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Microtransactions64/src/audio/synthesis.c
Gregory Heskett 5064fcfb69 Merge pull request #757 from HackerN64/master 
Merge HackerSM64 2.1.3 release into 2.2.0
2024-01-31 01:23:18 -05:00

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#ifndef VERSION_SH
#include <ultra64.h>
#include "synthesis.h"
#include "heap.h"
#include "data.h"
#include "load.h"
#include "seqplayer.h"
#include "internal.h"
#include "external.h"
#include "game/game_init.h"
#include "game/debug.h"
#include "engine/math_util.h"
#define DMEM_ADDR_TEMP 0x0
#define DMEM_ADDR_RESAMPLED 0x20
#define DMEM_ADDR_RESAMPLED2 0x160
#define DMEM_ADDR_UNCOMPRESSED_NOTE 0x180
#define DMEM_ADDR_NOTE_PAN_TEMP 0x200
#define DMEM_ADDR_STEREO_STRONG_TEMP_DRY 0x200
#define DMEM_ADDR_STEREO_STRONG_TEMP_WET 0x340
#define DMEM_ADDR_COMPRESSED_ADPCM_DATA 0x3f0
#define DMEM_ADDR_LEFT_CH 0x4c0
#define DMEM_ADDR_RIGHT_CH 0x600
#define DMEM_ADDR_WET_LEFT_CH 0x740
#define DMEM_ADDR_WET_RIGHT_CH 0x880
#define aSetLoadBufferPair(pkt, c, off) \
aSetBuffer(pkt, 0, c + DMEM_ADDR_WET_LEFT_CH, 0, DEFAULT_LEN_1CH - c); \
aLoadBuffer(pkt, VIRTUAL_TO_PHYSICAL2(gSynthesisReverb.ringBuffer.left + (off))); \
aSetBuffer(pkt, 0, c + DMEM_ADDR_WET_RIGHT_CH, 0, DEFAULT_LEN_1CH - c); \
aLoadBuffer(pkt, VIRTUAL_TO_PHYSICAL2(gSynthesisReverb.ringBuffer.right + (off)));
#define aSetSaveBufferPair(pkt, c, d, off) \
aSetBuffer(pkt, 0, 0, c + DMEM_ADDR_WET_LEFT_CH, d); \
aSaveBuffer(pkt, VIRTUAL_TO_PHYSICAL2(gSynthesisReverb.ringBuffer.left + (off))); \
aSetBuffer(pkt, 0, 0, c + DMEM_ADDR_WET_RIGHT_CH, d); \
aSaveBuffer(pkt, VIRTUAL_TO_PHYSICAL2(gSynthesisReverb.ringBuffer.right + (off)));
#define VOLRAMPING_MASK (~(0x8000 | ((1 << (15 - VOL_RAMPING_EXPONENT)) - 1)))
#ifdef BETTER_REVERB
// Do not touch these values manually, unless you want potential for problems.
u8 gBetterReverbPresetValue = 0;
u8 toggleBetterReverb = FALSE;
u8 betterReverbLightweight = FALSE;
u8 monoReverb;
s8 betterReverbDownsampleRate;
static s32 reverbMults[SYNTH_CHANNEL_STEREO_COUNT][NUM_ALLPASS / 3] = {0};
static s32 allpassIdx[SYNTH_CHANNEL_STEREO_COUNT][NUM_ALLPASS] = {0};
static s32 betterReverbDelays[SYNTH_CHANNEL_STEREO_COUNT][NUM_ALLPASS] = {0};
static s32 historySamplesLight[SYNTH_CHANNEL_STEREO_COUNT];
static s16 **delayBufs[SYNTH_CHANNEL_STEREO_COUNT];
u8 *gReverbMults[SYNTH_CHANNEL_STEREO_COUNT];
s32 reverbLastFilterIndex;
s32 reverbFilterCount;
s32 betterReverbWindowsSize;
s32 betterReverbRevIndex; // This one is okay to adjust whenever
s32 betterReverbGainIndex; // This one is okay to adjust whenever
#endif
struct VolumeChange {
u16 sourceLeft;
u16 sourceRight;
u16 targetLeft;
u16 targetRight;
};
u64 *synthesis_do_one_audio_update(s16 *aiBuf, u32 bufLen, u64 *cmd, s32 updateIndex);
u64 *synthesis_process_notes(s16 *aiBuf, u32 bufLen, u64 *cmd);
u64 *load_wave_samples(u64 *cmd, struct Note *note, s32 nSamplesToLoad);
#ifdef ENABLE_STEREO_HEADSET_EFFECTS
u64 *process_envelope(u64 *cmd, struct Note *note, s32 nSamples, u16 inBuf, s32 headsetPanSettings);
u64 *note_apply_headset_pan_effects(u64 *cmd, struct Note *note, s32 bufLen, s32 flags, s32 leftRight);
#else
u64 *process_envelope(u64 *cmd, struct Note *note, s32 nSamples, u16 inBuf);
#endif
struct SynthesisReverb gSynthesisReverb;
f32 *currentRampingTableLeft;
f32 *currentRampingTableRight;
#ifdef BETTER_REVERB
static void reverb_samples(s16 *start, s16 *end, s16 *downsampleBuffer, s32 channel) {
s16 *curDelaySample;
s32 historySample;
s32 tmpCarryover;
s32 outSampleTotal;
s32 i;
s32 j;
s32 k;
s32 downsampleIncrement = gReverbDownsampleRate;
s32 *delaysLocal = betterReverbDelays[channel];
s32 *reverbMultsLocal = reverbMults[channel];
s32 *allpassIdxLocal = allpassIdx[channel];
s16 **delayBufsLocal = delayBufs[channel];
s32 lastFilterIndex = reverbLastFilterIndex;
s32 revIndex = betterReverbRevIndex;
s32 gainIndex = betterReverbGainIndex;
// Set j outside of the loop only. Because we're forcing filter count to always be a multiple of 3, we can count on j always being 0 when exiting the second for loop.
