevAddr;
|
|
/* 0x0C */ u8* curRamAddr;
|
|
/* 0x10 */ u8* ramAddr;
|
|
/* 0x14 */ s32 state;
|
|
/* 0x18 */ s32 bytesRemaining;
|
|
/* 0x1C */ s8* status; // write-only
|
|
/* 0x20 */ Sample sample;
|
|
/* 0x30 */ OSMesgQueue msgQueue;
|
|
/* 0x48 */ OSMesg msg;
|
|
/* 0x4C */ OSIoMesg ioMesg;
|
|
} AudioSlowLoad; // size = 0x64
|
|
|
|
typedef struct AudioTableHeader {
|
|
/* 0x00 */ s16 numEntries;
|
|
/* 0x02 */ s16 unkMediumParam;
|
|
/* 0x04 */ uintptr_t romAddr;
|
|
/* 0x08 */ char pad[0x8];
|
|
} AudioTableHeader; // size = 0x10
|
|
|
|
typedef struct AudioTableEntry {
|
|
/* 0x00 */ u32 romAddr;
|
|
/* 0x04 */ u32 size;
|
|
/* 0x08 */ s8 medium;
|
|
/* 0x09 */ s8 cachePolicy;
|
|
/* 0x0A */ s16 shortData1;
|
|
/* 0x0C */ s16 shortData2;
|
|
/* 0x0E */ s16 shortData3;
|
|
} AudioTableEntry; // size = 0x10
|
|
|
|
typedef struct AudioTable {
|
|
/* 0x00 */ AudioTableHeader header;
|
|
/* 0x10 */ AudioTableEntry entries[1]; // (dynamic size)
|
|
} AudioTable; // size >= 0x20
|
|
|
|
typedef struct SampleDma {
|
|
/* 0x00 */ u8* ramAddr;
|
|
/* 0x04 */ u32 devAddr;
|
|
/* 0x08 */ u16 sizeUnused;
|
|
/* 0x0A */ u16 size;
|
|
/* 0x0C */ u8 unused;
|
|
/* 0x0D */ u8 reuseIndex; // position in sSampleDmaReuseQueue1/2, if ttl == 0
|
|
/* 0x0E */ u8 ttl; // duration after which the DMA can be discarded
|
|
} SampleDma; // size = 0x10
|
|
|
|
typedef struct AudioTask {
|
|
/* 0x00 */ OSTask task;
|
|
/* 0x40 */ OSMesgQueue* msgQueue;
|
|
/* 0x44 */ void* unk_44; // probably a message that gets unused.
|
|
/* 0x48 */ char unk_48[0x8];
|
|
} AudioTask; // size = 0x50
|
|
|
|
typedef struct AudioContext {
|
|
/* 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[0x0A];
|
|
/* 0x0010 */ s16* curLoadedBook;
|
|
/* 0x0014 */ NoteSubEu* noteSubsEu;
|
|
/* 0x0018 */ SynthesisReverb synthesisReverbs[4];
|
|
/* 0x0B38 */ char unk_0B38[0x30];
|
|
/* 0x0B68 */ Sample* usedSamples[128];
|
|
/* 0x0D68 */ AudioPreloadReq preloadSampleStack[128];
|
|
/* 0x1768 */ s32 numUsedSamples;
|
|
/* 0x176C */ s32 preloadSampleStackTop;
|
|
/* 0x1770 */ AudioAsyncLoad asyncLoads[0x10];
|
|
/* 0x1CF0 */ OSMesgQueue asyncLoadUnkMediumQueue;
|
|
/* 0x1D08 */ char unk_1D08[0x40];
|
|
/* 0x1D48 */ AudioAsyncLoad* curUnkMediumLoad;
|
|
/* 0x1D4C */ u32 slowLoadPos;
|
|
/* 0x1D50 */ AudioSlowLoad slowLoads[2];
|
|
/* 0x1E18 */ OSPiHandle* cartHandle;
|
|
/* 0x1E1C */ OSPiHandle* driveHandle;
|
|
/* 0x1E20 */ OSMesgQueue externalLoadQueue;
|
|
/* 0x1E38 */ OSMesg externalLoadMsgBuf[16];
|
|
/* 0x1E78 */ OSMesgQueue preloadSampleQueue;
|
|
/* 0x1E90 */ OSMesg preloadSampleMsgBuf[16];
|
|
/* 0x1ED0 */ OSMesgQueue curAudioFrameDmaQueue;
|
|
/* 0x1EE8 */ OSMesg curAudioFrameDmaMsgBuf[64];
|
|
/* 0x1FE8 */ OSIoMesg curAudioFrameDmaIoMsgBuf[64];
|
|
/* 0x25E8 */ OSMesgQueue syncDmaQueue;
|
|
/* 0x2600 */ OSMesg syncDmaMesg;
|
|
/* 0x2604 */ OSIoMesg syncDmaIoMesg;
