Files
FlaxEngine/Source/Engine/Audio/XAudio2/AudioBackendXAudio2.cpp
Wojtek Figat 1ffe8a7b60 Add OpenAL AL_SOFT_source_spatialize extension support for stereo spatial audio playback
Add `AllowSpatialization` option to `AudioSource`
2023-04-20 15:01:22 +02:00

858 lines
24 KiB
C++

// Copyright (c) 2012-2023 Wojciech Figat. All rights reserved.
#if AUDIO_API_XAUDIO2
#include "AudioBackendXAudio2.h"
#include "Engine/Audio/AudioSettings.h"
#include "Engine/Core/Collections/Array.h"
#include "Engine/Core/Log.h"
#include "Engine/Audio/Audio.h"
#include "Engine/Audio/AudioSource.h"
#include "Engine/Audio/AudioListener.h"
#if PLATFORM_WINDOWS
// Tweak Win ver
#define _WIN32_WINNT 0x0602
//#include "Engine/Platform/Windows/IncludeWindowsHeaders.h"
#endif
// Include XAudio library
// Documentation: https://docs.microsoft.com/en-us/windows/desktop/xaudio2/xaudio2-apis-portal
#include <xaudio2.h>
//#include <xaudio2fx.h>
#include <x3daudio.h>
#define MAX_INPUT_CHANNELS 2
#define MAX_OUTPUT_CHANNELS 8
#define MAX_CHANNELS_MATRIX_SIZE (MAX_INPUT_CHANNELS*MAX_OUTPUT_CHANNELS)
#define FLAX_COORD_SCALE 0.01f // units are meters
#define FLAX_DST_TO_XAUDIO(x) x * FLAX_COORD_SCALE
#define FLAX_POS_TO_XAUDIO(vec) X3DAUDIO_VECTOR(vec.X * FLAX_COORD_SCALE, vec.Y * FLAX_COORD_SCALE, vec.Z * FLAX_COORD_SCALE)
#define FLAX_VEL_TO_XAUDIO(vec) X3DAUDIO_VECTOR(vec.X * (FLAX_COORD_SCALE*FLAX_COORD_SCALE), vec.Y * (FLAX_COORD_SCALE*FLAX_COORD_SCALE), vec.Z * (FLAX_COORD_SCALE*FLAX_COORD_SCALE))
#define FLAX_VEC_TO_XAUDIO(vec) (*((X3DAUDIO_VECTOR*)&vec))
namespace XAudio2
{
struct Listener
{
AudioListener* AudioListener;
X3DAUDIO_LISTENER Data;
Listener()
{
Init();
}
void Init()
{
AudioListener = nullptr;
Data.pCone = nullptr;
}
bool IsFree() const
{
return AudioListener == nullptr;
}
void UpdateTransform()
{
const Vector3& position = AudioListener->GetPosition();
const Quaternion& orientation = AudioListener->GetOrientation();
const Vector3 front = orientation * Vector3::Forward;
const Vector3 top = orientation * Vector3::Up;
Data.OrientFront = FLAX_VEC_TO_XAUDIO(front);
Data.OrientTop = FLAX_VEC_TO_XAUDIO(top);
Data.Position = FLAX_POS_TO_XAUDIO(position);
}
void UpdateVelocity()
{
const Vector3& velocity = AudioListener->GetVelocity();
Data.Velocity = FLAX_VEL_TO_XAUDIO(velocity);
}
};
class VoiceCallback : public IXAudio2VoiceCallback
{
COM_DECLSPEC_NOTHROW void STDMETHODCALLTYPE OnVoiceProcessingPassStart(THIS_ UINT32 BytesRequired) override
{
}
COM_DECLSPEC_NOTHROW void STDMETHODCALLTYPE OnVoiceProcessingPassEnd(THIS) override
{
}
COM_DECLSPEC_NOTHROW void STDMETHODCALLTYPE OnStreamEnd(THIS_) override
