1
0
mirror of https://github.com/zeldaret/oot.git synced 2024-09-22 21:35:27 +00:00
oot/src/code/audio_synthesis.c
krimtonz d1a5ea5110
Audio WIP (#836)
* being code_800EC960

* wip

* wip

* more audio progress

* migrate data in code_800E11F0

* wip

* make ok

* remove asm

* wip

* move some variables outside of gAudioContext to the AudioContext structure due to the size used in func_800E3094

* more progress

* wip

* wip

* split code_800E11F0

* migrate rodata in code_800E11F0

* match functions that couldn't because of data issues

* move code_800E4FE0 asm files

* wip

* more wip

* fix global functions, and forward declarations

* wip

* wip

* wip

* ocarina wip

* match a couple functions

* some progress

* separate some bss

* match func_800EDA3C

* some matching

* more matches

* migrate audio rodata

* some matches

* more matchess

* start on synthesis

* work on synthesis

* fix function declaration

* Merge branch 'master' into audio

* match a few more functions

* wip

* wip

* more matching, rename Audio_SetBGM to Audio_QueueSeqCmd

* name several audio functions, and audiocontext members

* more naming, rename code_800E11F0 to audio_load, code_800DAAC0 to audio_synthesis

* audio wip

* match a few more functions.

* wip

* add missing NON_MATCHING directive

* wip

* some matching, data reogranization

* match cursed function

* wip

* wip

* formatting

* remove prefix from struct memebers

* missed function rename

* review

Co-authored-by: fig02 <fig02srl@gmail.com>
2021-07-27 19:44:58 -04:00

1200 lines
41 KiB
C

#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[];
#pragma GLOBAL_ASM("asm/non_matchings/code/audio_synthesis/pad_800DACB0.s")
void AudioSynth_InitNextRingBuf(s32 chunkSize, s32 bufIdx, s32 reverbIdx) {
ReverbRingBufferItem* bufItem;
s32 pad[3];
SynthesisReverb* reverb = &gAudioContext.synthesisReverbs[reverbIdx];
s32 temp_a0_2;
s32 temp_a0_4;
s32 sampleCnt;
s32 extraSamples;
s32 i;
s32 j;
if (reverb->downsampleRate >= 2) {
if (reverb->framesToIgnore == 0) {
bufItem = &reverb->items[reverb->curFrame][bufIdx];
// inval dcache
Audio_osInvalDCache(bufItem->toDownsampleLeft, 0x340);
for (j = 0, i = 0; i < bufItem->lengthA / 2; j += reverb->downsampleRate, i++) {
reverb->leftRingBuf[bufItem->startPos + i] = bufItem->toDownsampleLeft[j];
reverb->rightRingBuf[bufItem->startPos + i] = bufItem->toDownsampleRight[j];
}
for (i = 0; i < bufItem->lengthB / 2; j += reverb->downsampleRate, i++) {
reverb->leftRingBuf[i] = bufItem->toDownsampleLeft[j];
reverb->rightRingBuf[i] = bufItem->toDownsampleRight[j];
}
}
}
bufItem = &reverb->items[reverb->curFrame][bufIdx];
sampleCnt = chunkSize / reverb->downsampleRate;
extraSamples = (sampleCnt + reverb->nextRingBufPos) - reverb->bufSizePerChan;
temp_a0_2 = reverb->nextRingBufPos;
if (extraSamples < 0) {