j = 0;
for (; start < end; start++, downsampleBuffer += downsampleIncrement) {
// Mix the very last filter output with new incoming sample
tmpCarryover = ((delayBufsLocal[lastFilterIndex][allpassIdxLocal[lastFilterIndex]] * revIndex) >> 8) + *downsampleBuffer;
outSampleTotal = 0;
i = 0;
k = 0;
for (; i <= lastFilterIndex; ++i, ++j) {
curDelaySample = &delayBufsLocal[i][allpassIdxLocal[i]];
historySample = *curDelaySample;
if (j == 2) {
j = -1;
outSampleTotal += ((historySample * reverbMultsLocal[k++]) >> 8);
*curDelaySample = CLAMP_S16(tmpCarryover);
if (i != lastFilterIndex)
tmpCarryover = ((historySample * revIndex) >> 8)/* + *downsampleBuffer*/;
} else {
tmpCarryover += (historySample * (-gainIndex)) >> 8;
*curDelaySample = CLAMP_S16(tmpCarryover);
tmpCarryover = ((tmpCarryover * gainIndex) >> 8) + historySample;
}
if (++allpassIdxLocal[i] == delaysLocal[i]) allpassIdxLocal[i] = 0;
}
*start = CLAMP_S16(outSampleTotal);
}
}
static void reverb_samples_light(s16 *start, s16 *end, s16 *downsampleBuffer, s32 channel) {
s16 *curDelaySample;
s32 historySample;
s32 tmpCarryover;
s32 i;
s32 downsampleIncrement = gReverbDownsampleRate;
s32 *delaysLocal = betterReverbDelays[channel];
s32 *allpassIdxLocal = allpassIdx[channel];
s16 **delayBufsLocal = delayBufs[channel];
// Get history sample from last processing tick
tmpCarryover = historySamplesLight[channel];
for (; start < end; start++, downsampleBuffer += downsampleIncrement) {
// Mix previous sample with new incoming sample
tmpCarryover = ((tmpCarryover * BETTER_REVERB_REVERB_INDEX_LIGHT) >> 8) + *downsampleBuffer;
for (i = 0; i < BETTER_REVERB_FILTER_COUNT_LIGHT; ++i) {
curDelaySample = &delayBufsLocal[i][allpassIdxLocal[i]];
historySample = *curDelaySample;
tmpCarryover += ((historySample * (-BETTER_REVERB_GAIN_INDEX_LIGHT)) >> 8);
*curDelaySample = CLAMP_S16(tmpCarryover);
tmpCarryover = ((tmpCarryover * BETTER_REVERB_GAIN_INDEX_LIGHT) >> 8) + historySample;
if (++allpassIdxLocal[i] == delaysLocal[i]) allpassIdxLocal[i] = 0;
}
// Lightweight does not use the final filter type at all, unlike standard reverb processing
*start = CLAMP_S16(tmpCarryover);
}
// Copy history sample to temporary buffer for processing next tick
historySamplesLight[channel] = tmpCarryover;
}
void initialize_better_reverb_buffers(void) {
delayBufs[SYNTH_CHANNEL_LEFT] = (s16**) soundAlloc(&gBetterReverbPool, BETTER_REVERB_PTR_SIZE);
delayBufs[SYNTH_CHANNEL_RIGHT] = &delayBufs[SYNTH_CHANNEL_LEFT][NUM_ALLPASS];
}
void set_better_reverb_buffers(u32 *inputDelaysL, u32 *inputDelaysR) {
s32 bufOffset = 0;
s32 filterCount = reverbFilterCount;
u32 *inputDelayPtrs[SYNTH_CHANNEL_STEREO_COUNT] = {
[SYNTH_CHANNEL_LEFT] = inputDelaysL,
[SYNTH_CHANNEL_RIGHT] = inputDelaysR,
};
if (betterReverbLightweight)
filterCount = BETTER_REVERB_FILTER_COUNT_LIGHT;
gBetterReverbPool.cur = gBetterReverbPool.start + BETTER_REVERB_PTR_SIZE; // Reset reverb data pool
// Don't bother setting any buffers if BETTER_REVERB is disabled
if (!toggleBetterReverb)
return;
// NOTE: Using filterCount over NUM_ALLPASS will report less memory usage with fewer filters, but poses an additional
// risk to anybody testing on console with performance compromises, as emulator can be easily overlooked.