{
}
COM_DECLSPEC_NOTHROW void STDMETHODCALLTYPE OnBufferStart(THIS_ void* pBufferContext) override
{
PeekSamples();
}
COM_DECLSPEC_NOTHROW void STDMETHODCALLTYPE OnBufferEnd(THIS_ void* pBufferContext) override;
COM_DECLSPEC_NOTHROW void STDMETHODCALLTYPE OnLoopEnd(THIS_ void* pBufferContext) override
{
}
COM_DECLSPEC_NOTHROW void STDMETHODCALLTYPE OnVoiceError(THIS_ void* pBufferContext, HRESULT Error) override
{
LOG(Warning, "IXAudio2VoiceCallback::OnVoiceError! Error: 0x{0:x}", Error);
}
public:
AudioSource* Source;
void PeekSamples();
};
struct Source
{
IXAudio2SourceVoice* Voice;
X3DAUDIO_EMITTER Data;
WAVEFORMATEX Format;
XAUDIO2_SEND_DESCRIPTOR Destination;
float Pitch;
float StartTime;
uint64 LastBufferStartSamplesPlayed;
int32 BuffersProcessed;
bool IsDirty;
bool Is3D;
bool IsPlaying;
VoiceCallback Callback;
Source()
{
Init();
}
void Init()
{
Voice = nullptr;
Platform::MemoryClear(&Data, sizeof(Data));
Data.CurveDistanceScaler = 1.0f;
Destination.Flags = 0;
Destination.pOutputVoice = nullptr;
Pitch = 1.0f;
StartTime = 0.0f;
IsDirty = false;
Is3D = false;
IsPlaying = false;
LastBufferStartSamplesPlayed = 0;
BuffersProcessed = 0;
}
bool IsFree() const
{
return Voice == nullptr;
}
void UpdateTransform(const AudioSource* source)
{
const Vector3& position = source->GetPosition();
const Quaternion& orientation = source->GetOrientation();
const Vector3 front = orientation * Vector3::Forward;
const Vector3 top = orientation * Vector3::Up;
Data.OrientFront = FLAX_VEC_TO_XAUDIO(front);
Data.OrientTop = FLAX_VEC_TO_XAUDIO(top);
Data.Position = FLAX_POS_TO_XAUDIO(position);
}
void UpdateVelocity(const AudioSource* source)
{
const Vector3& velocity = source->GetVelocity();
Data.Velocity = FLAX_VEL_TO_XAUDIO(velocity);
}
};
struct Buffer
{
AudioDataInfo Info;
Array<byte> Data;
};
class EngineCallback : public IXAudio2EngineCallback
{
COM_DECLSPEC_NOTHROW void STDMETHODCALLTYPE OnProcessingPassEnd() override
{
}
COM_DECLSPEC_NOTHROW void STDMETHODCALLTYPE OnProcessingPassStart() override
{
}
COM_DECLSPEC_NOTHROW void STDMETHODCALLTYPE OnCriticalError(HRESULT error) override
{
LOG(Warning, "IXAudio2EngineCallback::OnCriticalError! Error: 0x{0:x}", error);
}
};
IXAudio2* Instance = nullptr;
IXAudio2MasteringVoice* MasteringVoice = nullptr;
X3DAUDIO_HANDLE X3DInstance;
DWORD ChannelMask;
UINT32 SampleRate;
UINT32 Channels;
bool ForceDirty = true;
Listener Listeners[AUDIO_MAX_LISTENERS];
Array<Source> Sources(32); // TODO: use ChunkedArray for better performance
Array<Buffer*> Buffers(64); // TODO: use ChunkedArray for better performance or use buffers pool?
