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zelda2774 2021-08-30 16:54:04 +02:00
parent 65f324c3c9
commit 559be01e03
3 changed files with 21 additions and 18 deletions
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1
include/macros.h
View file

@ -10,6 +10,7 @@
#define ALIGN16(val) (((val) + 0xF) & ~0xF)
#define ALIGN32(val) (((val) + 0x1F) & ~0x1F)
#define ALIGN64(val) (((val) + 0x3F) & ~0x3F)
#define ALIGN256(val) (((val) + 0xFF) & ~0xFF)
#define SQ(x) ((x)*(x))

9
include/z64audio.h
View file

@ -35,9 +35,6 @@
#define CODEC_ADPCM 0
#define CODEC_S8 1
#define DEFAULT_LEN_1CH 0x1A0
#define DEFAULT_LEN_2CH 0x340
struct Note;
struct NotePool;
struct SequenceChannel;
@ -95,7 +92,7 @@ typedef struct {
/* 0x00 */ s32 order;
/* 0x04 */ s32 npredictors;
/* 0x08 */ s16 book[1]; // size 8 * order * npredictors. 8-byte aligned
} AdpcmBook;
} AdpcmBook; // size >= 0x8
typedef struct {
/* 0x00 */ u32 codec : 4;
@ -175,7 +172,7 @@ typedef struct {
/* 0x08 */ AudioBankSound lowNotesSound;
/* 0x10 */ AudioBankSound normalNotesSound;
/* 0x18 */ AudioBankSound highNotesSound;
} Instrument; // size >= 0x20
} Instrument; // size = 0x20
typedef struct {
/* 0x00 */ u8 releaseRate;
@ -183,7 +180,7 @@ typedef struct {
/* 0x02 */ u8 loaded;
/* 0x04 */ AudioBankSound sound;
/* 0x14 */ AdsrEnvelope* envelope;
} Drum; // size >= 0x14
} Drum; // size = 0x14
typedef struct {
/* 0x00 */ u8 numInstruments;

