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cleanup
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6c76c1875d
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1b057824f7
7 changed files with 120 additions and 150 deletions
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@ -621,46 +621,46 @@ void func_800DF1D8(f32 arg0, f32 arg1, u16* arg2) {
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}
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}
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void func_800DF5AC(s16* arg0) {
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void func_800DF5AC(s16* filter) {
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s32 i;
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for (i = 0; i < 8; i++) {
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arg0[i] = 0;
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filter[i] = 0;
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}
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}
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void func_800DF5DC(s16* arg0, s32 arg1) {
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void func_800DF5DC(s16* filter, s32 arg1) {
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s32 i;
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s16* ptr = &D_80130228[8 * arg1];
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for (i = 0; i < 8; i++) {
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arg0[i] = ptr[i];
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filter[i] = ptr[i];
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}
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}
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void func_800DF630(s16* arg0, s32 arg1) {
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void func_800DF630(s16* filter, s32 arg1) {
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s32 i;
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s16* ptr = &D_80130328[8 * (arg1 - 1)];
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for (i = 0; i < 8; i++) {
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arg0[i] = ptr[i];
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filter[i] = ptr[i];
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}
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}
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void func_800DF688(s16* arg0, s32 arg1, s32 arg2) {
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void func_800DF688(s16* filter, s32 arg1, s32 arg2) {
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s32 i;
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if (arg1 == 0 && arg2 == 0) {
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func_800DF5DC(arg0, 0);
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func_800DF5DC(filter, 0);
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} else if (arg2 == 0) {
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func_800DF5DC(arg0, arg1);
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func_800DF5DC(filter, arg1);
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} else if (arg1 == 0) {
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func_800DF630(arg0, arg2);
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func_800DF630(filter, arg2);
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} else {
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s16* ptr1 = &D_80130228[8 * arg1];
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s16* ptr2 = &D_80130328[8 * (arg2 - 1)];
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for (i = 0; i < 8; i++) {
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arg0[i] = (ptr1[i] + ptr2[i]) / 2;
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filter[i] = (ptr1[i] + ptr2[i]) / 2;
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}
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}
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}
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@ -6,7 +6,7 @@ typedef enum { LOAD_STATUS_WAITING, LOAD_STATUS_START, LOAD_STATUS_LOADING, LOAD
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#define RELOC(v, base) (reloc = (void*)((u32)v + (u32)base))
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/* forward declarations */
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s32 func_800E217C(s32 playerIdx, s32, s32);
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s32 func_800E217C(s32 playerIndex, s32, s32);
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unk_ldr* func_800E2454(u32 bankId);
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AudioBankSample* Audio_GetBankSample(s32 bankId, s32 sfxId);
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void Audio_ProcessAsyncLoads(s32 arg0);
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@ -381,27 +381,27 @@ void func_800E202C(s32 arg0) {
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Audio_DiscardBank(arg0);
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}
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s32 func_800E20D4(s32 playerIdx, s32 seqId, s32 arg2) {
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s32 func_800E20D4(s32 playerIndex, s32 seqId, s32 arg2) {
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if (gAudioContext.resetTimer != 0) {
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return 0;
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} else {
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gAudioContext.seqPlayers[playerIdx].unk_DC = 0;
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return func_800E217C(playerIdx, seqId, arg2);
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gAudioContext.seqPlayers[playerIndex].unk_DC = 0;
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return func_800E217C(playerIndex, seqId, arg2);
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}
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}
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s32 func_800E2124(s32 playerIdx, s32 seqId, s32 arg2) {
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s32 func_800E2124(s32 playerIndex, s32 seqId, s32 arg2) {
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if (gAudioContext.resetTimer != 0) {
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return 0;
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}
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gAudioContext.seqPlayers[playerIdx].unk_DC = arg2;
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return func_800E217C(playerIdx, seqId, 0);
