mirror of
https://github.com/zeldaret/oot.git
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bd4912a1bd
* Audio Cleanup * more * missed one * Audio docs imported from MM * Suggestions from Discord * Small fix * Fix sizes in z64audio.h * numSamples * Cleanup comments * Revert overlap with soundFont PR * revert Audio_AdsrUpdate cleanup (move to different PR) * PR Suggestion * Mainly typos * PR suggestion, vol -> volume * Better description * Name sampling frequency * A good number of PR suggestions * Missed a suggestion * More PR Suggestions * Cleanup NotePlaybackState * one more * Add comments for new variable names * Reverb samplePosInt * Missed one * Remove `Always set to 0`
1272 lines
48 KiB
C
1272 lines
48 KiB
C
#include "ultra64.h"
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#include "global.h"
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#define DEFAULT_LEN_1CH 0x1A0
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#define DEFAULT_LEN_2CH 0x340
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// DMEM Addresses for the RSP
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#define DMEM_TEMP 0x3C0
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#define DMEM_UNCOMPRESSED_NOTE 0x580
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#define DMEM_NOTE_PAN_TEMP 0x5C0
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#define DMEM_SCRATCH2 0x760 // = DMEM_TEMP + DEFAULT_LEN_2CH + a bit more
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#define DMEM_COMPRESSED_ADPCM_DATA 0x940 // = DMEM_LEFT_CH
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#define DMEM_LEFT_CH 0x940
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#define DMEM_RIGHT_CH 0xAE0
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#define DMEM_WET_TEMP 0x3E0
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#define DMEM_WET_SCRATCH 0x720 // = DMEM_WET_TEMP + DEFAULT_LEN_2CH
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#define DMEM_WET_LEFT_CH 0xC80
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#define DMEM_WET_RIGHT_CH 0xE20 // = DMEM_WET_LEFT_CH + DEFAULT_LEN_1CH
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Acmd* AudioSynth_LoadRingBufferPart(Acmd* cmd, u16 dmem, u16 startPos, s32 length, SynthesisReverb* reverb);
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Acmd* AudioSynth_SaveBufferOffset(Acmd* cmd, u16 dmem, u16 offset, s32 length, s16* buf);
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Acmd* AudioSynth_SaveRingBufferPart(Acmd* cmd, u16 dmem, u16 startPos, s32 length, SynthesisReverb* reverb);
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Acmd* AudioSynth_DoOneAudioUpdate(s16* aiBuf, s32 aiBufLen, Acmd* cmd, s32 updateIndex);
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Acmd* AudioSynth_ProcessNote(s32 noteIndex, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s16* aiBuf,
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s32 aiBufLen, Acmd* cmd, s32 updateIndex);
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Acmd* AudioSynth_LoadWaveSamples(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s32 nSamplesToLoad);
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Acmd* AudioSynth_NoteApplyHeadsetPanEffects(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s32 bufLen,
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s32 flags, s32 side);
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Acmd* AudioSynth_ProcessEnvelope(Acmd* cmd, NoteSubEu* noteSubEu, NoteSynthesisState* synthState, s32 aiBufLen,
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u16 inBuf, s32 headsetPanSettings, s32 flags);
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Acmd* AudioSynth_FinalResample(Acmd* cmd, NoteSynthesisState* synthState, s32 count, u16 pitch, u16 inpDmem,
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s32 resampleFlags);
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u32 D_801304A0 = _SHIFTL(A_ENVMIXER, 24, 8);
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u32 D_801304A4 = MK_CMD(DMEM_NOTE_PAN_TEMP >> 4, DMEM_RIGHT_CH >> 4, DMEM_WET_LEFT_CH >> 4, DMEM_WET_RIGHT_CH >> 4);
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u32 D_801304A8 = MK_CMD(DMEM_LEFT_CH >> 4, DMEM_NOTE_PAN_TEMP >> 4, DMEM_WET_LEFT_CH >> 4, DMEM_WET_RIGHT_CH >> 4);
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u32 D_801304AC = MK_CMD(DMEM_LEFT_CH >> 4, DMEM_RIGHT_CH >> 4, DMEM_WET_LEFT_CH >> 4, DMEM_WET_RIGHT_CH >> 4);
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u16 D_801304B0[] = {
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0x7FFF, 0xD001, 0x3FFF, 0xF001, 0x5FFF, 0x9001, 0x7FFF, 0x8001,
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};
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u8 D_801304C0[] = { 0x40, 0x20, 0x10, 0x8 };
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void AudioSynth_InitNextRingBuf(s32 chunkLen, s32 updateIndex, 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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s32 temp_a0_2;
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s32 temp_a0_4;
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s32 numSamples;
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s32 extraSamples;
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s32 i;
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s32 j;
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if (reverb->downsampleRate >= 2) {
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if (reverb->framesToIgnore == 0) {
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bufItem = &reverb->items[reverb->curFrame][updateIndex];
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Audio_InvalDCache(bufItem->toDownsampleLeft, DEFAULT_LEN_2CH);
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for (j = 0, i = 0; i < bufItem->lengthA / 2; j += reverb->downsampleRate, i++) {
