mirror/oot
1
0
Fork 0
mirror of https://github.com/zeldaret/oot synced 2026-05-22 22:44:26 -04:00
Code Issues Packages Projects Releases Wiki Activity

Rename NoteSubEu -> NoteSampleState (#2738)

Browse Source
This commit is contained in:
Tharo
2026-04-24 20:15:28 +01:00
committed by GitHub
parent 072eb2dc00
commit 9c0dafe8d2
8 changed files with 192 additions and 191 deletions
Download Patch File Download Diff File Expand all files Collapse all files
include/audio.h
+12 -12
View File
@@ -31,7 +31,7 @@ typedef void (*AudioCustomUpdateFunction)(void);
#define SEQ_NUM_CHANNELS 16
#define SEQ_IO_VAL_NONE -1
#define MUTE_BEHAVIOR_3 (1 << 3) // prevent further noteSubEus from playing
#define MUTE_BEHAVIOR_3 (1 << 3) // prevent further NoteSampleStates from playing
#define MUTE_BEHAVIOR_4 (1 << 4) // stop something in seqLayer scripts
#define MUTE_BEHAVIOR_SOFTEN (1 << 5) // lower volume, by default to half
#define MUTE_BEHAVIOR_STOP_NOTES (1 << 6) // prevent further notes from playing
@@ -587,7 +587,7 @@ typedef struct NotePlaybackState {
} NotePlaybackState; // size = 0x88
// JAudio: "commonch"
typedef struct NoteSubEu {
typedef struct NoteSampleState {
struct {
/* 0x00 */ volatile u8 enabled : 1;
/* 0x00 */ u8 needsInit : 1;
@@ -621,7 +621,7 @@ typedef struct NoteSubEu {
};
/* 0x14 */ s16* filter;
/* 0x18 */ char pad_18[0x8];
} NoteSubEu; // size = 0x20
} NoteSampleState; // size = 0x20
// JAudio: "channel"
typedef struct Note {
@@ -630,7 +630,7 @@ typedef struct Note {
/* 0x30 */ NotePlaybackState playbackState;
/* 0xB8 */ char unk_B8[0x4];
/* 0xBC */ u32 startSamplePos; // initial position/index to start processing s16 samples
/* 0xC0 */ NoteSubEu noteSubEu;
/* 0xC0 */ NoteSampleState sampleState;
} Note; // size = 0xE0
typedef struct ReverbSettings {
@@ -888,7 +888,7 @@ typedef struct AudioContext {
/* 0x0004 */ u16 unk_4;
/* 0x0006 */ char unk_0006[0x0A];
/* 0x0010 */ s16* curLoadedBook;
/* 0x0014 */ NoteSubEu* noteSubsEu;
/* 0x0014 */ NoteSampleState* sampleStates;
/* 0x0018 */ SynthesisReverb synthesisReverbs[4];
/* 0x0B38 */ char unk_0B38[0x30];
/* 0x0B68 */ Sample* usedSamples[128];
@@ -988,7 +988,7 @@ typedef struct AudioContext {
/* 0x3530 */ SequencePlayer seqPlayers[4];
/* 0x3AB0 */ SequenceLayer sequenceLayers[64];
/* 0x5AB0 */ SequenceChannel sequenceChannelNone;
/* 0x5B84 */ s32 noteSubEuOffset;
/* 0x5B84 */ s32 sampleStateOffset;
/* 0x5B88 */ AudioListItem layerFreeList;
/* 0x5B98 */ NotePool noteFreeLists;
/* 0x5BD8 */ u8 threadCmdWritePos;
@@ -1007,7 +1007,7 @@ typedef struct AudioContext {
/* 0x5C50 */ AudioCmd threadCmdBuf[0x100]; // Audio thread commands used to transfer audio requests from the graph thread to the audio thread
} AudioContext; // size = 0x6450
typedef struct NoteSubAttributes {
typedef struct NoteSampleStateAttributes {
/* 0x00 */ u8 reverbVol;
/* 0x01 */ u8 gain; // Increases volume by a multiplicative scaling factor. Represented as a UQ4.4 number
/* 0x02 */ u8 pan;
@@ -1018,7 +1018,7 @@ typedef struct NoteSubAttributes {
/* 0x10 */ s16* filter;
/* 0x14 */ u8 combFilterSize;
/* 0x16 */ u16 combFilterGain;
} NoteSubAttributes; // size = 0x18
} NoteSampleStateAttributes; // size = 0x18
typedef struct TempoData {
/* 0x0 */ s16 unk_00; // set to 0x1C00, unused
@@ -1103,8 +1103,8 @@ void AudioThread_InitMesgQueues(void);
void Audio_InvalDCache(void* buf, s32 size);
void Audio_WritebackDCache(void* buf, s32 size);
s32 osAiSetNextBuffer(void*, u32);
void Audio_InitNoteSub(Note* note, NoteSubEu* sub, NoteSubAttributes* attrs);
void Audio_NoteSetResamplingRate(NoteSubEu* noteSubEu, f32 resamplingRateInput);
void Audio_InitSampleState(Note* note, NoteSampleState* sampleState, NoteSampleStateAttributes* attrs);
void Audio_NoteSetResamplingRate(NoteSampleState* sampleState, f32 resamplingRateInput);
void Audio_NoteInit(Note* note);
void Audio_NoteDisable(Note* note);
void Audio_ProcessNotes(void);
@@ -1230,8 +1230,8 @@ extern f32 gPitchFrequencies[];
extern u8 gDefaultShortNoteVelocityTable[16];
extern u8 gDefaultShortNoteGateTimeTable[16];
extern EnvelopePoint gDefaultEnvelope[4];
extern NoteSubEu gZeroNoteSub;
extern NoteSubEu gDefaultNoteSub;
extern NoteSampleState gZeroNoteSampleState;
extern NoteSampleState gDefaultNoteSampleState;
extern u16 gHaasEffectDelaySizes[64];
extern s16 D_8012FBA8[];
extern f32 gHeadsetPanVolume[128];
src/audio/game/debug.inc.c
+1 -1
View File
@@ -109,7 +109,7 @@ void AudioDebug_Draw(GfxPrint* printer) {
GfxPrint_Printf(printer, "- %s -", sAudioDebugPageNames[sAudioDebugPage]);
for (i = 0; i < gAudioSpecs[gAudioSpecId].numNotes; i++) {
if (gAudioCtx.notes[i].noteSubEu.bitField0.enabled == 1) {
if (gAudioCtx.notes[i].sampleState.bitField0.enabled == 1) {
numEnabledNotes++;
}
}
src/audio/internal/data.c
+2 -2
View File
@@ -552,9 +552,9 @@ EnvelopePoint gDefaultEnvelope[] = {
{ ADSR_DISABLE, 0 },
};
NoteSubEu gZeroNoteSub = { 0 };
NoteSampleState gZeroNoteSampleState = { 0 };
NoteSubEu gDefaultNoteSub = {
NoteSampleState gDefaultNoteSampleState = {
{ 1, 1, 0, 0, 0, 0, 0, 0 }, { 0 }, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, NULL, NULL, 0,