bufItem->lengthA = sampleCnt * 2;
bufItem->lengthB = 0;
bufItem->startPos = reverb->nextRingBufPos;
reverb->nextRingBufPos += sampleCnt;
} else {
bufItem->lengthA = (sampleCnt - extraSamples) * 2;
bufItem->lengthB = extraSamples * 2;
bufItem->startPos = reverb->nextRingBufPos;
reverb->nextRingBufPos = extraSamples;
}
bufItem->numSamplesAfterDownsampling = sampleCnt;
bufItem->chunkLen = chunkSize;
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;
extraSamples = (temp_a0_4 + sampleCnt) - reverb->bufSizePerChan;
if (extraSamples < 0) {
bufItem->lengthA = sampleCnt * 2;
bufItem->lengthB = 0;
bufItem->startPos = temp_a0_4;
} else {
bufItem->lengthA = (sampleCnt - extraSamples) * 2;
bufItem->lengthB = extraSamples * 2;
bufItem->startPos = temp_a0_4;
}
bufItem->numSamplesAfterDownsampling = sampleCnt;
bufItem->chunkLen = chunkSize;
}
}
void func_800DB03C(s32 arg0) {
NoteSubEu* temp_v0_2;
NoteSubEu* subEu;
s32 t;
s32 i;
t = gAudioContext.maxSimultaneousNotes * arg0;
for (i = 0; i < gAudioContext.maxSimultaneousNotes; i++) {
subEu = &gAudioContext.notes[i].noteSubEu;
temp_v0_2 = &gAudioContext.noteSubsEu[t + i];
if (subEu->bitField0.s.enabled) {
subEu->bitField0.s.needsInit = 0;
} else {
temp_v0_2->bitField0.s.enabled = 0;
}
subEu->unk_06 = 0;
}
}
Acmd* AudioSynth_Update(Acmd* cmdStart, s32* cmdCnt, s16* aiStart, s32 aiBufLen) {
s32 chunkSize;
s16* aiBufP;
Acmd* cmdP;
s32 i;
s32 j;
SynthesisReverb* reverb;
cmdP = cmdStart;
for (i = gAudioContext.audioBufferParameters.updatesPerFrame; i > 0; i--) {
Audio_ProcessSequences(i - 1);
func_800DB03C(gAudioContext.audioBufferParameters.updatesPerFrame - i);
}
aiBufP = aiStart;
gAudioContext.unk_0x10 = 0;
for (i = gAudioContext.audioBufferParameters.updatesPerFrame; i > 0; i--) {
if (i == 1) {
chunkSize = aiBufLen;
} else if ((aiBufLen / i) >= gAudioContext.audioBufferParameters.samplesPerUpdateMax) {
chunkSize = gAudioContext.audioBufferParameters.samplesPerUpdateMax;
} else if (gAudioContext.audioBufferParameters.samplesPerUpdateMin >= (aiBufLen / i)) {
chunkSize = gAudioContext.audioBufferParameters.samplesPerUpdateMin;
} else {
chunkSize = gAudioContext.audioBufferParameters.samplesPerUpdate;
}
for (j = 0; j < gAudioContext.numSynthesisReverbs; j++) {
if (gAudioContext.synthesisReverbs[j].useReverb) {
AudioSynth_InitNextRingBuf(chunkSize, gAudioContext.audioBufferParameters.updatesPerFrame - i, j);
}
}
cmdP = func_800DC384(aiBufP, chunkSize, cmdP, gAudioContext.audioBufferParameters.updatesPerFrame - i);
aiBufLen -= chunkSize;
aiBufP += chunkSize * 2;
}
for (j = 0; j < gAudioContext.numSynthesisReverbs; j++) {
if (gAudioContext.synthesisReverbs[j].framesToIgnore != 0) {
gAudioContext.synthesisReverbs[j].framesToIgnore--;
}
gAudioContext.synthesisReverbs[j].curFrame ^= 1;
}
*cmdCnt = cmdP - cmdStart;
return cmdP;
}
void func_800DB2C0(s32 arg0, s32 arg1) {