EngineCallback Callback;
Listener* GetListener()
{
for (int32 i = 0; i < AUDIO_MAX_LISTENERS; i++)
{
if (Listeners[i].AudioListener)
return &Listeners[i];
}
return nullptr;
}
Listener* GetListener(const AudioListener* listener)
{
for (int32 i = 0; i < AUDIO_MAX_LISTENERS; i++)
{
if (Listeners[i].AudioListener == listener)
return &Listeners[i];
}
return nullptr;
}
Source* GetSource(const AudioSource* source)
{
if (source->SourceIDs.Count() == 0)
return nullptr;
const AUDIO_SOURCE_ID_TYPE sourceId = source->SourceIDs[0];
// 0 is invalid ID so shift them
return &Sources[sourceId - 1];
}
void MarkAllDirty()
{
ForceDirty = true;
}
void QueueBuffer(Source* aSource, const AudioSource* source, const int32 bufferId, XAUDIO2_BUFFER& buffer)
{
Buffer* aBuffer = Buffers[bufferId - 1];
buffer.pAudioData = aBuffer->Data.Get();
buffer.AudioBytes = aBuffer->Data.Count();
if (aSource->StartTime > ZeroTolerance)
{
buffer.PlayBegin = (UINT32)(aSource->StartTime * (aBuffer->Info.SampleRate * aBuffer->Info.NumChannels));
buffer.PlayLength = aBuffer->Info.NumSamples / aBuffer->Info.NumChannels - buffer.PlayBegin;
aSource->StartTime = 0;
}
const HRESULT hr = aSource->Voice->SubmitSourceBuffer(&buffer);
if (FAILED(hr))
{
LOG(Warning, "XAudio2: Failed to submit source buffer (error: 0x{0:x})", hr);
}
}
void VoiceCallback::OnBufferEnd(void* pBufferContext)
{
auto aSource = GetSource(Source);
if (aSource->IsPlaying)
aSource->BuffersProcessed++;
}
void VoiceCallback::PeekSamples()
{
auto aSource = GetSource(Source);
XAUDIO2_VOICE_STATE state;
aSource->Voice->GetState(&state);
aSource->LastBufferStartSamplesPlayed = state.SamplesPlayed;
}
}
void AudioBackendXAudio2::Listener_OnAdd(AudioListener* listener)
{
// Get first free listener
XAudio2::Listener* aListener = nullptr;
for (int32 i = 0; i < AUDIO_MAX_LISTENERS; i++)
{
if (XAudio2::Listeners[i].IsFree())
{
aListener = &XAudio2::Listeners[i];
break;
}
}
ASSERT(aListener);
// Setup
aListener->AudioListener = listener;
aListener->UpdateTransform();
aListener->UpdateVelocity();
XAudio2::MarkAllDirty();
}
void AudioBackendXAudio2::Listener_OnRemove(AudioListener* listener)
{
// Free listener
XAudio2::Listener* aListener = XAudio2::GetListener(listener);
if (aListener)
{
aListener->Init();
XAudio2::MarkAllDirty();
}
}
void AudioBackendXAudio2::Listener_VelocityChanged(AudioListener* listener)
{
XAudio2::Listener* aListener = XAudio2::GetListener(listener);
if (aListener)
{
aListener->UpdateVelocity();
XAudio2::MarkAllDirty();
}
}
void AudioBackendXAudio2::Listener_TransformChanged(AudioListener* listener)
{
XAudio2::Listener* aListener = XAudio2::GetListener(listener);
if (aListener)
{
aListener->UpdateTransform();
XAudio2::MarkAllDirty();
}
}
void AudioBackendXAudio2::Listener_ReinitializeAll()
{
// TODO: Implement XAudio2 reinitialization; read HRTF audio value from Audio class
}
void AudioBackendXAudio2::Source_OnAdd(AudioSource* source)
{
// Skip if has no clip (needs audio data to create a source - needs data format information)
if (source->Clip == nullptr || !source->Clip->IsLoaded())
return;
auto clip = source->Clip.Get();
// Get first free source
XAudio2::Source* aSource = nullptr;
AUDIO_SOURCE_ID_TYPE sourceID;
