// Copyright (c) 2012-2023 Wojciech Figat. All rights reserved. #if AUDIO_API_OPENAL #include "AudioBackendOAL.h" #include "Engine/Platform/StringUtils.h" #include "Engine/Core/Log.h" #include "Engine/Tools/AudioTool/AudioTool.h" #include "Engine/Profiler/ProfilerCPU.h" #include "Engine/Audio/Audio.h" #include "Engine/Audio/AudioListener.h" #include "Engine/Audio/AudioSource.h" #include "Engine/Audio/AudioSettings.h" // Include OpenAL library // Source: https://github.com/kcat/openal-soft //#define AL_LIBTYPE_STATIC #include #include #include #define ALC_MULTIPLE_LISTENERS 0 #define FLAX_COORD_SCALE 0.01f // units are meters #define FLAX_DST_TO_OAL(x) x * FLAX_COORD_SCALE #define FLAX_POS_TO_OAL(vec) ((ALfloat)vec.X * -FLAX_COORD_SCALE), ((ALfloat)vec.Y * FLAX_COORD_SCALE), ((ALfloat)vec.Z * FLAX_COORD_SCALE) #define FLAX_VEL_TO_OAL(vec) ((ALfloat)vec.X * -(FLAX_COORD_SCALE*FLAX_COORD_SCALE)), ((ALfloat)vec.Y * (FLAX_COORD_SCALE*FLAX_COORD_SCALE)), ((ALfloat)vec.Z * (FLAX_COORD_SCALE*FLAX_COORD_SCALE)) #if BUILD_RELEASE #define ALC_CHECK_ERROR(method) #else #define ALC_CHECK_ERROR(method) \ { \ int alError = alGetError(); \ if (alError != 0) \ { \ LOG(Error, "OpenAL method {0} failed with error 0x{1:X} (at line {2})", TEXT(#method), alError, __LINE__ - 1); \ } \ } #endif #if ALC_MULTIPLE_LISTENERS #define ALC_FOR_EACH_CONTEXT() \ for (int32 i = 0; i < Contexts.Count(); i++) { \ if (Contexts.Count() > 1) \ alcMakeContextCurrent(Contexts[i]); #define ALC_GET_DEFAULT_CONTEXT() \ if (Contexts.Count() > 1) \ alcMakeContextCurrent(Contexts[0]); #define ALC_GET_LISTENER_CONTEXT(listener) \ if (Contexts.Count() > 1) \ alcMakeContextCurrent(ALC::GetContext(listener))); #else #define ALC_FOR_EACH_CONTEXT() { int32 i = 0; #define ALC_GET_DEFAULT_CONTEXT() #define ALC_GET_LISTENER_CONTEXT(listener) #endif namespace ALC { ALCdevice* Device = nullptr; Array> Contexts; AudioBackend::FeatureFlags Features = AudioBackend::FeatureFlags::None; bool IsExtensionSupported(const char* extension) { if (Device == nullptr) return false; const int32 length = StringUtils::Length(extension); if ((length > 2) && (StringUtils::Compare(extension, "ALC", 3) == 0)) return alcIsExtensionPresent(Device, extension) != AL_FALSE; return alIsExtensionPresent(extension) != AL_FALSE; } ALCcontext* GetContext(const class AudioListener* listener) { #if ALC_MULTIPLE_LISTENERS const auto& listeners = Audio::Listeners; if (listeners.HasItems()) { ASSERT(listeners.Count() == Contexts.Count()); const int32 numContexts = Contexts.Count(); ALC_FOR_EACH_CONTEXT() { if (listeners[i] == listener) return Contexts[i]; } } ASSERT(Contexts.HasItems()); #else ASSERT(Contexts.Count() == 1); #endif return Contexts[0]; } FORCE_INLINE const Array>& GetContexts() { return Contexts; } void ClearContexts() { alcMakeContextCurrent(nullptr); for (ALCcontext* context : Contexts) alcDestroyContext(context); Contexts.Clear(); } namespace Listener { void Rebuild(AudioListener* listener) { AudioBackend::Listener::TransformChanged(listener); const Vector3 velocity = listener->GetVelocity(); alListener3f(AL_VELOCITY, FLAX_VEL_TO_OAL(velocity)); alListenerf(AL_GAIN, Audio::GetVolume()); } } namespace