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Merge branch 'master' into puppycamera2

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Fazana
2021-08-04 15:19:04 +01:00
parent e117bdf476 49715e5204
commit 0a816814a5
8 changed files with 334 additions and 8 deletions
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1
README.md
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@@ -75,6 +75,7 @@ This is a fork of the ultrasm64 repo by CrashOveride which includes the followin
- ia8 coins (64x64), the vanilla coin texture is upgraded to accomodate. *
- Skybox size modifier. You can have 2x, 3x and 4x size skyboxes (you can select the skybox size in `config.h`.) Please note that this might affect console performance, especially 4x mode. 2x or 3x mode is recommended if aiming for console. By CowQuack *
- You can set the black border size to different values for console and emulator. It's set to 0 by default for both. *
- This repo supports much better implementation of reverb over vanilla's fake echo reverb. Great for caves or eerie levels, as well as just a better audio experience in general. See `audio/synthesis.c` for more configuration info. (By ArcticJaguar725) *
# UltraSM64

3
include/config.h
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@@ -153,6 +153,9 @@
#define DISABLE_AA
// Makes the coins ia8 64x64 instead of ia16 32x32. Uses new ia8 textures so that vanilla coins look better.
#define IA8_COINS
// Use a much better implementation of reverb over vanilla's fake echo reverb. Great for caves or eerie levels, as well as just a better audio experience in general.
// Reverb parameters can be configured in audio/synthesis.c to meet desired aesthetic/performance needs.
//#define BETTER_REVERB
// If you want to change the extended boundaries mode, go to engine/extended_bounds.h and change EXTENDED_BOUNDS_MODE

5
src/audio/data.h
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@@ -5,6 +5,7 @@
#include "internal.h"
#include "types.h"
#include "synthesis.h"
#define AUDIO_LOCK_UNINITIALIZED 0
#define AUDIO_LOCK_NOT_LOADING 0x76557364
@@ -150,11 +151,11 @@ extern OSMesgQueue *D_SH_80350FA8;
#if defined(VERSION_EU) || defined(VERSION_SH)
#define UNUSED_COUNT_80333EE8 24
#define AUDIO_HEAP_SIZE (0x2c500 + EXT_AUDIO_HEAP_SIZE + EXT_AUDIO_INIT_POOL_SIZE)
#define AUDIO_HEAP_SIZE (0x2c500 + EXT_AUDIO_HEAP_SIZE + EXT_AUDIO_INIT_POOL_SIZE + BETTER_REVERB_SIZE)
#define AUDIO_INIT_POOL_SIZE (0x2c00 + EXT_AUDIO_INIT_POOL_SIZE)
#else
#define UNUSED_COUNT_80333EE8 16
#define AUDIO_HEAP_SIZE (0x31150 + EXT_AUDIO_HEAP_SIZE + EXT_AUDIO_INIT_POOL_SIZE)
#define AUDIO_HEAP_SIZE (0x31150 + EXT_AUDIO_HEAP_SIZE + EXT_AUDIO_INIT_POOL_SIZE + BETTER_REVERB_SIZE)
#define AUDIO_INIT_POOL_SIZE (0x2500 + EXT_AUDIO_INIT_POOL_SIZE)
#endif

47
src/audio/heap.c
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@@ -6,6 +6,7 @@
#include "synthesis.h"
#include "seqplayer.h"
#include "effects.h"
#include "game/game_init.h"
#define ALIGN16(val) (((val) + 0xF) & ~0xF)
@@ -1122,6 +1123,9 @@ void audio_reset_session(void) {
#if defined(VERSION_JP) || defined(VERSION_US)
s32 frames;
s32 remainingDmas;
#ifdef BETTER_REVERB
s8 reverbConsole;
#endif
#else
struct SynthesisReverb *reverb;
#endif
@@ -1234,6 +1238,29 @@ void audio_reset_session(void) {
gMaxSimultaneousNotes = preset->maxSimultaneousNotes;
gSamplesPerFrameTarget = ALIGN16(gAiFrequency / 60);
gReverbDownsampleRate = preset->reverbDownsampleRate;
#ifdef BETTER_REVERB
if (gIsConsole)
reverbConsole = betterReverbConsoleDownsample; // Console!