|
|
/* 0x261C */ SampleDma* sampleDmas;
|
|
/* 0x2620 */ u32 sampleDmaCount;
|
|
/* 0x2624 */ u32 sampleDmaListSize1;
|
|
/* 0x2628 */ s32 unused2628;
|
|
/* 0x262C */ u8 sampleDmaReuseQueue1[0x100]; // read pos <= write pos, wrapping mod 256
|
|
/* 0x272C */ u8 sampleDmaReuseQueue2[0x100];
|
|
/* 0x282C */ u8 sampleDmaReuseQueue1RdPos; // Read position for short-lived sampleDma
|
|
/* 0x282D */ u8 sampleDmaReuseQueue2RdPos; // Read position for long-lived sampleDma
|
|
/* 0x282E */ u8 sampleDmaReuseQueue1WrPos; // Write position for short-lived sampleDma
|
|
/* 0x282F */ u8 sampleDmaReuseQueue2WrPos; // Write position for long-lived sampleDma
|
|
/* 0x2830 */ AudioTable* sequenceTable;
|
|
/* 0x2834 */ AudioTable* soundFontTable;
|
|
/* 0x2838 */ AudioTable* sampleBankTable;
|
|
/* 0x283C */ u8* sequenceFontTable;
|
|
/* 0x2840 */ u16 numSequences;
|
|
/* 0x2844 */ SoundFont* soundFontList;
|
|
/* 0x2848 */ AudioBufferParameters audioBufferParameters;
|
|
/* 0x2870 */ f32 unk_2870;
|
|
/* 0x2874 */ s32 sampleDmaBufSize1;
|
|
/* 0x2874 */ s32 sampleDmaBufSize2;
|
|
/* 0x287C */ char unk_287C[0x10];
|
|
/* 0x288C */ s32 sampleDmaBufSize;
|
|
/* 0x2890 */ s32 maxAudioCmds;
|
|
/* 0x2894 */ s32 numNotes;
|
|
/* 0x2898 */ s16 maxTempo; // Maximum possible tempo (seqTicks per minute), using every tick as a seqTick to process a .seq file
|
|
/* 0x289A */ s8 soundMode;
|
|
/* 0x289C */ s32 totalTaskCount; // The total number of times the top-level function on the audio thread has run since audio was initialized
|
|
/* 0x28A0 */ s32 curAudioFrameDmaCount;
|
|
/* 0x28A4 */ s32 rspTaskIndex;
|
|
/* 0x28A8 */ s32 curAiBufIndex;
|
|
/* 0x28AC */ Acmd* abiCmdBufs[2]; // Pointer to audio heap where the audio binary interface command lists (for the rsp) are stored. Two lists that alternate every frame
|
|
/* 0x28B4 */ Acmd* curAbiCmdBuf; // Pointer to the currently active abiCmdBufs
|
|
/* 0x28B8 */ AudioTask* curTask;
|
|
/* 0x28BC */ char unk_28BC[0x4];
|
|
/* 0x28C0 */ AudioTask rspTask[2];
|
|
/* 0x2960 */ f32 maxTempoTvTypeFactors; // tvType factors that impact maxTempo, in units of milliseconds/frame
|
|
/* 0x2964 */ s32 refreshRate;
|
|
/* 0x2968 */ s16* aiBuffers[3];
|
|
/* 0x2974 */ s16 aiBufLengths[3];
|
|
/* 0x297C */ u32 audioRandom;
|
|
/* 0x2980 */ s32 audioErrorFlags;
|
|
/* 0x2984 */ volatile u32 resetTimer;
|
|
/* 0x2988 */ char unk_2988[0x8];
|
|
/* 0x2990 */ AudioAllocPool sessionPool; // A sub-pool to main pool, contains all sub-pools and data that changes every audio reset
|
|
/* 0x29A0 */ AudioAllocPool externalPool; // pool allocated externally to the audio heap. Never used in game
|
|
/* 0x29B0 */ AudioAllocPool initPool;// A sub-pool to the main pool, contains all sub-pools and data that persists every audio reset
|
|
/* 0x29C0 */ AudioAllocPool miscPool; // A sub-pool to the session pool.