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gAudioContext.seqPlayers[playerIndex].unk_DC = arg2;
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return func_800E217C(playerIndex, seqId, 0);
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}
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// InitSeqPlayer
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s32 func_800E217C(s32 playerIdx, s32 seqId, s32 arg2) {
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SequencePlayer* seqPlayer = &gAudioContext.seqPlayers[playerIdx];
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s32 func_800E217C(s32 playerIndex, s32 seqId, s32 arg2) {
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SequencePlayer* seqPlayer = &gAudioContext.seqPlayers[playerIndex];
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u8* seqData;
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s32 phi_s0;
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s32 phi_s1;
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@ -437,7 +437,7 @@ s32 func_800E217C(s32 playerIdx, s32 seqId, s32 arg2) {
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seqPlayer->scriptState.depth = 0;
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seqPlayer->delay = 0;
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seqPlayer->finished = 0;
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seqPlayer->seqVariationEu = playerIdx;
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seqPlayer->playerIndex = playerIndex;
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Audio_ProcessSequence(seqPlayer);
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}
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@ -95,7 +95,7 @@ void Audio_NoteSetVelPanReverb(Note* note, NoteSubEu* sub, Reverb* reverb) {
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sub->targetVolRight = (s32)((vel * volRight) * 4095.999f);
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sub->unk_2 = reverb->unk_1;
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sub->unk_14 = reverb->unk_10;
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sub->filter = reverb->filter;
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sub->unk_07 = reverb->unk_14;
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sub->unk_0E = reverb->unk_16;
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sub->reverbVol = reverbVol;
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@ -256,7 +256,7 @@ void Audio_ProcessNotes(void) {
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reverb.vol = attributes->reverb;
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reverb.reverbBits = attributes->reverbBits;
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reverb.unk_1 = attributes->unk_1;
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reverb.unk_10 = attributes->unk_10;
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reverb.filter = attributes->filter;
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reverb.unk_14 = attributes->unk_4;
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reverb.unk_16 = attributes->unk_6;
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bookOffset = noteSubEu->bitField1.s.bookOffset;
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@ -274,7 +274,7 @@ void Audio_ProcessNotes(void) {
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}
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reverb.vol = channel->reverb;
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reverb.unk_1 = channel->unk_0C;
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reverb.unk_10 = channel->unk_CC;
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reverb.filter = channel->filter;
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reverb.unk_14 = channel->unk_0F;
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reverb.unk_16 = channel->unk_20;
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bookOffset = channel->bookOffset & 0x7;
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@ -471,13 +471,13 @@ void Audio_SeqChanLayerDecayRelease(SequenceChannelLayer* seqLayer, s32 target)
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chan = seqLayer->seqChannel;
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attributes->reverb = chan->reverb;
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attributes->unk_1 = chan->unk_0C;
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attributes->unk_10 = chan->unk_CC;
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attributes->filter = chan->filter;
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if (attributes->unk_10 != NULL) {
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if (attributes->filter != NULL) {
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for (i = 0; i < 8; i++) {
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attributes->unk_14[i] = attributes->unk_10[i];
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attributes->filterBuf[i] = attributes->filter[i];
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}
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attributes->unk_10 = attributes->unk_14;
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attributes->filter = attributes->filterBuf;
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}
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attributes->unk_6 = chan->unk_20;
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@ -144,7 +144,7 @@ void Audio_SequenceChannelInit(SequenceChannel* seqChannel) {
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seqChannel->vibratoRateChangeDelay = 0;
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seqChannel->vibratoExtentChangeDelay = 0;
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seqChannel->vibratoDelay = 0;
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seqChannel->unk_CC = NULL;
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seqChannel->filter = NULL;
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seqChannel->unk_20 = 0;
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seqChannel->unk_0F = 0;
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seqChannel->volume = 1.0f;
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@ -1275,7 +1275,7 @@ void Audio_SequenceChannelProcessScript(SequenceChannel* channel) {
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channel->vibratoRateTarget = 0;
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channel->vibratoRateStart = 0;
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channel->vibratoRateChangeDelay = 0;
