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Bugfix, refactor, and slightly improve BETTER_REVERB runtime (#391)
* Buxfix, refactor, and slightly improve BETTER_REVERB runtime * Update BETTER_REVERB defaults and add some more customizability * Improve BETTER_REVERB runtime even further * Rename some reverb variables to make more sense in context
This commit is contained in:
Gregory Heskett
GitHub
@@ -9,6 +9,11 @@
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*/
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#define CASTLE_MUSIC_FIX
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/**
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* Do not restart the music on cap grabs
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*/
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#define PERSISTENT_CAP_MUSIC
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/**
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* Increase audio heap size to allow for larger/more custom sequences/banks/sfx to be imported without causing issues (not supported for SH).
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*/
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@@ -28,8 +33,3 @@
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* Reverb parameters can be configured in audio/synthesis.c to meet desired aesthetic/performance needs. Currently US/JP only. Hurts emulator and console performance.
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*/
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// #define BETTER_REVERB
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/**
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* Do not restart the music on cap grabs
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*/
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#define PERSISTENT_CAP_MUSIC
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@@ -1059,7 +1059,6 @@ void init_reverb_us(s32 presetId) {
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s32 i;
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#ifdef BETTER_REVERB
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s8 reverbConsole;
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s32 bufOffset = 0;
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#endif
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s32 reverbWindowSize = gReverbSettings[presetId].windowSize;
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@@ -1112,9 +1111,7 @@ void init_reverb_us(s32 presetId) {
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gSynthesisReverb.items[1][i].toDownsampleRight = (mem + (DEFAULT_LEN_1CH / sizeof(s16)));
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}
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#ifdef BETTER_REVERB
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delayBufsL = (s32**) soundAlloc(&gBetterReverbPool, BETTER_REVERB_PTR_SIZE);
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delayBufsR = &delayBufsL[NUM_ALLPASS];
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delayBufsL[0] = (s32*) soundAlloc(&gBetterReverbPool, BETTER_REVERB_SIZE - BETTER_REVERB_PTR_SIZE);
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initialize_better_reverb_buffers();
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#endif
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} else {
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bzero(gSynthesisReverb.ringBuffer.left, (REVERB_WINDOW_SIZE_MAX * 2 * sizeof(s16)));
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@@ -1146,17 +1143,10 @@ void init_reverb_us(s32 presetId) {
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#ifdef BETTER_REVERB
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if (toggleBetterReverb) {
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if (sAudioIsInitialized) {
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bzero(delayBufsL[0], (BETTER_REVERB_SIZE - BETTER_REVERB_PTR_SIZE));
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}
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for (i = 0; i < NUM_ALLPASS; ++i) {
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delaysL[i] = (delaysBaselineL[i] / gReverbDownsampleRate);
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delaysR[i] = (delaysBaselineR[i] / gReverbDownsampleRate);
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delayBufsL[i] = (s32*) &delayBufsL[0][bufOffset];
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bufOffset += delaysL[i];
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delayBufsR[i] = (s32*) &delayBufsL[0][bufOffset]; // L and R buffers are interpolated adjacently in memory; not a bug
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bufOffset += delaysR[i];
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clear_better_reverb_buffers();
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}
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set_better_reverb_buffers();
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}
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#endif
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}
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@@ -93,6 +93,9 @@ extern struct SoundAllocPool gAudioInitPool;
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extern struct SoundAllocPool gNotesAndBuffersPool;
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extern struct SoundAllocPool gPersistentCommonPool;
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extern struct SoundAllocPool gTemporaryCommonPool;
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#ifdef BETTER_REVERB
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extern struct SoundAllocPool gBetterReverbPool;
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#endif
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extern struct SoundMultiPool gSeqLoadedPool;
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extern struct SoundMultiPool gBankLoadedPool;
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#ifdef VERSION_SH
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@@ -69,7 +69,7 @@
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// 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.
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// 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).
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// Higher downsample values also result in slightly shorter reverb decay times.
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s8 betterReverbDownsampleConsole = 3;
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s8 betterReverbDownsampleConsole = 2;
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// 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.