for (int32 i = 0; i < XAudio2::Sources.Count(); i++)
{
if (XAudio2::Sources[i].IsFree())
{
sourceID = i;
aSource = &XAudio2::Sources[i];
break;
}
}
if (aSource == nullptr)
{
// Add new
const XAudio2::Source src;
sourceID = XAudio2::Sources.Count();
XAudio2::Sources.Add(src);
aSource = &XAudio2::Sources[sourceID];
}
// Initialize audio data format information (from clip)
auto& header = clip->AudioHeader;
auto& format = aSource->Format;
format.wFormatTag = WAVE_FORMAT_PCM;
format.nChannels = source->Is3D() ? 1 : header.Info.NumChannels; // 3d audio is always mono (AudioClip auto-converts before buffer write)
format.nSamplesPerSec = header.Info.SampleRate;
format.wBitsPerSample = header.Info.BitDepth;
format.nBlockAlign = (WORD)(format.nChannels * (format.wBitsPerSample / 8));
format.nAvgBytesPerSec = format.nSamplesPerSec * format.nBlockAlign;
format.cbSize = 0;
// Setup dry effect
aSource->Destination.pOutputVoice = XAudio2::MasteringVoice;
// Create voice
const XAUDIO2_VOICE_SENDS sendList =
{
1,
&aSource->Destination
};
const HRESULT hr = XAudio2::Instance->CreateSourceVoice(&aSource->Voice, &aSource->Format, 0, 2.0f, &aSource->Callback, &sendList);
if (FAILED(hr))
{
LOG(Error, "Failed to create XAudio2 voice. Error: 0x{0:x}", hr);
return;
}
// Prepare source state
aSource->Callback.Source = source;
aSource->IsDirty = true;
aSource->Data.ChannelCount = format.nChannels;
aSource->Data.InnerRadius = FLAX_DST_TO_XAUDIO(source->GetMinDistance());
aSource->Is3D = source->Is3D();
aSource->Pitch = source->GetPitch();
aSource->UpdateTransform(source);
aSource->UpdateVelocity(source);
// 0 is invalid ID so shift them
sourceID++;
source->SourceIDs.Add(sourceID);
source->Restore();
}
void AudioBackendXAudio2::Source_OnRemove(AudioSource* source)
{
source->Cleanup();
}
void AudioBackendXAudio2::Source_VelocityChanged(AudioSource* source)
{
auto aSource = XAudio2::GetSource(source);
if (aSource)
{
aSource->UpdateVelocity(source);
aSource->IsDirty = true;
}
}
void AudioBackendXAudio2::Source_TransformChanged(AudioSource* source)
{
auto aSource = XAudio2::GetSource(source);
if (aSource)
{
aSource->UpdateTransform(source);
aSource->IsDirty = true;
}
}
void AudioBackendXAudio2::Source_VolumeChanged(AudioSource* source)
{
auto aSource = XAudio2::GetSource(source);
if (aSource && aSource->Voice)
{
aSource->Voice->SetVolume(source->GetVolume());
}
}
void AudioBackendXAudio2::Source_PitchChanged(AudioSource* source)
{
auto aSource = XAudio2::GetSource(source);
if (aSource)
{
aSource->Pitch = source->GetPitch();
aSource->IsDirty = true;
}
}
void AudioBackendXAudio2::Source_IsLoopingChanged(AudioSource* source)
{
auto aSource = XAudio2::GetSource(source);
if (!aSource || !aSource->Voice)
return;
// Skip if has no buffers (waiting for data or sth)
XAUDIO2_VOICE_STATE state;
aSource->Voice->GetState(&state);
if (state.BuffersQueued == 0)
return;
// Looping is defined during buffer submission so reset source buffer (this is called only for non-streamable sources that ue single buffer)
const uint32 bufferId = source->Clip->Buffers[0];
XAudio2::Buffer* aBuffer = XAudio2::Buffers[bufferId - 1];
const bool isPlaying = source->IsActuallyPlayingSth();
if (isPlaying)