29
src/code/audio_synthesis.c
View file

@ -1,6 +1,9 @@
#include "ultra64.h"
#include "global.h"
#define DEFAULT_LEN_1CH 0x1A0
#define DEFAULT_LEN_2CH 0x340
#define DMEM_TEMP 0x3C0
#define DMEM_UNCOMPRESSED_NOTE 0x580
#define DMEM_NOTE_PAN_TEMP 0x5C0
@ -48,7 +51,6 @@ void AudioSynth_InitNextRingBuf(s32 chunkLen, s32 bufIndex, s32 reverbIndex) {
if (reverb->downsampleRate >= 2) {
if (reverb->framesToIgnore == 0) {
bufItem = &reverb->items[reverb->curFrame][bufIndex];
// inval dcache
Audio_osInvalDCache(bufItem->toDownsampleLeft, DEFAULT_LEN_2CH);
for (j = 0, i = 0; i < bufItem->lengthA / 2; j += reverb->downsampleRate, i++) {
@ -550,7 +552,7 @@ Acmd* AudioSynth_SaveRingBuffer2(Acmd* cmd, SynthesisReverb* reverb, s16 bufInde
}
Acmd* AudioSynth_DoOneAudioUpdate(s16* aiBuf, s32 aiBufLen, Acmd* cmd, s32 updateIndex) {
u8 noteInds[0x5C];
u8 noteIndices[0x5C];
s16 count;
s16 reverbIndex;
SynthesisReverb* reverb;
@ -566,7 +568,7 @@ Acmd* AudioSynth_DoOneAudioUpdate(s16* aiBuf, s32 aiBufLen, Acmd* cmd, s32 updat
if (gAudioContext.numSynthesisReverbs == 0) {
for (i = 0; i < gAudioContext.maxSimultaneousNotes; i++) {
if (gAudioContext.noteSubsEu[t + i].bitField0.s.enabled) {
noteInds[count++] = i;
noteIndices[count++] = i;
}
}
} else {
@ -574,7 +576,7 @@ Acmd* AudioSynth_DoOneAudioUpdate(s16* aiBuf, s32 aiBufLen, Acmd* cmd, s32 updat
for (i = 0; i < gAudioContext.maxSimultaneousNotes; i++) {
noteSubEu = &gAudioContext.noteSubsEu[t + i];
if (noteSubEu->bitField0.s.enabled && noteSubEu->bitField1.s.reverbIndex == reverbIndex) {
noteInds[count++] = i;
noteIndices[count++] = i;
}
}
}
@ -583,7 +585,7 @@ Acmd* AudioSynth_DoOneAudioUpdate(s16* aiBuf, s32 aiBufLen, Acmd* cmd, s32 updat
noteSubEu = &gAudioContext.noteSubsEu[t + i];
if (noteSubEu->bitField0.s.enabled &&
noteSubEu->bitField1.s.reverbIndex >= gAudioContext.numSynthesisReverbs) {
noteInds[count++] = i;
noteIndices[count++] = i;
}
}
}
@ -617,10 +619,11 @@ Acmd* AudioSynth_DoOneAudioUpdate(s16* aiBuf, s32 aiBufLen, Acmd* cmd, s32 updat
}
while (i < count) {
noteSubEu2 = &gAudioContext.noteSubsEu[noteInds[i] + t];
noteSubEu2 = &gAudioContext.noteSubsEu[noteIndices[i] + t];
if (noteSubEu2->bitField1.s.reverbIndex == reverbIndex) {
cmd = AudioSynth_ProcessNote(noteInds[i], noteSubEu2, &gAudioContext.notes[noteInds[i]].synthesisState,
aiBuf, aiBufLen, cmd, updateIndex);
cmd = AudioSynth_ProcessNote(noteIndices[i], noteSubEu2,
&gAudioContext.notes[noteIndices[i]].synthesisState, aiBuf, aiBufLen, cmd,
updateIndex);
} else {
break;
}
@ -643,9 +646,9 @@ Acmd* AudioSynth_DoOneAudioUpdate(s16* aiBuf, s32 aiBufLen, Acmd* cmd, s32 updat
}
while (i < count) {
cmd =
AudioSynth_ProcessNote(noteInds[i], &gAudioContext.noteSubsEu[t + noteInds[i]],
&gAudioContext.notes[noteInds[i]].synthesisState, aiBuf, aiBufLen, cmd, updateIndex);
cmd = AudioSynth_ProcessNote(noteIndices[i], &gAudioContext.noteSubsEu[t + noteIndices[i]],
&gAudioContext.notes[noteIndices[i]].synthesisState, aiBuf, aiBufLen, cmd,
updateIndex);
i++;
}
@ -736,6 +739,8 @@ Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSubEu, NoteSynthesisS
nSamplesToLoad = samplesLenFixedPoint >> 16;
synthState->samplePosFrac = samplesLenFixedPoint & 0xFFFF;
// Partially-optimized out no-op ifs required for matching. SM64 decomp
// makes it clear that this is how it should look.
if (synthState->numAdpcmParts == 1 && nParts == 2) {
} else if (synthState->numAdpcmParts == 2 && nParts == 1) {
} else {
@ -1223,6 +1228,6 @@ Acmd* AudioSynth_NoteApplyHeadsetPanEffects(Acmd* cmd, NoteSubEu* noteSubEu, Not
aSaveBuffer(cmd++, DMEM_NOTE_PAN_TEMP + bufLen, &synthState->synthesisBuffers->panResampleState[0x8],
ALIGN16(panShift));
}
aAddMixer(cmd++, ((bufLen + 0x3F) & ~0x3F), DMEM_NOTE_PAN_TEMP, dest, 0x7FFF);
aAddMixer(cmd++, ALIGN64(bufLen), DMEM_NOTE_PAN_TEMP, dest, 0x7FFF);
return cmd;
}