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// Using a value of 1 is not recommended on emulator unless reducing other parameters to compensate. If you do decide to use 1 here, you must adjust BETTER_REVERB_SIZE appropriately.
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@@ -80,22 +80,22 @@ s8 betterReverbDownsampleEmulator = 2;
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// This value represents the number of filters to use with the reverb. This can be decreased to improve performance, but at the cost of a lesser presence of reverb in the final audio.
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// Filter count should always be a multiple of 3. Never ever set this value to be greater than NUM_ALLPASS.
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// Setting it to anything less than 3 will disable reverb outright.
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// This value cannot be less than 3. Setting it to anything lower will act as if it was set to 3.
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// This can be changed at any time, but is best set immediately before calling audio_reset_session.
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u32 reverbFilterCountConsole = (NUM_ALLPASS - 6);
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s32 reverbFilterCountConsole = (NUM_ALLPASS - 9);
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// This value represents the number of filters to use with the reverb. This can be decreased to improve performance, but at the cost of a lesser presence of reverb in the final audio.
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// Filter count should always be a multiple of 3. Never ever set this value to be greater than NUM_ALLPASS.
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// Setting it to anything less than 3 will disable reverb outright.
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// This value cannot be less than 3. Setting it to anything lower will act as if it was set to 3.
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// This can be changed at any time, but is best set immediately before calling audio_reset_session.
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u32 reverbFilterCountEmulator = NUM_ALLPASS;
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s32 reverbFilterCountEmulator = NUM_ALLPASS;
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// Set this to TRUE to use mono over stereo for reverb. This should increase performance, but at the cost of a less fullfilling reverb experience.
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// Set this to TRUE to use mono over stereo for reverb. This should increase performance, but at the cost of a less fulfilling reverb experience.
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// If performance is desirable, it is recommended to change reverbFilterCountConsole or betterReverbDownsampleConsole first.
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// This can be changed at any time, but is best set immediately before calling audio_reset_session.
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u8 monoReverbConsole = FALSE;
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// Set this to TRUE to use mono over stereo for reverb. This should increase performance, but at the cost of a less fullfilling reverb experience.
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// Set this to TRUE to use mono over stereo for reverb. This should increase performance, but at the cost of a less fulfilling reverb experience.
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// If performance is desirable, it is recommended to change reverbFilterCountEmulator or betterReverbDownsampleEmulator first.
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// This can be changed at any time, but is best set immediately before calling audio_reset_session.
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u8 monoReverbEmulator = FALSE;
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@@ -111,7 +111,7 @@ s32 betterReverbWindowsSize = -1;
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// Setting these to values larger than 0xFF (255) or less than 0 may cause issues and is not recommended.
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#define REVERB_REV_INDEX 0x60 // Affects decay time mostly (large values can cause terrible feedback!); can be messed with at any time
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#define REVERB_GAIN_INDEX 0xA0 // Affects signal immediately retransmitted back into buffers (mid-high values yield the strongest effect); can be messed with at any time
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#define REVERB_WET_SIGNAL 0xE8 // Amount of reverb specific output in final signal (also affects decay); can be messed with at any time, also very easy to control
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#define REVERB_WET_SIGNAL 0xE0 // Amount of reverb specific output in final signal (also affects decay); can be messed with at any time, also very easy to control
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// #define REVERB_DRY_SIGNAL = 0x00; // Amount of original input in final signal (large values can cause terrible feedback!); declaration and uses commented out by default to improve compiler optimization
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@@ -136,26 +136,25 @@ s32 delaysBaselineR[NUM_ALLPASS] = {
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1200, 1432, 1232
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};
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// These values affect reverb decay depending on the filter index; can be messed with at any time
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s32 reverbMultsL[NUM_ALLPASS / 3] = {0xD7, 0x6F, 0x36, 0x22};
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s32 reverbMultsR[NUM_ALLPASS / 3] = {0xCF, 0x73, 0x38, 0x1F};
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// These values affect reverb decay depending on the filter index; can be messed with at any time, and will have effects updated in real time
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s32 gReverbMultsL[NUM_ALLPASS / 3] = {0xD7, 0x6F, 0x36, 0x22};
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s32 gReverbMultsR[NUM_ALLPASS / 3] = {0xCF, 0x73, 0x38, 0x1F};
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/* -----------------------------------------------------------------------END REVERB PARAMETERS----------------------------------------------------------------------- */
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// Do not touch these values manually, unless you want potential for problems.
