Add samplerate argument to NativeMix

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);
+int NativeMix(short *audio, int num_samples, int sampleRateHz);
-void NativeSetMixer(void* mixer);
 
 // 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

@@ -482,6 +482,7 @@ void __PushExternalAudio(const s32 *audio, int numSamples) {
 		resampler.Clear();
 	}
 }
+
 #ifndef MOBILE_DEVICE
 void __StartLogAudio(const Path& filename) {
 	if (!m_logAudio) {

Core/HLE/__sceAudio.h

@@ -41,6 +41,9 @@ void __AudioShutdown();
 void __AudioSetOutputFrequency(int freq);
 void __AudioSetSRCFrequency(int freq);
 
+typedef void(*AudioUserCallback);
+void __AudioSetUserCallback(AudioUserCallback callback);
+
 // May return SCE_ERROR_AUDIO_CHANNEL_BUSY if buffer too large
 u32 __AudioEnqueue(AudioChannel &chan, int chanNum, bool blocking);
 void __AudioWakeThreads(AudioChannel &chan, int result, int step);

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/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;
@@ -235,9 +229,8 @@ std::string NativeQueryConfig(std::string query) {
 	}
 }
 
-int NativeMix(short *audio, int num_samples) {
+int NativeMix(short *audio, int numSamples, int sampleRateHz) {
-	int sample_rate = System_GetPropertyInt(SYSPROP_AUDIO_SAMPLE_RATE);
+	return __AudioMix(audio, numSamples, sampleRateHz);
-	return __AudioMix(audio, num_samples, sample_rate > 0 ? sample_rate : 44100);
 }
 
 // This is called before NativeInit so we do a little bit of initialization here.
@@ -859,7 +852,7 @@ bool NativeInitGraphics(GraphicsContext *graphicsContext) {
 #if PPSSPP_PLATFORM(UWP)
 	winAudioBackend->Init(0, &Win32Mix, 44100);
 #else
-	winAudioBackend->Init(MainWindow::GetHWND(), &Win32Mix, 44100);
+	winAudioBackend->Init(MainWindow::GetHWND(), &NativeMix, 44100);
 #endif
 #endif
 

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.
+// Always 2 channels, 16-bit audio.
-typedef int(*StreamCallback)(short *buffer, int numSamples, int bits, int rate);
+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