764 lines
24 KiB
C++
764 lines
24 KiB
C++
// Copyright (c) 2012-2024 Wojciech Figat. All rights reserved.
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#if AUDIO_API_OPENAL
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#include "AudioBackendOAL.h"
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#include "Engine/Platform/StringUtils.h"
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#include "Engine/Core/Log.h"
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#include "Engine/Core/Collections/Dictionary.h"
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#include "Engine/Tools/AudioTool/AudioTool.h"
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#include "Engine/Engine/Units.h"
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#include "Engine/Profiler/ProfilerCPU.h"
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#include "Engine/Audio/Audio.h"
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#include "Engine/Audio/AudioListener.h"
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#include "Engine/Audio/AudioSource.h"
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#include "Engine/Audio/AudioSettings.h"
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// Include OpenAL library
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// Source: https://github.com/kcat/openal-soft
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//#define AL_LIBTYPE_STATIC
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#include <OpenAL/al.h>
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#include <OpenAL/alc.h>
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#include <OpenAL/alext.h>
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#define FLAX_DST_TO_OAL(x) x * UNITS_TO_METERS_SCALE
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#define FLAX_POS_TO_OAL(vec) ((ALfloat)vec.X * -UNITS_TO_METERS_SCALE), ((ALfloat)vec.Y * UNITS_TO_METERS_SCALE), ((ALfloat)vec.Z * UNITS_TO_METERS_SCALE)
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#define FLAX_VEL_TO_OAL(vec) ((ALfloat)vec.X * -(UNITS_TO_METERS_SCALE*UNITS_TO_METERS_SCALE)), ((ALfloat)vec.Y * (UNITS_TO_METERS_SCALE*UNITS_TO_METERS_SCALE)), ((ALfloat)vec.Z * (UNITS_TO_METERS_SCALE*UNITS_TO_METERS_SCALE))
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#if BUILD_RELEASE
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#define ALC_CHECK_ERROR(method)
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#else
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#define ALC_CHECK_ERROR(method) \
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{ \
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int alError = alGetError(); \
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if (alError != 0) \
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{ \
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const Char* errorStr = GetOpenALErrorString(alError); \
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LOG(Error, "OpenAL method {0} failed with error 0x{1:X}:{2} (at line {3})", TEXT(#method), alError, errorStr, __LINE__ - 1); \
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} \
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}
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#endif
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namespace ALC
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{
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ALCdevice* Device = nullptr;
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ALCcontext* Context = nullptr;
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AudioBackend::FeatureFlags Features = AudioBackend::FeatureFlags::None;
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CriticalSection Locker;
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Dictionary<uint32, AudioDataInfo> SourceIDtoFormat;
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bool IsExtensionSupported(const char* extension)
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{
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if (Device == nullptr)
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return false;
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const int32 length = StringUtils::Length(extension);
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if ((length > 2) && (StringUtils::Compare(extension, "ALC", 3) == 0))
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return alcIsExtensionPresent(Device, extension) != AL_FALSE;
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return alIsExtensionPresent(extension) != AL_FALSE;
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}
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void ClearContext()
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{
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if (Context)
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{
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alcMakeContextCurrent(nullptr);
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alcDestroyContext(Context);
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Context = nullptr;
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}
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}
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namespace Listener
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{
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void Rebuild(const AudioListener* listener)
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{
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AudioBackend::Listener::Reset();
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AudioBackend::Listener::TransformChanged(listener->GetPosition(), listener->GetOrientation());
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AudioBackend::Listener::VelocityChanged(listener->GetVelocity());
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}
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}
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namespace Source
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{
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void Rebuild(uint32& sourceID, const Vector3& position, const Quaternion& orientation, float volume, float pitch, float pan, bool loop, bool spatial, float attenuation, float minDistance, float doppler)
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{
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ASSERT_LOW_LAYER(sourceID == 0);
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alGenSources(1, &sourceID);
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ASSERT_LOW_LAYER(sourceID != 0);
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alSourcef(sourceID, AL_GAIN, volume);
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alSourcef(sourceID, AL_PITCH, pitch);
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alSourcef(sourceID, AL_SEC_OFFSET, 0.0f);
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alSourcei(sourceID, AL_LOOPING, loop);