NoteSubEu* temp_v1;
s32 i;
for (i = arg0 + 1; i < gAudioContext.audioBufferParameters.updatesPerFrame; i++) {
temp_v1 = &gAudioContext.noteSubsEu[(gAudioContext.maxSimultaneousNotes * i) + arg1];
if (!temp_v1->bitField0.s.needsInit) {
temp_v1->bitField0.s.enabled = 0;
} else {
break;
}
}
}
Acmd* func_800DB330(Acmd* arg0, SynthesisReverb* reverb, s16 arg2) {
ReverbRingBufferItem* bufItem = &reverb->items[reverb->curFrame][arg2];
arg0 = AudioSynth_LoadRingBuffer(arg0, 0x3E0, bufItem->startPos, bufItem->lengthA, reverb);
if (bufItem->lengthB != 0) {
arg0 = AudioSynth_LoadRingBuffer(arg0, bufItem->lengthA + 0x3E0, 0, bufItem->lengthB, reverb);
}
return arg0;
}
Acmd* func_800DB3D8(Acmd* arg0, SynthesisReverb* arg1, s16 arg2) {
ReverbRingBufferItem* temp_v1;
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);
}
return arg0;
}
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* func_800DB4E4(Acmd* arg0, s32 arg1, SynthesisReverb* arg2, s16 arg3) {
ReverbRingBufferItem* temp_v1;
s16 temp_t0_2;
s16 temp_t1_2;
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);
}
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;
}
Acmd* func_800DB680(Acmd* cmd, SynthesisReverb* reverb, s16 bufIdx) {
ReverbRingBufferItem* bufItem = &reverb->items[reverb->curFrame][bufIdx];
aSetBuffer(cmd++, 0, 0xC80, 0x720, 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);
if (bufItem->lengthB != 0) {
cmd = AudioSynth_SaveBuffer(cmd, bufItem->lengthA + 0x720, 0, bufItem->lengthB, reverb->leftRingBuf);
}
aSetBuffer(cmd++, 0, 0xE20, 0x720, 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);
if (bufItem->lengthB != 0) {
cmd = AudioSynth_SaveBuffer(cmd, bufItem->lengthA + 0x720, 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;
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);
}
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;
}
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);
}
if (arg2->unk_274 != NULL) {
aFilter(arg0++, 2, arg1, arg2->unk_274);
aFilter(arg0++, arg2->resampleFlags, 0xE20, arg2->unk_27C);
}
return arg0;
}
Acmd* func_800DBAE8(Acmd* arg0, SynthesisReverb* arg1, s32 arg2) {
SynthesisReverb* temp_a3;
temp_a3 = &gAudioContext.synthesisReverbs[arg1->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);
}
return arg0;
}
void func_800DBB94(void) {
}
void func_800DBB9C(Acmd* arg0, s32 arg1, s32 arg2) {
aClearBuffer(arg0, arg1, arg2);
}
void func_800DBBBC(void) {
}
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 func_800DBC08(void) {
}
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 func_800DBC54(void) {
}
void func_800DBC5C(void) {
}
// possible fake match?
void AudioSynth_DMemMove(Acmd* cmd, s32 dmemIn, s32 dmemOut, s32 count) {
cmd->words.w0 = _SHIFTL(A_DMEMMOVE, 24, 8) | _SHIFTL(dmemIn, 0, 24);
cmd->words.w1 = _SHIFTL(dmemOut, 16, 16) | _SHIFTL(count, 0, 16);
