More HLE DSP work

include/audio/hle_core.hpp

@@ -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:

src/core/audio/hle_core.cpp

@@ -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][0] += s32(source.currentFrame[sampleIndex][0] * gains[0]);
-						intermediateMix[sampleIndex][1] += s32(source.currentSamples[sampleIndex][1] * gains[1]);
+						intermediateMix[sampleIndex][1] += s32(source.currentFrame[sampleIndex][1] * gains[1]);
-						intermediateMix[sampleIndex][2] += s32(source.currentSamples[sampleIndex][0] * gains[2]);
+						intermediateMix[sampleIndex][2] += s32(source.currentFrame[sampleIndex][0] * gains[2]);
-						intermediateMix[sampleIndex][3] += s32(source.currentSamples[sampleIndex][1] * gains[3]);
+						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