902 lines
38 KiB
C
902 lines
38 KiB
C
#include <ultra64.h>
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#include <macros.h>
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#include "synthesis.h"
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#include "memory.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 "external.h"
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#define DMEM_ADDR_TEMP 0x0
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#define DMEM_ADDR_UNCOMPRESSED_NOTE 0x180
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#define DMEM_ADDR_ADPCM_RESAMPLED 0x20
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#define DMEM_ADDR_ADPCM_RESAMPLED2 0x160
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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 SynthesisReverb gSynthesisReverb;
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u8 sAudioSynthesisPad[0x20];
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struct VolumeChange {
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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(u16 *aiBuf, s32 bufLen, u64 *cmd, u32 updateIndex);
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u64 *synthesis_process_notes(u16 *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(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(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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void prepare_reverb_ring_buffer(s32 chunkLen, u32 updateIndex) {
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struct ReverbRingBufferItem *item;
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s32 srcPos;
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s32 dstPos;
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s32 nSamples;
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s32 numSamplesAfterDownsampling;
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s32 excessiveSamples;
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if (gReverbDownsampleRate != 1) {
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if (gSynthesisReverb.framesLeftToIgnore == 0) {
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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->lengths[0] / 2;
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srcPos += gReverbDownsampleRate, dstPos++) {
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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->lengths[1] / 2; srcPos += gReverbDownsampleRate, dstPos++) {
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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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numSamplesAfterDownsampling = nSamples = chunkLen / gReverbDownsampleRate;
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if (((nSamples + gSynthesisReverb.nextRingBufferPos) - gSynthesisReverb.bufSizePerChannel) < 0) {
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// There is space in the ring buffer before it wraps around
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item->lengths[0] = nSamples * 2;
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item->lengths[1] = 0;
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item->startPos = (s32) gSynthesisReverb.nextRingBufferPos;
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gSynthesisReverb.nextRingBufferPos += nSamples;
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} else {
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// Ring buffer wrapped around
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excessiveSamples =
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(nSamples + gSynthesisReverb.nextRingBufferPos) - gSynthesisReverb.bufSizePerChannel;
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nSamples = numSamplesAfterDownsampling - excessiveSamples;
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item->lengths[0] = nSamples * 2;
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item->lengths[1] = 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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s32 get_volume_ramping(u16 sourceVol, u16 targetVol, s32 arg2) {
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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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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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// bufLen will be divisible by 16
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u64 *synthesis_execute(u64 *cmdBuf, s32 *writtenCmds, u16 *aiBuf, s32 bufLen) {
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s32 chunkLen;
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s32 i;
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s32 remaining = bufLen;
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u32 *aiBufPtr = (u32 *) aiBuf;
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u64 *cmd = cmdBuf;
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s32 v0;
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aSegment(cmd++, 0, 0);
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for (i = gAudioUpdatesPerFrame; i > 0; i--) {
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if (i == 1) {
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// 'remaining' will automatically be divisible by 8, no need to round
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chunkLen = remaining;
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} else {
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v0 = remaining / 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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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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prepare_reverb_ring_buffer(chunkLen, gAudioUpdatesPerFrame - i);
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}
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cmd = synthesis_do_one_audio_update((u16 *) aiBufPtr, chunkLen, cmd, gAudioUpdatesPerFrame - i);
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remaining -= chunkLen;
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aiBufPtr += chunkLen;
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}
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if (gSynthesisReverb.framesLeftToIgnore != 0) {
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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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u64 *synthesis_do_one_audio_update(u16 *aiBuf, s32 bufLen, u64 *cmd, u32 updateIndex) {
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UNUSED s32 pad1[1];
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s16 ra;
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s16 t4;
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UNUSED s32 pad[2];
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struct ReverbRingBufferItem *v1;
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UNUSED s32 pad2[2];
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v1 = &gSynthesisReverb.items[gSynthesisReverb.curFrame][updateIndex];
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if (gSynthesisReverb.useReverb == 0) {
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aClearBuffer(cmd++, DMEM_ADDR_LEFT_CH, DEFAULT_LEN_2CH);
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cmd = synthesis_process_notes(aiBuf, bufLen, cmd);
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} else {
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if (gReverbDownsampleRate == 1) {
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// Put the oldest samples in the ring buffer into the wet channels
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aSetLoadBufferPair(cmd++, 0, v1->startPos);
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if (v1->lengths[1] != 0) {
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// Ring buffer wrapped
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aSetLoadBufferPair(cmd++, v1->lengths[0], 0);
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}
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// Use the reverb sound as initial sound for this audio update
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aDMEMMove(cmd++, DMEM_ADDR_WET_LEFT_CH, DMEM_ADDR_LEFT_CH, DEFAULT_LEN_2CH);
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// (Hopefully) lower the volume of the wet channels. New reverb will later be mixed into
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// these channels.
