diff --git a/src/audio/audio_seqplayer.c b/src/audio/audio_seqplayer.c
index 7832890b..5e60cd9b 100644
--- a/src/audio/audio_seqplayer.c
+++ b/src/audio/audio_seqplayer.c
@@ -784,7 +784,7 @@ void AudioSeq_SequenceChannelProcessScript(SequenceChannel* channel) {
s8 sp4B;
u8* seqData;
s32 pad;
-
+
if (!channel->enabled) {
return;
}
diff --git a/src/audio/audio_synthesis.c b/src/audio/audio_synthesis.c
index f547eb86..2a637695 100644
--- a/src/audio/audio_synthesis.c
+++ b/src/audio/audio_synthesis.c
@@ -3,6 +3,7 @@
#include "audio/mixer.h"
#include "endianness.h"
+#define DMEM_WET_SCRATCH 0x470
#define DMEM_COMPRESSED_ADPCM_DATA 0x990
#define DMEM_LEFT_CH 0x990
#define DMEM_RIGHT_CH 0xB10
@@ -47,19 +48,20 @@ static const char devstr9[] = "S-Resample Pitch %x (old %d -> delay %d)\n";
void AudioSynth_DisableSampleStates(s32 updateIndex, s32 noteIndex);
void AudioSynth_SyncSampleStates(s32 updateIndex);
-Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSub, NoteSynthesisState* synthState, s16* aiBuf, s32 aiBufLen,
- Acmd* aList, s32 updateIndex);
+Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSub, NoteSynthesisState* synthState, s16* aiBuf,
+ s32 aiBufLen, Acmd* aList, s32 updateIndex);
Acmd* AudioSynth_DoOneAudioUpdate(s16* aiBuf, s32 aiBufLen, Acmd* aList, s32 updateIndex);
Acmd* AudioSynth_LoadRingBufferPart(Acmd* aList, u16 dmem, u16 startPos, s32 size, s32 reverbIndex);
Acmd* AudioSynth_SaveRingBufferPart(Acmd* aList, u16 dmem, u16 startPos, s32 size, s32 reverbIndex);
Acmd* AudioSynth_LoadReverbSamples(Acmd* aList, s32 aiBufLen, s16 reverbIndex, s16 updateIndex);
Acmd* AudioSynth_SaveReverbSamples(Acmd* aList, s16 reverbIndex, s16 updateIndex);
Acmd* AudioSynth_LoadWaveSamples(Acmd* aList, NoteSubEu* noteSub, NoteSynthesisState* synthState, s32 numSamplesToLoad);
-Acmd* AudioSynth_FinalResample(Acmd* aList, NoteSynthesisState* synthState, s32 size, u16 pitch, u16 inpDmem, u32 resampleFlags);
-Acmd* AudioSynth_ProcessEnvelope(Acmd* aList, NoteSubEu* noteSub, NoteSynthesisState* synthState, s32 aiBufLen, u16 dmemSrc,
- s32 delaySide, s32 flags);
+Acmd* AudioSynth_FinalResample(Acmd* aList, NoteSynthesisState* synthState, s32 size, u16 pitch, u16 inpDmem,
+ u32 resampleFlags);
+Acmd* AudioSynth_ProcessEnvelope(Acmd* aList, NoteSubEu* noteSub, NoteSynthesisState* synthState, s32 aiBufLen,
+ u16 dmemSrc, s32 delaySide, s32 flags);
Acmd* AudioSynth_ApplyHaasEffect(Acmd* aList, NoteSubEu* noteSub, NoteSynthesisState* synthState, s32 size, s32 flags,
- s32 delaySide);
+ s32 delaySide);
void AudioSynth_InitNextRingBuf(s32 sampleCount, s32 itemIndex, s32 reverbIndex) {
ReverbRingBufferItem* ringItem;
@@ -96,7 +98,7 @@ void AudioSynth_InitNextRingBuf(s32 sampleCount, s32 itemIndex, s32 reverbIndex)
ringItem->startPos = reverb->nextRingBufPos;
reverb->nextRingBufPos += numSamples;
} else {