};
src/audio/internal/heap.c
+8 -7
View File
@@ -442,7 +442,7 @@ void* AudioHeap_AllocCached(s32 tableType, s32 size, s32 cache, s32 id) {
if (loadStatusEntry0 == LOAD_STATUS_MAYBE_DISCARDABLE) {
for (i = 0; i < gAudioCtx.numNotes; i++) {
if (gAudioCtx.notes[i].playbackState.fontId == temporaryCache->entries[0].id &&
gAudioCtx.notes[i].noteSubEu.bitField0.enabled) {
gAudioCtx.notes[i].sampleState.bitField0.enabled) {
break;
}
}
@@ -456,7 +456,7 @@ void* AudioHeap_AllocCached(s32 tableType, s32 size, s32 cache, s32 id) {
if (loadStatusEntry1 == LOAD_STATUS_MAYBE_DISCARDABLE) {
for (i = 0; i < gAudioCtx.numNotes; i++) {
if (gAudioCtx.notes[i].playbackState.fontId == temporaryCache->entries[1].id &&
gAudioCtx.notes[i].noteSubEu.bitField0.enabled) {
gAudioCtx.notes[i].sampleState.bitField0.enabled) {
break;
}
}
@@ -512,7 +512,7 @@ void* AudioHeap_AllocCached(s32 tableType, s32 size, s32 cache, s32 id) {
if (loadStatusEntry0 == LOAD_STATUS_COMPLETE) {
for (i = 0; i < gAudioCtx.numNotes; i++) {
if (gAudioCtx.notes[i].playbackState.fontId == temporaryCache->entries[0].id &&
gAudioCtx.notes[i].noteSubEu.bitField0.enabled) {
gAudioCtx.notes[i].sampleState.bitField0.enabled) {
break;
}
}
@@ -525,7 +525,7 @@ void* AudioHeap_AllocCached(s32 tableType, s32 size, s32 cache, s32 id) {
if (loadStatusEntry1 == LOAD_STATUS_COMPLETE) {
for (i = 0; i < gAudioCtx.numNotes; i++) {
if (gAudioCtx.notes[i].playbackState.fontId == temporaryCache->entries[1].id &&
gAudioCtx.notes[i].noteSubEu.bitField0.enabled) {
gAudioCtx.notes[i].sampleState.bitField0.enabled) {
break;
}
}
@@ -891,7 +891,7 @@ s32 AudioHeap_ResetStep(void) {
AudioHeap_UpdateReverbs();
} else {
for (i = 0; i < gAudioCtx.numNotes; i++) {
if (gAudioCtx.notes[i].noteSubEu.bitField0.enabled &&
if (gAudioCtx.notes[i].sampleState.bitField0.enabled &&
gAudioCtx.notes[i].playbackState.adsr.action.s.state != ADSR_STATE_DISABLED) {
gAudioCtx.notes[i].playbackState.adsr.fadeOutVel =
gAudioCtx.audioBufferParameters.ticksPerUpdateInv;
@@ -1056,8 +1056,9 @@ void AudioHeap_Init(void) {
gAudioCtx.notes = AudioHeap_AllocZeroed(&gAudioCtx.miscPool, gAudioCtx.numNotes * sizeof(Note));
Audio_NoteInitAll();
Audio_InitNoteFreeList();
gAudioCtx.noteSubsEu = AudioHeap_AllocZeroed(&gAudioCtx.miscPool, gAudioCtx.audioBufferParameters.ticksPerUpdate *
gAudioCtx.numNotes * sizeof(NoteSubEu));
gAudioCtx.sampleStates =
AudioHeap_AllocZeroed(&gAudioCtx.miscPool, gAudioCtx.audioBufferParameters.ticksPerUpdate * gAudioCtx.numNotes *
sizeof(NoteSampleState));
// Initialize audio binary interface command list buffers
for (i = 0; i != 2; i++) {
gAudioCtx.abiCmdBufs[i] =
src/audio/internal/playback.c
+88 -88
View File
@@ -7,7 +7,7 @@
/**
* original name: Nas_smzSetParam
*/
void Audio_InitNoteSub(Note* note, NoteSubEu* sub, NoteSubAttributes* attrs) {
void Audio_InitSampleState(Note* note, NoteSampleState* sampleState, NoteSampleStateAttributes* attrs) {
f32 volLeft;
f32 volRight;
s32 halfPanIndex;
@@ -25,36 +25,36 @@ void Audio_InitNoteSub(Note* note, NoteSubEu* sub, NoteSubAttributes* attrs) {
reverbVol = attrs->reverbVol;
stereoData = attrs->stereo.s;
sub->bitField0 = note->noteSubEu.bitField0;
sub->bitField1 = note->noteSubEu.bitField1;
sub->waveSampleAddr = note->noteSubEu.waveSampleAddr;
sub->harmonicIndexCurAndPrev = note->noteSubEu.harmonicIndexCurAndPrev;
sampleState->bitField0 = note->sampleState.bitField0;
sampleState->bitField1 = note->sampleState.bitField1;
sampleState->waveSampleAddr = note->sampleState.waveSampleAddr;
sampleState->harmonicIndexCurAndPrev = note->sampleState.harmonicIndexCurAndPrev;
Audio_NoteSetResamplingRate(sub, attrs->frequency);
Audio_NoteSetResamplingRate(sampleState, attrs->frequency);
pan &= 0x7F;
sub->bitField0.stereoStrongRight = false;
sub->bitField0.stereoStrongLeft = false;
sub->bitField0.stereoHeadsetEffects = stereoData.stereoHeadsetEffects;
sub->bitField0.usesHeadsetPanEffects = stereoData.usesHeadsetPanEffects;
sampleState->bitField0.stereoStrongRight = false;
sampleState->bitField0.stereoStrongLeft = false;
sampleState->bitField0.stereoHeadsetEffects = stereoData.stereoHeadsetEffects;
sampleState->bitField0.usesHeadsetPanEffects = stereoData.usesHeadsetPanEffects;
if (stereoHeadsetEffects && (gAudioCtx.soundOutputMode == SOUND_OUTPUT_HEADSET)) {
halfPanIndex = pan >> 1;
if (halfPanIndex > 0x3F) {
halfPanIndex = 0x3F;
}
sub->haasEffectRightDelaySize = gHaasEffectDelaySizes[halfPanIndex];
sub->haasEffectLeftDelaySize = gHaasEffectDelaySizes[0x3F - halfPanIndex];
sub->bitField1.useHaasEffect = true;
sampleState->haasEffectRightDelaySize = gHaasEffectDelaySizes[halfPanIndex];
sampleState->haasEffectLeftDelaySize = gHaasEffectDelaySizes[0x3F - halfPanIndex];
sampleState->bitField1.useHaasEffect = true;
volLeft = gHeadsetPanVolume[pan];
volRight = gHeadsetPanVolume[0x7F - pan];
} else if (stereoHeadsetEffects && (gAudioCtx.soundOutputMode == SOUND_OUTPUT_STEREO)) {
strongLeft = strongRight = 0;
sub->haasEffectLeftDelaySize = 0;
sub->haasEffectRightDelaySize = 0;
sub->bitField1.useHaasEffect = false;
