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Shipwright/soh/src/code/audio_synthesis.c
Garrett Cox 7a41bd3878
Mirrored world enhancement (#1569) 
* Mirrored world PoC

* invert culling for health meter and A button action

* A few more fixes

* Fix for item equip animations

* Fix for pause triforce

* Mirror scenes with static backgrounds

* mirror minimap for mirror world

* mirror dungeon maps and icons on the pause menu

* mirror overworld map and icons on the pause menu

* mirror debug world movement

* mirror shops cursor and movement

* use flip flag

* Reverse crouch stab x axis for mirror mode

* use invert culling command and clean up culling logic

* Move mirror mode handler to mods and support random modes

* Small cvar tweaks

* mirror billboard score numbers and fix gyro horse mirrored inputs

---------

Co-authored-by: Adam Bird <archez39@me.com>
2023-06-13 08:46:15 -04:00

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#include <libultraship/libultra.h>
#include "global.h"
#include "soh/mixer.h"
#define DEFAULT_LEN_1CH 0x1A0
#define DEFAULT_LEN_2CH 0x340
#define DMEM_TEMP 0x3C0
#define DMEM_UNCOMPRESSED_NOTE 0x580
#define DMEM_NOTE_PAN_TEMP 0x5C0
#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);
u32 D_801304A0 = 0x13000000;
u32 D_801304A4 = 0x5CAEC8E2;
u32 D_801304A8 = 0x945CC8E2;
u32 D_801304AC = 0x94AEC8E2;
u16 D_801304B0[] = {
0x7FFF, 0xD001, 0x3FFF, 0xF001, 0x5FFF, 0x9001, 0x7FFF, 0x8001,
};
u8 D_801304C0[] = { 0x40, 0x20, 0x10, 0x8 };
void AudioSynth_InitNextRingBuf(s32 chunkLen, s32 bufIndex, s32 reverbIndex) {
ReverbRingBufferItem* bufItem;
s32 pad[3];
SynthesisReverb* reverb = &gAudioContext.synthesisReverbs[reverbIndex];
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][bufIndex];
Audio_InvalDCache(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];
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][bufIndex];
sampleCnt = chunkLen / 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 = 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][bufIndex];
sampleCnt = chunkLen / 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 = chunkLen;
}
}
void func_800DB03C(s32 arg0) {
NoteSubEu* subEu;
NoteSubEu* subEu2;
s32 baseIndex;
s32 i;
baseIndex = gAudioContext.numNotes * arg0;
for (i = 0; i < gAudioContext.numNotes; i++) {
subEu = &gAudioContext.notes[i].noteSubEu;
subEu2 = &gAudioContext.noteSubsEu[baseIndex + i];
if (subEu->bitField0.enabled) {
subEu->bitField0.needsInit = false;
} else {
subEu2->bitField0.enabled = false;
}
subEu->unk_06 = 0;
}
}
Acmd* AudioSynth_Update(Acmd* cmdStart, s32* cmdCnt, s16* aiStart, s32 aiBufLen) {
s32 chunkLen;
s16* aiBufP;
Acmd* cmdP;
s32 i;
s32 j;
SynthesisReverb* reverb;
cmdP = cmdStart;
for (i = gAudioContext.audioBufferParameters.updatesPerFrame; i > 0; i--) {
AudioSeq_ProcessSequences(i - 1);
func_800DB03C(gAudioContext.audioBufferParameters.updatesPerFrame - i);
}
aiBufP = aiStart;
gAudioContext.curLoadedBook = NULL;
for (i = gAudioContext.audioBufferParameters.updatesPerFrame; i > 0; i--) {
if (i == 1) {
chunkLen = aiBufLen;
} else if ((aiBufLen / i) >= gAudioContext.audioBufferParameters.samplesPerUpdateMax) {
chunkLen = gAudioContext.audioBufferParameters.samplesPerUpdateMax;
} else if (gAudioContext.audioBufferParameters.samplesPerUpdateMin >= (aiBufLen / i)) {
chunkLen = gAudioContext.audioBufferParameters.samplesPerUpdateMin;
} else {
chunkLen = gAudioContext.audioBufferParameters.samplesPerUpdate;
}