- ringItem->lengthA = (numSamples - extraSamples) * 2;
+ ringItem->lengthA = (numSamples - extraSamples) * SAMPLE_SIZE;
ringItem->lengthB = extraSamples * SAMPLE_SIZE;
ringItem->startPos = reverb->nextRingBufPos;
reverb->nextRingBufPos = extraSamples;
@@ -657,16 +659,14 @@ void AudioSynth_SyncSampleStates(s32 updateIndex) {
}
}
}
-extern GameState gGameState;
+
Acmd* AudioSynth_Update(Acmd* aList, s32* cmdCount, s16* aiBufStart, s32 aiBufLen) {
Acmd* aCmdPtr;
s16* aiBufPtr;
s32 chunkLen;
- volatile s32 chunkLentemp = aiBufLen;
s32 i;
s32 j;
- //if (gAudioBufferParams.ticksPerUpdate > 3 && gGameState != GSTATE_PLAY) return;
aCmdPtr = aList;
for (i = gAudioBufferParams.ticksPerUpdate; i > 0; i--) {
AudioSeq_ProcessSequences(i - 1);
@@ -674,20 +674,14 @@ Acmd* AudioSynth_Update(Acmd* aList, s32* cmdCount, s16* aiBufStart, s32 aiBufLe
}
aiBufPtr = aiBufStart;
- // @port: i = gAudioBufferParams.ticksPerUpdate - 1 // this change is necessary to avoid a crash, sounds like a
- // problem
for (i = gAudioBufferParams.ticksPerUpdate; i > 0; i--) {
if (i == 1) {
- //printf("func_80009B64 i == 1\n");
chunkLen = aiBufLen;
} else if ((aiBufLen / i) >= gAudioBufferParams.samplesPerTickMax) {
- //printf("func_80009B64 (aiBufLen / i) >= gAudioBufferParams.samplesPerTickMax\n");
chunkLen = gAudioBufferParams.samplesPerTickMax;
} else if (gAudioBufferParams.samplesPerTickMin >= (aiBufLen / i)) {
- //printf("func_80009B64 gAudioBufferParams.samplesPerTickMin >= (aiBufLen / i)\n");
chunkLen = gAudioBufferParams.samplesPerTickMin;
} else {
- //printf("func_80009B64 else\n");
chunkLen = gAudioBufferParams.samplesPerTick;
}
@@ -696,12 +690,10 @@ Acmd* AudioSynth_Update(Acmd* aList, s32* cmdCount, s16* aiBufStart, s32 aiBufLe
AudioSynth_InitNextRingBuf(chunkLen, gAudioBufferParams.ticksPerUpdate - i, j);
}
}
- //printf("chunkLen: %d, aiBufLen: %d \n", chunkLen, aiBufLen);
aCmdPtr =
AudioSynth_DoOneAudioUpdate((s16*) aiBufPtr, chunkLen, aCmdPtr, gAudioBufferParams.ticksPerUpdate - i);
aiBufLen -= chunkLen;
- // if (aiBufLen < 0) aiBufLen = 0;
aiBufPtr += chunkLen * 2;
}
@@ -735,11 +727,12 @@ Acmd* AudioSynth_LoadReverbSamples(Acmd* aList, s32 aiBufLen, s16 reverbIndex, s
} else {
sp62 = (sp64->startPos & 7) * 2;
sp60 = ALIGN16(sp62 + sp64->lengthA);
- aList = AudioSynth_LoadRingBufferPart(aList, 0x470, sp64->startPos - (sp62 / 2), DMEM_1CH_SIZE, reverbIndex);
+ aList = AudioSynth_LoadRingBufferPart(aList, DMEM_WET_SCRATCH, sp64->startPos - (sp62 / 2), DMEM_1CH_SIZE,
+ reverbIndex);
if (sp64->lengthB != 0) {
- aList = AudioSynth_LoadRingBufferPart(aList, sp60 + 0x470, 0, DMEM_1CH_SIZE - sp60, reverbIndex);
+ aList = AudioSynth_LoadRingBufferPart(aList, sp60 + DMEM_WET_SCRATCH, 0, DMEM_1CH_SIZE - sp60, reverbIndex);
}
- aSetBuffer(aList++, 0, sp62 + 0x470, DMEM_WET_LEFT_CH, aiBufLen * 2);
+ aSetBuffer(aList++, 0, sp62 + DMEM_WET_SCRATCH, DMEM_WET_LEFT_CH, aiBufLen * 2);
