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More optimizations to better reverb

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gheskett
2021-08-03 22:30:11 -05:00
parent 15f29f3814
commit 86fe86e2d6
3 changed files with 100 additions and 39 deletions
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5
src/audio/heap.c
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@@ -1235,7 +1235,7 @@ void audio_reset_session(void) {
if (gIsConsole)
reverbConsole = betterReverbConsoleDownsample; // Console!
else
reverbConsole = 2; // Setting this to 1 will crash unless you increase the better reverb buffer (in which case it will just freeze instead)
reverbConsole = 2; // Setting this to 1 is REALLY slow, please use sparingly!
if (reverbConsole <= 0) {
reverbConsole = 1;
@@ -1440,6 +1440,9 @@ void audio_reset_session(void) {
gSynthesisReverb.items[1][i].toDownsampleRight = mem + DEFAULT_LEN_1CH / sizeof(s16);
}
}
// This does not have to be reset after being initialized for the first time, which would speed up load times dramatically.
// However, reseting this allows for proper clearing of the reverb buffers, as well as dynamic customization of the delays array.
#ifdef BETTER_REVERB
if (consoleBetterReverb) {
for (i = 0; i < NUM_ALLPASS; ++i)

127
src/audio/synthesis.c
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@@ -54,13 +54,14 @@ s32 gReverbDrySignal = 0x26; // Amount of original input in final signal (large
// These values affect reverb delays, bigger values result in fatter echo (and more memory); must be cumulatively smaller than BETTER_REVERB_SIZE/8.
// If setting reverb downsample value to 1 (which currently does not work anyway), this must be BETTER_REVERB_SIZE/16.
// These values should never be changed unless in this declaration or during a call to audio_reset_session, or it could lead to a major memory leak or garbage audio.
u32 delays[NUM_ALLPASS] = {
// These values should also never be negative; these are just s32s to avoid typecasts
s32 delays[NUM_ALLPASS] = {
1080, 1352, 1200,
1384, 1048, 1352,
1200, 1232, 1432,
928, 1504, 1512
};
const u32 delaysBaseline[NUM_ALLPASS] = { // Like delays variable, but represent max values that never change (also probably somewhat redundant)
const s32 delaysBaseline[NUM_ALLPASS] = { // Like delays variable, but represent max values that never change (also probably somewhat redundant)
1080, 1352, 1200,
1384, 1048, 1352,
1200, 1232, 1432,
@@ -70,11 +71,12 @@ const s32 reverbMults[2][NUM_ALLPASS / 3] = { // These values affect reverb deca
{0xD2, 0x6E, 0x36, 0x1F},
{0x38, 0x26, 0xCF, 0x71}
};
u32 allpassIdx[2][NUM_ALLPASS] = {
s32 allpassIdx[2][NUM_ALLPASS] = {
{0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0},
{0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}
};
s32 tmpBuf[NUM_ALLPASS] = {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0};
s32 tmpBufL[NUM_ALLPASS] = {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0};
s32 tmpBufR[NUM_ALLPASS] = {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0};
s32 ***delayBufs;
u8 consoleBetterReverb = TRUE; // Do not change this line unless you know what you're doing; Please use the variable below instead.