for (j = 0; j < gAudioContext.numSynthesisReverbs; j++) {
if (gAudioContext.synthesisReverbs[j].useReverb) {
AudioSynth_InitNextRingBuf(chunkLen, gAudioContext.audioBufferParameters.updatesPerFrame - i, j);
}
}
cmdP = AudioSynth_DoOneAudioUpdate(aiBufP, chunkLen, cmdP, gAudioContext.audioBufferParameters.updatesPerFrame - i);
aiBufLen -= chunkLen;
aiBufP += chunkLen * 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 updateIndexStart, s32 noteIndex) {
NoteSubEu* temp_v1;
s32 i;
for (i = updateIndexStart + 1; i < gAudioContext.audioBufferParameters.updatesPerFrame; i++) {
temp_v1 = &gAudioContext.noteSubsEu[(gAudioContext.numNotes * i) + noteIndex];
if (!temp_v1->bitField0.needsInit) {
temp_v1->bitField0.enabled = 0;
} else {
break;
}
}
}
Acmd* AudioSynth_LoadRingBuffer1AtTemp(Acmd* cmd, SynthesisReverb* reverb, s16 bufIndex) {
ReverbRingBufferItem* bufItem = &reverb->items[reverb->curFrame][bufIndex];
cmd = AudioSynth_LoadRingBufferPart(cmd, DMEM_WET_TEMP, bufItem->startPos, bufItem->lengthA, reverb);
if (bufItem->lengthB != 0) {
// Ring buffer wrapped
cmd = AudioSynth_LoadRingBufferPart(cmd, DMEM_WET_TEMP + bufItem->lengthA, 0, bufItem->lengthB, reverb);
}
return cmd;
}
Acmd* AudioSynth_SaveRingBuffer1AtTemp(Acmd* cmd, SynthesisReverb* reverb, s16 bufIndex) {
ReverbRingBufferItem* bufItem = &reverb->items[reverb->curFrame][bufIndex];
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 cmd;
}
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* cmd, s32 arg1, SynthesisReverb* reverb, s16 arg3) {
ReverbRingBufferItem* item = &reverb->items[reverb->curFrame][arg3];
s16 offsetA;
s16 offsetB;
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(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 bufIndex) {
ReverbRingBufferItem* bufItem = &reverb->items[reverb->curFrame][bufIndex];
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_SaveBufferOffset(cmd, DMEM_WET_SCRATCH, bufItem->startPos, bufItem->lengthA, reverb->leftRingBuf);
if (bufItem->lengthB != 0) {
// Ring buffer wrapped
cmd = AudioSynth_SaveBufferOffset(cmd, DMEM_WET_SCRATCH + bufItem->lengthA, 0, bufItem->lengthB,
reverb->leftRingBuf);
}
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_SaveBufferOffset(cmd, DMEM_WET_SCRATCH, bufItem->startPos, bufItem->lengthA, reverb->rightRingBuf);
if (bufItem->lengthB != 0) {
// Ring buffer wrapped
cmd = AudioSynth_SaveBufferOffset(cmd, DMEM_WET_SCRATCH + bufItem->lengthA, 0, bufItem->lengthB,
reverb->rightRingBuf);
}
return cmd;
}
Acmd* func_800DB828(Acmd* cmd, s32 arg1, SynthesisReverb* reverb, s16 arg3) {
ReverbRingBufferItem* item = &reverb->items[reverb->curFrame][arg3];
s16 offsetA;
s16 offsetB;
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(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* 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 (reverb->filterRight != NULL) {
aFilter(cmd++, 2, count, reverb->filterRight);
aFilter(cmd++, reverb->resampleFlags, DMEM_WET_RIGHT_CH, reverb->filterRightState);
}
return cmd;
}
Acmd* AudioSynth_MaybeMixRingBuffer1(Acmd* cmd, SynthesisReverb* reverb, s32 arg2) {
SynthesisReverb* temp_a3;
temp_a3 = &gAudioContext.synthesisReverbs[reverb->unk_05];
if (temp_a3->downsampleRate == 1) {
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 cmd;
}
void func_800DBB94(void) {
}
void AudioSynth_ClearBuffer(Acmd* cmd, s32 arg1, s32 arg2) {
aClearBuffer(cmd, arg1, arg2);
}
void func_800DBBBC(void) {
}
void func_800DBBC4(void) {
}
void func_800DBBCC(void) {