}
void func_800DBC90(void) {
}
void func_800DBC98(void) {
}
void func_800DBCA0(void) {
}
void func_800DBCA8(void) {
}
void AudioSynth_InterL(Acmd* cmd, s32 dmemIn, s32 dmemOut, s32 count) {
cmd->words.w0 = _SHIFTL(A_INTERL, 24, 8) | _SHIFTL(count, 0, 16);
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 func_800DBD08(void) {
}
void func_800DBD10(Acmd* arg0, s32 arg1, s32 arg2, s32 arg3) {
aLoadBuffer(arg0, arg3, arg1, arg2);
}
void func_800DBD38(Acmd* arg0, s32 arg1, s32 arg2, s32 arg3) {
aSaveBuffer(arg0, arg1, arg3, arg2);
}
void AudioSynth_EnvSetup2(Acmd* cmd, s32 volLeft, s32 volRight) {
cmd->words.w0 = _SHIFTL(A_ENVSETUP2, 24, 8);
cmd->words.w1 = _SHIFTL(volLeft, 16, 16) | _SHIFTL(volRight, 0, 16);
}
void func_800DBD7C(void) {
}
void func_800DBD84(void) {
}
void func_800DBD8C(void) {
}
void AudioSynth_UnkCmd17(Acmd* arg0, s32 arg1, s32 arg2) {
aUnkCmd17(arg0, arg1, arg2);
}
void AudioSynth_HiLoGain(Acmd* cmd, s32 gain, s32 dmemIn, s32 dmemOut, s32 count) {
cmd->words.w0 = _SHIFTL(A_HILOGAIN, 24, 8) | _SHIFTL(gain, 16, 8) | _SHIFTL(count, 0, 16);
cmd->words.w1 = _SHIFTL(dmemIn, 16, 16) | _SHIFTL(dmemOut, 0, 16);
}
void AudioSynth_UnkCmd19(Acmd* cmd, s32 arg1, s32 arg2, s32 arg3, s32 arg4) {
cmd->words.w0 = _SHIFTL(A_UNK19, 24, 8) | _SHIFTL(arg4, 16, 8) | _SHIFTL(arg3, 0, 16);
cmd->words.w1 = _SHIFTL(arg1, 16, 16) | _SHIFTL(arg2, 0, 16);
}
void func_800DBE18(void) {
}
void func_800DBE20(void) {
}
void func_800DBE28(void) {
}
void func_800DBE30(void) {
}
void AudioSynth_UnkCmd3(Acmd* cmd, s32 arg1, s32 arg2, s32 arg3) {
cmd->words.w0 = _SHIFTL(A_UNK3, 24, 8) | _SHIFTL(arg3, 0, 16);
cmd->words.w1 = _SHIFTL(arg1, 16, 16) | _SHIFTL(arg2, 0, 16);
}
void func_800DBE5C(void) {
}
void func_800DBE64(void) {
}
void func_800DBE6C(void) {
}
void AudioSynth_SetFilter(Acmd* cmd, s32 flags, s32 countOrBuf, s32 addr) {
aFilter(cmd, flags, countOrBuf, addr);
}
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];
cmd = AudioSynth_LoadRingBuffer(cmd, 0xC80, ringBufferItem->startPos, ringBufferItem->lengthA, reverb);
if (ringBufferItem->lengthB != 0) {
cmd = AudioSynth_LoadRingBuffer(cmd, ringBufferItem->lengthA + 0xC80, 0, ringBufferItem->lengthB, reverb);
}
return cmd;
}
Acmd* func_800DBF5C(Acmd* cmd, s32 arg1, SynthesisReverb* reverb, s16 bufIdx) {
ReverbRingBufferItem* bufItem = &reverb->items2[reverb->curFrame][bufIdx];
cmd = AudioSynth_LoadRingBuffer(cmd, 0xC80, bufItem->startPos, bufItem->lengthA, reverb);
if (bufItem->lengthB != 0) {
cmd = AudioSynth_LoadRingBuffer(cmd, bufItem->lengthA + 0xC80, 0, bufItem->lengthB, reverb);
}
return cmd;
}
Acmd* AudioSynth_LoadRingBuffer(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);
return cmd;
}
Acmd* AudioSynth_SaveRingBuffer(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);
return cmd;
}