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reverb->leftRingBuf[bufItem->startPos + i] = bufItem->toDownsampleLeft[j];
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reverb->rightRingBuf[bufItem->startPos + i] = bufItem->toDownsampleRight[j];
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}
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for (i = 0; i < bufItem->lengthB / 2; j += reverb->downsampleRate, i++) {
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reverb->leftRingBuf[i] = bufItem->toDownsampleLeft[j];
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reverb->rightRingBuf[i] = bufItem->toDownsampleRight[j];
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}
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}
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}
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bufItem = &reverb->items[reverb->curFrame][updateIndex];
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numSamples = chunkLen / reverb->downsampleRate;
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extraSamples = (numSamples + reverb->nextRingBufPos) - reverb->bufSizePerChan;
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temp_a0_2 = reverb->nextRingBufPos;
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if (extraSamples < 0) {
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bufItem->lengthA = numSamples * 2;
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bufItem->lengthB = 0;
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bufItem->startPos = reverb->nextRingBufPos;
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reverb->nextRingBufPos += numSamples;
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} else {
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// End of the buffer is reached. Loop back around
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bufItem->lengthA = (numSamples - extraSamples) * 2;
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bufItem->lengthB = extraSamples * 2;
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bufItem->startPos = reverb->nextRingBufPos;
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reverb->nextRingBufPos = extraSamples;
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}
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bufItem->numSamplesAfterDownsampling = numSamples;
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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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if (temp_a0_4 >= reverb->bufSizePerChan) {
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temp_a0_4 -= reverb->bufSizePerChan;
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}
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bufItem = &reverb->items2[reverb->curFrame][updateIndex];
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numSamples = chunkLen / reverb->downsampleRate;
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extraSamples = (temp_a0_4 + numSamples) - reverb->bufSizePerChan;
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if (extraSamples < 0) {
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bufItem->lengthA = numSamples * 2;
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bufItem->lengthB = 0;
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bufItem->startPos = temp_a0_4;
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} else {
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// End of the buffer is reached. Loop back around
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bufItem->lengthA = (numSamples - extraSamples) * 2;
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bufItem->lengthB = extraSamples * 2;
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bufItem->startPos = temp_a0_4;
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}
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bufItem->numSamplesAfterDownsampling = numSamples;
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bufItem->chunkLen = chunkLen;
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}
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}
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void func_800DB03C(s32 updateIndex) {
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NoteSubEu* subEu;
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NoteSubEu* subEu2;
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s32 baseIndex;
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s32 i;
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baseIndex = gAudioContext.numNotes * updateIndex;
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for (i = 0; i < gAudioContext.numNotes; i++) {
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subEu = &gAudioContext.notes[i].noteSubEu;
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subEu2 = &gAudioContext.noteSubsEu[baseIndex + i];
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if (subEu->bitField0.enabled) {
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subEu->bitField0.needsInit = false;
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} else {
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subEu2->bitField0.enabled = false;
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}
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subEu->unk_06 = 0;
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}
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}
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Acmd* AudioSynth_Update(Acmd* cmdStart, s32* cmdCnt, s16* aiStart, s32 aiBufLen) {
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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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s32 j;
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SynthesisReverb* reverb;
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cmdP = cmdStart;