Source { void Rebuild(AudioSource* source) { ASSERT(source->SourceIDs.IsEmpty()); const bool is3D = source->Is3D(); const bool loop = source->GetIsLooping() && !source->UseStreaming(); ALC_FOR_EACH_CONTEXT() uint32 sourceID = 0; alGenSources(1, &sourceID); source->SourceIDs.Add(sourceID); } ALC_FOR_EACH_CONTEXT() const uint32 sourceID = source->SourceIDs[i]; alSourcef(sourceID, AL_GAIN, source->GetVolume()); alSourcef(sourceID, AL_PITCH, source->GetPitch()); alSourcef(sourceID, AL_SEC_OFFSET, 0.0f); alSourcei(sourceID, AL_LOOPING, loop); alSourcei(sourceID, AL_SOURCE_RELATIVE, !is3D); alSourcei(sourceID, AL_BUFFER, 0); if (is3D) { #ifdef AL_SOFT_source_spatialize alSourcei(sourceID, AL_SOURCE_SPATIALIZE_SOFT, AL_TRUE); #endif alSourcef(sourceID, AL_ROLLOFF_FACTOR, source->GetAttenuation()); alSourcef(sourceID, AL_DOPPLER_FACTOR, source->GetDopplerFactor()); alSourcef(sourceID, AL_REFERENCE_DISTANCE, FLAX_DST_TO_OAL(source->GetMinDistance())); alSource3f(sourceID, AL_POSITION, FLAX_POS_TO_OAL(source->GetPosition())); alSource3f(sourceID, AL_VELOCITY, FLAX_VEL_TO_OAL(source->GetVelocity())); } else { alSourcef(sourceID, AL_ROLLOFF_FACTOR, 0.0f); alSourcef(sourceID, AL_DOPPLER_FACTOR, 1.0f); alSourcef(sourceID, AL_REFERENCE_DISTANCE, 0.0f); alSource3f(sourceID, AL_POSITION, 0.0f, 0.0f, 0.0f); alSource3f(sourceID, AL_VELOCITY, 0.0f, 0.0f, 0.0f); } #ifdef AL_EXT_STEREO_ANGLES const float panAngle = source->GetPan() * PI_HALF; const ALfloat panAngles[2] = { (ALfloat)(PI / 6.0 - panAngle), (ALfloat)(-PI / 6.0 - panAngle) }; // Angles are specified counter-clockwise in radians alSourcefv(sourceID, AL_STEREO_ANGLES, panAngles); #endif } // Restore state after Cleanup source->Restore(); } } void RebuildContexts(bool isChangingDevice) { LOG(Info, "Rebuilding audio contexts"); if (!isChangingDevice) { for (AudioSource* source : Audio::Sources) source->Cleanup(); } ClearContexts(); if (Device == nullptr) return; ALCint attrsHrtf[] = { ALC_HRTF_SOFT, ALC_TRUE }; const ALCint* attrList = nullptr; if (Audio::GetEnableHRTF()) { LOG(Info, "Enabling OpenAL HRTF"); attrList = attrsHrtf; } #if ALC_MULTIPLE_LISTENERS const int32 numListeners = Audio::Listeners.Count(); const int32 numContexts = numListeners > 1 ? numListeners : 1; Contexts.Resize(numContexts); ALC_FOR_EACH_CONTEXT() ALCcontext* context = alcCreateContext(Device, attrList); Contexts[i] = context; } #else Contexts.Resize(1); Contexts[0] = alcCreateContext(Device, attrList); #endif // If only one context is available keep it active as an optimization. // Audio listeners and sources will avoid excessive context switching in such case. alcMakeContextCurrent(Contexts[0]); for (AudioListener* listener : Audio::Listeners) Listener::Rebuild(listener); for (AudioSource* source : Audio::Sources) Source::Rebuild(source); } } ALenum GetOpenALBufferFormat(uint32 numChannels, uint32 bitDepth) { // TODO: cache enum values in Init()?? switch (bitDepth) { case 8: { switch (numChannels) { case 1: return AL_FORMAT_MONO8; case 2: return AL_FORMAT_STEREO8; case 4: return alGetEnumValue("AL_FORMAT_QUAD8"); case 6: return alGetEnumValue("AL_FORMAT_51CHN8"); case 7: return alGetEnumValue("AL_FORMAT_61CHN8"); case 