Acmd* AudioSynth_SaveBuffer(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) {
if (reverb->downsampleRate == 1) {
cmd = func_800DBF5C(cmd, arg1, reverb, arg3);
}
return cmd;
}
Acmd* func_800DC164(Acmd* cmd, s32 arg1, SynthesisReverb* reverb, s16 arg3) {
if (reverb->downsampleRate == 1) {
if (reverb->unk_18 != 0) {
cmd = func_800DB828(cmd, arg1, reverb, arg3);
} else {
cmd = func_800DBEB0(cmd, arg1, reverb, arg3);
}
} else {
cmd = func_800DB4E4(cmd, arg1, reverb, arg3);
}
return cmd;
}
Acmd* func_800DC1D8(Acmd* cmd, SynthesisReverb* reverb, s16 bufIdx) {
ReverbRingBufferItem* bufItem = &reverb->items[reverb->curFrame][bufIdx];
if (reverb->downsampleRate == 1) {
if (reverb->unk_18 != 0) {
cmd = func_800DB680(cmd, reverb, bufIdx);
} else {
cmd = AudioSynth_SaveRingBuffer(cmd, 0xC80, bufItem->startPos, bufItem->lengthA, reverb);
if (bufItem->lengthB != 0) {
cmd = AudioSynth_SaveRingBuffer(cmd, bufItem->lengthA + 0xC80, 0, bufItem->lengthB, reverb);
}
}
} else {
func_800DBD38(cmd++, 0xC80, 0x340, reverb->items[reverb->curFrame][bufIdx].toDownsampleLeft);
}
reverb->resampleFlags = 0;
return cmd;
}
Acmd* func_800DC2DC(Acmd* cmd, SynthesisReverb* reverb, s16 bufIdx) {
ReverbRingBufferItem* bufItem = &reverb->items2[reverb->curFrame][bufIdx];
cmd = AudioSynth_SaveRingBuffer(cmd, 0xC80, bufItem->startPos, bufItem->lengthA, reverb);
if (bufItem->lengthB != 0) {
cmd = AudioSynth_SaveRingBuffer(cmd, bufItem->lengthA + 0xC80, 0, bufItem->lengthB, reverb);
}
return cmd;
}
#ifdef NON_MATCHING
Acmd* func_800DC384(s16* aiBuf, s32 aiBufLen, Acmd* cmd, s32 updateIdx) {
NoteSubEu* phi_v0;
u8 sp9C[88];
s16 phi_s2;
s16 phi_s4;
s32 t;
s32 useReverb;
SynthesisReverb* temp_t8;
u8 t2;
s32 i;
s32 unk14;
t = gAudioContext.maxSimultaneousNotes * updateIdx;
phi_s2 = 0;
if (gAudioContext.numSynthesisReverbs == 0) {
for (i = 0; i < gAudioContext.maxSimultaneousNotes; i++) {
phi_v0 = &gAudioContext.noteSubsEu[t + i];
if (phi_v0->bitField0.s.enabled) {
sp9C[phi_s2++] = i;
}
}
i = 0;
} else {
for (phi_s4 = 0; phi_s4 < gAudioContext.numSynthesisReverbs; phi_s4++) {
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;
}
}
}
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;
}
}
i = 0;
}
aClearBuffer(cmd++, 0x940, 0x340);
for (phi_s4 = 0; phi_s4 < gAudioContext.numSynthesisReverbs; phi_s4++) {
temp_t8 = &gAudioContext.synthesisReverbs[phi_s4];
useReverb = temp_t8->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;
if (unk14) {
aDMEMMove(cmd++, 0xC80, 0x3E0, 0x340);
}
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);
}
if (unk14) {
cmd = func_800DC1D8(cmd, temp_t8, updateIdx);
if (temp_t8->unk_05 != -1) {
cmd = func_800DBAE8(cmd, temp_t8, updateIdx);
}
cmd = func_800DC124(cmd, aiBufLen, temp_t8, updateIdx);
aMix(cmd++, 0x34, temp_t8->unk_16, 0x3E0, 0xC80);
}
}
while (i < phi_s2) {
phi_v0 = &gAudioContext.noteSubsEu[sp9C[i] + t];