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for (i = gAudioContext.audioBufferParameters.updatesPerFrame; i > 0; i--) {
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AudioSeq_ProcessSequences(i - 1);
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func_800DB03C(gAudioContext.audioBufferParameters.updatesPerFrame - i);
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}
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aiBufP = aiStart;
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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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chunkLen = aiBufLen;
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} else if ((aiBufLen / i) >= 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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chunkLen = gAudioContext.audioBufferParameters.samplesPerUpdateMin;
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} else {
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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(chunkLen, gAudioContext.audioBufferParameters.updatesPerFrame - i, j);
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}
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}
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cmdP = AudioSynth_DoOneAudioUpdate(aiBufP, chunkLen, cmdP,
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gAudioContext.audioBufferParameters.updatesPerFrame - i);
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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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if (gAudioContext.synthesisReverbs[j].framesToIgnore != 0) {
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gAudioContext.synthesisReverbs[j].framesToIgnore--;
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}
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gAudioContext.synthesisReverbs[j].curFrame ^= 1;
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}
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*cmdCnt = cmdP - cmdStart;
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return cmdP;
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}
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void func_800DB2C0(s32 updateIndex, s32 noteIndex) {
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NoteSubEu* noteSubEu;
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s32 i;
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for (i = updateIndex + 1; i < gAudioContext.audioBufferParameters.updatesPerFrame; i++) {
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noteSubEu = &gAudioContext.noteSubsEu[(gAudioContext.numNotes * i) + noteIndex];
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if (!noteSubEu->bitField0.needsInit) {
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noteSubEu->bitField0.enabled = false;
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} else {
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break;
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}
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}
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}
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Acmd* AudioSynth_LoadRingBuffer1AtTemp(Acmd* cmd, SynthesisReverb* reverb, s16 updateIndex) {
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ReverbRingBufferItem* bufItem = &reverb->items[reverb->curFrame][updateIndex];
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cmd = AudioSynth_LoadRingBufferPart(cmd, DMEM_WET_TEMP, bufItem->startPos, bufItem->lengthA, reverb);
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if (bufItem->lengthB != 0) {
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// Ring buffer wrapped
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cmd = AudioSynth_LoadRingBufferPart(cmd, DMEM_WET_TEMP + bufItem->lengthA, 0, bufItem->lengthB, reverb);
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}
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return cmd;
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}
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Acmd* AudioSynth_SaveRingBuffer1AtTemp(Acmd* cmd, SynthesisReverb* reverb, s16 updateIndex) {
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ReverbRingBufferItem* bufItem = &reverb->items[reverb->curFrame][updateIndex];
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cmd = AudioSynth_SaveRingBufferPart(cmd, DMEM_WET_TEMP, bufItem->startPos, bufItem->lengthA, reverb);
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if (bufItem->lengthB != 0) {
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// Ring buffer wrapped
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cmd = AudioSynth_SaveRingBufferPart(cmd, DMEM_WET_TEMP + bufItem->lengthA, 0, bufItem->lengthB, reverb);
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}
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return cmd;
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}
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/**
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* Leak some audio from the left reverb channel into the right reverb channel and vice versa (pan)
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*/
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Acmd* AudioSynth_LeakReverb(Acmd* cmd, SynthesisReverb* reverb) {
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aDMEMMove(cmd++, DMEM_WET_LEFT_CH, DMEM_WET_SCRATCH, DEFAULT_LEN_1CH);
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aMix(cmd++, 0x1A, reverb->leakRtl, DMEM_WET_RIGHT_CH, DMEM_WET_LEFT_CH);
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aMix(cmd++, 0x1A, reverb->leakLtr, DMEM_WET_SCRATCH, DMEM_WET_RIGHT_CH);
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return cmd;
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}