else
reverbConsole = betterReverbEmulatorDownsample; // Setting this to 1 is REALLY slow, please use sparingly!
if (reverbConsole <= 0) {
reverbConsole = 1;
toggleBetterReverb = FALSE;
}
else {
toggleBetterReverb = TRUE;
}
if (toggleBetterReverb && betterReverbWindowsSize >= 0)
reverbWindowSize = betterReverbWindowsSize;
if (gReverbDownsampleRate < (1 << (reverbConsole - 1)))
gReverbDownsampleRate = (1 << (reverbConsole - 1));
reverbWindowSize /= gReverbDownsampleRate;
if (reverbWindowSize < DEFAULT_LEN_2CH) // Minimum window size to not overflow
reverbWindowSize = DEFAULT_LEN_2CH;
#endif
switch (gReverbDownsampleRate) {
case 1:
@@ -1255,7 +1282,6 @@ void audio_reset_session(void) {
sReverbDownsampleRateLog = 0;
}
gReverbDownsampleRate = preset->reverbDownsampleRate;
gVolume = preset->volume;
gMinAiBufferLength = gSamplesPerFrameTarget - 0x10;
updatesPerFrame = gSamplesPerFrameTarget / 160 + 1;
@@ -1283,7 +1309,7 @@ void audio_reset_session(void) {
temporaryMem = DOUBLE_SIZE_ON_64_BIT(preset->temporaryBankMem + preset->temporarySeqMem);
#endif
totalMem = persistentMem + temporaryMem;
wantMisc = gAudioSessionPool.size - totalMem - 0x100;
wantMisc = gAudioSessionPool.size - totalMem - 0x100 - BETTER_REVERB_SIZE;
sSessionPoolSplit.wantSeq = wantMisc;
sSessionPoolSplit.wantCustom = totalMem;
session_pools_init(&sSessionPoolSplit);
@@ -1426,6 +1452,23 @@ 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 (toggleBetterReverb) {
for (i = 0; i < NUM_ALLPASS; ++i)
delays[i] = delaysBaseline[i] / gReverbDownsampleRate;
delayBufs = (s32***) soundAlloc(&gAudioSessionPool, 2 * sizeof(s32**));
delayBufs[0] = (s32**) soundAlloc(&gAudioSessionPool, NUM_ALLPASS * sizeof(s32*));
delayBufs[1] = (s32**) soundAlloc(&gAudioSessionPool, NUM_ALLPASS * sizeof(s32*));
for (i = 0; i < NUM_ALLPASS; ++i) {
delayBufs[0][i] = (s32*) soundAlloc(&gAudioSessionPool, delays[i] * sizeof(s32));
delayBufs[1][i] = (s32*) soundAlloc(&gAudioSessionPool, delays[i] * sizeof(s32));
}
}
#endif
}
#endif

261
src/audio/synthesis.c
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@@ -37,6 +37,83 @@
#define AUDIO_ALIGN(val, amnt) (((val) + (1 << amnt) - 1) & ~((1 << amnt) - 1))
/* -------------------------------------------------------BEGIN REVERB PARAMETERS-------------------------------------------------------------- */
/**
* This reverb is a much more natural, ambient implementation over vanilla's, though at the cost of some memory and performance.
* These parameters are here to provide maximum control over the usage of the reverb effect, as well as with game performance.
*
* To take advantage of the reverb effect, you can change the echo parameters set in levels/level_defines.h to tailor the reverb to each specific level area.
* To adjust reverb presence with individual sound effects, apply the .set_reverb command within sound/sequences/00_sound_player.s (see examples of other sounds that use it).
* To use with M64 sequences, set the Effect parameter for each channel accordingly (CC 91 for MIDI files).