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alSourcei(sourceID, AL_SOURCE_RELATIVE, !spatial);
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alSourcei(sourceID, AL_BUFFER, 0);
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if (spatial)
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{
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#ifdef AL_SOFT_source_spatialize
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alSourcei(sourceID, AL_SOURCE_SPATIALIZE_SOFT, AL_TRUE);
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#endif
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alSourcef(sourceID, AL_ROLLOFF_FACTOR, attenuation);
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alSourcef(sourceID, AL_DOPPLER_FACTOR, doppler);
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alSourcef(sourceID, AL_REFERENCE_DISTANCE, FLAX_DST_TO_OAL(minDistance));
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alSource3f(sourceID, AL_POSITION, FLAX_POS_TO_OAL(position));
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alSource3f(sourceID, AL_VELOCITY, FLAX_VEL_TO_OAL(Vector3::Zero));
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}
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else
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{
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alSourcef(sourceID, AL_ROLLOFF_FACTOR, 0.0f);
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alSourcef(sourceID, AL_DOPPLER_FACTOR, 1.0f);
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alSourcef(sourceID, AL_REFERENCE_DISTANCE, 0.0f);
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alSource3f(sourceID, AL_POSITION, 0.0f, 0.0f, 0.0f);
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alSource3f(sourceID, AL_VELOCITY, 0.0f, 0.0f, 0.0f);
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}
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#ifdef AL_EXT_STEREO_ANGLES
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const float panAngle = pan * PI_HALF;
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const ALfloat panAngles[2] = { (ALfloat)(PI / 6.0 - panAngle), (ALfloat)(-PI / 6.0 - panAngle) }; // Angles are specified counter-clockwise in radians
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alSourcefv(sourceID, AL_STEREO_ANGLES, panAngles);
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#endif
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}
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}
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struct AudioSourceState
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{
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AudioSource::States State;
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float Time;
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};
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void RebuildContext(const Array<AudioSourceState>& states)
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{
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LOG(Info, "Rebuilding audio contexts");
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ClearContext();
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if (Device == nullptr)
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return;
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ALCint attrsHrtf[] = { ALC_HRTF_SOFT, ALC_TRUE };
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const ALCint* attrList = nullptr;
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if (Audio::GetEnableHRTF())
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{
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LOG(Info, "Enabling OpenAL HRTF");
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attrList = attrsHrtf;
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}
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Context = alcCreateContext(Device, attrList);
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alcMakeContextCurrent(Context);
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for (AudioListener* listener : Audio::Listeners)
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Listener::Rebuild(listener);
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for (int32 i = 0; i < states.Count(); i++)
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{
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AudioSource* source = Audio::Sources[i];
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Source::Rebuild(source->SourceID, source->GetPosition(), source->GetOrientation(), source->GetVolume(), source->GetPitch(), source->GetPan(), source->GetIsLooping() && !source->UseStreaming(), source->Is3D(), source->GetAttenuation(), source->GetMinDistance(), source->GetDopplerFactor());
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if (states.HasItems())
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{
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// Restore playback state
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auto& state = states[i];
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if (state.State != AudioSource::States::Stopped)
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source->Play();
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if (state.State == AudioSource::States::Paused)
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source->Pause();
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if (state.State != AudioSource::States::Stopped)
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source->SetTime(state.Time);
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}
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}
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}
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void RebuildContext(bool isChangingDevice)
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{
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Array<AudioSourceState> states;
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if (!isChangingDevice)
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{
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states.EnsureCapacity(Audio::Sources.Count());
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for (AudioSource* source : Audio::Sources)
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{
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states.Add({ source->GetState(), source->GetTime() });
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source->Stop();
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}
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}
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RebuildContext(states);
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}
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}
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ALenum GetOpenALBufferFormat(uint32 numChannels, uint32 bitDepth)
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{
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// TODO: cache enum values in Init()??