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Acmd* func_800DB4E4(Acmd* cmd, s32 aiBufLen, SynthesisReverb* reverb, s16 updateIndex) {
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ReverbRingBufferItem* item = &reverb->items[reverb->curFrame][updateIndex];
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s16 offsetA;
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s16 offsetB;
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offsetA = (item->startPos & 7) * 2;
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offsetB = ALIGN16(offsetA + item->lengthA);
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cmd = AudioSynth_LoadRingBufferPart(cmd, DMEM_WET_TEMP, item->startPos - (offsetA / 2), DEFAULT_LEN_1CH, reverb);
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if (item->lengthB != 0) {
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// Ring buffer wrapped
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cmd = AudioSynth_LoadRingBufferPart(cmd, DMEM_WET_TEMP + offsetB, 0, DEFAULT_LEN_1CH - offsetB, reverb);
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}
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aSetBuffer(cmd++, 0, DMEM_WET_TEMP + offsetA, DMEM_WET_LEFT_CH, aiBufLen * 2);
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aResample(cmd++, reverb->resampleFlags, reverb->unk_0E, reverb->unk_30);
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aSetBuffer(cmd++, 0, DMEM_WET_TEMP + DEFAULT_LEN_1CH + offsetA, DMEM_WET_RIGHT_CH, aiBufLen * 2);
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aResample(cmd++, reverb->resampleFlags, reverb->unk_0E, reverb->unk_34);
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return cmd;
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}
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Acmd* func_800DB680(Acmd* cmd, SynthesisReverb* reverb, s16 updateIndex) {
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ReverbRingBufferItem* bufItem = &reverb->items[reverb->curFrame][updateIndex];
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aSetBuffer(cmd++, 0, DMEM_WET_LEFT_CH, DMEM_WET_SCRATCH, bufItem->unk_18 * 2);
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aResample(cmd++, reverb->resampleFlags, bufItem->unk_16, reverb->unk_38);
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cmd = AudioSynth_SaveBufferOffset(cmd, DMEM_WET_SCRATCH, bufItem->startPos, bufItem->lengthA, reverb->leftRingBuf);
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if (bufItem->lengthB != 0) {
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// Ring buffer wrapped
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cmd = AudioSynth_SaveBufferOffset(cmd, DMEM_WET_SCRATCH + bufItem->lengthA, 0, bufItem->lengthB,
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reverb->leftRingBuf);
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}
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aSetBuffer(cmd++, 0, DMEM_WET_RIGHT_CH, DMEM_WET_SCRATCH, bufItem->unk_18 * 2);
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aResample(cmd++, reverb->resampleFlags, bufItem->unk_16, reverb->unk_3C);
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cmd = AudioSynth_SaveBufferOffset(cmd, DMEM_WET_SCRATCH, bufItem->startPos, bufItem->lengthA, reverb->rightRingBuf);
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if (bufItem->lengthB != 0) {
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// Ring buffer wrapped
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cmd = AudioSynth_SaveBufferOffset(cmd, DMEM_WET_SCRATCH + bufItem->lengthA, 0, bufItem->lengthB,
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reverb->rightRingBuf);
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}
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return cmd;
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}
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Acmd* func_800DB828(Acmd* cmd, s32 aiBufLen, SynthesisReverb* reverb, s16 updateIndex) {
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ReverbRingBufferItem* item = &reverb->items[reverb->curFrame][updateIndex];
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s16 offsetA;
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s16 offsetB;
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item->unk_14 = (item->unk_18 << 0xF) / aiBufLen;
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offsetA = (item->startPos & 7) * 2;
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item->unk_16 = (aiBufLen << 0xF) / item->unk_18;
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offsetB = ALIGN16(offsetA + item->lengthA);
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cmd = AudioSynth_LoadRingBufferPart(cmd, DMEM_WET_TEMP, item->startPos - (offsetA / 2), DEFAULT_LEN_1CH, reverb);
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if (item->lengthB != 0) {
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// Ring buffer wrapped
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cmd = AudioSynth_LoadRingBufferPart(cmd, DMEM_WET_TEMP + offsetB, 0, DEFAULT_LEN_1CH - offsetB, reverb);
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}
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aSetBuffer(cmd++, 0, DMEM_WET_TEMP + offsetA, DMEM_WET_LEFT_CH, aiBufLen * 2);
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aResample(cmd++, reverb->resampleFlags, item->unk_14, reverb->unk_30);
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aSetBuffer(cmd++, 0, DMEM_WET_TEMP + DEFAULT_LEN_1CH + offsetA, DMEM_WET_RIGHT_CH, aiBufLen * 2);
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aResample(cmd++, reverb->resampleFlags, item->unk_14, reverb->unk_34);
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return cmd;
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}
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/**
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* Apply a filter (convolution) to each reverb channel.
