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1568 lines
57 KiB
C
1568 lines
57 KiB
C
#ifndef VERSION_SH
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#include <ultra64.h>
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#include "../../debug_print.h"
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#include "synthesis.h"
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#include "heap.h"
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#include "data.h"
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#include "load.h"
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#include "seqplayer.h"
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#include "internal.h"
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#include "external.h"
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#ifndef TARGET_N64
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#include "../pc/mixer.h"
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#endif
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#define DMEM_ADDR_TEMP 0x0
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#define DMEM_ADDR_RESAMPLED 0x20
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#define DMEM_ADDR_RESAMPLED2 0x160
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#define DMEM_ADDR_UNCOMPRESSED_NOTE 0x180
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#define DMEM_ADDR_NOTE_PAN_TEMP 0x200
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#define DMEM_ADDR_STEREO_STRONG_TEMP_DRY 0x200
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#define DMEM_ADDR_STEREO_STRONG_TEMP_WET 0x340
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#define DMEM_ADDR_COMPRESSED_ADPCM_DATA 0x3f0
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#define DMEM_ADDR_LEFT_CH 0x4c0
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#define DMEM_ADDR_RIGHT_CH 0x600
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#define DMEM_ADDR_WET_LEFT_CH 0x740
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#define DMEM_ADDR_WET_RIGHT_CH 0x880
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#define aSetLoadBufferPair(pkt, c, off) \
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aSetBuffer(pkt, 0, c + DMEM_ADDR_WET_LEFT_CH, 0, DEFAULT_LEN_1CH - c); \
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aLoadBuffer(pkt, VIRTUAL_TO_PHYSICAL2(gSynthesisReverb.ringBuffer.left + (off))); \
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aSetBuffer(pkt, 0, c + DMEM_ADDR_WET_RIGHT_CH, 0, DEFAULT_LEN_1CH - c); \
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aLoadBuffer(pkt, VIRTUAL_TO_PHYSICAL2(gSynthesisReverb.ringBuffer.right + (off)))
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#define aSetSaveBufferPair(pkt, c, d, off) \
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aSetBuffer(pkt, 0, 0, c + DMEM_ADDR_WET_LEFT_CH, d); \
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aSaveBuffer(pkt, VIRTUAL_TO_PHYSICAL2(gSynthesisReverb.ringBuffer.left + (off))); \
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aSetBuffer(pkt, 0, 0, c + DMEM_ADDR_WET_RIGHT_CH, d); \
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aSaveBuffer(pkt, VIRTUAL_TO_PHYSICAL2(gSynthesisReverb.ringBuffer.right + (off)));
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#define ALIGN(val, amnt) (((val) + (1 << amnt) - 1) & ~((1 << amnt) - 1))
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struct VolumeChange
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{
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u16 sourceLeft;
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u16 sourceRight;
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u16 targetLeft;
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u16 targetRight;
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};
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u64 *synthesis_do_one_audio_update(s16 *aiBuf, s32 bufLen, u64 *cmd, s32 updateIndex);
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#ifdef VERSION_EU
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u64 *synthesis_process_note(struct Note *note, struct NoteSubEu *noteSubEu, struct NoteSynthesisState *synthesisState, s16 *aiBuf, s32 bufLen, u64 *cmd);
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u64 *load_wave_samples(u64 *cmd, struct NoteSubEu *noteSubEu, struct NoteSynthesisState *synthesisState, s32 nSamplesToLoad);
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u64 *final_resample(u64 *cmd, struct NoteSynthesisState *synthesisState, s32 count, u16 pitch, u16 dmemIn, u32 flags);
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u64 *process_envelope(u64 *cmd, struct NoteSubEu *noteSubEu, struct NoteSynthesisState *synthesisState, s32 nSamples, u16 inBuf, s32 headsetPanSettings, u32 flags);
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u64 *note_apply_headset_pan_effects(u64 *cmd, struct NoteSubEu *noteSubEu, struct NoteSynthesisState *note, s32 bufLen, s32 flags, s32 leftRight);
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#else
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u64 *synthesis_process_notes(s16 *aiBuf, s32 bufLen, u64 *cmd);
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u64 *load_wave_samples(u64 *cmd, struct Note *note, s32 nSamplesToLoad);
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u64 *final_resample(u64 *cmd, struct Note *note, s32 count, u16 pitch, u16 dmemIn, u32 flags);
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u64 *process_envelope(
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u64 *cmd, struct Note *note, s32 nSamples, u16 inBuf, s32 headsetPanSettings,
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u32 flags);
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u64 *process_envelope_inner(
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u64 *cmd, struct Note *note, s32 nSamples, u16 inBuf,
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s32 headsetPanSettings, struct VolumeChange *vol);
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u64 *note_apply_headset_pan_effects(u64 *cmd, struct Note *note, s32 bufLen, s32 flags, s32 leftRight);
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#endif
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#ifdef VERSION_EU
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struct SynthesisReverb gSynthesisReverbs[4];
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u8 sAudioSynthesisPad[0x10];
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#else
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struct SynthesisReverb gSynthesisReverb;
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u8 sAudioSynthesisPad[0x20];
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#endif
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#ifdef VERSION_EU
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s16 gVolume;
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s8 gUseReverb;
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s8 gNumSynthesisReverbs;
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struct NoteSubEu *gNoteSubsEu;
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#endif
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#ifdef VERSION_EU
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f32 gLeftVolRampings[3][1024];
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f32 gRightVolRampings[3][1024];
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f32 *gCurrentLeftVolRamping; // Points to any of the three left buffers above
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f32 *gCurrentRightVolRamping; // Points to any of the three right buffers above
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u8 audioString1[] = "pitch %x: delaybytes %d : olddelay %d\n";
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u8 audioString2[] = "cont %x: delaybytes %d : olddelay %d\n";
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#endif
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#ifdef VERSION_EU
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// Equivalent functionality as the US/JP version,
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// just that the reverb structure is chosen from an array with index
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void prepare_reverb_ring_buffer(s32 chunkLen, u32 updateIndex, s32 reverbIndex)
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{
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struct ReverbRingBufferItem *item;
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struct SynthesisReverb *reverb = &gSynthesisReverbs[reverbIndex];
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s32 srcPos;
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s32 dstPos;
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s32 nSamples;
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s32 excessiveSamples;
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s32 UNUSED pad[3];
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if (reverb->downsampleRate != 1)
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{
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if (reverb->framesLeftToIgnore == 0)
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{
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// Now that the RSP has finished, downsample the samples produced two frames ago by skipping
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// samples.
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item = &reverb->items[reverb->curFrame][updateIndex];
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// Touches both left and right since they are adjacent in memory
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osInvalDCache(item->toDownsampleLeft, DEFAULT_LEN_2CH);
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for (srcPos = 0, dstPos = 0; dstPos < item->lengthA / 2;
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srcPos += reverb->downsampleRate, dstPos++)
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{
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reverb->ringBuffer.left[item->startPos + dstPos] =
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item->toDownsampleLeft[srcPos];
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reverb->ringBuffer.right[item->startPos + dstPos] =
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item->toDownsampleRight[srcPos];
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}
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for (dstPos = 0; dstPos < item->lengthB / 2; srcPos += reverb->downsampleRate, dstPos++)
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{
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reverb->ringBuffer.left[dstPos] = item->toDownsampleLeft[srcPos];
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reverb->ringBuffer.right[dstPos] = item->toDownsampleRight[srcPos];
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}
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}
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}
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item = &reverb->items[reverb->curFrame][updateIndex];
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nSamples = chunkLen / reverb->downsampleRate;
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excessiveSamples = (nSamples + reverb->nextRingBufferPos) - reverb->bufSizePerChannel;
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if (excessiveSamples < 0)
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{
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// There is space in the ring buffer before it wraps around
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item->lengthA = nSamples * 2;
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item->lengthB = 0;
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item->startPos = (s32)reverb->nextRingBufferPos;
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reverb->nextRingBufferPos += nSamples;
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}
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else
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{
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// Ring buffer wrapped around
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item->lengthA = (nSamples - excessiveSamples) * 2;
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item->lengthB = excessiveSamples * 2;
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item->startPos = reverb->nextRingBufferPos;
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reverb->nextRingBufferPos = excessiveSamples;
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}
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// These fields are never read later
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item->numSamplesAfterDownsampling = nSamples;
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item->chunkLen = chunkLen;
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}
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#else
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void prepare_reverb_ring_buffer(s32 chunkLen, u32 updateIndex)
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{
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struct ReverbRingBufferItem *item;
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s32 srcPos;
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s32 dstPos;
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if (gReverbDownsampleRate != 1)
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{
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if (gSynthesisReverb.framesLeftToIgnore == 0)
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{
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// Now that the RSP has finished, downsample the samples produced two frames ago by skipping
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// samples.