8: return alGetEnumValue("AL_FORMAT_71CHN8"); default: CRASH; return 0; } } case 16: { switch (numChannels) { case 1: return AL_FORMAT_MONO16; case 2: return AL_FORMAT_STEREO16; case 4: return alGetEnumValue("AL_FORMAT_QUAD16"); case 6: return alGetEnumValue("AL_FORMAT_51CHN16"); case 7: return alGetEnumValue("AL_FORMAT_61CHN16"); case 8: return alGetEnumValue("AL_FORMAT_71CHN16"); default: CRASH; return 0; } } case 32: { switch (numChannels) { case 1: return alGetEnumValue("AL_FORMAT_MONO_FLOAT32"); case 2: return alGetEnumValue("AL_FORMAT_STEREO_FLOAT32"); case 4: return alGetEnumValue("AL_FORMAT_QUAD32"); case 6: return alGetEnumValue("AL_FORMAT_51CHN32"); case 7: return alGetEnumValue("AL_FORMAT_61CHN32"); case 8: return alGetEnumValue("AL_FORMAT_71CHN32"); default: CRASH; return 0; } } default: CRASH; return 0; } } void AudioBackendOAL::Listener_OnAdd(AudioListener* listener) { #if ALC_MULTIPLE_LISTENERS ALC::RebuildContexts(false); #else AudioBackend::Listener::TransformChanged(listener); const Vector3 velocity = listener->GetVelocity(); alListener3f(AL_VELOCITY, FLAX_VEL_TO_OAL(velocity)); alListenerf(AL_GAIN, Audio::GetVolume()); #endif } void AudioBackendOAL::Listener_OnRemove(AudioListener* listener) { #if ALC_MULTIPLE_LISTENERS ALC::RebuildContexts(false); #endif } void AudioBackendOAL::Listener_VelocityChanged(AudioListener* listener) { ALC_GET_LISTENER_CONTEXT(listener) const Vector3 velocity = listener->GetVelocity(); alListener3f(AL_VELOCITY, FLAX_VEL_TO_OAL(velocity)); } void AudioBackendOAL::Listener_TransformChanged(AudioListener* listener) { ALC_GET_LISTENER_CONTEXT(listener) const Vector3 position = listener->GetPosition(); const Quaternion orientation = listener->GetOrientation(); const Vector3 flipX(-1, 1, 1); const Vector3 alOrientation[2] = { // Forward orientation * Vector3::Forward * flipX, // Up orientation * Vector3::Up * flipX }; alListenerfv(AL_ORIENTATION, (float*)alOrientation); alListener3f(AL_POSITION, FLAX_POS_TO_OAL(position)); } void AudioBackendOAL::Listener_ReinitializeAll() { ALC::RebuildContexts(false); } void AudioBackendOAL::Source_OnAdd(AudioSource* source) { ALC::Source::Rebuild(source); } void AudioBackendOAL::Source_OnRemove(AudioSource* source) { source->Cleanup(); } void AudioBackendOAL::Source_VelocityChanged(AudioSource* source) { if (!source->Is3D()) return; ALC_FOR_EACH_CONTEXT() const uint32 sourceID = source->SourceIDs[i]; alSource3f(sourceID, AL_VELOCITY, FLAX_VEL_TO_OAL(source->GetVelocity())); } } void AudioBackendOAL::Source_TransformChanged(AudioSource* source) { if (!source->Is3D()) return; ALC_FOR_EACH_CONTEXT() const uint32 sourceID = source->SourceIDs[i]; alSource3f(sourceID, AL_POSITION, FLAX_POS_TO_OAL(source->GetPosition())); } } void AudioBackendOAL::Source_VolumeChanged(AudioSource* source) { ALC_FOR_EACH_CONTEXT() const uint32 sourceID = source->SourceIDs[i]; alSourcef(sourceID, AL_GAIN, source->GetVolume()); } } void AudioBackendOAL::Source_PitchChanged(AudioSource* source) { ALC_FOR_EACH_CONTEXT() const uint32 sourceID = source->SourceIDs[i]; alSourcef(sourceID, AL_PITCH, source->GetPitch()); } } void