|
|
/* 0x29D0 */ char unk_29D0[0x20]; // probably two unused pools
|
|
/* 0x29F0 */ AudioAllocPool cachePool; // The common pool for cache entries
|
|
/* 0x2A00 */ AudioAllocPool persistentCommonPool; // A sub-pool to the cache pool, contains caches for data stored persistently
|
|
/* 0x2A10 */ AudioAllocPool temporaryCommonPool; // A sub-pool to the cache pool, contains caches for data stored temporarily
|
|
/* 0x2A20 */ AudioCache seqCache; // Cache to store sequences
|
|
/* 0x2B30 */ AudioCache fontCache; // Cache to store soundFonts
|
|
/* 0x2C40 */ AudioCache sampleBankCache; // Cache for loading entire sample banks
|
|
/* 0x2D50 */ AudioAllocPool permanentPool; // Pool to store audio data that is always loaded. Used for sfxs
|
|
/* 0x2D60 */ AudioCacheEntry permanentCache[32]; // individual entries to the permanent pool
|
|
/* 0x2EE0 */ AudioSampleCache persistentSampleCache; // Stores individual samples persistently
|
|
/* 0x3174 */ AudioSampleCache temporarySampleCache; // Stores individual samples temporarily
|
|
/* 0x3408 */ AudioSessionPoolSplit sessionPoolSplit; // splits session pool into the cache pool and misc pool
|
|
/* 0x3418 */ AudioCachePoolSplit cachePoolSplit; // splits cache pool into the persistent & temporary common pools
|
|
/* 0x3420 */ AudioCommonPoolSplit persistentCommonPoolSplit;// splits persistent common pool into caches for sequences, soundFonts, sample banks
|
|
/* 0x342C */ AudioCommonPoolSplit temporaryCommonPoolSplit; // splits temporary common pool into caches for sequences, soundFonts, sample banks
|
|
/* 0x3438 */ u8 sampleFontLoadStatus[0x30];
|
|
/* 0x3468 */ u8 fontLoadStatus[0x30];
|
|
/* 0x3498 */ u8 seqLoadStatus[0x80];
|
|
/* 0x3518 */ volatile u8 resetStatus;
|
|
/* 0x3519 */ u8 specId;
|
|
/* 0x351C */ s32 audioResetFadeOutFramesLeft;
|
|
/* 0x3520 */ f32* adsrDecayTable; // A table on the audio heap that stores decay rates used for adsr
|
|
/* 0x3524 */ u8* audioHeap;
|
|
/* 0x3528 */ u32 audioHeapSize;
|
|
/* 0x352C */ Note* notes;
|
|
/* 0x3530 */ SequencePlayer seqPlayers[4];
|
|
/* 0x3AB0 */ SequenceLayer sequenceLayers[64];
|
|
/* 0x5AB0 */ SequenceChannel sequenceChannelNone;
|
|
/* 0x5B84 */ s32 noteSubEuOffset;
|
|
/* 0x5B88 */ AudioListItem layerFreeList;
|
|
/* 0x5B98 */ NotePool noteFreeLists;
|
|
/* 0x5BD8 */ u8 threadCmdWritePos;
|
|
/* 0x5BD9 */ u8 threadCmdReadPos;
|
|
/* 0x5BDA */ u8 threadCmdQueueFinished;
|
|
/* 0x5BDC */ u16 threadCmdChannelMask[4]; // bitfield for 16 channels. When processing an audio thread channel command on all channels, only process channels with their bit set.