for (s32 channel = 0; channel < SYNTH_CHANNEL_STEREO_COUNT; channel++) {
historySamplesLight[channel] = 0;
for (s32 filter = 0; filter < filterCount; filter++) {
betterReverbDelays[channel][filter] = (s32) (inputDelayPtrs[channel][filter] / gReverbDownsampleRate);
delayBufs[channel][filter] = soundAlloc(&gBetterReverbPool, betterReverbDelays[channel][filter] * sizeof(s16));
bufOffset += betterReverbDelays[channel][filter];
}
}
aggress(bufOffset * sizeof(s16) <= BETTER_REVERB_SIZE - BETTER_REVERB_PTR_SIZE, "BETTER_REVERB_SIZE is too small for this preset!");
bzero(allpassIdx, sizeof(allpassIdx));
}
#endif
void prepare_reverb_ring_buffer(s32 chunkLen, u32 updateIndex) {
struct ReverbRingBufferItem *item;
s32 srcPos, dstPos;
s32 nSamples;
s32 excessiveSamples;
if (gSynthesisReverb.framesLeftToIgnore == 0) {
#ifdef BETTER_REVERB
if (!toggleBetterReverb && gReverbDownsampleRate != 1) {
#else
if (gReverbDownsampleRate != 1) {
#endif
// Now that the RSP has finished, downsample the samples produced two frames ago by skipping
// samples.
item = &gSynthesisReverb.items[gSynthesisReverb.curFrame][updateIndex];
// Touches both left and right since they are adjacent in memory
osInvalDCache(item->toDownsampleLeft, DEFAULT_LEN_2CH);
for (srcPos = 0, dstPos = 0; dstPos < item->lengthA / 2;
srcPos += gReverbDownsampleRate, dstPos++) {
gSynthesisReverb.ringBuffer.left[dstPos + item->startPos] = item->toDownsampleLeft[srcPos];
gSynthesisReverb.ringBuffer.right[dstPos + item->startPos] = item->toDownsampleRight[srcPos];
}
for (dstPos = 0; dstPos < item->lengthB / 2; srcPos += gReverbDownsampleRate, dstPos++) {
gSynthesisReverb.ringBuffer.left[dstPos] = item->toDownsampleLeft[srcPos];
gSynthesisReverb.ringBuffer.right[dstPos] = item->toDownsampleRight[srcPos];
}
}
#ifdef BETTER_REVERB
else if (toggleBetterReverb) {
s32 loopCounts[2];
s16 *betterReverbDownsampleBuffers[SYNTH_CHANNEL_STEREO_COUNT][ARRAY_COUNT(loopCounts)]; // StartA and StartB for both channels
s16 *betterReverbSampleBuffers[SYNTH_CHANNEL_STEREO_COUNT][ARRAY_COUNT(loopCounts)]; // Output reverb buffers
void (*reverbFunc)(s16*, s16*, s16*, s32) = betterReverbLightweight ? reverb_samples_light : reverb_samples; // Function pointers for both heavy and lightweight reverb functions
item = &gSynthesisReverb.items[gSynthesisReverb.curFrame][updateIndex];
loopCounts[0] = item->lengthA / 2;
loopCounts[1] = item->lengthB / 2;
if (gReverbDownsampleRate != 1) {
// NOTE: / HACKERSM64 TODO: Commenting this check seems to improve runtime by about 100 microseconds (per 30fps frame),
// but idk enough about why it was added here in vanilla to comfortably remove it. Is it supposed to act as an
// optimization (that isn't actually an optimization) or is it a safety measure since it's loaded from the RSP?
osInvalDCache(item->toDownsampleLeft, DEFAULT_LEN_2CH);
}
betterReverbSampleBuffers[SYNTH_CHANNEL_LEFT][0] = &gSynthesisReverb.ringBuffer.left[item->startPos];
betterReverbSampleBuffers[SYNTH_CHANNEL_LEFT][1] = &gSynthesisReverb.ringBuffer.left[0];
betterReverbSampleBuffers[SYNTH_CHANNEL_RIGHT][0] = &gSynthesisReverb.ringBuffer.right[item->startPos];
betterReverbSampleBuffers[SYNTH_CHANNEL_RIGHT][1] = &gSynthesisReverb.ringBuffer.right[0];
if (gReverbDownsampleRate > 1) {
betterReverbDownsampleBuffers[SYNTH_CHANNEL_LEFT][0] = &item->toDownsampleLeft[0];
betterReverbDownsampleBuffers[SYNTH_CHANNEL_LEFT][1] = &item->toDownsampleLeft[loopCounts[0] * gReverbDownsampleRate];
betterReverbDownsampleBuffers[SYNTH_CHANNEL_RIGHT][0] = &item->toDownsampleRight[0];
betterReverbDownsampleBuffers[SYNTH_CHANNEL_RIGHT][1] = &item->toDownsampleRight[loopCounts[0] * gReverbDownsampleRate];
} else {
betterReverbDownsampleBuffers[SYNTH_CHANNEL_LEFT][0] = betterReverbSampleBuffers[SYNTH_CHANNEL_LEFT][0];
betterReverbDownsampleBuffers[SYNTH_CHANNEL_LEFT][1] = betterReverbSampleBuffers[SYNTH_CHANNEL_LEFT][1];