@@ -121,26 +123,72 @@ inline s16 clamp16(s32 x) {
return (s16) x;
}
inline void reverb_samples(s16 *outSampleL, s16 *outSampleR, s32 inSampleL, s32 inSampleR) {
u32 i = 0;
s32 j = 0;
u8 k = 0;
s32 outTmpL = 0;
s32 outTmpR = 0;
s32 tmpCarryoverL = ((tmpBufL[NUM_ALLPASS-1] * gReverbRevIndex) / 256) + inSampleL; // Unique to left channel
s32 tmpCarryoverR = ((tmpBufR[NUM_ALLPASS-1] * gReverbRevIndex) / 256);
for (; i < NUM_ALLPASS; ++i, ++j) {
tmpBufL[i] = delayBufs[0][i][allpassIdx[0][i]];
tmpBufR[i] = delayBufs[1][i][allpassIdx[1][i]];
if (j == 2) {
j = -1;
outTmpL += (tmpBufL[i] * reverbMults[0][k]) / 256;
outTmpR += (tmpBufR[i] * reverbMults[1][k++]) / 256;
delayBufs[0][i][allpassIdx[0][i]] = tmpCarryoverL;
delayBufs[1][i][allpassIdx[1][i]] = tmpCarryoverR;
if (i != NUM_ALLPASS - 1) {
tmpCarryoverL = (tmpBufL[i] * gReverbRevIndex) / 256;
tmpCarryoverR = (tmpBufR[i] * gReverbRevIndex) / 256;
}
}
else {
delayBufs[0][i][allpassIdx[0][i]] = (tmpBufL[i] * (-gReverbGainIndex)) / 256 + tmpCarryoverL;
delayBufs[1][i][allpassIdx[1][i]] = (tmpBufR[i] * (-gReverbGainIndex)) / 256 + tmpCarryoverR;
if (i == 6)
delayBufs[1][i][allpassIdx[1][i]] += inSampleR; // Unique to right channel
tmpCarryoverL = (delayBufs[0][i][allpassIdx[0][i]] * gReverbGainIndex) / 256 + tmpBufL[i];
tmpCarryoverR = (delayBufs[1][i][allpassIdx[1][i]] * gReverbGainIndex) / 256 + tmpBufR[i];
}
if (++allpassIdx[0][i] == delays[i]) {
allpassIdx[0][i] = 0;
allpassIdx[1][i] = -1; // To avoid an else branch
}
++allpassIdx[1][i];
}
*outSampleL = clamp16((outTmpL * gReverbWetSignal + inSampleL * gReverbDrySignal) / 256);
*outSampleR = clamp16((outTmpR * gReverbWetSignal + inSampleR * gReverbDrySignal) / 256);
}
inline s16 reverb_sample_left(s32 inSample) {
u32 i = 0;
s32 j = 0;
u8 k = 0;
s32 outTmp = 0;
s32 tmpCarryover = ((tmpBuf[NUM_ALLPASS-1] * gReverbRevIndex) / 256) + inSample;
s32 tmpCarryover = ((tmpBufL[NUM_ALLPASS-1] * gReverbRevIndex) / 256) + inSample;
for (; i < NUM_ALLPASS; ++i, ++j) {
tmpBuf[i] = delayBufs[0][i][allpassIdx[0][i]];
tmpBufL[i] = delayBufs[0][i][allpassIdx[0][i]];
if (j == 2) {
j = -1;
outTmp += (tmpBuf[i] * reverbMults[0][k++]) / 256;
outTmp += (tmpBufL[i] * reverbMults[0][k++]) / 256;
delayBufs[0][i][allpassIdx[0][i]] = tmpCarryover;
if (i != NUM_ALLPASS - 1)
tmpCarryover = (tmpBuf[i] * gReverbRevIndex) / 256;
tmpCarryover = (tmpBufL[i] * gReverbRevIndex) / 256;
}
else {
delayBufs[0][i][allpassIdx[0][i]] = (tmpBuf[i] * (-gReverbGainIndex)) / 256 + tmpCarryover;
tmpCarryover = (delayBufs[0][i][allpassIdx[0][i]] * gReverbGainIndex) / 256 + tmpBuf[i];
delayBufs[0][i][allpassIdx[0][i]] = (tmpBufL[i] * (-gReverbGainIndex)) / 256 + tmpCarryover;
tmpCarryover = (delayBufs[0][i][allpassIdx[0][i]] * gReverbGainIndex) / 256 + tmpBufL[i];
}
if (++allpassIdx[0][i] == delays[i])
@@ -155,25 +203,25 @@ inline s16 reverb_sample_right(s32 inSample) {
s32 j = 0;
u8 k = 0;
s32 outTmp = 0;
s32 tmpCarryover = ((tmpBuf[NUM_ALLPASS-1] * gReverbRevIndex) / 256);
s32 tmpCarryover = ((tmpBufR[NUM_ALLPASS-1] * gReverbRevIndex) / 256);