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channel->unk_CC = NULL;
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channel->filter = NULL;
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channel->unk_0C = 0;
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channel->adsr.sustain = 0;
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channel->velocityRandomVariance = 0;
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@ -1295,18 +1295,18 @@ void Audio_SequenceChannelProcessScript(SequenceChannel* channel) {
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case 0xB0:
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offset = (u16)parameters[0];
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data = seqPlayer->seqData + offset;
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channel->unk_CC = (s16*)data;
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channel->filter = (s16*)data;
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break;
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case 0xB1:
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channel->unk_CC = NULL;
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channel->filter = NULL;
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break;
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case 0xB3:
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command = parameters[0];
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if (channel->unk_CC != NULL) {
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if (channel->filter != NULL) {
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lowBits = (command >> 4) & 0xF;
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command &= 0xF;
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func_800DF688(channel->unk_CC, lowBits, command);
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func_800DF688(channel->filter, lowBits, command);
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}
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break;
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case 0xB2:
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@ -1676,11 +1676,11 @@ void Audio_SequencePlayerProcessSequence(SequencePlayer* seqPlayer) {
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case 0xC4:
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command = Audio_M64ReadU8(seqScript);
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if (command == 0xFF) {
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command = seqPlayer->seqVariationEu;
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command = seqPlayer->playerIndex;
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}
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commandLow = Audio_M64ReadU8(seqScript);
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func_800E20D4(command, commandLow, 0);
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if (command == (u8)seqPlayer->seqVariationEu) {
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if (command == (u8)seqPlayer->playerIndex) {
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return;
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}
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break;
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@ -34,7 +34,7 @@ extern u32 D_801304A8;
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extern u32 D_801304AC;
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extern u8 D_801304C0[];
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void AudioSynth_InitNextRingBuf(s32 chunkSize, s32 bufIndex, s32 reverbIndex) {
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void AudioSynth_InitNextRingBuf(s32 chunkLen, s32 bufIndex, s32 reverbIndex) {
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ReverbRingBufferItem* bufItem;
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s32 pad[3];
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SynthesisReverb* reverb = &gAudioContext.synthesisReverbs[reverbIndex];
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@ -64,7 +64,7 @@ void AudioSynth_InitNextRingBuf(s32 chunkSize, s32 bufIndex, s32 reverbIndex) {
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}
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bufItem = &reverb->items[reverb->curFrame][bufIndex];
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sampleCnt = chunkSize / reverb->downsampleRate;
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sampleCnt = chunkLen / reverb->downsampleRate;
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extraSamples = (sampleCnt + reverb->nextRingBufPos) - reverb->bufSizePerChan;
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temp_a0_2 = reverb->nextRingBufPos;
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if (extraSamples < 0) {
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@ -80,7 +80,7 @@ void AudioSynth_InitNextRingBuf(s32 chunkSize, s32 bufIndex, s32 reverbIndex) {
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}
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bufItem->numSamplesAfterDownsampling = sampleCnt;
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bufItem->chunkLen = chunkSize;
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bufItem->chunkLen = chunkLen;
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if (reverb->unk_14 != 0) {
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temp_a0_4 = reverb->unk_14 + temp_a0_2;
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@ -88,7 +88,7 @@ void AudioSynth_InitNextRingBuf(s32 chunkSize, s32 bufIndex, s32 reverbIndex) {
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temp_a0_4 -= reverb->bufSizePerChan;
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}
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bufItem = &reverb->items2[reverb->curFrame][bufIndex];
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sampleCnt = chunkSize / reverb->downsampleRate;
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sampleCnt = chunkLen / reverb->downsampleRate;
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extraSamples = (temp_a0_4 + sampleCnt) - reverb->bufSizePerChan;
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if (extraSamples < 0) {
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bufItem->lengthA = sampleCnt * 2;
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@ -100,7 +100,7 @@ void AudioSynth_InitNextRingBuf(s32 chunkSize, s32 bufIndex, s32 reverbIndex) {
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bufItem->startPos = temp_a0_4;
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}