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s32 reverbFilterCount = NUM_ALLPASS;
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s32 reverbFilterCountm1 = (NUM_ALLPASS - 1);
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u8 monoReverb = FALSE;
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u8 toggleBetterReverb = TRUE;
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s32 allpassIdxL[NUM_ALLPASS] = {0};
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s32 allpassIdxR[NUM_ALLPASS] = {0};
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s32 tmpBufL[NUM_ALLPASS] = {0};
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s32 tmpBufR[NUM_ALLPASS] = {0};
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s32 delaysL[NUM_ALLPASS] = {0};
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s32 delaysR[NUM_ALLPASS] = {0};
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s32 **delayBufsL;
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s32 **delayBufsR;
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static u8 monoReverb = FALSE;
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static s32 reverbFilterCount = NUM_ALLPASS;
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static s32 allpassIdxL[NUM_ALLPASS] = {0};
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static s32 allpassIdxR[NUM_ALLPASS] = {0};
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static s32 delaysL[NUM_ALLPASS] = {0};
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static s32 delaysR[NUM_ALLPASS] = {0};
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static u8 reverbMultsL[NUM_ALLPASS / 3] = {0};
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static u8 reverbMultsR[NUM_ALLPASS / 3] = {0};
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static s32 **delayBufsL;
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static s32 **delayBufsR;
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#endif
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@@ -193,72 +192,109 @@ u8 sAudioSynthesisPad[0x20];
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#endif
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#ifdef BETTER_REVERB
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static inline void reverb_samples(s16 *outSampleL, s16 *outSampleR, s32 inSampleL, s32 inSampleR) {
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static void reverb_samples(s16 *outSampleL, s16 *outSampleR, s32 inSampleL, s32 inSampleR) {
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s32 *curDelaySampleL;
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s32 *curDelaySampleR;
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s32 historySampleL;
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s32 historySampleR;
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s32 i = 0;
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s32 j = 0;
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s32 k = 0;
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s32 outTmpL = 0;
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s32 outTmpR = 0;
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s32 tmpCarryoverL = (((tmpBufL[reverbFilterCountm1] * REVERB_REV_INDEX) / 256) + inSampleL);
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s32 tmpCarryoverR = (((tmpBufR[reverbFilterCountm1] * REVERB_REV_INDEX) / 256) + inSampleR);
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s32 tmpCarryoverL = (((delayBufsL[reverbFilterCount][allpassIdxL[reverbFilterCount]] * REVERB_REV_INDEX) >> 8) + inSampleL);
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s32 tmpCarryoverR = (((delayBufsR[reverbFilterCount][allpassIdxR[reverbFilterCount]] * REVERB_REV_INDEX) >> 8) + inSampleR);
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for (; i != reverbFilterCount; ++i, ++j) {
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tmpBufL[i] = delayBufsL[i][allpassIdxL[i]];
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tmpBufR[i] = delayBufsR[i][allpassIdxR[i]];
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for (; i <= reverbFilterCount; ++i, ++j) {
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curDelaySampleL = &delayBufsL[i][allpassIdxL[i]];
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curDelaySampleR = &delayBufsR[i][allpassIdxR[i]];
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historySampleL = *curDelaySampleL;
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historySampleR = *curDelaySampleR;
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if (j == 2) {
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j = -1;
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outTmpL += ((tmpBufL[i] * reverbMultsL[k ]) / 256);
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outTmpR += ((tmpBufR[i] * reverbMultsR[k++]) / 256);
