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authorzelda2774 <69368340+zelda2774@users.noreply.github.com>2021-09-01 00:53:35 +0200
committerGitHub <noreply@github.com>2021-08-31 18:53:35 -0400
commit4444c5ea35a272e9e9ea157df92d7bb3eac8fc05 (patch)
tree169ac947769558790055b7ee54fce13d5d154812 /src/code
parent430a172183fe6f7fe5781325386803e3f1f975cd (diff)
audio_synthesis OK + partially documented (#920)
* Match func_800DDB64 * Match func_800DC910 * remove stack comments * Move nop padding to ucode_disas * Partially label audio_synthesis To some extent copied from sm64. * cleanup * Reverb and ReverbBits have nothing to do with reverbs * review * naming * Undo changes to permuter_settings.toml Co-authored-by: zelda2774 <zelda2774@invalid>
Diffstat (limited to 'src/code')
-rw-r--r--src/code/audio_heap.c50
-rw-r--r--src/code/audio_load.c60
-rw-r--r--src/code/audio_playback.c128
-rw-r--r--src/code/audio_seqplayer.c28
-rw-r--r--src/code/audio_synthesis.c1220
-rw-r--r--src/code/code_800E4FE0.c10
-rw-r--r--src/code/code_800EC960.c22
-rw-r--r--src/code/ucode_disas.c5
8 files changed, 783 insertions, 740 deletions
diff --git a/src/code/audio_heap.c b/src/code/audio_heap.c
index 8c4ea64f7..1d264a1e7 100644
--- a/src/code/audio_heap.c
+++ b/src/code/audio_heap.c
@@ -621,46 +621,46 @@ void func_800DF1D8(f32 arg0, f32 arg1, u16* arg2) {
}
}
-void func_800DF5AC(s16* arg0) {
+void func_800DF5AC(s16* filter) {
s32 i;
for (i = 0; i < 8; i++) {
- arg0[i] = 0;
+ filter[i] = 0;
}
}
-void func_800DF5DC(s16* arg0, s32 arg1) {
+void func_800DF5DC(s16* filter, s32 arg1) {
s32 i;
s16* ptr = &D_80130228[8 * arg1];
for (i = 0; i < 8; i++) {
- arg0[i] = ptr[i];
+ filter[i] = ptr[i];
}
}
-void func_800DF630(s16* arg0, s32 arg1) {
+void func_800DF630(s16* filter, s32 arg1) {
s32 i;
s16* ptr = &D_80130328[8 * (arg1 - 1)];
for (i = 0; i < 8; i++) {
- arg0[i] = ptr[i];
+ filter[i] = ptr[i];
}
}
-void func_800DF688(s16* arg0, s32 arg1, s32 arg2) {
+void func_800DF688(s16* filter, s32 arg1, s32 arg2) {
s32 i;
if (arg1 == 0 && arg2 == 0) {
- func_800DF5DC(arg0, 0);
+ func_800DF5DC(filter, 0);
} else if (arg2 == 0) {
- func_800DF5DC(arg0, arg1);
+ func_800DF5DC(filter, arg1);
} else if (arg1 == 0) {
- func_800DF630(arg0, arg2);
+ func_800DF630(filter, arg2);
} else {
s16* ptr1 = &D_80130228[8 * arg1];
s16* ptr2 = &D_80130328[8 * (arg2 - 1)];
for (i = 0; i < 8; i++) {
- arg0[i] = (ptr1[i] + ptr2[i]) / 2;
+ filter[i] = (ptr1[i] + ptr2[i]) / 2;
}
}
}
@@ -898,8 +898,8 @@ void Audio_InitHeap(void) {
reverb->unk_14 = settings->unk_6 * 64;
reverb->unk_16 = settings->unk_8;
reverb->unk_18 = 0;
- reverb->unk_10 = settings->unk_C;
- reverb->unk_12 = settings->unk_E;
+ reverb->leakRtl = settings->leakRtl;
+ reverb->leakLtr = settings->leakLtr;
reverb->unk_05 = settings->unk_10;
reverb->unk_08 = settings->unk_12;
reverb->useReverb = 8;
@@ -914,8 +914,8 @@ void Audio_InitHeap(void) {
reverb->sound.sample = &reverb->sample;
reverb->sample.loop = &reverb->loop;
reverb->sound.tuning = 1.0f;
- reverb->sample.bits4 = 4;
- reverb->sample.bits2 = 0;
+ reverb->sample.codec = 4;
+ reverb->sample.medium = 0;
reverb->sample.size = reverb->windowSize * 2;
reverb->sample.sampleAddr = (u8*)reverb->leftRingBuf;
reverb->loop.start = 0;
@@ -939,19 +939,19 @@ void Audio_InitHeap(void) {
}
if (settings->unk_14 != 0) {
- reverb->unk_278 = Audio_AllocDmaMemoryZeroed(&gAudioContext.notesAndBuffersPool, 0x40);
- reverb->unk_270 = Audio_AllocDmaMemory(&gAudioContext.notesAndBuffersPool, 8 * sizeof(s16));
- func_800DF5DC(reverb->unk_270, settings->unk_14);
+ reverb->filterLeftState = Audio_AllocDmaMemoryZeroed(&gAudioContext.notesAndBuffersPool, 0x40);
+ reverb->filterLeft = Audio_AllocDmaMemory(&gAudioContext.notesAndBuffersPool, 8 * sizeof(s16));
+ func_800DF5DC(reverb->filterLeft, settings->unk_14);
} else {
- reverb->unk_270 = NULL;
+ reverb->filterLeft = NULL;
}
if (settings->unk_16 != 0) {
- reverb->unk_27C = Audio_AllocDmaMemoryZeroed(&gAudioContext.notesAndBuffersPool, 0x40);
- reverb->unk_274 = Audio_AllocDmaMemory(&gAudioContext.notesAndBuffersPool, 8 * sizeof(s16));
- func_800DF5DC(reverb->unk_274, settings->unk_16);
+ reverb->filterRightState = Audio_AllocDmaMemoryZeroed(&gAudioContext.notesAndBuffersPool, 0x40);
+ reverb->filterRight = Audio_AllocDmaMemory(&gAudioContext.notesAndBuffersPool, 8 * sizeof(s16));
+ func_800DF5DC(reverb->filterRight, settings->unk_16);
} else {
- reverb->unk_274 = NULL;
+ reverb->filterRight = NULL;
}
}
@@ -1181,7 +1181,7 @@ void func_800E0BB4(UnkHeapEntry* entry, AudioBankSample* sample) {
if (sample != NULL) {
if (sample->sampleAddr == entry->unk_08) {
sample->sampleAddr = entry->unk_0C;
- sample->bits2 = entry->unk_01;
+ sample->medium = entry->unk_01;
}
}
}
@@ -1251,7 +1251,7 @@ void func_800E0E0C(Struct_800E0E0C* arg0, AudioBankSample* sample) {
u8* sampleAddr = sample->sampleAddr;
if (start <= sampleAddr && sampleAddr < end) {
sample->sampleAddr = sampleAddr - start + arg0->unk_4;
- sample->bits2 = arg0->unk_C & 0xFF;
+ sample->medium = arg0->unk_C & 0xFF;
}
}
}
diff --git a/src/code/audio_load.c b/src/code/audio_load.c
index 62ffc50a5..3fa8a0788 100644
--- a/src/code/audio_load.c
+++ b/src/code/audio_load.c
@@ -6,7 +6,7 @@ typedef enum { LOAD_STATUS_WAITING, LOAD_STATUS_START, LOAD_STATUS_LOADING, LOAD
#define RELOC(v, base) (reloc = (void*)((u32)v + (u32)base))
/* forward declarations */
-s32 func_800E217C(s32 playerIdx, s32, s32);
+s32 func_800E217C(s32 playerIndex, s32, s32);
unk_ldr* func_800E2454(u32 bankId);
AudioBankSample* Audio_GetBankSample(s32 bankId, s32 sfxId);
void Audio_ProcessAsyncLoads(s32 arg0);
@@ -72,7 +72,7 @@ void func_800E11F0(void) {
gAudioContext.unk_2628 = 0;
}
-#pragma GLOBAL_ASM("asm/non_matchings/code/audio_load/func_800E12DC.s")
+#pragma GLOBAL_ASM("asm/non_matchings/code/audio_load/Audio_DmaSampleData.s")
void func_800E1618(s32 arg0) {
SampleDmaReq* temp_s0;
@@ -268,18 +268,18 @@ s32 func_800E1C78(AudioBankSample* sample, s32 arg1) {
void* sampleAddr;
if (sample->unk_bits25 == 1) {
- if (sample->bits2 != 0) {
- sampleAddr = func_800E05C4(sample->size, arg1, (void*)sample->sampleAddr, sample->bits2, 1);
+ if (sample->medium != 0) {
+ sampleAddr = func_800E05C4(sample->size, arg1, (void*)sample->sampleAddr, sample->medium, 1);
if (sampleAddr == NULL) {
return -1;
}
- if (sample->bits2 == 1) {
+ if (sample->medium == 1) {
func_800E2BCC(sample->sampleAddr, sampleAddr, sample->size, gAudioContext.audioTable->header.unk_02);
} else {
- Audio_DMAFastCopy(sample->sampleAddr, sampleAddr, sample->size, sample->bits2);
+ Audio_DMAFastCopy(sample->sampleAddr, sampleAddr, sample->size, sample->medium);
}
- sample->bits2 = 0;
+ sample->medium = 0;
sample->sampleAddr = sampleAddr;
}
}
@@ -381,27 +381,27 @@ void func_800E202C(s32 arg0) {
Audio_DiscardBank(arg0);
}
-s32 func_800E20D4(s32 playerIdx, s32 seqId, s32 arg2) {
+s32 func_800E20D4(s32 playerIndex, s32 seqId, s32 arg2) {
if (gAudioContext.resetTimer != 0) {
return 0;
} else {
- gAudioContext.seqPlayers[playerIdx].unk_DC = 0;
- return func_800E217C(playerIdx, seqId, arg2);
+ gAudioContext.seqPlayers[playerIndex].unk_DC = 0;
+ return func_800E217C(playerIndex, seqId, arg2);
}
}
-s32 func_800E2124(s32 playerIdx, s32 seqId, s32 arg2) {
+s32 func_800E2124(s32 playerIndex, s32 seqId, s32 arg2) {
if (gAudioContext.resetTimer != 0) {
return 0;
}
- gAudioContext.seqPlayers[playerIdx].unk_DC = arg2;
- return func_800E217C(playerIdx, seqId, 0);
+ gAudioContext.seqPlayers[playerIndex].unk_DC = arg2;
+ return func_800E217C(playerIndex, seqId, 0);
}
// InitSeqPlayer
-s32 func_800E217C(s32 playerIdx, s32 seqId, s32 arg2) {
- SequencePlayer* seqPlayer = &gAudioContext.seqPlayers[playerIdx];
+s32 func_800E217C(s32 playerIndex, s32 seqId, s32 arg2) {
+ SequencePlayer* seqPlayer = &gAudioContext.seqPlayers[playerIndex];
u8* seqData;
s32 phi_s0;
s32 phi_s1;
@@ -437,7 +437,7 @@ s32 func_800E217C(s32 playerIdx, s32 seqId, s32 arg2) {
seqPlayer->scriptState.depth = 0;
seqPlayer->delay = 0;
seqPlayer->finished = 0;
- seqPlayer->seqVariationEu = playerIdx;
+ seqPlayer->playerIndex = playerIndex;
Audio_ProcessSequence(seqPlayer);
}
@@ -1055,7 +1055,7 @@ s32 Audio_SyncLoadSample(s32 arg0, s32 arg1, u8* isDone) {
return -1;
}
- if (sample->bits2 == 0) {
+ if (sample->medium == 0) {
*isDone = 2;
return 0;
}
@@ -1067,10 +1067,10 @@ s32 Audio_SyncLoadSample(s32 arg0, s32 arg1, u8* isDone) {
syncLoad->sample = *sample;
syncLoad->isDone = isDone;
- syncLoad->ramAddr = func_800E05C4(sample->size, arg0, sample->sampleAddr, sample->bits2, 0);
+ syncLoad->ramAddr = func_800E05C4(sample->size, arg0, sample->sampleAddr, sample->medium, 0);
if (syncLoad->ramAddr == NULL) {
- if (sample->bits2 == 1 || sample->bits4 == 2) {
+ if (sample->medium == 1 || sample->codec == 2) {
*isDone = 0;
return -1;
} else {
@@ -1083,7 +1083,7 @@ s32 Audio_SyncLoadSample(s32 arg0, s32 arg1, u8* isDone) {
syncLoad->size = ALIGN16(sample->size);
syncLoad->unk_10 = syncLoad->ramAddr;
syncLoad->devAddr = sample->sampleAddr;
- syncLoad->unk_00 = sample->bits2;
+ syncLoad->unk_00 = sample->medium;
syncLoad->unk_01 = arg0;
syncLoad->unk_02 = arg1;
if (syncLoad->unk_00 == 1) {
@@ -1136,7 +1136,7 @@ void func_800E3678(AudioSyncLoad* syncLoad) {
syncLoad->sample = *sample;
sample->sampleAddr = syncLoad->unk_10;
- sample->bits2 = 0;
+ sample->medium = 0;
}
void Audio_ProcessSyncLoads(s32 resetStatus) {
@@ -1430,14 +1430,14 @@ void Audio_SampleReloc(AudioBankSound* sound, u32 arg1, RelocInfo* arg2) {
if (sample->size != 0 && sample->unk_bits25 != 1) {
sample->loop = RELOC(sample->loop, arg1);
sample->book = RELOC(sample->book, arg1);
- switch (sample->bits2) {
+ switch (sample->medium) {
case 0:
sample->sampleAddr = RELOC(sample->sampleAddr, arg2->unk_08);
- sample->bits2 = arg2->unk_10;
+ sample->medium = arg2->unk_10;
break;
case 1:
sample->sampleAddr = RELOC(sample->sampleAddr, arg2->unk_0C);
- sample->bits2 = arg2->unk_14;
+ sample->medium = arg2->unk_14;
break;
case 2:
case 3:
@@ -1445,7 +1445,7 @@ void Audio_SampleReloc(AudioBankSound* sound, u32 arg1, RelocInfo* arg2) {
}
sample->unk_bits25 = 1;
- if (sample->unk_bits26 && (sample->bits2 != 0)) {
+ if (sample->unk_bits26 && (sample->medium != 0)) {
gAudioContext.unk_B68[gAudioContext.unk_1768++] = sample;
}
}
@@ -1477,10 +1477,10 @@ s32 func_800E4590(s32 resetStatus) {
if (item->isFree == false) {
sample = item->sample;
- key = (u32)sample->sampleAddr + sample->size + sample->bits2;
+ key = (u32)sample->sampleAddr + sample->size + sample->medium;
if (key == item->endAndMediumKey) {
sample->sampleAddr = item->ramAddr;
- sample->bits2 = 0;
+ sample->medium = 0;
}
item->isFree = true;
}
@@ -1497,12 +1497,12 @@ s32 func_800E4590(s32 resetStatus) {
sample = item->sample;
unk = (sample->size >> 12) + 1;
- key = (u32)sample->sampleAddr + sample->size + sample->bits2;
+ key = (u32)sample->sampleAddr + sample->size + sample->medium;
if (key != item->endAndMediumKey) {
item->isFree = true;
gAudioContext.unk_176C--;
} else {
- Audio_InitAsyncReq((u32)sample->sampleAddr, item->ramAddr, sample->size, sample->bits2, unk,
+ Audio_InitAsyncReq((u32)sample->sampleAddr, item->ramAddr, sample->size, sample->medium, unk,
&gAudioContext.unk_1E78, item->encodedInfo);
break;
}
@@ -1563,7 +1563,7 @@ s32 func_800E478C(s32 bankId, AudioBankSample** sampleList) {
void func_800E48C0(AudioBankSound* sound) {
AudioBankSample* sample = sound->sample;
- if ((sample->size != 0) && (sample->unk_bits26) && (sample->bits2)) {
+ if ((sample->size != 0) && (sample->unk_bits26) && (sample->medium)) {
gAudioContext.unk_B68[gAudioContext.unk_1768++] = sample;
}
}
diff --git a/src/code/audio_playback.c b/src/code/audio_playback.c
index a9ce8e9c2..970aaf8f2 100644
--- a/src/code/audio_playback.c
+++ b/src/code/audio_playback.c
@@ -1,6 +1,6 @@
#include "global.h"
-void Audio_NoteSetVelPanReverb(Note* note, NoteSubEu* sub, Reverb* reverb) {
+void Audio_InitNoteSub(Note* note, NoteSubEu* sub, NoteSubAttributes* attrs) {
f32 volRight, volLeft;
s32 smallPanIndex;
u64 pad;
@@ -9,20 +9,20 @@ void Audio_NoteSetVelPanReverb(Note* note, NoteSubEu* sub, Reverb* reverb) {
f32 vel;
u8 pan;
u8 reverbVol;
- ReverbBitsData sp24;
+ StereoData sp24;
s32 stereoHeadsetEffects = note->playbackState.stereoHeadsetEffects;
- vel = reverb->velocity;
- pan = reverb->pan;
- reverbVol = reverb->reverb;
- sp24 = reverb->reverbBits.s;
+ vel = attrs->velocity;
+ pan = attrs->pan;
+ reverbVol = attrs->reverbVol;
+ sp24 = attrs->stereo.s;
sub->bitField0.s = note->noteSubEu.bitField0.s;
sub->bitField1.s = note->noteSubEu.bitField1.s;
sub->sound.samples = note->noteSubEu.sound.samples;
sub->unk_06 = note->noteSubEu.unk_06;
- Audio_NoteSetResamplingRate(sub, reverb->frequency);
+ Audio_NoteSetResamplingRate(sub, attrs->frequency);
pan &= 0x7F;
@@ -38,7 +38,7 @@ void Audio_NoteSetVelPanReverb(Note* note, NoteSubEu* sub, Reverb* reverb) {
sub->headsetPanLeft = gHeadsetPanQuantization[smallPanIndex];
sub->headsetPanRight = gHeadsetPanQuantization[0x3f - smallPanIndex];
- sub->bitField1.s.hasTwoAdpcmParts = true;
+ sub->bitField1.s.usesHeadsetPanEffects2 = true;
volLeft = gHeadsetPanVolume[pan];
volRight = gHeadsetPanVolume[0x7f - pan];
@@ -46,7 +46,7 @@ void Audio_NoteSetVelPanReverb(Note* note, NoteSubEu* sub, Reverb* reverb) {