betterReverbDownsampleBuffers[SYNTH_CHANNEL_RIGHT][0] = betterReverbSampleBuffers[SYNTH_CHANNEL_RIGHT][0];
betterReverbDownsampleBuffers[SYNTH_CHANNEL_RIGHT][1] = betterReverbSampleBuffers[SYNTH_CHANNEL_RIGHT][1];
}
if (gSoundMode == SOUND_MODE_MONO || monoReverb) {
for (srcPos = 0; srcPos < ARRAY_COUNT(loopCounts); srcPos++) { // LengthA and LengthB processing
s16 *downsampleBufferL = betterReverbDownsampleBuffers[SYNTH_CHANNEL_LEFT][srcPos];
s16 *downsampleBufferR = betterReverbDownsampleBuffers[SYNTH_CHANNEL_RIGHT][srcPos];
for (dstPos = 0; dstPos < loopCounts[srcPos]; dstPos += gReverbDownsampleRate) { // Individual sample processing
downsampleBufferL[dstPos] = ((s32) downsampleBufferL[dstPos] + (s32) downsampleBufferR[dstPos]) >> 1; // Merge stereo samples into left channel
}
}
for (srcPos = 0; srcPos < ARRAY_COUNT(loopCounts); srcPos++) { // LengthA and LengthB processing
// Call core reverb processing function, either reverb_samples() or reverb_samples_light()
(*reverbFunc)(betterReverbSampleBuffers[SYNTH_CHANNEL_LEFT][srcPos], betterReverbSampleBuffers[SYNTH_CHANNEL_LEFT][srcPos] + loopCounts[srcPos], betterReverbDownsampleBuffers[SYNTH_CHANNEL_LEFT][srcPos], SYNTH_CHANNEL_LEFT);
bcopy(betterReverbSampleBuffers[SYNTH_CHANNEL_LEFT][srcPos], betterReverbSampleBuffers[SYNTH_CHANNEL_RIGHT][srcPos], loopCounts[srcPos] * sizeof(s16));
}
} else {
for (dstPos = 0; dstPos < SYNTH_CHANNEL_STEREO_COUNT; dstPos++) { // left and right channels
for (srcPos = 0; srcPos < ARRAY_COUNT(loopCounts); srcPos++) { // LengthA and LengthB processing
// Call core reverb processing function, either reverb_samples() or reverb_samples_light()
(*reverbFunc)(betterReverbSampleBuffers[dstPos][srcPos], betterReverbSampleBuffers[dstPos][srcPos] + loopCounts[srcPos], betterReverbDownsampleBuffers[dstPos][srcPos], dstPos);
}
}
}
}
#endif
}
item = &gSynthesisReverb.items[gSynthesisReverb.curFrame][updateIndex];
s32 numSamplesAfterDownsampling = chunkLen / gReverbDownsampleRate;
if (((numSamplesAfterDownsampling + gSynthesisReverb.nextRingBufferPos) - gSynthesisReverb.bufSizePerChannel) < 0) {
// There is space in the ring buffer before it wraps around
item->lengthA = numSamplesAfterDownsampling * 2;
item->lengthB = 0;
item->startPos = (s32) gSynthesisReverb.nextRingBufferPos;
gSynthesisReverb.nextRingBufferPos += numSamplesAfterDownsampling;
} else {
// Ring buffer wrapped around
excessiveSamples =
(numSamplesAfterDownsampling + gSynthesisReverb.nextRingBufferPos) - gSynthesisReverb.bufSizePerChannel;
nSamples = numSamplesAfterDownsampling - excessiveSamples;
item->lengthA = nSamples * 2;
item->lengthB = excessiveSamples * 2;
item->startPos = gSynthesisReverb.nextRingBufferPos;
gSynthesisReverb.nextRingBufferPos = excessiveSamples;
}
// These fields are never read later
item->numSamplesAfterDownsampling = numSamplesAfterDownsampling;
item->chunkLen = chunkLen;
}
// bufLen will be divisible by 16
u64 *synthesis_execute(u64 *cmdBuf, s32 *writtenCmds, s16 *aiBuf, s32 bufLen) {
u32 chunkLen;
s32 i;
u32 *aiBufPtr = (u32 *) aiBuf;
u64 *cmd = cmdBuf + 1;
s32 v0;
aSegment(cmdBuf, 0, 0);
#ifdef BETTER_REVERB
s32 filterCountDiv3 = reverbFilterCount / 3;
reverbFilterCount = filterCountDiv3 * 3; // reverbFilterCount should always be a multiple of 3.
if (reverbFilterCount > NUM_ALLPASS) {
reverbFilterCount = NUM_ALLPASS;
} else if (reverbFilterCount < 3) {
reverbFilterCount = 3;
}
reverbLastFilterIndex = reverbFilterCount - 1;
// Update reverbMults every audio frame just in case gReverbMults is ever to change.
if (gReverbMults[SYNTH_CHANNEL_LEFT] != NULL && gReverbMults[SYNTH_CHANNEL_RIGHT] != NULL) {
for (i = 0; i < filterCountDiv3; ++i) {
reverbMults[SYNTH_CHANNEL_LEFT][i] = gReverbMults[SYNTH_CHANNEL_LEFT][i];
reverbMults[SYNTH_CHANNEL_RIGHT][i] = gReverbMults[SYNTH_CHANNEL_RIGHT][i];
}
}
// If there's only one reverb multiplier set, adjust these to match so one channel doesn't end up potentially overpowering the other.
