Merge pull request #17175 from hrydgard/more-audio-refactor

Some more audio refactoring

Common/System/NativeApp.h

@@ -75,8 +75,7 @@ void NativeRender(GraphicsContext *graphicsContext);
 // the rest of the game, so be careful with synchronization.
 // Returns the number of samples actually output. The app should do everything it can
 // to fill the buffer completely.
-int NativeMix(short *audio, int num_samples);
-void NativeSetMixer(void* mixer);
+int NativeMix(short *audio, int num_samples, int sampleRateHz);
 
 // Called when it's time to shutdown. After this has been called,
 // no more calls to any other function will be made from the framework

Core/HLE/__sceAudio.cpp

@@ -49,6 +49,33 @@
 
 StereoResampler resampler;
 
+// numFrames is number of stereo frames.
+// This is called from *outside* the emulator thread.
+int __AudioMix(short *outstereo, int numFrames, int sampleRate) {
+	return resampler.Mix(outstereo, numFrames, false, sampleRate);
+}
+
+void __AudioGetDebugStats(char *buf, size_t bufSize) {
+	resampler.GetAudioDebugStats(buf, bufSize);
+}
+
+void __AudioClear() {
+	resampler.Clear();
+}
+
+void __AudioPushSamples(const s32 *audio, int numSamples) {
+	if (audio) {
+		resampler.PushSamples(audio, numSamples);
+	} else {
+		resampler.Clear();
+	}
+}
+
+void __AudioResetStatCounters() {
+	resampler.ResetStatCounters();
+}
+
+
 // Should be used to lock anything related to the outAudioQueue.
 // atomic locks are used on the lock. TODO: make this lock-free
 std::atomic_flag atomicLock_;
@@ -109,7 +136,7 @@ static void __AudioCPUMHzChange() {
 
 
 void __AudioInit() {
-	resampler.ResetStatCounters();
+	__AudioResetStatCounters();
 	mixFrequency = 44100;
 	srcFrequency = 0;
 
@@ -132,7 +159,7 @@ void __AudioInit() {
 	clampedMixBuffer = new s16[hwBlockSize * 2];
 	memset(mixBuffer, 0, hwBlockSize * 2 * sizeof(s32));
 
-	resampler.Clear();
+	__AudioClear();
 	CoreTiming::RegisterMHzChangeCallback(&__AudioCPUMHzChange);
 }
 
@@ -155,15 +182,18 @@ void __AudioDoState(PointerWrap &p) {
 		mixFrequency = 44100;
 	}
 
-	// TODO: This never happens because maxVer=1.
 	if (s >= 2) {
-		resampler.DoState(p);
+		// TODO: Next time we bump, get rid of this. It's kinda useless.
+		StereoResampler::DoState(p);
+		if (p.mode == p.MODE_READ) {
+			__AudioClear();
+		}
 	} else {
 		// Only to preserve the previous file format. Might cause a slight audio glitch on upgrades?
 		FixedSizeQueue<s16, 512 * 16> outAudioQueue;
 		outAudioQueue.DoState(p);
 
-		resampler.Clear();
+		__AudioClear();
 	}
 
 	int chanCount = ARRAY_SIZE(chans);
@@ -333,9 +363,7 @@ void __AudioSetSRCFrequency(int freq) {
 	srcFrequency = freq;
 }
 
-// Mix samples from the various audio channels into a single sample queue.
-// This single sample queue is where __AudioMix should read from. If the sample queue is full, we should
-// just sleep the main emulator thread a little.
+// Mix samples from the various audio channels into a single sample queue, managed by the backend implementation.
 void __AudioUpdate(bool resetRecording) {
 	// Audio throttle doesn't really work on the PSP since the mixing intervals are so closely tied
 	// to the CPU. Much better to throttle the frame rate on frame display and just throw away audio
@@ -427,7 +455,7 @@ void __AudioUpdate(bool resetRecording) {
 	}
 
 	if (g_Config.bEnableSound) {
-		resampler.PushSamples(mixBuffer, hwBlockSize);
+		__AudioPushSamples(mixBuffer, hwBlockSize);
 #ifndef MOBILE_DEVICE
 		if (g_Config.bSaveLoadResetsAVdumping && resetRecording) {
 			__StopLogAudio();
@@ -465,23 +493,6 @@ void __AudioUpdate(bool resetRecording) {
 	}
 }
 
