765 lines
24 KiB
C++
765 lines
24 KiB
C++
// Copyright (c) 2012-2024 Wojciech Figat. All rights reserved.
|
|
|
|
#if AUDIO_API_OPENAL
|
|
|
|
#include "AudioBackendOAL.h"
|
|
#include "Engine/Platform/StringUtils.h"
|
|
#include "Engine/Core/Log.h"
|
|
#include "Engine/Core/Collections/Dictionary.h"
|
|
#include "Engine/Tools/AudioTool/AudioTool.h"
|
|
#include "Engine/Engine/Units.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 <OpenAL/al.h>
|
|
#include <OpenAL/alc.h>
|
|
#include <OpenAL/alext.h>
|
|
|
|
#define FLAX_DST_TO_OAL(x) x * UNITS_TO_METERS_SCALE
|
|
#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)
|
|
#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))
|
|
#if BUILD_RELEASE
|
|
#define ALC_CHECK_ERROR(method)
|
|
#else
|
|
#define ALC_CHECK_ERROR(method) \
|
|
{ \
|
|
int alError = alGetError(); \
|
|
if (alError != 0) \
|
|
{ \
|
|
const Char* errorStr = GetOpenALErrorString(alError); \
|
|
LOG(Error, "OpenAL method {0} failed with error 0x{1:X}:{2} (at line {3})", TEXT(#method), alError, errorStr, __LINE__ - 1); \
|
|
} \
|
|
}
|
|
#endif
|
|
|
|
namespace ALC
|
|
{
|
|
ALCdevice* Device = nullptr;
|
|
ALCcontext* Context = nullptr;
|
|
AudioBackend::FeatureFlags Features = AudioBackend::FeatureFlags::None;
|
|
CriticalSection Locker;
|
|
Dictionary<uint32, AudioDataInfo> SourceIDtoFormat;
|
|
|
|
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;
|
|
}
|
|
|
|
void ClearContext()
|
|
{
|
|
if (Context)
|
|
{
|
|
alcMakeContextCurrent(nullptr);
|
|
alcDestroyContext(Context);
|
|
Context = nullptr;
|
|
}
|
|
}
|
|
|
|
namespace Listener
|
|
{
|
|
void Rebuild(const AudioListener* listener)
|
|
{
|
|
AudioBackend::Listener::Reset();
|
|
AudioBackend::Listener::TransformChanged(listener->GetPosition(), listener->GetOrientation());
|
|
AudioBackend::Listener::VelocityChanged(listener->GetVelocity());
|
|
}
|
|
}
|
|
|
|
namespace Source
|
|
{
|
|
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)
|
|
{
|
|
ASSERT_LOW_LAYER(sourceID == 0);
|
|
alGenSources(1, &sourceID);
|
|
ASSERT_LOW_LAYER(sourceID != 0);
|
|
|
|
alSourcef(sourceID, AL_GAIN, volume);
|
|
alSourcef(sourceID, AL_PITCH, pitch);
|
|
alSourcef(sourceID, AL_SEC_OFFSET, 0.0f);
|
|
alSourcei(sourceID, AL_LOOPING, loop);
|
|
alSourcei(sourceID, AL_SOURCE_RELATIVE, !spatial);
|
|
alSourcei(sourceID, AL_BUFFER, 0);
|
|
if (spatial)
|
|
{
|
|
#ifdef AL_SOFT_source_spatialize
|
|
alSourcei(sourceID, AL_SOURCE_SPATIALIZE_SOFT, AL_TRUE);
|
|
#endif
|
|
alSourcef(sourceID, AL_ROLLOFF_FACTOR, attenuation);
|
|
alSourcef(sourceID, AL_DOPPLER_FACTOR, doppler);
|
|
alSourcef(sourceID, AL_REFERENCE_DISTANCE, FLAX_DST_TO_OAL(minDistance));
|
|
alSource3f(sourceID, AL_POSITION, FLAX_POS_TO_OAL(position));
|
|
alSource3f(sourceID, AL_VELOCITY, FLAX_VEL_TO_OAL(Vector3::Zero));
|
