bsnes/ruby/audio/xaudio2.cpp

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#include "xaudio2.hpp"
#undef interface
struct AudioXAudio2 : AudioDriver, public IXAudio2VoiceCallback {
enum : uint { Buffers = 32 };
AudioXAudio2& self = *this;
AudioXAudio2(Audio& super) : AudioDriver(super) { construct(); }
~AudioXAudio2() { destruct(); }
auto create() -> bool override {
super.setDevice(hasDevices().first());
super.setChannels(2);
super.setFrequency(48000);
super.setLatency(40);
return initialize();
}
auto driver() -> string override { return "XAudio 2.1"; }
auto ready() -> bool override { return self.isReady; }
auto hasBlocking() -> bool override { return true; }
auto hasDynamic() -> bool override { return true; }
auto hasDevices() -> vector<string> override {
vector<string> devices;
for(auto& device : self.devices) devices.append(device.name);
return devices;
}
Update to v104r06 release. byuu says: Changelog: - gba,ws: removed Thread::step() override¹ - processor/m68k: move.b (a7)+ and move.b (a7)- adjust a7 by two, not by one² - tomoko: created new initialize(Video,Audio,Input)Driver() functions³ - ruby/audio: split Audio::information into Audio::available(Devices,Frequencies,Latencies,Channels)³ - ws: added Model::(WonderSwan,WonderSwanColor,SwanCrystal)() functions for consistency with other cores ¹: this should hopefully fix GBA Pokemon Pinball. Thanks to SuperMikeMan for pointing out the underlying cause. ²: this fixes A Ressaha de Ikou, Mega Bomberman, and probably more games. ³: this is the big change: so there was a problem with WASAPI where you might change your device under the audio settings panel. And your new device may not support the frequency that your old device used. This would end up not updating the frequency, and the pitch would be distorted. The old Audio::information() couldn't tell you what frequencies, latencies, or channels were available for all devices simultaneously, so I had to split them up. The new initializeAudioDriver() function validates you have a correct driver, or it defaults to none. Then it validates a correct device name, or it defaults to the first entry in the list. Then it validates a correct frequency, or defaults to the first in the list. Then finally it validates a correct latency, or defaults to the first in the list. In this way ... we have a clear path now with no API changes required to select default devices, frequencies, latencies, channel counts: they need to be the first items in their respective lists. So, what we need to do now is go through and for every audio driver that enumerates devices, we need to make sure the default device gets added to the top of the list. I'm ... not really sure how to do this with most drivers, so this is definitely going to take some time. Also, when you change a device, initializeAudioDriver() is called again, so if it's a bad device, it will disable the audio driver instead of continuing to send samples at it and hoping that the driver blocked those API calls when it failed to initialize properly. Now then ... since it was a decently-sized API change, it's possible I've broken compilation of the Linux drivers, so please report any compilation errors so that I can fix them.
2017-08-26 01:15:49 +00:00
auto hasFrequencies() -> vector<uint> override {
return {44100, 48000, 96000};
Update to v104r06 release. byuu says: Changelog: - gba,ws: removed Thread::step() override¹ - processor/m68k: move.b (a7)+ and move.b (a7)- adjust a7 by two, not by one² - tomoko: created new initialize(Video,Audio,Input)Driver() functions³ - ruby/audio: split Audio::information into Audio::available(Devices,Frequencies,Latencies,Channels)³ - ws: added Model::(WonderSwan,WonderSwanColor,SwanCrystal)() functions for consistency with other cores ¹: this should hopefully fix GBA Pokemon Pinball. Thanks to SuperMikeMan for pointing out the underlying cause. ²: this fixes A Ressaha de Ikou, Mega Bomberman, and probably more games. ³: this is the big change: so there was a problem with WASAPI where you might change your device under the audio settings panel. And your new device may not support the frequency that your old device used. This would end up not updating the frequency, and the pitch would be distorted. The old Audio::information() couldn't tell you what frequencies, latencies, or channels were available for all devices simultaneously, so I had to split them up. The new initializeAudioDriver() function validates you have a correct driver, or it defaults to none. Then it validates a correct device name, or it defaults to the first entry in the list. Then it validates a correct frequency, or defaults to the first in the list. Then finally it validates a correct latency, or defaults to the first in the list. In this way ... we have a clear path now with no API changes required to select default devices, frequencies, latencies, channel counts: they need to be the first items in their respective lists. So, what we need to do now is go through and for every audio driver that enumerates devices, we need to make sure the default device gets added to the top of the list. I'm ... not really sure how to do this with most drivers, so this is definitely going to take some time. Also, when you change a device, initializeAudioDriver() is called again, so if it's a bad device, it will disable the audio driver instead of continuing to send samples at it and hoping that the driver blocked those API calls when it failed to initialize properly. Now then ... since it was a decently-sized API change, it's possible I've broken compilation of the Linux drivers, so please report any compilation errors so that I can fix them.