for (; i < NUM_ALLPASS; ++i, ++j) {
tmpBuf[i] = delayBufs[1][i][allpassIdx[1][i]];
tmpBufR[i] = delayBufs[1][i][allpassIdx[1][i]];
if (j == 2) {
j = -1;
outTmp += (tmpBuf[i] * reverbMults[1][k++]) / 256;
outTmp += (tmpBufR[i] * reverbMults[1][k++]) / 256;
delayBufs[1][i][allpassIdx[1][i]] = tmpCarryover;
if (i != NUM_ALLPASS - 1)
tmpCarryover = (tmpBuf[i] * gReverbRevIndex) / 256;
tmpCarryover = (tmpBufR[i] * gReverbRevIndex) / 256;
}
else {
delayBufs[1][i][allpassIdx[1][i]] = (tmpBuf[i] * (-gReverbGainIndex)) / 256 + tmpCarryover;
delayBufs[1][i][allpassIdx[1][i]] = (tmpBufR[i] * (-gReverbGainIndex)) / 256 + tmpCarryover;
if (i == 6)
delayBufs[1][i][allpassIdx[1][i]] += inSample;
tmpCarryover = (delayBufs[1][i][allpassIdx[1][i]] * gReverbGainIndex) / 256 + tmpBuf[i];
tmpCarryover = (delayBufs[1][i][allpassIdx[1][i]] * gReverbGainIndex) / 256 + tmpBufR[i];
}
if (++allpassIdx[1][i] == delays[i])
@@ -294,26 +342,35 @@ void prepare_reverb_ring_buffer(s32 chunkLen, u32 updateIndex) {
item = &gSynthesisReverb.items[gSynthesisReverb.curFrame][updateIndex];
if (gReverbDownsampleRate != 1) {
osInvalDCache(item->toDownsampleLeft, DEFAULT_LEN_2CH);
for (srcPos = 0, dstPos = item->startPos; dstPos < item->lengthA / 2 + item->startPos;
srcPos += gReverbDownsampleRate, dstPos++) {
gSynthesisReverb.ringBuffer.left[dstPos] = reverb_sample_left(item->toDownsampleLeft[srcPos]);
gSynthesisReverb.ringBuffer.right[dstPos] = reverb_sample_right(item->toDownsampleRight[srcPos]);
}
for (dstPos = 0; dstPos < item->lengthB / 2; srcPos += gReverbDownsampleRate, dstPos++) {
gSynthesisReverb.ringBuffer.left[dstPos] = reverb_sample_left(item->toDownsampleLeft[srcPos]);
gSynthesisReverb.ringBuffer.right[dstPos] = reverb_sample_right(item->toDownsampleRight[srcPos]);
}
for (srcPos = 0, dstPos = item->startPos; dstPos < item->lengthA / 2 + item->startPos; srcPos += gReverbDownsampleRate, dstPos++)
reverb_samples(&gSynthesisReverb.ringBuffer.left[dstPos], &gSynthesisReverb.ringBuffer.right[dstPos], item->toDownsampleLeft[srcPos], item->toDownsampleRight[srcPos]);
for (dstPos = 0; dstPos < item->lengthB / 2; srcPos += gReverbDownsampleRate, dstPos++)
reverb_samples(&gSynthesisReverb.ringBuffer.left[dstPos], &gSynthesisReverb.ringBuffer.right[dstPos], item->toDownsampleLeft[srcPos], item->toDownsampleRight[srcPos]);
// for (srcPos = 0, dstPos = item->startPos; dstPos < item->lengthA / 2 + item->startPos;
// srcPos += gReverbDownsampleRate, dstPos++) {
// gSynthesisReverb.ringBuffer.left[dstPos] = reverb_sample_left(item->toDownsampleLeft[srcPos]);
// gSynthesisReverb.ringBuffer.right[dstPos] = reverb_sample_right(item->toDownsampleRight[srcPos]);
// }
// for (dstPos = 0; dstPos < item->lengthB / 2; srcPos += gReverbDownsampleRate, dstPos++) {
// gSynthesisReverb.ringBuffer.left[dstPos] = reverb_sample_left(item->toDownsampleLeft[srcPos]);
// gSynthesisReverb.ringBuffer.right[dstPos] = reverb_sample_right(item->toDownsampleRight[srcPos]);
// }
}
else { // Too slow for practical use?
