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optimize dsp by 27X

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This commit is contained in:
madeline
2026-04-09 08:25:06 -07:00
parent 755239f280
commit 4ed0909085
7 changed files with 144 additions and 92 deletions
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src/dusk/audio/DuskDsp.cpp
+52 -50
View File
@@ -9,6 +9,7 @@
#include <span>
#include "Adpcm.hpp"
#include "freeverb/revmodel.hpp"
#include "JSystem/JAudio2/JASDriverIF.h"
#include "dusk/audio/DuskAudioSystem.h"
#include "dusk/endian.h"
@@ -18,6 +19,9 @@ using namespace dusk::audio;
ChannelAuxData dusk::audio::ChannelAux[DSP_CHANNELS] = {};
static revmodel SharedReverb;
static bool ReverbHasTail = false;
static bool sDumpWasActive = false;
static FILE* sChannelDumpFiles[DSP_CHANNELS] = {};
@@ -140,37 +144,37 @@ void dusk::audio::DspRender(OutputSubframe& subframe) {
std::span channels(JASDsp::CH_BUF, DSP_CHANNELS);
DspSubframe reverbInputL = {};
DspSubframe reverbInputR = {};
bool anyReverbInput = false;
for (int i = 0; i < channels.size(); i++) {
auto& channel = channels[i];
auto& channelAux = ChannelAux[i];
bool skipRender = false;
if (!channel.mIsActive) {
skipRender = true;
continue;
}
else if (channel.mPauseFlag) {
// Not really sure what the practical difference between pause and
// deactivation is. Either avoids clearing state or allows the DSP to avoid popping?
skipRender = true;
continue;
}
else if (channel.mForcedStop) {
channel.mIsFinished = true;
skipRender = true;
continue;
}
else if (channel.mWaveAramAddress == 0) {
// I think these are oscillator channels? Not backed by audio.
// No idea how to implement these yet, so skip them.
channel.mIsFinished = true;
skipRender = true;
continue;
}
ValidateChannel(channel);
OutputSubframe channelSubframe = {};
if (!skipRender) {
ValidateChannel(channel);
RenderChannel(channel, channelAux, channelSubframe);
}
RenderChannel(channel, channelAux, channelSubframe);
if (EnableReverb) {
// scale the input to the reverb rather than using wet/dry on the output.
@@ -178,31 +182,23 @@ void dusk::audio::DspRender(OutputSubframe& subframe) {
// so any tail always decays at the correct level regardless of mAutoMixerFxMix changes
// prevents transients when the next sound starts playing with a different reverb level
// 600.0f was pulled out of my ass and just sounds good enough for console
f32 inputGain = (!skipRender) ? (channel.mAutoMixerFxMix >> 8) / 600.0f : 0.0f;
OutputSubframe reverbSubframe = {};
for (int j = 0; j < DSP_SUBFRAME_SIZE; j++) {
reverbSubframe.channels[0][j] = channelSubframe.channels[0][j] * inputGain;
reverbSubframe.channels[1][j] = channelSubframe.channels[1][j] * inputGain;
}
channelAux.reverb.processreplace(
reverbSubframe.channels[0].data(), reverbSubframe.channels[1].data(),
reverbSubframe.channels[0].data(), reverbSubframe.channels[1].data(),
DSP_SUBFRAME_SIZE, 1
);
for (int j = 0; j < DSP_SUBFRAME_SIZE; j++) {
channelSubframe.channels[0][j] += reverbSubframe.channels[0][j];
channelSubframe.channels[1][j] += reverbSubframe.channels[1][j];
f32 inputGain = (channel.mAutoMixerFxMix >> 8) / 600.0f;
if (inputGain > 0) {
anyReverbInput = true;
for (int j = 0; j < DSP_SUBFRAME_SIZE; j++) {
reverbInputL[j] += channelSubframe.channels[0][j] * inputGain;
reverbInputR[j] += channelSubframe.channels[1][j] * inputGain;
}
}
}
if (DumpAudio && sChannelDumpFiles[i]) {
f32 interleaved[DSP_SUBFRAME_SIZE * 2];
for (int j = 0; j < DSP_SUBFRAME_SIZE; j++) {
fwrite(&channelSubframe.channels[0][j], sizeof(f32), 1, sChannelDumpFiles[i]);