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item = &gSynthesisReverb.items[gSynthesisReverb.curFrame][updateIndex];
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// Touches both left and right since they are adjacent in memory
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osInvalDCache(item->toDownsampleLeft, DEFAULT_LEN_2CH);
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for (srcPos = 0, dstPos = 0; dstPos < item->lengthA / 2;
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srcPos += gReverbDownsampleRate, dstPos++)
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{
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gSynthesisReverb.ringBuffer.left[dstPos + item->startPos] =
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item->toDownsampleLeft[srcPos];
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gSynthesisReverb.ringBuffer.right[dstPos + item->startPos] =
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item->toDownsampleRight[srcPos];
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}
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for (dstPos = 0; dstPos < item->lengthB / 2; srcPos += gReverbDownsampleRate, dstPos++)
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{
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gSynthesisReverb.ringBuffer.left[dstPos] = item->toDownsampleLeft[srcPos];
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gSynthesisReverb.ringBuffer.right[dstPos] = item->toDownsampleRight[srcPos];
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}
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}
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}
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item = &gSynthesisReverb.items[gSynthesisReverb.curFrame][updateIndex];
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s32 numSamplesAfterDownsampling = chunkLen / gReverbDownsampleRate;
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if (numSamplesAfterDownsampling + gSynthesisReverb.nextRingBufferPos - gSynthesisReverb.bufSizePerChannel < 0)
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{
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// There is space in the ring buffer before it wraps around
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item->lengthA = numSamplesAfterDownsampling * 2;
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item->lengthB = 0;
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item->startPos = gSynthesisReverb.nextRingBufferPos;
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gSynthesisReverb.nextRingBufferPos += numSamplesAfterDownsampling;
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}
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else
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{
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// Ring buffer wrapped around
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s32 excessiveSamples = numSamplesAfterDownsampling + gSynthesisReverb.nextRingBufferPos - gSynthesisReverb.bufSizePerChannel;
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s32 nSamples = numSamplesAfterDownsampling - excessiveSamples;
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item->lengthA = nSamples * 2;
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item->lengthB = excessiveSamples * 2;
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item->startPos = gSynthesisReverb.nextRingBufferPos;
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gSynthesisReverb.nextRingBufferPos = excessiveSamples;
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}
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// These fields are never read later
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item->numSamplesAfterDownsampling = numSamplesAfterDownsampling;
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item->chunkLen = chunkLen;
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}
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#endif
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#ifdef VERSION_EU
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u64 *synthesis_load_reverb_ring_buffer(u64 *cmd, u16 addr, u16 srcOffset, s32 len, s32 reverbIndex)
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{
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aSetBuffer(cmd++, 0, addr, 0, len);
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aLoadBuffer(cmd++, VIRTUAL_TO_PHYSICAL2(&gSynthesisReverbs[reverbIndex].ringBuffer.left[srcOffset]));
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aSetBuffer(cmd++, 0, addr + DEFAULT_LEN_1CH, 0, len);
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aLoadBuffer(cmd++, VIRTUAL_TO_PHYSICAL2(&gSynthesisReverbs[reverbIndex].ringBuffer.right[srcOffset]));
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return cmd;
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}
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#endif
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#ifdef VERSION_EU
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u64 *synthesis_save_reverb_ring_buffer(u64 *cmd, u16 addr, u16 destOffset, s32 len, s32 reverbIndex)
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{
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aSetBuffer(cmd++, 0, 0, addr, len);
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aSaveBuffer(cmd++, VIRTUAL_TO_PHYSICAL2(&gSynthesisReverbs[reverbIndex].ringBuffer.left[destOffset]));
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aSetBuffer(cmd++, 0, 0, addr + DEFAULT_LEN_1CH, len);
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aSaveBuffer(cmd++, VIRTUAL_TO_PHYSICAL2(&gSynthesisReverbs[reverbIndex].ringBuffer.right[destOffset]));
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return cmd;
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}
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#endif
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#ifdef VERSION_EU
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void synthesis_load_note_subs_eu(s32 updateIndex)
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{
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struct NoteSubEu *src;
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struct NoteSubEu *dest;
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s32 i;
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for (i = 0; i < gMaxSimultaneousNotes; i++)
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{
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src = &gNotes[i].noteSubEu;
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dest = &gNoteSubsEu[gMaxSimultaneousNotes * updateIndex + i];
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if (src->enabled)
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{
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*dest = *src;
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src->needsInit = FALSE;
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}
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else
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{
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dest->enabled = FALSE;
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}
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}
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}
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#endif
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#ifndef VERSION_EU
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s32 get_volume_ramping(u16 sourceVol, u16 targetVol, s32 arg2)
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{
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// This roughly computes 2^16 * (targetVol / sourceVol) ^ (8 / arg2),
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// but with discretizations of targetVol, sourceVol and arg2.
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f32 ret;
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switch (arg2)
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{
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default:
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ret = gVolRampingLhs136[targetVol >> 8] * gVolRampingRhs136[sourceVol >> 8];
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break;
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case 128:
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ret = gVolRampingLhs128[targetVol >> 8] * gVolRampingRhs128[sourceVol >> 8];
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break;
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case 136:
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ret = gVolRampingLhs136[targetVol >> 8] * gVolRampingRhs136[sourceVol >> 8];
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break;
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case 144:
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ret = gVolRampingLhs144[targetVol >> 8] * gVolRampingRhs144[sourceVol >> 8];
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break;
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}
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return ret;
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}
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#endif
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#ifdef VERSION_EU
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// TODO: (Scrub C) pointless mask and whitespace
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u64 *synthesis_execute(u64 *cmdBuf, s32 *writtenCmds, s16 *aiBuf, s32 bufLen)
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{
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s32 i, j;
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f32 *leftVolRamp;
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f32 *rightVolRamp;
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u32 *aiBufPtr;
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u64 *cmd = cmdBuf;
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s32 chunkLen;
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s32 nextVolRampTable;
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for (i = gAudioBufferParameters.updatesPerFrame; i > 0; i--)
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{
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process_sequences(i - 1);
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synthesis_load_note_subs_eu(gAudioBufferParameters.updatesPerFrame - i);
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}
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aSegment(cmd++, 0, 0);
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aiBufPtr = (u32 *)aiBuf;
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for (i = gAudioBufferParameters.updatesPerFrame; i > 0; i--)
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{
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if (i == 1) {
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#pragma GCC diagnostic push
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#if defined(__clang__)
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#pragma GCC diagnostic ignored "-Wself-assign"
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#endif
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// self-assignment has no affect when added here, could possibly simplify a macro definition
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chunkLen= bufLen; nextVolRampTable= nextVolRampTable; leftVolRamp= gLeftVolRampings[nextVolRampTable]; rightVolRamp= gRightVolRampings[nextVolRampTable & 0xFFFFFFFF];
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#pragma GCC diagnostic pop
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} else {
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if (bufLen / i >= gAudioBufferParameters.samplesPerUpdateMax) {
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chunkLen = gAudioBufferParameters.samplesPerUpdateMax; nextVolRampTable = 2; leftVolRamp = gLeftVolRampings[2]; rightVolRamp = gRightVolRampings[2];
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} else if (bufLen / i <= gAudioBufferParameters.samplesPerUpdateMin) {
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chunkLen = gAudioBufferParameters.samplesPerUpdateMin; nextVolRampTable = 0; leftVolRamp = gLeftVolRampings[0]; rightVolRamp = gRightVolRampings[0];
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} else {
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chunkLen = gAudioBufferParameters.samplesPerUpdate; nextVolRampTable = 1; leftVolRamp = gLeftVolRampings[1]; rightVolRamp = gRightVolRampings[1];
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}
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}
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gCurrentLeftVolRamping= leftVolRamp;