|
}
|
|
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 = pan * 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
|
|
}
|
|
}
|
|
|
|
struct AudioSourceState
|
|
{
|
|
AudioSource::States State;
|
|
float Time;
|
|
};
|
|
|
|
void RebuildContext(const Array<AudioSourceState>& states)
|
|
{
|
|
LOG(Info, "Rebuilding audio contexts");
|
|
|
|
ClearContext();
|
|
|
|
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;
|
|
}
|
|
|
|
Context = alcCreateContext(Device, attrList);
|
|
alcMakeContextCurrent(Context);
|
|
|
|
for (AudioListener* listener : Audio::Listeners)
|
|
Listener::Rebuild(listener);
|
|
|
|
for (int32 i = 0; i < states.Count(); i++)
|
|
{
|
|
AudioSource* source = Audio::Sources[i];
|
|
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());
|
|
|
|
if (states.HasItems())
|
|
{
|
|
// Restore playback state
|
|
auto& state = states[i];
|
|
if (state.State != AudioSource::States::Stopped)
|
|
source->Play();
|
|
if (state.State == AudioSource::States::Paused)
|
|
source->Pause();
|
|
if (state.State != AudioSource::States::Stopped)
|
|
source->SetTime(state.Time);
|
|
}
|
|
}
|
|
}
|
|
|
|
void RebuildContext(bool isChangingDevice)
|
|
{
|
|
Array<AudioSourceState> states;
|
|
if (!isChangingDevice)
|
|
{
|
|
states.EnsureCapacity(Audio::Sources.Count());
|
|
for (AudioSource* source : Audio::Sources)
|
|
{
|
|
states.Add({ source->GetState(), source->GetTime() });
|
|
source->Stop();
|
|
}
|
|
}
|
|
|
|
RebuildContext(states);
|
|
}
|
|
}
|
|
|
|
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");
|
|
}
|
|
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");
|
|
}
|
|
case 32:
|
|
switch (numChannels)
|
|
{
|
|
case 1:
|
|
#ifdef AL_FORMAT_MONO_FLOAT32
|
|
return AL_FORMAT_MONO_FLOAT32;
|
|
#else
|
|
return alGetEnumValue("AL_FORMAT_MONO_FLOAT32");
|
|
#endif
|
|
case 2:
|
|
#ifdef AL_FORMAT_STEREO_FLOAT32
|
|
return AL_FORMAT_STEREO_FLOAT32;
|
|
#else
|
|
return alGetEnumValue("AL_FORMAT_STEREO_FLOAT32");
|
|
#endif
|
|
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");
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
const Char* GetOpenALErrorString(int error)
|
|
{
|
|
switch (error)
|
|
{
|
|
case AL_NO_ERROR:
|
|
return TEXT("AL_NO_ERROR");
|
|
case AL_INVALID_NAME:
|
|
return TEXT("AL_INVALID_NAME");
|
|
case AL_INVALID_ENUM:
|
|
return TEXT("AL_INVALID_ENUM");
|
|
case AL_INVALID_VALUE:
|
|
return TEXT("AL_INVALID_VALUE");
|
|
case AL_INVALID_OPERATION:
|
|
return TEXT("AL_INVALID_OPERATION");
|
|
case AL_OUT_OF_MEMORY:
|
|
return TEXT("AL_OUT_OF_MEMORY");
|
|
default:
|
|
break;
|
|
}
|
|
return TEXT("???");
|
|
}
|
|
|
|
void AudioBackendOAL::Listener_Reset()
|
|
{
|
|
alListenerf(AL_GAIN, Audio::GetVolume());
|
|
}
|
|
|
|
void AudioBackendOAL::Listener_VelocityChanged(const Vector3& velocity)
|
|
{