aSource->Voice->Stop();
aSource->Voice->FlushSourceBuffers();
aSource->LastBufferStartSamplesPlayed = 0;
aSource->BuffersProcessed = 0;
XAUDIO2_BUFFER buffer = { 0 };
buffer.pContext = aBuffer;
buffer.Flags = XAUDIO2_END_OF_STREAM;
if (source->GetIsLooping())
buffer.LoopCount = XAUDIO2_LOOP_INFINITE;
// Restore play position
const UINT32 totalSamples = aBuffer->Info.NumSamples / aBuffer->Info.NumChannels;
buffer.PlayBegin = state.SamplesPlayed % totalSamples;
buffer.PlayLength = totalSamples - buffer.PlayBegin;
aSource->StartTime = 0;
XAudio2::QueueBuffer(aSource, source, bufferId, buffer);
if (isPlaying)
aSource->Voice->Start();
}
void AudioBackendXAudio2::Source_SpatialSetupChanged(AudioSource* source)
{
auto aSource = XAudio2::GetSource(source);
if (aSource)
{
// TODO: implement attenuation settings for 3d audio
aSource->Data.InnerRadius = FLAX_DST_TO_XAUDIO(source->GetMinDistance());
aSource->IsDirty = true;
}
}
void AudioBackendXAudio2::Source_ClipLoaded(AudioSource* source)
{
auto aSource = XAudio2::GetSource(source);
if (!aSource)
{
// Register source if clip was missing
Source_OnAdd(source);
}
}
void AudioBackendXAudio2::Source_Cleanup(AudioSource* source)
{
auto aSource = XAudio2::GetSource(source);
if (!aSource)
return;
// Free source
if (aSource->Voice)
{
aSource->Voice->DestroyVoice();
}
aSource->Init();
}
void AudioBackendXAudio2::Source_Play(AudioSource* source)
{
auto aSource = XAudio2::GetSource(source);
if (aSource && aSource->Voice && !aSource->IsPlaying)
{
// Play
aSource->Voice->Start();
aSource->IsPlaying = true;
}
}
void AudioBackendXAudio2::Source_Pause(AudioSource* source)
{
auto aSource = XAudio2::GetSource(source);
if (aSource && aSource->Voice && aSource->IsPlaying)
{
// Pause
aSource->Voice->Stop();
aSource->IsPlaying = false;
}
}
void AudioBackendXAudio2::Source_Stop(AudioSource* source)
{
auto aSource = XAudio2::GetSource(source);
if (aSource && aSource->Voice)
{
aSource->StartTime = 0.0f;
// Pause
aSource->Voice->Stop();
aSource->IsPlaying = false;
// Unset streaming buffers to rewind
aSource->Voice->FlushSourceBuffers();
aSource->BuffersProcessed = 0;
aSource->Callback.PeekSamples();
}
}
void AudioBackendXAudio2::Source_SetCurrentBufferTime(AudioSource* source, float value)
{
const auto aSource = XAudio2::GetSource(source);
if (aSource)
{
// Store start time so next buffer submitted will start from here (assumes audio is stopped)
aSource->StartTime = value;
}
}
float AudioBackendXAudio2::Source_GetCurrentBufferTime(const AudioSource* source)
{
float time = 0;
auto aSource = XAudio2::GetSource(source);
if (aSource)
{
ASSERT(source->Clip && source->Clip->IsLoaded());
const auto& clipInfo = source->Clip->AudioHeader.Info;
XAUDIO2_VOICE_STATE state;
aSource->Voice->GetState(&state);
const uint32 numChannels = source->Is3D() ? 1 : clipInfo.NumChannels; // 3d audio is always mono (AudioClip auto-converts before buffer write)
const UINT32 totalSamples = clipInfo.NumSamples / numChannels;
state.SamplesPlayed -= aSource->LastBufferStartSamplesPlayed % totalSamples; // Offset by the last buffer start to get time relative to its begin
time = aSource->StartTime + (state.SamplesPlayed % totalSamples) / static_cast<float>(Math::Max(1U, clipInfo.SampleRate));