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gCurrentRightVolRamping= rightVolRamp;
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for (j =0; j<gNumSynthesisReverbs; j++) {
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if (gSynthesisReverbs[j].useReverb != 0) {
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prepare_reverb_ring_buffer(chunkLen, gAudioBufferParameters.updatesPerFrame - i, j);
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}
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}
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cmd= synthesis_do_one_audio_update((s16 *) aiBufPtr, chunkLen, cmd, gAudioBufferParameters.updatesPerFrame- i);
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bufLen-= chunkLen;
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aiBufPtr+= chunkLen;
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}
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for (j =0; j<gNumSynthesisReverbs; j++) {
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if (gSynthesisReverbs[j].framesLeftToIgnore != 0) {
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gSynthesisReverbs[j].framesLeftToIgnore--;
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}
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gSynthesisReverbs[j].curFrame ^= 1;
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}
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*writtenCmds= cmd- cmdBuf;
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return cmd;
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}
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#else
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// bufLen will be divisible by 16
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u64 *synthesis_execute(u64 *cmdBuf, s32 *writtenCmds, s16 *aiBuf, s32 bufLen)
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{
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s32 chunkLen;
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u32 *aiBufPtr = (u32 *)aiBuf;
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u64 *cmd = cmdBuf + 1;
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aSegment(cmdBuf, 0, 0);
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for (s32 i = gAudioUpdatesPerFrame; i > 0; i--)
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{
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if (i == 1)
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{
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// 'bufLen' will automatically be divisible by 8, no need to round
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chunkLen = bufLen;
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}
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else
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{
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s32 v0 = bufLen / i;
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// chunkLen = v0 rounded to nearest multiple of 8
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chunkLen = v0 - (v0 & 7);
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if ((v0 & 7) >= 4)
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{
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chunkLen += 8;
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}
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}
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process_sequences(i - 1);
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if (gSynthesisReverb.useReverb != 0)
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{
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prepare_reverb_ring_buffer(chunkLen, gAudioUpdatesPerFrame - i);
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}
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cmd = synthesis_do_one_audio_update((s16 *)aiBufPtr, chunkLen, cmd, gAudioUpdatesPerFrame - i);
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bufLen -= chunkLen;
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aiBufPtr += chunkLen;
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}
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if (gSynthesisReverb.framesLeftToIgnore != 0)
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{
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gSynthesisReverb.framesLeftToIgnore--;
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}
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gSynthesisReverb.curFrame ^= 1;
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*writtenCmds = cmd - cmdBuf;
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return cmd;
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}
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#endif
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#ifdef VERSION_EU
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u64 *synthesis_resample_and_mix_reverb(u64 *cmd, s32 bufLen, s16 reverbIndex, s16 updateIndex)
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{
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struct ReverbRingBufferItem *item;
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s16 startPad;
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s16 paddedLengthA;
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item = &gSynthesisReverbs[reverbIndex].items[gSynthesisReverbs[reverbIndex].curFrame][updateIndex];
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aClearBuffer(cmd++, DMEM_ADDR_WET_LEFT_CH, DEFAULT_LEN_2CH);
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if (gSynthesisReverbs[reverbIndex].downsampleRate == 1)
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{
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cmd = synthesis_load_reverb_ring_buffer(cmd, DMEM_ADDR_WET_LEFT_CH, item->startPos, item->lengthA, reverbIndex);
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if (item->lengthB != 0)
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{
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cmd = synthesis_load_reverb_ring_buffer(cmd, DMEM_ADDR_WET_LEFT_CH + item->lengthA, 0, item->lengthB, reverbIndex);
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}
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aSetBuffer(cmd++, 0, 0, 0, DEFAULT_LEN_2CH);
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aMix(cmd++, 0, 0x7fff, DMEM_ADDR_WET_LEFT_CH, DMEM_ADDR_LEFT_CH);
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aMix(cmd++, 0, 0x8000 + gSynthesisReverbs[reverbIndex].reverbGain, DMEM_ADDR_WET_LEFT_CH, DMEM_ADDR_WET_LEFT_CH);
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}
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else
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{
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startPad = (item->startPos % 8u) * 2;
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paddedLengthA = ALIGN(startPad + item->lengthA, 4);
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|
|
cmd = synthesis_load_reverb_ring_buffer(cmd, DMEM_ADDR_RESAMPLED, (item->startPos - startPad / 2), DEFAULT_LEN_1CH, reverbIndex);
|
|
if (item->lengthB != 0)
|
|
{
|
|
cmd = synthesis_load_reverb_ring_buffer(cmd, DMEM_ADDR_RESAMPLED + paddedLengthA, 0, DEFAULT_LEN_1CH - paddedLengthA, reverbIndex);
|
|
}
|
|
|
|
aSetBuffer(cmd++, 0, DMEM_ADDR_RESAMPLED + startPad, DMEM_ADDR_WET_LEFT_CH, bufLen * 2);
|
|
aResample(cmd++, gSynthesisReverbs[reverbIndex].resampleFlags, gSynthesisReverbs[reverbIndex].resampleRate, VIRTUAL_TO_PHYSICAL2(gSynthesisReverbs[reverbIndex].resampleStateLeft));
|
|
|
|
aSetBuffer(cmd++, 0, DMEM_ADDR_RESAMPLED2 + startPad, DMEM_ADDR_WET_RIGHT_CH, bufLen * 2);
|
|
aResample(cmd++, gSynthesisReverbs[reverbIndex].resampleFlags, gSynthesisReverbs[reverbIndex].resampleRate, VIRTUAL_TO_PHYSICAL2(gSynthesisReverbs[reverbIndex].resampleStateRight));
|
|
|
|
aSetBuffer(cmd++, 0, 0, 0, DEFAULT_LEN_2CH);
|
|
aMix(cmd++, 0, 0x7fff, DMEM_ADDR_WET_LEFT_CH, DMEM_ADDR_LEFT_CH);
|
|
aMix(cmd++, 0, 0x8000 + gSynthesisReverbs[reverbIndex].reverbGain, DMEM_ADDR_WET_LEFT_CH, DMEM_ADDR_WET_LEFT_CH);
|
|
}
|
|
return cmd;
|
|
}
|
|
|
|
u64 *synthesis_save_reverb_samples(u64 *cmd, s16 reverbIndex, s16 updateIndex)
|
|
{
|
|
struct ReverbRingBufferItem *item;
|
|
|
|
item = &gSynthesisReverbs[reverbIndex].items[gSynthesisReverbs[reverbIndex].curFrame][updateIndex];
|
|
if (gSynthesisReverbs[reverbIndex].useReverb != 0)
|
|
{
|
|
switch (gSynthesisReverbs[reverbIndex].downsampleRate)
|
|
{
|
|
case 1:
|
|
// Put the oldest samples in the ring buffer into the wet channels
|
|
cmd = synthesis_save_reverb_ring_buffer(cmd, DMEM_ADDR_WET_LEFT_CH, item->startPos, item->lengthA, reverbIndex);
|
|
if (item->lengthB != 0)
|
|
{
|
|
// Ring buffer wrapped
|
|
cmd = synthesis_save_reverb_ring_buffer(cmd, DMEM_ADDR_WET_LEFT_CH + item->lengthA, 0, item->lengthB, reverbIndex);
|
|
}
|
|
break;
|
|
|
|
default:
|
|
// 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.
|
|
aSetBuffer(cmd++, 0, 0, DMEM_ADDR_WET_LEFT_CH, DEFAULT_LEN_2CH);
|
|
aSaveBuffer(cmd++, VIRTUAL_TO_PHYSICAL2(gSynthesisReverbs[reverbIndex].items[gSynthesisReverbs[reverbIndex].curFrame][updateIndex].toDownsampleLeft));
|
|
gSynthesisReverbs[reverbIndex].resampleFlags = 0;
|
|
break;
|
|
}
|
|
}
|
|
return cmd;
|
|
}
|
|
#endif
|
|
|
|
#ifdef VERSION_EU
|
|
u64 *synthesis_do_one_audio_update(s16 *aiBuf, s32 bufLen, u64 *cmd, s32 updateIndex)
|
|
{
|
|
struct NoteSubEu *noteSubEu;
|
|
u8 noteIndices[56];
|
|
s32 temp;
|
|
s32 i;
|
|
s16 j;
|
|
s16 notePos = 0;
|
|
|
|
if (gNumSynthesisReverbs == 0)
|
|
{
|
|
for (i = 0; i < gMaxSimultaneousNotes; i++)
|
|
{
|
|
if (gNoteSubsEu[gMaxSimultaneousNotes * updateIndex + i].enabled)
|
|
{
|
|
noteIndices[notePos++] = i;
|
|
}
|
|
}
|
|
}
|
|
else
|
|
{
|
|
for (j = 0; j < gNumSynthesisReverbs; j++)
|
|
{
|
|
for (i = 0; i < gMaxSimultaneousNotes; i++)
|
|
{
|
|
noteSubEu = &gNoteSubsEu[gMaxSimultaneousNotes * updateIndex + i];
|
|
if (noteSubEu->enabled && j == noteSubEu->reverbIndex)
|
|
{
|
|
noteIndices[notePos++] = i;
|
|
}
|
|
}
|
|
}
|
|
|
|
for (i = 0; i < gMaxSimultaneousNotes; i++)
|
|
{
|
|
noteSubEu = &gNoteSubsEu[gMaxSimultaneousNotes * updateIndex + i];
|
|
if (noteSubEu->enabled && noteSubEu->reverbIndex >= gNumSynthesisReverbs)
|
|
{
|
|
noteIndices[notePos++] = i;
|
|
}
|
|
}
|
|
}
|
|
aClearBuffer(cmd++, DMEM_ADDR_LEFT_CH, DEFAULT_LEN_2CH);
|
|
i = 0;
|
|
for (j = 0; j < gNumSynthesisReverbs; j++)
|
|
{
|
|
gUseReverb = gSynthesisReverbs[j].useReverb;
|
|
if (gUseReverb != 0)
|
|
{
|
|
cmd = synthesis_resample_and_mix_reverb(cmd, bufLen, j, updateIndex);
|
|
}
|
|
for (; i < notePos; i++)
|
|
{
|
|
temp = updateIndex * gMaxSimultaneousNotes;
|
|
if (j == gNoteSubsEu[temp + noteIndices[i]].reverbIndex)
|
|
{
|
|
cmd = synthesis_process_note(&gNotes[noteIndices[i]],
|
|
&gNoteSubsEu[temp + noteIndices[i]],
|
|
&gNotes[noteIndices[i]].synthesisState,
|
|
aiBuf, bufLen, cmd);
|
|
continue;
|
|
}
|
|
else
|
|
{
|
|
break;
|
|
}
|
|
}
|
|
if (gSynthesisReverbs[j].useReverb != 0)
|
|
{
|
|
cmd = synthesis_save_reverb_samples(cmd, j, updateIndex);
|
|
}
|
|
}
|
|
for (; i < notePos; i++)
|
|
{
|
|
temp = updateIndex * gMaxSimultaneousNotes;
|
|
if (IS_BANK_LOAD_COMPLETE(gNoteSubsEu[temp + noteIndices[i]].bankId) == TRUE)
|
|
{
|
|
cmd = synthesis_process_note(&gNotes[noteIndices[i]],
|
|
&gNoteSubsEu[temp + noteIndices[i]],
|
|
&gNotes[noteIndices[i]].synthesisState,
|
|
aiBuf, bufLen, cmd);
|
|
}
|
|
else
|
|
{
|
|
gAudioErrorFlags = (gNoteSubsEu[temp + noteIndices[i]].bankId + (i << 8)) + 0x10000000;
|
|
}
|
|
}
|
|
|
|
temp = bufLen * 2;
|
|
aSetBuffer(cmd++, 0, 0, DMEM_ADDR_TEMP, temp);
|
|
aInterleave(cmd++, DMEM_ADDR_LEFT_CH, DMEM_ADDR_RIGHT_CH);
|
|
aSetBuffer(cmd++, 0, 0, DMEM_ADDR_TEMP, temp * 2);
|
|
aSaveBuffer(cmd++, VIRTUAL_TO_PHYSICAL2(aiBuf));
|
|
return cmd;
|
|
}
|
|
#else
|
|
u64 *synthesis_do_one_audio_update(s16 *aiBuf, s32 bufLen, u64 *cmd, s32 updateIndex)
|
|
{
|
|
|
|
UNUSED s32 pad1[1];
|
|
UNUSED s32 pad[2];
|
|
UNUSED s32 pad2[1];
|
|
s16 temp;
|
|
|
|
|
|
struct ReverbRingBufferItem *v1 = &gSynthesisReverb.items[gSynthesisReverb.curFrame][updateIndex];
|
|
|
|
if (gSynthesisReverb.useReverb == 0)
|
|
{
|
|
aClearBuffer(cmd++, DMEM_ADDR_LEFT_CH, DEFAULT_LEN_2CH);
|
|
cmd = synthesis_process_notes(aiBuf, bufLen, cmd);
|
|
}
|
|
else
|
|
{
|
|
if (gReverbDownsampleRate == 1)
|
|
{
|
|
|
|
// Put the oldest samples in the ring buffer into the wet channels
|
|
aSetLoadBufferPair(cmd++, 0, v1->startPos);
|
|
if (v1->lengthB != 0)
|
|
{
|
|
// Ring buffer wrapped
|
|
aSetLoadBufferPair(cmd++, v1->lengthA, 0);
|
|
temp = 0;
|
|
}
|
|
|
|
// Use the reverb sound as initial sound for this audio update
|
|
aDMEMMove(cmd++, DMEM_ADDR_WET_LEFT_CH, DMEM_ADDR_LEFT_CH, DEFAULT_LEN_2CH);
|
|
|
|
// (Hopefully) lower the volume of the wet channels. New reverb will later be mixed into
|
|
// these channels.