|
|
/* 0x5BE4 */ OSMesgQueue* audioResetQueueP;
|
|
/* 0x5BE8 */ OSMesgQueue* taskStartQueueP;
|
|
/* 0x5BEC */ OSMesgQueue* threadCmdProcQueueP;
|
|
/* 0x5BF0 */ OSMesgQueue taskStartQueue;
|
|
/* 0x5C08 */ OSMesgQueue threadCmdProcQueue;
|
|
/* 0x5C20 */ OSMesgQueue audioResetQueue;
|
|
/* 0x5C38 */ OSMesg taskStartMsgBuf[1];
|
|
/* 0x5C3C */ OSMesg audioResetMsgBuf[1];
|
|
/* 0x5C40 */ OSMesg threadCmdProcMsgBuf[4];
|
|
/* 0x5C50 */ AudioCmd threadCmdBuf[0x100]; // Audio thread commands used to transfer audio requests from the graph thread to the audio thread
|
|
} AudioContext; // size = 0x6450
|
|
|
|
typedef struct NoteSubAttributes {
|
|
/* 0x00 */ u8 reverbVol;
|
|
/* 0x01 */ u8 gain; // Increases volume by a multiplicative scaling factor. Represented as a UQ4.4 number
|
|
/* 0x02 */ u8 pan;
|
|
/* 0x03 */ Stereo stereo;
|
|
/* 0x04 */ f32 frequency;
|
|
/* 0x08 */ f32 velocity;
|
|
/* 0x0C */ char unk_0C[0x4];
|
|
/* 0x10 */ s16* filter;
|
|
/* 0x14 */ u8 combFilterSize;
|
|
/* 0x16 */ u16 combFilterGain;
|
|
} NoteSubAttributes; // size = 0x18
|
|
|
|
typedef struct TempoData {
|
|
/* 0x0 */ s16 unk_00; // set to 0x1C00, unused
|
|
/* 0x2 */ s16 seqTicksPerBeat;
|
|
} TempoData; // size = 0x4
|
|
|
|
typedef struct AudioHeapInitSizes {
|
|
/* 0x00 */ u32 heapSize; // total number of bytes allocated to the audio heap. Must be <= the size of `gAudioHeap` (ideally about the same size)
|
|
/* 0x04 */ u32 initPoolSize; // The entire audio heap is split into two pools.
|
|
/* 0x08 */ u32 permanentPoolSize;
|
|
} AudioHeapInitSizes; // size = 0xC
|
|
|
|
// TODO these prototypes should be sorted into the relevant audio header files
|
|
|
|
Acmd* AudioSynth_Update(Acmd* cmdStart, s32* cmdCnt, s16* aiStart, s32 aiBufLen);
|
|
void AudioHeap_DiscardFont(s32 fontId);
|
|
void AudioHeap_ReleaseNotesForFont(s32 fontId);
|
|
void AudioHeap_DiscardSequence(s32 seqId);
|
|
void AudioHeap_WritebackDCache(void* ramAddr, u32 size);
|
|
void* AudioHeap_AllocZeroedAttemptExternal(AudioAllocPool* pool, u32 size);
|
|
void* AudioHeap_AllocAttemptExternal(AudioAllocPool* pool, u32 size);
|
|
void* AudioHeap_AllocDmaMemory(AudioAllocPool* pool, u32 size);
|
|
void* AudioHeap_AllocDmaMemoryZeroed(AudioAllocPool* pool, u32 size);
|
|
void* AudioHeap_AllocZeroed(AudioAllocPool* pool, u32 size);
|
|
void* AudioHeap_Alloc(AudioAllocPool* pool, u32 size);
|
|
void AudioHeap_InitPool(AudioAllocPool* pool, void* ramAddr, u32 size);
|
|
void AudioHeap_PopPersistentCache(s32 tableType);
|
|
void AudioHeap_InitMainPools(s32 initPoolSize);
|
|
void* AudioHeap_AllocCached(s32 tableType, s32 size, s32 cache, s32 id);
|
|
void* AudioHeap_SearchCaches(s32 tableType, s32 cache, s32 id);
|
|
void* AudioHeap_SearchRegularCaches(s32 tableType, s32 cache, s32 id);
|
|
void AudioHeap_LoadFilter(s16* filter, s32 lowPassCutoff, s32 highPassCutoff);
|
|
s32 AudioHeap_ResetStep(void);
|
|
void AudioHeap_Init(void);
|
|
void* AudioHeap_SearchPermanentCache(s32 tableType, s32 id);
|
|
void* AudioHeap_AllocPermanent(s32 tableType, s32 id, u32 size);
|
|
void* AudioHeap_AllocSampleCache(u32 size, s32 fontId, void* sampleAddr, s8 medium, s32 cache);
|
|
void AudioHeap_ApplySampleBankCache(s32 sampleBankId);
|
|
void AudioLoad_DecreaseSampleDmaTtls(void);
|
|
void* AudioLoad_DmaSampleData(u32 devAddr, u32 size, s32 arg2, u8* dmaIndexRef, s32 medium);
|
|
void AudioLoad_InitSampleDmaBuffers(s32 numNotes);