*/
s32 gReverbRevIndex = 0x7F; // Affects decay time mostly; can be messed with at any time (and also probably the most useful parameter here)
s32 gReverbGainIndex = 0xA3; // Affects signal immediately retransmitted back into buffers; can be messed with at any time
s32 gReverbWetSignal = 0xEF; // Amount of reverb specific output in final signal; can be messed with at any time
s32 gReverbDrySignal = 0x00; // Amount of original input in final signal (large values can cause terrible feedback!); can be messed with at any time
// This value controls the size of the reverb buffer. It affects the global reverb delay time. This is probably the easiest parameter to control usefully.
// It is not recommended setting this to values greater than 0x1000 * 2^(downsample factor - 1), as you run the risk of running into a memory issue (though this is far from a guarantee).
// Setting the value lower than the downsample buffer size will destroy the game audio, so this is taken into account automatically but may limit the lower possibilities.
// If this value is changed, it will go into effect the next time audio_reset_session is called.
// Set to -1 to use a default preset instead. Higher values represent more audio delay (usually better for echoey spaces).
s32 betterReverbWindowsSize = -1;
// These values affect filter delays. Bigger values will result in fatter echo (and more memory); must be cumulatively smaller than BETTER_REVERB_SIZE/4.
// If setting a reverb downsample value to 1, this must be smaller than BETTER_REVERB_SIZE/8.
// These values should never be changed unless in this declaration or during a call to audio_reset_session, as it could otherwise lead to a major memory leak or garbage audio.
// None of the delay values should ever be smaller than 1 either; these are s32s purely to avoid typecasts.
// These values are currently set by using delaysBaseline in the audio_reset_session function, so its behavior must be overridden to use dynamically (or at all).
s32 delays[NUM_ALLPASS] = {
1080, 1352, 1200,
1384, 1048, 1352,
1200, 1232, 1432,
928, 1504, 1512
};
// Like the delays array, but represents default max values that don't change (also probably somewhat redundant)
// Change this array rather than the delays array to customize reverb delay times globally.
// Similarly to delays, these should be kept within the memory constraints defined by BETTER_REVERB_SIZE.
const s32 delaysBaseline[NUM_ALLPASS] = {
1080, 1352, 1200,
1384, 1048, 1352,
1200, 1232, 1432,
928, 1504, 1512
};
// These values affect reverb decay depending on the filter index; can be messed with at any time
const s32 reverbMults[2][NUM_ALLPASS / 3] = {
{0xD2, 0x6E, 0x36, 0x1F}, // Left Channel
{0x38, 0x26, 0xCF, 0x71} // Right Channel
};
// Setting this to 4 corrupts the game, so set this value to -1 to use vanilla reverb if this is too slow, or if it just doesn't fit the desired aesthetic of a level.
// You can change this value before audio_reset_session gets called if different levels can tolerate the demand better than others or just have different reverb goals.
// A higher downsample value hits the game's frequency limit sooner, which can cause the reverb sometimes to be off pitch. This is a vanilla level issue (and also counter intuitive).
// Higher downsample values also result in slightly shorter reverb decay times.
s8 betterReverbConsoleDownsample = 3;
// Most emulators can handle a default value of 2, but 3 may be advisable in some cases if targeting older emulators (e.g. PJ64 1.6). Setting this to -1 also uses vanilla reverb.
// Using a value of 1 is not recommended except in very specific situations. If you do decide to use 1 here, you must adjust BETTER_REVERB_SIZE appropriately.
// You can change this value before audio_reset_session gets called if different levels can tolerate the demand better than others or just have different reverb goals.
// A higher downsample value hits the game's frequency limit sooner, which can cause the reverb sometimes to be off pitch. This is a vanilla level issue (and also counter intuitive).
// Higher downsample values also result in slightly shorter reverb decay times.
s8 betterReverbEmulatorDownsample = 2;
/* --------------------------------------------------------END REVERB PARAMETERS--------------------------------------------------------------- */
// Do not touch these values.