}
return time;
}
void AudioBackendXAudio2::Source_SetNonStreamingBuffer(AudioSource* source)
{
auto aSource = XAudio2::GetSource(source);
if (!aSource)
return;
const uint32 bufferId = source->Clip->Buffers[0];
XAudio2::Buffer* aBuffer = XAudio2::Buffers[bufferId - 1];
XAUDIO2_BUFFER buffer = { 0 };
buffer.pContext = aBuffer;
buffer.Flags = XAUDIO2_END_OF_STREAM;
if (source->GetIsLooping())
buffer.LoopCount = XAUDIO2_LOOP_INFINITE;
// Queue single buffer
XAudio2::QueueBuffer(aSource, source, bufferId, buffer);
}
void AudioBackendXAudio2::Source_GetProcessedBuffersCount(AudioSource* source, int32& processedBuffersCount)
{
processedBuffersCount = 0;
auto aSource = XAudio2::GetSource(source);
if (aSource && aSource->Voice)
{
processedBuffersCount = aSource->BuffersProcessed;
}
}
void AudioBackendXAudio2::Source_GetQueuedBuffersCount(AudioSource* source, int32& queuedBuffersCount)
{
queuedBuffersCount = 0;
auto aSource = XAudio2::GetSource(source);
if (aSource && aSource->Voice)
{
XAUDIO2_VOICE_STATE state;
aSource->Voice->GetState(&state, XAUDIO2_VOICE_NOSAMPLESPLAYED);
queuedBuffersCount = state.BuffersQueued;
}
}
void AudioBackendXAudio2::Source_QueueBuffer(AudioSource* source, uint32 bufferId)
{
auto aSource = XAudio2::GetSource(source);
if (!aSource)
return;
XAudio2::Buffer* aBuffer = XAudio2::Buffers[bufferId - 1];
XAUDIO2_BUFFER buffer = { 0 };
buffer.pContext = aBuffer;
XAudio2::QueueBuffer(aSource, source, bufferId, buffer);
}
void AudioBackendXAudio2::Source_DequeueProcessedBuffers(AudioSource* source)
{
auto aSource = XAudio2::GetSource(source);
if (aSource && aSource->Voice)
{
const HRESULT hr = aSource->Voice->FlushSourceBuffers();
if (FAILED(hr))
{
LOG(Warning, "XAudio2: FlushSourceBuffers failed. Error: 0x{0:x}", hr);
}
aSource->BuffersProcessed = 0;
}
}
void AudioBackendXAudio2::Buffer_Create(uint32& bufferId)
{
// Get first free buffer slot
XAudio2::Buffer* aBuffer = nullptr;
for (int32 i = 0; i < XAudio2::Buffers.Count(); i++)
{
if (XAudio2::Buffers[i] == nullptr)
{
aBuffer = New<XAudio2::Buffer>();
XAudio2::Buffers[i] = aBuffer;
bufferId = i + 1;
break;
}
}
if (!aBuffer)
{
// Add new slot
aBuffer = New<XAudio2::Buffer>();
XAudio2::Buffers.Add(aBuffer);
bufferId = XAudio2::Buffers.Count();
}
aBuffer->Data.Resize(0);
}
void AudioBackendXAudio2::Buffer_Delete(uint32& bufferId)
{
XAudio2::Buffer*& aBuffer = XAudio2::Buffers[bufferId - 1];
aBuffer->Data.Resize(0);
Delete(aBuffer);
aBuffer = nullptr;
}
void AudioBackendXAudio2::Buffer_Write(uint32 bufferId, byte* samples, const AudioDataInfo& info)
{
XAudio2::Buffer* aBuffer = XAudio2::Buffers[bufferId - 1];
const uint32 bytesPerSample = info.BitDepth / 8;
const int32 samplesLength = info.NumSamples * bytesPerSample;
aBuffer->Info = info;
aBuffer->Data.Set(samples, samplesLength);
}
const Char* AudioBackendXAudio2::Base_Name()
{
return TEXT("XAudio2");
}
AudioBackend::FeatureFlags AudioBackendXAudio2::Base_Features()
{
return FeatureFlags::None;
}
void AudioBackendXAudio2::Base_OnActiveDeviceChanged()
{
}
void AudioBackendXAudio2::Base_SetDopplerFactor(float value)
{
XAudio2::MarkAllDirty();
}
void AudioBackendXAudio2::Base_SetVolume(float value)