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switch (bitDepth)
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{
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case 8:
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switch (numChannels)
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{
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case 1:
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return AL_FORMAT_MONO8;
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case 2:
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return AL_FORMAT_STEREO8;
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case 4:
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return alGetEnumValue("AL_FORMAT_QUAD8");
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case 6:
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return alGetEnumValue("AL_FORMAT_51CHN8");
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case 7:
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return alGetEnumValue("AL_FORMAT_61CHN8");
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case 8:
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return alGetEnumValue("AL_FORMAT_71CHN8");
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}
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case 16:
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switch (numChannels)
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{
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case 1:
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return AL_FORMAT_MONO16;
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case 2:
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return AL_FORMAT_STEREO16;
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case 4:
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return alGetEnumValue("AL_FORMAT_QUAD16");
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case 6:
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return alGetEnumValue("AL_FORMAT_51CHN16");
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case 7:
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return alGetEnumValue("AL_FORMAT_61CHN16");
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case 8:
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return alGetEnumValue("AL_FORMAT_71CHN16");
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}
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case 32:
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switch (numChannels)
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{
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case 1:
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#ifdef AL_FORMAT_MONO_FLOAT32
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return AL_FORMAT_MONO_FLOAT32;
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#else
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return alGetEnumValue("AL_FORMAT_MONO_FLOAT32");
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#endif
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case 2:
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#ifdef AL_FORMAT_STEREO_FLOAT32
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return AL_FORMAT_STEREO_FLOAT32;
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#else
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return alGetEnumValue("AL_FORMAT_STEREO_FLOAT32");
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#endif
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case 4:
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return alGetEnumValue("AL_FORMAT_QUAD32");
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case 6:
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return alGetEnumValue("AL_FORMAT_51CHN32");
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case 7:
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return alGetEnumValue("AL_FORMAT_61CHN32");
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case 8:
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return alGetEnumValue("AL_FORMAT_71CHN32");
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}
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}
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return 0;
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}
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const Char* GetOpenALErrorString(int error)
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{
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switch (error)
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{
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case AL_NO_ERROR:
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return TEXT("AL_NO_ERROR");
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case AL_INVALID_NAME:
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return TEXT("AL_INVALID_NAME");
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case AL_INVALID_ENUM:
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return TEXT("AL_INVALID_ENUM");
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case AL_INVALID_VALUE:
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return TEXT("AL_INVALID_VALUE");
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case AL_INVALID_OPERATION:
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return TEXT("AL_INVALID_OPERATION");
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case AL_OUT_OF_MEMORY:
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return TEXT("AL_OUT_OF_MEMORY");
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default:
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break;
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}
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return TEXT("???");
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}
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void AudioBackendOAL::Listener_Reset()
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{
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alListenerf(AL_GAIN, Audio::GetVolume());
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}
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void AudioBackendOAL::Listener_VelocityChanged(const Vector3& velocity)
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{
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alListener3f(AL_VELOCITY, FLAX_VEL_TO_OAL(velocity));
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}
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void AudioBackendOAL::Listener_TransformChanged(const Vector3& position, const Quaternion& orientation)
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{
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const Float3 flipX(-1, 1, 1);
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const Float3 alOrientation[2] =
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{
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orientation * Float3::Forward * flipX,
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orientation * Float3::Up * flipX
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};
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alListenerfv(AL_ORIENTATION, (float*)alOrientation);
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alListener3f(AL_POSITION, FLAX_POS_TO_OAL(position));
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}
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void AudioBackendOAL::Listener_ReinitializeAll()
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{
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ALC::RebuildContext(false);
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}
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uint32 AudioBackendOAL::Source_Add(const AudioDataInfo& format, const Vector3& position, const Quaternion& orientation, float volume, float pitch, float pan, bool loop, bool spatial, float attenuation, float minDistance, float doppler)
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{