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bufItem->numSamplesAfterDownsampling = sampleCnt;
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bufItem->chunkLen = chunkSize;
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bufItem->chunkLen = chunkLen;
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}
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}
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@ -125,7 +125,7 @@ void func_800DB03C(s32 arg0) {
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}
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Acmd* AudioSynth_Update(Acmd* cmdStart, s32* cmdCnt, s16* aiStart, s32 aiBufLen) {
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s32 chunkSize;
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s32 chunkLen;
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s16* aiBufP;
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Acmd* cmdP;
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s32 i;
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@ -142,25 +142,25 @@ Acmd* AudioSynth_Update(Acmd* cmdStart, s32* cmdCnt, s16* aiStart, s32 aiBufLen)
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gAudioContext.curLoadedBook = NULL;
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for (i = gAudioContext.audioBufferParameters.updatesPerFrame; i > 0; i--) {
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if (i == 1) {
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chunkSize = aiBufLen;
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chunkLen = aiBufLen;
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} else if ((aiBufLen / i) >= gAudioContext.audioBufferParameters.samplesPerUpdateMax) {
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chunkSize = gAudioContext.audioBufferParameters.samplesPerUpdateMax;
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chunkLen = gAudioContext.audioBufferParameters.samplesPerUpdateMax;
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} else if (gAudioContext.audioBufferParameters.samplesPerUpdateMin >= (aiBufLen / i)) {
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chunkSize = gAudioContext.audioBufferParameters.samplesPerUpdateMin;
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chunkLen = gAudioContext.audioBufferParameters.samplesPerUpdateMin;
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} else {
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chunkSize = gAudioContext.audioBufferParameters.samplesPerUpdate;
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chunkLen = gAudioContext.audioBufferParameters.samplesPerUpdate;
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}
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for (j = 0; j < gAudioContext.numSynthesisReverbs; j++) {
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if (gAudioContext.synthesisReverbs[j].useReverb) {
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AudioSynth_InitNextRingBuf(chunkSize, gAudioContext.audioBufferParameters.updatesPerFrame - i, j);
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AudioSynth_InitNextRingBuf(chunkLen, gAudioContext.audioBufferParameters.updatesPerFrame - i, j);
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}
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}
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cmdP = AudioSynth_DoOneAudioUpdate(aiBufP, chunkSize, cmdP,
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cmdP = AudioSynth_DoOneAudioUpdate(aiBufP, chunkLen, cmdP,
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gAudioContext.audioBufferParameters.updatesPerFrame - i);
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aiBufLen -= chunkSize;
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aiBufP += chunkSize * 2;
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aiBufLen -= chunkLen;
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aiBufP += chunkLen * 2;
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}
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for (j = 0; j < gAudioContext.numSynthesisReverbs; j++) {
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@ -702,7 +702,7 @@ Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSubEu, NoteSynthesisS
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u32 nSamplesToLoad;
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u16 unk7;
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u16 unkE;
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s16* thing4;
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s16* filter;
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s32 bookOffset;
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s32 finished;
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s32 aligned;
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@ -745,7 +745,6 @@ Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSubEu, NoteSynthesisS
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if (noteSubEu->bitField1.s.isSyntheticWave) {
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cmd = AudioSynth_LoadWaveSamples(cmd, noteSubEu, synthState, nSamplesToLoad);
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// loopInfo = loopInfo;
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noteSamplesDmemAddrBeforeResampling = DMEM_UNCOMPRESSED_NOTE + (synthState->samplePosInt * 2);
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synthState->samplePosInt += nSamplesToLoad;
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} else {
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@ -1014,9 +1013,9 @@ Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSubEu, NoteSynthesisS
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AudioSynth_HiLoGain(cmd++, phi_a1_2, DMEM_TEMP, 0, (aiBufLen * 2) + 0x20);
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}
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thing4 = noteSubEu->unk_14;
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if (thing4 != 0) {
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AudioSynth_SetFilterCount(cmd++, aiBufLen * 2, thing4);
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filter = noteSubEu->filter;
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if (filter != 0) {
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AudioSynth_SetFilterCount(cmd++, aiBufLen * 2, filter);
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AudioSynth_SetFilter(cmd++, flags, DMEM_TEMP, synthState->synthesisBuffers->mixEnvelopeState);
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}
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