|
|
s32 AudioLoad_IsFontLoadComplete(s32 fontId);
|
|
s32 AudioLoad_IsSeqLoadComplete(s32 seqId);
|
|
void AudioLoad_SetFontLoadStatus(s32 fontId, s32 loadStatus);
|
|
void AudioLoad_SetSeqLoadStatus(s32 seqId, s32 loadStatus);
|
|
void AudioLoad_SyncLoadSeqParts(s32 seqId, s32 arg1);
|
|
s32 AudioLoad_SyncLoadInstrument(s32 fontId, s32 instId, s32 drumId);
|
|
void AudioLoad_AsyncLoadSeq(s32 seqId, s32 arg1, s32 retData, OSMesgQueue* retQueue);
|
|
void AudioLoad_AsyncLoadSampleBank(s32 sampleBankId, s32 arg1, s32 retData, OSMesgQueue* retQueue);
|
|
void AudioLoad_AsyncLoadFont(s32 fontId, s32 arg1, s32 retData, OSMesgQueue* retQueue);
|
|
u8* AudioLoad_GetFontsForSequence(s32 seqId, u32* outNumFonts);
|
|
void AudioLoad_DiscardSeqFonts(s32 seqId);
|
|
s32 AudioLoad_SyncInitSeqPlayer(s32 playerIdx, s32 seqId, s32 arg2);
|
|
s32 AudioLoad_SyncInitSeqPlayerSkipTicks(s32 playerIdx, s32 seqId, s32 skipTicks);
|
|
void AudioLoad_ProcessLoads(s32 resetStatus);
|
|
void AudioLoad_SetDmaHandler(DmaHandler callback);
|
|
void AudioLoad_Init(void* heap, u32 heapSize);
|
|
void AudioLoad_InitSlowLoads(void);
|
|
s32 AudioLoad_SlowLoadSample(s32 fontId, s32 instId, s8* status);
|
|
s32 AudioLoad_SlowLoadSeq(s32 seqId, u8* ramAddr, s8* status);
|
|
void AudioLoad_InitAsyncLoads(void);
|
|
void AudioLoad_LoadPermanentSamples(void);
|
|
void AudioLoad_ScriptLoad(s32 tableType, s32 id, s8* status);
|
|
void AudioLoad_ProcessScriptLoads(void);
|
|
void AudioLoad_InitScriptLoads(void);
|
|
|
|
AudioTask* AudioThread_Update(void);
|
|
void AudioThread_QueueCmdF32(u32 opArgs, f32 data);
|
|
void AudioThread_QueueCmdS32(u32 opArgs, s32 data);
|
|
void AudioThread_QueueCmdS8(u32 opArgs, s8 data);
|
|
void AudioThread_QueueCmdU16(u32 opArgs, u16 data);
|
|
s32 AudioThread_ScheduleProcessCmds(void);
|
|
u32 func_800E5E20(u32* out);
|
|
u8* AudioThread_GetFontsForSequence(s32 seqId, u32* outNumFonts);
|
|
s32 func_800E5EDC(void);
|
|
s32 AudioThread_ResetAudioHeap(s32 specId);
|
|
void AudioThread_PreNMIInternal(void);
|
|
s32 func_800E6680(void);
|
|
u32 AudioThread_NextRandom(void);
|
|
void AudioThread_InitMesgQueues(void);
|
|
|
|
void Audio_InvalDCache(void* buf, s32 size);
|
|
void Audio_WritebackDCache(void* buf, s32 size);
|
|
s32 osAiSetNextBuffer(void*, u32);
|
|
void Audio_InitNoteSub(Note* note, NoteSubEu* sub, NoteSubAttributes* attrs);
|
|
void Audio_NoteSetResamplingRate(NoteSubEu* noteSubEu, f32 resamplingRateInput);
|
|
void Audio_NoteInit(Note* note);
|
|
void Audio_NoteDisable(Note* note);
|
|
void Audio_ProcessNotes(void);
|
|
TunedSample* Audio_GetInstrumentTunedSample(Instrument* instrument, s32 semitone);
|
|
Instrument* Audio_GetInstrumentInner(s32 fontId, s32 instId);
|
|
Drum* Audio_GetDrum(s32 fontId, s32 drumId);
|
|
SoundEffect* Audio_GetSoundEffect(s32 fontId, s32 sfxId);
|
|
s32 Audio_SetFontInstrument(s32 instrumentType, s32 fontId, s32 index, void* value);
|
|
void Audio_SeqLayerDecayRelease(SequenceLayer* layer, s32 target);
|
|
void Audio_SeqLayerNoteDecay(SequenceLayer* layer);
|
|
void Audio_SeqLayerNoteRelease(SequenceLayer* layer);
|
|
s32 Audio_BuildSyntheticWave(Note* note, SequenceLayer* layer, s32 waveId);
|
|
void Audio_InitSyntheticWave(Note* note, SequenceLayer* layer);
|
|
void Audio_InitNoteList(AudioListItem* list);
|
|
void Audio_InitNoteLists(NotePool* pool);
|
|
void Audio_InitNoteFreeList(void);