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*/
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Acmd* AudioSynth_FilterReverb(Acmd* cmd, s32 count, SynthesisReverb* reverb) {
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if (reverb->filterLeft != NULL) {
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aFilter(cmd++, 2, count, reverb->filterLeft);
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aFilter(cmd++, reverb->resampleFlags, DMEM_WET_LEFT_CH, reverb->filterLeftState);
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}
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if (reverb->filterRight != NULL) {
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aFilter(cmd++, 2, count, reverb->filterRight);
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aFilter(cmd++, reverb->resampleFlags, DMEM_WET_RIGHT_CH, reverb->filterRightState);
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}
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return cmd;
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}
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Acmd* AudioSynth_MaybeMixRingBuffer1(Acmd* cmd, SynthesisReverb* reverb, s32 updateIndex) {
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SynthesisReverb* temp_a3;
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temp_a3 = &gAudioContext.synthesisReverbs[reverb->unk_05];
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if (temp_a3->downsampleRate == 1) {
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cmd = AudioSynth_LoadRingBuffer1AtTemp(cmd, temp_a3, updateIndex);
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aMix(cmd++, 0x34, reverb->unk_08, DMEM_WET_LEFT_CH, DMEM_WET_TEMP);
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cmd = AudioSynth_SaveRingBuffer1AtTemp(cmd, temp_a3, updateIndex);
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}
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return cmd;
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}
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void func_800DBB94(void) {
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}
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void AudioSynth_ClearBuffer(Acmd* cmd, s32 arg1, s32 arg2) {
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aClearBuffer(cmd, arg1, arg2);
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}
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void func_800DBBBC(void) {
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}
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void func_800DBBC4(void) {
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}
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void func_800DBBCC(void) {
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}
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void AudioSynth_Mix(Acmd* cmd, s32 arg1, s32 arg2, s32 arg3, s32 arg4) {
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aMix(cmd, arg1, arg2, arg3, arg4);
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}
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void func_800DBC08(void) {
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}
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void func_800DBC10(void) {
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}
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void func_800DBC18(void) {
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}
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void AudioSynth_SetBuffer(Acmd* cmd, s32 flags, s32 dmemIn, s32 dmemOut, s32 count) {
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aSetBuffer(cmd, flags, dmemIn, dmemOut, count);
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}
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void func_800DBC54(void) {
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}
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void func_800DBC5C(void) {
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}
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// possible fake match?
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void AudioSynth_DMemMove(Acmd* cmd, s32 dmemIn, s32 dmemOut, s32 count) {