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delayBufsL[i][allpassIdxL[i]] = tmpCarryoverL;
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delayBufsR[i][allpassIdxR[i]] = tmpCarryoverR;
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if (i != reverbFilterCountm1) {
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tmpCarryoverL = ((tmpBufL[i] * REVERB_REV_INDEX) / 256);
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tmpCarryoverR = ((tmpBufR[i] * REVERB_REV_INDEX) / 256);
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outTmpL += ((historySampleL * reverbMultsL[k ]) >> 8);
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outTmpR += ((historySampleR * reverbMultsR[k++]) >> 8);
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*curDelaySampleL = tmpCarryoverL;
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*curDelaySampleR = tmpCarryoverR;
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if (i != reverbFilterCount) {
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tmpCarryoverL = ((historySampleL * REVERB_REV_INDEX) >> 8);
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tmpCarryoverR = ((historySampleR * REVERB_REV_INDEX) >> 8);
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}
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} else {
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delayBufsL[i][allpassIdxL[i]] = (((tmpBufL[i] * (-REVERB_GAIN_INDEX)) / 256) + tmpCarryoverL);
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delayBufsR[i][allpassIdxR[i]] = (((tmpBufR[i] * (-REVERB_GAIN_INDEX)) / 256) + tmpCarryoverR);
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tmpCarryoverL = (((delayBufsL[i][allpassIdxL[i]] * REVERB_GAIN_INDEX) / 256) + tmpBufL[i]);
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tmpCarryoverR = (((delayBufsR[i][allpassIdxR[i]] * REVERB_GAIN_INDEX) / 256) + tmpBufR[i]);
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*curDelaySampleL = (((historySampleL * (-REVERB_GAIN_INDEX)) >> 8) + tmpCarryoverL);
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*curDelaySampleR = (((historySampleR * (-REVERB_GAIN_INDEX)) >> 8) + tmpCarryoverR);
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tmpCarryoverL = (((*curDelaySampleL * REVERB_GAIN_INDEX) >> 8) + historySampleL);
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tmpCarryoverR = (((*curDelaySampleR * REVERB_GAIN_INDEX) >> 8) + historySampleR);
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}
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if (++allpassIdxL[i] == delaysL[i]) allpassIdxL[i] = 0;
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if (++allpassIdxR[i] == delaysR[i]) allpassIdxR[i] = 0;
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}
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s32 outUnclamped = ((outTmpL * REVERB_WET_SIGNAL/* + inSampleL * REVERB_DRY_SIGNAL*/) / 256);
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s32 outUnclamped = ((outTmpL * REVERB_WET_SIGNAL/* + inSampleL * REVERB_DRY_SIGNAL*/) >> 8);
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*outSampleL = CLAMP_S16(outUnclamped);
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outUnclamped = ((outTmpR * REVERB_WET_SIGNAL/* + inSampleL * REVERB_DRY_SIGNAL*/) / 256);
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outUnclamped = ((outTmpR * REVERB_WET_SIGNAL/* + inSampleL * REVERB_DRY_SIGNAL*/) >> 8);
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*outSampleR = CLAMP_S16(outUnclamped);
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}
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static inline void reverb_mono_sample(s16 *outSample, s32 inSample) {
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static void reverb_mono_sample(s16 *outSample, s32 inSample) {
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s32 *curDelaySample;
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s32 historySample;
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s32 i = 0;
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s32 j = 0;
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s32 k = 0;
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s32 outTmp = 0;
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s32 tmpCarryover = (((tmpBufL[reverbFilterCountm1] * REVERB_REV_INDEX) / 256) + inSample);
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s32 tmpCarryover = (((delayBufsL[reverbFilterCount][allpassIdxL[reverbFilterCount]] * REVERB_REV_INDEX) >> 8) + inSample);
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for (; i != reverbFilterCount; ++i, ++j) {
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tmpBufL[i] = delayBufsL[i][allpassIdxL[i]];
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for (; i <= reverbFilterCount; ++i, ++j) {