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aSetBuffer(cmd++, 0, 0, 0, DEFAULT_LEN_2CH);
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// 0x8000 here is -100%
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aMix(cmd++, 0, /*gain*/ 0x8000 + gSynthesisReverb.reverbGain, /*in*/ DMEM_ADDR_WET_LEFT_CH,
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/*out*/ DMEM_ADDR_WET_LEFT_CH);
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} else {
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// Same as above but upsample the previously downsampled samples used for reverb first
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t4 = (v1->startPos & 7) * 2;
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ra = ALIGN(v1->lengths[0] + t4, 4);
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aSetLoadBufferPair(cmd++, 0, v1->startPos - t4 / 2);
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if (v1->lengths[1] != 0) {
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// Ring buffer wrapped
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aSetLoadBufferPair(cmd++, ra, 0);
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}
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aSetBuffer(cmd++, 0, t4 + DMEM_ADDR_WET_LEFT_CH, DMEM_ADDR_LEFT_CH, bufLen << 1);
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aResample(cmd++, gSynthesisReverb.resampleFlags, (u16) gSynthesisReverb.resampleRate,
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VIRTUAL_TO_PHYSICAL2(gSynthesisReverb.resampleStateLeft));
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aSetBuffer(cmd++, 0, t4 + DMEM_ADDR_WET_RIGHT_CH, DMEM_ADDR_RIGHT_CH, bufLen << 1);
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aResample(cmd++, gSynthesisReverb.resampleFlags, (u16) gSynthesisReverb.resampleRate,
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VIRTUAL_TO_PHYSICAL2(gSynthesisReverb.resampleStateRight));
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aSetBuffer(cmd++, 0, 0, 0, DEFAULT_LEN_2CH);
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aMix(cmd++, 0, /*gain*/ 0x8000 + gSynthesisReverb.reverbGain, /*in*/ DMEM_ADDR_LEFT_CH,
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/*out*/ DMEM_ADDR_LEFT_CH);
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aDMEMMove(cmd++, DMEM_ADDR_LEFT_CH, DMEM_ADDR_WET_LEFT_CH, DEFAULT_LEN_2CH);
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}
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cmd = synthesis_process_notes(aiBuf, bufLen, cmd);
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if (gReverbDownsampleRate == 1) {
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aSetSaveBufferPair(cmd++, 0, v1->lengths[0], v1->startPos);
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if (v1->lengths[1] != 0) {
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// Ring buffer wrapped
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aSetSaveBufferPair(cmd++, v1->lengths[0], v1->lengths[1], 0);
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}
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} else {
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// Downsampling is done later by CPU when RSP is done, therefore we need to have double
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// buffering. Left and right buffers are adjacent in memory.