|
|
alListener3f(AL_VELOCITY, FLAX_VEL_TO_OAL(velocity));
|
|
}
|
|
|
|
void AudioBackendOAL::Listener_TransformChanged(const Vector3& position, const Quaternion& orientation)
|
|
{
|
|
const Float3 flipX(-1, 1, 1);
|
|
const Float3 alOrientation[2] =
|
|
{
|
|
orientation * Float3::Forward * flipX,
|
|
orientation * Float3::Up * flipX
|
|
};
|
|
alListenerfv(AL_ORIENTATION, (float*)alOrientation);
|
|
alListener3f(AL_POSITION, FLAX_POS_TO_OAL(position));
|
|
}
|
|
|
|
void AudioBackendOAL::Listener_ReinitializeAll()
|
|
{
|
|
ALC::RebuildContext(false);
|
|
}
|
|
|
|
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)
|
|
{
|
|
uint32 sourceID = 0;
|
|
ALC::Source::Rebuild(sourceID, position, orientation, volume, pitch, pan, loop, spatial, attenuation, minDistance, doppler);
|
|
|
|
// Cache audio data format assigned on source (used in Source_GetCurrentBufferTime)
|
|
ALC::Locker.Lock();
|
|
ALC::SourceIDtoFormat[sourceID] = format;
|
|
ALC::Locker.Unlock();
|
|
|
|
return sourceID;
|
|
}
|
|
|
|
void AudioBackendOAL::Source_Remove(uint32 sourceID)
|
|
{
|
|
alSourcei(sourceID, AL_BUFFER, 0);
|
|
ALC_CHECK_ERROR(alSourcei);
|
|
alDeleteSources(1, &sourceID);
|
|
ALC_CHECK_ERROR(alDeleteSources);
|
|
|
|
ALC::Locker.Lock();
|
|
ALC::SourceIDtoFormat.Remove(sourceID);
|
|
ALC::Locker.Unlock();
|
|
}
|
|
|
|
void AudioBackendOAL::Source_VelocityChanged(uint32 sourceID, const Vector3& velocity)
|
|
{
|
|
alSource3f(sourceID, AL_VELOCITY, FLAX_VEL_TO_OAL(velocity));
|
|
}
|
|
|
|
void AudioBackendOAL::Source_TransformChanged(uint32 sourceID, const Vector3& position, const Quaternion& orientation)
|
|
{
|
|
alSource3f(sourceID, AL_POSITION, FLAX_POS_TO_OAL(position));
|
|
}
|
|
|
|
void AudioBackendOAL::Source_VolumeChanged(uint32 sourceID, float volume)
|
|
{
|
|
alSourcef(sourceID, AL_GAIN, volume);
|
|
}
|
|
|
|
void AudioBackendOAL::Source_PitchChanged(uint32 sourceID, float pitch)
|
|
{
|
|
alSourcef(sourceID, AL_PITCH, pitch);
|
|
}
|
|
|
|
void AudioBackendOAL::Source_PanChanged(uint32 sourceID, float pan)
|
|
{
|
|
#ifdef AL_EXT_STEREO_ANGLES
|
|
const float panAngle = pan * 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
|
|
}
|
|
|
|
void AudioBackendOAL::Source_IsLoopingChanged(uint32 sourceID, bool loop)
|
|
{
|
|
alSourcei(sourceID, AL_LOOPING, loop);
|
|
}
|
|
|
|
void AudioBackendOAL::Source_SpatialSetupChanged(uint32 sourceID, bool spatial, float attenuation, float minDistance, float doppler)
|
|
{
|
|
alSourcei(sourceID, AL_SOURCE_RELATIVE, !spatial);
|
|
if (spatial)
|
|
{
|
|
#ifdef AL_SOFT_source_spatialize
|
|
alSourcei(sourceID, AL_SOURCE_SPATIALIZE_SOFT, AL_TRUE);
|
|
#endif
|
|
alSourcef(sourceID, AL_ROLLOFF_FACTOR, attenuation);
|
|
alSourcef(sourceID, AL_DOPPLER_FACTOR, doppler);
|
|
alSourcef(sourceID, AL_REFERENCE_DISTANCE, FLAX_DST_TO_OAL(minDistance));