AudioBackendOAL::Source_PanChanged(AudioSource* source) { #ifdef AL_EXT_STEREO_ANGLES const float panAngle = source->GetPan() * PI_HALF; const ALfloat panAngles[2] = { (ALfloat)(PI / 6.0 - panAngle), (ALfloat)(-PI / 6.0 - panAngle) }; // Angles are specified counter-clockwise in radians ALC_FOR_EACH_CONTEXT() const uint32 sourceID = source->SourceIDs[i]; alSourcefv(sourceID, AL_STEREO_ANGLES, panAngles); } #endif } void AudioBackendOAL::Source_IsLoopingChanged(AudioSource* source) { const bool loop = source->GetIsLooping() && !source->UseStreaming(); ALC_FOR_EACH_CONTEXT() const uint32 sourceID = source->SourceIDs[i]; alSourcei(sourceID, AL_LOOPING, loop); } } void AudioBackendOAL::Source_SpatialSetupChanged(AudioSource* source) { const bool is3D = source->Is3D(); ALC_FOR_EACH_CONTEXT() const uint32 sourceID = source->SourceIDs[i]; alSourcei(sourceID, AL_SOURCE_RELATIVE, !is3D); if (is3D) { #ifdef AL_SOFT_source_spatialize alSourcei(sourceID, AL_SOURCE_SPATIALIZE_SOFT, AL_TRUE); #endif alSourcef(sourceID, AL_ROLLOFF_FACTOR, source->GetAttenuation()); alSourcef(sourceID, AL_DOPPLER_FACTOR, source->GetDopplerFactor()); alSourcef(sourceID, AL_REFERENCE_DISTANCE, FLAX_DST_TO_OAL(source->GetMinDistance())); } else { alSourcef(sourceID, AL_ROLLOFF_FACTOR, 0.0f); alSourcef(sourceID, AL_DOPPLER_FACTOR, 1.0f); alSourcef(sourceID, AL_REFERENCE_DISTANCE, 0.0f); } } } void AudioBackendOAL::Source_ClipLoaded(AudioSource* source) { if (source->SourceIDs.Count() < ALC::Contexts.Count()) return; const auto clip = source->Clip.Get(); const bool is3D = source->Is3D(); const bool loop = source->GetIsLooping() && !clip->IsStreamable(); ALC_FOR_EACH_CONTEXT() const uint32 sourceID = source->SourceIDs[i]; alSourcei(sourceID, AL_SOURCE_RELATIVE, !is3D); alSourcei(sourceID, AL_LOOPING, loop); } } void AudioBackendOAL::Source_Cleanup(AudioSource* source) { ALC_FOR_EACH_CONTEXT() const uint32 sourceID = source->SourceIDs[i]; alSourcei(sourceID, AL_BUFFER, 0); ALC_CHECK_ERROR(alSourcei); alDeleteSources(1, &sourceID); ALC_CHECK_ERROR(alDeleteSources); } } void AudioBackendOAL::Source_Play(AudioSource* source) { ALC_FOR_EACH_CONTEXT() const uint32 sourceID = source->SourceIDs[i]; // Play alSourcePlay(sourceID); ALC_CHECK_ERROR(alSourcePlay); } } void AudioBackendOAL::Source_Pause(AudioSource* source) { ALC_FOR_EACH_CONTEXT() const uint32 sourceID = source->SourceIDs[i]; // Pause alSourcePause(sourceID); ALC_CHECK_ERROR(alSourcePause); } } void AudioBackendOAL::Source_Stop(AudioSource* source) { ALC_FOR_EACH_CONTEXT() const uint32 sourceID = source->SourceIDs[i]; // Stop and rewind alSourceRewind(sourceID); ALC_CHECK_ERROR(alSourceRewind); alSourcef(sourceID, AL_SEC_OFFSET, 0.0f); // Unset streaming buffers alSourcei(sourceID, AL_BUFFER, 0); ALC_CHECK_ERROR(alSourcei); } } void AudioBackendOAL::Source_SetCurrentBufferTime(AudioSource* source, float value) { ALC_FOR_EACH_CONTEXT() const uint32 sourceID = source->SourceIDs[i]; alSourcef(sourceID, AL_SEC_OFFSET, value); } } float AudioBackendOAL::Source_GetCurrentBufferTime(const AudioSource* source) { ALC_GET_DEFAULT_CONTEXT() #if 0 float time; alGetSourcef(source->SourceIDs[0], AL_SEC_OFFSET, &time); #else ASSERT(source->Clip && source->Clip->IsLoaded()); const AudioDataInfo& clipInfo = source->Clip->AudioHeader.Info; ALint samplesPlayed; alGetSourcei(source->SourceIDs[0], AL_SAMPLE_OFFSET, &samplesPlayed); const uint32 totalSamples = clipInfo.NumSamples / clipInfo.NumChannels; const float time = (samplesPlayed % totalSamples) / static_cast(Math::Max(1U, clipInfo.SampleRate)); #endif return time; } void AudioBackendOAL::Source_SetNonStreamingBuffer(AudioSource* source) { const uint32 bufferId = source->Clip->Buffers[0]; ALC_FOR_EACH_CONTEXT() const uint32 sourceID = source->SourceIDs[i]; alSourcei(sourceID, AL_BUFFER, bufferId); ALC_CHECK_ERROR(alSourcei); } } void AudioBackendOAL::Source_GetProcessedBuffersCount(AudioSource* source, int32& processedBuffersCount) { ALC_GET_DEFAULT_CONTEXT() // Check the first context only const uint32 sourceID = source->SourceIDs[0]; alGetSourcei(sourceID, AL_BUFFERS_PROCESSED, &processedBuffersCount); ALC_CHECK_ERROR(alGetSourcei); } void AudioBackendOAL::Source_GetQueuedBuffersCount(AudioSource* source, int32& queuedBuffersCount) { ALC_GET_DEFAULT_CONTEXT() // Check the first context only const uint32 sourceID = source->SourceIDs[0]; alGetSourcei(sourceID, AL_BUFFERS_QUEUED, &queuedBuffersCount); ALC_CHECK_ERROR(alGetSourcei); } void AudioBackendOAL::Source_QueueBuffer(AudioSource* source, uint32 bufferId) { ALC_FOR_EACH_CONTEXT() const uint32 sourceID = source->SourceIDs[i]; // Queue new buffer alSourceQueueBuffers(sourceID, 1, &bufferId); ALC_CHECK_ERROR(alSourceQueueBuffers); } } void AudioBackendOAL::Source_DequeueProcessedBuffers(AudioSource* source) { ALuint buffers[AUDIO_MAX_SOURCE_BUFFERS]; ALC_FOR_EACH_CONTEXT() const uint32 sourceID = source->SourceIDs[i]; int32 numProcessedBuffers; alGetSourcei(sourceID, AL_BUFFERS_PROCESSED, &numProcessedBuffers); alSourceUnqueueBuffers(sourceID, numProcessedBuffers, buffers); ALC_CHECK_ERROR(alSourceUnqueueBuffers); } } uint32 AudioBackendOAL::Buffer_Create() { uint32 bufferId; alGenBuffers(1, &bufferId); ALC_CHECK_ERROR(alGenBuffers); return bufferId; } void AudioBackendOAL::Buffer_Delete(uint32 bufferId) { alDeleteBuffers(1, &bufferId); ALC_CHECK_ERROR(alDeleteBuffers); } void AudioBackendOAL::Buffer_Write(uint32 bufferId, byte* samples, const AudioDataInfo& info) { PROFILE_CPU(); // TODO: maybe use temporary buffers per thread to reduce dynamic allocations when uploading data to OpenAL? // Mono or stereo if (info.NumChannels <= 2) { if (info.BitDepth > 16) { if (ALC::IsExtensionSupported("AL_EXT_float32")) { const uint32 bufferSize = info.NumSamples * sizeof(float); float* sampleBufferFloat = (float*)Allocator::Allocate(bufferSize); AudioTool::ConvertToFloat(samples, info.BitDepth, sampleBufferFloat, info.NumSamples); const ALenum format = GetOpenALBufferFormat(info.NumChannels, info.BitDepth); alBufferData(bufferId, format, sampleBufferFloat, bufferSize, info.SampleRate); ALC_CHECK_ERROR(alBufferData); Allocator::Free(sampleBufferFloat); } else { LOG(Warning, "OpenAL doesn't support bit depth larger than 16. Your audio data will be truncated."); const uint32 bufferSize = info.NumSamples * 2; byte* sampleBuffer16 = (byte*)Allocator::Allocate(bufferSize); AudioTool::ConvertBitDepth(samples, info.BitDepth, sampleBuffer16, 16, info.NumSamples); const ALenum format = GetOpenALBufferFormat(info.NumChannels, 16); alBufferData(bufferId, format, sampleBuffer16, bufferSize, info.SampleRate); ALC_CHECK_ERROR(alBufferData); Allocator::Free(sampleBuffer16); } } else if (info.BitDepth == 8) { // OpenAL expects unsigned 8-bit data, but engine stores it as signed, so convert const uint32 bufferSize = info.NumSamples * (info.BitDepth / 8); byte* sampleBuffer = (byte*)Allocator::Allocate(bufferSize); for (uint32 i = 0; i < info.NumSamples; i++) sampleBuffer[i] = ((int8*)samples)[i] + 128; const ALenum format = GetOpenALBufferFormat(info.NumChannels, 16); alBufferData(bufferId, format, sampleBuffer, bufferSize, info.SampleRate); ALC_CHECK_ERROR(alBufferData); Allocator::Free(sampleBuffer); } else { const ALenum format = GetOpenALBufferFormat(info.NumChannels, info.BitDepth); alBufferData(bufferId, format, samples, info.NumSamples * (info.BitDepth / 8), info.SampleRate); ALC_CHECK_ERROR(alBufferData); } } // Multichannel else { // Note: Assuming AL_EXT_MCFORMATS is supported. If it's not, channels should be reduced to mono or stereo. // 24-bit not supported, convert to 32-bit if (info.BitDepth == 24) { const uint32 bufferSize = info.NumChannels * sizeof(int32); byte* sampleBuffer32 = (byte*)Allocator::Allocate(bufferSize); AudioTool::ConvertBitDepth(samples, info.BitDepth, sampleBuffer32, 32, info.NumSamples); const ALenum format = GetOpenALBufferFormat(info.NumChannels, 32); alBufferData(bufferId, format, sampleBuffer32, bufferSize, info.SampleRate); ALC_CHECK_ERROR(alBufferData); Allocator::Free(sampleBuffer32); } else if (info.BitDepth == 8) { // OpenAL expects unsigned 8-bit data, but engine stores it as signed, so convert const uint32 bufferSize = info.NumSamples * (info.BitDepth / 8); byte* sampleBuffer = (byte*)Allocator::Allocate(bufferSize); for (uint32 i = 0; i < info.NumSamples; i++) sampleBuffer[i] = ((int8*)samples)[i] + 128; const ALenum format = GetOpenALBufferFormat(info.NumChannels, 16); alBufferData(bufferId, format, sampleBuffer, bufferSize, info.SampleRate); ALC_CHECK_ERROR(alBufferData); Allocator::Free(sampleBuffer); } else { const ALenum format = GetOpenALBufferFormat(info.NumChannels, info.BitDepth); alBufferData(bufferId, format, samples, info.NumSamples * (info.BitDepth / 8), info.SampleRate); ALC_CHECK_ERROR(alBufferData); } } } const Char* AudioBackendOAL::Base_Name() { return TEXT("OpenAL"); } AudioBackend::FeatureFlags AudioBackendOAL::Base_Features() { return ALC::Features; } void AudioBackendOAL::Base_OnActiveDeviceChanged() { // Cleanup for (AudioSource* source : Audio::Sources) source->Cleanup(); ALC::ClearContexts(); if (ALC::Device != nullptr) { alcCloseDevice(ALC::Device); ALC::Device = nullptr; } // Open device const StringAnsi& name = Audio::GetActiveDevice()->InternalName; ALC::Device = alcOpenDevice(name.Get()); if (ALC::Device == nullptr) { LOG(Fatal, "Failed to open OpenAL device ({0}).", String(name)); return; } // Setup ALC::RebuildContexts(true); } void AudioBackendOAL::Base_SetDopplerFactor(float value) { alDopplerFactor(value); } void