u8 toggleBetterReverb = TRUE;
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 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;
struct VolumeChange {
u16 sourceLeft;
u16 sourceRight;
@@ -70,6 +147,123 @@ struct SynthesisReverb gSynthesisReverb;
u8 sAudioSynthesisPad[0x20];
#endif
inline s16 clamp16(s32 x) {
if (x >= 32767)
return 32767;
if (x <= -32768)
return -32768;
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 = ((tmpBufL[NUM_ALLPASS-1] * gReverbRevIndex) / 256) + inSample;
for (; i < NUM_ALLPASS; ++i, ++j) {
tmpBufL[i] = delayBufs[0][i][allpassIdx[0][i]];
if (j == 2) {
j = -1;
outTmp += (tmpBufL[i] * reverbMults[0][k++]) / 256;
delayBufs[0][i][allpassIdx[0][i]] = tmpCarryover;
if (i != NUM_ALLPASS - 1)
tmpCarryover = (tmpBufL[i] * gReverbRevIndex) / 256;
}
else {
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])
allpassIdx[0][i] = 0;
}
return clamp16((outTmp * gReverbWetSignal + inSample * gReverbDrySignal) / 256);
}
inline s16 reverb_sample_right(s32 inSample) {
u32 i = 0;
s32 j = 0;
u8 k = 0;
s32 outTmp = 0;
s32 tmpCarryover = ((tmpBufR[NUM_ALLPASS-1] * gReverbRevIndex) / 256);
for (; i < NUM_ALLPASS; ++i, ++j) {
tmpBufR[i] = delayBufs[1][i][allpassIdx[1][i]];
if (j == 2) {
j = -1;
outTmp += (tmpBufR[i] * reverbMults[1][k++]) / 256;
delayBufs[1][i][allpassIdx[1][i]] = tmpCarryover;
if (i != NUM_ALLPASS - 1)
tmpCarryover = (tmpBufR[i] * gReverbRevIndex) / 256;
}
else {
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 + tmpBufR[i];
}
if (++allpassIdx[1][i] == delays[i])
allpassIdx[1][i] = 0;
}
return clamp16((outTmp * gReverbWetSignal + inSample * gReverbDrySignal) / 256);
}
#ifdef VERSION_EU
s16 gVolume;
s8 gUseReverb;
@@ -149,7 +343,12 @@ void prepare_reverb_ring_buffer(s32 chunkLen, u32 updateIndex) {
s32 nSamples;
s32 numSamplesAfterDownsampling;
s32 excessiveSamples;
#ifdef BETTER_REVERB
if (!toggleBetterReverb && gReverbDownsampleRate != 1) {
#else
if (gReverbDownsampleRate != 1) {
#endif
if (gSynthesisReverb.framesLeftToIgnore == 0) {
// Now that the RSP has finished, downsample the samples produced two frames ago by skipping
// samples.
@@ -171,6 +370,68 @@ void prepare_reverb_ring_buffer(s32 chunkLen, u32 updateIndex) {
}
}
}
#ifdef BETTER_REVERB
else if (toggleBetterReverb) {
item = &gSynthesisReverb.items[gSynthesisReverb.curFrame][updateIndex];
if (gSoundMode == SOUND_MODE_MONO) {
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(((s32) item->toDownsampleLeft[srcPos] + (s32) item->toDownsampleRight[srcPos]) / 2);
gSynthesisReverb.ringBuffer.right[dstPos] = gSynthesisReverb.ringBuffer.left[dstPos];
}
for (dstPos = 0; dstPos < item->lengthB / 2; srcPos += gReverbDownsampleRate, dstPos++) {
gSynthesisReverb.ringBuffer.left[dstPos] = reverb_sample_left(((s32) item->toDownsampleLeft[srcPos] + (s32) item->toDownsampleRight[srcPos]) / 2);
gSynthesisReverb.ringBuffer.right[dstPos] = gSynthesisReverb.ringBuffer.left[dstPos];
}
}
else { // Too slow for practical use, not recommended most of the time.