|
|
aSetBuffer(cmd++, 0, 0, 0, DEFAULT_LEN_2CH);
|
|
// 0x8000 here is -100%
|
|
aMix(cmd++, 0, /*gain*/ 0x8000 + gSynthesisReverb.reverbGain, /*in*/ DMEM_ADDR_WET_LEFT_CH,
|
|
/*out*/ DMEM_ADDR_WET_LEFT_CH);
|
|
}
|
|
else
|
|
{
|
|
|
|
// Same as above but upsample the previously downsampled samples used for reverb first
|
|
temp = 0; //! jesus christ
|
|
s16 t4 = (v1->startPos & 7) * 2;
|
|
s16 ra = ALIGN(v1->lengthA + t4, 4);
|
|
aSetLoadBufferPair(cmd++, 0, v1->startPos - t4 / 2);
|
|
if (v1->lengthB != 0)
|
|
{
|
|
// Ring buffer wrapped
|
|
aSetLoadBufferPair(cmd++, ra, 0);
|
|
//! We need an empty statement (even an empty ';') here to make the function match (because IDO).
|
|
//! However, copt removes extraneous statements and dead code. So we need to trick copt
|
|
//! into thinking 'temp' could be undefined, and luckily the compiler optimizes out the
|
|
//! useless assignment.
|
|
ra = ra + temp;
|
|
}
|
|
aSetBuffer(cmd++, 0, t4 + DMEM_ADDR_WET_LEFT_CH, DMEM_ADDR_LEFT_CH, bufLen << 1);
|
|
aResample(cmd++, gSynthesisReverb.resampleFlags, gSynthesisReverb.resampleRate, VIRTUAL_TO_PHYSICAL2(gSynthesisReverb.resampleStateLeft));
|
|
aSetBuffer(cmd++, 0, t4 + DMEM_ADDR_WET_RIGHT_CH, DMEM_ADDR_RIGHT_CH, bufLen << 1);
|
|
aResample(cmd++, gSynthesisReverb.resampleFlags, gSynthesisReverb.resampleRate, VIRTUAL_TO_PHYSICAL2(gSynthesisReverb.resampleStateRight));
|
|
aSetBuffer(cmd++, 0, 0, 0, DEFAULT_LEN_2CH);
|
|
aMix(cmd++, 0, /*gain*/ 0x8000 + gSynthesisReverb.reverbGain, /*in*/ DMEM_ADDR_LEFT_CH, /*out*/ DMEM_ADDR_LEFT_CH);
|
|
aDMEMMove(cmd++, DMEM_ADDR_LEFT_CH, DMEM_ADDR_WET_LEFT_CH, DEFAULT_LEN_2CH);
|
|
}
|
|
cmd = synthesis_process_notes(aiBuf, bufLen, cmd);
|
|
if (gReverbDownsampleRate == 1)
|
|
{
|
|
aSetSaveBufferPair(cmd++, 0, v1->lengthA, v1->startPos);
|
|
if (v1->lengthB != 0)
|
|
{
|
|
// Ring buffer wrapped
|
|
aSetSaveBufferPair(cmd++, v1->lengthA, v1->lengthB, 0);
|
|
}
|
|
}
|
|
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.
|
|
aSetBuffer(cmd++, 0, 0, DMEM_ADDR_WET_LEFT_CH, DEFAULT_LEN_2CH);
|
|
aSaveBuffer(cmd++, VIRTUAL_TO_PHYSICAL2(gSynthesisReverb.items[gSynthesisReverb.curFrame][updateIndex].toDownsampleLeft));
|
|
gSynthesisReverb.resampleFlags = 0;
|
|
}
|
|
}
|
|
return cmd;
|
|
}
|
|
#endif
|
|
|
|
#ifdef VERSION_EU
|
|
// Processes just one note, not all
|
|
u64 *synthesis_process_note(struct Note *note, struct NoteSubEu *noteSubEu, struct NoteSynthesisState *synthesisState, UNUSED s16 *aiBuf, s32 bufLen, u64 *cmd)
|
|
{
|
|
UNUSED s32 pad0[3];
|
|
#else
|
|
u64 *synthesis_process_notes(s16 *aiBuf, s32 bufLen, u64 *cmd)
|
|
{
|
|
|
|
s32 noteIndex; // sp174
|
|
struct Note *note; // s7
|
|
UNUSED u8 pad0[0x08];
|
|
#endif
|
|
struct AudioBankSample *audioBookSample; // sp164, sp138
|
|
struct AdpcmLoop *loopInfo; // sp160, sp134
|
|
s16 *curLoadedBook = NULL; // sp154, sp130
|
|
#ifdef VERSION_EU
|
|
UNUSED u8 padEU[0x04];
|
|
#endif
|
|
UNUSED u8 pad8[0x04];
|
|
#ifndef VERSION_EU
|
|
u16 resamplingRateFixedPoint; // sp5c, sp11A
|
|
#endif
|
|
s32 noteFinished; // 150 t2, sp124
|
|
s32 restart; // 14c t3, sp120
|
|
s32 flags; // sp148, sp11C
|
|
#ifdef VERSION_EU
|
|
u16 resamplingRateFixedPoint; // sp5c, sp11A
|
|
#endif
|
|
UNUSED u8 pad7[0x0c]; // sp100
|
|
UNUSED s32 tempBufLen;
|
|
#ifdef VERSION_EU
|
|
s32 sp130; //sp128, sp104
|
|
UNUSED u32 pad9;
|
|
#else
|
|
UNUSED u32 pad9;
|
|
s32 sp130; //sp128, sp104
|
|
#endif
|
|
s32 nAdpcmSamplesProcessed; // signed required for US
|
|
s32 t0;
|
|
#ifdef VERSION_EU
|
|
u8 *sampleAddr; // sp120, spF4
|
|
s32 s6;
|
|
#else
|
|
s32 s6;
|
|
u8 *sampleAddr; // sp120, spF4
|
|
#endif
|
|
|
|
#ifdef VERSION_EU
|
|
s32 samplesLenAdjusted; // 108, spEC
|
|
// Might have been used to store (samplesLenFixedPoint >> 0x10), but doing so causes strange
|
|
// behavior with the break near the end of the loop, causing US and JP to need a goto instead
|
|
UNUSED s32 samplesLenInt;
|
|
s32 endPos; // sp110, spE4
|
|
s32 nSamplesToProcess; // sp10c/a0, spE0
|
|
s32 s2;
|
|
#else
|
|
// Might have been used to store (samplesLenFixedPoint >> 0x10), but doing so causes strange
|
|
// behavior with the break near the end of the loop, causing US and JP to need a goto instead
|
|
UNUSED s32 samplesLenInt;
|
|
s32 samplesLenAdjusted; // 108
|
|
s32 s2;
|
|
s32 endPos; // sp110, spE4
|
|
s32 nSamplesToProcess; // sp10c/a0, spE0
|
|
#endif
|
|
|
|
s32 leftRight;
|
|
s32 s3;
|
|
s32 s5; //s4
|
|
|
|
u32 samplesLenFixedPoint; // v1_1
|
|
s32 nSamplesInThisIteration; // v1_2
|
|
u32 a3;
|
|
#ifndef VERSION_EU
|
|
s32 t9;
|
|
#endif
|
|
u8 *v0_2;
|
|
s32 nParts; // spE8, spBC
|
|
s32 curPart; // spE4, spB8
|
|
|
|
#ifndef VERSION_EU
|
|
f32 resamplingRate; // f12
|
|
#endif
|
|
s32 temp;
|
|
|
|
#ifdef VERSION_EU
|
|
s32 s5Aligned;
|
|
#endif
|
|
s32 resampledTempLen; // spD8, spAC
|
|
u16 noteSamplesDmemAddrBeforeResampling; // spD6, spAA
|
|
|
|
|
|
for (noteIndex = 0; noteIndex < gMaxSimultaneousNotes; noteIndex++)
|
|
{
|
|
|
|
note = &gNotes[noteIndex];
|
|
|
|
//! This function requires note->enabled to be volatile, but it breaks other functions like note_enable.