if (filterCountDiv3 == 1) {
reverbMults[SYNTH_CHANNEL_LEFT][0] = (reverbMults[SYNTH_CHANNEL_RIGHT][0] + reverbMults[SYNTH_CHANNEL_LEFT][0]) / 2;
reverbMults[SYNTH_CHANNEL_RIGHT][0] = reverbMults[SYNTH_CHANNEL_LEFT][0];
}
#endif
for (i = gAudioUpdatesPerFrame; i > 0; i--) {
if (i == 1) {
// 'bufLen' will automatically be divisible by 8, no need to round
chunkLen = bufLen;
} else {
v0 = bufLen / i;
// chunkLen = v0 rounded to nearest multiple of 8
chunkLen = v0 - (v0 & 7);
if ((v0 & 7) >= 4) {
chunkLen += 8;
}
}
AUDIO_PROFILER_SWITCH(PROFILER_TIME_SUB_AUDIO_UPDATE, PROFILER_TIME_SUB_AUDIO_SEQUENCES);
AUDIO_PROFILER_START_SHARED(PROFILER_TIME_SUB_AUDIO_SEQUENCES, PROFILER_TIME_SUB_AUDIO_SEQUENCES_SCRIPT);
process_sequences(i - 1);
AUDIO_PROFILER_COMPLETE_AND_SWITCH(PROFILER_TIME_SUB_AUDIO_SEQUENCES_PROCESSING, PROFILER_TIME_SUB_AUDIO_SEQUENCES, PROFILER_TIME_SUB_AUDIO_SYNTHESIS);
AUDIO_PROFILER_START_SHARED(PROFILER_TIME_SUB_AUDIO_SYNTHESIS, PROFILER_TIME_SUB_AUDIO_SYNTHESIS_ENVELOPE_REVERB);
if (gSynthesisReverb.useReverb) {
prepare_reverb_ring_buffer(chunkLen, gAudioUpdatesPerFrame - i);
}
cmd = synthesis_do_one_audio_update((s16 *) aiBufPtr, chunkLen * 2, cmd, gAudioUpdatesPerFrame - i);
AUDIO_PROFILER_COMPLETE_AND_SWITCH(PROFILER_TIME_SUB_AUDIO_SYNTHESIS_ENVELOPE_REVERB, PROFILER_TIME_SUB_AUDIO_SYNTHESIS, PROFILER_TIME_SUB_AUDIO_UPDATE);
bufLen -= chunkLen;
aiBufPtr += chunkLen;
}
if (gSynthesisReverb.framesLeftToIgnore != 0) {
gSynthesisReverb.framesLeftToIgnore--;
}
gSynthesisReverb.curFrame ^= 1;
*writtenCmds = cmd - cmdBuf;
return cmd;
}
u64 *synthesis_do_one_audio_update(s16 *aiBuf, u32 bufLen, u64 *cmd, s32 updateIndex) {
s16 ra;
s16 t4;
struct ReverbRingBufferItem *v1;
v1 = &gSynthesisReverb.items[gSynthesisReverb.curFrame][updateIndex];
if (!gSynthesisReverb.useReverb) {
aClearBuffer(cmd++, DMEM_ADDR_LEFT_CH, DEFAULT_LEN_2CH);
AUDIO_PROFILER_SWITCH(PROFILER_TIME_SUB_AUDIO_SYNTHESIS_ENVELOPE_REVERB, PROFILER_TIME_SUB_AUDIO_SYNTHESIS_PROCESSING);
cmd = synthesis_process_notes(aiBuf, bufLen, cmd);
AUDIO_PROFILER_SWITCH(PROFILER_TIME_SUB_AUDIO_SYNTHESIS_PROCESSING, PROFILER_TIME_SUB_AUDIO_SYNTHESIS_ENVELOPE_REVERB);
} 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);
}
// 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
t4 = (v1->startPos & 7) * 2;
ra = ALIGN16(v1->lengthA + t4);
aSetLoadBufferPair(cmd++, 0, v1->startPos - t4 / 2);
if (v1->lengthB != 0) {
// Ring buffer wrapped
aSetLoadBufferPair(cmd++, ra, 0);
}
aSetBuffer(cmd++, 0, t4 + DMEM_ADDR_WET_LEFT_CH, DMEM_ADDR_LEFT_CH, bufLen);
aResample(cmd++, gSynthesisReverb.resampleFlags, (u16) gSynthesisReverb.resampleRate, VIRTUAL_TO_PHYSICAL2(gSynthesisReverb.resampleStateLeft));
aSetBuffer(cmd++, 0, t4 + DMEM_ADDR_WET_RIGHT_CH, DMEM_ADDR_RIGHT_CH, bufLen);
aResample(cmd++, gSynthesisReverb.resampleFlags, (u16) gSynthesisReverb.resampleRate, VIRTUAL_TO_PHYSICAL2(gSynthesisReverb.resampleStateRight));
#ifdef BETTER_REVERB
// NOTE: Technically using an if/else here means using BETTER_REVERB vanilla presets with downsampling won't match 1-to-1 in volume with BETTER_REVERB being disabled.
// This chunk is actually preferable to what vanilla uses, but was mainly ifdef'd here as a means of documenting BETTER_REVERB changes for other non-HackerSM64 repos.