sampleState->haasEffectLeftDelaySize = 0;
sampleState->haasEffectRightDelaySize = 0;
sampleState->bitField1.useHaasEffect = false;
volLeft = gStereoPanVolume[pan];
volRight = gStereoPanVolume[0x7F - pan];
@@ -64,37 +64,37 @@ void Audio_InitNoteSub(Note* note, NoteSubEu* sub, NoteSubAttributes* attrs) {
strongRight = 1;
}
sub->bitField0.stereoStrongRight = strongRight;
sub->bitField0.stereoStrongLeft = strongLeft;
sampleState->bitField0.stereoStrongRight = strongRight;
sampleState->bitField0.stereoStrongLeft = strongLeft;
switch (stereoData.bit2) {
case 0:
break;
case 1:
sub->bitField0.stereoStrongRight = stereoData.strongRight;
sub->bitField0.stereoStrongLeft = stereoData.strongLeft;
sampleState->bitField0.stereoStrongRight = stereoData.strongRight;
sampleState->bitField0.stereoStrongLeft = stereoData.strongLeft;
break;
case 2:
sub->bitField0.stereoStrongRight = stereoData.strongRight | strongRight;
sub->bitField0.stereoStrongLeft = stereoData.strongLeft | strongLeft;
sampleState->bitField0.stereoStrongRight = stereoData.strongRight | strongRight;
sampleState->bitField0.stereoStrongLeft = stereoData.strongLeft | strongLeft;
break;
case 3:
sub->bitField0.stereoStrongRight = stereoData.strongRight ^ strongRight;
sub->bitField0.stereoStrongLeft = stereoData.strongLeft ^ strongLeft;
sampleState->bitField0.stereoStrongRight = stereoData.strongRight ^ strongRight;
sampleState->bitField0.stereoStrongLeft = stereoData.strongLeft ^ strongLeft;
break;
}
} else if (gAudioCtx.soundOutputMode == SOUND_OUTPUT_MONO) {
sub->bitField0.stereoHeadsetEffects = false;
sub->bitField0.usesHeadsetPanEffects = false;
sampleState->bitField0.stereoHeadsetEffects = false;
sampleState->bitField0.usesHeadsetPanEffects = false;
volLeft = 0.707f; // approx 1/sqrt(2)
volRight = 0.707f;
} else {
sub->bitField0.stereoStrongRight = stereoData.strongRight;
sub->bitField0.stereoStrongLeft = stereoData.strongLeft;
sampleState->bitField0.stereoStrongRight = stereoData.strongRight;
sampleState->bitField0.stereoStrongLeft = stereoData.strongLeft;
volLeft = gDefaultPanVolume[pan];
volRight = gDefaultPanVolume[0x7F - pan];
}
@@ -102,35 +102,35 @@ void Audio_InitNoteSub(Note* note, NoteSubEu* sub, NoteSubAttributes* attrs) {
vel = 0.0f > vel ? 0.0f : vel;
vel = 1.0f < vel ? 1.0f : vel;
sub->targetVolLeft = (s32)((vel * volLeft) * (0x1000 - 0.001f));
sub->targetVolRight = (s32)((vel * volRight) * (0x1000 - 0.001f));
sampleState->targetVolLeft = (s32)((vel * volLeft) * (0x1000 - 0.001f));
sampleState->targetVolRight = (s32)((vel * volRight) * (0x1000 - 0.001f));
sub->gain = attrs->gain;
sub->filter = attrs->filter;
sub->combFilterSize = attrs->combFilterSize;
sub->combFilterGain = attrs->combFilterGain;
sub->reverbVol = reverbVol;
sampleState->gain = attrs->gain;
sampleState->filter = attrs->filter;
sampleState->combFilterSize = attrs->combFilterSize;
sampleState->combFilterGain = attrs->combFilterGain;
sampleState->reverbVol = reverbVol;
}
/**
* original name: Nas_smzSetPitch
*/
void Audio_NoteSetResamplingRate(NoteSubEu* noteSubEu, f32 resamplingRateInput) {
void Audio_NoteSetResamplingRate(NoteSampleState* sampleState, f32 resamplingRateInput) {
f32 resamplingRate = 0.0f;
if (resamplingRateInput < 2.0f) {
noteSubEu->bitField1.hasTwoParts = false;
sampleState->bitField1.hasTwoParts = false;
resamplingRate = CLAMP_MAX(resamplingRateInput, 1.99998f);
} else {
noteSubEu->bitField1.hasTwoParts = true;
sampleState->bitField1.hasTwoParts = true;
if (resamplingRateInput > 3.99996f) {
resamplingRate = 1.99998f;
} else {
resamplingRate = resamplingRateInput * 0.5f;
}
}
noteSubEu->resamplingRateFixedPoint = (s32)(resamplingRate * 32768.0f);
sampleState->resamplingRateFixedPoint = (s32)(resamplingRate * 32768.0f);
}
/**
@@ -147,20 +147,20 @@ void Audio_NoteInit(Note* note) {
note->playbackState.unk_04 = 0;
note->playbackState.adsr.action.s.state = ADSR_STATE_INITIAL;
note->noteSubEu = gDefaultNoteSub;
note->sampleState = gDefaultNoteSampleState;
}
/**
* original name: Nas_StopVoice
*/
void Audio_NoteDisable(Note* note) {
if (note->noteSubEu.bitField0.needsInit == true) {
note->noteSubEu.bitField0.needsInit = false;
if (note->sampleState.bitField0.needsInit == true) {
note->sampleState.bitField0.needsInit = false;
}
note->playbackState.priority = 0;
note->noteSubEu.bitField0.enabled = false;
note->sampleState.bitField0.enabled = false;
note->playbackState.unk_04 = 0;
note->noteSubEu.bitField0.finished = false;
note->sampleState.bitField0.finished = false;
note->playbackState.parentLayer = NO_LAYER;
note->playbackState.prevParentLayer = NO_LAYER;
note->playbackState.adsr.action.s.state = ADSR_STATE_DISABLED;
@@ -173,18 +173,18 @@ void Audio_NoteDisable(Note* note) {
void Audio_ProcessNotes(void) {
s32 pad[2];
NoteAttributes* attrs;
NoteSubEu* noteSubEu2;
NoteSubEu* noteSubEu;
NoteSampleState* sampleState2;
NoteSampleState* sampleState;
Note* note;
NotePlaybackState* playbackState;
NoteSubAttributes subAttrs;
NoteSampleStateAttributes sampleStateAttrs;
u8 bookOffset;
f32 scale;
s32 i;
for (i = 0; i < gAudioCtx.numNotes; i++) {
note = &gAudioCtx.notes[i];