|
|
//! Casting to a struct with just the volatile bitfield works, but there may be a better way to match.
|
|
if (((struct vNote *)note)->enabled && IS_BANK_LOAD_COMPLETE(note->bankId) == FALSE)
|
|
{
|
|
gAudioErrorFlags = (note->bankId << 8) + noteIndex + 0x1000000;
|
|
continue;
|
|
}
|
|
|
|
if (((struct vNote *)note)->enabled)
|
|
{
|
|
|
|
flags = 0;
|
|
|
|
if (note->needsInit == TRUE)
|
|
{
|
|
flags = A_INIT;
|
|
note->samplePosInt = 0;
|
|
note->samplePosFrac = 0;
|
|
}
|
|
|
|
if (note->frequency < US_FLOAT(2.0))
|
|
{
|
|
nParts = 1;
|
|
if (note->frequency > US_FLOAT(1.99996))
|
|
{
|
|
note->frequency = US_FLOAT(1.99996);
|
|
}
|
|
resamplingRate = note->frequency;
|
|
}
|
|
else
|
|
{
|
|
// If frequency is > 2.0, the processing must be split into two parts
|
|
nParts = 2;
|
|
if (note->frequency >= US_FLOAT(3.99993))
|
|
{
|
|
note->frequency = US_FLOAT(3.99993);
|
|
}
|
|
resamplingRate = note->frequency * US_FLOAT(.5);
|
|
}
|
|
|
|
resamplingRateFixedPoint = (u16)(s32)(resamplingRate * 32768.0f);
|
|
samplesLenFixedPoint = note->samplePosFrac + resamplingRateFixedPoint * bufLen * 2;
|
|
note->samplePosFrac = samplesLenFixedPoint & 0xFFFF; // 16-bit store, can't reuse
|
|
|
|
if (note->sound == NULL)
|
|
{
|
|
// A wave synthesis note (not ADPCM)
|
|
|
|
cmd = load_wave_samples(cmd, note, samplesLenFixedPoint >> 0x10);
|
|
noteSamplesDmemAddrBeforeResampling = DMEM_ADDR_UNCOMPRESSED_NOTE + note->samplePosInt * 2;
|
|
note->samplePosInt += samplesLenFixedPoint >> 0x10;
|
|
flags = 0;
|
|
}
|
|
else
|
|
{
|
|
// ADPCM note
|
|
|
|
audioBookSample = note->sound->sample;
|
|
|
|
loopInfo = audioBookSample->loop;
|
|
endPos = loopInfo->end;
|
|
sampleAddr = audioBookSample->sampleAddr;
|
|
resampledTempLen = 0;
|
|
for (curPart = 0; curPart < nParts; curPart++)
|
|
{
|
|
nAdpcmSamplesProcessed = 0; // s8
|
|
s5 = 0; // s4
|
|
|
|
if (nParts == 1)
|
|
{
|
|
samplesLenAdjusted = samplesLenFixedPoint >> 0x10;
|
|
}
|
|
else if (samplesLenFixedPoint >> 0x10 & 1)
|
|
{
|
|
samplesLenAdjusted = (samplesLenFixedPoint >> 0x10 & ~1) + curPart * 2;
|
|
}
|
|
else
|
|
{
|
|
samplesLenAdjusted = samplesLenFixedPoint >> 0x10;
|
|
}
|
|
|
|
if (curLoadedBook != audioBookSample->book->book)
|
|
{
|
|
u32 nEntries; // v1
|
|
curLoadedBook = audioBookSample->book->book;
|
|
nEntries = audioBookSample->book->order * audioBookSample->book->npredictors;
|
|
aLoadADPCM(cmd++, nEntries * 16, VIRTUAL_TO_PHYSICAL2(curLoadedBook));
|
|
}
|
|
|
|
while (nAdpcmSamplesProcessed != samplesLenAdjusted)
|
|
{
|
|
s32 samplesRemaining; // v1
|
|
s32 s0;
|
|
|
|
noteFinished = FALSE;
|
|
restart = FALSE;
|
|
nSamplesToProcess = samplesLenAdjusted - nAdpcmSamplesProcessed;
|
|
s2 = note->samplePosInt & 0xf;
|
|
samplesRemaining = endPos - note->samplePosInt;
|
|
|
|
if (s2 == 0 && note->restart == FALSE)
|
|
{
|
|
s2 = 16;
|
|
}
|
|
s6 = 16 - s2; // a1
|
|
|
|
if (nSamplesToProcess < samplesRemaining)
|
|
{
|
|
t0 = (nSamplesToProcess - s6 + 0xf) / 16;
|
|
s0 = t0 * 16;
|
|
s3 = s6 + s0 - nSamplesToProcess;
|
|
}
|
|
else
|
|
{
|
|
s0 = samplesRemaining + s2 - 0x10;
|
|
s3 = 0;
|
|
if (s0 <= 0)
|
|
{
|
|
s0 = 0;
|
|
s6 = samplesRemaining;
|
|
}
|
|
t0 = (s0 + 0xf) / 16;
|
|
if (loopInfo->count != 0)
|
|
{
|
|
// Loop around and restart
|
|
restart = 1;
|
|
}
|
|
else
|
|
{
|
|
noteFinished = 1;
|
|
}
|
|
}
|
|
|
|
if (t0 != 0)
|
|
{
|
|
temp = (note->samplePosInt - s2 + 0x10) / 16;
|
|
v0_2 = dma_sample_data(
|
|
(uintptr_t)(sampleAddr + temp * 9),
|
|
t0 * 9, flags, ¬e->sampleDmaIndex);
|
|
a3 = (u32)((uintptr_t)v0_2 & 0xf);
|
|
aSetBuffer(cmd++, 0, DMEM_ADDR_COMPRESSED_ADPCM_DATA, 0, t0 * 9 + a3);
|
|
aLoadBuffer(cmd++, VIRTUAL_TO_PHYSICAL2(v0_2 - a3));
|
|
}
|
|
else
|
|
{
|
|
s0 = 0;
|
|
a3 = 0;
|
|
}
|
|
|
|
if (note->restart != FALSE)
|
|
{
|
|
aSetLoop(cmd++, VIRTUAL_TO_PHYSICAL2(audioBookSample->loop->state));
|
|
flags = A_LOOP; // = 2
|
|
note->restart = FALSE;
|
|
}
|
|
|
|
nSamplesInThisIteration = s0 + s6 - s3;
|
|
if (nAdpcmSamplesProcessed == 0)
|
|
{
|
|
aSetBuffer(cmd++, 0, DMEM_ADDR_COMPRESSED_ADPCM_DATA + a3, DMEM_ADDR_UNCOMPRESSED_NOTE, s0 * 2);
|
|
aADPCMdec(cmd++, flags, VIRTUAL_TO_PHYSICAL2(note->synthesisBuffers->adpcmdecState));
|
|
sp130 = s2 * 2;
|
|
}
|
|
else
|
|
{
|
|
aSetBuffer(cmd++, 0, DMEM_ADDR_COMPRESSED_ADPCM_DATA + a3, DMEM_ADDR_UNCOMPRESSED_NOTE + ALIGN(s5, 5), s0 * 2);
|
|
aADPCMdec(cmd++, flags, VIRTUAL_TO_PHYSICAL2(note->synthesisBuffers->adpcmdecState));
|
|
aDMEMMove(cmd++, DMEM_ADDR_UNCOMPRESSED_NOTE + ALIGN(s5, 5) + s2 * 2, DMEM_ADDR_UNCOMPRESSED_NOTE + s5, nSamplesInThisIteration * 2);
|
|
}
|
|
|
|
nAdpcmSamplesProcessed += nSamplesInThisIteration;
|
|
|
|
switch (flags)
|
|
{
|
|
case A_INIT: // = 1
|
|
sp130 = 0;
|
|
s5 = s0 * 2 + s5;
|
|
break;
|
|
|
|
case A_LOOP: // = 2
|
|
s5 = nSamplesInThisIteration * 2 + s5;
|
|
break;
|
|
|
|
default:
|
|
if (s5 != 0)
|
|
{
|
|
s5 = nSamplesInThisIteration * 2 + s5;
|
|
}
|
|
else
|
|
{
|
|
s5 = (s2 + nSamplesInThisIteration) * 2;
|
|
}
|
|
break;
|
|
}
|
|
flags = 0;
|
|
|
|
if (noteFinished)
|
|
{
|
|
aClearBuffer(cmd++, DMEM_ADDR_UNCOMPRESSED_NOTE + s5,
|
|
(samplesLenAdjusted - nAdpcmSamplesProcessed) * 2);
|
|
note->samplePosInt = 0;
|
|
note->finished = 1;
|
|
((struct vNote *)note)->enabled = 0;
|
|
break;
|
|
}
|
|
|
|
if (restart)
|
|
{
|
|
note->restart = TRUE;
|
|
note->samplePosInt = loopInfo->start;
|
|
}
|
|
else
|
|
{
|
|
note->samplePosInt += nSamplesToProcess;
|
|
}
|
|
}
|
|
|
|
switch (nParts)
|
|
{
|
|
case 1:
|
|
noteSamplesDmemAddrBeforeResampling = DMEM_ADDR_UNCOMPRESSED_NOTE + sp130;
|
|
break;
|
|
|
|
case 2:
|
|
switch (curPart)
|
|
{
|
|
case 0:
|
|
aSetBuffer(cmd++, 0, DMEM_ADDR_UNCOMPRESSED_NOTE + sp130, DMEM_ADDR_RESAMPLED, samplesLenAdjusted + 4);