// Please use this chunk over the latter if matching BETTER_REVERB behavior ever becomes a future priority.
aDMEMMove(cmd++, DMEM_ADDR_LEFT_CH, DMEM_ADDR_WET_LEFT_CH, DEFAULT_LEN_2CH);
aSetBuffer(cmd++, 0, 0, 0, DEFAULT_LEN_2CH);
aMix(cmd++, 0, /*gain*/ 0x8000 + gSynthesisReverb.reverbGain, /*in*/ DMEM_ADDR_WET_LEFT_CH, /*out*/ DMEM_ADDR_WET_LEFT_CH);
#else
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);
#endif
}
AUDIO_PROFILER_SWITCH(PROFILER_TIME_SUB_AUDIO_SYNTHESIS_ENVELOPE_REVERB, PROFILER_TIME_SUB_AUDIO_SYNTHESIS_PROCESSING);
cmd = synthesis_process_notes(aiBuf, bufLen, cmd);
AUDIO_PROFILER_SWITCH(PROFILER_TIME_SUB_AUDIO_SYNTHESIS_PROCESSING, PROFILER_TIME_SUB_AUDIO_SYNTHESIS_ENVELOPE_REVERB);
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;
}
u64 *synthesis_process_notes(s16 *aiBuf, u32 bufLen, u64 *cmd) {
s32 noteIndex; // sp174
struct Note *note; // s7
struct AudioBankSample *audioBookSample; // sp164, sp138
struct AdpcmLoop *loopInfo; // sp160, sp134
s16 *curLoadedBook = NULL; // sp154, sp130
s32 noteFinished; // 150 t2, sp124
s32 restart; // 14c t3, sp120
s32 flags; // sp148, sp11C
s32 sp130 = 0; //sp128, sp104
s32 nAdpcmSamplesProcessed; // signed required for US
s32 t0;
u8 *sampleAddr; // sp120, spF4
s32 s6;
// Might have been used to store (samplesLenFixedPoint >> 16), but doing so causes strange
// behavior with the break near the end of the loop, causing US and JP to need a goto instead
s32 samplesLenAdjusted; // 108
s32 s2;
s32 endPos; // sp110, spE4
s32 nSamplesToProcess; // sp10c/a0, spE0
#ifdef ENABLE_STEREO_HEADSET_EFFECTS
s32 leftRight;
#endif
s32 s3;
s32 s5; //s4
u32 samplesLenFixedPoint; // v1_1
s32 nSamplesInThisIteration; // v1_2
u32 a3;
u8 *v0_2;
s32 nParts; // spE8, spBC
s32 curPart; // spE4, spB8
f32 resamplingRate; // f12
s32 temp;
s32 s5Aligned;
s32 resampledTempLen; // spD8, spAC
u16 noteSamplesDmemAddrBeforeResampling = 0; // spD6, spAA
u16 resamplingRateFixedPoint; // sp5c, sp11A
switch (bufLen) {
case (128 * 2):
currentRampingTableLeft = gVolRampingLhs128;
currentRampingTableRight = gVolRampingRhs128;
break;
case (144 * 2):
currentRampingTableLeft = gVolRampingLhs144;
currentRampingTableRight = gVolRampingRhs144;
break;
case (136 * 2):
default:
currentRampingTableLeft = gVolRampingLhs136;
currentRampingTableRight = gVolRampingRhs136;
break;
}
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)) {
gAudioErrorFlags = (note->bankId << 8) + noteIndex + 0x1000000;
} else if (((struct vNote *)note)->enabled) {
flags = 0;
if (note->needsInit == TRUE) {
flags = A_INIT;
note->samplePosInt = 0;
note->samplePosFrac = 0;
}
if (note->frequency < 2.0f) {
nParts = 1;
if (note->frequency > 1.99996f) {
note->frequency = 1.99996f;
}
resamplingRate = note->frequency;
} else {
// If frequency is > 2.0, the processing must be split into two parts
nParts = 2;
if (note->frequency >= 3.99993f) {
note->frequency = 3.99993f;
}
resamplingRate = note->frequency * 0.5f;
}
resamplingRateFixedPoint = (u16)(s32)(resamplingRate * 32768.0f);
samplesLenFixedPoint = note->samplePosFrac + (resamplingRateFixedPoint * bufLen);
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 >> 16);
noteSamplesDmemAddrBeforeResampling = DMEM_ADDR_UNCOMPRESSED_NOTE + note->samplePosInt * 2;
note->samplePosInt += (samplesLenFixedPoint >> 16);
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 >> 16;
} else if ((samplesLenFixedPoint >> 16) & 1) {
samplesLenAdjusted = ((samplesLenFixedPoint >> 16) & ~1) + (curPart * 2);
}
else {
samplesLenAdjusted = (samplesLenFixedPoint >> 16);
}
if (curLoadedBook != audioBookSample->book->book) {
u32 nEntries; // v1
curLoadedBook = audioBookSample->book->book;
nEntries = audioBookSample->book->order * audioBookSample->book->npredictors * 16U;
aLoadADPCM(cmd++, nEntries, 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 - s6;
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 + 16) / 16;
AUDIO_PROFILER_SWITCH(PROFILER_TIME_SUB_AUDIO_SYNTHESIS_PROCESSING, PROFILER_TIME_SUB_AUDIO_SYNTHESIS_DMA);
v0_2 = dma_sample_data(
(uintptr_t) (sampleAddr + temp * 9),
t0 * 9, flags, &note->sampleDmaIndex);
AUDIO_PROFILER_SWITCH(PROFILER_TIME_SUB_AUDIO_SYNTHESIS_DMA, PROFILER_TIME_SUB_AUDIO_SYNTHESIS_PROCESSING);
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 {
s5Aligned = ALIGN32(s5);
aSetBuffer(cmd++, 0, DMEM_ADDR_COMPRESSED_ADPCM_DATA + a3, DMEM_ADDR_UNCOMPRESSED_NOTE + s5Aligned, s0 * 2);
aADPCMdec(cmd++, flags, VIRTUAL_TO_PHYSICAL2(note->synthesisBuffers->adpcmdecState));
aDMEMMove(cmd++, DMEM_ADDR_UNCOMPRESSED_NOTE + s5Aligned + (s2 * 2), DMEM_ADDR_UNCOMPRESSED_NOTE + s5, (nSamplesInThisIteration) * 2);
}
nAdpcmSamplesProcessed += nSamplesInThisIteration;
switch (flags) {
case A_INIT: // = 1
/**
* !NOTE: Removing this seems to produce a more accurate waveform, however I have no idea why Nintendo decided to add this originally.