{
if (XAudio2::MasteringVoice)
{
XAudio2::MasteringVoice->SetVolume(value);
}
}
bool AudioBackendXAudio2::Base_Init()
{
auto& devices = Audio::Devices;
// Initialize XAudio backend
HRESULT hr = XAudio2Create(&XAudio2::Instance, 0, XAUDIO2_DEFAULT_PROCESSOR);
if (FAILED(hr))
{
LOG(Error, "Failed to initalize XAudio2. Error: 0x{0:x}", hr);
return true;
}
XAudio2::Instance->RegisterForCallbacks(&XAudio2::Callback);
// Initialize master voice
hr = XAudio2::Instance->CreateMasteringVoice(&XAudio2::MasteringVoice);
if (FAILED(hr))
{
LOG(Error, "Failed to initalize XAudio2 mastering voice. Error: 0x{0:x}", hr);
return true;
}
XAUDIO2_VOICE_DETAILS details;
XAudio2::MasteringVoice->GetVoiceDetails(&details);
XAudio2::SampleRate = details.InputSampleRate;
XAudio2::Channels = details.InputChannels;
hr = XAudio2::MasteringVoice->GetChannelMask(&XAudio2::ChannelMask);
if (FAILED(hr))
{
LOG(Error, "Failed to get XAudio2 mastering voice channel mask. Error: 0x{0:x}", hr);
return true;
}
// Initialize spatial audio subsystem
DWORD dwChannelMask;
XAudio2::MasteringVoice->GetChannelMask(&dwChannelMask);
hr = X3DAudioInitialize(dwChannelMask, X3DAUDIO_SPEED_OF_SOUND, XAudio2::X3DInstance);
if (FAILED(hr))
{
LOG(Error, "Failed to initalize XAudio2 3D support. Error: 0x{0:x}", hr);
return true;
}
// Info
LOG(Info, "XAudio2: {0} channels at {1} kHz (channel mask {2})", XAudio2::Channels, XAudio2::SampleRate / 1000.0f, XAudio2::ChannelMask);
// Dummy device
devices.Resize(1);
devices[0].Name = TEXT("XAudio2 device");
Audio::SetActiveDeviceIndex(0);
return false;
}
void AudioBackendXAudio2::Base_Update()
{
// Update dirty voices
const auto listener = XAudio2::GetListener();
const float dopplerFactor = AudioSettings::Get()->DopplerFactor;
float matrixCoefficients[MAX_CHANNELS_MATRIX_SIZE];
X3DAUDIO_DSP_SETTINGS dsp = { 0 };
dsp.DstChannelCount = XAudio2::Channels;
dsp.pMatrixCoefficients = matrixCoefficients;
for (int32 i = 0; i < XAudio2::Sources.Count(); i++)
{
auto& source = XAudio2::Sources[i];
if (source.IsFree() || !(source.IsDirty || XAudio2::ForceDirty))
continue;
dsp.SrcChannelCount = source.Data.ChannelCount;
if (source.Is3D && listener)
{
X3DAudioCalculate(XAudio2::X3DInstance, &listener->Data, &source.Data, X3DAUDIO_CALCULATE_MATRIX | X3DAUDIO_CALCULATE_DOPPLER, &dsp);
}
else
{
// Stereo
dsp.DopplerFactor = 1.0f;
Platform::MemoryClear(dsp.pMatrixCoefficients, sizeof(matrixCoefficients));
dsp.pMatrixCoefficients[0] = 1.0f;
if (source.Format.nChannels == 1)
{
dsp.pMatrixCoefficients[1] = 1.0f;
}
else
{
dsp.pMatrixCoefficients[3] = 1.0f;
}
}
const float frequencyRatio = dopplerFactor * source.Pitch * dsp.DopplerFactor;
source.Voice->SetFrequencyRatio(frequencyRatio);
source.Voice->SetOutputMatrix(XAudio2::MasteringVoice, dsp.SrcChannelCount, dsp.DstChannelCount, dsp.pMatrixCoefficients);
source.IsDirty = false;
}
// Clear flag
XAudio2::ForceDirty = false;
}
void AudioBackendXAudio2::Base_Dispose()
{
// Cleanup stuff
if (XAudio2::MasteringVoice)
{
XAudio2::MasteringVoice->DestroyVoice();
XAudio2::MasteringVoice = nullptr;
}
if (XAudio2::Instance)
{
XAudio2::Instance->StopEngine();
XAudio2::Instance->Release();
XAudio2::Instance = nullptr;
}
}
#endif