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curDelaySample = &delayBufsL[i][allpassIdxL[i]];
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historySample = *curDelaySample;
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if (j == 2) {
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j = -1;
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outTmp += ((tmpBufL[i] * reverbMultsL[k++]) / 256);
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delayBufsL[i][allpassIdxL[i]] = tmpCarryover;
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if (i != reverbFilterCountm1)
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tmpCarryover = ((tmpBufL[i] * REVERB_REV_INDEX) / 256);
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outTmp += ((historySample * reverbMultsL[k++]) >> 8);
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*curDelaySample = tmpCarryover;
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if (i != reverbFilterCount)
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tmpCarryover = ((historySample * REVERB_REV_INDEX) >> 8);
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} else {
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delayBufsL[i][allpassIdxL[i]] = (((tmpBufL[i] * (-REVERB_GAIN_INDEX)) / 256) + tmpCarryover);
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tmpCarryover = (((delayBufsL[i][allpassIdxL[i]] * REVERB_GAIN_INDEX) / 256) + tmpBufL[i]);
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*curDelaySample = (((historySample * (-REVERB_GAIN_INDEX)) >> 8) + tmpCarryover);
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tmpCarryover = (((*curDelaySample * REVERB_GAIN_INDEX) >> 8) + historySample);
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}
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if (++allpassIdxL[i] == delaysL[i])
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allpassIdxL[i] = 0;
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if (++allpassIdxL[i] == delaysL[i]) allpassIdxL[i] = 0;
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}
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s32 outUnclamped = ((outTmp * REVERB_WET_SIGNAL/* + inSample * REVERB_DRY_SIGNAL*/) / 256);
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s32 outUnclamped = ((outTmp * REVERB_WET_SIGNAL/* + inSample * REVERB_DRY_SIGNAL*/) >> 8);
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*outSample = CLAMP_S16(outUnclamped);
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}
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void initialize_better_reverb_buffers(void) {
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delayBufsL = (s32**) soundAlloc(&gBetterReverbPool, BETTER_REVERB_PTR_SIZE);
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delayBufsR = &delayBufsL[NUM_ALLPASS];
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delayBufsL[0] = (s32*) soundAlloc(&gBetterReverbPool, BETTER_REVERB_SIZE - BETTER_REVERB_PTR_SIZE);
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}
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void clear_better_reverb_buffers(void) {
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bzero(delayBufsL[0], (BETTER_REVERB_SIZE - BETTER_REVERB_PTR_SIZE));
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bzero(allpassIdxL, sizeof(allpassIdxL));
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bzero(allpassIdxR, sizeof(allpassIdxR));
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}
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void set_better_reverb_buffers(void) {
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s32 bufOffset = 0;
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s32 i;
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for (i = 0; i < NUM_ALLPASS; ++i) {
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delaysL[i] = (delaysBaselineL[i] / gReverbDownsampleRate);
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delaysR[i] = (delaysBaselineR[i] / gReverbDownsampleRate);
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delayBufsL[i] = (s32*) &delayBufsL[0][bufOffset];
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bufOffset += delaysL[i];
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delayBufsR[i] = (s32*) &delayBufsL[0][bufOffset]; // L and R buffers are interpolated adjacently in memory; not a bug
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bufOffset += delaysR[i];
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}
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}
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#endif
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#ifdef VERSION_EU
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@@ -361,6 +397,7 @@ void prepare_reverb_ring_buffer(s32 chunkLen, u32 updateIndex) {