noteSubEu2 = &gAudioCtx.noteSubsEu[gAudioCtx.noteSubEuOffset + i];
sampleState2 = &gAudioCtx.sampleStates[gAudioCtx.sampleStateOffset + i];
playbackState = &note->playbackState;
if (playbackState->parentLayer != NO_LAYER) {
if ((u32)playbackState->parentLayer < 0x7FFFFFFF) {
@@ -224,9 +224,9 @@ void Audio_ProcessNotes(void) {
out:
if (playbackState->priority != 0) {
if (1) {}
noteSubEu = &note->noteSubEu;
if (playbackState->unk_04 >= 1 || noteSubEu->bitField0.finished) {
if (playbackState->adsr.action.s.state == ADSR_STATE_DISABLED || noteSubEu->bitField0.finished) {
sampleState = &note->sampleState;
if (playbackState->unk_04 >= 1 || sampleState->bitField0.finished) {
if (playbackState->adsr.action.s.state == ADSR_STATE_DISABLED || sampleState->bitField0.finished) {
if (playbackState->wantedParentLayer != NO_LAYER) {
Audio_NoteDisable(note);
if (playbackState->wantedParentLayer->channel != NULL) {
@@ -268,46 +268,46 @@ void Audio_ProcessNotes(void) {
Audio_NoteVibratoUpdate(note);
attrs = &playbackState->attributes;
if (playbackState->unk_04 == 1 || playbackState->unk_04 == 2) {
subAttrs.frequency = attrs->freqScale;
subAttrs.velocity = attrs->velocity;
subAttrs.pan = attrs->pan;
subAttrs.reverbVol = attrs->reverb;
subAttrs.stereo = attrs->stereo;
subAttrs.gain = attrs->gain;
subAttrs.filter = attrs->filter;
subAttrs.combFilterSize = attrs->combFilterSize;
subAttrs.combFilterGain = attrs->combFilterGain;
bookOffset = noteSubEu->bitField1.bookOffset;
sampleStateAttrs.frequency = attrs->freqScale;
sampleStateAttrs.velocity = attrs->velocity;
sampleStateAttrs.pan = attrs->pan;
sampleStateAttrs.reverbVol = attrs->reverb;
sampleStateAttrs.stereo = attrs->stereo;
sampleStateAttrs.gain = attrs->gain;
sampleStateAttrs.filter = attrs->filter;
sampleStateAttrs.combFilterSize = attrs->combFilterSize;
sampleStateAttrs.combFilterGain = attrs->combFilterGain;
bookOffset = sampleState->bitField1.bookOffset;
} else {
SequenceLayer* layer = playbackState->parentLayer;
SequenceChannel* channel = layer->channel;
subAttrs.frequency = layer->noteFreqScale;
subAttrs.velocity = layer->noteVelocity;
subAttrs.pan = layer->notePan;
sampleStateAttrs.frequency = layer->noteFreqScale;
sampleStateAttrs.velocity = layer->noteVelocity;
sampleStateAttrs.pan = layer->notePan;
if (layer->stereo.asByte == 0) {
subAttrs.stereo = channel->stereo;
sampleStateAttrs.stereo = channel->stereo;
} else {
subAttrs.stereo = layer->stereo;
sampleStateAttrs.stereo = layer->stereo;
}
subAttrs.reverbVol = channel->targetReverbVol;
subAttrs.gain = channel->gain;
subAttrs.filter = channel->filter;
subAttrs.combFilterSize = channel->combFilterSize;
subAttrs.combFilterGain = channel->combFilterGain;
sampleStateAttrs.reverbVol = channel->targetReverbVol;
sampleStateAttrs.gain = channel->gain;
sampleStateAttrs.filter = channel->filter;
sampleStateAttrs.combFilterSize = channel->combFilterSize;
sampleStateAttrs.combFilterGain = channel->combFilterGain;
bookOffset = channel->bookOffset & 0x7;
if (channel->seqPlayer->muted && (channel->muteBehavior & MUTE_BEHAVIOR_3)) {
subAttrs.frequency = 0.0f;
subAttrs.velocity = 0.0f;
sampleStateAttrs.frequency = 0.0f;
sampleStateAttrs.velocity = 0.0f;
}
}
subAttrs.frequency *= playbackState->vibratoFreqScale * playbackState->portamentoFreqScale;
subAttrs.frequency *= gAudioCtx.audioBufferParameters.resampleRate;
subAttrs.velocity *= scale;
Audio_InitNoteSub(note, noteSubEu2, &subAttrs);
noteSubEu->bitField1.bookOffset = bookOffset;
sampleStateAttrs.frequency *= playbackState->vibratoFreqScale * playbackState->portamentoFreqScale;
sampleStateAttrs.frequency *= gAudioCtx.audioBufferParameters.resampleRate;
sampleStateAttrs.velocity *= scale;
Audio_InitSampleState(note, sampleState2, &sampleStateAttrs);
sampleState->bitField1.bookOffset = bookOffset;
skip:;
}
}
@@ -520,7 +520,7 @@ void Audio_SeqLayerDecayRelease(SequenceLayer* layer, s32 target) {
attrs->combFilterGain = channel->combFilterGain;
attrs->combFilterSize = channel->combFilterSize;
if (channel->seqPlayer->muted && (channel->muteBehavior & MUTE_BEHAVIOR_3)) {
note->noteSubEu.bitField0.finished = true;
note->sampleState.bitField0.finished = true;
}
if (layer->stereo.asByte == 0) {
@@ -617,7 +617,7 @@ s32 Audio_BuildSyntheticWave(Note* note, SequenceLayer* layer, s32 waveId) {
// Save the pointer to the synthethic wave
// waveId index starts at 128, there are WAVE_SAMPLE_COUNT samples to read from
note->noteSubEu.waveSampleAddr = &gWaveSamples[waveId - 128][harmonicIndex * WAVE_SAMPLE_COUNT];
note->sampleState.waveSampleAddr = &gWaveSamples[waveId - 128][harmonicIndex * WAVE_SAMPLE_COUNT];
return harmonicIndex;
}
@@ -635,7 +635,7 @@ void Audio_InitSyntheticWave(Note* note, SequenceLayer* layer) {
curHarmonicIndex = Audio_BuildSyntheticWave(note, layer, waveId);
if (curHarmonicIndex != prevHarmonicIndex) {
note->noteSubEu.harmonicIndexCurAndPrev = (curHarmonicIndex << 2) + prevHarmonicIndex;
note->sampleState.harmonicIndexCurAndPrev = (curHarmonicIndex << 2) + prevHarmonicIndex;
}
}
@@ -831,7 +831,7 @@ void Audio_NoteInitForLayer(Note* note, SequenceLayer* layer) {