|
|
void Audio_NotePoolClear(NotePool* pool);
|
|
void Audio_NotePoolFill(NotePool* pool, s32 count);
|
|
void Audio_AudioListPushFront(AudioListItem* list, AudioListItem* item);
|
|
void Audio_AudioListRemove(AudioListItem* item);
|
|
Note* Audio_FindNodeWithPrioLessThan(AudioListItem* list, s32 limit);
|
|
void Audio_NoteInitForLayer(Note* note, SequenceLayer* layer);
|
|
void func_800E82C0(Note* note, SequenceLayer* layer);
|
|
void Audio_NoteReleaseAndTakeOwnership(Note* note, SequenceLayer* layer);
|
|
Note* Audio_AllocNoteFromDisabled(NotePool* pool, SequenceLayer* layer);
|
|
Note* Audio_AllocNoteFromDecaying(NotePool* pool, SequenceLayer* layer);
|
|
Note* Audio_AllocNoteFromActive(NotePool* pool, SequenceLayer* layer);
|
|
Note* Audio_AllocNote(SequenceLayer* layer);
|
|
void Audio_NoteInitAll(void);
|
|
void Audio_SequenceChannelProcessSound(SequenceChannel* channel, s32 recalculateVolume, s32 applyBend);
|
|
void Audio_SequencePlayerProcessSound(SequencePlayer* seqPlayer);
|
|
f32 Audio_GetPortamentoFreqScale(Portamento* portamento);
|
|
s16 Audio_GetVibratoPitchChange(VibratoState* vib);
|
|
f32 Audio_GetVibratoFreqScale(VibratoState* vib);
|
|
void Audio_NoteVibratoUpdate(Note* note);
|
|
void Audio_NoteVibratoInit(Note* note);
|
|
void Audio_NotePortamentoInit(Note* note);
|
|
void Audio_AdsrInit(AdsrState* adsr, EnvelopePoint* envelope, s16* volOut);
|
|
f32 Audio_AdsrUpdate(AdsrState* adsr);
|
|
void AudioSeq_SequenceChannelDisable(SequenceChannel* channel);
|
|
void AudioSeq_SequencePlayerDisableAsFinished(SequencePlayer* seqPlayer);
|
|
void AudioSeq_SequencePlayerDisable(SequencePlayer* seqPlayer);
|
|
void AudioSeq_AudioListPushBack(AudioListItem* list, AudioListItem* item);
|
|
void* AudioSeq_AudioListPopBack(AudioListItem* list);
|
|
void AudioSeq_ProcessSequences(s32 arg0);
|
|
void AudioSeq_SkipForwardSequence(SequencePlayer* seqPlayer);
|
|
void AudioSeq_ResetSequencePlayer(SequencePlayer* seqPlayer);
|
|
void AudioSeq_InitSequencePlayerChannels(s32 playerIdx);
|
|
void AudioSeq_InitSequencePlayers(void);
|
|
|
|
void AudioDebug_Draw(struct GfxPrint* printer);
|
|
void AudioDebug_ScrPrt(const char* str, u16 num);
|
|
void Audio_Update(void);
|
|
void Audio_SetSfxProperties(u8 bankId, u8 entryIdx, u8 channelIndex);
|
|
void Audio_PlayCutsceneEffectsSequence(u8 csEffectType);
|
|
void func_800F4010(Vec3f* pos, u16 sfxId, f32);
|
|
void Audio_PlaySfxRandom(Vec3f* pos, u16 baseSfxId, u8 randLim);
|
|
void func_800F4138(Vec3f* pos, u16 sfxId, f32);
|
|
void func_800F4190(Vec3f* pos, u16 sfxId);
|
|
void func_800F436C(Vec3f* pos, u16 sfxId, f32 arg2);
|
|
void func_800F4414(Vec3f* pos, u16 sfxId, f32);
|
|
void func_800F44EC(s8 arg0, s8 arg1);
|
|
void func_800F4524(Vec3f* pos, u16 sfxId, s8 arg2);
|
|
void func_800F4254(Vec3f* pos, u8 level);
|
|
void Audio_PlaySfxRiver(Vec3f* pos, f32 freqScale);
|
|
void Audio_PlaySfxWaterfall(Vec3f* pos, f32 freqScale);
|
|
void Audio_SetBgmVolumeOffDuringFanfare(void);
|
|
void Audio_SetBgmVolumeOnDuringFanfare(void);
|
|
void Audio_SetMainBgmVolume(u8 targetVol, u8 volFadeTimer);
|
|
void Audio_SetGanonsTowerBgmVolumeLevel(u8 ganonsTowerLevel);
|
|
void Audio_LowerMainBgmVolume(u8 volume);
|
|
void Audio_PlaySfxIncreasinglyTransposed(Vec3f* pos, s16 sfxId, u8* semitones);
|
|
void Audio_ResetIncreasingTranspose(void);
|
|
void Audio_PlaySfxTransposed(Vec3f* pos, u16 sfxId, s8 semitone);