|
|
aResample(cmd++, A_INIT, 0xff60, VIRTUAL_TO_PHYSICAL2(note->synthesisBuffers->dummyResampleState));
|
|
resampledTempLen = samplesLenAdjusted + 4;
|
|
noteSamplesDmemAddrBeforeResampling = DMEM_ADDR_RESAMPLED + 4;
|
|
if (note->finished != FALSE)
|
|
{
|
|
aClearBuffer(cmd++, DMEM_ADDR_RESAMPLED + resampledTempLen, samplesLenAdjusted + 0x10);
|
|
}
|
|
break;
|
|
|
|
case 1:
|
|
aSetBuffer(cmd++, 0, DMEM_ADDR_UNCOMPRESSED_NOTE + sp130,
|
|
DMEM_ADDR_RESAMPLED2,
|
|
samplesLenAdjusted + 8);
|
|
aResample(cmd++, A_INIT, 0xff60,
|
|
VIRTUAL_TO_PHYSICAL2(
|
|
note->synthesisBuffers->dummyResampleState));
|
|
aDMEMMove(cmd++, DMEM_ADDR_RESAMPLED2 + 4,
|
|
DMEM_ADDR_RESAMPLED + resampledTempLen,
|
|
samplesLenAdjusted + 4);
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (note->finished != FALSE)
|
|
{
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
flags = 0;
|
|
|
|
if (note->needsInit == TRUE)
|
|
{
|
|
flags = A_INIT;
|
|
note->needsInit = FALSE;
|
|
}
|
|
|
|
cmd = final_resample(cmd, note, bufLen * 2, resamplingRateFixedPoint,
|
|
noteSamplesDmemAddrBeforeResampling, flags);
|
|
|
|
if (note->headsetPanRight != 0 || note->prevHeadsetPanRight != 0)
|
|
{
|
|
leftRight = 1;
|
|
}
|
|
else if (note->headsetPanLeft != 0 || note->prevHeadsetPanLeft != 0)
|
|
{
|
|
leftRight = 2;
|
|
}
|
|
else
|
|
{
|
|
leftRight = 0;
|
|
}
|
|
|
|
cmd = process_envelope(cmd, note, bufLen, 0, leftRight, flags);
|
|
|
|
if (note->usesHeadsetPanEffects)
|
|
{
|
|
cmd = note_apply_headset_pan_effects(cmd, note, bufLen * 2, flags, leftRight);
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
t9 = bufLen * 2;
|
|
aSetBuffer(cmd++, 0, 0, DMEM_ADDR_TEMP, t9);
|
|
aInterleave(cmd++, DMEM_ADDR_LEFT_CH, DMEM_ADDR_RIGHT_CH);
|
|
t9 *= 2;
|
|
aSetBuffer(cmd++, 0, 0, DMEM_ADDR_TEMP, t9);
|
|
aSaveBuffer(cmd++, VIRTUAL_TO_PHYSICAL2(aiBuf));
|
|
|
|
return cmd;
|
|
}
|
|
|
|
|
|
u64 *load_wave_samples(u64 *cmd, struct Note *note, s32 nSamplesToLoad)
|
|
{
|
|
aSetBuffer(cmd++, /*flags*/ 0, /*dmemin*/ DMEM_ADDR_UNCOMPRESSED_NOTE, /*dmemout*/ 0,
|
|
/*count*/ sizeof(note->synthesisBuffers->samples));
|
|
aLoadBuffer(cmd++, VIRTUAL_TO_PHYSICAL2(note->synthesisBuffers->samples));
|
|
note->samplePosInt &= note->sampleCount - 1;
|
|
s32 a3 = 64 - note->samplePosInt;
|
|
if (a3 < nSamplesToLoad)
|
|
{
|
|
for (s32 i = 0; i <= (nSamplesToLoad - a3 + 63) / 64 - 1; i++)
|
|
{
|
|
aDMEMMove(cmd++, /*dmemin*/ DMEM_ADDR_UNCOMPRESSED_NOTE, /*dmemout*/ DMEM_ADDR_UNCOMPRESSED_NOTE + (1 + i) * sizeof(note->synthesisBuffers->samples), /*count*/ sizeof(note->synthesisBuffers->samples));
|
|
}
|
|
}
|
|
return cmd;
|
|
}
|
|
|
|
u64 *final_resample(u64 *cmd, struct Note *note, s32 count, u16 pitch, u16 dmemIn, u32 flags)
|
|
{
|
|
aSetBuffer(cmd++, /*flags*/ 0, dmemIn, /*dmemout*/ DMEM_ADDR_TEMP, count);
|
|
aResample(cmd++, flags, pitch, VIRTUAL_TO_PHYSICAL2(note->synthesisBuffers->finalResampleState));
|
|
return cmd;
|
|
}
|
|
|
|
#ifndef VERSION_EU
|
|
u64 *process_envelope(
|
|
u64 *cmd, struct Note *note, s32 nSamples, u16 inBuf, s32 headsetPanSettings,
|
|
UNUSED u32 flags)
|
|
{
|
|
UNUSED u8 pad[16];
|
|
struct VolumeChange vol;
|
|
vol.sourceLeft = note->curVolLeft;
|
|
vol.sourceRight = note->curVolRight;
|
|
vol.targetLeft = note->targetVolLeft;
|
|
vol.targetRight = note->targetVolRight;
|
|
note->curVolLeft = vol.targetLeft;
|
|
note->curVolRight = vol.targetRight;
|
|
return process_envelope_inner(cmd, note, nSamples, inBuf, headsetPanSettings, &vol);
|
|
}
|
|
|
|
u64 *process_envelope_inner(
|
|
u64 *cmd, struct Note *note, s32 nSamples, u16 inBuf,
|
|
s32 headsetPanSettings, struct VolumeChange *vol)
|
|
{
|
|
UNUSED u8 pad[3];
|
|
u8 mixerFlags;
|
|
UNUSED u8 pad2[8];
|
|
s32 rampLeft, rampRight;
|
|
#elif defined(VERSION_EU)
|
|
u64 *process_envelope(u64 *cmd, struct NoteSubEu *note, struct NoteSynthesisState *synthesisState, s32 nSamples, u16 inBuf, s32 headsetPanSettings, UNUSED u32 flags)
|
|
{
|
|
UNUSED u8 pad1[20];
|
|
u16 sourceRight;
|
|
u16 sourceLeft;
|
|
UNUSED u8 pad2[4];
|
|
u16 targetLeft;
|
|
u16 targetRight;
|
|
s32 mixerFlags;
|
|
s32 rampLeft;
|
|
s32 rampRight;
|
|
|
|
sourceLeft = synthesisState->curVolLeft;
|
|
sourceRight = synthesisState->curVolRight;
|
|
targetLeft = (note->targetVolLeft << 5);
|
|
targetRight = (note->targetVolRight << 5);
|
|
if (targetLeft == 0)
|
|
{
|
|
targetLeft++;
|
|
}
|
|
if (targetRight == 0)
|
|
{
|
|
targetRight++;
|
|
}
|
|
synthesisState->curVolLeft = targetLeft;
|
|
synthesisState->curVolRight = targetRight;
|
|
#endif
|
|
|
|
// For aEnvMixer, five buffers and count are set using aSetBuffer.
|
|
// in, dry left, count without A_AUX flag.
|
|
// dry right, wet left, wet right with A_AUX flag.
|
|
|
|
if (note->usesHeadsetPanEffects)
|
|
{
|
|
aClearBuffer(cmd++, DMEM_ADDR_NOTE_PAN_TEMP, DEFAULT_LEN_1CH);
|
|
|
|
switch (headsetPanSettings)
|
|
{
|
|
case 1:
|
|
aSetBuffer(cmd++, 0, inBuf, DMEM_ADDR_NOTE_PAN_TEMP, nSamples * 2);
|
|
aSetBuffer(cmd++, A_AUX, DMEM_ADDR_RIGHT_CH, DMEM_ADDR_WET_LEFT_CH,
|
|
DMEM_ADDR_WET_RIGHT_CH);
|
|
break;
|
|
case 2:
|
|
aSetBuffer(cmd++, 0, inBuf, DMEM_ADDR_LEFT_CH, nSamples * 2);
|
|
aSetBuffer(cmd++, A_AUX, DMEM_ADDR_NOTE_PAN_TEMP, DMEM_ADDR_WET_LEFT_CH,
|
|
DMEM_ADDR_WET_RIGHT_CH);
|
|
break;
|
|
default:
|
|
aSetBuffer(cmd++, 0, inBuf, DMEM_ADDR_LEFT_CH, nSamples * 2);
|
|
aSetBuffer(cmd++, A_AUX, DMEM_ADDR_RIGHT_CH, DMEM_ADDR_WET_LEFT_CH,
|
|
DMEM_ADDR_WET_RIGHT_CH);
|
|
break;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
// It's a bit unclear what the "stereo strong" concept does.
|
|
// Instead of mixing the opposite channel to the normal buffers, the sound is first
|
|
// mixed into a temporary buffer and then subtracted from the normal buffer.