s32 pad[3];
s16 instId;
NotePlaybackState* playbackState = &note->playbackState;
NoteSubEu* sub = &note->noteSubEu;
NoteSampleState* sampleState = &note->sampleState;
note->playbackState.prevParentLayer = NO_LAYER;
note->playbackState.parentLayer = layer;
@@ -848,21 +848,21 @@ void Audio_NoteInitForLayer(Note* note, SequenceLayer* layer) {
if (instId == 0xFF) {
instId = layer->channel->instOrWave;
}
sub->tunedSample = layer->tunedSample;
sampleState->tunedSample = layer->tunedSample;
if (instId >= 0x80 && instId < 0xC0) {
sub->bitField1.isSyntheticWave = true;
sampleState->bitField1.isSyntheticWave = true;
} else {
sub->bitField1.isSyntheticWave = false;
sampleState->bitField1.isSyntheticWave = false;
}
if (sub->bitField1.isSyntheticWave) {
if (sampleState->bitField1.isSyntheticWave) {
Audio_BuildSyntheticWave(note, layer, instId);
}
playbackState->fontId = layer->channel->fontId;
playbackState->stereoHeadsetEffects = layer->channel->stereoHeadsetEffects;
sub->bitField1.reverbIndex = layer->channel->reverbIndex & 3;
sampleState->bitField1.reverbIndex = layer->channel->reverbIndex & 3;
}
/**
@@ -1016,7 +1016,7 @@ void Audio_NoteInitAll(void) {
for (i = 0; i < gAudioCtx.numNotes; i++) {
note = &gAudioCtx.notes[i];
note->noteSubEu = gZeroNoteSub;
note->sampleState = gZeroNoteSampleState;
note->playbackState.priority = 0;
note->playbackState.unk_04 = 0;
note->playbackState.parentLayer = NO_LAYER;
src/audio/internal/seqplayer.c
+1 -1
View File
@@ -2151,7 +2151,7 @@ void AudioSeq_ProcessSequences(s32 arg0) {
SequencePlayer* seqPlayer;
u32 i;
gAudioCtx.noteSubEuOffset = (gAudioCtx.audioBufferParameters.ticksPerUpdate - arg0 - 1) * gAudioCtx.numNotes;
gAudioCtx.sampleStateOffset = (gAudioCtx.audioBufferParameters.ticksPerUpdate - arg0 - 1) * gAudioCtx.numNotes;
for (i = 0; i < (u32)gAudioCtx.audioBufferParameters.numSequencePlayers; i++) {
seqPlayer = &gAudioCtx.seqPlayers[i];
src/audio/internal/synthesis.c
+71 -71
View File
@@ -28,12 +28,13 @@ Acmd* AudioSynth_LoadRingBufferPart(Acmd* cmd, u16 dmem, u16 startPos, s32 size,
Acmd* AudioSynth_SaveBufferOffset(Acmd* cmd, u16 dmem, u16 offset, s32 size, s16* buf);
Acmd* AudioSynth_SaveRingBufferPart(Acmd* cmd, u16 dmem, u16 startPos, s32 size, SynthesisReverb* reverb);
Acmd* AudioSynth_DoOneAudioUpdate(s16* aiBuf, s32 aiBufLen, Acmd* cmd, s32 updateIndex);
Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s16* aiBuf,
Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSampleState* sampleState, NoteSynthesisState* synthState, s16* aiBuf,
s32 aiBufLen, Acmd* cmd, s32 updateIndex);
Acmd* AudioSynth_LoadWaveSamples(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s32 numSamplesToLoad);
Acmd* AudioSynth_ApplyHaasEffect(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s32 size, s32 flags,
s32 haasEffectDelaySide);
Acmd* AudioSynth_ProcessEnvelope(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s32 aiBufLen,
Acmd* AudioSynth_LoadWaveSamples(Acmd* cmd, NoteSampleState* sampleState, NoteSynthesisState* synthState,
s32 numSamplesToLoad);
Acmd* AudioSynth_ApplyHaasEffect(Acmd* cmd, NoteSampleState* sampleState, NoteSynthesisState* synthState, s32 size,
s32 flags, s32 haasEffectDelaySide);
Acmd* AudioSynth_ProcessEnvelope(Acmd* cmd, NoteSampleState* sampleState, NoteSynthesisState* synthState, s32 aiBufLen,
u16 dmemSrc, s32 haasEffectDelaySide, s32 flags);
Acmd* AudioSynth_FinalResample(Acmd* cmd, NoteSynthesisState* synthState, s32 size, u16 pitch, u16 inpDmem,
s32 resampleFlags);
@@ -140,22 +141,21 @@ void AudioSynth_InitNextRingBuf(s32 chunkLen, s32 updateIndex, s32 reverbIndex)
* original name: __Nas_PushDrvReg
*/
void func_800DB03C(s32 updateIndex) {
NoteSubEu* subEu;
NoteSubEu* subEu2;
NoteSampleState* sampleState;
NoteSampleState* sampleState2;
s32 baseIndex;
s32 i;
baseIndex = gAudioCtx.numNotes * updateIndex;
for (i = 0; i < gAudioCtx.numNotes; i++) {
subEu = &gAudioCtx.notes[i].noteSubEu;
subEu2 = &gAudioCtx.noteSubsEu[baseIndex + i];
if (subEu->bitField0.enabled) {
subEu->bitField0.needsInit = false;
sampleState = &gAudioCtx.notes[i].sampleState;
sampleState2 = &gAudioCtx.sampleStates[baseIndex + i];
if (sampleState->bitField0.enabled) {
sampleState->bitField0.needsInit = false;
} else {
subEu2->bitField0.enabled = false;
sampleState2->bitField0.enabled = false;
}
subEu->harmonicIndexCurAndPrev = 0;
sampleState->harmonicIndexCurAndPrev = 0;
}
}
@@ -216,13 +216,13 @@ Acmd* AudioSynth_Update(Acmd* cmdStart, s32* cmdCnt, s16* aiStart, s32 aiBufLen)
* original name: __Nas_WaveTerminateProcess
*/
void func_800DB2C0(s32 updateIndex, s32 noteIndex) {
NoteSubEu* noteSubEu;
NoteSampleState* sampleState;
s32 i;
for (i = updateIndex + 1; i < gAudioCtx.audioBufferParameters.ticksPerUpdate; i++) {
noteSubEu = &gAudioCtx.noteSubsEu[(gAudioCtx.numNotes * i) + noteIndex];
if (!noteSubEu->bitField0.needsInit) {
noteSubEu->bitField0.enabled = false;
sampleState = &gAudioCtx.sampleStates[(gAudioCtx.numNotes * i) + noteIndex];
if (!sampleState->bitField0.needsInit) {
sampleState->bitField0.enabled = false;
} else {
break;
}