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cmd->words.w0 = _SHIFTL(A_DMEMMOVE, 24, 8) | _SHIFTL(dmemIn, 0, 24);
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cmd->words.w1 = _SHIFTL(dmemOut, 16, 16) | _SHIFTL(count, 0, 16);
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}
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void func_800DBC90(void) {
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}
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void func_800DBC98(void) {
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}
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void func_800DBCA0(void) {
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}
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void func_800DBCA8(void) {
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}
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void AudioSynth_InterL(Acmd* cmd, s32 dmemIn, s32 dmemOut, s32 count) {
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cmd->words.w0 = _SHIFTL(A_INTERL, 24, 8) | _SHIFTL(count, 0, 16);
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cmd->words.w1 = _SHIFTL(dmemIn, 16, 16) | _SHIFTL(dmemOut, 0, 16);
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}
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void AudioSynth_EnvSetup1(Acmd* cmd, s32 arg1, s32 arg2, s32 arg3, s32 arg4) {
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aEnvSetup1(cmd, arg1, arg2, arg3, arg4);
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}
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void func_800DBD08(void) {
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}
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void AudioSynth_LoadBuffer(Acmd* cmd, s32 arg1, s32 arg2, s32 arg3) {
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aLoadBuffer(cmd, arg3, arg1, arg2);
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}
|
|
|
|
void AudioSynth_SaveBuffer(Acmd* cmd, s32 arg1, s32 arg2, s32 arg3) {
|
|
aSaveBuffer(cmd, arg1, arg3, arg2);
|
|
}
|
|
|
|
void AudioSynth_EnvSetup2(Acmd* cmd, s32 volLeft, s32 volRight) {
|
|
cmd->words.w0 = _SHIFTL(A_ENVSETUP2, 24, 8);
|
|
cmd->words.w1 = _SHIFTL(volLeft, 16, 16) | _SHIFTL(volRight, 0, 16);
|
|
}
|
|
|
|
void func_800DBD7C(void) {
|
|
}
|
|
|
|
void func_800DBD84(void) {
|
|
}
|
|
|
|
void func_800DBD8C(void) {
|
|
}
|
|
|
|
void AudioSynth_S8Dec(Acmd* cmd, s32 flags, s16* state) {
|
|
aS8Dec(cmd, flags, state);
|
|
}
|
|
|
|
void AudioSynth_HiLoGain(Acmd* cmd, s32 gain, s32 dmemIn, s32 dmemOut, s32 count) {
|
|
cmd->words.w0 = _SHIFTL(A_HILOGAIN, 24, 8) | _SHIFTL(gain, 16, 8) | _SHIFTL(count, 0, 16);
|
|
cmd->words.w1 = _SHIFTL(dmemIn, 16, 16) | _SHIFTL(dmemOut, 0, 16);
|
|
}
|
|
|
|
void AudioSynth_UnkCmd19(Acmd* cmd, s32 arg1, s32 arg2, s32 arg3, s32 arg4) {
|
|
cmd->words.w0 = _SHIFTL(A_UNK19, 24, 8) | _SHIFTL(arg4, 16, 8) | _SHIFTL(arg3, 0, 16);
|
|
cmd->words.w1 = _SHIFTL(arg1, 16, 16) | _SHIFTL(arg2, 0, 16);
|
|
}
|
|
|
|
void func_800DBE18(void) {
|
|
}
|
|
|
|
void func_800DBE20(void) {
|
|
}
|
|
|
|
void func_800DBE28(void) {
|
|
}
|
|
|
|
void func_800DBE30(void) {
|
|
}
|
|
|
|
void AudioSynth_UnkCmd3(Acmd* cmd, s32 arg1, s32 arg2, s32 arg3) {
|
|
cmd->words.w0 = _SHIFTL(A_UNK3, 24, 8) | _SHIFTL(arg3, 0, 16);
|
|
cmd->words.w1 = _SHIFTL(arg1, 16, 16) | _SHIFTL(arg2, 0, 16);
|
|
}
|
|
|
|
void func_800DBE5C(void) {
|
|
}
|
|
|
|
void func_800DBE64(void) {
|
|
}
|
|
|
|
void func_800DBE6C(void) {
|
|
}
|
|
|
|
void AudioSynth_LoadFilter(Acmd* cmd, s32 flags, s32 countOrBuf, s32 addr) {
|
|
aFilter(cmd, flags, countOrBuf, addr);
|
|
}
|
|
|
|
void AudioSynth_LoadFilterCount(Acmd* cmd, s32 count, s32 addr) {
|
|
aFilter(cmd, 2, count, addr);
|
|
}
|
|
|
|
Acmd* AudioSynth_LoadRingBuffer1(Acmd* cmd, s32 aiBufLen, SynthesisReverb* reverb, s16 updateIndex) {
|
|
ReverbRingBufferItem* ringBufferItem = &reverb->items[reverb->curFrame][updateIndex];
|
|
|
|
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 aiBufLen, SynthesisReverb* reverb, s16 updateIndex) {
|
|
ReverbRingBufferItem* bufItem = &reverb->items2[reverb->curFrame][updateIndex];
|
|
|
|
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 aiBufLen, SynthesisReverb* reverb, s16 updateIndex) {