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uint32 sourceID = 0;
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ALC::Source::Rebuild(sourceID, position, orientation, volume, pitch, pan, loop, spatial, attenuation, minDistance, doppler);
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// Cache audio data format assigned on source (used in Source_GetCurrentBufferTime)
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ALC::Locker.Lock();
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ALC::SourceIDtoFormat[sourceID] = format;
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ALC::Locker.Unlock();
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return sourceID;
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}
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void AudioBackendOAL::Source_Remove(uint32 sourceID)
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{
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alSourcei(sourceID, AL_BUFFER, 0);
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ALC_CHECK_ERROR(alSourcei);
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alDeleteSources(1, &sourceID);
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ALC_CHECK_ERROR(alDeleteSources);
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ALC::Locker.Lock();
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ALC::SourceIDtoFormat.Remove(sourceID);
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ALC::Locker.Unlock();
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}
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void AudioBackendOAL::Source_VelocityChanged(uint32 sourceID, const Vector3& velocity)
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{
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alSource3f(sourceID, AL_VELOCITY, FLAX_VEL_TO_OAL(velocity));
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}
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void AudioBackendOAL::Source_TransformChanged(uint32 sourceID, const Vector3& position, const Quaternion& orientation)
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{
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alSource3f(sourceID, AL_POSITION, FLAX_POS_TO_OAL(position));
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}
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void AudioBackendOAL::Source_VolumeChanged(uint32 sourceID, float volume)
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{
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alSourcef(sourceID, AL_GAIN, volume);
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}
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void AudioBackendOAL::Source_PitchChanged(uint32 sourceID, float pitch)
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{
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alSourcef(sourceID, AL_PITCH, pitch);
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}
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void AudioBackendOAL::Source_PanChanged(uint32 sourceID, float pan)
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{
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#ifdef AL_EXT_STEREO_ANGLES
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const float panAngle = pan * PI_HALF;
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const ALfloat panAngles[2] = { (ALfloat)(PI / 6.0 - panAngle), (ALfloat)(-PI / 6.0 - panAngle) }; // Angles are specified counter-clockwise in radians
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alSourcefv(sourceID, AL_STEREO_ANGLES, panAngles);
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#endif
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}
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void AudioBackendOAL::Source_IsLoopingChanged(uint32 sourceID, bool loop)
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{
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alSourcei(sourceID, AL_LOOPING, loop);
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}
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void AudioBackendOAL::Source_SpatialSetupChanged(uint32 sourceID, bool spatial, float attenuation, float minDistance, float doppler)
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{
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alSourcei(sourceID, AL_SOURCE_RELATIVE, !spatial);
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if (spatial)
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{
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#ifdef AL_SOFT_source_spatialize
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alSourcei(sourceID, AL_SOURCE_SPATIALIZE_SOFT, AL_TRUE);
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#endif
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alSourcef(sourceID, AL_ROLLOFF_FACTOR, attenuation);
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alSourcef(sourceID, AL_DOPPLER_FACTOR, doppler);
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alSourcef(sourceID, AL_REFERENCE_DISTANCE, FLAX_DST_TO_OAL(minDistance));
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}
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else
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{
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alSourcef(sourceID, AL_ROLLOFF_FACTOR, 0.0f);
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alSourcef(sourceID, AL_DOPPLER_FACTOR, 1.0f);
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alSourcef(sourceID, AL_REFERENCE_DISTANCE, 0.0f);
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}
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}
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void AudioBackendOAL::Source_Play(uint32 sourceID)
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{
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alSourcePlay(sourceID);
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ALC_CHECK_ERROR(alSourcePlay);
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}
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void AudioBackendOAL::Source_Pause(uint32 sourceID)
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{
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alSourcePause(sourceID);
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ALC_CHECK_ERROR(alSourcePause);
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}
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void AudioBackendOAL::Source_Stop(uint32 sourceID)
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{
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// Stop and rewind
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alSourceRewind(sourceID);
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ALC_CHECK_ERROR(alSourceRewind);
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alSourcef(sourceID, AL_SEC_OFFSET, 0.0f);
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// Unset streaming buffers
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alSourcei(sourceID, AL_BUFFER, 0);
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ALC_CHECK_ERROR(alSourcei);
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}
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void AudioBackendOAL::Source_SetCurrentBufferTime(uint32 sourceID, float value)
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{
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alSourcef(sourceID, AL_SEC_OFFSET, value);
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}
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float AudioBackendOAL::Source_GetCurrentBufferTime(uint32 sourceID)
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{
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#if 0
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float time;
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alGetSourcef(sourceID, AL_SEC_OFFSET, &time);
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#else
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ALC::Locker.Lock();
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AudioDataInfo clipInfo = ALC::SourceIDtoFormat[sourceID];