aResample(aList++, gSynthReverbs[reverbIndex].resampleFlags, gSynthReverbs[reverbIndex].unk_0A,
OS_K0_TO_PHYSICAL(gSynthReverbs[reverbIndex].unk_30));
aSetBuffer(aList++, 0, sp62 + DMEM_UNCOMPRESSED_NOTE, DMEM_WET_RIGHT_CH, aiBufLen * 2);
@@ -750,16 +743,15 @@ Acmd* AudioSynth_LoadReverbSamples(Acmd* aList, s32 aiBufLen, s16 reverbIndex, s
}
if ((gSynthReverbs[reverbIndex].leakRtL != 0) || (gSynthReverbs[reverbIndex].leakLtR != 0)) {
- aDMEMMove(aList++, DMEM_WET_LEFT_CH, 0x470, DMEM_1CH_SIZE);
+ aDMEMMove(aList++, DMEM_WET_LEFT_CH, DMEM_WET_SCRATCH, DMEM_1CH_SIZE);
aMix(aList++, DMEM_1CH_SIZE >> 4, gSynthReverbs[reverbIndex].leakRtL, DMEM_WET_RIGHT_CH, DMEM_WET_LEFT_CH);
- aMix(aList++, DMEM_1CH_SIZE >> 4, gSynthReverbs[reverbIndex].leakLtR, 0x470, DMEM_WET_RIGHT_CH);
+ aMix(aList++, DMEM_1CH_SIZE >> 4, gSynthReverbs[reverbIndex].leakLtR, DMEM_WET_SCRATCH, DMEM_WET_RIGHT_CH);
}
return aList;
}
Acmd* AudioSynth_SaveReverbSamples(Acmd* aList, s16 reverbIndex, s16 updateIndex) {
ReverbRingBufferItem* sp24;
-
sp24 = &gSynthReverbs[reverbIndex].items[gSynthReverbs[reverbIndex].curFrame][updateIndex];
switch (gSynthReverbs[reverbIndex].downsampleRate) {
case 1:
@@ -792,7 +784,6 @@ Acmd* AudioSynth_DoOneAudioUpdate(s16* aiBuf, s32 aiBufLen, Acmd* aList, s32 upd
count = 0;
if (gNumSynthReverbs == 0) {
- if (gSynthReverbs[i].useReverb) {} // fake?
for (j = 0; j < gNumNotes; j++) {
if (gNoteSubsEu[gNumNotes * updateIndex + j].bitField0.enabled) {
sp84[count++] = j;
@@ -850,48 +841,26 @@ Acmd* AudioSynth_DoOneAudioUpdate(s16* aiBuf, s32 aiBufLen, Acmd* aList, s32 upd
return aList;
}
-Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSub, NoteSynthesisState* synthState, s16* aiBuf, s32 aiBufLen,
- Acmd* aList, s32 updateIndex) {
- s32 pad11C;
- s32 pad118;
- s32 pad114;
+Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSub, NoteSynthesisState* synthState, s16* aiBuf,
+ s32 aiBufLen, Acmd* aList, s32 updateIndex) {
Sample* bookSample;
AdpcmLoop* loopInfo;
void* currentBook;
- s32 pad104;
- s32 pad100;
s32 sampleFinished;
u32 loopToPoint;
s32 flags;
u16 resampleRateFixedPoint;
s32 numSamplesToLoad;
- s32 padE8;
- s32 padE4;
- s32 padE0;
s32 skipBytes = 0;
- s32 padD8;
- s32 padD4;
- s32 padD0;
uintptr_t sampleAddr;
- s32 padC8;
s32 numSamplesToLoadAdj;
s32 numSamplesProcessed;
u32 endPos;
s32 nSamplesToProcess;
- s32 padB4;
- s32 padB0;
- s32 padAC;
- s32 padA4;
- s32 padA8;
s32 numTrailingSamplesToIgnore;
- s32 pad9C;
- s32 pad98;
- s32 pad94;
s32 frameSize;
- s32 pad8C;
s32 skipInitialSamples;
s32 sampleDmaStart;
- s32 pad80;
s32 numParts;
s32 curPart;
s32 numSamplesInThisIteration;
@@ -905,27 +874,23 @@ Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSub, NoteSynthesisSta
u16 sp56;
s32 numSamplesInFirstFrame;
s32 delaySide;
- s32 padasdsa;
s32 nFramesToDecode;