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aSetBuffer(cmd++, 0, 0, DMEM_ADDR_WET_LEFT_CH, DEFAULT_LEN_2CH);
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aSaveBuffer(
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cmd++,
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VIRTUAL_TO_PHYSICAL2(
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gSynthesisReverb.items[gSynthesisReverb.curFrame][updateIndex].toDownsampleLeft));
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gSynthesisReverb.resampleFlags = 0;
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}
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}
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return cmd;
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}
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#ifdef NON_MATCHING
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u64 *synthesis_process_notes(u16 *aiBuf, s32 bufLen, u64 *cmd) {
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s32 noteIndex; // sp174
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struct Note *note; // s7
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struct AudioBankSample *audioBookSample; // sp164
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struct AdpcmLoop *loopInfo; // sp160
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s16 *curLoadedBook; // sp15C
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s32 noteFinished; // 150 t2
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s32 restart; // 14c t3
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s32 flags; // sp148
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UNUSED u8 pad8[0x14];
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s32 sp130;
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UNUSED u8 pad7[0xC];
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u8 *sampleAddr; // sp120
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u32 samplesLenAdjusted; // 108, definitely unsigned, t5
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// UNUSED u8 pad6[4];
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s32 endPos; // sp110
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s32 nSamplesToProcess; // 10c a0
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// UNUSED u8 pad5[0x10c - 0xe8 - 4];
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s32 nParts; // spE8
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s32 curPart; // spE4
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u32 nAdpcmSamplesProcessed; // probably unsigned, fp
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s32 t0;
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s32 resampledTempLen; // spD8
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u16 noteSamplesDmemAddrBeforeResampling; // spD6
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// sp6c is a temporary!
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u16 resamplingRateFixedPoint; // sp5c
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s32 samplesLenInt; // sp58
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s32 onePart; // sp54
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s32 s6;
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s32 s6_2;
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s32 s2;
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s32 s0;
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s32 s3;
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s32 s5;
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// s32 v0;
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u32 samplesLenFixedPoint; // v1_1
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s32 nSamplesInThisIteration; // v1_2
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u32 a3;
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s32 t9;
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u8 *v0_2;
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f32 resamplingRate; // f12
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UNUSED s32 temp;
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for (noteIndex = 0, curLoadedBook = NULL; noteIndex < gMaxSimultaneousNotes; noteIndex++) {
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note = &gNotes[noteIndex];
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if (IS_BANK_LOAD_COMPLETE(note->bankId) == FALSE) {
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gAudioErrorFlags = (note->bankId << 8) + noteIndex + 0x1000000;
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} else if (note->enabled) {
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// This matches much much better if enabled is volatile... but that
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// breaks other functions (e.g. note_enable). Can we achieve the
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// volatile effect in some other way?
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flags = 0;
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if (note->needsInit == TRUE) {
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flags = A_INIT;
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note->samplePosInt = 0;
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note->samplePosFrac = 0;
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}
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if (note->frequency < US_FLOAT(2.0)) {
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nParts = 1;
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if (note->frequency > US_FLOAT(1.99996)) {
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note->frequency = US_FLOAT(1.99996);
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}
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resamplingRate = note->frequency;
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} else {
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// If frequency is > 2.0, the processing must be split into two parts
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nParts = 2;
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if (note->frequency >= US_FLOAT(3.99993)) {
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note->frequency = US_FLOAT(3.99993);
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}
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resamplingRate = note->frequency * US_FLOAT(.5);
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}
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resamplingRateFixedPoint = (u16)(s32)(resamplingRate * 32768.0f);
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samplesLenFixedPoint = note->samplePosFrac + (resamplingRateFixedPoint * bufLen) * 2;
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note->samplePosFrac = samplesLenFixedPoint; // 16-bit store, can't reuse
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if (note->sound == NULL) {
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// A wave synthesis note (not ADPCM)
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// samplesLenFixedPoint >> 0x10 stored in s0
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cmd = load_wave_samples(cmd, note, samplesLenFixedPoint >> 0x10);
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noteSamplesDmemAddrBeforeResampling =
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DMEM_ADDR_UNCOMPRESSED_NOTE + note->samplePosInt * 2;
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note->samplePosInt += (samplesLenFixedPoint >> 0x10);
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flags = 0;
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} else {
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// ADPCM note
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audioBookSample = note->sound->sample;
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// sp58 is a low-numbered register, so possibly a temporary.