AudioBackendOAL::Base_SetVolume(float value) { ALC_FOR_EACH_CONTEXT() alListenerf(AL_GAIN, value); } } bool AudioBackendOAL::Base_Init() { auto& devices = Audio::Devices; #if 0 // Use it for ALSOFT errors debugging (build OpenAL-Soft in Debug) Platform::SetEnvironmentVariable(TEXT("ALSOFT_TRAP_ERROR"), TEXT("1")); Platform::SetEnvironmentVariable(TEXT("ALSOFT_LOGLEVEL"), TEXT("9")); Platform::SetEnvironmentVariable(TEXT("ALSOFT_LOGFILE"), TEXT("alc_log.txt")); #endif // Initialization (use the preferred device) int32 activeDeviceIndex; ALC::Device = alcOpenDevice(nullptr); if (ALC::Device == nullptr) { activeDeviceIndex = -1; const auto err = alGetError(); LOG(Warning, "Failed to open default OpenAL device. Error: 0x{0:X}", err); } else { activeDeviceIndex = 0; } // Get audio devices #if ALC_ENUMERATE_ALL_EXT const ALCchar* defaultDevice = alcGetString(nullptr, ALC_DEFAULT_ALL_DEVICES_SPECIFIER); if (ALC::IsExtensionSupported("ALC_ENUMERATE_ALL_EXT") && defaultDevice != nullptr) { const ALCchar* devicesStr = alcGetString(nullptr, ALC_ALL_DEVICES_SPECIFIER); const StringAnsi defaultDeviceName(defaultDevice); devices.Clear(); devices.EnsureCapacity(8); activeDeviceIndex = -1; while (devicesStr && *devicesStr) { const int32 i = devices.Count(); devices.Resize(i + 1); auto& device = devices[i]; device.InternalName = devicesStr; device.Name = String(device.InternalName).TrimTrailing(); device.Name.Replace(TEXT("OpenAL Soft on "), TEXT("")); if (device.InternalName == defaultDeviceName) { activeDeviceIndex = i; } devicesStr += (device.InternalName.Length() + 1) * sizeof(ALCchar); } if (activeDeviceIndex == -1) { LOG(Warning, "Failed to pick a default device"); LOG_STR(Warning, String(defaultDeviceName)); for (int32 i = 0; i < devices.Count(); i++) LOG_STR(Warning, devices[i].Name); if (devices.IsEmpty()) return true; LOG(Warning, "Using the first audio device"); activeDeviceIndex = 0; } // Open default device if (ALC::Device) alcCloseDevice(ALC::Device); const auto& name = devices[activeDeviceIndex].InternalName; ALC::Device = alcOpenDevice(name.Get()); if (ALC::Device == nullptr) { LOG(Warning, "Failed to open OpenAL device ({0}).", String(name)); return true; } } else #endif { if (ALC::Device) { // Single device devices.Resize(1); devices[0].Name = TEXT("Default device"); } else { // No device devices.Resize(0); } } // Init Base_SetDopplerFactor(AudioSettings::Get()->DopplerFactor); alDistanceModel(AL_INVERSE_DISTANCE_CLAMPED); // Default attenuation model ALC::RebuildContexts(true); Audio::SetActiveDeviceIndex(activeDeviceIndex); #ifdef AL_SOFT_source_spatialize if (ALC::IsExtensionSupported("AL_SOFT_source_spatialize")) ALC::Features = EnumAddFlags(ALC::Features, FeatureFlags::SpatialMultiChannel); #endif // Log service info LOG(Info, "{0} ({1})", String(alGetString(AL_RENDERER)), String(alGetString(AL_VERSION))); for (int32 i = 0; i < devices.Count(); i++) { LOG(Info, "{0}{1}", i == activeDeviceIndex ? TEXT("[active] ") : TEXT(""), devices[i].Name); } return false; } void AudioBackendOAL::Base_Update() { } void AudioBackendOAL::Base_Dispose() { if (ALC::Device != nullptr) { alcCloseDevice(ALC::Device); ALC::Device = nullptr; } } #endif