for (srcPos = 0, dstPos = item->startPos; dstPos < item->lengthA / 2 + item->startPos;
srcPos += gReverbDownsampleRate, dstPos++) {
gSynthesisReverb.ringBuffer.left[dstPos] = reverb_sample_left(gSynthesisReverb.ringBuffer.left[dstPos]);
gSynthesisReverb.ringBuffer.right[dstPos] = reverb_sample_right(gSynthesisReverb.ringBuffer.right[dstPos]);
}
for (dstPos = 0; dstPos < item->lengthB / 2; srcPos += gReverbDownsampleRate, dstPos++) {
gSynthesisReverb.ringBuffer.left[dstPos] = reverb_sample_left(gSynthesisReverb.ringBuffer.left[dstPos]);
gSynthesisReverb.ringBuffer.right[dstPos] = reverb_sample_right(gSynthesisReverb.ringBuffer.right[dstPos]);
}
else { // Too slow for practical use, not recommended most of the time.
for (dstPos = item->startPos; dstPos < item->lengthA / 2 + item->startPos; dstPos++)
reverb_samples(&gSynthesisReverb.ringBuffer.left[dstPos], &gSynthesisReverb.ringBuffer.right[dstPos], gSynthesisReverb.ringBuffer.left[dstPos], gSynthesisReverb.ringBuffer.right[dstPos]);
for (dstPos = 0; dstPos < item->lengthB / 2; srcPos += gReverbDownsampleRate, dstPos++)
reverb_samples(&gSynthesisReverb.ringBuffer.left[dstPos], &gSynthesisReverb.ringBuffer.right[dstPos], gSynthesisReverb.ringBuffer.left[dstPos], gSynthesisReverb.ringBuffer.right[dstPos]);
// for (dstPos = item->startPos; dstPos < item->lengthA / 2 + item->startPos; dstPos++) {
// gSynthesisReverb.ringBuffer.left[dstPos] = reverb_sample_left(gSynthesisReverb.ringBuffer.left[dstPos]);
// gSynthesisReverb.ringBuffer.right[dstPos] = reverb_sample_right(gSynthesisReverb.ringBuffer.right[dstPos]);
// }
// for (dstPos = 0; dstPos < item->lengthB / 2; srcPos += gReverbDownsampleRate, dstPos++) {
// gSynthesisReverb.ringBuffer.left[dstPos] = reverb_sample_left(gSynthesisReverb.ringBuffer.left[dstPos]);
// gSynthesisReverb.ringBuffer.right[dstPos] = reverb_sample_right(gSynthesisReverb.ringBuffer.right[dstPos]);
// }
}
}
#endif

7
src/audio/synthesis.h
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@@ -18,15 +18,16 @@
#endif
#ifdef BETTER_REVERB
#define BETTER_REVERB_SIZE 0xF000 // Size of all delaysBaseline values * 8 (plus array pointers)
#define BETTER_REVERB_SIZE 0xF200 // Size of all delaysBaseline values * 8 / 2^downsampleFactor (plus array pointers)
// #define BETTER_REVERB_SIZE 0x1E200 // For use with no downsampling (Warning: very slow!)
#else
#define BETTER_REVERB_SIZE 0
#endif
#define NUM_ALLPASS 12
extern const u32 delaysBaseline[NUM_ALLPASS];
extern u32 delays[NUM_ALLPASS];
extern const s32 delaysBaseline[NUM_ALLPASS];
extern s32 delays[NUM_ALLPASS];
extern s32 ***delayBufs;
extern u8 consoleBetterReverb;