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Add samplerate argument to NativeMix

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This commit is contained in:
Henrik Rydgård 2023-03-24 13:59:27 +01:00
parent 8421c1e2a1
commit a5fdf3d05b
13 changed files with 38 additions and 37 deletions
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3
Common/System/NativeApp.h
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@ -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

1
Core/HLE/__sceAudio.cpp
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@ -482,6 +482,7 @@ void __PushExternalAudio(const s32 *audio, int numSamples) {
resampler.Clear();
}
}
#ifndef MOBILE_DEVICE
void __StartLogAudio(const Path& filename) {
if (!m_logAudio) {

3
Core/HLE/__sceAudio.h
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@ -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);

17
Qt/QtMain.cpp
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@ -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);
}

5
SDL/SDLMain.cpp
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@ -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;

13
UI/NativeApp.cpp
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@ -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 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.
@ -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

2
Windows/DSoundStream.cpp
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@ -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;

8
Windows/WASAPIStream.cpp
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@ -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;
}
}

4
Windows/WindowsAudio.h
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@ -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.

4
android/jni/AndroidAudio.h
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@ -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:

2
android/jni/OpenSLContext.cpp
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@ -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;

9
ios/AudioEngine.mm
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@ -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);

4
ios/iOSCoreAudio.mm
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@ -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