}
void AudioSynth_Mix(Acmd* cmd, s32 arg1, s32 arg2, s32 arg3, s32 arg4) {
aMix(cmd, arg1, arg2, arg3, arg4);
}
void func_800DBC08(void) {
}
void func_800DBC10(void) {
}
void func_800DBC18(void) {
}
void AudioSynth_SetBuffer(Acmd* cmd, s32 flags, s32 dmemIn, s32 dmemOut, s32 count) {
aSetBuffer(cmd, flags, 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) {
aDMEMMove(cmd, dmemIn, dmemOut, count);
}
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) {
aInterl(cmd, dmemIn, dmemOut, count);
}
void AudioSynth_EnvSetup1(Acmd* cmd, s32 arg1, s32 arg2, s32 arg3, s32 arg4) {
aEnvSetup1(cmd, arg1, arg2, arg3, arg4);
}
void func_800DBD08(void) {
}
void AudioSynth_LoadBuffer(Acmd* cmd, s32 arg1, s32 arg2, uintptr_t arg3) {
aLoadBuffer(cmd, arg3, arg1, arg2);
}
void AudioSynth_SaveBuffer(Acmd* cmd, s32 arg1, s32 arg2, uintptr_t arg3) {
aSaveBuffer(cmd, arg1, arg3, arg2);
}
void AudioSynth_EnvSetup2(Acmd* cmd, s32 volLeft, s32 volRight) {
aEnvSetup2(cmd, volLeft, volRight);
}
void func_800DBD7C(void) {
}
void func_800DBD84(void) {
}
void func_800DBD8C(void) {
}
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) {
aHiLoGain(cmd, gain, count, dmemIn, dmemOut);
}
void AudioSynth_UnkCmd19(Acmd* cmd, s32 arg1, s32 arg2, s32 arg3, s32 arg4) {
aUnkCmd19(cmd, arg1, arg2, arg3, arg4);
}
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) {
aUnkCmd3(cmd, arg1, arg2, arg3);
}
void func_800DBE5C(void) {
}
void func_800DBE64(void) {
}
void func_800DBE6C(void) {
}
void AudioSynth_LoadFilter(Acmd* cmd, s32 flags, s32 countOrBuf, uintptr_t addr) {
aFilter(cmd, flags, countOrBuf, addr);
}
void AudioSynth_LoadFilterCount(Acmd* cmd, s32 count, uintptr_t addr) {
aFilter(cmd, 2, count, addr);
}
Acmd* AudioSynth_LoadRingBuffer1(Acmd* cmd, s32 arg1, SynthesisReverb* reverb, s16 bufIndex) {
ReverbRingBufferItem* ringBufferItem = &reverb->items[reverb->curFrame][bufIndex];
cmd =
AudioSynth_LoadRingBufferPart(cmd, DMEM_WET_LEFT_CH, ringBufferItem->startPos, ringBufferItem->lengthA, reverb);
if (ringBufferItem->lengthB != 0) {
// Ring buffer wrapped
cmd = AudioSynth_LoadRingBufferPart(cmd, DMEM_WET_LEFT_CH + ringBufferItem->lengthA, 0, ringBufferItem->lengthB,
reverb);
}
return cmd;
}
Acmd* AudioSynth_LoadRingBuffer2(Acmd* cmd, s32 arg1, SynthesisReverb* reverb, s16 bufIndex) {
ReverbRingBufferItem* bufItem = &reverb->items2[reverb->curFrame][bufIndex];
cmd = AudioSynth_LoadRingBufferPart(cmd, DMEM_WET_LEFT_CH, bufItem->startPos, bufItem->lengthA, reverb);
if (bufItem->lengthB != 0) {
// Ring buffer wrapped
cmd = AudioSynth_LoadRingBufferPart(cmd, DMEM_WET_LEFT_CH + bufItem->lengthA, 0, bufItem->lengthB, reverb);
}
return cmd;
}
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 + DEFAULT_LEN_1CH, length);
return cmd;
}
Acmd* AudioSynth_SaveRingBufferPart(Acmd* cmd, u16 dmem, u16 startPos, s32 length, SynthesisReverb* reverb) {
aSaveBuffer(cmd++, dmem, &reverb->leftRingBuf[startPos], length);
aSaveBuffer(cmd++, dmem + DEFAULT_LEN_1CH, &reverb->rightRingBuf[startPos], length);
return cmd;
}
Acmd* AudioSynth_SaveBufferOffset(Acmd* cmd, u16 dmem, u16 offset, s32 length, s16* buf) {
aSaveBuffer(cmd++, dmem, &buf[offset], length);
return cmd;
}
Acmd* AudioSynth_MaybeLoadRingBuffer2(Acmd* cmd, s32 arg1, SynthesisReverb* reverb, s16 bufIndex) {
if (reverb->downsampleRate == 1) {
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 = AudioSynth_LoadRingBuffer1(cmd, arg1, reverb, arg3);
}
} else {
cmd = func_800DB4E4(cmd, arg1, reverb, arg3);