|
|
void func_800F4C58(Vec3f* pos, u16 sfxId, u8);
|
|
void func_800F4E30(Vec3f* pos, f32);
|
|
void Audio_ClearSariaBgm(void);
|
|
void Audio_ClearSariaBgmAtPos(Vec3f* pos);
|
|
void Audio_PlaySariaBgm(Vec3f* pos, u16 seqId, u16 distMax);
|
|
void Audio_ClearSariaBgm2(void);
|
|
void Audio_PlayMorningSceneSequence(u16 seqId);
|
|
void Audio_PlaySceneSequence(u16 seqId);
|
|
void Audio_SetMainBgmTempoFreqAfterFanfare(f32 scaleTempoAndFreq, u8 duration);
|
|
void Audio_PlayWindmillBgm(void);
|
|
void Audio_SetFastTempoForTimedMinigame(void);
|
|
void Audio_PlaySequenceInCutscene(u16 seqId);
|
|
void Audio_StopSequenceInCutscene(u16 seqId);
|
|
s32 Audio_IsSequencePlaying(u16 seqId);
|
|
void func_800F5ACC(u16 seqId);
|
|
void func_800F5B58(void);
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void func_800F5BF0(u8 natureAmbienceId);
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void Audio_PlayFanfare(u16);
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void func_800F5C2C(void);
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void Audio_PlaySequenceWithSeqPlayerIO(u8 seqPlayerIndex, u16 seqId, u8 fadeInDuration, s8 ioPort, s8 ioData);
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void Audio_SetSequenceMode(u8 seqMode);
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void Audio_SetBgmEnemyVolume(f32 dist);
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void Audio_UpdateMalonSinging(f32 dist, u16 seqId);
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void func_800F64E0(u8 arg0);
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void Audio_ToggleMalonSinging(u8 malonSingingDisabled);
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void Audio_SetEnvReverb(s8 reverb);
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void Audio_SetCodeReverb(s8 reverb);
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void func_800F6700(s8 audioSetting);
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void Audio_SetBaseFilter(u8);
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void Audio_SetExtraFilter(u8);
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void Audio_SetCutsceneFlag(s8 flag);
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void Audio_PlaySfxIfNotInCutscene(u16 sfxId);
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void func_800F6964(u16);
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void Audio_StopBgmAndFanfare(u16 fadeOutDuration);
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void func_800F6B3C(void);
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void func_800F6BDC(void);
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void Audio_PreNMI(void);
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void Audio_SetNatureAmbienceChannelIO(u8 channelIdxRange, u8 ioPort, u8 ioData);
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void Audio_PlayNatureAmbienceSequence(u8 natureAmbienceId);
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void Audio_Init(void);
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void Audio_InitSound(void);
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void func_800F7170(void);
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void func_800F71BC(s32 arg0);
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#endif
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