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// Should it be used for samplesLenFixedPoint >> 0x10 above as well? But then
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// the asm matches worse. This variable seems to highly involved
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// in causing this function not to match...
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samplesLenInt = samplesLenFixedPoint >> 0x10; // v0 // sp58
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loopInfo = audioBookSample->loop;
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endPos = loopInfo->end;
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sampleAddr = audioBookSample->sampleAddr;
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onePart = (nParts == 1);
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resampledTempLen = 0;
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for (curPart = 0; curPart < nParts; curPart++) {
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nAdpcmSamplesProcessed = 0;
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s5 = 0;
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// This whole if-else if chain is weird. First it uses onePart
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// instead of nParts == 1, and it needs a weird if to not
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// induce non-matchings all over the rest of the function.
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// Then it induces a bunch of stack-relative loads that
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// shouldn't be there. Finally, it relates to sp58, which
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// behaves very oddly...
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if (onePart) { // nParts == 1
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if (1) { // shouldn't be here, but it makes things line up better...
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samplesLenAdjusted = samplesLenInt;
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}
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} else if (samplesLenInt & 1) {
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samplesLenAdjusted = (samplesLenInt & ~1) + (curPart * 2);
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} else {
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samplesLenAdjusted = samplesLenInt;
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}
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if (curLoadedBook != audioBookSample->book->book) {
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u32 nEntries; // v1
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curLoadedBook = audioBookSample->book->book;
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nEntries = audioBookSample->book->order * audioBookSample->book->npredictors;
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aLoadADPCM(cmd++, nEntries * 16, VIRTUAL_TO_PHYSICAL2(curLoadedBook));
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}
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while (nAdpcmSamplesProcessed != samplesLenAdjusted) {
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s32 samplesRemaining; // v1
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s32 s0;
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// sp58 = sp58; here, doesn't happen
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noteFinished = FALSE;
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restart = FALSE;
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nSamplesToProcess = samplesLenAdjusted - nAdpcmSamplesProcessed;
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s2 = note->samplePosInt & 0xf;
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samplesRemaining = endPos - note->samplePosInt;
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if (s2 == 0 && !note->restart) {
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s2 = 16;
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}
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s6 = 16 - s2; // a1
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if (nSamplesToProcess < samplesRemaining) {
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t0 = (nSamplesToProcess - s6 + 0xf) / 16;
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s0 = t0 * 16;
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s3 = s6 + s0 - nSamplesToProcess;
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} else {
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s0 = samplesRemaining + s2 - 0x10;
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s3 = 0;
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if (s0 <= 0) {
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s0 = 0;
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s6 = samplesRemaining;
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}
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t0 = (s0 + 0xf) / 16;
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if (loopInfo->count != 0) {
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// Loop around and restart
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restart = 1;
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} else {
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noteFinished = 1;
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}
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}
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// Improve regalloc for saved registers. Probably
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// shouldn't be here, but it gives nicer diffs for now.
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s6_2 = s6;
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if (t0 != 0) {
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// maybe keep a var for t0 * 9?