* I can only speculate (and hope) that this was just an oversight on their part and this has no reason to exist, given my testing.
* I'm leaving it commented out here just in case though.
*/
// 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 + 16);
}
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;
}
// final resample
aSetBuffer(cmd++, /*flags*/ 0, noteSamplesDmemAddrBeforeResampling, /*dmemout*/ DMEM_ADDR_TEMP, bufLen);
aResample(cmd++, flags, resamplingRateFixedPoint, VIRTUAL_TO_PHYSICAL2(note->synthesisBuffers->finalResampleState));
#ifdef ENABLE_STEREO_HEADSET_EFFECTS
if (note->headsetPanRight != 0 || note->prevHeadsetPanRight != 0) {
leftRight = 1;
} else if (note->headsetPanLeft != 0 || note->prevHeadsetPanLeft != 0) {
leftRight = 2;
} else {
leftRight = 0;
}
AUDIO_PROFILER_SWITCH(PROFILER_TIME_SUB_AUDIO_SYNTHESIS_PROCESSING, PROFILER_TIME_SUB_AUDIO_SYNTHESIS_ENVELOPE_REVERB);
cmd = process_envelope(cmd, note, bufLen, 0, leftRight);
AUDIO_PROFILER_SWITCH(PROFILER_TIME_SUB_AUDIO_SYNTHESIS_ENVELOPE_REVERB, PROFILER_TIME_SUB_AUDIO_SYNTHESIS_PROCESSING);
if (note->usesHeadsetPanEffects) {
cmd = note_apply_headset_pan_effects(cmd, note, bufLen, flags, leftRight);
}
#else
AUDIO_PROFILER_SWITCH(PROFILER_TIME_SUB_AUDIO_SYNTHESIS_PROCESSING, PROFILER_TIME_SUB_AUDIO_SYNTHESIS_ENVELOPE_REVERB);
cmd = process_envelope(cmd, note, bufLen, 0);
AUDIO_PROFILER_SWITCH(PROFILER_TIME_SUB_AUDIO_SYNTHESIS_ENVELOPE_REVERB, PROFILER_TIME_SUB_AUDIO_SYNTHESIS_PROCESSING);
#endif
}
}
aSetBuffer(cmd++, 0, 0, DMEM_ADDR_TEMP, bufLen);
aInterleave(cmd++, DMEM_ADDR_LEFT_CH, DMEM_ADDR_RIGHT_CH);
aSetBuffer(cmd++, 0, 0, DMEM_ADDR_TEMP, bufLen * 2);
aSaveBuffer(cmd++, VIRTUAL_TO_PHYSICAL2(aiBuf));
return cmd;
}
u64 *load_wave_samples(u64 *cmd, struct Note *note, s32 nSamplesToLoad) {
s32 a3;
s32 repeats;
s32 i;
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);
a3 = 64 - note->samplePosInt;
if (a3 < nSamplesToLoad) {
repeats = (nSamplesToLoad - a3 + 63) / 64;
for (i = 0; i < repeats; 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;
}
#ifdef ENABLE_STEREO_HEADSET_EFFECTS
u64 *process_envelope(u64 *cmd, struct Note *note, s32 nSamples, u16 inBuf, s32 headsetPanSettings) {
#else
u64 *process_envelope(u64 *cmd, struct Note *note, s32 nSamples, u16 inBuf) {
#endif
u8 mixerFlags;
s32 rampLeft;
s32 rampRight;
struct VolumeChange vol;
if (note->initFullVelocity) {
note->initFullVelocity = FALSE;
vol.sourceLeft = note->targetVolLeft;
vol.sourceRight = note->targetVolRight;
} else {
vol.sourceLeft = note->curVolLeft;
vol.sourceRight = note->curVolRight;
}
vol.targetLeft = note->targetVolLeft;
vol.targetRight = note->targetVolRight;
note->curVolLeft = vol.targetLeft;
note->curVolRight = vol.targetRight;
// 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.