|
|
if (reverb->downsampleRate == 1) {
|
|
cmd = AudioSynth_LoadRingBuffer2(cmd, aiBufLen, reverb, updateIndex);
|
|
}
|
|
|
|
return cmd;
|
|
}
|
|
|
|
Acmd* AudioSynth_LoadReverbSamples(Acmd* cmd, s32 aiBufLen, SynthesisReverb* reverb, s16 updateIndex) {
|
|
// Sets DMEM_WET_{LEFT,RIGHT}_CH, clobbers DMEM_TEMP
|
|
if (reverb->downsampleRate == 1) {
|
|
if (reverb->unk_18 != 0) {
|
|
cmd = func_800DB828(cmd, aiBufLen, reverb, updateIndex);
|
|
} else {
|
|
cmd = AudioSynth_LoadRingBuffer1(cmd, aiBufLen, reverb, updateIndex);
|
|
}
|
|
} else {
|
|
cmd = func_800DB4E4(cmd, aiBufLen, reverb, updateIndex);
|
|
}
|
|
return cmd;
|
|
}
|
|
|
|
Acmd* AudioSynth_SaveReverbSamples(Acmd* cmd, SynthesisReverb* reverb, s16 updateIndex) {
|
|
ReverbRingBufferItem* bufItem = &reverb->items[reverb->curFrame][updateIndex];
|
|
|
|
if (reverb->downsampleRate == 1) {
|
|
if (reverb->unk_18 != 0) {
|
|
cmd = func_800DB680(cmd, reverb, updateIndex);
|
|
} 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][updateIndex].toDownsampleLeft);
|
|
}
|
|
|
|
reverb->resampleFlags = 0;
|
|
return cmd;
|
|
}
|
|
|
|
Acmd* AudioSynth_SaveRingBuffer2(Acmd* cmd, SynthesisReverb* reverb, s16 updateIndex) {
|
|
ReverbRingBufferItem* bufItem = &reverb->items2[reverb->curFrame][updateIndex];
|
|
|
|
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;
|
|
|
|
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) {
|
|
|
|
// Loads reverb samples from RDRAM (ringBuffer) into DMEM (DMEM_WET_LEFT_CH)
|
|
cmd = AudioSynth_LoadReverbSamples(cmd, aiBufLen, reverb, updateIndex);
|
|
|
|
// Mixes reverb sample into the main dry channel
|
|
// reverb->volume is always set to 0x7FFF (audio spec), and DMEM_LEFT_CH is cleared before the loop.
|
|
// So for the first iteration, this is essentially a DMEMmove from DMEM_WET_LEFT_CH to DMEM_LEFT_CH
|
|
aMix(cmd++, DEFAULT_LEN_2CH >> 4, reverb->volume, 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);
|
|
}
|
|
|
|
// Decays reverb over time. The (+ 0x8000) here is -100%
|
|
aMix(cmd++, DEFAULT_LEN_2CH >> 4, reverb->decayRatio + 0x8000, DMEM_WET_LEFT_CH, DMEM_WET_LEFT_CH);
|
|
|
|
// Leak reverb between the left and right channels
|
|
if (reverb->leakRtl != 0 || reverb->leakLtr != 0) {
|
|
cmd = AudioSynth_LeakReverb(cmd, reverb);
|
|
}
|
|
|
|
if (unk14) {
|
|
// Saves the wet channel sample from DMEM (DMEM_WET_LEFT_CH) into RDRAM (ringBuffer) for future use
|
|
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);
|
|
}
|
|
|
|
// Saves the wet channel sample from DMEM (DMEM_WET_LEFT_CH) into RDRAM (ringBuffer) for future use
|
|
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;
|
|
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 gain;
|
|
s32 frameIndex;
|
|
s32 skipBytes;
|
|
s32 temp_v1_6;
|
|
void* buf;
|
|
s32 nSamplesToDecode;
|
|
u32 sampleAddr;
|
|
u32 samplesLenFixedPoint;
|
|
s32 samplesLenAdjusted;
|
|
s32 nSamplesProcessed;
|
|
s32 loopEndPos;
|
|
s32 nSamplesToProcess;
|
|
s32 phi_s4;
|
|
s32 nFirstFrameSamplesToIgnore;
|
|
s32 pad2[7];
|
|
s32 frameSize;
|
|
s32 nFramesToDecode;
|
|
s32 skipInitialSamples;
|
|
s32 sampleDataStart;
|
|
u8* sampleData;
|
|
s32 nParts;
|
|
s32 curPart;
|
|
s32 sampleDataStartPad;
|
|
s32 side;
|
|
s32 resampledTempLen;
|
|
u16 noteSamplesDmemAddrBeforeResampling;
|
|
s32 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->startSamplePos;
|
|
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;
|
|
}
|
|
if (1) {}
|
|
if (1) {}
|
|
if (1) {}
|
|
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);
|
|
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 = (u32)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) {
|
|
if (1) {}
|
|
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,
|
|
ALIGN8(samplesLenAdjusted / 2));
|
|
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, ALIGN8(samplesLenAdjusted / 2));
|
|
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);
|
|
}
|
|
|
|
gain = noteSubEu->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) {
|
|
gain = 0x10;
|
|
}
|
|
AudioSynth_HiLoGain(cmd++, gain, 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, 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, 0x7FFF);
|
|
return cmd;
|
|
}
|