2017-08-26 01:15:49 +00:00
}
auto hasLatencies() -> vector<uint> override {
Update to v104r06 release. byuu says: Changelog: - gba,ws: removed Thread::step() override¹ - processor/m68k: move.b (a7)+ and move.b (a7)- adjust a7 by two, not by one² - tomoko: created new initialize(Video,Audio,Input)Driver() functions³ - ruby/audio: split Audio::information into Audio::available(Devices,Frequencies,Latencies,Channels)³ - ws: added Model::(WonderSwan,WonderSwanColor,SwanCrystal)() functions for consistency with other cores ¹: this should hopefully fix GBA Pokemon Pinball. Thanks to SuperMikeMan for pointing out the underlying cause. ²: this fixes A Ressaha de Ikou, Mega Bomberman, and probably more games. ³: this is the big change: so there was a problem with WASAPI where you might change your device under the audio settings panel. And your new device may not support the frequency that your old device used. This would end up not updating the frequency, and the pitch would be distorted. The old Audio::information() couldn't tell you what frequencies, latencies, or channels were available for all devices simultaneously, so I had to split them up. The new initializeAudioDriver() function validates you have a correct driver, or it defaults to none. Then it validates a correct device name, or it defaults to the first entry in the list. Then it validates a correct frequency, or defaults to the first in the list. Then finally it validates a correct latency, or defaults to the first in the list. In this way ... we have a clear path now with no API changes required to select default devices, frequencies, latencies, channel counts: they need to be the first items in their respective lists. So, what we need to do now is go through and for every audio driver that enumerates devices, we need to make sure the default device gets added to the top of the list. I'm ... not really sure how to do this with most drivers, so this is definitely going to take some time. Also, when you change a device, initializeAudioDriver() is called again, so if it's a bad device, it will disable the audio driver instead of continuing to send samples at it and hoping that the driver blocked those API calls when it failed to initialize properly. Now then ... since it was a decently-sized API change, it's possible I've broken compilation of the Linux drivers, so please report any compilation errors so that I can fix them.
2017-08-26 01:15:49 +00:00
return {20, 40, 60, 80, 100};
}
auto setDevice(string device) -> bool override { return initialize(); }
auto setBlocking(bool blocking) -> bool override { return true; }
auto setFrequency(uint frequency) -> bool override { return initialize(); }
auto setLatency(uint latency) -> bool override { return initialize(); }
auto clear() -> void override {
self.sourceVoice->Stop(0);
self.sourceVoice->FlushSourceBuffers(); //calls OnBufferEnd for all currently submitted buffers
self.index = 0;
self.queue = 0;
for(uint n : range(Buffers)) self.buffers[n].fill();
self.sourceVoice->Start(0);
}
auto level() -> double override {
XAUDIO2_VOICE_STATE state{};
self.sourceVoice->GetState(&state);
uint level = state.BuffersQueued * self.period + buffers[self.index].size() - state.SamplesPlayed % self.period;
uint limit = Buffers * self.period;
return (double)level / limit;
}
auto output(const double samples[]) -> void override {
uint32_t frame = 0;
frame |= (uint16_t)sclamp<16>(samples[0] * 32767.0) << 0;
frame |= (uint16_t)sclamp<16>(samples[1] * 32767.0) << 16;
auto& buffer = self.buffers[self.index];
buffer.write(frame);
if(!buffer.full()) return;
buffer.flush();
if(self.queue == Buffers - 1) {
if(self.blocking) {
//wait until there is at least one other free buffer for the next sample
while(self.queue == Buffers - 1);
} else {
//there is no free buffer for the next block, so ignore the current contents