strongLeft = strongRight = 0;
sub->headsetPanRight = 0;
sub->headsetPanLeft = 0;
- sub->bitField1.s.hasTwoAdpcmParts = false;
+ sub->bitField1.s.usesHeadsetPanEffects2 = false;
volLeft = gStereoPanVolume[pan];
volRight = gStereoPanVolume[0x7f - pan];
@@ -94,10 +94,10 @@ void Audio_NoteSetVelPanReverb(Note* note, NoteSubEu* sub, Reverb* reverb) {
sub->targetVolLeft = (s32)((vel * volLeft) * 4095.999f);
sub->targetVolRight = (s32)((vel * volRight) * 4095.999f);
- sub->unk_2 = reverb->unk_1;
- sub->unk_14 = reverb->unk_10;
- sub->unk_07 = reverb->unk_14;
- sub->unk_10 = reverb->unk_16;
+ sub->unk_2 = attrs->unk_1;
+ sub->filter = attrs->filter;
+ sub->unk_07 = attrs->unk_14;
+ sub->unk_0E = attrs->unk_16;
sub->reverbVol = reverbVol;
}
@@ -105,7 +105,7 @@ void Audio_NoteSetResamplingRate(NoteSubEu* noteSubEu, f32 resamplingRateInput)
f32 resamplingRate = 0.0f;
if (resamplingRateInput < 2.0f) {
- noteSubEu->bitField1.s.isSyntheticWave = false;
+ noteSubEu->bitField1.s.hasTwoAdpcmParts = false;
if (1.99998f < resamplingRateInput) {
resamplingRate = 1.99998f;
@@ -114,7 +114,7 @@ void Audio_NoteSetResamplingRate(NoteSubEu* noteSubEu, f32 resamplingRateInput)
}
} else {
- noteSubEu->bitField1.s.isSyntheticWave = true;
+ noteSubEu->bitField1.s.hasTwoAdpcmParts = true;
if (3.99996f < resamplingRateInput) {
resamplingRate = 1.99998f;
} else {
@@ -154,12 +154,12 @@ void Audio_NoteDisable(Note* note) {
void Audio_ProcessNotes(void) {
s32 pad[2];
- NoteAttributes* attributes;
+ NoteAttributes* attrs;
NoteSubEu* noteSubEu2;
NoteSubEu* noteSubEu;
Note* note;
NotePlaybackState* playbackState;
- Reverb reverb;
+ NoteSubAttributes subAttrs;
u8 bookOffset;
f32 scale;
s32 i;
@@ -248,47 +248,47 @@ void Audio_ProcessNotes(void) {
scale = Audio_AdsrUpdate(&playbackState->adsr);
Audio_NoteVibratoUpdate(note);
- attributes = &playbackState->attributes;
+ attrs = &playbackState->attributes;
if (playbackState->unk_04 == 1 || playbackState->unk_04 == 2) {
- reverb.frequency = attributes->freqScale;
- reverb.velocity = attributes->velocity;
- reverb.pan = attributes->pan;
- reverb.reverb = attributes->reverb;
- reverb.reverbBits = attributes->reverbBits;
- reverb.unk_1 = attributes->unk_1;
- reverb.unk_10 = attributes->unk_10;
- reverb.unk_14 = attributes->unk_4;
- reverb.unk_16 = attributes->unk_6;
+ subAttrs.frequency = attrs->freqScale;
+ subAttrs.velocity = attrs->velocity;
+ subAttrs.pan = attrs->pan;
+ subAttrs.reverbVol = attrs->reverb;
+ subAttrs.stereo = attrs->stereo;
+ subAttrs.unk_1 = attrs->unk_1;
+ subAttrs.filter = attrs->filter;
+ subAttrs.unk_14 = attrs->unk_4;
+ subAttrs.unk_16 = attrs->unk_6;
bookOffset = noteSubEu->bitField1.s.bookOffset;
} else {
SequenceChannelLayer* layer = playbackState->parentLayer;
SequenceChannel* channel = layer->seqChannel;
- reverb.frequency = layer->noteFreqScale;
- reverb.velocity = layer->noteVelocity;
- reverb.pan = layer->notePan;
- if (layer->reverbBits.asByte == 0) {
- reverb.reverbBits = channel->reverbBits;
+ subAttrs.frequency = layer->noteFreqScale;
+ subAttrs.velocity = layer->noteVelocity;
+ subAttrs.pan = layer->notePan;
+ if (layer->stereo.asByte == 0) {
+ subAttrs.stereo = channel->stereo;
} else {
- reverb.reverbBits = layer->reverbBits;
+ subAttrs.stereo = layer->stereo;
}
- reverb.reverb = channel->reverb;
- reverb.unk_1 = channel->unk_0C;
- reverb.unk_10 = channel->unk_CC;
- reverb.unk_14 = channel->unk_0F;
- reverb.unk_16 = channel->unk_20;
+ subAttrs.reverbVol = channel->reverb;
+ subAttrs.unk_1 = channel->unk_0C;
+ subAttrs.filter = channel->filter;
+ subAttrs.unk_14 = channel->unk_0F;
+ subAttrs.unk_16 = channel->unk_20;
bookOffset = channel->bookOffset & 0x7;
if (channel->seqPlayer->muted && (channel->muteBehavior & 8)) {
- reverb.frequency = 0.0f;
- reverb.velocity = 0.0f;
+ subAttrs.frequency = 0.0f;
+ subAttrs.velocity = 0.0f;
}
}
- reverb.frequency *= playbackState->vibratoFreqScale * playbackState->portamentoFreqScale;
- reverb.frequency *= gAudioContext.audioBufferParameters.resampleRate;
- reverb.velocity *= scale;
- Audio_NoteSetVelPanReverb(note, noteSubEu2, &reverb);
+ subAttrs.frequency *= playbackState->vibratoFreqScale * playbackState->portamentoFreqScale;
+ subAttrs.frequency *= gAudioContext.audioBufferParameters.resampleRate;
+ subAttrs.velocity *= scale;
+ Audio_InitNoteSub(note, noteSubEu2, &subAttrs);
noteSubEu->bitField1.s.bookOffset = bookOffset;
skip:;
}
@@ -432,7 +432,7 @@ s32 func_800E7744(s32 instrument, s32 bankId, s32 instId, void* arg3) {
void Audio_SeqChanLayerDecayRelease(SequenceChannelLayer* seqLayer, s32 target) {
Note* note;
- NoteAttributes* attributes;
+ NoteAttributes* attrs;
SequenceChannel* chan;
s32 i;
@@ -447,7 +447,7 @@ void Audio_SeqChanLayerDecayRelease(SequenceChannelLayer* seqLayer, s32 target)
}
note = seqLayer->note;
- attributes = &note->playbackState.attributes;
+ attrs = &note->playbackState.attributes;
if (note->playbackState.wantedParentLayer == seqLayer) {
note->playbackState.wantedParentLayer = NO_LAYER;
@@ -463,37 +463,37 @@ void Audio_SeqChanLayerDecayRelease(SequenceChannelLayer* seqLayer, s32 target)
}
if (note->playbackState.adsr.action.s.state != ADSR_STATE_DECAY) {
- attributes->freqScale = seqLayer->noteFreqScale;
- attributes->velocity = seqLayer->noteVelocity;
- attributes->pan = seqLayer->notePan;
+ attrs->freqScale = seqLayer->noteFreqScale;
+ attrs->velocity = seqLayer->noteVelocity;
+ attrs->pan = seqLayer->notePan;
if (seqLayer->seqChannel != NULL) {
chan = seqLayer->seqChannel;
- attributes->reverb = chan->reverb;
- attributes->unk_1 = chan->unk_0C;
- attributes->unk_10 = chan->unk_CC;
+ attrs->reverb = chan->reverb;
+ attrs->unk_1 = chan->unk_0C;
+ attrs->filter = chan->filter;
- if (attributes->unk_10 != NULL) {
+ if (attrs->filter != NULL) {
for (i = 0; i < 8; i++) {
- attributes->unk_14[i] = attributes->unk_10[i];
+ attrs->filterBuf[i] = attrs->filter[i];
}
- attributes->unk_10 = attributes->unk_14;
+ attrs->filter = attrs->filterBuf;
}
- attributes->unk_6 = chan->unk_20;
- attributes->unk_4 = chan->unk_0F;
+ attrs->unk_6 = chan->unk_20;
+ attrs->unk_4 = chan->unk_0F;
if (chan->seqPlayer->muted && (chan->muteBehavior & 8)) {
note->noteSubEu.bitField0.s.finished = true;
}
- if (seqLayer->reverbBits.asByte == 0) {
- attributes->reverbBits = chan->reverbBits;
+ if (seqLayer->stereo.asByte == 0) {
+ attrs->stereo = chan->stereo;
} else {
- attributes->reverbBits = seqLayer->reverbBits;
+ attrs->stereo = seqLayer->stereo;
}
note->playbackState.priority = chan->someOtherPriority;
} else {
- attributes->reverbBits = seqLayer->reverbBits;
+ attrs->stereo = seqLayer->stereo;
note->playbackState.priority = 1;
}
@@ -767,12 +767,12 @@ void Audio_NoteInitForLayer(Note* note, SequenceChannelLayer* seqLayer) {
sub->sound.audioBankSound = seqLayer->sound;
if (instId >= 0x80 && instId < 0xC0) {
- sub->bitField1.s.bit2 = true;
+ sub->bitField1.s.isSyntheticWave = true;
} else {
- sub->bitField1.s.bit2 = false;
+ sub->bitField1.s.isSyntheticWave = false;
}
- if (sub->bitField1.s.bit2) {
+ if (sub->bitField1.s.isSyntheticWave) {
Audio_BuildSyntheticWave(note, seqLayer, instId);
}
diff --git a/src/code/audio_seqplayer.c b/src/code/audio_seqplayer.c
index c4940a6b6..64b8b9531 100644
--- a/src/code/audio_seqplayer.c
+++ b/src/code/audio_seqplayer.c
@@ -120,7 +120,7 @@ void Audio_SequenceChannelInit(SequenceChannel* seqChannel) {
seqChannel->transposition = 0;
seqChannel->largeNotes = false;
seqChannel->bookOffset = 0;
- seqChannel->reverbBits.asByte = 0;
+ seqChannel->stereo.asByte = 0;
seqChannel->changes.asByte = 0xFF;
seqChannel->scriptState.depth = 0;
seqChannel->newPan = 0x40;
@@ -144,7 +144,7 @@ void Audio_SequenceChannelInit(SequenceChannel* seqChannel) {
seqChannel->vibratoRateChangeDelay = 0;
seqChannel->vibratoExtentChangeDelay = 0;
seqChannel->vibratoDelay = 0;
- seqChannel->unk_CC = NULL;
+ seqChannel->filter = NULL;
seqChannel->unk_20 = 0;
seqChannel->unk_0F = 0;
seqChannel->volume = 1.0f;
@@ -186,7 +186,7 @@ s32 Audio_SeqChannelSetLayer(SequenceChannel* seqChannel, s32 layerIdx) {
layer->ignoreDrumPan = false;
layer->bit1 = false;
layer->notePropertiesNeedInit = false;
- layer->reverbBits.asByte = 0;
+ layer->stereo.asByte = 0;
layer->portamento.mode = 0;
layer->scriptState.depth = 0;
layer->noteDuration = 0x80;
@@ -427,8 +427,8 @@ void func_800E9ED8(SequenceChannelLayer* layer) {
}
s32 func_800E9F64(SequenceChannelLayer* layer, s32 arg1) {
- if (!layer->stopSomething && layer->sound != NULL && layer->sound->sample->bits4 == 2 &&
- layer->sound->sample->bits2 != 0) {
+ if (!layer->stopSomething && layer->sound != NULL && layer->sound->sample->codec == 2 &&
+ layer->sound->sample->medium != 0) {
layer->stopSomething = true;
return -1;
}
@@ -587,7 +587,7 @@ s32 func_800EA0C0(SequenceChannelLayer* layer) {
break;
case 0xCD:
- layer->reverbBits.asByte = Audio_M64ReadU8(state);
+ layer->stereo.asByte = Audio_M64ReadU8(state);
break;
case 0xCE: {
@@ -1212,7 +1212,7 @@ void Audio_SequenceChannelProcessScript(SequenceChannel* channel) {
} else {
channel->stereoHeadsetEffects = false;
}
- channel->reverbBits.asByte = command & 0x7F;
+ channel->stereo.asByte = command & 0x7F;
break;
case 0xD1:
command = (u8)parameters[0];
@@ -1275,7 +1275,7 @@ void Audio_SequenceChannelProcessScript(SequenceChannel* channel) {
channel->vibratoRateTarget = 0;
channel->vibratoRateStart = 0;
channel->vibratoRateChangeDelay = 0;
- channel->unk_CC = NULL;
+ channel->filter = NULL;
channel->unk_0C = 0;
channel->adsr.sustain = 0;
channel->velocityRandomVariance = 0;
@@ -1295,18 +1295,18 @@ void Audio_SequenceChannelProcessScript(SequenceChannel* channel) {
case 0xB0:
offset = (u16)parameters[0];
data = seqPlayer->seqData + offset;
- channel->unk_CC = (s16*)data;
+ channel->filter = (s16*)data;
break;
case 0xB1:
- channel->unk_CC = NULL;
+ channel->filter = NULL;
break;
case 0xB3:
command = parameters[0];
- if (channel->unk_CC != NULL) {
+ if (channel->filter != NULL) {
lowBits = (command >> 4) & 0xF;
command &= 0xF;
- func_800DF688(channel->unk_CC, lowBits, command);
+ func_800DF688(channel->filter, lowBits, command);
}
break;
case 0xB2:
@@ -1676,11 +1676,11 @@ void Audio_SequencePlayerProcessSequence(SequencePlayer* seqPlayer) {
case 0xC4:
command = Audio_M64ReadU8(seqScript);
if (command == 0xFF) {
- command = seqPlayer->seqVariationEu;
+ command = seqPlayer->playerIndex;
}
commandLow = Audio_M64ReadU8(seqScript);
func_800E20D4(command, commandLow, 0);
- if (command == (u8)seqPlayer->seqVariationEu) {
+ if (command == (u8)seqPlayer->playerIndex) {
return;
}
break;
diff --git a/src/code/audio_synthesis.c b/src/code/audio_synthesis.c
index 8e07ef0d8..cd686bf0b 100644
--- a/src/code/audio_synthesis.c
+++ b/src/code/audio_synthesis.c
@@ -1,25 +1,46 @@
#include "ultra64.h"
#include "global.h"
-Acmd* AudioSynth_LoadRingBuffer(Acmd* arg0, u16 arg1, u16 arg2, s32 arg3, SynthesisReverb* arg4);
-Acmd* AudioSynth_SaveBuffer(Acmd* arg0, u16 arg1, u16 arg2, s32 arg3, s16* arg4);
-Acmd* AudioSynth_SaveRingBuffer(Acmd* arg0, u16 arg1, u16 arg2, s32 arg3, SynthesisReverb* arg4);
-Acmd* func_800DC384(s16* aiBuf, s32 aiBufLen, Acmd* cmd, s32 updateIdx);
-Acmd* func_800DC910(s32 noteIdx, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s16* aiBuf, s32 aiBufLen,
- Acmd* cmd, s32 updateIdx);
-Acmd* func_800DD9F4(Acmd* arg0, NoteSubEu* arg1, NoteSynthesisState* arg2, s32 arg3);
-Acmd* func_800DDB64(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s32 arg3, s32 arg4, s32 arg5);
-Acmd* func_800DD6CC(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s32 aiBufLen, u16, s32, s32);
-Acmd* func_800DD62C(Acmd* cmd, NoteSynthesisState* synthState, s32 count, u16 pitch, u16 inpDmem, s32 resampleFlags);
-extern s16 D_8012FBAA[];
+#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
+#define DMEM_SCRATCH2 0x760 // = DMEM_TEMP + DEFAULT_LEN_2CH + a bit more
+#define DMEM_COMPRESSED_ADPCM_DATA 0x940 // = DMEM_LEFT_CH
+#define DMEM_LEFT_CH 0x940
+#define DMEM_RIGHT_CH 0xAE0
+#define DMEM_WET_TEMP 0x3E0
+#define DMEM_WET_SCRATCH 0x720 // = DMEM_WET_TEMP + DEFAULT_LEN_2CH
+#define DMEM_WET_LEFT_CH 0xC80
+#define DMEM_WET_RIGHT_CH 0xE20 // = DMEM_WET_LEFT_CH + DEFAULT_LEN_1CH
+
+Acmd* AudioSynth_LoadRingBufferPart(Acmd* cmd, u16 dmem, u16 startPos, s32 length, SynthesisReverb* reverb);
+Acmd* AudioSynth_SaveBufferOffset(Acmd* cmd, u16 dmem, u16 offset, s32 length, s16* buf);
+Acmd* AudioSynth_SaveRingBufferPart(Acmd* cmd, u16 dmem, u16 startPos, s32 length, SynthesisReverb* reverb);
+Acmd* AudioSynth_DoOneAudioUpdate(s16* aiBuf, s32 aiBufLen, Acmd* cmd, s32 updateIndex);
+Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s16* aiBuf,
+ s32 aiBufLen, Acmd* cmd, s32 updateIndex);
+Acmd* AudioSynth_LoadWaveSamples(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s32 nSamplesToLoad);
+Acmd* AudioSynth_NoteApplyHeadsetPanEffects(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s32 bufLen,
+ s32 flags, s32 side);
+Acmd* AudioSynth_ProcessEnvelope(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s32 aiBufLen,
+ u16 inBuf, s32 headsetPanSettings, s32 flags);
+Acmd* AudioSynth_FinalResample(Acmd* cmd, NoteSynthesisState* synthState, s32 count, u16 pitch, u16 inpDmem,
+ s32 resampleFlags);
-// 4 nops, part of ucode_disas?