-// numFrames is number of stereo frames.
-// This is called from *outside* the emulator thread.
-int __AudioMix(short *outstereo, int numFrames, int sampleRate) {
-	return resampler.Mix(outstereo, numFrames, false, sampleRate);
-}
-
-void __AudioGetDebugStats(char *buf, size_t bufSize) {
-	resampler.GetAudioDebugStats(buf, bufSize);
-}
-
-void __PushExternalAudio(const s32 *audio, int numSamples) {
-	if (audio) {
-		resampler.PushSamples(audio, numSamples);
-	} else {
-		resampler.Clear();
-	}
-}
 #ifndef MOBILE_DEVICE
 void __StartLogAudio(const Path& filename) {
 	if (!m_logAudio) {

Core/HLE/__sceAudio.h

@@ -46,9 +46,12 @@ u32 __AudioEnqueue(AudioChannel &chan, int chanNum, bool blocking);
 void __AudioWakeThreads(AudioChannel &chan, int result, int step);
 void __AudioWakeThreads(AudioChannel &chan, int result);
 
+// Resampler API, to be extracted
 int __AudioMix(short *outstereo, int numSamples, int sampleRate);
 void __AudioGetDebugStats(char *buf, size_t bufSize);
-void __PushExternalAudio(const s32 *audio, int numSamples);  // Should not be used in-game, only at the menu!
+void __AudioClear();
+void __AudioPushSamples(const s32 *audio, int numSamples);  // Should not be used in-game, only at the menu!
+void __AudioResetStatCounters();
 
 int __AudioGetHostAttemptBlockSize();
 

Core/HW/StereoResampler.cpp

@@ -346,8 +346,4 @@ void StereoResampler::ResetStatCounters() {
 
 void StereoResampler::DoState(PointerWrap &p) {
 	auto s = p.Section("resampler", 1);
-	if (!s)
-		return;
-	if (p.mode == p.MODE_READ)
-		Clear();
 }

Core/HW/StereoResampler.h

@@ -41,7 +41,8 @@ public:
 
 	void Clear();
 
-	void DoState(PointerWrap &p);
+	// TODO: Get rid of this.
+	static void DoState(PointerWrap &p);
 
 	void GetAudioDebugStats(char *buf, size_t bufSize);
 	void ResetStatCounters();

Qt/QtMain.cpp

@@ -54,6 +54,13 @@
 #include <signal.h>
 #include <string.h>
 
+// Audio
+#define AUDIO_FREQ 44100
+#define AUDIO_CHANNELS 2
+#define AUDIO_SAMPLES 2048
+#define AUDIO_SAMPLESIZE 16
+#define AUDIO_BUFFERS 5
+
 MainUI *emugl = nullptr;
 static float refreshRate = 60.f;
 static int browseFileEvent = -1;
@@ -69,7 +76,7 @@ SDL_AudioSpec g_retFmt;
 static SDL_AudioDeviceID audioDev = 0;
 
 extern void mixaudio(void *userdata, Uint8 *stream, int len) {
-	NativeMix((short *)stream, len / 4);
+	NativeMix((short *)stream, len / 4, AUDIO_FREQ);
 }
 
 static void InitSDLAudioDevice() {
@@ -726,12 +733,6 @@ void MainUI::updateAccelerometer() {
 }
 
 #ifndef SDL
-// Audio
-#define AUDIO_FREQ 44100
-#define AUDIO_CHANNELS 2
-#define AUDIO_SAMPLES 2048
-#define AUDIO_SAMPLESIZE 16
-#define AUDIO_BUFFERS 5
 
 MainAudio::~MainAudio() {
 	if (feed != nullptr) {
@@ -770,7 +771,7 @@ void MainAudio::run() {
 
 void MainAudio::timerEvent(QTimerEvent *) {
 	memset(mixbuf, 0, mixlen);
-	size_t frames = NativeMix((short *)mixbuf, AUDIO_BUFFERS*AUDIO_SAMPLES);
+	size_t frames = NativeMix((short *)mixbuf, AUDIO_BUFFERS*AUDIO_SAMPLES, AUDIO_FREQ);
 	if (frames > 0)
 		feed->write(mixbuf, sizeof(short) * AUDIO_CHANNELS * frames);
 }