s32 nFirstFrameSamplesToIgnore;
s32 dmemUncompressedAddrOffset1;
u32 sampleslenFixedPoint;
u8* samplesToLoadAddr;
- uintptr_t temp;
- s32 temp2;
- unsigned int new_var2;
+ s32 gain;
u32 nEntries;
s32 aligned;
s32 align2;
- uintptr_t addr;
- u8* new_var;
+ u32 addr;
s32 samplesRemaining;
s32 numSamplesToDecode;
- s32 sega;
+
currentBook = NULL;
note = &gNotes[noteIndex];
flags = 0;
- if (new_var2 = noteSub->bitField0.needsInit == 1) {
+ if (noteSub->bitField0.needsInit == 1) {
flags = 1;
synthState->restart = 0;
synthState->samplePosInt = 0;
@@ -937,11 +902,13 @@ Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSub, NoteSynthesisSta
synthState->numParts = 0;
note->noteSubEu.bitField0.finished = 0;
}
+
resampleRateFixedPoint = noteSub->resampleRate;
numParts = noteSub->bitField1.hasTwoParts + 1;
sampleslenFixedPoint = ((resampleRateFixedPoint * aiBufLen) * 2) + synthState->samplePosFrac;
numSamplesToLoad = sampleslenFixedPoint >> 0x10;
synthState->samplePosFrac = sampleslenFixedPoint & 0xFFFF;
+
if ((synthState->numParts == 1) && (numParts == 2)) {
numSamplesToLoad += 2;
sp56 = 2;
@@ -951,13 +918,14 @@ Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSub, NoteSynthesisSta
} else {
sp56 = 0;
}
+
synthState->numParts = numParts;
if (noteSub->bitField1.isSyntheticWave) {
aList = AudioSynth_LoadWaveSamples(aList, noteSub, synthState, numSamplesToLoad);
- noteSamplesDmemAddrBeforeResampling = (synthState->samplePosInt * ((char) 2)) + 0x5F0;
+ noteSamplesDmemAddrBeforeResampling = DMEM_UNCOMPRESSED_NOTE + (synthState->samplePosInt * SAMPLE_SIZE);
synthState->samplePosInt += numSamplesToLoad;
} else {
- bookSample = *((Sample**) noteSub->waveSampleAddr);
+ bookSample = *(noteSub->waveSampleAddr);
loopInfo = bookSample->loop;
endPos = loopInfo->end;
@@ -1049,7 +1017,8 @@ Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSub, NoteSynthesisSta
dmemUncompressedAddrOffset1 = numSamplesToLoadAdj;
size_t bytesToRead;
- if (((synthState->samplePosInt * 2) + (numSamplesToLoadAdj + SAMPLES_PER_FRAME) * SAMPLE_SIZE) < bookSample->size) {
+ if (((synthState->samplePosInt * 2) + (numSamplesToLoadAdj + SAMPLES_PER_FRAME) * SAMPLE_SIZE) <
+ bookSample->size) {
bytesToRead = (numSamplesToLoadAdj + SAMPLES_PER_FRAME) * SAMPLE_SIZE;
} else {
bytesToRead = bookSample->size - (synthState->samplePosInt * 2);
@@ -1067,15 +1036,11 @@ Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSub, NoteSynthesisSta
aligned = ALIGN16((nFramesToDecode * frameSize) + 0x10);
addr = 0x990 - aligned;
+
if (nFramesToDecode != 0) {
- if (1) {}
frameIndex = (synthState->samplePosInt + skipInitialSamples - nFirstFrameSamplesToIgnore) / 16;
sampleDataOffset = frameIndex * frameSize;
- if (bookSample->medium == 0) {
- samplesToLoadAddr = (u8*) (sampleDmaStart + sampleDataOffset + sampleAddr);
- } else {