-#pragma GLOBAL_ASM("asm/non_matchings/code/audio_synthesis/pad_800DACB0.s")
+extern s16 D_8012FBAA[];
+extern u32 D_801304A0;
+extern u32 D_801304A4;
+extern u32 D_801304A8;
+extern u32 D_801304AC;
+extern u8 D_801304C0[];
-void AudioSynth_InitNextRingBuf(s32 chunkSize, s32 bufIdx, s32 reverbIdx) {
+void AudioSynth_InitNextRingBuf(s32 chunkLen, s32 bufIndex, s32 reverbIndex) {
ReverbRingBufferItem* bufItem;
s32 pad[3];
- SynthesisReverb* reverb = &gAudioContext.synthesisReverbs[reverbIdx];
+ SynthesisReverb* reverb = &gAudioContext.synthesisReverbs[reverbIndex];
s32 temp_a0_2;
s32 temp_a0_4;
s32 sampleCnt;
@@ -29,9 +50,8 @@ void AudioSynth_InitNextRingBuf(s32 chunkSize, s32 bufIdx, s32 reverbIdx) {
if (reverb->downsampleRate >= 2) {
if (reverb->framesToIgnore == 0) {
- bufItem = &reverb->items[reverb->curFrame][bufIdx];
- // inval dcache
- Audio_osInvalDCache(bufItem->toDownsampleLeft, 0x340);
+ bufItem = &reverb->items[reverb->curFrame][bufIndex];
+ Audio_osInvalDCache(bufItem->toDownsampleLeft, DEFAULT_LEN_2CH);
for (j = 0, i = 0; i < bufItem->lengthA / 2; j += reverb->downsampleRate, i++) {
reverb->leftRingBuf[bufItem->startPos + i] = bufItem->toDownsampleLeft[j];
@@ -45,8 +65,8 @@ void AudioSynth_InitNextRingBuf(s32 chunkSize, s32 bufIdx, s32 reverbIdx) {
}
}
- bufItem = &reverb->items[reverb->curFrame][bufIdx];
- sampleCnt = chunkSize / reverb->downsampleRate;
+ bufItem = &reverb->items[reverb->curFrame][bufIndex];
+ sampleCnt = chunkLen / reverb->downsampleRate;
extraSamples = (sampleCnt + reverb->nextRingBufPos) - reverb->bufSizePerChan;
temp_a0_2 = reverb->nextRingBufPos;
if (extraSamples < 0) {
@@ -62,15 +82,15 @@ void AudioSynth_InitNextRingBuf(s32 chunkSize, s32 bufIdx, s32 reverbIdx) {
}
bufItem->numSamplesAfterDownsampling = sampleCnt;
- bufItem->chunkLen = chunkSize;
+ bufItem->chunkLen = chunkLen;
if (reverb->unk_14 != 0) {
temp_a0_4 = reverb->unk_14 + temp_a0_2;
if (temp_a0_4 >= reverb->bufSizePerChan) {
temp_a0_4 -= reverb->bufSizePerChan;
}
- bufItem = &reverb->items2[reverb->curFrame][bufIdx];
- sampleCnt = chunkSize / reverb->downsampleRate;
+ bufItem = &reverb->items2[reverb->curFrame][bufIndex];
+ sampleCnt = chunkLen / reverb->downsampleRate;
extraSamples = (temp_a0_4 + sampleCnt) - reverb->bufSizePerChan;
if (extraSamples < 0) {
bufItem->lengthA = sampleCnt * 2;
@@ -82,24 +102,24 @@ void AudioSynth_InitNextRingBuf(s32 chunkSize, s32 bufIdx, s32 reverbIdx) {
bufItem->startPos = temp_a0_4;
}
bufItem->numSamplesAfterDownsampling = sampleCnt;
- bufItem->chunkLen = chunkSize;
+ bufItem->chunkLen = chunkLen;
}
}
void func_800DB03C(s32 arg0) {
- NoteSubEu* temp_v0_2;
NoteSubEu* subEu;
- s32 t;
+ NoteSubEu* subEu2;
+ s32 baseIndex;
s32 i;
- t = gAudioContext.maxSimultaneousNotes * arg0;
+ baseIndex = gAudioContext.maxSimultaneousNotes * arg0;
for (i = 0; i < gAudioContext.maxSimultaneousNotes; i++) {
subEu = &gAudioContext.notes[i].noteSubEu;
- temp_v0_2 = &gAudioContext.noteSubsEu[t + i];
+ subEu2 = &gAudioContext.noteSubsEu[baseIndex + i];
if (subEu->bitField0.s.enabled) {
- subEu->bitField0.s.needsInit = 0;
+ subEu->bitField0.s.needsInit = false;
} else {
- temp_v0_2->bitField0.s.enabled = 0;
+ subEu2->bitField0.s.enabled = false;
}
subEu->unk_06 = 0;
@@ -107,7 +127,7 @@ void func_800DB03C(s32 arg0) {
}
Acmd* AudioSynth_Update(Acmd* cmdStart, s32* cmdCnt, s16* aiStart, s32 aiBufLen) {
- s32 chunkSize;
+ s32 chunkLen;
s16* aiBufP;
Acmd* cmdP;
s32 i;
@@ -121,27 +141,28 @@ Acmd* AudioSynth_Update(Acmd* cmdStart, s32* cmdCnt, s16* aiStart, s32 aiBufLen)
}
aiBufP = aiStart;
- gAudioContext.unk_0x10 = 0;
+ gAudioContext.curLoadedBook = NULL;
for (i = gAudioContext.audioBufferParameters.updatesPerFrame; i > 0; i--) {
if (i == 1) {
- chunkSize = aiBufLen;
+ chunkLen = aiBufLen;
} else if ((aiBufLen / i) >= gAudioContext.audioBufferParameters.samplesPerUpdateMax) {
- chunkSize = gAudioContext.audioBufferParameters.samplesPerUpdateMax;
+ chunkLen = gAudioContext.audioBufferParameters.samplesPerUpdateMax;
} else if (gAudioContext.audioBufferParameters.samplesPerUpdateMin >= (aiBufLen / i)) {
- chunkSize = gAudioContext.audioBufferParameters.samplesPerUpdateMin;
+ chunkLen = gAudioContext.audioBufferParameters.samplesPerUpdateMin;
} else {
- chunkSize = gAudioContext.audioBufferParameters.samplesPerUpdate;
+ chunkLen = gAudioContext.audioBufferParameters.samplesPerUpdate;
}
for (j = 0; j < gAudioContext.numSynthesisReverbs; j++) {
if (gAudioContext.synthesisReverbs[j].useReverb) {
- AudioSynth_InitNextRingBuf(chunkSize, gAudioContext.audioBufferParameters.updatesPerFrame - i, j);
+ AudioSynth_InitNextRingBuf(chunkLen, gAudioContext.audioBufferParameters.updatesPerFrame - i, j);
}
}
- cmdP = func_800DC384(aiBufP, chunkSize, cmdP, gAudioContext.audioBufferParameters.updatesPerFrame - i);
- aiBufLen -= chunkSize;
- aiBufP += chunkSize * 2;
+ cmdP = AudioSynth_DoOneAudioUpdate(aiBufP, chunkLen, cmdP,
+ gAudioContext.audioBufferParameters.updatesPerFrame - i);
+ aiBufLen -= chunkLen;
+ aiBufP += chunkLen * 2;
}
for (j = 0; j < gAudioContext.numSynthesisReverbs; j++) {
@@ -155,12 +176,12 @@ Acmd* AudioSynth_Update(Acmd* cmdStart, s32* cmdCnt, s16* aiStart, s32 aiBufLen)
return cmdP;
}
-void func_800DB2C0(s32 arg0, s32 arg1) {
+void func_800DB2C0(s32 updateIndexStart, s32 noteIndex) {
NoteSubEu* temp_v1;
s32 i;
- for (i = arg0 + 1; i < gAudioContext.audioBufferParameters.updatesPerFrame; i++) {
- temp_v1 = &gAudioContext.noteSubsEu[(gAudioContext.maxSimultaneousNotes * i) + arg1];
+ for (i = updateIndexStart + 1; i < gAudioContext.audioBufferParameters.updatesPerFrame; i++) {
+ temp_v1 = &gAudioContext.noteSubsEu[(gAudioContext.maxSimultaneousNotes * i) + noteIndex];
if (!temp_v1->bitField0.s.needsInit) {
temp_v1->bitField0.s.enabled = 0;
} else {
@@ -169,128 +190,132 @@ void func_800DB2C0(s32 arg0, s32 arg1) {
}
}
-Acmd* func_800DB330(Acmd* arg0, SynthesisReverb* reverb, s16 arg2) {
- ReverbRingBufferItem* bufItem = &reverb->items[reverb->curFrame][arg2];
+Acmd* AudioSynth_LoadRingBuffer1AtTemp(Acmd* cmd, SynthesisReverb* reverb, s16 bufIndex) {
+ ReverbRingBufferItem* bufItem = &reverb->items[reverb->curFrame][bufIndex];
- arg0 = AudioSynth_LoadRingBuffer(arg0, 0x3E0, bufItem->startPos, bufItem->lengthA, reverb);
+ cmd = AudioSynth_LoadRingBufferPart(cmd, DMEM_WET_TEMP, bufItem->startPos, bufItem->lengthA, reverb);
if (bufItem->lengthB != 0) {
- arg0 = AudioSynth_LoadRingBuffer(arg0, bufItem->lengthA + 0x3E0, 0, bufItem->lengthB, reverb);
+ // Ring buffer wrapped
+ cmd = AudioSynth_LoadRingBufferPart(cmd, DMEM_WET_TEMP + bufItem->lengthA, 0, bufItem->lengthB, reverb);
}
- return arg0;
+ return cmd;
}
-Acmd* func_800DB3D8(Acmd* arg0, SynthesisReverb* arg1, s16 arg2) {
- ReverbRingBufferItem* temp_v1;
+Acmd* AudioSynth_SaveRingBuffer1AtTemp(Acmd* cmd, SynthesisReverb* reverb, s16 bufIndex) {
+ ReverbRingBufferItem* bufItem = &reverb->items[reverb->curFrame][bufIndex];
- temp_v1 = &arg1->items[arg1->curFrame][arg2];
- arg0 = AudioSynth_SaveRingBuffer(arg0, 0x3E0, temp_v1->startPos, temp_v1->lengthA, arg1);
- if (temp_v1->lengthB != 0) {
- arg0 = AudioSynth_SaveRingBuffer(arg0, temp_v1->lengthA + 0x3E0, 0, temp_v1->lengthB, arg1);
+ cmd = AudioSynth_SaveRingBufferPart(cmd, DMEM_WET_TEMP, bufItem->startPos, bufItem->lengthA, reverb);
+ if (bufItem->lengthB != 0) {
+ // Ring buffer wrapped
+ cmd = AudioSynth_SaveRingBufferPart(cmd, DMEM_WET_TEMP + bufItem->lengthA, 0, bufItem->lengthB, reverb);
}
- return arg0;
+ return cmd;
}
-Acmd* func_800DB480(Acmd* arg0, SynthesisReverb* arg1) {
- void* temp_a2;
- void* temp_a3;
-
- aDMEMMove(arg0++, 0xC80, 0x720, 0x1A0);
- aMix(arg0++, 0x1A, arg1->unk_10, 0xE20, 0xC80);
- aMix(arg0++, 0x1A, arg1->unk_12, 0x720, 0xE20);
- return arg0;
+Acmd* AudioSynth_LeakReverb(Acmd* cmd, SynthesisReverb* reverb) {
+ // Leak some audio from the left reverb channel into the right reverb channel and vice versa (pan)
+ aDMEMMove(cmd++, DMEM_WET_LEFT_CH, DMEM_WET_SCRATCH, DEFAULT_LEN_1CH);
+ aMix(cmd++, 0x1A, reverb->leakRtl, DMEM_WET_RIGHT_CH, DMEM_WET_LEFT_CH);
+ aMix(cmd++, 0x1A, reverb->leakLtr, DMEM_WET_SCRATCH, DMEM_WET_RIGHT_CH);
+ return cmd;
}
-Acmd* func_800DB4E4(Acmd* arg0, s32 arg1, SynthesisReverb* arg2, s16 arg3) {
- ReverbRingBufferItem* temp_v1;
- s16 temp_t0_2;
- s16 temp_t1_2;
+Acmd* func_800DB4E4(Acmd* cmd, s32 arg1, SynthesisReverb* reverb, s16 arg3) {
+ ReverbRingBufferItem* item = &reverb->items[reverb->curFrame][arg3];
+ s16 offsetA;
+ s16 offsetB;
- temp_v1 = &arg2->items[arg2->curFrame][arg3];
- temp_t0_2 = (temp_v1->startPos & 7) * 2;
- temp_t1_2 = ALIGN16(temp_t0_2 + temp_v1->lengthA);
- arg0 = AudioSynth_LoadRingBuffer(arg0, 0x3E0, temp_v1->startPos - (temp_t0_2 / 2), 0x1A0, arg2);
- if (temp_v1->lengthB != 0) {
- arg0 = AudioSynth_LoadRingBuffer(arg0, temp_t1_2 + 0x3E0, 0, 0x1A0 - temp_t1_2, arg2);
+ offsetA = (item->startPos & 7) * 2;
+ offsetB = ALIGN16(offsetA + item->lengthA);
+ cmd = AudioSynth_LoadRingBufferPart(cmd, DMEM_WET_TEMP, item->startPos - (offsetA / 2), DEFAULT_LEN_1CH, reverb);
+ if (item->lengthB != 0) {
+ // Ring buffer wrapped
+ cmd = AudioSynth_LoadRingBufferPart(cmd, DMEM_WET_TEMP + offsetB, 0, DEFAULT_LEN_1CH - offsetB, reverb);
}
- aSetBuffer(arg0++, 0, temp_t0_2 + 0x3E0, 0xC80, arg1 * 2);
- aResample(arg0++, arg2->resampleFlags, arg2->unk_0E, arg2->unk_30);
- aSetBuffer(arg0++, 0, temp_t0_2 + 0x580, 0xE20, arg1 * 2);
- aResample(arg0++, arg2->resampleFlags, arg2->unk_0E, arg2->unk_34);
- return arg0;
+ aSetBuffer(cmd++, 0, DMEM_WET_TEMP + offsetA, DMEM_WET_LEFT_CH, arg1 * 2);
+ aResample(cmd++, reverb->resampleFlags, reverb->unk_0E, reverb->unk_30);
+ aSetBuffer(cmd++, 0, DMEM_WET_TEMP + DEFAULT_LEN_1CH + offsetA, DMEM_WET_RIGHT_CH, arg1 * 2);
+ aResample(cmd++, reverb->resampleFlags, reverb->unk_0E, reverb->unk_34);
+ return cmd;
}
-Acmd* func_800DB680(Acmd* cmd, SynthesisReverb* reverb, s16 bufIdx) {
- ReverbRingBufferItem* bufItem = &reverb->items[reverb->curFrame][bufIdx];
+Acmd* func_800DB680(Acmd* cmd, SynthesisReverb* reverb, s16 bufIndex) {
+ ReverbRingBufferItem* bufItem = &reverb->items[reverb->curFrame][bufIndex];