}
return cmd;
}
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, bufIndex);
} else {
// 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) {
// Ring buffer wrapped
cmd = AudioSynth_SaveRingBufferPart(cmd, DMEM_WET_LEFT_CH + bufItem->lengthA, 0, bufItem->lengthB,
reverb);
}
}
} else {
// 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* AudioSynth_SaveRingBuffer2(Acmd* cmd, SynthesisReverb* reverb, s16 bufIndex) {
ReverbRingBufferItem* bufItem = &reverb->items2[reverb->curFrame][bufIndex];
cmd = AudioSynth_SaveRingBufferPart(cmd, DMEM_WET_LEFT_CH, bufItem->startPos, bufItem->lengthA, reverb);
if (bufItem->lengthB != 0) {
// Ring buffer wrapped
cmd = AudioSynth_SaveRingBufferPart(cmd, DMEM_WET_LEFT_CH + bufItem->lengthA, 0, bufItem->lengthB, reverb);
}
return cmd;
}
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* noteSubEu;
NoteSubEu* noteSubEu2;
s32 unk14;
//if (aiBufLen == 0)
//return;
t = gAudioContext.numNotes * updateIndex;
count = 0;
if (gAudioContext.numSynthesisReverbs == 0) {
for (i = 0; i < gAudioContext.numNotes; i++) {
if (gAudioContext.noteSubsEu[t + i].bitField0.enabled) {
noteIndices[count++] = i;
}
}
} else {
for (reverbIndex = 0; reverbIndex < gAudioContext.numSynthesisReverbs; reverbIndex++) {
for (i = 0; i < gAudioContext.numNotes; i++) {
noteSubEu = &gAudioContext.noteSubsEu[t + i];
if (noteSubEu->bitField0.enabled && noteSubEu->bitField1.reverbIndex == reverbIndex) {
noteIndices[count++] = i;
}
}
}
for (i = 0; i < gAudioContext.numNotes; i++) {
noteSubEu = &gAudioContext.noteSubsEu[t + i];
if (noteSubEu->bitField0.enabled && noteSubEu->bitField1.reverbIndex >= gAudioContext.numSynthesisReverbs) {
noteIndices[count++] = i;
}
}
}
aClearBuffer(cmd++, DMEM_LEFT_CH, DEFAULT_LEN_2CH);
i = 0;
for (reverbIndex = 0; reverbIndex < gAudioContext.numSynthesisReverbs; reverbIndex++) {
reverb = &gAudioContext.synthesisReverbs[reverbIndex];
useReverb = reverb->useReverb;
if (useReverb) {
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++, DMEM_WET_LEFT_CH, DMEM_WET_TEMP, DEFAULT_LEN_2CH);
}
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 = AudioSynth_SaveReverbSamples(cmd, reverb, updateIndex);
if (reverb->unk_05 != -1) {
cmd = AudioSynth_MaybeMixRingBuffer1(cmd, reverb, updateIndex);
}
cmd = AudioSynth_MaybeLoadRingBuffer2(cmd, aiBufLen, reverb, updateIndex);
aMix(cmd++, 0x34, reverb->unk_16, DMEM_WET_TEMP, DMEM_WET_LEFT_CH);
}
}
while (i < count) {
noteSubEu2 = &gAudioContext.noteSubsEu[noteIndices[i] + t];
if (noteSubEu2->bitField1.reverbIndex == reverbIndex) {
cmd = AudioSynth_ProcessNote(noteIndices[i], noteSubEu2,
&gAudioContext.notes[noteIndices[i]].synthesisState, aiBuf, aiBufLen, cmd,
updateIndex);
} else {
break;
}
i++;
}
if (useReverb) {
if (reverb->filterLeft != NULL || reverb->filterRight != NULL) {
cmd = AudioSynth_FilterReverb(cmd, aiBufLen * 2, reverb);
}
if (unk14) {
cmd = AudioSynth_SaveRingBuffer2(cmd, reverb, updateIndex);
} else {
cmd = AudioSynth_SaveReverbSamples(cmd, reverb, updateIndex);
if (reverb->unk_05 != -1) {
cmd = AudioSynth_MaybeMixRingBuffer1(cmd, reverb, updateIndex);
}
}
}
}
while (i < count) {
cmd = AudioSynth_ProcessNote(noteIndices[i], &gAudioContext.noteSubsEu[t + noteIndices[i]],
&gAudioContext.notes[noteIndices[i]].synthesisState, aiBuf, aiBufLen, cmd,
updateIndex);
i++;
}
updateIndex = aiBufLen * 2;
if (CVarGetInteger("gMirroredWorld", 0)) {
aInterleave(cmd++, DMEM_TEMP, DMEM_RIGHT_CH, DMEM_LEFT_CH, updateIndex);