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v0_2 = dma_sample_data(
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(uintptr_t) (sampleAddr + (note->samplePosInt - s2 + 0x10) / 16 * 9),
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t0 * 9, flags, ¬e->sampleDmaIndex);
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a3 = (u32)((uintptr_t) v0_2 & 0xf);
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aSetBuffer(cmd++, 0, DMEM_ADDR_COMPRESSED_ADPCM_DATA, 0, t0 * 9 + a3);
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aLoadBuffer(cmd++, VIRTUAL_TO_PHYSICAL2(v0_2 - a3));
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} else {
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s0 = 0;
|
|
a3 = 0;
|
|
}
|
|
|
|
if (note->restart != FALSE) {
|
|
aSetLoop(cmd++, VIRTUAL_TO_PHYSICAL2(audioBookSample->loop->state));
|
|
flags = A_LOOP; // = 2
|
|
note->restart = FALSE;
|
|
}
|
|
|
|
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, (s0 + s6_2 - s3) * 2);
|
|
}
|
|
|
|
nAdpcmSamplesProcessed = nAdpcmSamplesProcessed + s0 + s6_2 - s3;
|
|
nSamplesInThisIteration = s0 + s6_2 - s3;
|
|
switch (flags) {
|
|
case A_INIT: // = 1
|
|
sp130 = 0;
|
|
s5 += s0 * 2;
|
|
break;
|
|
|
|
case A_LOOP: // = 2
|
|
s5 += nSamplesInThisIteration * 2;
|
|
break;
|
|
|
|
default:
|
|
if (s5 != 0) {
|
|
s5 += nSamplesInThisIteration * 2;
|
|
} else {
|
|
s5 = (nSamplesInThisIteration + s2) * 2;
|
|
}
|
|
break;
|
|
}
|
|
flags = 0;
|
|
|
|
if (noteFinished) {
|
|
aClearBuffer(cmd++, DMEM_ADDR_UNCOMPRESSED_NOTE + s5,
|
|
(samplesLenAdjusted - nAdpcmSamplesProcessed) * 2);
|
|
note->samplePosInt = 0;
|
|
note->finished = 1;
|
|
note->enabled = 0;
|
|
break; // goto? doesn't matter, though
|
|
}
|
|
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_ADPCM_RESAMPLED, samplesLenAdjusted + 4);
|
|
aResample(cmd++, A_INIT, 0xff60,
|
|
VIRTUAL_TO_PHYSICAL2(
|
|
note->synthesisBuffers->dummyResampleState));
|
|
resampledTempLen = samplesLenAdjusted + 4;
|
|
noteSamplesDmemAddrBeforeResampling = DMEM_ADDR_ADPCM_RESAMPLED + 4;
|
|
if (note->finished != 0) {
|
|
aClearBuffer(cmd++,
|
|
DMEM_ADDR_ADPCM_RESAMPLED + samplesLenAdjusted + 4,
|
|
samplesLenAdjusted + 0x10);
|
|
}
|
|
break;
|
|
|
|
case 1:
|
|
aSetBuffer(cmd++, 0, DMEM_ADDR_UNCOMPRESSED_NOTE + sp130,
|
|
DMEM_ADDR_ADPCM_RESAMPLED2, samplesLenAdjusted + 8);
|
|
aResample(cmd++, A_INIT, 0xff60,
|
|
VIRTUAL_TO_PHYSICAL2(
|
|
note->synthesisBuffers->dummyResampleState));
|
|
aDMEMMove(cmd++, DMEM_ADDR_ADPCM_RESAMPLED2 + 4,
|
|
DMEM_ADDR_ADPCM_RESAMPLED + resampledTempLen,
|
|
samplesLenAdjusted + 4);
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (note->finished != 0) {
|
|
// ("break;" doesn't match)
|
|
flags = 0;
|
|
goto out;
|
|
}
|
|
}
|
|
|
|
flags = 0;
|
|
}
|
|
out:
|
|
|
|
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) {
|
|
s0 = 1;
|
|
} else if (note->headsetPanLeft != 0 || note->prevHeadsetPanLeft != 0) {
|
|
s0 = 2;
|
|
} else {
|
|
s0 = 0;
|
|
}
|
|
|
|
cmd = process_envelope(cmd, note, bufLen, 0, s0, flags);
|
|
if (note->usesHeadsetPanEffects) {
|
|
cmd = note_apply_headset_pan_effects(cmd, note, bufLen * 2, flags, s0);
|
|
}
|
|
}
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
#elif defined(VERSION_JP)
|
|
GLOBAL_ASM("asm/non_matchings/synthesis_process_notes_jp.s")
|
|
#else
|
|
GLOBAL_ASM("asm/non_matchings/synthesis_process_notes_us.s")
|
|
#endif
|
|
|
|
u64 *load_wave_samples(u64 *cmd, struct Note *note, s32 nSamplesToLoad) {
|
|
s32 a3;
|
|
s32 i;
|
|
aSetBuffer(cmd++, /*flags*/ 0, /*dmemin*/ DMEM_ADDR_UNCOMPRESSED_NOTE, /*dmemout*/ 0,
|
|
/*count*/ sizeof(note->synthesisBuffers->samples)); // interesting that it's 128...