- aSetBuffer(cmd++, 0, 0xC80, 0x720, bufItem->unk_18 * 2);
+ aSetBuffer(cmd++, 0, DMEM_WET_LEFT_CH, DMEM_WET_SCRATCH, bufItem->unk_18 * 2);
aResample(cmd++, reverb->resampleFlags, bufItem->unk_16, reverb->unk_38);
- cmd = AudioSynth_SaveBuffer(cmd, 0x720, bufItem->startPos, bufItem->lengthA, reverb->leftRingBuf);
+ cmd = AudioSynth_SaveBufferOffset(cmd, DMEM_WET_SCRATCH, bufItem->startPos, bufItem->lengthA, reverb->leftRingBuf);
if (bufItem->lengthB != 0) {
- cmd = AudioSynth_SaveBuffer(cmd, bufItem->lengthA + 0x720, 0, bufItem->lengthB, reverb->leftRingBuf);
+ // Ring buffer wrapped
+ cmd = AudioSynth_SaveBufferOffset(cmd, DMEM_WET_SCRATCH + bufItem->lengthA, 0, bufItem->lengthB,
+ reverb->leftRingBuf);
}
- aSetBuffer(cmd++, 0, 0xE20, 0x720, bufItem->unk_18 * 2);
+ aSetBuffer(cmd++, 0, DMEM_WET_RIGHT_CH, DMEM_WET_SCRATCH, bufItem->unk_18 * 2);
aResample(cmd++, reverb->resampleFlags, bufItem->unk_16, reverb->unk_3C);
- cmd = AudioSynth_SaveBuffer(cmd, 0x720, bufItem->startPos, bufItem->lengthA, reverb->rightRingBuf);
+ cmd = AudioSynth_SaveBufferOffset(cmd, DMEM_WET_SCRATCH, bufItem->startPos, bufItem->lengthA, reverb->rightRingBuf);
if (bufItem->lengthB != 0) {
- cmd = AudioSynth_SaveBuffer(cmd, bufItem->lengthA + 0x720, 0, bufItem->lengthB, reverb->rightRingBuf);
+ // Ring buffer wrapped
+ cmd = AudioSynth_SaveBufferOffset(cmd, DMEM_WET_SCRATCH + bufItem->lengthA, 0, bufItem->lengthB,
+ reverb->rightRingBuf);
}
return cmd;
}
-Acmd* func_800DB828(Acmd* arg0, s32 arg1, SynthesisReverb* arg2, s16 arg3) {
- ReverbRingBufferItem* temp_v1;
- s16 temp_t1;
- s16 temp_t2_2;
+Acmd* func_800DB828(Acmd* cmd, s32 arg1, SynthesisReverb* reverb, s16 arg3) {
+ ReverbRingBufferItem* item = &reverb->items[reverb->curFrame][arg3];
+ s16 offsetA;
+ s16 offsetB;
- temp_v1 = &arg2->items[arg2->curFrame][arg3];
- temp_v1->unk_14 = (temp_v1->unk_18 << 0xF) / arg1;
- temp_t1 = (temp_v1->startPos & 7) * 2;
- temp_v1->unk_16 = (arg1 << 0xF) / temp_v1->unk_18;
- temp_t2_2 = ALIGN16(temp_t1 + temp_v1->lengthA);
- arg0 = AudioSynth_LoadRingBuffer(arg0, 0x3E0, temp_v1->startPos - (temp_t1 / 2), 0x1A0, arg2);
- if (temp_v1->lengthB != 0) {
- arg0 = AudioSynth_LoadRingBuffer(arg0, temp_t2_2 + 0x3E0, 0, 0x1A0 - temp_t2_2, arg2);
+ item->unk_14 = (item->unk_18 << 0xF) / arg1;
+ offsetA = (item->startPos & 7) * 2;
+ item->unk_16 = (arg1 << 0xF) / item->unk_18;
+ offsetB = ALIGN16(offsetA + item->lengthA);
+ cmd = AudioSynth_LoadRingBufferPart(cmd, DMEM_WET_TEMP, item->startPos - (offsetA / 2), DEFAULT_LEN_1CH, reverb);
+ if (item->lengthB != 0) {
+ // Ring buffer wrapped
+ cmd = AudioSynth_LoadRingBufferPart(cmd, DMEM_WET_TEMP + offsetB, 0, DEFAULT_LEN_1CH - offsetB, reverb);
}
- aSetBuffer(arg0++, 0, temp_t1 + 0x3E0, 0xC80, arg1 * 2);
- aResample(arg0++, arg2->resampleFlags, temp_v1->unk_14, arg2->unk_30);
- aSetBuffer(arg0++, 0, temp_t1 + 0x580, 0xE20, arg1 * 2);
- aResample(arg0++, arg2->resampleFlags, temp_v1->unk_14, arg2->unk_34);
- return arg0;
+ aSetBuffer(cmd++, 0, DMEM_WET_TEMP + offsetA, DMEM_WET_LEFT_CH, arg1 * 2);
+ aResample(cmd++, reverb->resampleFlags, item->unk_14, reverb->unk_30);
+ aSetBuffer(cmd++, 0, DMEM_WET_TEMP + DEFAULT_LEN_1CH + offsetA, DMEM_WET_RIGHT_CH, arg1 * 2);
+ aResample(cmd++, reverb->resampleFlags, item->unk_14, reverb->unk_34);
+ return cmd;
}
-Acmd* func_800DBA40(Acmd* arg0, s32 arg1, SynthesisReverb* arg2) {
- if (arg2->unk_270 != NULL) {
- aFilter(arg0++, 2, arg1, arg2->unk_270);
- aFilter(arg0++, arg2->resampleFlags, 0xC80, arg2->unk_278);
+Acmd* AudioSynth_FilterReverb(Acmd* cmd, s32 count, SynthesisReverb* reverb) {
+ // Apply a filter (convolution) to each reverb channel.
+ if (reverb->filterLeft != NULL) {
+ aFilter(cmd++, 2, count, reverb->filterLeft);
+ aFilter(cmd++, reverb->resampleFlags, DMEM_WET_LEFT_CH, reverb->filterLeftState);
}
- if (arg2->unk_274 != NULL) {
- aFilter(arg0++, 2, arg1, arg2->unk_274);
- aFilter(arg0++, arg2->resampleFlags, 0xE20, arg2->unk_27C);
+ if (reverb->filterRight != NULL) {
+ aFilter(cmd++, 2, count, reverb->filterRight);
+ aFilter(cmd++, reverb->resampleFlags, DMEM_WET_RIGHT_CH, reverb->filterRightState);
}
- return arg0;
+ return cmd;
}
-Acmd* func_800DBAE8(Acmd* arg0, SynthesisReverb* arg1, s32 arg2) {
+Acmd* AudioSynth_MaybeMixRingBuffer1(Acmd* cmd, SynthesisReverb* reverb, s32 arg2) {
SynthesisReverb* temp_a3;
- temp_a3 = &gAudioContext.synthesisReverbs[arg1->unk_05];
+ temp_a3 = &gAudioContext.synthesisReverbs[reverb->unk_05];
if (temp_a3->downsampleRate == 1) {
- arg0 = func_800DB330(arg0, temp_a3, arg2);
- aMix(arg0++, 0x34, arg1->unk_08, 0xC80, 0x3E0);
- arg0 = func_800DB3D8(arg0, temp_a3, arg2);
+ cmd = AudioSynth_LoadRingBuffer1AtTemp(cmd, temp_a3, arg2);
+ aMix(cmd++, 0x34, reverb->unk_08, DMEM_WET_LEFT_CH, DMEM_WET_TEMP);
+ cmd = AudioSynth_SaveRingBuffer1AtTemp(cmd, temp_a3, arg2);
}
- return arg0;
+ return cmd;
}
void func_800DBB94(void) {
}
-void func_800DBB9C(Acmd* arg0, s32 arg1, s32 arg2) {
- aClearBuffer(arg0, arg1, arg2);
+void AudioSynth_ClearBuffer(Acmd* cmd, s32 arg1, s32 arg2) {
+ aClearBuffer(cmd, arg1, arg2);
}
void func_800DBBBC(void) {
@@ -302,8 +327,8 @@ void func_800DBBC4(void) {
void func_800DBBCC(void) {
}
-void func_800DBBD4(Acmd* arg0, s32 arg1, s32 arg2, s32 arg3, s32 arg4) {
- aMix(arg0, arg1, arg2, arg3, arg4);
+void AudioSynth_Mix(Acmd* cmd, s32 arg1, s32 arg2, s32 arg3, s32 arg4) {
+ aMix(cmd, arg1, arg2, arg3, arg4);
}
void func_800DBC08(void) {
@@ -315,8 +340,8 @@ void func_800DBC10(void) {
void func_800DBC18(void) {
}
-void AudioSynth_SetBuffer(Acmd* cmd, s32 flag, s32 dmemIn, s32 dmemOut, s32 count) {
- aSetBuffer(cmd, flag, dmemIn, dmemOut, count);
+void AudioSynth_SetBuffer(Acmd* cmd, s32 flags, s32 dmemIn, s32 dmemOut, s32 count) {
+ aSetBuffer(cmd, flags, dmemIn, dmemOut, count);
}
void func_800DBC54(void) {
@@ -348,19 +373,19 @@ void AudioSynth_InterL(Acmd* cmd, s32 dmemIn, s32 dmemOut, s32 count) {
cmd->words.w1 = _SHIFTL(dmemIn, 16, 16) | _SHIFTL(dmemOut, 0, 16);
}
-void func_800DBCD4(Acmd* arg0, s32 arg1, s32 arg2, s32 arg3, s32 arg4) {
- aEnvSetup1(arg0, arg1, arg2, arg3, arg4);
+void AudioSynth_EnvSetup1(Acmd* cmd, s32 arg1, s32 arg2, s32 arg3, s32 arg4) {
+ aEnvSetup1(cmd, arg1, arg2, arg3, arg4);
}
void func_800DBD08(void) {
}
-void func_800DBD10(Acmd* arg0, s32 arg1, s32 arg2, s32 arg3) {
- aLoadBuffer(arg0, arg3, arg1, arg2);
+void AudioSynth_LoadBuffer(Acmd* cmd, s32 arg1, s32 arg2, s32 arg3) {
+ aLoadBuffer(cmd, arg3, arg1, arg2);
}
-void func_800DBD38(Acmd* arg0, s32 arg1, s32 arg2, s32 arg3) {
- aSaveBuffer(arg0, arg1, arg3, arg2);
+void AudioSynth_SaveBuffer(Acmd* cmd, s32 arg1, s32 arg2, s32 arg3) {
+ aSaveBuffer(cmd, arg1, arg3, arg2);
}
void AudioSynth_EnvSetup2(Acmd* cmd, s32 volLeft, s32 volRight) {
@@ -377,8 +402,8 @@ void func_800DBD84(void) {
void func_800DBD8C(void) {
}
-void AudioSynth_UnkCmd17(Acmd* arg0, s32 arg1, s32 arg2) {
- aUnkCmd17(arg0, arg1, arg2);
+void AudioSynth_S8Dec(Acmd* cmd, s32 flags, s16* state) {
+ aS8Dec(cmd, flags, state);
}
void AudioSynth_HiLoGain(Acmd* cmd, s32 gain, s32 dmemIn, s32 dmemOut, s32 count) {
@@ -425,58 +450,63 @@ void AudioSynth_SetFilterCount(Acmd* cmd, s32 count, s32 addr) {
aFilter(cmd, 2, count, addr);
}
-Acmd* func_800DBEB0(Acmd* cmd, s32 arg1, SynthesisReverb* reverb, s16 arg3) {
- ReverbRingBufferItem* ringBufferItem = &reverb->items[reverb->curFrame][arg3];
+Acmd* AudioSynth_LoadRingBuffer1(Acmd* cmd, s32 arg1, SynthesisReverb* reverb, s16 bufIndex) {
+ ReverbRingBufferItem* ringBufferItem = &reverb->items[reverb->curFrame][bufIndex];
- cmd = AudioSynth_LoadRingBuffer(cmd, 0xC80, ringBufferItem->startPos, ringBufferItem->lengthA, reverb);
+ cmd =
+ AudioSynth_LoadRingBufferPart(cmd, DMEM_WET_LEFT_CH, ringBufferItem->startPos, ringBufferItem->lengthA, reverb);
if (ringBufferItem->lengthB != 0) {
- cmd = AudioSynth_LoadRingBuffer(cmd, ringBufferItem->lengthA + 0xC80, 0, ringBufferItem->lengthB, reverb);
+ // Ring buffer wrapped
+ cmd = AudioSynth_LoadRingBufferPart(cmd, DMEM_WET_LEFT_CH + ringBufferItem->lengthA, 0, ringBufferItem->lengthB,
+ reverb);
}
return cmd;
}
-Acmd* func_800DBF5C(Acmd* cmd, s32 arg1, SynthesisReverb* reverb, s16 bufIdx) {
- ReverbRingBufferItem* bufItem = &reverb->items2[reverb->curFrame][bufIdx];
+Acmd* AudioSynth_LoadRingBuffer2(Acmd* cmd, s32 arg1, SynthesisReverb* reverb, s16 bufIndex) {
+ ReverbRingBufferItem* bufItem = &reverb->items2[reverb->curFrame][bufIndex];
- cmd = AudioSynth_LoadRingBuffer(cmd, 0xC80, bufItem->startPos, bufItem->lengthA, reverb);
+ cmd = AudioSynth_LoadRingBufferPart(cmd, DMEM_WET_LEFT_CH, bufItem->startPos, bufItem->lengthA, reverb);
if (bufItem->lengthB != 0) {
- cmd = AudioSynth_LoadRingBuffer(cmd, bufItem->lengthA + 0xC80, 0, bufItem->lengthB, reverb);
+ // Ring buffer wrapped
+ cmd = AudioSynth_LoadRingBufferPart(cmd, DMEM_WET_LEFT_CH + bufItem->lengthA, 0, bufItem->lengthB, reverb);
}
return cmd;
}
-Acmd* AudioSynth_LoadRingBuffer(Acmd* cmd, u16 dmem, u16 startPos, s32 length, SynthesisReverb* reverb) {
+Acmd* AudioSynth_LoadRingBufferPart(Acmd* cmd, u16 dmem, u16 startPos, s32 length, SynthesisReverb* reverb) {
aLoadBuffer(cmd++, &reverb->leftRingBuf[startPos], dmem, length);
- aLoadBuffer(cmd++, &reverb->rightRingBuf[startPos], dmem + 0x1A0, length);
+ aLoadBuffer(cmd++, &reverb->rightRingBuf[startPos], dmem + DEFAULT_LEN_1CH, length);
return cmd;
}
-Acmd* AudioSynth_SaveRingBuffer(Acmd* cmd, u16 dmem, u16 startPos, s32 length, SynthesisReverb* reverb) {
+Acmd* AudioSynth_SaveRingBufferPart(Acmd* cmd, u16 dmem, u16 startPos, s32 length, SynthesisReverb* reverb) {
aSaveBuffer(cmd++, dmem, &reverb->leftRingBuf[startPos], length);
- aSaveBuffer(cmd++, dmem + 0x1A0, &reverb->rightRingBuf[startPos], length);
+ aSaveBuffer(cmd++, dmem + DEFAULT_LEN_1CH, &reverb->rightRingBuf[startPos], length);
return cmd;
}
-Acmd* AudioSynth_SaveBuffer(Acmd* cmd, u16 dmem, u16 offset, s32 length, s16* buf) {
+Acmd* AudioSynth_SaveBufferOffset(Acmd* cmd, u16 dmem, u16 offset, s32 length, s16* buf) {
aSaveBuffer(cmd++, dmem, &buf[offset], length);
return cmd;
}