|
|
if (note->stereoStrongRight)
|
|
{
|
|
aClearBuffer(cmd++, DMEM_ADDR_STEREO_STRONG_TEMP_DRY, DEFAULT_LEN_2CH);
|
|
aSetBuffer(cmd++, 0, inBuf, DMEM_ADDR_STEREO_STRONG_TEMP_DRY, nSamples * 2);
|
|
aSetBuffer(cmd++, A_AUX, DMEM_ADDR_RIGHT_CH, DMEM_ADDR_STEREO_STRONG_TEMP_WET,
|
|
DMEM_ADDR_WET_RIGHT_CH);
|
|
}
|
|
else if (note->stereoStrongLeft)
|
|
{
|
|
aClearBuffer(cmd++, DMEM_ADDR_STEREO_STRONG_TEMP_DRY, DEFAULT_LEN_2CH);
|
|
aSetBuffer(cmd++, 0, inBuf, DMEM_ADDR_LEFT_CH, nSamples * 2);
|
|
aSetBuffer(cmd++, A_AUX, DMEM_ADDR_STEREO_STRONG_TEMP_DRY, DMEM_ADDR_WET_LEFT_CH,
|
|
DMEM_ADDR_STEREO_STRONG_TEMP_WET);
|
|
}
|
|
else
|
|
{
|
|
aSetBuffer(cmd++, 0, inBuf, DMEM_ADDR_LEFT_CH, nSamples * 2);
|
|
aSetBuffer(cmd++, A_AUX, DMEM_ADDR_RIGHT_CH, DMEM_ADDR_WET_LEFT_CH, DMEM_ADDR_WET_RIGHT_CH);
|
|
}
|
|
}
|
|
|
|
#ifdef VERSION_EU
|
|
if (targetLeft == sourceLeft && targetRight == sourceRight && !note->envMixerNeedsInit) {
|
|
#else
|
|
if (vol->targetLeft == vol->sourceLeft && vol->targetRight == vol->sourceRight
|
|
&& !note->envMixerNeedsInit)
|
|
{
|
|
#endif
|
|
mixerFlags = A_CONTINUE;
|
|
}
|
|
else
|
|
{
|
|
mixerFlags = A_INIT;
|
|
|
|
#ifdef VERSION_EU
|
|
rampLeft = gCurrentLeftVolRamping[targetLeft >> 5] * gCurrentRightVolRamping[sourceLeft >> 5];
|
|
rampRight = gCurrentLeftVolRamping[targetRight >> 5] * gCurrentRightVolRamping[sourceRight >> 5];
|
|
#else
|
|
rampLeft = get_volume_ramping(vol->sourceLeft, vol->targetLeft, nSamples);
|
|
rampRight = get_volume_ramping(vol->sourceRight, vol->targetRight, nSamples);
|
|
#endif
|
|
|
|
// The operation's parameters change meanings depending on flags
|
|
#ifdef VERSION_EU
|
|
aSetVolume(cmd++, A_VOL | A_LEFT, sourceLeft, 0, 0);
|
|
aSetVolume(cmd++, A_VOL | A_RIGHT, sourceRight, 0, 0);
|
|
aSetVolume32(cmd++, A_RATE | A_LEFT, targetLeft, rampLeft);
|
|
aSetVolume32(cmd++, A_RATE | A_RIGHT, targetRight, rampRight);
|
|
aSetVolume(cmd++, A_AUX, gVolume, 0, note->reverbVol << 8);
|
|
#else
|
|
aSetVolume(cmd++, A_VOL | A_LEFT, vol->sourceLeft, 0, 0);
|
|
aSetVolume(cmd++, A_VOL | A_RIGHT, vol->sourceRight, 0, 0);
|
|
aSetVolume32(cmd++, A_RATE | A_LEFT, vol->targetLeft, rampLeft);
|
|
aSetVolume32(cmd++, A_RATE | A_RIGHT, vol->targetRight, rampRight);
|
|
aSetVolume(cmd++, A_AUX, gVolume, 0, note->reverbVolShifted);
|
|
#endif
|
|
}
|
|
|
|
#ifdef VERSION_EU
|
|
if (gUseReverb &¬e->reverbVol != 0) {
|
|
aEnvMixer(cmd++, mixerFlags | A_AUX,
|
|
VIRTUAL_TO_PHYSICAL2(synthesisState->synthesisBuffers->mixEnvelopeState));
|
|
#else
|
|
if (gSynthesisReverb.useReverb && note->reverbVol != 0)
|
|
{
|
|
aEnvMixer(cmd++, mixerFlags | A_AUX,
|
|
VIRTUAL_TO_PHYSICAL2(note->synthesisBuffers->mixEnvelopeState));
|
|
#endif
|
|
if (note->stereoStrongRight)
|
|
{
|
|
aSetBuffer(cmd++, 0, 0, 0, nSamples * 2);
|
|
// 0x8000 is -100%, so subtract sound instead of adding...
|
|
aMix(cmd++, 0, /*gain*/ 0x8000, /*in*/ DMEM_ADDR_STEREO_STRONG_TEMP_DRY,
|
|
/*out*/ DMEM_ADDR_LEFT_CH);
|
|
aMix(cmd++, 0, /*gain*/ 0x8000, /*in*/ DMEM_ADDR_STEREO_STRONG_TEMP_WET,
|
|
/*out*/ DMEM_ADDR_WET_LEFT_CH);
|
|
}
|
|
else if (note->stereoStrongLeft)
|
|
{
|
|
aSetBuffer(cmd++, 0, 0, 0, nSamples * 2);
|
|
aMix(cmd++, 0, /*gain*/ 0x8000, /*in*/ DMEM_ADDR_STEREO_STRONG_TEMP_DRY,
|
|
/*out*/ DMEM_ADDR_RIGHT_CH);
|
|
aMix(cmd++, 0, /*gain*/ 0x8000, /*in*/ DMEM_ADDR_STEREO_STRONG_TEMP_WET,
|
|
/*out*/ DMEM_ADDR_WET_RIGHT_CH);
|
|
}
|
|
}
|
|
else
|
|
{
|
|
#ifdef VERSION_EU
|
|
aEnvMixer(cmd++, mixerFlags, VIRTUAL_TO_PHYSICAL2(synthesisState->synthesisBuffers->mixEnvelopeState));
|
|
#else
|
|
aEnvMixer(cmd++, mixerFlags, VIRTUAL_TO_PHYSICAL2(note->synthesisBuffers->mixEnvelopeState));
|
|
#endif
|
|
if (note->stereoStrongRight)
|
|
{
|
|
aSetBuffer(cmd++, 0, 0, 0, nSamples * 2);
|
|
aMix(cmd++, 0, /*gain*/ 0x8000, /*in*/ DMEM_ADDR_STEREO_STRONG_TEMP_DRY,
|
|
/*out*/ DMEM_ADDR_LEFT_CH);
|
|
}
|
|
else if (note->stereoStrongLeft)
|
|
{
|
|
aSetBuffer(cmd++, 0, 0, 0, nSamples * 2);
|
|
aMix(cmd++, 0, /*gain*/ 0x8000, /*in*/ DMEM_ADDR_STEREO_STRONG_TEMP_DRY,
|
|
/*out*/ DMEM_ADDR_RIGHT_CH);
|
|
}
|
|
}
|
|
return cmd;
|
|
}
|
|
|
|
#ifdef VERSION_EU
|
|
u64 *note_apply_headset_pan_effects(u64 *cmd, struct NoteSubEu *noteSubEu, struct NoteSynthesisState *note, s32 bufLen, s32 flags, s32 leftRight)
|
|
{
|
|
|
|
|
|
#else
|
|
u64 *note_apply_headset_pan_effects(u64 *cmd, struct Note *note, s32 bufLen, s32 flags, s32 leftRight)
|
|
{
|
|
#endif
|
|
u16 dest;
|
|
u16 pitch;
|
|
#ifdef VERSION_EU
|
|
u8 prevPanShift;
|
|
u8 panShift;
|
|
UNUSED u8 unkDebug;
|
|
#else
|
|
u16 prevPanShift;
|
|
u16 panShift;
|
|
#endif
|
|
|
|
switch (leftRight)
|
|
{
|
|
case 1:
|
|
dest = DMEM_ADDR_LEFT_CH;
|
|
#ifdef VERSION_EU
|
|
panShift = noteSubEu->headsetPanRight;
|
|
#else
|
|
panShift = note->headsetPanRight;
|
|
#endif
|
|
note->prevHeadsetPanLeft = 0;
|
|
prevPanShift = note->prevHeadsetPanRight;
|
|
note->prevHeadsetPanRight = panShift;
|
|
break;
|
|
case 2:
|
|
dest = DMEM_ADDR_RIGHT_CH;
|
|
#ifdef VERSION_EU
|
|
panShift = noteSubEu->headsetPanLeft;
|
|
#else
|
|
panShift = note->headsetPanLeft;
|
|
#endif
|
|
note->prevHeadsetPanRight = 0;
|
|
|
|
prevPanShift = note->prevHeadsetPanLeft;
|
|
note->prevHeadsetPanLeft = panShift;
|
|
break;
|
|
default:
|
|
return cmd;
|
|
}
|
|
|
|
if (flags != 1)
|
|
{ // A_INIT?
