Format

include/audio/hle_core.hpp

@@ -21,7 +21,7 @@ namespace Audio {
 			u32 paddr;        // Physical address of the buffer
 			u32 sampleCount;  // Total number of samples
 			u8 adpcmScale;    // ADPCM predictor/scale
-			u8 pad1;           // Unknown
+			u8 pad1;          // Unknown
 
 			std::array<s16, 2> previousSamples;  // ADPCM y[n-1] and y[n-2]
 			bool adpcmDirty;
@@ -59,8 +59,8 @@ namespace Audio {
 		// Where y[n] is the output sample we're generating, x[n] is the ADPCM "differential" of the current sample
 		// And coeff1/coeff2 are the coefficients from this array that are used for weighing the history samples
 		std::array<s16, 16> adpcmCoefficients;
-		s16 history1; // y[n-1], the previous output sample
+		s16 history1;  // y[n-1], the previous output sample
-		s16 history2; // y[n-2], the previous previous output sample
+		s16 history2;  // y[n-2], the previous previous output sample
 
 		SampleBuffer currentSamples;
 		int index = 0;  // Index of the voice in [0, 23] for debugging
@@ -69,7 +69,7 @@ namespace Audio {
 		// Pop a buffer from the buffer queue and return it
 		Buffer popBuffer() {
 			assert(!buffers.empty());
-			
+
 			Buffer ret = buffers.top();
 			buffers.pop();
 
@@ -87,7 +87,7 @@ namespace Audio {
 
 		template <typename T, usize channelCount>
 		using Frame = std::array<Sample<T, channelCount>, 160>;
-		
+
 		template <typename T>
 		using MonoFrame = Frame<T, 1>;
 
@@ -99,6 +99,7 @@ namespace Audio {
 
 		using Source = Audio::DSPSource;
 		using SampleBuffer = Source::SampleBuffer;
+
 	  private:
 		enum class DSPState : u32 {
 			Off,
@@ -118,7 +119,7 @@ namespace Audio {
 		SourceType sourceType = SourceType::Stereo;
 
 		void resetAudioPipe();
-		bool loaded = false; // Have we loaded a component?
+		bool loaded = false;  // Have we loaded a component?
 
 		// Get the index for the current region we'll be reading. Returns the region with the highest frame counter
 		// Accounting for whether one of the frame counters has wrapped around

src/core/audio/hle_core.cpp

@@ -107,7 +107,7 @@ namespace Audio {
 		outputFrame();
 		scheduler.addEvent(Scheduler::EventType::RunDSP, scheduler.currentTimestamp + Audio::cyclesPerFrame);
 	}
-	
+
 	u16 HLE_DSP::recvData(u32 regId) {
 		if (regId != 0) {
 			Helpers::panic("Audio: invalid register in HLE frontend");
@@ -141,14 +141,11 @@ namespace Audio {
 							// TODO: Other initialization stuff here
 							dspState = DSPState::On;
 							resetAudioPipe();
-							
+
 							dspService.triggerPipeEvent(DSPPipeType::Audio);
 							break;
 
-						case StateChange::Shutdown:
+						case StateChange::Shutdown: dspState = DSPState::Off; break;
-							dspState = DSPState::Off;
-							break;
-
 						default: Helpers::panic("Unimplemented DSP audio pipe state change %d", state);
 					}
 				}
@@ -216,11 +213,13 @@ namespace Audio {
 
 			// Generate audio
 			if (source.enabled && !source.buffers.empty()) {
+				static int aaaa = 0;
 				const auto& buffer = source.buffers.top();
 				const u8* data = getPointerPhys<u8>(buffer.paddr);
 
 				if (data != nullptr) {
 					// TODO
+					}
 				}
 			}
 
@@ -265,7 +264,7 @@ namespace Audio {
 			config.partialResetFlag = 0;
 			source.buffers = {};
 		}
-		
+
 		// TODO: Should we check bufferQueueDirty here too?
 		if (config.formatDirty || config.embeddedBufferDirty) {
 			sampleFormat = config.format;
@@ -341,14 +340,15 @@ namespace Audio {
 
 	HLE_DSP::SampleBuffer HLE_DSP::decodeADPCM(const u8* data, usize sampleCount, Source& source) {
 		static constexpr uint samplesPerBlock = 14;
-		// An ADPCM block is comprised of a single header which contains the scale and predictor value for the block, and then 14 4bpp samples (hence the / 2)
+		// An ADPCM block is comprised of a single header which contains the scale and predictor value for the block, and then 14 4bpp samples (hence
+		// the / 2)
 		static constexpr usize blockSize = sizeof(u8) + samplesPerBlock / 2;
- 
+
 		// How many ADPCM blocks we'll be consuming. It's sampleCount / samplesPerBlock, rounded up.
 		const usize blockCount = (sampleCount + (samplesPerBlock - 1)) / samplesPerBlock;
 		const usize outputSize = sampleCount + (sampleCount & 1);  // Bump the output size to a multiple of 2
 
-		usize outputCount = 0; // How many stereo samples have we output thus far?
+		usize outputCount = 0;  // How many stereo samples have we output thus far?
 		SampleBuffer decodedSamples(outputSize);
 
 		s16 history1 = source.history1;
@@ -371,8 +371,8 @@ namespace Audio {
 			// So each byte of ADPCM data ends up generating 2 stereo samples
 			for (uint sampleIndex = 0; sampleIndex < samplesPerBlock && outputCount < sampleCount; sampleIndex += 2) {
 				const auto decode = [&](s32 nibble) -> s16 {
-					static constexpr s32 ONE = 0x800; // 1.0 in S5.11 fixed point
+					static constexpr s32 ONE = 0x800;     // 1.0 in S5.11 fixed point
-					static constexpr s32 HALF = ONE / 2; // 0.5 similarly
+					static constexpr s32 HALF = ONE / 2;  // 0.5 similarly
 
 					// Sign extend our nibble from s4 to s32
 					nibble = (nibble << 28) >> 28;