@@ -656,31 +656,31 @@ Acmd* AudioSynth_DoOneAudioUpdate(s16* aiBuf, s32 aiBufLen, Acmd* cmd, s32 updat
s32 useReverb;
s32 t;
s32 i;
NoteSubEu* noteSubEu;
NoteSubEu* noteSubEu2;
NoteSampleState* sampleState;
NoteSampleState* sampleState2;
s32 unk14;
t = gAudioCtx.numNotes * updateIndex;
count = 0;
if (gAudioCtx.numSynthesisReverbs == 0) {
for (i = 0; i < gAudioCtx.numNotes; i++) {
if (gAudioCtx.noteSubsEu[t + i].bitField0.enabled) {
if (gAudioCtx.sampleStates[t + i].bitField0.enabled) {
noteIndices[count++] = i;
}
}
} else {
for (reverbIndex = 0; reverbIndex < gAudioCtx.numSynthesisReverbs; reverbIndex++) {
for (i = 0; i < gAudioCtx.numNotes; i++) {
noteSubEu = &gAudioCtx.noteSubsEu[t + i];
if (noteSubEu->bitField0.enabled && noteSubEu->bitField1.reverbIndex == reverbIndex) {
sampleState = &gAudioCtx.sampleStates[t + i];
if (sampleState->bitField0.enabled && sampleState->bitField1.reverbIndex == reverbIndex) {
noteIndices[count++] = i;
}
}
}
for (i = 0; i < gAudioCtx.numNotes; i++) {
noteSubEu = &gAudioCtx.noteSubsEu[t + i];
if (noteSubEu->bitField0.enabled && noteSubEu->bitField1.reverbIndex >= gAudioCtx.numSynthesisReverbs) {
sampleState = &gAudioCtx.sampleStates[t + i];
if (sampleState->bitField0.enabled && sampleState->bitField1.reverbIndex >= gAudioCtx.numSynthesisReverbs) {
noteIndices[count++] = i;
}
}
@@ -733,11 +733,11 @@ Acmd* AudioSynth_DoOneAudioUpdate(s16* aiBuf, s32 aiBufLen, Acmd* cmd, s32 updat
}
while (i < count) {
noteSubEu2 = &gAudioCtx.noteSubsEu[noteIndices[i] + t];
if (noteSubEu2->bitField1.reverbIndex == reverbIndex) {
cmd =
AudioSynth_ProcessNote(noteIndices[i], noteSubEu2, &gAudioCtx.notes[noteIndices[i]].synthesisState,
aiBuf, aiBufLen, cmd, updateIndex);
sampleState2 = &gAudioCtx.sampleStates[noteIndices[i] + t];
if (sampleState2->bitField1.reverbIndex == reverbIndex) {
cmd = AudioSynth_ProcessNote(noteIndices[i], sampleState2,
&gAudioCtx.notes[noteIndices[i]].synthesisState, aiBuf, aiBufLen, cmd,
updateIndex);
} else {
break;
}
@@ -763,7 +763,7 @@ Acmd* AudioSynth_DoOneAudioUpdate(s16* aiBuf, s32 aiBufLen, Acmd* cmd, s32 updat
while (i < count) {
cmd =
AudioSynth_ProcessNote(noteIndices[i], &gAudioCtx.noteSubsEu[t + noteIndices[i]],
AudioSynth_ProcessNote(noteIndices[i], &gAudioCtx.sampleStates[t + noteIndices[i]],
&gAudioCtx.notes[noteIndices[i]].synthesisState, aiBuf, aiBufLen, cmd, updateIndex);
i++;
}
@@ -777,7 +777,7 @@ Acmd* AudioSynth_DoOneAudioUpdate(s16* aiBuf, s32 aiBufLen, Acmd* cmd, s32 updat
/**
* original name: Nas_SynthMain
*/
Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s16* aiBuf,
Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSampleState* sampleState, NoteSynthesisState* synthState, s16* aiBuf,
s32 aiBufLen, Acmd* cmd, s32 updateIndex) {
s32 pad1[3];
Sample* sample;
@@ -830,12 +830,12 @@ Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSubEu, NoteSynthesisS
s16 addr;
u16 unused;
bookOffset = noteSubEu->bitField1.bookOffset;
finished = noteSubEu->bitField0.finished;
bookOffset = sampleState->bitField1.bookOffset;
finished = sampleState->bitField0.finished;
note = &gAudioCtx.notes[noteIndex];
flags = A_CONTINUE;
if (noteSubEu->bitField0.needsInit == true) {
if (sampleState->bitField0.needsInit == true) {
flags = A_INIT;
synthState->restart = 0;
synthState->samplePosInt = note->startSamplePos;
@@ -844,15 +844,15 @@ Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSubEu, NoteSynthesisS
synthState->curVolRight = 0;
synthState->prevHaasEffectLeftDelaySize = 0;
synthState->prevHaasEffectRightDelaySize = 0;
synthState->reverbVol = noteSubEu->reverbVol;
synthState->reverbVol = sampleState->reverbVol;
synthState->numParts = 0;
synthState->combFilterNeedsInit = true;
note->noteSubEu.bitField0.finished = false;
note->sampleState.bitField0.finished = false;
finished = false;
}
resamplingRateFixedPoint = noteSubEu->resamplingRateFixedPoint;
nParts = noteSubEu->bitField1.hasTwoParts + 1;
resamplingRateFixedPoint = sampleState->resamplingRateFixedPoint;
nParts = sampleState->bitField1.hasTwoParts + 1;
samplesLenFixedPoint = (resamplingRateFixedPoint * aiBufLen * 2) + synthState->samplePosFrac;
numSamplesToLoad = samplesLenFixedPoint >> 16;
@@ -873,12 +873,12 @@ Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSubEu, NoteSynthesisS
synthState->numParts = nParts;
if (noteSubEu->bitField1.isSyntheticWave) {
cmd = AudioSynth_LoadWaveSamples(cmd, noteSubEu, synthState, numSamplesToLoad);
if (sampleState->bitField1.isSyntheticWave) {
cmd = AudioSynth_LoadWaveSamples(cmd, sampleState, synthState, numSamplesToLoad);
sampleDmemBeforeResampling = DMEM_UNCOMPRESSED_NOTE + (synthState->samplePosInt * (s32)SAMPLE_SIZE);
synthState->samplePosInt += numSamplesToLoad;
} else {
sample = noteSubEu->tunedSample->sample;
sample = sampleState->tunedSample->sample;
loopInfo = sample->loop;
loopEndPos = loopInfo->header.end;
sampleAddr = (u32)sample->sampleAddr;
@@ -1097,7 +1097,7 @@ Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSubEu, NoteSynthesisS
AudioSynth_ClearBuffer(cmd++, DMEM_UNCOMPRESSED_NOTE + s5,
(samplesLenAdjusted - nSamplesProcessed) * SAMPLE_SIZE);