for (dstPos = item->startPos; dstPos < item->lengthA / 2 + item->startPos; dstPos++) {
gSynthesisReverb.ringBuffer.left[dstPos] = reverb_sample_left(((s32) gSynthesisReverb.ringBuffer.left[dstPos] + (s32) gSynthesisReverb.ringBuffer.right[dstPos]) / 2);
gSynthesisReverb.ringBuffer.right[dstPos] = gSynthesisReverb.ringBuffer.left[dstPos];
}
for (dstPos = 0; dstPos < item->lengthB / 2; dstPos++) {
gSynthesisReverb.ringBuffer.left[dstPos] = reverb_sample_left(((s32) gSynthesisReverb.ringBuffer.left[dstPos] + (s32) gSynthesisReverb.ringBuffer.right[dstPos]) / 2);
gSynthesisReverb.ringBuffer.right[dstPos] = gSynthesisReverb.ringBuffer.left[dstPos];
}
}
}
else {
if (gReverbDownsampleRate != 1) {
osInvalDCache(item->toDownsampleLeft, DEFAULT_LEN_2CH);
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, 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; 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
item = &gSynthesisReverb.items[gSynthesisReverb.curFrame][updateIndex];
numSamplesAfterDownsampling = chunkLen / gReverbDownsampleRate;

18
src/audio/synthesis.h
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@@ -17,6 +17,24 @@
#define MAX_UPDATES_PER_FRAME 4
#endif
#ifdef BETTER_REVERB
#define BETTER_REVERB_SIZE 0xF200 // Size determined by ((all delaysBaseline values * 16) / (2 ^ Minimum Downsample Factor)) + array pointers; can be increased if needed
// #define BETTER_REVERB_SIZE 0x1E200 // For use with a downsampling value of 1 (i.e. no downsampling at all)
#else
#define BETTER_REVERB_SIZE 0
#endif
#define NUM_ALLPASS 12 // Number of delay filters to use with better reverb; do not change this value if you don't know what you're doing.
extern s32 betterReverbWindowsSize;
extern const s32 delaysBaseline[NUM_ALLPASS];
extern s32 delays[NUM_ALLPASS];
extern s32 ***delayBufs;
extern u8 toggleBetterReverb;
extern s8 betterReverbConsoleDownsample;
extern s8 betterReverbEmulatorDownsample;
struct ReverbRingBufferItem
{
s16 numSamplesAfterDownsampling;

3
src/buffers/buffers.c
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@@ -7,6 +7,7 @@
#include <hvqm/hvqm.h>
#endif
#include "config.h"
#include "audio/synthesis.h"
ALIGNED8 u8 gDecompressionHeap[0xD000];
#if defined(VERSION_EU)
@@ -14,7 +15,7 @@ ALIGNED16 u8 gAudioHeap[DOUBLE_SIZE_ON_64_BIT(0x31200 + EXT_AUDIO_HEAP_SIZE + EX
#elif defined(VERSION_SH)
ALIGNED16 u8 gAudioHeap[DOUBLE_SIZE_ON_64_BIT(0x31200 + EXT_AUDIO_HEAP_SIZE + EXT_AUDIO_INIT_POOL_SIZE) - 0x4800];
#else
ALIGNED16 u8 gAudioHeap[DOUBLE_SIZE_ON_64_BIT(0x31200 + EXT_AUDIO_HEAP_SIZE + EXT_AUDIO_INIT_POOL_SIZE)];
ALIGNED16 u8 gAudioHeap[DOUBLE_SIZE_ON_64_BIT(0x31200 + EXT_AUDIO_HEAP_SIZE + EXT_AUDIO_INIT_POOL_SIZE + BETTER_REVERB_SIZE)];
#endif
ALIGNED8 u8 gIdleThreadStack[0x800];

4
src/game/game_init.c
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@@ -45,7 +45,7 @@ struct GfxPool *gGfxPool;
OSContStatus gControllerStatuses[4];
OSContPad gControllerPads[4];
u8 gControllerBits;
u8 gIsConsole;
u8 gIsConsole = TRUE; // Needs to be initialized before audio_reset_session is called
u8 gBorderHeight;
#ifdef EEP
s8 gEepromProbe;
@@ -77,11 +77,9 @@ UNUSED static s32 sUnusedGameInitValue = 0;
// General timer that runs as the game starts
u32 gGlobalTimer = 0;
u8 gIsConsole;
#ifdef WIDE
u8 gWidescreen;
#endif
u8 gBorderHeight;
// Framebuffer rendering values (max 3)
u16 sRenderedFramebuffer = 0;