- samplesToLoadAddr = (u8*) (sampleDmaStart + sampleDataOffset + sampleAddr);
- }
+ samplesToLoadAddr = (u8*) (sampleDmaStart + sampleDataOffset + sampleAddr);
sampleDataChunkAlignPad = ((uintptr_t) samplesToLoadAddr) % 16;
aLoadBuffer(aList++, OS_K0_TO_PHYSICAL(samplesToLoadAddr - sampleDataChunkAlignPad), addr, aligned);
@@ -1097,13 +1062,15 @@ Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSub, NoteSynthesisSta
switch (bookSample->codec) {
case 0:
- aSetBuffer(aList++, 0, addr + sampleDataChunkAlignPad, 0x5F0, numSamplesToDecode * 2);
+ aSetBuffer(aList++, 0, addr + sampleDataChunkAlignPad, DMEM_UNCOMPRESSED_NOTE,
+ numSamplesToDecode * 2);
aADPCMdec(aList++, flags, OS_K0_TO_PHYSICAL(synthState->synthesisBuffers));
break;
case 1:
- aSetBuffer(aList++, 0, addr + sampleDataChunkAlignPad, 0x5F0, numSamplesToDecode * 2);
+ aSetBuffer(aList++, 0, addr + sampleDataChunkAlignPad, DMEM_UNCOMPRESSED_NOTE,
+ numSamplesToDecode * 2);
aS8Dec(aList++, flags, OS_K0_TO_PHYSICAL(synthState->synthesisBuffers));
break;
}
@@ -1113,20 +1080,20 @@ Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSub, NoteSynthesisSta
align2 = ALIGN16(dmemUncompressedAddrOffset1 + 0x10);
switch (bookSample->codec) {
case 0:
- aSetBuffer(aList++, 0, addr + sampleDataChunkAlignPad, align2 + 0x5F0,
+ aSetBuffer(aList++, 0, addr + sampleDataChunkAlignPad, align2 + DMEM_UNCOMPRESSED_NOTE,
numSamplesToDecode * 2);
aADPCMdec(aList++, flags, OS_K0_TO_PHYSICAL(synthState->synthesisBuffers));
break;
case 1:
- aSetBuffer(aList++, 0, addr + sampleDataChunkAlignPad, align2 + 0x5F0,
+ aSetBuffer(aList++, 0, addr + sampleDataChunkAlignPad, align2 + DMEM_UNCOMPRESSED_NOTE,
numSamplesToDecode * 2);
aS8Dec(aList++, flags, OS_K0_TO_PHYSICAL(synthState->synthesisBuffers));
break;
}
- aDMEMMove(aList++, 0x5F0 + align2 + (nFirstFrameSamplesToIgnore * 2),
- 0x5F0 + dmemUncompressedAddrOffset1, numSamplesInThisIteration * 2);
+ aDMEMMove(aList++, DMEM_UNCOMPRESSED_NOTE + align2 + (nFirstFrameSamplesToIgnore * 2),
+ DMEM_UNCOMPRESSED_NOTE + dmemUncompressedAddrOffset1, numSamplesInThisIteration * 2);
}
numSamplesProcessed += numSamplesInThisIteration;
@@ -1154,7 +1121,7 @@ Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSub, NoteSynthesisSta
flags = 0;
if (sampleFinished) {
- aClearBuffer(aList++, dmemUncompressedAddrOffset1 + 0x5F0,
+ aClearBuffer(aList++, dmemUncompressedAddrOffset1 + DMEM_UNCOMPRESSED_NOTE,
(numSamplesToLoadAdj - numSamplesProcessed) * 2);
noteSub->bitField0.finished = 1;
note->noteSubEu.bitField0.finished = 1;
@@ -1171,15 +1138,16 @@ Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSub, NoteSynthesisSta
switch (numParts) {
case 1:
- noteSamplesDmemAddrBeforeResampling = skipBytes + 0x5F0;
+ noteSamplesDmemAddrBeforeResampling = skipBytes + DMEM_UNCOMPRESSED_NOTE;
break;
case 2:
switch (curPart) {
case 0:
- aInterl(aList++, skipBytes + 0x5F0, 0x470, ALIGN8(numSamplesToLoadAdj / 2));