|
|
// Slightly adjust the sample rate in order to fit a change in pan shift
|
|
if (prevPanShift == 0)
|
|
{
|
|
// Kind of a hack that moves the first samples into the resample state
|
|
aDMEMMove(cmd++, DMEM_ADDR_NOTE_PAN_TEMP, DMEM_ADDR_TEMP, 8);
|
|
aClearBuffer(cmd++, 8, 8); // Set pitch accumulator to 0 in the resample state
|
|
aDMEMMove(cmd++, DMEM_ADDR_NOTE_PAN_TEMP, DMEM_ADDR_TEMP + 0x10,
|
|
0x10); // No idea, result seems to be overwritten later
|
|
|
|
aSetBuffer(cmd++, 0, 0, DMEM_ADDR_TEMP, 32);
|
|
aSaveBuffer(cmd++, VIRTUAL_TO_PHYSICAL2(note->synthesisBuffers->panResampleState));
|
|
|
|
#ifdef VERSION_EU
|
|
pitch = (bufLen << 0xf) / (bufLen + panShift - prevPanShift + 8);
|
|
if (pitch)
|
|
{
|
|
}
|
|
#else
|
|
pitch = (bufLen << 0xf) / (panShift + bufLen - prevPanShift + 8);
|
|
#endif
|
|
aSetBuffer(cmd++, 0, DMEM_ADDR_NOTE_PAN_TEMP + 8, DMEM_ADDR_TEMP, panShift + bufLen - prevPanShift);
|
|
aResample(cmd++, 0, pitch, VIRTUAL_TO_PHYSICAL2(note->synthesisBuffers->panResampleState));
|
|
}
|
|
else
|
|
{
|
|
if (panShift == 0)
|
|
{
|
|
pitch = (bufLen << 0xf) / (bufLen - prevPanShift - 4);
|
|
}
|
|
else
|
|
{
|
|
pitch = (bufLen << 0xf) / (bufLen + panShift - prevPanShift);
|
|
}
|
|
|
|
#if defined(VERSION_EU) && !defined(AVOID_UB)
|
|
if (unkDebug)
|
|
{ // UB
|
|
}
|
|
#endif
|
|
aSetBuffer(cmd++, 0, DMEM_ADDR_NOTE_PAN_TEMP, DMEM_ADDR_TEMP, panShift + bufLen - prevPanShift);
|
|
aResample(cmd++, 0, pitch, VIRTUAL_TO_PHYSICAL2(note->synthesisBuffers->panResampleState));
|
|
}
|
|
|
|
if (prevPanShift != 0)
|
|
{
|
|
aSetBuffer(cmd++, 0, DMEM_ADDR_NOTE_PAN_TEMP, 0, prevPanShift);
|
|
aLoadBuffer(cmd++, VIRTUAL_TO_PHYSICAL2(note->synthesisBuffers->panSamplesBuffer));
|
|
aDMEMMove(cmd++, DMEM_ADDR_TEMP, DMEM_ADDR_NOTE_PAN_TEMP + prevPanShift, panShift + bufLen - prevPanShift);
|
|
}
|
|
else
|
|
{
|
|
aDMEMMove(cmd++, DMEM_ADDR_TEMP, DMEM_ADDR_NOTE_PAN_TEMP, panShift + bufLen - prevPanShift);
|
|
}
|
|
}
|
|
else
|
|
{
|
|
// Just shift right
|
|
aDMEMMove(cmd++, DMEM_ADDR_NOTE_PAN_TEMP, DMEM_ADDR_TEMP, bufLen);
|
|
aDMEMMove(cmd++, DMEM_ADDR_TEMP, DMEM_ADDR_NOTE_PAN_TEMP + panShift, bufLen);
|
|
aClearBuffer(cmd++, DMEM_ADDR_NOTE_PAN_TEMP, panShift);
|
|
}
|
|
|
|
if (panShift)
|
|
{
|
|
// Save excessive samples for next iteration
|
|
aSetBuffer(cmd++, 0, 0, DMEM_ADDR_NOTE_PAN_TEMP + bufLen, panShift);
|
|
aSaveBuffer(cmd++, VIRTUAL_TO_PHYSICAL2(note->synthesisBuffers->panSamplesBuffer));
|
|
}
|
|
|
|
aSetBuffer(cmd++, 0, 0, 0, bufLen);
|
|
aMix(cmd++, 0, /*gain*/ 0x7fff, /*in*/ DMEM_ADDR_NOTE_PAN_TEMP, /*out*/ dest);
|
|
|
|
return cmd;
|
|
}
|
|
|
|
#ifndef VERSION_EU
|
|
// Moved to playback.c in EU
|
|
|
|
void note_init_volume(struct Note *note)
|
|
{
|
|
note->targetVolLeft = 0;
|
|
note->targetVolRight = 0;
|
|
note->reverbVol = 0;
|
|
note->reverbVolShifted = 0;
|
|
note->unused2 = 0;
|
|
note->curVolLeft = 1;
|
|
note->curVolRight = 1;
|
|
note->frequency = 0.0f;
|
|
}
|
|
|
|
void note_set_vel_pan_reverb(struct Note *note, f32 velocity, f32 pan, u8 reverbVol)
|
|
{
|
|
s32 panIndex;
|
|
f32 volLeft;
|
|
f32 volRight;
|
|
// Anding with 127 avoids out-of-bounds reads when pan is outside of [0, 1].
|
|
// This can occur during PU movement -- see the bug comment in get_sound_pan
|
|
// in external.c. An out-of-bounds read by itself doesn't crash, but if the
|
|
// resulting value is a nan or denormal, performing arithmetic on it crashes
|
|
// on console.
|
|
#ifdef VERSION_JP
|
|
panIndex = MIN((s32)(pan * 127.5), 127);
|
|
#else
|
|
panIndex = (s32)(pan * 127.5f) & 127;
|
|
#endif
|
|
if (note->stereoHeadsetEffects && gSoundMode == SOUND_MODE_HEADSET)
|
|
{
|
|
s8 smallPanIndex;
|
|
s8 temp = (s8)(pan * 10.0f);
|
|
if (temp < 9)
|
|
{
|
|
smallPanIndex = temp;
|
|
}
|
|
else
|
|
{
|
|
smallPanIndex = 9;
|
|
}
|
|
note->headsetPanLeft = gHeadsetPanQuantization[smallPanIndex];
|
|
note->headsetPanRight = gHeadsetPanQuantization[9 - smallPanIndex];
|
|
note->stereoStrongRight = FALSE;
|
|
note->stereoStrongLeft = FALSE;
|
|
note->usesHeadsetPanEffects = TRUE;
|
|
volLeft = gHeadsetPanVolume[panIndex];
|
|
volRight = gHeadsetPanVolume[127 - panIndex];
|
|
}
|
|
else if (note->stereoHeadsetEffects && gSoundMode == SOUND_MODE_STEREO)
|
|
{
|
|
u8 strongLeft = FALSE;
|
|
u8 strongRight = FALSE;
|
|
note->headsetPanLeft = 0;
|
|
note->headsetPanRight = 0;
|
|
note->usesHeadsetPanEffects = FALSE;
|
|
volLeft = gStereoPanVolume[panIndex];
|
|
volRight = gStereoPanVolume[127 - panIndex];
|
|
if (panIndex < 0x20)
|
|
{
|
|
strongLeft = TRUE;
|
|
}
|
|
else if (panIndex > 0x60)
|
|
{
|
|
strongRight = TRUE;
|
|
}
|
|
note->stereoStrongRight = strongRight;
|
|
note->stereoStrongLeft = strongLeft;
|
|
}
|
|
else if (gSoundMode == SOUND_MODE_MONO)
|
|
{
|
|
volLeft = .707f;
|
|
volRight = .707f;
|
|
}
|
|
else
|
|
{
|
|
volLeft = gDefaultPanVolume[panIndex];
|
|
volRight = gDefaultPanVolume[127 - panIndex];
|
|
}
|
|
|
|
if (velocity < 0)
|
|
{
|
|
velocity = 0;
|
|
}
|
|
#ifdef VERSION_JP
|
|
note->targetVolLeft = (u16)(velocity * volLeft) & ~0x80FF; // 0x7F00, but that doesn't match
|
|
note->targetVolRight = (u16)(velocity * volRight) & ~0x80FF;
|
|
#else
|
|
note->targetVolLeft = (u16)(s32)(velocity * volLeft) & ~0x80FF;
|
|
note->targetVolRight = (u16)(s32)(velocity * volRight) & ~0x80FF;
|
|
#endif
|
|
if (note->targetVolLeft == 0)
|
|
{
|
|
note->targetVolLeft++;
|
|
}
|
|
if (note->targetVolRight == 0)
|
|
{
|
|
note->targetVolRight++;
|
|
}
|
|
if (note->reverbVol != reverbVol)
|
|
{
|
|
note->reverbVol = reverbVol;
|
|
note->reverbVolShifted = reverbVol << 8;
|
|
note->envMixerNeedsInit = TRUE;
|
|
return;
|
|
}
|
|
|
|
if (note->needsInit)
|
|
{
|
|
note->envMixerNeedsInit = TRUE;
|
|
}
|
|
else
|
|
{
|
|
note->envMixerNeedsInit = FALSE;
|
|
}
|
|
}
|
|
|
|
void note_set_frequency(struct Note *note, f32 frequency)
|
|
{
|
|
note->frequency = frequency;
|
|
}
|
|
|
|
void note_enable(struct Note *note)
|
|
{
|
|
note->enabled = TRUE;
|
|
note->needsInit = TRUE;
|
|
note->restart = FALSE;
|
|
note->finished = FALSE;
|
|
note->stereoStrongRight = FALSE;
|
|
note->stereoStrongLeft = FALSE;
|
|
note->usesHeadsetPanEffects = FALSE;
|
|
note->headsetPanLeft = 0;
|
|
note->headsetPanRight = 0;
|
|
note->prevHeadsetPanRight = 0;
|
|
note->prevHeadsetPanLeft = 0;
|
|
}
|
|
|
|
void note_disable(struct Note *note)
|
|
{
|
|
if (note->needsInit == TRUE)
|
|
{
|
|
note->needsInit = FALSE;
|
|
}
|
|
else
|
|
{
|
|
note_set_vel_pan_reverb(note, 0, .5, 0);
|
|
}
|
|
note->priority = NOTE_PRIORITY_DISABLED;
|
|
note->enabled = FALSE;
|
|
note->finished = FALSE;
|
|
note->parentLayer = NO_LAYER;
|
|
note->prevParentLayer = NO_LAYER;
|
|
}
|
|
#endif
|
|
#endif
|