} else {
aInterleave(cmd++, DMEM_TEMP, DMEM_LEFT_CH, DMEM_RIGHT_CH, updateIndex);
}
aSaveBuffer(cmd++, DMEM_TEMP, aiBuf, updateIndex * 2);
return cmd;
}
Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s16* aiBuf,
s32 aiBufLen, Acmd* cmd, s32 updateIndex) {
s32 pad1[3];
SoundFontSample* audioFontSample;
AdpcmLoop* loopInfo;
s32 nSamplesUntilLoopEnd;
s32 nSamplesInThisIteration;
s32 noteFinished;
s32 restart;
s32 flags;
u16 resamplingRateFixedPoint;
s32 nSamplesInFirstFrame;
s32 nTrailingSamplesToIgnore;
s32 phi_a1_2;
s32 frameIndex;
s32 skipBytes;
s32 temp_v1_6;
void* buf;
s32 nSamplesToDecode;
uintptr_t sampleAddr;
u32 samplesLenFixedPoint;
s32 samplesLenAdjusted;
s32 nSamplesProcessed;
uintptr_t loopEndPos;
s32 nSamplesToProcess;
s32 phi_s4;
s32 nFirstFrameSamplesToIgnore;
s32 pad2[7];
s32 frameSize;
s32 nFramesToDecode;
s32 skipInitialSamples;
intptr_t sampleDataStart;
u8* sampleData;
s32 nParts;
s32 curPart;
intptr_t sampleDataStartPad;
s32 side;
s32 resampledTempLen;
u16 noteSamplesDmemAddrBeforeResampling;
intptr_t sampleDataOffset;
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.bookOffset;
finished = noteSubEu->bitField0.finished;
note = &gAudioContext.notes[noteIndex];
flags = A_CONTINUE;
if (noteSubEu->bitField0.needsInit == true) {
flags = A_INIT;
synthState->restart = 0;
synthState->samplePosInt = note->unk_BC;
synthState->samplePosFrac = 0;
synthState->curVolLeft = 0;
synthState->curVolRight = 0;
synthState->prevHeadsetPanRight = 0;
synthState->prevHeadsetPanLeft = 0;
synthState->reverbVol = noteSubEu->reverbVol;
synthState->numParts = 0;
synthState->unk_1A = 1;
note->noteSubEu.bitField0.finished = false;
finished = false;
}
resamplingRateFixedPoint = noteSubEu->resamplingRateFixedPoint;
nParts = noteSubEu->bitField1.hasTwoParts + 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->numParts == 1 && nParts == 2) {
} else if (synthState->numParts == 2 && nParts == 1) {
} else {
}
synthState->numParts = nParts;
if (noteSubEu->bitField1.isSyntheticWave) {
cmd = AudioSynth_LoadWaveSamples(cmd, noteSubEu, synthState, nSamplesToLoad);
noteSamplesDmemAddrBeforeResampling = DMEM_UNCOMPRESSED_NOTE + (synthState->samplePosInt * 2);
synthState->samplePosInt += nSamplesToLoad;
} else
{
audioFontSample = noteSubEu->sound.soundFontSound->sample;
loopInfo = audioFontSample->loop;
loopEndPos = loopInfo->end;
sampleAddr = audioFontSample->sampleAddr;
resampledTempLen = 0;
for (curPart = 0; curPart < nParts; curPart++) {
nSamplesProcessed = 0;
s5 = 0;
if (nParts == 1) {
samplesLenAdjusted = nSamplesToLoad;
} else if (nSamplesToLoad & 1) {
samplesLenAdjusted = (nSamplesToLoad & ~1) + (curPart * 2);
} else {
samplesLenAdjusted = nSamplesToLoad;
}
if (audioFontSample->codec == CODEC_ADPCM || audioFontSample->codec == CODEC_SMALL_ADPCM) {
if (gAudioContext.curLoadedBook != audioFontSample->book->book) {
u32 nEntries;
switch (bookOffset) {
case 1:
gAudioContext.curLoadedBook = &D_8012FBA8[1];
break;
case 2:
case 3:
default:
gAudioContext.curLoadedBook = audioFontSample->book->book;
break;
}
nEntries = 16 * audioFontSample->book->order * audioFontSample->book->npredictors;
aLoadADPCM(cmd++, nEntries, gAudioContext.curLoadedBook);
}
}
while (nSamplesProcessed != samplesLenAdjusted) {
noteFinished = false;
restart = false;
phi_s4 = 0;
nFirstFrameSamplesToIgnore = synthState->samplePosInt & 0xF;
nSamplesUntilLoopEnd = loopEndPos - synthState->samplePosInt;
nSamplesToProcess = samplesLenAdjusted - nSamplesProcessed;