SDL/SDLMain.cpp

@@ -74,6 +74,7 @@ static int g_QuitRequested = 0;
 static int g_DesktopWidth = 0;
 static int g_DesktopHeight = 0;
 static float g_RefreshRate = 60.f;
+static int g_sampleRate = 44100;
 
 static SDL_AudioSpec g_retFmt;
 
@@ -92,7 +93,7 @@ int getDisplayNumber(void) {
 }
 
 void sdl_mixaudio_callback(void *userdata, Uint8 *stream, int len) {
-	NativeMix((short *)stream, len / (2 * 2));
+	NativeMix((short *)stream, len / (2 * 2), g_sampleRate);
 }
 
 static SDL_AudioDeviceID audioDev = 0;
@@ -101,7 +102,7 @@ static SDL_AudioDeviceID audioDev = 0;
 static void InitSDLAudioDevice(const std::string &name = "") {
 	SDL_AudioSpec fmt;
 	memset(&fmt, 0, sizeof(fmt));
-	fmt.freq = 44100;
+	fmt.freq = g_sampleRate;
 	fmt.format = AUDIO_S16;
 	fmt.channels = 2;
 	fmt.samples = 256;

UI/BackgroundAudio.cpp

@@ -12,6 +12,7 @@
 #include "Common/Data/Collections/FixedSizeQueue.h"
 #include "Core/HW/SimpleAudioDec.h"
 #include "Core/HLE/__sceAudio.h"
+#include "Core/System.h"
 #include "GameInfoCache.h"
 #include "Core/Config.h"
 #include "UI/BackgroundAudio.h"
@@ -333,14 +334,18 @@ void BackgroundAudio::SetGame(const Path &path) {
 	bgGamePath_ = path;
 }
 
-int BackgroundAudio::Play() {
+bool BackgroundAudio::Play() {
+	if (GetUIState() == UISTATE_INGAME) {
+		return false;
+	}
+
 	std::lock_guard<std::mutex> lock(mutex_);
 
 	// Immediately stop the sound if it is turned off while playing.
 	if (!g_Config.bEnableSound) {
 		Clear(true);
-		__PushExternalAudio(0, 0);
-		return 0;
+		__AudioClear();
+		return true;
 	}
 
 	double now = time_now_d();
@@ -388,7 +393,7 @@ int BackgroundAudio::Play() {
 		}
 	}
 
-	__PushExternalAudio(buffer, sz);
+	__AudioPushSamples(buffer, sz);
 
 	if (at3Reader_ && fadingOut_ && volume_ <= 0.0f) {
 		Clear(true);
@@ -398,7 +403,7 @@ int BackgroundAudio::Play() {
 
 	lastPlaybackTime_ = now;
 
-	return 0;
+	return true;
 }
 
 void BackgroundAudio::Update() {

UI/BackgroundAudio.h

@@ -17,7 +17,7 @@ public:
 
 	void SetGame(const Path &path);
 	void Update();
-	int Play();
+	bool Play();
 
 	void LoadSamples();
 	void PlaySFX(UI::UISound sfx);

UI/NativeApp.cpp

@@ -203,12 +203,6 @@ public:
 	}
 };
 
-#ifdef _WIN32
-int Win32Mix(short *buffer, int numSamples, int bits, int rate) {
-	return NativeMix(buffer, numSamples);
-}
-#endif
-
 // globals
 static LogListener *logger = nullptr;
 Path boot_filename;
@@ -235,12 +229,8 @@ std::string NativeQueryConfig(std::string query) {
 	}
 }
 
-int NativeMix(short *audio, int num_samples) {
-	if (GetUIState() != UISTATE_INGAME) {
-		g_BackgroundAudio.Play();
-	}
-	int sample_rate = System_GetPropertyInt(SYSPROP_AUDIO_SAMPLE_RATE);
-	return __AudioMix(audio, num_samples, sample_rate > 0 ? sample_rate : 44100);
+int NativeMix(short *audio, int numSamples, int sampleRateHz) {
+	return __AudioMix(audio, numSamples, sampleRateHz);
 }
 