if (phi_v0->bitField1.s.reverbIndex == phi_s4) {
cmd = func_800DC910(sp9C[i], phi_v0, &gAudioContext.notes[sp9C[i]].synthesisState, aiBuf, aiBufLen,
cmd, updateIdx);
} else {
break;
}
i++;
}
if (useReverb) {
if ((temp_t8->unk_270 != NULL) || (temp_t8->unk_274 != NULL)) {
cmd = func_800DBA40(cmd, aiBufLen * 2, temp_t8);
}
if (unk14) {
cmd = func_800DC2DC(cmd, temp_t8, updateIdx);
} else {
cmd = func_800DC1D8(cmd, temp_t8, updateIdx);
if (temp_t8->unk_05 != -1) {
cmd = func_800DBAE8(cmd, temp_t8, updateIdx);
}
}
}
}
while (i < phi_s2) {
cmd = func_800DC910(sp9C[i], &gAudioContext.noteSubsEu[t + sp9C[i]],
&gAudioContext.notes[sp9C[i]].synthesisState, aiBuf, aiBufLen, cmd, updateIdx);
i++;
}
updateIdx = aiBufLen * 2;
aInterleave(cmd++, updateIdx, 0x3C0, 0x940, 0xAE0);
aSaveBuffer(cmd++, 0x3C0, aiBuf, updateIdx * 2);
return cmd;
}
#else
#pragma GLOBAL_ASM("asm/non_matchings/code/audio_synthesis/func_800DC384.s")
#endif
#ifdef NON_MATCHING
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;
s32 temp_v1_5;
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;
s32 phi_s3;
s32 phi_s1;
s32 phi_a1;
s32 phi_t1_3;
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) {
synthState->restart = 0;
synthState->unk_06 = 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->unk_1A = 1;
sp9C->noteSubEu.bitField0.s.finished = 0;
sp88 = 0;
sp138 = 1;
}
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;
// OLD BLOCK 94
} 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);
} else {
phi_fp = sp50;
}
if (sp150->bits4 == 0 || sp150->bits4 == 3) {
if (gAudioContext.unk_0x10 != sp150->bits4 + 8) {
switch (sp8C) {
case 1:
gAudioContext.unk_0x10 = D_8012FBAA;
;
break;
case 2:
gAudioContext.unk_0x10 = sp150->bits4 + 8;
break;
case 3:
default:
break;
}
aLoadADPCM(cmd++, sp150->book->order * 0x10 * sp150->book->npredictors, gAudioContext.unk_0x10);
}
}
sp104 = 0;
phi_s5 = 0;
while (sp104 != phi_fp) {
sp140 = 0;
sp13C = 0;
spFC = phi_fp - sp104;
phi_s3 = synthState->samplePosInt & 0xF;
if (phi_s3 == 0 && synthState->restart == 0) {
phi_s3 = 0x10;
}
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;
} else {
phi_s0 = (sp14C->end - synthState->samplePosInt) - (0x10 - phi_s3);
phi_s6 = (0x10 - phi_s3);
if (phi_s0 <= 0) {
phi_s0 = 0;
phi_s6 = (sp14C->end - synthState->samplePosInt);
}
phi_s1 = (s32)(phi_s0 + 0xF) / 0x10;
if (sp14C->count != 0) {
sp13C = 1;
phi_s7 = 0;
} else {
sp140 = 1;
phi_s7 = 0;
}
}
switch (sp150->bits4) {
case 0:
spCC = 0x10;
spC8 = 0;
spD4 = 9;
goto block_39;
case 3:
spCC = 0x10;
spC8 = 0;
spD4 = 5;
goto block_39;
case 1:
spCC = 0x10;
spC8 = 0;
spD4 = 0x10;
goto block_39;
case 2:
sp138 = 0;
sp120 = 0;
sp104 = phi_fp;
func_800DBB9C(cmd++, 0x580, (phi_fp * 2) + 0x20);
phi_s5 = phi_fp;
break;
case 5:
sp138 = 0;
sp120 = 0;
sp104 = phi_fp;