+ aInterl(aList++, skipBytes + DMEM_UNCOMPRESSED_NOTE, DMEM_WET_SCRATCH,
+ ALIGN8(numSamplesToLoadAdj / 2));
resampledTempLen = numSamplesToLoadAdj;
- noteSamplesDmemAddrBeforeResampling = 0x470;
+ noteSamplesDmemAddrBeforeResampling = DMEM_WET_SCRATCH;
if (noteSub->bitField0.finished) {
aClearBuffer(aList++, resampledTempLen + noteSamplesDmemAddrBeforeResampling,
numSamplesToLoadAdj + 0x10);
@@ -1187,7 +1155,7 @@ Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSub, NoteSynthesisSta
break;
case 1:
- aInterl(aList++, skipBytes + 0x5F0, resampledTempLen + 0x470,
+ aInterl(aList++, skipBytes + DMEM_UNCOMPRESSED_NOTE, resampledTempLen + DMEM_WET_SCRATCH,
ALIGN8(numSamplesToLoadAdj / 2));
break;
}
@@ -1206,20 +1174,20 @@ Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSub, NoteSynthesisSta
noteSub->bitField0.needsInit = 0;
}
flags = sp56 | flags;
- aList = AudioSynth_FinalResample(aList, synthState, aiBufLen * 2, resampleRateFixedPoint, noteSamplesDmemAddrBeforeResampling,
- flags);
+ aList = AudioSynth_FinalResample(aList, synthState, aiBufLen * 2, resampleRateFixedPoint,
+ noteSamplesDmemAddrBeforeResampling, flags);
if (flags & 1) {
flags = 1;
}
if (noteSub->bitField1.bookOffset == 3) {
aUnkCmd19(aList++, 0, aiBufLen * 2, 0x450, 0x450);
}
- temp2 = noteSub->gain;
- if (temp2 != 0) {
- if (temp2 < 0x10) {
- temp2 = 0x10;
+ gain = noteSub->gain;
+ if (gain != 0) {
+ if (gain < 0x10) {
+ gain = 0x10;
}
- aHiLoGain(aList++, temp2, (aiBufLen + 0x10) * 2, 0x450, 0);
+ aHiLoGain(aList++, gain, (aiBufLen + 0x10) * 2, 0x450, 0);
}
if ((noteSub->leftDelaySize != 0) || (synthState->prevHaasEffectLeftDelaySize != 0)) {
delaySide = 1;
diff --git a/src/audio/audio_thread.c b/src/audio/audio_thread.c
index 8c50e928..f38e8935 100644
--- a/src/audio/audio_thread.c
+++ b/src/audio/audio_thread.c
@@ -83,14 +83,11 @@ void AudioThread_CreateNextAudioBuffer(s16* samples, u32 num_samples) {
// Spectrum Analyzer fix
memcpy(gAiBuffers[gCurAiBuffIndex], samples, num_samples);
-
- gAudioRandom = osGetCount() * (gAudioRandom + gAudioTaskCountQ);
- if (gMaxAbiCmdCnt < abiCmdCount) {
- gMaxAbiCmdCnt = abiCmdCount;
- }
+ gAudioRandom = osGetCount() * (gAudioRandom + gAudioTaskCountQ);
}
+// Original function, unused in the port.
SPTask* AudioThread_CreateTask() {
static s32 gMaxAbiCmdCnt = 128;
static SPTask* gWaitingAudioTask = NULL;
diff --git a/src/port/Engine.cpp b/src/port/Engine.cpp
index 19597759..57ca68ff 100644
--- a/src/port/Engine.cpp
+++ b/src/port/Engine.cpp
@@ -203,8 +203,17 @@ void GameEngine::StartFrame() const {
this->context->GetWindow()->StartFrame();
}
+#if 1
+// Values for 44100 hz
#define SAMPLES_HIGH 752
#define SAMPLES_LOW 720
+#else
+// Values for 32000 hz
+#define SAMPLES_HIGH 560
+#define SAMPLES_LOW 544
+
+#endif
+
#define NUM_AUDIO_CHANNELS 2
extern "C" u16 audBuffer = 0;