-Acmd* func_800DC124(Acmd* cmd, s32 arg1, SynthesisReverb* reverb, s16 arg3) {
+Acmd* AudioSynth_MaybeLoadRingBuffer2(Acmd* cmd, s32 arg1, SynthesisReverb* reverb, s16 bufIndex) {
if (reverb->downsampleRate == 1) {
- cmd = func_800DBF5C(cmd, arg1, reverb, arg3);
+ cmd = AudioSynth_LoadRingBuffer2(cmd, arg1, reverb, bufIndex);
}
return cmd;
}
Acmd* func_800DC164(Acmd* cmd, s32 arg1, SynthesisReverb* reverb, s16 arg3) {
+ // Sets DMEM_WET_{LEFT,RIGHT}_CH, clobbers DMEM_TEMP
if (reverb->downsampleRate == 1) {
if (reverb->unk_18 != 0) {
cmd = func_800DB828(cmd, arg1, reverb, arg3);
} else {
- cmd = func_800DBEB0(cmd, arg1, reverb, arg3);
+ cmd = AudioSynth_LoadRingBuffer1(cmd, arg1, reverb, arg3);
}
} else {
cmd = func_800DB4E4(cmd, arg1, reverb, arg3);
@@ -484,107 +514,116 @@ Acmd* func_800DC164(Acmd* cmd, s32 arg1, SynthesisReverb* reverb, s16 arg3) {
return cmd;
}
-Acmd* func_800DC1D8(Acmd* cmd, SynthesisReverb* reverb, s16 bufIdx) {
- ReverbRingBufferItem* bufItem = &reverb->items[reverb->curFrame][bufIdx];
+Acmd* AudioSynth_SaveReverbSamples(Acmd* cmd, SynthesisReverb* reverb, s16 bufIndex) {
+ ReverbRingBufferItem* bufItem = &reverb->items[reverb->curFrame][bufIndex];
if (reverb->downsampleRate == 1) {
if (reverb->unk_18 != 0) {
- cmd = func_800DB680(cmd, reverb, bufIdx);
+ cmd = func_800DB680(cmd, reverb, bufIndex);
} else {
- cmd = AudioSynth_SaveRingBuffer(cmd, 0xC80, bufItem->startPos, bufItem->lengthA, reverb);
+ // Put the oldest samples in the ring buffer into the wet channels
+ cmd = AudioSynth_SaveRingBufferPart(cmd, DMEM_WET_LEFT_CH, bufItem->startPos, bufItem->lengthA, reverb);
if (bufItem->lengthB != 0) {
- cmd = AudioSynth_SaveRingBuffer(cmd, bufItem->lengthA + 0xC80, 0, bufItem->lengthB, reverb);
+ // Ring buffer wrapped
+ cmd = AudioSynth_SaveRingBufferPart(cmd, DMEM_WET_LEFT_CH + bufItem->lengthA, 0, bufItem->lengthB,
+ reverb);
}
}
} else {
- func_800DBD38(cmd++, 0xC80, 0x340, reverb->items[reverb->curFrame][bufIdx].toDownsampleLeft);
+ // Downsampling is done later by CPU when RSP is done, therefore we need to have
+ // double buffering. Left and right buffers are adjacent in memory.
+ AudioSynth_SaveBuffer(cmd++, DMEM_WET_LEFT_CH, DEFAULT_LEN_2CH,
+ reverb->items[reverb->curFrame][bufIndex].toDownsampleLeft);
}
reverb->resampleFlags = 0;
return cmd;
}
-Acmd* func_800DC2DC(Acmd* cmd, SynthesisReverb* reverb, s16 bufIdx) {
- ReverbRingBufferItem* bufItem = &reverb->items2[reverb->curFrame][bufIdx];
+Acmd* AudioSynth_SaveRingBuffer2(Acmd* cmd, SynthesisReverb* reverb, s16 bufIndex) {
+ ReverbRingBufferItem* bufItem = &reverb->items2[reverb->curFrame][bufIndex];
- cmd = AudioSynth_SaveRingBuffer(cmd, 0xC80, bufItem->startPos, bufItem->lengthA, reverb);
+ cmd = AudioSynth_SaveRingBufferPart(cmd, DMEM_WET_LEFT_CH, bufItem->startPos, bufItem->lengthA, reverb);
if (bufItem->lengthB != 0) {
- cmd = AudioSynth_SaveRingBuffer(cmd, bufItem->lengthA + 0xC80, 0, bufItem->lengthB, reverb);
+ // Ring buffer wrapped
+ cmd = AudioSynth_SaveRingBufferPart(cmd, DMEM_WET_LEFT_CH + bufItem->lengthA, 0, bufItem->lengthB, reverb);
}
return cmd;
}
-Acmd* func_800DC384(s16* aiBuf, s32 aiBufLen, Acmd* cmd, s32 updateIdx) {
- u8 sp9C[0x5C];
- s16 phi_s2;
- s16 phi_s4;
- SynthesisReverb* temp_t8;
+Acmd* AudioSynth_DoOneAudioUpdate(s16* aiBuf, s32 aiBufLen, Acmd* cmd, s32 updateIndex) {
+ u8 noteIndices[0x5C];
+ s16 count;
+ s16 reverbIndex;
+ SynthesisReverb* reverb;
s32 useReverb;
s32 t;
s32 i;
- NoteSubEu* phi_v0;
- NoteSubEu* phi_v0_2;
+ NoteSubEu* noteSubEu;
+ NoteSubEu* noteSubEu2;
s32 unk14;
- t = gAudioContext.maxSimultaneousNotes * updateIdx;
- phi_s2 = 0;
+ t = gAudioContext.maxSimultaneousNotes * updateIndex;
+ count = 0;
if (gAudioContext.numSynthesisReverbs == 0) {
for (i = 0; i < gAudioContext.maxSimultaneousNotes; i++) {
if (gAudioContext.noteSubsEu[t + i].bitField0.s.enabled) {
- sp9C[phi_s2++] = i;
+ noteIndices[count++] = i;
}
}
} else {
- for (phi_s4 = 0; phi_s4 < gAudioContext.numSynthesisReverbs; phi_s4++) {
+ for (reverbIndex = 0; reverbIndex < gAudioContext.numSynthesisReverbs; reverbIndex++) {
for (i = 0; i < gAudioContext.maxSimultaneousNotes; i++) {
- phi_v0 = &gAudioContext.noteSubsEu[t + i];
- if (phi_v0->bitField0.s.enabled && phi_v0->bitField1.s.reverbIndex == phi_s4) {
- sp9C[phi_s2++] = i;
+ noteSubEu = &gAudioContext.noteSubsEu[t + i];
+ if (noteSubEu->bitField0.s.enabled && noteSubEu->bitField1.s.reverbIndex == reverbIndex) {
+ noteIndices[count++] = i;
}
}
}
for (i = 0; i < gAudioContext.maxSimultaneousNotes; i++) {
- phi_v0 = &gAudioContext.noteSubsEu[t + i];
- if (phi_v0->bitField0.s.enabled && phi_v0->bitField1.s.reverbIndex >= gAudioContext.numSynthesisReverbs) {
- sp9C[phi_s2++] = i;
+ noteSubEu = &gAudioContext.noteSubsEu[t + i];
+ if (noteSubEu->bitField0.s.enabled &&
+ noteSubEu->bitField1.s.reverbIndex >= gAudioContext.numSynthesisReverbs) {
+ noteIndices[count++] = i;
}
}
}
- aClearBuffer(cmd++, 0x940, 0x340);
+ aClearBuffer(cmd++, DMEM_LEFT_CH, DEFAULT_LEN_2CH);
i = 0;
- for (phi_s4 = 0; phi_s4 < gAudioContext.numSynthesisReverbs; phi_s4++) {
- temp_t8 = &gAudioContext.synthesisReverbs[phi_s4];
- useReverb = temp_t8->useReverb;
+ for (reverbIndex = 0; reverbIndex < gAudioContext.numSynthesisReverbs; reverbIndex++) {
+ reverb = &gAudioContext.synthesisReverbs[reverbIndex];
+ useReverb = reverb->useReverb;
if (useReverb) {
- cmd = func_800DC164(cmd, aiBufLen, temp_t8, updateIdx);
- aMix(cmd++, 0x34, temp_t8->unk_0A, 0xC80, 0x940);
- unk14 = temp_t8->unk_14;
+ cmd = func_800DC164(cmd, aiBufLen, reverb, updateIndex);
+ aMix(cmd++, 0x34, reverb->unk_0A, DMEM_WET_LEFT_CH, DMEM_LEFT_CH);
+ unk14 = reverb->unk_14;
if (unk14) {
- aDMEMMove(cmd++, 0xC80, 0x3E0, 0x340);
+ aDMEMMove(cmd++, DMEM_WET_LEFT_CH, DMEM_WET_TEMP, DEFAULT_LEN_2CH);
}
- aMix(cmd++, 0x34, temp_t8->unk_0C + 0x8000, 0xC80, 0xC80);
- if ((temp_t8->unk_10 != 0) || (temp_t8->unk_12 != 0)) {
- cmd = func_800DB480(cmd, temp_t8);
+ aMix(cmd++, 0x34, reverb->unk_0C + 0x8000, DMEM_WET_LEFT_CH, DMEM_WET_LEFT_CH);
+ if (reverb->leakRtl != 0 || reverb->leakLtr != 0) {
+ cmd = AudioSynth_LeakReverb(cmd, reverb);
}
if (unk14) {
- cmd = func_800DC1D8(cmd, temp_t8, updateIdx);
- if (temp_t8->unk_05 != -1) {
- cmd = func_800DBAE8(cmd, temp_t8, updateIdx);
+ cmd = AudioSynth_SaveReverbSamples(cmd, reverb, updateIndex);
+ if (reverb->unk_05 != -1) {
+ cmd = AudioSynth_MaybeMixRingBuffer1(cmd, reverb, updateIndex);
}
- cmd = func_800DC124(cmd, aiBufLen, temp_t8, updateIdx);
- aMix(cmd++, 0x34, temp_t8->unk_16, 0x3E0, 0xC80);
+ cmd = AudioSynth_MaybeLoadRingBuffer2(cmd, aiBufLen, reverb, updateIndex);
+ aMix(cmd++, 0x34, reverb->unk_16, DMEM_WET_TEMP, DMEM_WET_LEFT_CH);
}
}
- while (i < phi_s2) {
- phi_v0_2 = &gAudioContext.noteSubsEu[sp9C[i] + t];
- if (phi_v0_2->bitField1.s.reverbIndex == phi_s4) {
- cmd = func_800DC910(sp9C[i], phi_v0_2, &gAudioContext.notes[sp9C[i]].synthesisState, aiBuf, aiBufLen,
- cmd, updateIdx);
+ while (i < count) {
+ noteSubEu2 = &gAudioContext.noteSubsEu[noteIndices[i] + t];
+ if (noteSubEu2->bitField1.s.reverbIndex == reverbIndex) {
+ cmd = AudioSynth_ProcessNote(noteIndices[i], noteSubEu2,
+ &gAudioContext.notes[noteIndices[i]].synthesisState, aiBuf, aiBufLen, cmd,
+ updateIndex);
} else {
break;
}
@@ -592,394 +631,383 @@ Acmd* func_800DC384(s16* aiBuf, s32 aiBufLen, Acmd* cmd, s32 updateIdx) {
}
if (useReverb) {
- if ((temp_t8->unk_270 != NULL) || (temp_t8->unk_274 != NULL)) {
- cmd = func_800DBA40(cmd, aiBufLen * 2, temp_t8);
+ if (reverb->filterLeft != NULL || reverb->filterRight != NULL) {
+ cmd = AudioSynth_FilterReverb(cmd, aiBufLen * 2, reverb);
}
if (unk14) {
- cmd = func_800DC2DC(cmd, temp_t8, updateIdx);
+ cmd = AudioSynth_SaveRingBuffer2(cmd, reverb, updateIndex);
} else {
- cmd = func_800DC1D8(cmd, temp_t8, updateIdx);
- if (temp_t8->unk_05 != -1) {
- cmd = func_800DBAE8(cmd, temp_t8, updateIdx);
+ cmd = AudioSynth_SaveReverbSamples(cmd, reverb, updateIndex);
+ if (reverb->unk_05 != -1) {
+ cmd = AudioSynth_MaybeMixRingBuffer1(cmd, reverb, updateIndex);
}
}
}
}
- while (i < phi_s2) {
- cmd = func_800DC910(sp9C[i], &gAudioContext.noteSubsEu[t + sp9C[i]],
- &gAudioContext.notes[sp9C[i]].synthesisState, aiBuf, aiBufLen, cmd, updateIdx);
+ while (i < count) {
+ cmd = AudioSynth_ProcessNote(noteIndices[i], &gAudioContext.noteSubsEu[t + noteIndices[i]],
+ &gAudioContext.notes[noteIndices[i]].synthesisState, aiBuf, aiBufLen, cmd,
+ updateIndex);
i++;
}
- updateIdx = aiBufLen * 2;
- aInterleave(cmd++, updateIdx, 0x3C0, 0x940, 0xAE0);
- aSaveBuffer(cmd++, 0x3C0, aiBuf, updateIdx * 2);
+ updateIndex = aiBufLen * 2;
+ aInterleave(cmd++, DMEM_TEMP, DMEM_LEFT_CH, DMEM_RIGHT_CH, updateIndex);
+ aSaveBuffer(cmd++, DMEM_TEMP, aiBuf, updateIndex * 2);
return cmd;
}
-#ifdef NON_EQUIVALENT
-Acmd* func_800DC910(s32 noteIdx, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s16* aiBuf, s32 aiBufLen,
- Acmd* cmd, s32 updateIdx) {
- AudioBankSample* sp150;
- AdpcmLoop* sp14C;
- s32 sp140;
- s32 sp13C;
- s32 sp138;
- u16 sp136;
- s32 sp120;
- s32 sp110;
- s32 sp104;
- s32 sp100;
- s32 spFC;
- s32 spD4;
- s32 spCC;
- s32 spC8;
- s32 spC0;
- s32 spBC;
- s32 spB0;
- u16 spAE;
- Note* sp9C;
- u32 sp8C;
- u32 sp88;
- u32 sp50;
- Acmd* temp_a0_3;
- Acmd* temp_v0;
- Acmd* temp_v0_10;
- Acmd* temp_v0_11;
- Acmd* temp_v0_12;
- Acmd* temp_v0_13;
- Acmd* temp_v0_8;
- Acmd* temp_v0_9;
- Acmd* temp_v1_3;
- NoteSubEu* temp_a1;
- NoteSynthesisState* temp_a2;
- s32 temp_a1_3;
- s32 temp_a2_2;
- s32 temp_a2_3;
- s32 temp_a2_4;
- s32 temp_s1;
- s32 temp_s1_2;
- s32 temp_s1_3;
- s32 temp_s4;
- s32 temp_t0_3;
- s32 temp_t1;
- s32 temp_t9;
- s32 temp_v0_3;
- s32 temp_v0_5;
- s32 temp_v0_7;
- s32 temp_v1;
- s32 temp_v1_4;
+Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s16* aiBuf,
+ s32 aiBufLen, Acmd* cmd, s32 updateIndex) {
+ s32 pad1[3];
+ AudioBankSample* audioBankSample;
+ AdpcmLoop* loopInfo;
+ s32 samplesRemaining;
+ s32 nSamplesInThisIteration;
+ s32 noteFinished;
+ s32 restart;
+ s32 flags;
+ u16 resamplingRateFixedPoint;
+ s32 phi_s6;
+ s32 phi_s7;
+ s32 phi_a1_2;
s32 temp_v1_5;
+ s32 sp120;