func_800DBB9C(cmd++, 0x580, (phi_fp * 2) + 0x20);
phi_s5 = phi_fp;
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_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 (synthState->restart) {
aSetLoop(cmd++, sp150->loop);
sp138 = 2;
synthState->restart = 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 (sp104 != 0) {
aDMEMMove(cmd++, phi_s4 + (phi_s3 * 2) + 0x580, phi_s5 + 0x580,
(((phi_s0 + phi_s6) - phi_s7) * 2));
}
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;
}
sp138 = 0;
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);
} else {
if (sp13C != 0) {
synthState->restart = 1;
synthState->samplePosInt = sp14C->start;
} else {
synthState->samplePosInt += spFC;
}
}
phi_v1_2 = spC0;
}
switch (spC0) {
case 1:
spAE = sp120 + 0x580;
break;
case 2:
break;
default:
switch (spBC) {
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);
}
break;
case 1:
AudioSynth_InterL(cmd++, sp120 + 0x580, spB0 + 0x3E0, ((phi_fp / 2) + 7) & ~7);
break;
}
}
}
}
sp138 = 0;
if (noteSubEu->bitField0.s.needsInit == 1) {
noteSubEu->bitField0.s.needsInit = 0;
sp138 = 1;
}
cmd = func_800DD62C(cmd++, synthState, aiBufLen * 2, sp136, spAE, sp138);
if (sp8C == 3) {
AudioSynth_UnkCmd19(cmd++, 0x3C0, 0x3C0, aiBufLen * 2, 0);
}
if (sp8C == 2) {
AudioSynth_UnkCmd3(cmd++, 0x3C0, 0x3C0, aiBufLen * 2);
}
phi_a1_2 = noteSubEu->unk_2;
if (phi_a1_2 != 0) {
if (phi_a1_2 < 0x10) {
phi_a1_2 = 0x10;
}
AudioSynth_HiLoGain(cmd++, phi_a1_2, 0x3C0, 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);
}
if (noteSubEu->unk_07 != 0 && noteSubEu->unk_10 != 0) {
AudioSynth_DMemMove(cmd++, 0x3C0, 0x760, aiBufLen * 2);
if (synthState->unk_1A != 0) {
func_800DBB9C(cmd++, 0x760 - noteSubEu->unk_07, noteSubEu->unk_07);
synthState->unk_1A = 0;
} else {
func_800DBD10(cmd++, 0x760 - noteSubEu->unk_07, noteSubEu->unk_07,
&synthState->synthesisBuffers->panSamplesBuffer[0x18]);
}
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);
} 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;
} else {
phi_s0 = 0;
}
cmd = func_800DD6CC(cmd, noteSubEu, synthState, aiBufLen, 0x3C0, phi_s0, sp138);
if (noteSubEu->bitField0.s.usesHeadsetPanEffects) {
if (!(sp138 & 1)) {
sp138 = 0;
}
cmd = func_800DDB64(cmd, noteSubEu, synthState, aiBufLen * 2, sp138, phi_s0);
}
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) {
if (pitch == 0) {
func_800DBB9C(cmd++, 0x3C0, count);
} else {
aSetBuffer(cmd++, 0, inpDmem, 0x3C0, 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) {
u32 phi_a1;
u16 curVolLeft;
u16 targetVolLeft;
s32 phi_t1;
s16 reverVol;
u16 curVolRight;
s16 phi_a3;
s16 phi_t0;
s16 sp52;
s16 sp50;
u16 targetVolRight;
s32 pad;
curVolLeft = synthState->curVolLeft;
targetVolLeft = noteSubEu->targetVolLeft;
targetVolLeft *= 0x10;
reverVol = noteSubEu->reverbVol;
curVolRight = synthState->curVolRight;
targetVolRight = noteSubEu->targetVolRight;