|
|
aLoadBuffer(cmd++, VIRTUAL_TO_PHYSICAL2(note->synthesisBuffers->samples));
|
|
note->samplePosInt = (note->sampleCount - 1) & note->samplePosInt;
|
|
a3 = 64 - note->samplePosInt;
|
|
if (a3 < nSamplesToLoad) {
|
|
for (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*/ 0, count);
|
|
aResample(cmd++, flags, pitch, VIRTUAL_TO_PHYSICAL2(note->synthesisBuffers->finalResampleState));
|
|
return cmd;
|
|
}
|
|
|
|
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;
|
|
|
|
// 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);
|
|
}
|
|
}
|
|
|
|
if (vol->sourceLeft == vol->targetLeft && vol->sourceRight == vol->targetRight
|
|
&& !note->envMixerNeedsInit) {
|
|
mixerFlags = A_CONTINUE;
|
|
} else {
|
|
mixerFlags = A_INIT;
|
|
rampLeft = get_volume_ramping(vol->sourceLeft, vol->targetLeft, nSamples);
|
|
rampRight = get_volume_ramping(vol->sourceRight, vol->targetRight, nSamples);
|
|
|
|
// The operation's parameters change meanings depending on flags
|
|
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->reverbVol);
|
|
}
|
|
if (gSynthesisReverb.useReverb && note->reverb) {
|
|
aEnvMixer(cmd++, mixerFlags | A_AUX,
|
|
VIRTUAL_TO_PHYSICAL2(note->synthesisBuffers->mixEnvelopeState));
|
|
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 {
|
|
aEnvMixer(cmd++, mixerFlags, VIRTUAL_TO_PHYSICAL2(note->synthesisBuffers->mixEnvelopeState));
|
|
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;
|
|
}
|
|
|
|
u64 *note_apply_headset_pan_effects(u64 *cmd, struct Note *note, s32 bufLen, s32 flags, s32 leftRight) {
|
|
u16 dest;
|
|
u16 prevPanShift;
|
|
u16 panShift;
|
|
u16 pitch; // t2
|
|
UNUSED s32 padding[11];
|
|
|
|
switch (leftRight) {
|
|
case 1:
|
|
dest = DMEM_ADDR_LEFT_CH;
|
|
note->prevHeadsetPanLeft = 0;
|
|
panShift = note->headsetPanRight;
|
|
prevPanShift = note->prevHeadsetPanRight;
|
|
note->prevHeadsetPanRight = panShift;
|
|
break;
|
|
case 2:
|
|
dest = DMEM_ADDR_RIGHT_CH;
|
|
note->prevHeadsetPanRight = 0;
|
|
panShift = note->headsetPanLeft;
|
|
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));
|
|
|
|
pitch = (bufLen << 0xf) / (panShift + bufLen - prevPanShift + 8);
|
|
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 {
|
|
pitch = (panShift == 0) ? (bufLen << 0xf) / (bufLen - prevPanShift - 4)
|
|
: (bufLen << 0xf) / (bufLen + panShift - prevPanShift);
|
|
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;
|
|
}
|
|
|
|
void note_init_volume(struct Note *note) {
|
|
note->targetVolLeft = 0;
|
|
note->targetVolRight = 0;
|
|
note->reverb = 0;
|
|
note->reverbVol = 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 reverb) {
|
|
s32 panIndex;
|
|
f32 volLeft;
|
|
f32 volRight;
|
|
#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;
|
|
smallPanIndex = MIN((s8)(pan * 10.0f), 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];
|
|
}
|
|
|
|
velocity = MAX(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->reverb != reverb) {
|
|
note->reverb = reverb;
|
|
note->reverbVol = reverb << 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;
|
|
}
|