fwrite(&channelSubframe.channels[1][j], sizeof(f32), 1, sChannelDumpFiles[i]);
interleaved[j * 2 + 0] = channelSubframe.channels[0][j];
interleaved[j * 2 + 1] = channelSubframe.channels[1][j];
}
fwrite(interleaved, sizeof(f32), DSP_SUBFRAME_SIZE * 2, sChannelDumpFiles[i]);
}
for (int o = 0; o < subframe.channels.size(); o++) {
@@ -210,6 +206,17 @@ void dusk::audio::DspRender(OutputSubframe& subframe) {
}
}
if (EnableReverb && (anyReverbInput || ReverbHasTail)) {
// Equivalent to -80 dBFS: rms = 1e-4, rms^2 = 1e-8, sumSq = 2 * N * 1e-8
constexpr f32 REVERB_ENERGY_EPSILON = 2.0f * DSP_SUBFRAME_SIZE * 1e-8f;
f32 wetEnergy = SharedReverb.processmix(
reverbInputL.data(), reverbInputR.data(),
subframe.channels[0].data(), subframe.channels[1].data(),
DSP_SUBFRAME_SIZE, 1, 1.0f
);
ReverbHasTail = wetEnergy >= REVERB_ENERGY_EPSILON;
}
for (auto& channel : subframe.channels) {
ApplyVolume(channel, channel, PrevMasterVolume, MasterVolume);
}
@@ -341,7 +348,7 @@ static void FillDecodeBuf(JASDsp::TChannel& channel, ChannelAuxData& aux, int ne
}
aux.decodeBufCount += ReadChannelSamplesChunk(
channel, aux, std::min(remainingDecodeSpace, needed - aux.decodeBufCount),
channel, aux, std::min(remainingDecodeSpace, needed - aux.decodeBufCount),
aux.decodeBuf + aux.decodeBufCount, remainingDecodeSpace
);
}
@@ -491,17 +498,17 @@ static void RenderChannel(
}
DspSubframe audioLoadBuffer = {};
f64 pos = channelAux.resamplePos;
f32 pos = channelAux.resamplePos;
s16 prev = channelAux.resamplePrev;
s16 next = channelAux.decodeBufCount > 0 ? channelAux.decodeBuf[0] : prev;
int srcIdx = 0;
// linear resampling and f32 conversion
for (int i = 0; i < DSP_SUBFRAME_SIZE; i++) {
audioLoadBuffer[i] = static_cast<f32>(prev + pos * (next - prev)) / 32768.0f;
audioLoadBuffer[i] = (prev + pos * (next - prev)) / 32768.0f;
pos += step;
while (pos >= 1.0) {
pos -= 1.0;
while (pos >= 1.0f) {
pos -= 1.0f;
prev = next;
srcIdx++;
next = srcIdx < channelAux.decodeBufCount ? channelAux.decodeBuf[srcIdx] : prev;
@@ -529,16 +536,13 @@ static void RenderChannel(
}
void dusk::audio::DspInit() {
for (int i = 0; i < DSP_CHANNELS; i++) {
auto& channelAux = ChannelAux[i];
channelAux.reverb.setwet(1.0f);
channelAux.reverb.setdry(0.0f);
channelAux.reverb.setroomsize(0.5f);
channelAux.reverb.setdamp(0.7f);
channelAux.reverb.setwidth(1.0f);
channelAux.reverb.setmode(0.0f);
channelAux.reverb.mute();
}
SharedReverb.setwet(1.0f);
SharedReverb.setdry(0.0f);
SharedReverb.setroomsize(0.5f);
SharedReverb.setdamp(0.7f);
SharedReverb.setwidth(1.0f);
SharedReverb.setmode(0.0f);
SharedReverb.mute();
}
void dusk::audio::ApplyVolume(
@@ -549,15 +553,13 @@ void dusk::audio::ApplyVolume(
assert(dst.size() >= src.size());
if (startVolume == endVolume) {
for (int i = 0; i < src.size(); i++) {
for (int i = 0; i < (int)src.size(); i++) {
dst[i] = src[i] * startVolume;
}
} else {
const f32 step = (endVolume - startVolume) / static_cast<f32>(src.size());
auto curVolume = startVolume;
for (int i = 0; i < src.size(); i++) {
dst[i] = src[i] * curVolume;
curVolume += step;
for (int i = 0; i < (int)src.size(); i++) {
dst[i] = src[i] * (startVolume + i * step);
}
}
}
src/dusk/audio/DuskDsp.hpp
-3
View File
@@ -8,8 +8,6 @@
#include "SDL3/SDL_audio.h"
#include <span>
#include "freeverb/revmodel.hpp"
// ReSharper disable once CppUnusedIncludeDirective
#include "global.h"
@@ -31,7 +29,6 @@ namespace dusk::audio {
// Used for debugging tools.
u32 resetCount;
revmodel reverb;
/**
* Previous volume values, per output channel.