|
|
}
|
|
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_Play(uint32 sourceID)
|
|
{
|
|
alSourcePlay(sourceID);
|
|
ALC_CHECK_ERROR(alSourcePlay);
|
|
}
|
|
|
|
void AudioBackendOAL::Source_Pause(uint32 sourceID)
|
|
{
|
|
alSourcePause(sourceID);
|
|
ALC_CHECK_ERROR(alSourcePause);
|
|
}
|
|
|
|
void AudioBackendOAL::Source_Stop(uint32 sourceID)
|
|
{
|
|
// 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(uint32 sourceID, float value)
|
|
{
|
|
alSourcef(sourceID, AL_SEC_OFFSET, value);
|
|
}
|
|
|
|
float AudioBackendOAL::Source_GetCurrentBufferTime(uint32 sourceID)
|
|
{
|
|
#if 0
|
|
float time;
|
|
alGetSourcef(sourceID, AL_SEC_OFFSET, &time);
|
|
#else
|
|
ALC::Locker.Lock();
|
|
AudioDataInfo clipInfo = ALC::SourceIDtoFormat[sourceID];
|
|
ALC::Locker.Unlock();
|
|
ALint samplesPlayed;
|
|
alGetSourcei(sourceID, AL_SAMPLE_OFFSET, &samplesPlayed);
|
|
const uint32 totalSamples = clipInfo.NumSamples / clipInfo.NumChannels;
|
|
if (totalSamples > 0)
|
|
samplesPlayed %= totalSamples;
|
|
const float time = samplesPlayed / static_cast<float>(Math::Max(1U, clipInfo.SampleRate));
|
|
#endif
|
|
return time;
|
|
}
|
|
|
|
void AudioBackendOAL::Source_SetNonStreamingBuffer(uint32 sourceID, uint32 bufferID)
|
|
{
|
|
alSourcei(sourceID, AL_BUFFER, bufferID);
|
|
ALC_CHECK_ERROR(alSourcei);
|
|
}
|
|
|
|
void AudioBackendOAL::Source_GetProcessedBuffersCount(uint32 sourceID, int32& processedBuffersCount)
|
|
{
|
|
// Check the first context only
|
|
alGetSourcei(sourceID, AL_BUFFERS_PROCESSED, &processedBuffersCount);
|
|
ALC_CHECK_ERROR(alGetSourcei);
|
|
}
|
|
|
|
void AudioBackendOAL::Source_GetQueuedBuffersCount(uint32 sourceID, int32& queuedBuffersCount)
|
|
{
|
|
// Check the first context only
|
|
alGetSourcei(sourceID, AL_BUFFERS_QUEUED, &queuedBuffersCount);
|
|
ALC_CHECK_ERROR(alGetSourcei);
|
|
}
|
|
|
|
void AudioBackendOAL::Source_QueueBuffer(uint32 sourceID, uint32 bufferID)
|
|
{
|
|
// Queue new buffer
|
|
alSourceQueueBuffers(sourceID, 1, &bufferID);
|
|
ALC_CHECK_ERROR(alSourceQueueBuffers);
|
|
}
|
|
|
|
void AudioBackendOAL::Source_DequeueProcessedBuffers(uint32 sourceID)
|
|
{
|
|
int32 numProcessedBuffers;
|
|
alGetSourcei(sourceID, AL_BUFFERS_PROCESSED, &numProcessedBuffers);
|
|
Array<ALuint, InlinedAllocation<AUDIO_MAX_SOURCE_BUFFERS>> buffers;
|
|
buffers.Resize(numProcessedBuffers);
|
|
alSourceUnqueueBuffers(sourceID, numProcessedBuffers, buffers.Get());
|
|
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();
|
|
|
|
// Pick the format for the audio data (it might not be supported natively)
|
|
ALenum format = GetOpenALBufferFormat(info.NumChannels, info.BitDepth);
|
|
|
|
// 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);
|
|
|
|
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
|