s32 temp_v1_6;
- s32 temp_t0;
- u16 temp_t6;
- u16 temp_v1_7;
- u32 temp_a3;
- u32 temp_a3_3;
- u32 temp_a3_4;
- u32 temp_a3_5;
- u32 temp_t0_2;
- u32 temp_v0_6;
- u8 temp_a1_4;
- AdpcmLoop* temp_a1_2;
- AudioBankSample* temp_v0_2;
- AdpcmBook* temp_v0_4;
- s32 phi_t1;
- u8 phi_v1;
- s32 phi_fp;
- s32 phi_t1_2;
+ void* buf;
+ s32 phi_s0;
+ s32 sampleAddr;
+ u32 samplesLenFixedPoint;
+ s32 samplesLenAdjusted;
+ s32 nAdpcmSamplesProcessed;
+ s32 endPos;
+ s32 nSamplesToProcess;
+ s32 phi_s4;
s32 phi_s3;
+ s32 pad2[7];
+ s32 spD4;
s32 phi_s1;
- s32 phi_a1;
- s32 phi_t1_3;
+ s32 spCC;
+ s32 spC8;
+ u8* phi_a1;
+ s32 nParts;
+ s32 curPart;
s32 phi_t0;
- s32 phi_s4;
- s32 phi_s0;
- s32 phi_s5;
- s32 phi_s6;
- s32 phi_s7;
- s32 phi_t1_4;
- s32 phi_v1_2;
- u8 phi_v1_3;
- s32 phi_v1_4;
- s32 phi_t1_5;
- s32 phi_a1_2;
- NoteSubEu* phi_s6_2;
- s32 phi_s1_2;
- NoteSubEu* phi_s6_3;
-
- sp8C = noteSubEu->bitField1.s.bookOffset;
- sp88 = noteSubEu->bitField0.s.finished;
- sp9C = &gAudioContext.notes[noteIdx];
- sp138 = 0;
-
- if (noteSubEu->bitField0.s.needsInit == 1) {
+ s32 side;
+ s32 resampledTempLen;
+ u16 noteSamplesDmemAddrBeforeResampling;
+ s32 temp_mult;
+ s32 thing;
+ s32 s5;
+ Note* note;
+ u32 nSamplesToLoad;
+ u16 unk7;
+ u16 unkE;
+ s16* filter;
+ s32 bookOffset;
+ s32 finished;
+ s32 aligned;
+ s16 addr;
+ u16 unused;
+
+ bookOffset = noteSubEu->bitField1.s.bookOffset;
+ finished = noteSubEu->bitField0.s.finished;
+ note = &gAudioContext.notes[noteIndex];
+ flags = A_CONTINUE;
+
+ if (noteSubEu->bitField0.s.needsInit == true) {
+ flags = A_INIT;
synthState->restart = 0;
- synthState->unk_06 = 0;
+ synthState->samplePosInt = note->unk_BC;
+ synthState->samplePosFrac = 0;
synthState->curVolLeft = 0;
synthState->curVolRight = 0;
- synthState->samplePosInt = sp9C->unk_BC;
- synthState->samplePosFrac = noteSubEu->reverbVol;
- synthState->unk_05 = 0;
synthState->prevHeadsetPanRight = 0;
synthState->prevHeadsetPanLeft = 0;
+ synthState->reverbVol = noteSubEu->reverbVol;
+ synthState->numAdpcmParts = 0;
synthState->unk_1A = 1;
- sp9C->noteSubEu.bitField0.s.finished = 0;
- sp88 = 0;
- sp138 = 1;
+ note->noteSubEu.bitField0.s.finished = false;
+ finished = false;
}
- sp136 = noteSubEu->resamplingRateFixedPoint;
- spC0 = noteSubEu->bitField1.s.isSyntheticWave + 1;
- temp_t0 = (sp136 * aiBufLen * 2) + synthState->unk_06;
- synthState->unk_06 = temp_t0;
- if (1 != synthState->unk_05) {}
- synthState->unk_05 = spC0;
- if (noteSubEu->bitField1.s.bit2) {
- sp50 = temp_t0 >> 0x10;
- cmd = func_800DD9F4(cmd, noteSubEu, synthState, sp50);
- spAE = (synthState->samplePosInt * 2) + 0x580;
- synthState->samplePosInt += sp50;
+ resamplingRateFixedPoint = noteSubEu->resamplingRateFixedPoint;
+ nParts = noteSubEu->bitField1.s.hasTwoAdpcmParts + 1;
+ samplesLenFixedPoint = (resamplingRateFixedPoint * aiBufLen * 2) + synthState->samplePosFrac;
+ 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 {
+ }
- // OLD BLOCK 94
+ synthState->numAdpcmParts = nParts;
+ if (noteSubEu->bitField1.s.isSyntheticWave) {
+ cmd = AudioSynth_LoadWaveSamples(cmd, noteSubEu, synthState, nSamplesToLoad);
+ noteSamplesDmemAddrBeforeResampling = DMEM_UNCOMPRESSED_NOTE + (synthState->samplePosInt * 2);
+ synthState->samplePosInt += nSamplesToLoad;
} else {
- sp50 = temp_t0 >> 0x10;
- sp150 = noteSubEu->sound.audioBankSound->sample;
- sp14C = sp150->loop;
- sp100 = sp14C->end;
- spB0 = 0;
- spBC = 0;
- sp110 = sp150->sampleAddr;
- for (spBC = 0; !sp88 && spBC < spC0; spBC++) {
- if (spC0 == 1) {
- phi_fp = sp50;
- } else if (sp50 & 1) {
- phi_fp = (sp50 & ~1) + (spBC * 2);
+ // ADPCM note
+ audioBankSample = noteSubEu->sound.audioBankSound->sample;
+ loopInfo = audioBankSample->loop;
+ endPos = loopInfo->end;
+ sampleAddr = audioBankSample->sampleAddr;
+ resampledTempLen = 0;
+
+ for (curPart = 0; curPart < nParts; curPart++) {
+ // reordering in the loop prelude
+ nAdpcmSamplesProcessed = 0;
+ s5 = 0;
+
+ if (nParts == 1) {
+ samplesLenAdjusted = nSamplesToLoad;
+ } else if (nSamplesToLoad & 1) {
+ samplesLenAdjusted = (nSamplesToLoad & ~1) + (curPart * 2);
} else {
- phi_fp = sp50;
+ samplesLenAdjusted = nSamplesToLoad;
}
- if (sp150->bits4 == 0 || sp150->bits4 == 3) {
- if (gAudioContext.unk_0x10 != sp150->bits4 + 8) {
- switch (sp8C) {
+ if (audioBankSample->codec == 0 || audioBankSample->codec == 3) {
+ if (gAudioContext.curLoadedBook != (*audioBankSample->book).book) {
+ u32 nEntries;
+ switch (bookOffset) {
case 1:
- gAudioContext.unk_0x10 = D_8012FBAA;
- ;
+ gAudioContext.curLoadedBook = D_8012FBAA;
break;
case 2:
- gAudioContext.unk_0x10 = sp150->bits4 + 8;
- break;
case 3:
default:
+ gAudioContext.curLoadedBook = (*audioBankSample->book).book;
break;
}
-
- aLoadADPCM(cmd++, sp150->book->order * 0x10 * sp150->book->npredictors, gAudioContext.unk_0x10);
+ if (1) {}
+ if (1) {}
+ if (1) {}
+ nEntries = 16 * audioBankSample->book->order * audioBankSample->book->npredictors;
+ aLoadADPCM(cmd++, nEntries, gAudioContext.curLoadedBook);
}
}
- sp104 = 0;
- phi_s5 = 0;
- while (sp104 != phi_fp) {
- sp140 = 0;
- sp13C = 0;
- spFC = phi_fp - sp104;
+
+ while (nAdpcmSamplesProcessed != samplesLenAdjusted) {
+ noteFinished = false;
+ restart = false;
+ phi_s4 = 0;
+
phi_s3 = synthState->samplePosInt & 0xF;
- if (phi_s3 == 0 && synthState->restart == 0) {
- phi_s3 = 0x10;
+ samplesRemaining = endPos - synthState->samplePosInt;
+ nSamplesToProcess = samplesLenAdjusted - nAdpcmSamplesProcessed;
+
+ if (phi_s3 == 0 && synthState->restart == false) {
+ phi_s3 = 16;
}
- if (spFC < (sp14C->end - synthState->samplePosInt)) {
- phi_s1 = (s32)((spFC - (0x10 - phi_s3)) + 0xF) / 0x10;
- phi_s0 = phi_s1 * 0x10;
- phi_s6 = (0x10 - phi_s3);
- phi_s7 = ((0x10 - phi_s3) + phi_s0) - spFC;
+ phi_s6 = 16 - phi_s3;
+
+ if (nSamplesToProcess < samplesRemaining) {
+ phi_s1 = (s32)(nSamplesToProcess - phi_s6 + 15) / 16;
+ phi_s0 = phi_s1 * 16;
+ phi_s7 = phi_s6 + phi_s0 - nSamplesToProcess;
} else {
- phi_s0 = (sp14C->end - synthState->samplePosInt) - (0x10 - phi_s3);
- phi_s6 = (0x10 - phi_s3);
+ phi_s0 = samplesRemaining - phi_s6;
+ phi_s7 = 0;
if (phi_s0 <= 0) {
phi_s0 = 0;
- phi_s6 = (sp14C->end - synthState->samplePosInt);
+ phi_s6 = samplesRemaining;
}
- phi_s1 = (s32)(phi_s0 + 0xF) / 0x10;
- if (sp14C->count != 0) {
- sp13C = 1;
- phi_s7 = 0;
+ phi_s1 = (phi_s0 + 15) / 16;
+ if (loopInfo->count != 0) {
+ // Loop around and restart
+ restart = true;
} else {
- sp140 = 1;
- phi_s7 = 0;
+ noteFinished = true;
}
}
- switch (sp150->bits4) {
- case 0:
- spCC = 0x10;
- spC8 = 0;
+
+ switch (audioBankSample->codec) {
+ case CODEC_ADPCM:
spD4 = 9;
- goto block_39;
- case 3:
spCC = 0x10;
spC8 = 0;
+ break;
+ case 3:
spD4 = 5;
- goto block_39;
- case 1:
spCC = 0x10;
spC8 = 0;
+ break;
+ case CODEC_S8:
spD4 = 0x10;
- goto block_39;
+ spCC = 0x10;
+ spC8 = 0;
+ break;
case 2:
- sp138 = 0;
+ AudioSynth_ClearBuffer(cmd++, DMEM_UNCOMPRESSED_NOTE, (samplesLenAdjusted * 2) + 0x20);
+ flags = A_CONTINUE;
sp120 = 0;
- sp104 = phi_fp;
- func_800DBB9C(cmd++, 0x580, (phi_fp * 2) + 0x20);
- phi_s5 = phi_fp;
- break;
+ nAdpcmSamplesProcessed = samplesLenAdjusted;
+ s5 = samplesLenAdjusted;
+ goto skip;
case 5:
- sp138 = 0;
+ AudioSynth_ClearBuffer(cmd++, DMEM_UNCOMPRESSED_NOTE, (samplesLenAdjusted * 2) + 0x20);
+ flags = A_CONTINUE;
sp120 = 0;
- sp104 = phi_fp;
- func_800DBB9C(cmd++, 0x580, (phi_fp * 2) + 0x20);
- phi_s5 = phi_fp;
+ nAdpcmSamplesProcessed = samplesLenAdjusted;
+ s5 = samplesLenAdjusted;
+ goto skip;
+ case 4:
break;
- // OLD BLOCK 71
- // OLD BLOCK 78
+ }
- case 4:
- block_39:
- if (phi_s1 != 0) {
- temp_v1_5 = (s32)((synthState->samplePosInt + spCC) - phi_s3) / 0x10;
- if (sp150->bits2 == 0) {
- phi_a1 = spC8 + (temp_v1_5 * spD4) + sp110;
- } else if (sp150->bits2 != 1) {
- phi_a1 =
- func_800E12DC(spC8 + (temp_v1_5 * spD4) + sp110, ((phi_s1 * spD4) + 0x1F) & ~0xF,
- sp138, &synthState->sampleDmaIndex, sp150->bits2);
- } else {
- return cmd;
- }
+ if (phi_s1 != 0) {
+ temp_v1_5 = (synthState->samplePosInt + spCC - phi_s3) / 16;
+ temp_mult = (temp_v1_5 * spD4);
+ if (audioBankSample->medium == 0) {
+ phi_a1 = spC8 + temp_mult + sampleAddr;
+ } else if (audioBankSample->medium == 1) {
+ return cmd;
+ } else {
+ phi_a1 =
+ Audio_DmaSampleData((u32)(spC8 + temp_mult + sampleAddr), ALIGN16((phi_s1 * spD4) + 0x10),
+ flags, &synthState->sampleDmaIndex, audioBankSample->medium);
+ }
- if (phi_a1 != 0) {
- phi_t0 = phi_a1 & 0xF;
- temp_v0_7 = ((phi_s1 * spD4) + 0x1F) & 0xFFF0;
- aLoadBuffer(cmd++, phi_a1 - phi_t0, 0x940 - temp_v0_7, temp_v0_7);
- } else {
- phi_t0 = 0;
- phi_s0 = 0;
- }
+ if (phi_a1 == NULL) {
+ return cmd;
+ }
- if (synthState->restart) {
- aSetLoop(cmd++, sp150->loop);
- sp138 = 2;
- synthState->restart = 0;
- }
+ phi_t0 = (u32)phi_a1 & 0xF;
+ aligned = ALIGN16((phi_s1 * spD4) + 16);
+ addr = DMEM_COMPRESSED_ADPCM_DATA - aligned;
+ aLoadBuffer(cmd++, phi_a1 - phi_t0, addr, aligned);
+ } else {
+ phi_s0 = 0;
+ phi_t0 = 0;
+ }
- if (sp104 == 0) {
- sp120 = phi_s3 * 2;
- phi_s4 = 0;
- } else {
- phi_s4 = (phi_s5 + 0x1F) & ~0xF;
- }
- switch (sp150->bits4) {
- case 0:
- aSetBuffer(cmd++, 0, ((0x940 - (((phi_s1 * spD4) + 0x1F) & 0xFFF0)) + phi_t0),
- phi_s4 + 0x580, phi_s0 * 2);
- aADPCMdec(cmd++, sp138, synthState->synthesisBuffers);
- break;
- case 3:
- aSetBuffer(cmd++, 0, ((0x940 - (((phi_s1 * spD4) + 0x1F) & 0xFFF0)) + phi_t0),
- phi_s4 + 0x580, phi_s0 * 2);
- aADPCMdec(cmd++, sp138 | 4, synthState->synthesisBuffers);
- break;
- case 1:
- AudioSynth_SetBuffer(cmd++, 0, 0x940 - (((phi_s1 * spD4) + 0x1F) & 0xFFF0) + phi_t0,
- phi_s4 + 0x580, phi_s0 * 2);
- AudioSynth_UnkCmd17(cmd++, sp138, (s32)synthState->synthesisBuffers);
- break;
- }
+ if (synthState->restart) {
+ aSetLoop(cmd++, audioBankSample->loop->state);
+ flags = A_LOOP;
+ synthState->restart = false;
+ }
- if (sp104 != 0) {
- aDMEMMove(cmd++, phi_s4 + (phi_s3 * 2) + 0x580, phi_s5 + 0x580,
- (((phi_s0 + phi_s6) - phi_s7) * 2));
- }
+ nSamplesInThisIteration = phi_s0 + phi_s6 - phi_s7;
+ if (nAdpcmSamplesProcessed == 0) {
+ if (1) {}
+ sp120 = phi_s3 * 2;
+ } else {
+ phi_s4 = ALIGN16(s5 + 16);