 // This is called before NativeInit so we do a little bit of initialization here.
@@ -860,9 +850,9 @@ bool NativeInitGraphics(GraphicsContext *graphicsContext) {
 #ifdef _WIN32
 	winAudioBackend = CreateAudioBackend((AudioBackendType)g_Config.iAudioBackend);
 #if PPSSPP_PLATFORM(UWP)
-	winAudioBackend->Init(0, &Win32Mix, 44100);
+	winAudioBackend->Init(0, &NativeMix, 44100);
 #else
-	winAudioBackend->Init(MainWindow::GetHWND(), &Win32Mix, 44100);
+	winAudioBackend->Init(MainWindow::GetHWND(), &NativeMix, 44100);
 #endif
 #endif
 
@@ -1235,6 +1225,7 @@ void NativeUpdate() {
 	g_screenManager->update();
 
 	g_Discord.Update();
+	g_BackgroundAudio.Play();
 
 	UI::SetSoundEnabled(g_Config.bUISound);
 }

UWP/XAudioSoundStream.cpp

@@ -155,7 +155,7 @@ void XAudioBackend::PollLoop() {
 		// take ownership of the data. It needs to be big enough to fit the max number of buffers we check for
 		// above, which it is, easily.
 
-		int stereoSamplesRendered = (*callback_)((short*)&realtimeBuffer_[cursor_], readCount / 4, 16, sampleRate_);
+		int stereoSamplesRendered = (*callback_)((short*)&realtimeBuffer_[cursor_], readCount / 4, sampleRate_);
 		int numBytesRendered = 2 * sizeof(short) * stereoSamplesRendered;
 
 		XAUDIO2_BUFFER xaudioBuffer{};

Windows/DSoundStream.cpp

@@ -124,7 +124,7 @@ int DSoundAudioBackend::RunThread() {
 		int numBytesToRender = RoundDown128(ModBufferSize(currentPos_ - lastPos_)); 
 
 		if (numBytesToRender >= 256) {
-			int numBytesRendered = 4 * (*callback_)(realtimeBuffer_, numBytesToRender >> 2, 16, 44100);
+			int numBytesRendered = 4 * (*callback_)(realtimeBuffer_, numBytesToRender >> 2, 44100);
 			//We need to copy the full buffer, regardless of what the mixer claims to have filled
 			//If we don't do this then the sound will loop if the sound stops and the mixer writes only zeroes
 			numBytesRendered = numBytesToRender;

Windows/WASAPIStream.cpp

@@ -501,11 +501,11 @@ void WASAPIAudioThread::Run() {
 			int chans = deviceFormat_->Format.nChannels;
 			switch (format_) {
 			case Format::IEEE_FLOAT:
-				callback_(shortBuf_, pNumAvFrames, 16, sampleRate_);
+				callback_(shortBuf_, pNumAvFrames, sampleRate_);
 				if (chans == 1) {
 					float *ptr = (float *)pData;
 					memset(ptr, 0, pNumAvFrames * chans * sizeof(float));
-					for (UINT32 i = 0; i < pNumAvFrames; i++) {
+					for (uint32_t i = 0; i < pNumAvFrames; i++) {
 						ptr[i * chans + 0] = 0.5f * ((float)shortBuf_[i * 2] + (float)shortBuf_[i * 2 + 1]) * (1.0f / 32768.0f);
 					}
 				} else if (chans == 2) {
@@ -513,14 +513,14 @@ void WASAPIAudioThread::Run() {
 				} else if (chans > 2) {
 					float *ptr = (float *)pData;
 					memset(ptr, 0, pNumAvFrames * chans * sizeof(float));
-					for (UINT32 i = 0; i < pNumAvFrames; i++) {
+					for (uint32_t i = 0; i < pNumAvFrames; i++) {
 						ptr[i * chans + 0] = (float)shortBuf_[i * 2] * (1.0f / 32768.0f);
 						ptr[i * chans + 1] = (float)shortBuf_[i * 2 + 1] * (1.0f / 32768.0f);
 					}
 				}
 				break;
 			case Format::PCM16:
-				callback_((short *)pData, pNumAvFrames, 16, sampleRate_);
+				callback_((short *)pData, pNumAvFrames, sampleRate_);
 				break;
 			}
 		}