if (nFirstFrameSamplesToIgnore == 0 && !synthState->restart) {
nFirstFrameSamplesToIgnore = 16;
}
nSamplesInFirstFrame = 16 - nFirstFrameSamplesToIgnore;
if (nSamplesToProcess < nSamplesUntilLoopEnd) {
nFramesToDecode = (s32)(nSamplesToProcess - nSamplesInFirstFrame + 15) / 16;
nSamplesToDecode = nFramesToDecode * 16;
nTrailingSamplesToIgnore = nSamplesInFirstFrame + nSamplesToDecode - nSamplesToProcess;
} else {
nSamplesToDecode = nSamplesUntilLoopEnd - nSamplesInFirstFrame;
nTrailingSamplesToIgnore = 0;
if (nSamplesToDecode <= 0) {
nSamplesToDecode = 0;
nSamplesInFirstFrame = nSamplesUntilLoopEnd;
}
nFramesToDecode = (nSamplesToDecode + 15) / 16;
if (loopInfo->count != 0) {
// Loop around and restart
restart = true;
} else {
noteFinished = true;
}
}
switch (audioFontSample->codec) {
case CODEC_ADPCM:
frameSize = 9;
skipInitialSamples = 16;
sampleDataStart = 0;
break;
case CODEC_SMALL_ADPCM:
frameSize = 5;
skipInitialSamples = 16;
sampleDataStart = 0;
break;
case CODEC_S8:
frameSize = 16;
skipInitialSamples = 16;
sampleDataStart = 0;
break;
case CODEC_S16_INMEMORY:
AudioSynth_ClearBuffer(cmd++, DMEM_UNCOMPRESSED_NOTE, (samplesLenAdjusted * 2) + 0x20);
flags = A_CONTINUE;
skipBytes = 0;
nSamplesProcessed = samplesLenAdjusted;
s5 = samplesLenAdjusted;
goto skip;
case CODEC_S16:
AudioSynth_ClearBuffer(cmd++, DMEM_UNCOMPRESSED_NOTE, (samplesLenAdjusted * 2) + 0x20);
AudioSynth_LoadBuffer(cmd++, DMEM_UNCOMPRESSED_NOTE, ALIGN16(nSamplesToLoad * 2),
audioFontSample->sampleAddr + (synthState->samplePosInt * 2));
flags = A_CONTINUE;
skipBytes = 0;
nSamplesProcessed = samplesLenAdjusted;
s5 = samplesLenAdjusted;
goto skip;
case CODEC_REVERB:
break;
}
if (nFramesToDecode != 0) {
frameIndex = (synthState->samplePosInt + skipInitialSamples - nFirstFrameSamplesToIgnore) / 16;
sampleDataOffset = frameIndex * frameSize;
if (audioFontSample->medium == MEDIUM_RAM)
{
sampleData = (u8*)(sampleDataStart + sampleDataOffset + sampleAddr);
} else if (audioFontSample->medium == MEDIUM_UNK) {
return cmd;
} else {
sampleData = AudioLoad_DmaSampleData(sampleDataStart + sampleDataOffset + sampleAddr,
ALIGN16((nFramesToDecode * frameSize) + 0x10), flags,
&synthState->sampleDmaIndex, audioFontSample->medium);
}
if (sampleData == NULL) {
return cmd;
}
sampleDataStartPad = (uintptr_t)sampleData & 0xF;
aligned = ALIGN16((nFramesToDecode * frameSize) + 16);
addr = DMEM_COMPRESSED_ADPCM_DATA - aligned;
aLoadBuffer(cmd++, sampleData - sampleDataStartPad, addr, aligned);
} else {
nSamplesToDecode = 0;
sampleDataStartPad = 0;
}
if (synthState->restart) {
aSetLoop(cmd++, audioFontSample->loop->state);
flags = A_LOOP;
synthState->restart = false;
}
nSamplesInThisIteration = nSamplesToDecode + nSamplesInFirstFrame - nTrailingSamplesToIgnore;
if (nSamplesProcessed == 0) {
skipBytes = nFirstFrameSamplesToIgnore * 2;
} else {
phi_s4 = ALIGN16(s5 + 16);
}
switch (audioFontSample->codec) {
case CODEC_ADPCM:
aligned = ALIGN16((nFramesToDecode * frameSize) + 0x10);
addr = DMEM_COMPRESSED_ADPCM_DATA - aligned;
aSetBuffer(cmd++, 0, addr + sampleDataStartPad, DMEM_UNCOMPRESSED_NOTE + phi_s4,
nSamplesToDecode * 2);
aADPCMdec(cmd++, flags, synthState->synthesisBuffers->adpcmdecState);
break;
case CODEC_SMALL_ADPCM:
aligned = ALIGN16((nFramesToDecode * frameSize) + 0x10);
addr = DMEM_COMPRESSED_ADPCM_DATA - aligned;
aSetBuffer(cmd++, 0, addr + sampleDataStartPad, DMEM_UNCOMPRESSED_NOTE + phi_s4,
nSamplesToDecode * 2);
aADPCMdec(cmd++, flags | 4, synthState->synthesisBuffers->adpcmdecState);