+ }
+ switch (audioBankSample->codec) {
+ case CODEC_ADPCM:
+ aligned = ALIGN16((phi_s1 * spD4) + 0x10);
+ addr = DMEM_COMPRESSED_ADPCM_DATA - aligned;
+ aSetBuffer(cmd++, 0, addr + phi_t0, DMEM_UNCOMPRESSED_NOTE + phi_s4, phi_s0 * 2);
+ aADPCMdec(cmd++, flags, synthState->synthesisBuffers->adpcmdecState);
+ break;
+ case 3:
+ aligned = ALIGN16((phi_s1 * spD4) + 0x10);
+ addr = DMEM_COMPRESSED_ADPCM_DATA - aligned;
+ aSetBuffer(cmd++, 0, addr + phi_t0, DMEM_UNCOMPRESSED_NOTE + phi_s4, phi_s0 * 2);
+ aADPCMdec(cmd++, flags | 4, synthState->synthesisBuffers->adpcmdecState);
+ break;
+ case CODEC_S8:
+ aligned = ALIGN16((phi_s1 * spD4) + 0x10);
+ addr = DMEM_COMPRESSED_ADPCM_DATA - aligned;
+ AudioSynth_SetBuffer(cmd++, 0, addr + phi_t0, DMEM_UNCOMPRESSED_NOTE + phi_s4, phi_s0 * 2);
+ AudioSynth_S8Dec(cmd++, flags, synthState->synthesisBuffers->adpcmdecState);
+ break;
+ }
- temp_v1_6 = (phi_s0 + phi_s6) - phi_s7;
- temp_t1 = sp104 + temp_v1_6;
-
- switch (sp138) {
- case 1:
- sp120 = 0x20;
- phi_s5 = (phi_s0 * 2) + 0x20;
- break;
- case 2:
- phi_s5 += (temp_v1_6 * 2);
- break;
- default:
- if (phi_s5 != 0) {
- phi_s5 += (temp_v1_6 * 2);
- } else {
- phi_s5 = (phi_s3 + temp_v1_6) * 2;
- }
- break;
- }
+ if (nAdpcmSamplesProcessed != 0) {
+ aDMEMMove(cmd++, DMEM_UNCOMPRESSED_NOTE + phi_s4 + (phi_s3 * 2), DMEM_UNCOMPRESSED_NOTE + s5,
+ nSamplesInThisIteration * 2);
+ }
- sp138 = 0;
- break;
+ nAdpcmSamplesProcessed += nSamplesInThisIteration;
+
+ switch (flags) {
+ case A_INIT:
+ sp120 = 0x20;
+ s5 = (phi_s0 + 0x10) * 2;
+ break;
+ case A_LOOP:
+ s5 = nSamplesInThisIteration * 2 + s5;
+ break;
+ default:
+ if (s5 != 0) {
+ s5 = nSamplesInThisIteration * 2 + s5;
+ } else {
+ s5 = (phi_s3 + nSamplesInThisIteration) * 2;
}
+ break;
}
- if (sp140 != 0) {
- sp88 = 1;
- func_800DBB9C(cmd++, phi_s5 + 0x580, (phi_fp - sp104) * 2);
- sp9C->noteSubEu.bitField0.s.finished = 1;
- func_800DB2C0(updateIdx, noteIdx);
+ flags = A_CONTINUE;
+
+ skip:
+ if (noteFinished) {
+ AudioSynth_ClearBuffer(cmd++, DMEM_UNCOMPRESSED_NOTE + s5,
+ (samplesLenAdjusted - nAdpcmSamplesProcessed) * 2);
+ finished = true;
+ note->noteSubEu.bitField0.s.finished = true;
+ func_800DB2C0(updateIndex, noteIndex);
+ break;
} else {
- if (sp13C != 0) {
- synthState->restart = 1;
- synthState->samplePosInt = sp14C->start;
+ if (restart) {
+ synthState->restart = true;
+ synthState->samplePosInt = loopInfo->start;
} else {
- synthState->samplePosInt += spFC;
+ synthState->samplePosInt += nSamplesToProcess;
}
}
- phi_v1_2 = spC0;
}
- switch (spC0) {
+ switch (nParts) {
case 1:
- spAE = sp120 + 0x580;
+ noteSamplesDmemAddrBeforeResampling = DMEM_UNCOMPRESSED_NOTE + sp120;
break;
case 2:
- break;
- default:
- switch (spBC) {
+ switch (curPart) {
case 0:
- spAE = 0x3E0;
- AudioSynth_InterL(cmd++, sp120 + 0x580, 0x3E0, ((phi_fp / 2) + 7) & ~7);
- spB0 = phi_fp;
- if (sp88 != 0) {
- func_800DBB9C(cmd++, phi_fp + 0x3E0, phi_fp + 0x10);
+ AudioSynth_InterL(cmd++, DMEM_UNCOMPRESSED_NOTE + sp120, DMEM_TEMP + 0x20,
+ ((samplesLenAdjusted / 2) + 7) & ~7);
+ resampledTempLen = samplesLenAdjusted;
+ noteSamplesDmemAddrBeforeResampling = DMEM_TEMP + 0x20;
+ if (finished) {
+ AudioSynth_ClearBuffer(cmd++, noteSamplesDmemAddrBeforeResampling + resampledTempLen,
+ samplesLenAdjusted + 0x10);
}
break;
case 1:
- AudioSynth_InterL(cmd++, sp120 + 0x580, spB0 + 0x3E0, ((phi_fp / 2) + 7) & ~7);
+ AudioSynth_InterL(cmd++, DMEM_UNCOMPRESSED_NOTE + sp120,
+ DMEM_TEMP + 0x20 + resampledTempLen, ((samplesLenAdjusted / 2) + 7) & ~7);
break;
}
}
+ if (finished) {
+ break;
+ }
}
}
- sp138 = 0;
- if (noteSubEu->bitField0.s.needsInit == 1) {
- noteSubEu->bitField0.s.needsInit = 0;
- sp138 = 1;
+ flags = A_CONTINUE;
+ if (noteSubEu->bitField0.s.needsInit == true) {
+ noteSubEu->bitField0.s.needsInit = false;
+ flags = A_INIT;
}
- cmd = func_800DD62C(cmd++, synthState, aiBufLen * 2, sp136, spAE, sp138);
- if (sp8C == 3) {
- AudioSynth_UnkCmd19(cmd++, 0x3C0, 0x3C0, aiBufLen * 2, 0);
+ cmd = AudioSynth_FinalResample(cmd, synthState, aiBufLen * 2, resamplingRateFixedPoint,
+ noteSamplesDmemAddrBeforeResampling, flags);
+ if (bookOffset == 3) {
+ AudioSynth_UnkCmd19(cmd++, DMEM_TEMP, DMEM_TEMP, aiBufLen * 2, 0);
}
- if (sp8C == 2) {
- AudioSynth_UnkCmd3(cmd++, 0x3C0, 0x3C0, aiBufLen * 2);
+ if (bookOffset == 2) {
+ AudioSynth_UnkCmd3(cmd++, DMEM_TEMP, DMEM_TEMP, aiBufLen * 2);
}
phi_a1_2 = noteSubEu->unk_2;
@@ -987,109 +1015,114 @@ Acmd* func_800DC910(s32 noteIdx, NoteSubEu* noteSubEu, NoteSynthesisState* synth
if (phi_a1_2 < 0x10) {
phi_a1_2 = 0x10;
}
- AudioSynth_HiLoGain(cmd++, phi_a1_2, 0x3C0, 0, (aiBufLen * 2) + 0x20);
+ AudioSynth_HiLoGain(cmd++, phi_a1_2, DMEM_TEMP, 0, (aiBufLen * 2) + 0x20);
}
- if (noteSubEu->unk_14 != 0) {
- AudioSynth_SetFilterCount(cmd++, aiBufLen * 2, noteSubEu->unk_14);
- AudioSynth_SetFilter(cmd++, sp138, 0x3C0, synthState->synthesisBuffers->mixEnvelopeState);
+ filter = noteSubEu->filter;
+ if (filter != 0) {
+ AudioSynth_SetFilterCount(cmd++, aiBufLen * 2, filter);
+ AudioSynth_SetFilter(cmd++, flags, DMEM_TEMP, synthState->synthesisBuffers->mixEnvelopeState);
}
- if (noteSubEu->unk_07 != 0 && noteSubEu->unk_10 != 0) {
- AudioSynth_DMemMove(cmd++, 0x3C0, 0x760, aiBufLen * 2);
+ unk7 = noteSubEu->unk_07;
+ unkE = noteSubEu->unk_0E;
+ buf = &synthState->synthesisBuffers->panSamplesBuffer[0x18];
+ if (unk7 != 0 && noteSubEu->unk_0E != 0) {
+ AudioSynth_DMemMove(cmd++, DMEM_TEMP, DMEM_SCRATCH2, aiBufLen * 2);
+ thing = DMEM_SCRATCH2 - unk7;
if (synthState->unk_1A != 0) {
- func_800DBB9C(cmd++, 0x760 - noteSubEu->unk_07, noteSubEu->unk_07);
+ AudioSynth_ClearBuffer(cmd++, thing, unk7);
synthState->unk_1A = 0;
} else {
- func_800DBD10(cmd++, 0x760 - noteSubEu->unk_07, noteSubEu->unk_07,
- &synthState->synthesisBuffers->panSamplesBuffer[0x18]);
+ AudioSynth_LoadBuffer(cmd++, thing, unk7, buf);
}
- AudioSynth_SaveBuffer(cmd++, ((aiBufLen * 2) - noteSubEu->unk_07) + 0x3C0, noteSubEu->unk_07,
- &synthState->synthesisBuffers->panSamplesBuffer[0x18], NULL);
- func_800DBBD4(cmd++, (aiBufLen * 2) >> 4, noteSubEu->unk_10, 0x760, 0x760 - noteSubEu->unk_07);
- AudioSynth_DMemMove(cmd++, 0x760 - noteSubEu->unk_07, 0x3C0, aiBufLen * 2);
+ AudioSynth_SaveBuffer(cmd++, DMEM_TEMP + (aiBufLen * 2) - unk7, unk7, buf);
+ AudioSynth_Mix(cmd++, (aiBufLen * 2) >> 4, unkE, DMEM_SCRATCH2, thing);
+ AudioSynth_DMemMove(cmd++, thing, DMEM_TEMP, aiBufLen * 2);
} else {
synthState->unk_1A = 1;
}
- if ((noteSubEu->headsetPanRight != 0) || (synthState->prevHeadsetPanRight != 0)) {
- phi_s0 = 1;
- } else if ((noteSubEu->headsetPanLeft != 0) || synthState->prevHeadsetPanLeft != 0) {
- phi_s0 = 2;
+ if (noteSubEu->headsetPanRight != 0 || synthState->prevHeadsetPanRight != 0) {
+ side = 1;
+ } else if (noteSubEu->headsetPanLeft != 0 || synthState->prevHeadsetPanLeft != 0) {
+ side = 2;
} else {
- phi_s0 = 0;
+ side = 0;
}
- cmd = func_800DD6CC(cmd, noteSubEu, synthState, aiBufLen, 0x3C0, phi_s0, sp138);
- if (noteSubEu->bitField0.s.usesHeadsetPanEffects) {
- if (!(sp138 & 1)) {
- sp138 = 0;
+ cmd = AudioSynth_ProcessEnvelope(cmd, noteSubEu, synthState, aiBufLen, DMEM_TEMP, side, flags);
+ if (noteSubEu->bitField1.s.usesHeadsetPanEffects2) {
+ if (!(flags & A_INIT)) {
+ flags = A_CONTINUE;
}
- cmd = func_800DDB64(cmd, noteSubEu, synthState, aiBufLen * 2, sp138, phi_s0);
+ cmd = AudioSynth_NoteApplyHeadsetPanEffects(cmd, noteSubEu, synthState, aiBufLen * 2, flags, side);
}
return cmd;
}
-#else
-#pragma GLOBAL_ASM("asm/non_matchings/code/audio_synthesis/func_800DC910.s")
-#endif
-Acmd* func_800DD62C(Acmd* cmd, NoteSynthesisState* synthState, s32 count, u16 pitch, u16 inpDmem, s32 resampleFlags) {
+Acmd* AudioSynth_FinalResample(Acmd* cmd, NoteSynthesisState* synthState, s32 count, u16 pitch, u16 inpDmem,
+ s32 resampleFlags) {
if (pitch == 0) {
- func_800DBB9C(cmd++, 0x3C0, count);
+ AudioSynth_ClearBuffer(cmd++, DMEM_TEMP, count);
} else {
- aSetBuffer(cmd++, 0, inpDmem, 0x3C0, count);
+ aSetBuffer(cmd++, 0, inpDmem, DMEM_TEMP, count);
aResample(cmd++, resampleFlags, pitch, synthState->synthesisBuffers->finalResampleState);
}
return cmd;
}
-extern u32 D_801304A4;
-extern u32 D_801304A8;
-extern u32 D_801304AC;
-
-Acmd* func_800DD6CC(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s32 aiBufLen, u16 arg4, s32 arg5,
- s32 arg6) {
+Acmd* AudioSynth_ProcessEnvelope(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s32 aiBufLen,
+ u16 inBuf, s32 headsetPanSettings, s32 flags) {
u32 phi_a1;
u16 curVolLeft;
u16 targetVolLeft;
s32 phi_t1;
- s16 reverVol;
+ s16 reverbVol;
u16 curVolRight;
- s16 phi_a3;
- s16 phi_t0;
- s16 sp52;
- s16 sp50;
+ s16 rampLeft;
+ s16 rampRight;
+ s16 rampReverb;
+ s16 sourceReverbVol;
u16 targetVolRight;
s32 pad;
curVolLeft = synthState->curVolLeft;
targetVolLeft = noteSubEu->targetVolLeft;
- targetVolLeft *= 0x10;
- reverVol = noteSubEu->reverbVol;
+ targetVolLeft <<= 4;
+ reverbVol = noteSubEu->reverbVol;
curVolRight = synthState->curVolRight;
targetVolRight = noteSubEu->targetVolRight;
- targetVolRight *= 0x10;
+ targetVolRight <<= 4;
- phi_a3 = targetVolLeft != curVolLeft ? (targetVolLeft - curVolLeft) / (aiBufLen >> 3) : 0;
- phi_t0 = targetVolRight != curVolRight ? (targetVolRight - curVolRight) / (aiBufLen >> 3) : 0;
+ if (targetVolLeft != curVolLeft) {
+ rampLeft = (targetVolLeft - curVolLeft) / (aiBufLen >> 3);
+ } else {
+ rampLeft = 0;
+ }
+ if (targetVolRight != curVolRight) {
+ rampRight = (targetVolRight - curVolRight) / (aiBufLen >> 3);
+ } else {
+ rampRight = 0;
+ }
- sp50 = synthState->samplePosFrac;
- phi_t1 = sp50 & 0x7F;
+ sourceReverbVol = synthState->reverbVol;
+ phi_t1 = sourceReverbVol & 0x7F;
- if (sp50 != reverVol) {
- sp52 = (((reverVol & 0x7F) - phi_t1) << 9) / (aiBufLen >> 3);
- synthState->samplePosFrac = reverVol;
+ if (sourceReverbVol != reverbVol) {
+ rampReverb = (((reverbVol & 0x7F) - phi_t1) << 9) / (aiBufLen >> 3);
+ synthState->reverbVol = reverbVol;
} else {
- sp52 = 0;
+ rampReverb = 0;
}
- synthState->curVolLeft = curVolLeft + (phi_a3 * (aiBufLen >> 3));
- synthState->curVolRight = curVolRight + (phi_t0 * (aiBufLen >> 3));
