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More HLE DSP work

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wheremyfoodat 2024-11-06 19:26:57 +02:00
parent 07cee43a2b
commit b299609a9b
2 changed files with 60 additions and 48 deletions
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85
include/audio/hle_core.hpp
View file

@ -16,6 +16,46 @@ namespace Audio {
using SampleFormat = HLE::SourceConfiguration::Configuration::Format;
using SourceType = HLE::SourceConfiguration::Configuration::MonoOrStereo;
class DSPMixer {
public:
template <typename T, usize channelCount = 1>
using Sample = std::array<T, channelCount>;
template <typename T, usize channelCount>
using Frame = std::array<Sample<T, channelCount>, 160>;
template <typename T>
using MonoFrame = Frame<T, 1>;
template <typename T>
using StereoFrame = Frame<T, 2>;
template <typename T>
using QuadFrame = Frame<T, 4>;
// Internally the DSP uses four channels when mixing.
// Neatly, QuadFrame<s32> means that every sample is a uint32x4 value, which is particularly nice for SIMD mixing
using IntermediateMix = QuadFrame<s32>;
private:
using ChannelFormat = HLE::DspConfiguration::OutputFormat;
// The audio from each DSP voice is converted to quadraphonic and then fed into 3 intermediate mixing stages
// Two of these intermediate mixers (second and third) are used for effects, including custom effects done on the CPU
static constexpr usize mixerStageCount = 3;
public:
ChannelFormat channelFormat = ChannelFormat::Stereo;
std::array<float, mixerStageCount> volumes;
std::array<bool, 2> enableAuxStages;
void reset() {
channelFormat = ChannelFormat::Stereo;
volumes.fill(0.0);
enableAuxStages.fill(false);
}
};
struct DSPSource {
// Audio buffer information
// https://www.3dbrew.org/wiki/DSP_Memory_Region
@ -49,6 +89,7 @@ namespace Audio {
using SampleBuffer = std::deque<std::array<s16, 2>>;
using BufferQueue = std::priority_queue<Buffer>;
DSPMixer::StereoFrame<s16> currentFrame;
BufferQueue buffers;
SampleFormat sampleFormat = SampleFormat::ADPCM;
@ -98,46 +139,6 @@ namespace Audio {
DSPSource() { reset(); }
};
class DSPMixer {
public:
template <typename T, usize channelCount = 1>
using Sample = std::array<T, channelCount>;
template <typename T, usize channelCount>
using Frame = std::array<Sample<T, channelCount>, 160>;
template <typename T>
using MonoFrame = Frame<T, 1>;
template <typename T>
using StereoFrame = Frame<T, 2>;
template <typename T>
using QuadFrame = Frame<T, 4>;
// Internally the DSP uses four channels when mixing.
// Neatly, QuadFrame<s32> means that every sample is a uint32x4 value, which is particularly nice for SIMD mixing
using IntermediateMix = QuadFrame<s32>;
private:
using ChannelFormat = HLE::DspConfiguration::OutputFormat;
// The audio from each DSP voice is converted to quadraphonic and then fed into 3 intermediate mixing stages
// Two of these intermediate mixers (second and third) are used for effects, including custom effects done on the CPU
static constexpr usize mixerStageCount = 3;
public:
ChannelFormat channelFormat = ChannelFormat::Stereo;
std::array<float, mixerStageCount> volumes;
std::array<bool, 2> enableAuxStages;
void reset() {
channelFormat = ChannelFormat::Stereo;
volumes.fill(0.0);
enableAuxStages.fill(false);
}
};
class HLE_DSP : public DSPCore {
// The audio frame types are public in case we want to use them for unit tests
public:
@ -159,7 +160,7 @@ namespace Audio {
using Source = Audio::DSPSource;
using SampleBuffer = Source::SampleBuffer;
using IntermediateMix = DSPMixer::IntermediateMix;
private:
enum class DSPState : u32 {
Off,
@ -226,7 +227,7 @@ namespace Audio {
void outputFrame();
// Perform the final mix, mixing the quadraphonic samples from all voices into the output audio frame
void performMix(Audio::HLE::SharedMemory& readRegion, Audio::HLE::SharedMemory& writeRegion);
// Decode an entire buffer worth of audio
void decodeBuffer(DSPSource& source);

23
src/core/audio/hle_core.cpp
View file

@ -262,10 +262,10 @@ namespace Audio {
for (usize sampleIndex = 0; sampleIndex < Audio::samplesInFrame; sampleIndex++) {
// Mono samples are in the format: (l, r)
// When converting to quad, gain0 and gain2 are applied to the left sample, gain1 and gain3 to the right one
intermediateMix[sampleIndex][0] += s32(source.currentSamples[sampleIndex][0] * gains[0]);
intermediateMix[sampleIndex][1] += s32(source.currentSamples[sampleIndex][1] * gains[1]);
intermediateMix[sampleIndex][2] += s32(source.currentSamples[sampleIndex][0] * gains[2]);
intermediateMix[sampleIndex][3] += s32(source.currentSamples[sampleIndex][1] * gains[3]);
intermediateMix[sampleIndex][0] += s32(source.currentFrame[sampleIndex][0] * gains[0]);
intermediateMix[sampleIndex][1] += s32(source.currentFrame[sampleIndex][1] * gains[1]);
intermediateMix[sampleIndex][2] += s32(source.currentFrame[sampleIndex][0] * gains[2]);
intermediateMix[sampleIndex][3] += s32(source.currentFrame[sampleIndex][1] * gains[3]);
}
}
}
@ -467,6 +467,9 @@ namespace Audio {
}
void HLE_DSP::generateFrame(DSPSource& source) {
// Zero out all output samples at first. TODO: Don't zero out the entire frame initially, rather only zero-out the "unwritten" samples when the frame is done being processed.
source.currentFrame = {};
if (source.currentSamples.empty()) {
// There's no audio left to play, turn the voice off
if (source.buffers.empty()) {
@ -480,7 +483,7 @@ namespace Audio {
decodeBuffer(source);
} else {
uint maxSampleCount = uint(float(Audio::samplesInFrame) * source.rateMultiplier);
uint maxSampleCount = uint(float(Audio::samplesInFrame) * 1.0);
uint outputCount = 0;
while (outputCount < maxSampleCount) {
@ -494,8 +497,14 @@ namespace Audio {
const uint sampleCount = std::min<s32>(maxSampleCount - outputCount, source.currentSamples.size());
// samples.insert(samples.end(), source.currentSamples.begin(), source.currentSamples.begin() + sampleCount);
// Copy samples to current frame buffer
// TODO: Implement linear/polyphase interpolation
std::copy(
source.currentSamples.begin(), std::next(source.currentSamples.begin(), sampleCount), source.currentFrame.begin() + outputCount
);
// Remove samples from sample buffer
source.currentSamples.erase(source.currentSamples.begin(), std::next(source.currentSamples.begin(), sampleCount));
// Advance sample position
source.samplePosition += sampleCount;
outputCount += sampleCount;
}
@ -718,5 +727,7 @@ namespace Audio {
buffers = {};
currentSamples.clear();
gains.fill({});
}
} // namespace Audio