break;
case CODEC_S8:
aligned = ALIGN16((nFramesToDecode * frameSize) + 0x10);
addr = DMEM_COMPRESSED_ADPCM_DATA - aligned;
AudioSynth_SetBuffer(cmd++, 0, addr + sampleDataStartPad, DMEM_UNCOMPRESSED_NOTE + phi_s4,
nSamplesToDecode * 2);
AudioSynth_S8Dec(cmd++, flags, synthState->synthesisBuffers->adpcmdecState);
break;
}
if (nSamplesProcessed != 0) {
aDMEMMove(cmd++, DMEM_UNCOMPRESSED_NOTE + phi_s4 + (nFirstFrameSamplesToIgnore * 2),
DMEM_UNCOMPRESSED_NOTE + s5, nSamplesInThisIteration * 2);
}
nSamplesProcessed += nSamplesInThisIteration;
switch (flags) {
case A_INIT:
skipBytes = 0x20;
s5 = (nSamplesToDecode + 0x10) * 2;
break;
case A_LOOP:
s5 = nSamplesInThisIteration * 2 + s5;
break;
default:
if (s5 != 0) {
s5 = nSamplesInThisIteration * 2 + s5;
} else {
s5 = (nFirstFrameSamplesToIgnore + nSamplesInThisIteration) * 2;
}
break;
}
flags = A_CONTINUE;
skip:
if (noteFinished) {
AudioSynth_ClearBuffer(cmd++, DMEM_UNCOMPRESSED_NOTE + s5, (samplesLenAdjusted - nSamplesProcessed) * 2);
finished = true;
note->noteSubEu.bitField0.finished = true;
func_800DB2C0(updateIndex, noteIndex);
break;
} else {
if (restart) {
synthState->restart = true;
synthState->samplePosInt = loopInfo->start;
} else {
synthState->samplePosInt += nSamplesToProcess;
}
}
}
switch (nParts) {
case 1:
noteSamplesDmemAddrBeforeResampling = DMEM_UNCOMPRESSED_NOTE + skipBytes;
break;
case 2:
switch (curPart) {
case 0:
AudioSynth_InterL(cmd++, DMEM_UNCOMPRESSED_NOTE + skipBytes, 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++, DMEM_UNCOMPRESSED_NOTE + skipBytes,
DMEM_TEMP + 0x20 + resampledTempLen, ((samplesLenAdjusted / 2) + 7) & ~7);
break;
}
}
if (finished) {
break;
}
}
}
flags = A_CONTINUE;
if (noteSubEu->bitField0.needsInit == true) {
noteSubEu->bitField0.needsInit = false;
flags = A_INIT;
}
cmd = AudioSynth_FinalResample(cmd, synthState, aiBufLen * 2, resamplingRateFixedPoint,
noteSamplesDmemAddrBeforeResampling, flags);
if (bookOffset == 3) {
AudioSynth_UnkCmd19(cmd++, DMEM_TEMP, DMEM_TEMP, aiBufLen * 2, 0);
}
if (bookOffset == 2) {
AudioSynth_UnkCmd3(cmd++, DMEM_TEMP, DMEM_TEMP, 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, DMEM_TEMP, 0, (aiBufLen * 2) + 0x20);
}
filter = noteSubEu->filter;
if (filter != 0) {
AudioSynth_LoadFilterCount(cmd++, aiBufLen * 2, filter);
AudioSynth_LoadFilter(cmd++, flags, DMEM_TEMP, synthState->synthesisBuffers->mixEnvelopeState);
}
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) {
AudioSynth_ClearBuffer(cmd++, thing, unk7);
synthState->unk_1A = 0;
} else {
AudioSynth_LoadBuffer(cmd++, thing, unk7, buf);
}
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) {
side = 1;
} else if (noteSubEu->headsetPanLeft != 0 || synthState->prevHeadsetPanLeft != 0) {
side = 2;
} else {
side = 0;
}
cmd = AudioSynth_ProcessEnvelope(cmd, noteSubEu, synthState, aiBufLen, DMEM_TEMP, side, flags);
if (noteSubEu->bitField1.usesHeadsetPanEffects2) {
if (!(flags & A_INIT)) {
flags = A_CONTINUE;
}
cmd = AudioSynth_NoteApplyHeadsetPanEffects(cmd, noteSubEu, synthState, aiBufLen * 2, flags, side);
}
return cmd;
}
Acmd* AudioSynth_FinalResample(Acmd* cmd, NoteSynthesisState* synthState, s32 count, u16 pitch, u16 inpDmem,
s32 resampleFlags) {
if (pitch == 0) {
AudioSynth_ClearBuffer(cmd++, DMEM_TEMP, count);
} else {
aSetBuffer(cmd++, 0, inpDmem, DMEM_TEMP, count);
aResample(cmd++, resampleFlags, pitch, synthState->synthesisBuffers->finalResampleState);
}
return cmd;
}