Windows/WindowsAudio.h

@@ -3,8 +3,8 @@
 #include "Common/CommonWindows.h"
 #include "Core/ConfigValues.h"
 
-// Always 2 channels.
-typedef int(*StreamCallback)(short *buffer, int numSamples, int bits, int rate);
+// Always 2 channels, 16-bit audio.
+typedef int (*StreamCallback)(short *buffer, int numSamples, int rate);
 
 // Note that the backend may override the passed in sample rate. The actual sample rate
 // should be returned by GetSampleRate though.

android/jni/AndroidAudio.h

@@ -3,7 +3,7 @@
 #include <string>
 #include <mutex>
 
-typedef int (*AndroidAudioCallback)(short *buffer, int num_samples);
+typedef int (*AndroidAudioCallback)(short *buffer, int numSamples, int sampleRateHz);
 
 class AudioContext {
 public:
@@ -12,6 +12,8 @@ public:
 	virtual bool AudioRecord_Start(int sampleRate) { return false; };
 	virtual bool AudioRecord_Stop() { return false; };
 
+	int SampleRate() const { return sampleRate; }
+
 	virtual ~AudioContext() {}
 
 protected:

android/jni/OpenSLContext.cpp

@@ -50,7 +50,7 @@ void OpenSLContext::BqPlayerCallback(SLAndroidSimpleBufferQueueItf bq) {
 		return;
 	}
 
-	int renderedFrames = audioCallback(buffer[curBuffer], framesPerBuffer);
+	int renderedFrames = audioCallback(buffer[curBuffer], framesPerBuffer, SampleRate());
 
 	int sizeInBytes = framesPerBuffer * 2 * sizeof(short);
 	int byteCount = (framesPerBuffer - renderedFrames) * 4;

ios/AudioEngine.mm

@@ -19,7 +19,7 @@ static volatile BOOL done = 0;
 #define SAMPLE_SIZE 44100
 static short stream[SAMPLE_SIZE];
 
-int NativeMix(short *audio, int num_samples);
+int NativeMix(short *audio, int numSamples, int sampleRateHz);
 
 @interface AudioEngine ()
 
@@ -120,11 +120,12 @@ int NativeMix(short *audio, int num_samples);
 - (void)audioLoop
 {
     dispatch_async(dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_DEFAULT, 0), ^(void){
+        const int sampleRateHz = 44100;
         while (!done)
         {
             size_t frames_ready;
             if (![self playing])
-                frames_ready = NativeMix(stream, SAMPLE_SIZE / 2);
+                frames_ready = NativeMix(stream, SAMPLE_SIZE / 2, sampleRateHz);
             else
                 frames_ready = 0;
 
@@ -132,12 +133,12 @@ int NativeMix(short *audio, int num_samples);
             {
                 const size_t bytes_ready = frames_ready * sizeof(short) * 2;
                 alSourcei(source, AL_BUFFER, 0);
-                alBufferData(buffer, AL_FORMAT_STEREO16, stream, bytes_ready, 44100);
+                alBufferData(buffer, AL_FORMAT_STEREO16, stream, bytes_ready, sampleRateHz);
                 alSourcei(source, AL_BUFFER, buffer);
                 alSourcePlay(source);
 
                 // TODO: Maybe this could get behind?
-                usleep((1000000 * frames_ready) / 44100);
+                usleep((1000000 * frames_ready) / sampleRateHz);
             }
             else
                 usleep(100);

ios/iOSCoreAudio.mm

@@ -30,7 +30,7 @@
 
 AudioComponentInstance audioInstance = nil;
 
-int NativeMix(short *audio, int num_samples);
+int NativeMix(short *audio, int numSamples, int sampleRate);
 
 OSStatus iOSCoreAudioCallback(void *inRefCon,
 							  AudioUnitRenderActionFlags *ioActionFlags,
@@ -41,7 +41,7 @@ OSStatus iOSCoreAudioCallback(void *inRefCon,
 {
 	// see if we have any sound to play
 	short *output = (short *)ioData->mBuffers[0].mData;
-	UInt32 framesReady = NativeMix(output, inNumberFrames);
+	UInt32 framesReady = NativeMix(output, inNumberFrames, SAMPLE_RATE);
 	
 	if (framesReady == 0) {
 		// oops, we don't currently have any sound, so return silence