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}
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#ifdef BETTER_REVERB
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else if (toggleBetterReverb) {
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reverbFilterCount--; // Temporarily lower filter count for optimized bulk processing
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item = &gSynthesisReverb.items[gSynthesisReverb.curFrame][updateIndex];
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if (gSoundMode == SOUND_MODE_MONO || monoReverb) {
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if (gReverbDownsampleRate != 1) {
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@@ -397,6 +434,7 @@ void prepare_reverb_ring_buffer(s32 chunkLen, u32 updateIndex) {
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reverb_samples(&gSynthesisReverb.ringBuffer.left[dstPos], &gSynthesisReverb.ringBuffer.right[dstPos], gSynthesisReverb.ringBuffer.left[dstPos], gSynthesisReverb.ringBuffer.right[dstPos]);
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}
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}
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reverbFilterCount++; // Reset filter count to accurate numbers
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}
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#endif
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item = &gSynthesisReverb.items[gSynthesisReverb.curFrame][updateIndex];
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@@ -556,18 +594,31 @@ u64 *synthesis_execute(u64 *cmdBuf, s32 *writtenCmds, s16 *aiBuf, s32 bufLen) {
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#ifdef BETTER_REVERB
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if (gIsConsole) {
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reverbFilterCount = (s32) reverbFilterCountConsole;
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reverbFilterCount = reverbFilterCountConsole;
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monoReverb = monoReverbConsole;
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} else {
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reverbFilterCount = (s32) reverbFilterCountEmulator;
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reverbFilterCount = reverbFilterCountEmulator;
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monoReverb = monoReverbEmulator;
|
||||
}
|
||||
if (reverbFilterCount > NUM_ALLPASS) {
|
||||
reverbFilterCount = NUM_ALLPASS;
|
||||
} else if (reverbFilterCount < 3) {
|
||||
reverbFilterCount = 3;
|
||||
}
|
||||
reverbFilterCountm1 = (reverbFilterCount - 1);
|
||||
if (reverbFilterCount < 3) {
|
||||
reverbFilterCountm1 = 0;
|
||||
|
||||
s32 filterCountDiv3 = reverbFilterCount / 3;
|
||||
reverbFilterCount = filterCountDiv3 * 3; // reverbFilterCount should always be a multiple of 3.
|
||||
|
||||
// Update reverbMultsL every audio frame just in case gReverbMults is ever to change.
|
||||
for (i = 0; i < filterCountDiv3; ++i) {
|
||||
reverbMultsL[i] = gReverbMultsL[i];
|
||||
reverbMultsR[i] = gReverbMultsR[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) {
|
||||
reverbMultsL[0] = (reverbMultsR[0] + reverbMultsL[0]) / 2;
|
||||
reverbMultsR[0] = reverbMultsL[0];
|
||||
}
|
||||
#endif
|
||||
|
||||
|
||||
@@ -33,20 +33,16 @@
|
||||
|
||||
extern s8 betterReverbDownsampleConsole;
|
||||
extern s8 betterReverbDownsampleEmulator;
|
||||
extern u32 reverbFilterCountConsole;
|
||||
extern u32 reverbFilterCountEmulator;
|
||||
extern u8 monoReverbConsole;
|
||||
extern u8 monoReverbEmulator;
|
||||
extern s32 reverbFilterCountConsole;
|
||||
extern s32 reverbFilterCountEmulator;
|
||||
extern s32 betterReverbWindowsSize;
|
||||
|
||||
extern s32 delaysBaselineL[NUM_ALLPASS];
|
||||
extern s32 delaysBaselineR[NUM_ALLPASS];
|
||||
extern s32 delaysL[NUM_ALLPASS];
|
||||
extern s32 delaysR[NUM_ALLPASS];
|
||||
extern s32 reverbMultsL[NUM_ALLPASS / 3];
|
||||
extern s32 reverbMultsR[NUM_ALLPASS / 3];
|
||||
extern s32 **delayBufsL;
|
||||
extern s32 **delayBufsR;
|
||||
extern s32 gReverbMultsL[NUM_ALLPASS / 3];
|
||||
extern s32 gReverbMultsR[NUM_ALLPASS / 3];
|
||||
|
||||
extern u8 toggleBetterReverb;
|
||||
#define REVERB_WINDOW_SIZE_MAX 0x2000
|
||||
@@ -149,6 +145,12 @@ extern struct SynthesisReverb gSynthesisReverb;
|
||||
extern s16 D_SH_803479B4;
|
||||
#endif
|
||||
|
||||
#ifdef BETTER_REVERB
|
||||
void initialize_better_reverb_buffers(void);
|
||||
void clear_better_reverb_buffers(void);
|
||||
void set_better_reverb_buffers(void);
|
||||
#endif
|
||||
|
||||
u64 *synthesis_execute(u64 *cmdBuf, s32 *writtenCmds, s16 *aiBuf, s32 bufLen);
|
||||
#if defined(VERSION_JP) || defined(VERSION_US)
|
||||
void note_init_volume(struct Note *note);
|
||||
|
||||
Reference in New Issue
Block a user