finished = true;
note->noteSubEu.bitField0.finished = true;
note->sampleState.bitField0.finished = true;
func_800DB2C0(updateIndex, noteIndex);
break;
} else {
@@ -1143,8 +1143,8 @@ Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSubEu, NoteSynthesisS
}
flags = A_CONTINUE;
if (noteSubEu->bitField0.needsInit == true) {
noteSubEu->bitField0.needsInit = false;
if (sampleState->bitField0.needsInit == true) {
sampleState->bitField0.needsInit = false;
flags = A_INIT;
}
@@ -1158,7 +1158,7 @@ Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSubEu, NoteSynthesisS
AudioSynth_UnkCmd3(cmd++, DMEM_TEMP, DMEM_TEMP, aiBufLen * SAMPLE_SIZE);
}
gain = noteSubEu->gain;
gain = sampleState->gain;
if (gain != 0) {
// A gain of 0x10 (a UQ4.4 number) is equivalent to 1.0 and represents no volume change
if (gain < 0x10) {
@@ -1167,7 +1167,7 @@ Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSubEu, NoteSynthesisS
AudioSynth_HiLoGain(cmd++, gain, DMEM_TEMP, 0, (aiBufLen + SAMPLES_PER_FRAME) * SAMPLE_SIZE);
}
filter = noteSubEu->filter;
filter = sampleState->filter;
if (filter != NULL) {
AudioSynth_LoadFilterSize(cmd++, aiBufLen * SAMPLE_SIZE, filter);
AudioSynth_LoadFilterBuffer(cmd++, flags, DMEM_TEMP, synthState->synthesisBuffers->mixEnvelopeState);
@@ -1175,10 +1175,10 @@ Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSubEu, NoteSynthesisS
// Apply the comb filter to the mono-signal by taking the signal with a small temporal offset,
// and adding it back to itself
combFilterSize = noteSubEu->combFilterSize;
combFilterGain = noteSubEu->combFilterGain;
combFilterSize = sampleState->combFilterSize;
combFilterGain = sampleState->combFilterGain;
combFilterState = synthState->synthesisBuffers->combFilterState;
if ((combFilterSize != 0) && (noteSubEu->combFilterGain != 0)) {
if ((combFilterSize != 0) && (sampleState->combFilterGain != 0)) {
AudioSynth_DMemMove(cmd++, DMEM_TEMP, DMEM_COMB_TEMP, aiBufLen * SAMPLE_SIZE);
combFilterDmem = DMEM_COMB_TEMP - combFilterSize;
if (synthState->combFilterNeedsInit) {
@@ -1195,21 +1195,21 @@ Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSubEu, NoteSynthesisS
synthState->combFilterNeedsInit = true;
}
if ((noteSubEu->haasEffectLeftDelaySize != 0) || (synthState->prevHaasEffectLeftDelaySize != 0)) {
if ((sampleState->haasEffectLeftDelaySize != 0) || (synthState->prevHaasEffectLeftDelaySize != 0)) {
haasEffectDelaySide = HAAS_EFFECT_DELAY_LEFT;
} else if ((noteSubEu->haasEffectRightDelaySize != 0) || (synthState->prevHaasEffectRightDelaySize != 0)) {
} else if ((sampleState->haasEffectRightDelaySize != 0) || (synthState->prevHaasEffectRightDelaySize != 0)) {
haasEffectDelaySide = HAAS_EFFECT_DELAY_RIGHT;
} else {
haasEffectDelaySide = HAAS_EFFECT_DELAY_NONE;
}
cmd = AudioSynth_ProcessEnvelope(cmd, noteSubEu, synthState, aiBufLen, DMEM_TEMP, haasEffectDelaySide, flags);
cmd = AudioSynth_ProcessEnvelope(cmd, sampleState, synthState, aiBufLen, DMEM_TEMP, haasEffectDelaySide, flags);
if (noteSubEu->bitField1.useHaasEffect) {
if (sampleState->bitField1.useHaasEffect) {
if (!(flags & A_INIT)) {
flags = A_CONTINUE;
}
cmd = AudioSynth_ApplyHaasEffect(cmd, noteSubEu, synthState, aiBufLen * (s32)SAMPLE_SIZE, flags,
cmd = AudioSynth_ApplyHaasEffect(cmd, sampleState, synthState, aiBufLen * (s32)SAMPLE_SIZE, flags,
haasEffectDelaySide);
}
@@ -1233,7 +1233,7 @@ Acmd* AudioSynth_FinalResample(Acmd* cmd, NoteSynthesisState* synthState, s32 si
/**
* original name: Nas_Synth_Envelope
*/
Acmd* AudioSynth_ProcessEnvelope(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s32 aiBufLen,
Acmd* AudioSynth_ProcessEnvelope(Acmd* cmd, NoteSampleState* sampleState, NoteSynthesisState* synthState, s32 aiBufLen,
u16 dmemSrc, s32 haasEffectDelaySide, s32 flags) {
u32 dmemDests;
u16 curVolLeft;
@@ -1249,11 +1249,11 @@ Acmd* AudioSynth_ProcessEnvelope(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisS
s32 pad;
curVolLeft = synthState->curVolLeft;
targetVolLeft = noteSubEu->targetVolLeft;
targetVolLeft = sampleState->targetVolLeft;
targetVolLeft <<= 4;
reverbVol = noteSubEu->reverbVol;
reverbVol = sampleState->reverbVol;
curVolRight = synthState->curVolRight;
targetVolRight = noteSubEu->targetVolRight;
targetVolRight = sampleState->targetVolRight;
targetVolRight <<= 4;
if (targetVolLeft != curVolLeft) {
@@ -1280,7 +1280,7 @@ Acmd* AudioSynth_ProcessEnvelope(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisS
synthState->curVolLeft = curVolLeft + (rampLeft * (aiBufLen >> 3));
synthState->curVolRight = curVolRight + (rampRight * (aiBufLen >> 3));
if (noteSubEu->bitField1.useHaasEffect) {
if (sampleState->bitField1.useHaasEffect) {
AudioSynth_ClearBuffer(cmd++, DMEM_HAAS_TEMP, DMEM_1CH_SIZE);
AudioSynth_EnvSetup1(cmd++, phi_t1 * 2, rampReverb, rampLeft, rampRight);
AudioSynth_EnvSetup2(cmd++, curVolLeft, curVolRight);
@@ -1306,21 +1306,21 @@ Acmd* AudioSynth_ProcessEnvelope(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisS
dmemDests = sEnvMixerDefaultDmemDests;
}
aEnvMixer(cmd++, dmemSrc, aiBufLen, (sourceReverbVol & 0x80) >> 7, noteSubEu->bitField0.stereoHeadsetEffects,