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 reverbVol;
u16 curVolRight;
s16 rampLeft;
s16 rampRight;
s16 rampReverb;
s16 sourceReverbVol;
u16 targetVolRight;
s32 pad;
curVolLeft = synthState->curVolLeft;
targetVolLeft = noteSubEu->targetVolLeft;
targetVolLeft <<= 4;
reverbVol = noteSubEu->reverbVol;
curVolRight = synthState->curVolRight;
targetVolRight = noteSubEu->targetVolRight;
targetVolRight <<= 4;
if (targetVolLeft != curVolLeft) {
rampLeft = (targetVolLeft - curVolLeft) / (aiBufLen >> 3);
} else {
rampLeft = 0;
}
if (targetVolRight != curVolRight) {
rampRight = (targetVolRight - curVolRight) / (aiBufLen >> 3);
} else {
rampRight = 0;
}
sourceReverbVol = synthState->reverbVol;
phi_t1 = sourceReverbVol & 0x7F;
if (sourceReverbVol != reverbVol) {
rampReverb = (((reverbVol & 0x7F) - phi_t1) << 9) / (aiBufLen >> 3);
synthState->reverbVol = reverbVol;
} else {
rampReverb = 0;
}
synthState->curVolLeft = curVolLeft + (rampLeft * (aiBufLen >> 3));
synthState->curVolRight = curVolRight + (rampRight * (aiBufLen >> 3));
if (noteSubEu->bitField1.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 (headsetPanSettings) {
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, rampReverb, rampLeft, rampRight);
aEnvSetup2(cmd++, curVolLeft, curVolRight);
phi_a1 = D_801304AC;
}
aEnvMixer(cmd++, inBuf, aiBufLen, (sourceReverbVol & 0x80) >> 7, noteSubEu->bitField0.stereoHeadsetEffects,
noteSubEu->bitField0.usesHeadsetPanEffects, noteSubEu->bitField0.stereoStrongRight,
noteSubEu->bitField0.stereoStrongLeft, phi_a1, D_801304A0);
return cmd;
}
Acmd* AudioSynth_LoadWaveSamples(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s32 nSamplesToLoad) {
s32 temp_v0;
s32 unk6 = noteSubEu->unk_06;
s32 samplePosInt = synthState->samplePosInt;
s32 repeats;
if (noteSubEu->bitField1.bookOffset != 0) {
AudioSynth_LoadBuffer(cmd++, DMEM_UNCOMPRESSED_NOTE, ALIGN16(nSamplesToLoad * 2), gWaveSamples[8]);
gWaveSamples[8] += nSamplesToLoad * 2;
return cmd;
} else {
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 < nSamplesToLoad) {
repeats = ((nSamplesToLoad - temp_v0 + 0x3F) / 0x40);
if (repeats != 0) {
aDuplicate(cmd++, repeats, DMEM_UNCOMPRESSED_NOTE, DMEM_UNCOMPRESSED_NOTE + 0x80);
}
}
synthState->samplePosInt = samplePosInt;
}
return cmd;
}
Acmd* AudioSynth_NoteApplyHeadsetPanEffects(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s32 bufLen,
s32 flags, s32 side) {
u16 dest;
u16 pitch;
u8 prevPanShift;
u8 panShift;
switch (side) {
case 1:
dest = DMEM_LEFT_CH;
panShift = noteSubEu->headsetPanRight;
prevPanShift = synthState->prevHeadsetPanRight;
synthState->prevHeadsetPanLeft = 0;
synthState->prevHeadsetPanRight = panShift;
break;
case 2:
dest = DMEM_RIGHT_CH;
panShift = noteSubEu->headsetPanLeft;
prevPanShift = synthState->prevHeadsetPanLeft;
synthState->prevHeadsetPanLeft = panShift;
synthState->prevHeadsetPanRight = 0;
break;
default:
return cmd;
}
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++, DMEM_NOTE_PAN_TEMP, DMEM_TEMP, bufLen);
}
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++, DMEM_TEMP, DMEM_NOTE_PAN_TEMP, bufLen + panShift);
}
} else {
// 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 (panShift) {
// Save excessive samples for next iteration
aSaveBuffer(cmd++, DMEM_NOTE_PAN_TEMP + bufLen, &synthState->synthesisBuffers->panResampleState[0x8],
ALIGN16(panShift));
}
aAddMixer(cmd++, ALIGN64(bufLen), DMEM_NOTE_PAN_TEMP, dest);
return cmd;
}
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