+ synthState->curVolLeft = curVolLeft + (rampLeft * (aiBufLen >> 3));
+ synthState->curVolRight = curVolRight + (rampRight * (aiBufLen >> 3));
- if (noteSubEu->bitField1.s.hasTwoAdpcmParts) {
- func_800DBB9C(cmd++, 0x5C0, 0x1A0);
- func_800DBCD4(cmd++, phi_t1 * 2, sp52, phi_a3, phi_t0);
+ if (noteSubEu->bitField1.s.usesHeadsetPanEffects2) {
+ AudioSynth_ClearBuffer(cmd++, DMEM_NOTE_PAN_TEMP, DEFAULT_LEN_1CH);
+ AudioSynth_EnvSetup1(cmd++, phi_t1 * 2, rampReverb, rampLeft, rampRight);
AudioSynth_EnvSetup2(cmd++, curVolLeft, curVolRight);
- switch (arg5) {
+ switch (headsetPanSettings) {
case 1:
phi_a1 = D_801304A4;
break;
@@ -1101,95 +1134,100 @@ Acmd* func_800DD6CC(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisState* synthSt
break;
}
} else {
- aEnvSetup1(cmd++, phi_t1 * 2, sp52, phi_a3, phi_t0);
+ aEnvSetup1(cmd++, phi_t1 * 2, rampReverb, rampLeft, rampRight);
aEnvSetup2(cmd++, curVolLeft, curVolRight);
phi_a1 = D_801304AC;
}
- aEnvMixer(cmd++, arg4, aiBufLen, (sp50 & 0x80) >> 7, noteSubEu->bitField0.s.stereoHeadsetEffects,
+ aEnvMixer(cmd++, inBuf, aiBufLen, (sourceReverbVol & 0x80) >> 7, noteSubEu->bitField0.s.stereoHeadsetEffects,
noteSubEu->bitField0.s.usesHeadsetPanEffects, noteSubEu->bitField0.s.stereoStrongRight,
- noteSubEu->bitField0.s.stereoStrongLeft, phi_a1);
+ noteSubEu->bitField0.s.stereoStrongLeft, phi_a1, D_801304A0);
return cmd;
}
-extern u8 D_801304C0[];
-Acmd* func_800DD9F4(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s32 arg3) {
+Acmd* AudioSynth_LoadWaveSamples(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s32 nSamplesToLoad) {
s32 temp_v0;
- s32 temp_t0;
- s32 samplePosInt;
+ s32 unk6 = noteSubEu->unk_06;
+ s32 samplePosInt = synthState->samplePosInt;
+ s32 repeats;
- temp_t0 = noteSubEu->unk_06;
- samplePosInt = synthState->samplePosInt;
if (noteSubEu->bitField1.s.bookOffset != 0) {
- func_800DBD10(cmd++, 0x580, ALIGN16(arg3 * 2), gWaveSamples[8]);
- gWaveSamples[8] += arg3 * 2;
+ AudioSynth_LoadBuffer(cmd++, DMEM_UNCOMPRESSED_NOTE, ALIGN16(nSamplesToLoad * 2), gWaveSamples[8]);
+ gWaveSamples[8] += nSamplesToLoad * 2;
return cmd;
} else {
- aLoadBuffer(cmd++, noteSubEu->sound.samples, 0x580, 0x80);
- if (temp_t0 != 0) {
- samplePosInt = (samplePosInt * D_801304C0[temp_t0 >> 2]) / D_801304C0[temp_t0 & 3];
+ aLoadBuffer(cmd++, noteSubEu->sound.samples, DMEM_UNCOMPRESSED_NOTE, 0x80);
+ if (unk6 != 0) {
+ samplePosInt = (samplePosInt * D_801304C0[unk6 >> 2]) / D_801304C0[unk6 & 3];
}
samplePosInt &= 0x3F;
temp_v0 = 0x40 - samplePosInt;
- if (temp_v0 < arg3 && ((arg3 - temp_v0 + 0x3F) / 0x40) != 0) {
- aDuplicate(cmd++, ((arg3 - temp_v0 + 0x3F) / 0x40), 0x580, 0x600, 0x80);
+ if (temp_v0 < nSamplesToLoad) {
+ repeats = ((nSamplesToLoad - temp_v0 + 0x3F) / 0x40);
+ if (repeats != 0) {
+ aDuplicate(cmd++, repeats, DMEM_UNCOMPRESSED_NOTE, DMEM_UNCOMPRESSED_NOTE + 0x80, 0x80);
+ }
}
synthState->samplePosInt = samplePosInt;
}
return cmd;
}
-#ifdef NON_MATCHING
-Acmd* func_800DDB64(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s32 arg3, s32 arg4, s32 arg5) {
- s32 phi_v0;
- s32 phi_v1;
- s32 phi_t0;
+Acmd* AudioSynth_NoteApplyHeadsetPanEffects(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s32 bufLen,
+ s32 flags, s32 side) {
+ u16 dest;
+ u16 pitch;
+ u8 prevPanShift;
+ u8 panShift;
- switch (arg5) {
+ switch (side) {
case 1:
- phi_t0 = 0x940;
- phi_v1 = noteSubEu->headsetPanRight;
- phi_v0 = synthState->prevHeadsetPanRight;
+ dest = DMEM_LEFT_CH;
+ panShift = noteSubEu->headsetPanRight;
+ prevPanShift = synthState->prevHeadsetPanRight;
synthState->prevHeadsetPanLeft = 0;
- synthState->prevHeadsetPanRight = phi_v1;
+ synthState->prevHeadsetPanRight = panShift;
break;
case 2:
- phi_t0 = 0xAE0;
- phi_v1 = noteSubEu->headsetPanLeft;
- phi_v0 = synthState->prevHeadsetPanLeft;
- synthState->prevHeadsetPanLeft = phi_v1;
+ dest = DMEM_RIGHT_CH;
+ panShift = noteSubEu->headsetPanLeft;
+ prevPanShift = synthState->prevHeadsetPanLeft;
+ synthState->prevHeadsetPanLeft = panShift;
synthState->prevHeadsetPanRight = 0;
break;
default:
return cmd;
}
- if (arg4 != 1) {
- if (phi_v0 != phi_v1) {
- aSetBuffer(cmd++, 0, 0x5C0, 0x3C0, arg3 + phi_v1 - phi_v0);
- aResampleZOH(cmd++, (u16)((((arg3 << 0xF) / 2) - 1) / ((arg3 + phi_v1 - phi_v0 - 2) / 2)), 0);
+ if (flags != A_INIT) {
+ // Slightly adjust the sample rate in order to fit a change in pan shift
+ if (panShift != prevPanShift) {
+ pitch = (((bufLen << 0xF) / 2) - 1) / ((bufLen + panShift - prevPanShift - 2) / 2);
+ aSetBuffer(cmd++, 0, DMEM_NOTE_PAN_TEMP, DMEM_TEMP, bufLen + panShift - prevPanShift);
+ aResampleZoh(cmd++, pitch, 0);
} else {
- aDMEMMove(cmd++, 0x5C0, 0x3C0, arg3);
+ aDMEMMove(cmd++, DMEM_NOTE_PAN_TEMP, DMEM_TEMP, bufLen);
}
- if (phi_v0 != 0) {
- aLoadBuffer(cmd++, &synthState->synthesisBuffers->panResampleState[0x8], 0x5C0, ALIGN16(phi_v0));
- aDMEMMove(cmd++, 0x3C0, 0x5C0 + phi_v0, arg3 + phi_v1 - phi_v0);
+ if (prevPanShift != 0) {
+ aLoadBuffer(cmd++, &synthState->synthesisBuffers->panResampleState[0x8], DMEM_NOTE_PAN_TEMP,
+ ALIGN16(prevPanShift));
+ aDMEMMove(cmd++, DMEM_TEMP, DMEM_NOTE_PAN_TEMP + prevPanShift, bufLen + panShift - prevPanShift);
} else {
- aDMEMMove(cmd++, 0x3C0, 0x5C0, arg3 + phi_v1);
+ aDMEMMove(cmd++, DMEM_TEMP, DMEM_NOTE_PAN_TEMP, bufLen + panShift);
}
} else {
- aDMEMMove(cmd++, 0x5C0, 0x3C0, arg3);
- aClearBuffer(cmd++, 0x5C0, phi_v1);
- aDMEMMove(cmd++, 0x3C0, 0x5C0 + phi_v1, arg3);
+ // Just shift right
+ aDMEMMove(cmd++, DMEM_NOTE_PAN_TEMP, DMEM_TEMP, bufLen);
+ aClearBuffer(cmd++, DMEM_NOTE_PAN_TEMP, panShift);
+ aDMEMMove(cmd++, DMEM_TEMP, DMEM_NOTE_PAN_TEMP + panShift, bufLen);
}
- if (phi_v1 != 0) {
- aSaveBuffer(cmd++, 0x5C0 + arg3, &synthState->synthesisBuffers->panResampleState[0x8], ALIGN16(phi_v1));
+ if (panShift) {
+ // Save excessive samples for next iteration
+ aSaveBuffer(cmd++, DMEM_NOTE_PAN_TEMP + bufLen, &synthState->synthesisBuffers->panResampleState[0x8],
+ ALIGN16(panShift));
}
- aAddMixer(cmd++, ((arg3 + 0x3F) & ~0x3F), 0x5C0, phi_t0, 0x7FFF);
+ aAddMixer(cmd++, ALIGN64(bufLen), DMEM_NOTE_PAN_TEMP, dest, 0x7FFF);
return cmd;
}
-#else
-#pragma GLOBAL_ASM("asm/non_matchings/code/audio_synthesis/func_800DDB64.s")
-#endif
diff --git a/src/code/code_800E4FE0.c b/src/code/code_800E4FE0.c
index 70c4ab979..b36cd0bdc 100644
--- a/src/code/code_800E4FE0.c
+++ b/src/code/code_800E4FE0.c
@@ -18,7 +18,7 @@ typedef enum {
CHAN_UPD_VIBE_X32, // 11
CHAN_UPD_UNK_0F, // 12
CHAN_UPD_UNK_20, // 13
- CHAN_UPD_REVERB_FLG // 14
+ CHAN_UPD_STEREO // 14
} ChannelUpdateType;
void func_800E6300(SequenceChannel* channel, AudioCmd* arg1);
@@ -704,8 +704,8 @@ void func_800E6300(SequenceChannel* channel, AudioCmd* cmd) {
case CHAN_UPD_UNK_20:
channel->unk_20 = cmd->asUShort;
return;
- case CHAN_UPD_REVERB_FLG:
- channel->reverbBits.asByte = cmd->asUbyte;
+ case CHAN_UPD_STEREO:
+ channel->stereo.asByte = cmd->asUbyte;
return;
}
}
@@ -792,10 +792,10 @@ s32 func_800E66C0(s32 arg0) {
if (temp_a2->adsr.action.s.state != 0) {
if (arg0 >= 2) {
sound = temp_a3->sound.audioBankSound;
- if (sound == NULL || temp_a3->bitField1.s.bit2) {
+ if (sound == NULL || temp_a3->bitField1.s.isSyntheticWave) {
continue;
}
- if (sound->sample->bits2 == 0) {
+ if (sound->sample->medium == 0) {
continue;
}
}
diff --git a/src/code/code_800EC960.c b/src/code/code_800EC960.c
index 7b9ee4387..e107dd93a 100644
--- a/src/code/code_800EC960.c
+++ b/src/code/code_800EC960.c
@@ -6,7 +6,7 @@ typedef struct {
f32 freqScale;
s8 reverb;
s8 panSigned;
- s8 reverbFlg;
+ s8 stereoBits;
u8 unk_0B;
u8 unk_0C;
} unk_s1;
@@ -2754,7 +2754,7 @@ void Audio_SetSoundProperties(u8 bankIdx, u8 entryIdx, u8 channelIdx) {
s8 reverb = 0;
f32 freqScale = 1.0f;
s8 panSigned = 0x40;
- u8 reverbFlg = 0;
+ u8 stereoBits = 0;
u8 sp39 = 0;
s8 sp38 = 0;
f32 sp34;
@@ -2781,17 +2781,17 @@ void Audio_SetSoundProperties(u8 bankIdx, u8 entryIdx, u8 channelIdx) {
sp34 = D_801305C4[(entry->unk_26 & 0x400) >> 10];
if (!(entry->unk_26 & 0x800)) {
if (*entry->posZ < sp34) {
- reverbFlg = 0x10;
+ stereoBits = 0x10;
}
- if ((D_8016B8B8[channelIdx].reverbFlg ^ reverbFlg) & 0x10) {
+ if ((D_8016B8B8[channelIdx].stereoBits ^ stereoBits) & 0x10) {
if (panSigned < 0x40) {
- reverbFlg = D_8016B8B8[channelIdx].reverbFlg ^ 0x14;
+ stereoBits = D_8016B8B8[channelIdx].stereoBits ^ 0x14;
} else {
- reverbFlg = D_8016B8B8[channelIdx].reverbFlg ^ 0x18;
+ stereoBits = D_8016B8B8[channelIdx].stereoBits ^ 0x18;
}
} else {
- reverbFlg = D_8016B8B8[channelIdx].reverbFlg;
+ stereoBits = D_8016B8B8[channelIdx].stereoBits;
}
}
}
@@ -2828,9 +2828,9 @@ void Audio_SetSoundProperties(u8 bankIdx, u8 entryIdx, u8 channelIdx) {
Audio_QueueCmdF32(0x4020000 | (channelIdx << 8), freqScale);
D_8016B8B8[channelIdx].freqScale = freqScale;
}
- if (reverbFlg != D_8016B8B8[channelIdx].reverbFlg) {
- Audio_QueueCmdS8(0xE020000 | (channelIdx << 8), reverbFlg | 0x10);
- D_8016B8B8[channelIdx].reverbFlg = reverbFlg;
+ if (stereoBits != D_8016B8B8[channelIdx].stereoBits) {
+ Audio_QueueCmdS8(0xE020000 | (channelIdx << 8), stereoBits | 0x10);
+ D_8016B8B8[channelIdx].stereoBits = stereoBits;
}
if (sp39 != D_8016B8B8[channelIdx].unk_0B) {
// CHAN_UPD_SCRIPT_IO (slot 3)
@@ -2860,7 +2860,7 @@ void func_800F3ED4(void) {
t->freqScale = 1.0f;
t->reverb = 0;
t->panSigned = 0x40;
- t->reverbFlg = 0;
+ t->stereoBits = 0;
t->unk_0B = 0xFF;
t->unk_0C = 0xFF;
}
diff --git a/src/code/ucode_disas.c b/src/code/ucode_disas.c
index 81301f2cc..00335dd20 100644
--- a/src/code/ucode_disas.c
+++ b/src/code/ucode_disas.c
@@ -283,3 +283,8 @@ void UCodeDisas_RegisterUCode(UCodeDisas* this, s32 count, UCodeInfo* ucodeArray
void UCodeDisas_SetCurUCode(UCodeDisas* this, void* ptr) {
UCodeDisas_SetCurUCodeImpl(this, ptr);
}
+
+// 4 bytes of nops, separating this file from audio_synthesis and padding .text
+// to a 32-byte boundary. Unclear what this comes from... maybe the audio
+// library was built separately and aligned to 32 bytes?
+#pragma GLOBAL_ASM("asm/non_matchings/code/ucode_disas/pad_800DACB0.s")
generated by cgit v1.2.3 (git 2.25.1) at 2026-08-03 13:13:32 +0000