#ifdef ENABLE_STEREO_HEADSET_EFFECTS
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);
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);
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);
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);
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);
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);
aSetBuffer(cmd++, A_AUX, DMEM_ADDR_RIGHT_CH, DMEM_ADDR_WET_LEFT_CH, DMEM_ADDR_WET_RIGHT_CH);
}
}
#else
aSetBuffer(cmd++, 0, inBuf, DMEM_ADDR_LEFT_CH, nSamples);
aSetBuffer(cmd++, A_AUX, DMEM_ADDR_RIGHT_CH, DMEM_ADDR_WET_LEFT_CH, DMEM_ADDR_WET_RIGHT_CH);
#endif
if (vol.targetLeft == vol.sourceLeft && vol.targetRight == vol.sourceRight
&& !note->envMixerNeedsInit) {
mixerFlags = A_CONTINUE;
} else {
mixerFlags = A_INIT;
// volume ramping
// This roughly computes 2^16 * (targetVol / sourceVol) ^ (8 / arg2),
// but with discretizations of targetVol, sourceVol and arg2.
rampLeft = currentRampingTableLeft[vol.targetLeft >> (15 - VOL_RAMPING_EXPONENT)] * currentRampingTableRight[vol.sourceLeft >> (15 - VOL_RAMPING_EXPONENT)];
rampRight = currentRampingTableLeft[vol.targetRight >> (15 - VOL_RAMPING_EXPONENT)] * currentRampingTableRight[vol.sourceRight >> (15 - VOL_RAMPING_EXPONENT)];
// 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 << 8);
}
#ifdef ENABLE_STEREO_HEADSET_EFFECTS
if (gSynthesisReverb.useReverb && note->reverbVol != 0) {
aEnvMixer(cmd++, mixerFlags | A_AUX,
VIRTUAL_TO_PHYSICAL2(note->synthesisBuffers->mixEnvelopeState));
if (note->stereoStrongRight) {
aSetBuffer(cmd++, 0, 0, 0, nSamples);
// 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);
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);
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);
aMix(cmd++, 0, /*gain*/ 0x8000, /*in*/ DMEM_ADDR_STEREO_STRONG_TEMP_DRY,
/*out*/ DMEM_ADDR_RIGHT_CH);
}
}
#else
if (gSynthesisReverb.useReverb && note->reverbVol != 0) {
mixerFlags |= A_AUX;
}
aEnvMixer(cmd++, mixerFlags, VIRTUAL_TO_PHYSICAL2(note->synthesisBuffers->mixEnvelopeState));
#endif
return cmd;
}
#ifdef ENABLE_STEREO_HEADSET_EFFECTS
u64 *note_apply_headset_pan_effects(u64 *cmd, struct Note *note, s32 bufLen, s32 flags, s32 leftRight) {
u16 dest;
u16 pitch;
u16 prevPanShift;
u16 panShift;
switch (leftRight) {
case 1:
dest = DMEM_ADDR_LEFT_CH;
panShift = note->headsetPanRight;
note->prevHeadsetPanLeft = 0;
prevPanShift = note->prevHeadsetPanRight;
note->prevHeadsetPanRight = panShift;
break;
case 2:
dest = DMEM_ADDR_RIGHT_CH;
panShift = note->headsetPanLeft;
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 + 16,
16); // 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) / (bufLen + panShift - 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 {
if (panShift == 0) {
pitch = (bufLen << 0xf) / (bufLen - prevPanShift - 4);
} else {
pitch = (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;
}
#endif
void note_init_volume(struct Note *note) {
note->targetVolLeft = 0;
note->targetVolRight = 0;
note->reverbVol = 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) {
f32 volLeft, volRight;
s32 panIndex = (s32)(pan * 127.5f) & 127;
if (gSoundMode == SOUND_MODE_MONO) {
volLeft = 0.707f;
volRight = 0.707f;
#ifdef ENABLE_STEREO_HEADSET_EFFECTS
} else 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;
#endif
} else {
volLeft = gDefaultPanVolume[panIndex];
volRight = gDefaultPanVolume[127 - panIndex];
}
if (velocity < 0) {
velocity = 0;
}
note->targetVolLeft = (u16)(s32)(velocity * volLeft) & VOLRAMPING_MASK;
note->targetVolRight = (u16)(s32)(velocity * volRight) & VOLRAMPING_MASK;
if (note->targetVolLeft == 0) {
note->targetVolLeft++;
}
if (note->targetVolRight == 0) {
note->targetVolRight++;
}
if (note->reverbVol != reverbVol) {
note->reverbVol = reverbVol;
note->envMixerNeedsInit = TRUE;
return;
}
note->envMixerNeedsInit = note->needsInit;
}
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;
#ifdef ENABLE_STEREO_HEADSET_EFFECTS
note->stereoStrongRight = FALSE;
note->stereoStrongLeft = FALSE;
note->usesHeadsetPanEffects = FALSE;
note->initFullVelocity = FALSE;
note->headsetPanLeft = 0;
note->headsetPanRight = 0;
note->prevHeadsetPanRight = 0;
note->prevHeadsetPanLeft = 0;
#endif
}
void note_disable(struct Note *note) {
if (note->needsInit) {
note->needsInit = FALSE;
} else {
note_set_vel_pan_reverb(note, 0, 0.5f, 0);
}
note->priority = NOTE_PRIORITY_DISABLED;
note->enabled = FALSE;
note->finished = FALSE;
note->parentLayer = NO_LAYER;
note->prevParentLayer = NO_LAYER;
}
#endif
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