return;
}
}
write(buffer.data(), buffer.capacity<uint8_t>());
self.index = (self.index + 1) % Buffers;
}
private:
struct Device {
uint id = 0;
uint channels = 0;
uint frequency = 0;
Format format = Format::none;
string name;
};
vector<Device> devices;
auto construct() -> void {
XAudio2Create(&self.interface, 0 , XAUDIO2_DEFAULT_PROCESSOR);
uint deviceCount = 0;
self.interface->GetDeviceCount(&deviceCount);
for(uint deviceIndex : range(deviceCount)) {
XAUDIO2_DEVICE_DETAILS deviceDetails{};
self.interface->GetDeviceDetails(deviceIndex, &deviceDetails);
auto format = deviceDetails.OutputFormat.Format.wFormatTag;
auto bits = deviceDetails.OutputFormat.Format.wBitsPerSample;
Device device;
device.id = deviceIndex;
device.name = (const char*)utf8_t(deviceDetails.DisplayName);
device.channels = deviceDetails.OutputFormat.Format.nChannels;
device.frequency = deviceDetails.OutputFormat.Format.nSamplesPerSec;
if(format == WAVE_FORMAT_PCM) {
if(bits == 16) device.format = Format::int16;
if(bits == 32) device.format = Format::int32;
} else if(format == WAVE_FORMAT_IEEE_FLOAT) {
if(bits == 32) device.format = Format::float32;
}
//ensure devices.first() is the default device
if(deviceDetails.Role & DefaultGameDevice) {
devices.prepend(device);
} else {
devices.append(device);
}
}
}
auto destruct() -> void {
terminate();
if(self.interface) {
self.interface->Release();
self.interface = nullptr;
}
}
auto initialize() -> bool {
terminate();
if(!self.interface) return false;
self.period = self.frequency * self.latency / Buffers / 1000.0 + 0.5;
for(uint n : range(Buffers)) buffers[n].resize(self.period);
self.index = 0;
self.queue = 0;
if(!hasDevices().find(self.device)) self.device = hasDevices().first();
uint deviceID = devices[hasDevices().find(self.device)()].id;
if(FAILED(self.interface->CreateMasteringVoice(&self.masterVoice, self.channels, self.frequency, 0, deviceID, nullptr))) return terminate(), false;
WAVEFORMATEX waveFormat{};
waveFormat.wFormatTag = WAVE_FORMAT_PCM;
waveFormat.nChannels = self.channels;
waveFormat.nSamplesPerSec = self.frequency;
waveFormat.nBlockAlign = 4;
waveFormat.wBitsPerSample = 16;
waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
waveFormat.cbSize = 0;
if(FAILED(self.interface->CreateSourceVoice(&self.sourceVoice, (WAVEFORMATEX*)&waveFormat, XAUDIO2_VOICE_NOSRC, XAUDIO2_DEFAULT_FREQ_RATIO, this, nullptr, nullptr))) return terminate(), false;
clear();
return self.isReady = true;
}
auto terminate() -> void {
self.isReady = false;
if(self.sourceVoice) {
self.sourceVoice->Stop(0);
self.sourceVoice->DestroyVoice();
self.sourceVoice = nullptr;
}
if(self.masterVoice) {
self.masterVoice->DestroyVoice();
self.masterVoice = nullptr;
}
}
auto write(const uint32_t* audioData, uint bytes) -> void {
XAUDIO2_BUFFER buffer{};
buffer.AudioBytes = bytes;
buffer.pAudioData = (const BYTE*)audioData;
buffer.pContext = nullptr;
InterlockedIncrement(&self.queue);
self.sourceVoice->SubmitSourceBuffer(&buffer);
}
bool isReady = false;
queue<uint32_t> buffers[Buffers];
uint period = 0; //amount (in 32-bit frames) of samples per buffer
uint index = 0; //current buffer for writing samples to
volatile long queue = 0; //how many buffers are queued and ready for playback
IXAudio2* interface = nullptr;
IXAudio2MasteringVoice* masterVoice = nullptr;
IXAudio2SourceVoice* sourceVoice = nullptr;
//inherited from IXAudio2VoiceCallback
STDMETHODIMP_(void) OnBufferStart(void* pBufferContext){}
STDMETHODIMP_(void) OnLoopEnd(void* pBufferContext){}
STDMETHODIMP_(void) OnStreamEnd() {}
STDMETHODIMP_(void) OnVoiceError(void* pBufferContext, HRESULT Error) {}
STDMETHODIMP_(void) OnVoiceProcessingPassEnd() {}
STDMETHODIMP_(void) OnVoiceProcessingPassStart(UINT32 BytesRequired) {}
STDMETHODIMP_(void) OnBufferEnd(void* pBufferContext) {
InterlockedDecrement(&self.queue);
}
};