targetVolRight *= 0x10;
phi_a3 = targetVolLeft != curVolLeft ? (targetVolLeft - curVolLeft) / (aiBufLen >> 3) : 0;
phi_t0 = targetVolRight != curVolRight ? (targetVolRight - curVolRight) / (aiBufLen >> 3) : 0;
sp50 = synthState->samplePosFrac;
phi_t1 = sp50 & 0x7F;
if (sp50 != reverVol) {
sp52 = (((reverVol & 0x7F) - phi_t1) << 9) / (aiBufLen >> 3);
synthState->samplePosFrac = reverVol;
} else {
sp52 = 0;
}
synthState->curVolLeft = curVolLeft + (phi_a3 * (aiBufLen >> 3));
synthState->curVolRight = curVolRight + (phi_t0 * (aiBufLen >> 3));
if (noteSubEu->bitField1.s.hasTwoAdpcmParts) {
func_800DBB9C(cmd++, 0x5C0, 0x1A0);
func_800DBCD4(cmd++, phi_t1 * 2, sp52, phi_a3, phi_t0);
AudioSynth_EnvSetup2(cmd++, curVolLeft, curVolRight);
switch (arg5) {
case 1:
phi_a1 = D_801304A4;
break;
case 2:
phi_a1 = D_801304A8;
break;
default:
phi_a1 = D_801304AC;
break;
}
} else {
aEnvSetup1(cmd++, phi_t1 * 2, sp52, phi_a3, phi_t0);
aEnvSetup2(cmd++, curVolLeft, curVolRight);
phi_a1 = D_801304AC;
}
aEnvMixer(cmd++, arg4, aiBufLen, (sp50 & 0x80) >> 7, noteSubEu->bitField0.s.stereoHeadsetEffects,
noteSubEu->bitField0.s.usesHeadsetPanEffects, noteSubEu->bitField0.s.stereoStrongRight,
noteSubEu->bitField0.s.stereoStrongLeft, phi_a1);
return cmd;
}
extern u8 D_801304C0[];
Acmd* func_800DD9F4(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s32 arg3) {
s32 temp_v0;
s32 temp_t0;
s32 samplePosInt;
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;
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];
}
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);
}
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;
switch (arg5) {
case 1:
phi_v1 = noteSubEu->headsetPanRight;
phi_v0 = synthState->prevHeadsetPanRight;
synthState->prevHeadsetPanLeft = 0;
synthState->prevHeadsetPanRight = noteSubEu->headsetPanRight;
phi_t0 = 0x940;
break;
case 2:
phi_v1 = noteSubEu->headsetPanLeft;
phi_v0 = synthState->prevHeadsetPanLeft;
synthState->prevHeadsetPanLeft = noteSubEu->headsetPanLeft;
synthState->prevHeadsetPanRight = 0;
phi_t0 = 0xAE0;
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);
} else {
aDMEMMove(cmd++, 0x5C0, 0x3C0, arg3);
}
if (phi_v0 != 0) {
aLoadBuffer(cmd++, &synthState->synthesisBuffers->panResampleState[0x8], 0x5C0, ALIGN16(phi_v0));
aDMEMMove(cmd++, 0x3C0, phi_v0 + 0x5C0, (arg3 + phi_v1) - phi_v0);
} else {
aDMEMMove(cmd++, 0x3C0, 0x5C0, (arg3 + phi_v1));
}
} else {
aDMEMMove(cmd++, 0x5C0, 0x3C0, arg3);
aClearBuffer(cmd++, 0x5C0, phi_v1);
aDMEMMove(cmd++, 0x3C0, phi_v1 + 0x5C0, arg3);
}
if (phi_v1 != 0) {
aSaveBuffer(cmd++, arg3 + 0x5C0, &synthState->synthesisBuffers->panResampleState[0x8], ALIGN16(phi_v1));
}
aAddMixer(cmd++, ((arg3 + 0x3F) & ~0x3F), 0x5C0, phi_t0, 0x7FFF);
return cmd;
}
#else
#pragma GLOBAL_ASM("asm/non_matchings/code/audio_synthesis/func_800DDB64.s")
#endif