noteSubEu->bitField0.usesHeadsetPanEffects, noteSubEu->bitField0.stereoStrongRight,
noteSubEu->bitField0.stereoStrongLeft, dmemDests, sEnvMixerOp);
aEnvMixer(cmd++, dmemSrc, aiBufLen, (sourceReverbVol & 0x80) >> 7, sampleState->bitField0.stereoHeadsetEffects,
sampleState->bitField0.usesHeadsetPanEffects, sampleState->bitField0.stereoStrongRight,
sampleState->bitField0.stereoStrongLeft, dmemDests, sEnvMixerOp);
return cmd;
}
Acmd* AudioSynth_LoadWaveSamples(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisState* synthState,
Acmd* AudioSynth_LoadWaveSamples(Acmd* cmd, NoteSampleState* sampleState, NoteSynthesisState* synthState,
s32 numSamplesToLoad) {
s32 numSamplesAvail;
s32 harmonicIndexCurAndPrev = noteSubEu->harmonicIndexCurAndPrev;
s32 harmonicIndexCurAndPrev = sampleState->harmonicIndexCurAndPrev;
s32 samplePosInt = synthState->samplePosInt;
s32 numDuplicates;
if (noteSubEu->bitField1.bookOffset != 0) {
if (sampleState->bitField1.bookOffset != 0) {
// Move the noise wave (that reads compiled assembly as samples) from ram to dmem
AudioSynth_LoadBuffer(cmd++, DMEM_UNCOMPRESSED_NOTE, ALIGN16(numSamplesToLoad * SAMPLE_SIZE), gWaveSamples[8]);
// Offset the address for the samples read by gWaveSamples[8] to the next set of samples
@@ -1328,7 +1328,7 @@ Acmd* AudioSynth_LoadWaveSamples(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisS
return cmd;
} else {
// Move the synthetic wave from ram to dmem
aLoadBuffer(cmd++, noteSubEu->waveSampleAddr, DMEM_UNCOMPRESSED_NOTE, WAVE_SAMPLE_COUNT * SAMPLE_SIZE);
aLoadBuffer(cmd++, sampleState->waveSampleAddr, DMEM_UNCOMPRESSED_NOTE, WAVE_SAMPLE_COUNT * SAMPLE_SIZE);
// If the harmonic changes, map the offset in the wave from one harmonic to another for continuity
if (harmonicIndexCurAndPrev != 0) {
@@ -1364,8 +1364,8 @@ Acmd* AudioSynth_LoadWaveSamples(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisS
*
* original name: Nas_Synth_Delay
*/
Acmd* AudioSynth_ApplyHaasEffect(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s32 size, s32 flags,
s32 haasEffectDelaySide) {
Acmd* AudioSynth_ApplyHaasEffect(Acmd* cmd, NoteSampleState* sampleState, NoteSynthesisState* synthState, s32 size,
s32 flags, s32 haasEffectDelaySide) {
u16 dmemDest;
u16 pitch;
u8 prevHaasEffectDelaySize;
@@ -1376,7 +1376,7 @@ Acmd* AudioSynth_ApplyHaasEffect(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisS
// Delay the sample on the left channel
// This allows the right channel to be heard first
dmemDest = DMEM_LEFT_CH;
haasEffectDelaySize = noteSubEu->haasEffectLeftDelaySize;
haasEffectDelaySize = sampleState->haasEffectLeftDelaySize;
prevHaasEffectDelaySize = synthState->prevHaasEffectLeftDelaySize;
synthState->prevHaasEffectRightDelaySize = 0;
synthState->prevHaasEffectLeftDelaySize = haasEffectDelaySize;
@@ -1386,7 +1386,7 @@ Acmd* AudioSynth_ApplyHaasEffect(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisS
// Delay the sample on the right channel
// This allows the left channel to be heard first
dmemDest = DMEM_RIGHT_CH;
haasEffectDelaySize = noteSubEu->haasEffectRightDelaySize;
haasEffectDelaySize = sampleState->haasEffectRightDelaySize;
prevHaasEffectDelaySize = synthState->prevHaasEffectRightDelaySize;
synthState->prevHaasEffectRightDelaySize = haasEffectDelaySize;
synthState->prevHaasEffectLeftDelaySize = 0;
src/audio/internal/thread.c
+9 -9
View File
@@ -255,11 +255,11 @@ void AudioThread_ProcessGlobalCmd(AudioCmd* cmd) {
if (cmd->asUInt == 1) {
for (i = 0; i < gAudioCtx.numNotes; i++) {
Note* note = &gAudioCtx.notes[i];
NoteSubEu* subEu = &note->noteSubEu;
NoteSampleState* sampleState = &note->sampleState;
if (subEu->bitField0.enabled && (note->playbackState.unk_04 == 0) &&
if (sampleState->bitField0.enabled && (note->playbackState.unk_04 == 0) &&
(note->playbackState.parentLayer->channel->muteBehavior & MUTE_BEHAVIOR_3)) {
subEu->bitField0.finished = true;
sampleState->bitField0.finished = true;
}
}
}
@@ -874,7 +874,7 @@ s32 func_800E6590(s32 seqPlayerIndex, s32 channelIndex, s32 layerIndex) {
note = layer->note;
if (layer == note->playbackState.parentLayer) {
tunedSample = note->noteSubEu.tunedSample;
tunedSample = note->sampleState.tunedSample;
if (tunedSample == NULL) {
return 0;
}
@@ -905,7 +905,7 @@ void func_800E66A0(void) {
s32 func_800E66C0(s32 flags) {
s32 phi_v1;
NotePlaybackState* playbackState;
NoteSubEu* noteSubEu;
NoteSampleState* sampleState;
s32 i;
Note* note;
TunedSample* tunedSample;
@@ -914,12 +914,12 @@ s32 func_800E66C0(s32 flags) {
for (i = 0; i < gAudioCtx.numNotes; i++) {
note = &gAudioCtx.notes[i];
playbackState = &note->playbackState;
if (note->noteSubEu.bitField0.enabled) {
noteSubEu = &note->noteSubEu;
if (note->sampleState.bitField0.enabled) {
sampleState = &note->sampleState;
if (playbackState->adsr.action.s.state != 0) {
if (flags >= 2) {
tunedSample = noteSubEu->tunedSample;
if (tunedSample == NULL || noteSubEu->bitField1.isSyntheticWave) {
tunedSample = sampleState->tunedSample;
if (tunedSample == NULL || sampleState->bitField1.isSyntheticWave) {
continue;
}
if (tunedSample->sample->medium == MEDIUM_RAM) {