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ALC::Locker.Unlock();
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ALint samplesPlayed;
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alGetSourcei(sourceID, AL_SAMPLE_OFFSET, &samplesPlayed);
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const uint32 totalSamples = clipInfo.NumSamples / clipInfo.NumChannels;
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if (totalSamples > 0)
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samplesPlayed %= totalSamples;
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const float time = samplesPlayed / static_cast<float>(Math::Max(1U, clipInfo.SampleRate));
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#endif
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return time;
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}
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void AudioBackendOAL::Source_SetNonStreamingBuffer(uint32 sourceID, uint32 bufferID)
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{
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alSourcei(sourceID, AL_BUFFER, bufferID);
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ALC_CHECK_ERROR(alSourcei);
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}
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void AudioBackendOAL::Source_GetProcessedBuffersCount(uint32 sourceID, int32& processedBuffersCount)
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{
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// Check the first context only
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alGetSourcei(sourceID, AL_BUFFERS_PROCESSED, &processedBuffersCount);
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ALC_CHECK_ERROR(alGetSourcei);
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}
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void AudioBackendOAL::Source_GetQueuedBuffersCount(uint32 sourceID, int32& queuedBuffersCount)
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{
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// Check the first context only
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alGetSourcei(sourceID, AL_BUFFERS_QUEUED, &queuedBuffersCount);
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ALC_CHECK_ERROR(alGetSourcei);
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}
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void AudioBackendOAL::Source_QueueBuffer(uint32 sourceID, uint32 bufferID)
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{
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// Queue new buffer
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alSourceQueueBuffers(sourceID, 1, &bufferID);
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ALC_CHECK_ERROR(alSourceQueueBuffers);
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}
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void AudioBackendOAL::Source_DequeueProcessedBuffers(uint32 sourceID)
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{
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int32 numProcessedBuffers;
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ALuint buffers[AUDIO_MAX_SOURCE_BUFFERS];
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alGetSourcei(sourceID, AL_BUFFERS_PROCESSED, &numProcessedBuffers);
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alSourceUnqueueBuffers(sourceID, numProcessedBuffers, buffers);
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ALC_CHECK_ERROR(alSourceUnqueueBuffers);
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}
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uint32 AudioBackendOAL::Buffer_Create()
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{
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uint32 bufferID;
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alGenBuffers(1, &bufferID);
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ALC_CHECK_ERROR(alGenBuffers);
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return bufferID;
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}
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void AudioBackendOAL::Buffer_Delete(uint32 bufferID)
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{
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alDeleteBuffers(1, &bufferID);
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ALC_CHECK_ERROR(alDeleteBuffers);
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}
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void AudioBackendOAL::Buffer_Write(uint32 bufferID, byte* samples, const AudioDataInfo& info)
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{
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PROFILE_CPU();
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// Pick the format for the audio data (it might not be supported natively)
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ALenum format = GetOpenALBufferFormat(info.NumChannels, info.BitDepth);
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// Mono or stereo
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if (info.NumChannels <= 2)
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{
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if (info.BitDepth > 16)
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{
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if (ALC::IsExtensionSupported("AL_EXT_float32"))
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{
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const uint32 bufferSize = info.NumSamples * sizeof(float);
|
|
float* sampleBufferFloat = (float*)Allocator::Allocate(bufferSize);
|
|
AudioTool::ConvertToFloat(samples, info.BitDepth, sampleBufferFloat, info.NumSamples);
|
|
|
|
format = GetOpenALBufferFormat(info.NumChannels, 32);
|
|
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. 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);
|
|
|
|
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;
|
|
|
|
alBufferData(bufferID, format, sampleBuffer, bufferSize, info.SampleRate);
|
|
ALC_CHECK_ERROR(alBufferData);
|
|
Allocator::Free(sampleBuffer);
|
|
}
|
|
else if (format)
|
|
{
|
|
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);
|
|
|
|
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;
|
|
|
|
format = GetOpenALBufferFormat(info.NumChannels, 16);
|
|
alBufferData(bufferID, format, sampleBuffer, bufferSize, info.SampleRate);
|
|
ALC_CHECK_ERROR(alBufferData);
|
|
|
|
Allocator::Free(sampleBuffer);
|
|
}
|
|
else if (format)
|
|
{
|
|
alBufferData(bufferID, format, samples, info.NumSamples * (info.BitDepth / 8), info.SampleRate);
|
|
ALC_CHECK_ERROR(alBufferData);
|
|
}
|
|
}
|
|
|
|
if (!format)
|
|
{
|
|
LOG(Error, "Not suppported audio data format for OpenAL device: BitDepth={}, NumChannels={}", info.BitDepth, info.NumChannels);
|
|
}
|
|
}
|
|
|
|
const Char* AudioBackendOAL::Base_Name()
|
|
{
|
|
return TEXT("OpenAL");
|
|
}
|
|
|
|
AudioBackend::FeatureFlags AudioBackendOAL::Base_Features()
|
|
{
|
|
return ALC::Features;
|
|
}
|
|
|
|
void AudioBackendOAL::Base_OnActiveDeviceChanged()
|
|
{
|
|
// Cleanup
|
|
Array<ALC::AudioSourceState> states;
|
|
states.EnsureCapacity(Audio::Sources.Count());
|
|
for (AudioSource* source : Audio::Sources)
|
|
{
|
|
states.Add({ source->GetState(), source->GetTime() });
|
|
source->Stop();
|
|
if (source->SourceID)
|
|
{
|
|
Source_Remove(source->SourceID);
|
|
source->SourceID = 0;
|
|
}
|
|
}
|
|
ALC::ClearContext();
|
|
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::RebuildContext(states);
|
|
}
|
|
|
|
void AudioBackendOAL::Base_SetDopplerFactor(float value)
|
|
{
|
|
alDopplerFactor(value);
|
|
}
|
|
|
|
void AudioBackendOAL::Base_SetVolume(float value)
|
|
{
|
|
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
|
|
int32 clampedIndex = Math::Clamp(activeDeviceIndex, -1, Audio::Devices.Count() - 1);
|
|
if (clampedIndex == Audio::GetActiveDeviceIndex())
|
|
{
|
|
ALC::RebuildContext(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
|