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Fix various issues with OpenAL buffers playback (do proper bit convertion)

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This commit is contained in:
Wojtek Figat
2023-10-06 13:59:03 +02:00
parent 176123eb1f
commit 560f699dd8
3 changed files with 28 additions and 50 deletions
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6
Source/Engine/Audio/AudioClip.cpp
View File

@@ -454,14 +454,12 @@ bool AudioClip::WriteBuffer(int32 chunkIndex)
}
break;
case AudioFormat::Raw:
{
data = Span<byte>(chunk->Get(), chunk->Size());
}
break;
break;
default:
return true;
}
info.NumSamples = data.Length() / bytesPerSample;
info.NumSamples = Math::AlignDown(data.Length() / bytesPerSample, info.NumChannels * bytesPerSample);
// Convert to Mono if used as 3D source and backend doesn't support it
if (Is3D() && info.NumChannels > 1 && EnumHasNoneFlags(AudioBackend::Features(), AudioBackend::FeatureFlags::SpatialMultiChannel))

59
Source/Engine/Audio/OpenAL/AudioBackendOAL.cpp
View File

@@ -228,11 +228,9 @@ namespace ALC
ALenum GetOpenALBufferFormat(uint32 numChannels, uint32 bitDepth)
{
// TODO: cache enum values in Init()??
switch (bitDepth)
{
case 8:
{
switch (numChannels)
{
case 1:
@@ -247,13 +245,8 @@ ALenum GetOpenALBufferFormat(uint32 numChannels, uint32 bitDepth)
return alGetEnumValue("AL_FORMAT_61CHN8");
case 8:
return alGetEnumValue("AL_FORMAT_71CHN8");
default:
CRASH;
return 0;
}
}
case 16:
{
switch (numChannels)
{
case 1:
@@ -268,19 +261,22 @@ ALenum GetOpenALBufferFormat(uint32 numChannels, uint32 bitDepth)
return alGetEnumValue("AL_FORMAT_61CHN16");
case 8:
return alGetEnumValue("AL_FORMAT_71CHN16");
default:
CRASH;
return 0;
}
}
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:
@@ -289,15 +285,9 @@ ALenum GetOpenALBufferFormat(uint32 numChannels, uint32 bitDepth)
return alGetEnumValue("AL_FORMAT_61CHN32");
case 8:
return alGetEnumValue("AL_FORMAT_71CHN32");
default:
CRASH;
return 0;
}
}
default:
CRASH;
return 0;
}
return 0;
}
void AudioBackendOAL::Listener_OnAdd(AudioListener* listener)
@@ -607,7 +597,8 @@ void AudioBackendOAL::Buffer_Write(uint32 bufferId, byte* samples, const AudioDa
{
PROFILE_CPU();
// TODO: maybe use temporary buffers per thread to reduce dynamic allocations when uploading data to OpenAL?
// 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)
@@ -618,28 +609,23 @@ void AudioBackendOAL::Buffer_Write(uint32 bufferId, byte* samples, const AudioDa
{
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);
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. 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);
format = GetOpenALBufferFormat(info.NumChannels, 16);
alBufferData(bufferId, format, sampleBuffer16, bufferSize, info.SampleRate);
ALC_CHECK_ERROR(alBufferData);
Allocator::Free(sampleBuffer16);
}
}
@@ -648,19 +634,15 @@ void AudioBackendOAL::Buffer_Write(uint32 bufferId, byte* samples, const AudioDa
// 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
else if (format)
{
const ALenum format = GetOpenALBufferFormat(info.NumChannels, info.BitDepth);
alBufferData(bufferId, format, samples, info.NumSamples * (info.BitDepth / 8), info.SampleRate);
ALC_CHECK_ERROR(alBufferData);
}
@@ -675,10 +657,9 @@ void AudioBackendOAL::Buffer_Write(uint32 bufferId, byte* samples, const AudioDa
{
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);
format = GetOpenALBufferFormat(info.NumChannels, 32);
alBufferData(bufferId, format, sampleBuffer32, bufferSize, info.SampleRate);
ALC_CHECK_ERROR(alBufferData);
@@ -693,19 +674,23 @@ void AudioBackendOAL::Buffer_Write(uint32 bufferId, byte* samples, const AudioDa
for (uint32 i = 0; i < info.NumSamples; i++)
sampleBuffer[i] = ((int8*)samples)[i] + 128;
const ALenum format = GetOpenALBufferFormat(info.NumChannels, 16);
format = GetOpenALBufferFormat(info.NumChannels, 16);
alBufferData(bufferId, format, sampleBuffer, bufferSize, info.SampleRate);
ALC_CHECK_ERROR(alBufferData);
Allocator::Free(sampleBuffer);
}
else
else if (format)
{
const ALenum format = GetOpenALBufferFormat(info.NumChannels, info.BitDepth);
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()

13
Source/Engine/Tools/AudioTool/AudioTool.cpp
View File

@@ -231,8 +231,7 @@ void AudioTool::ConvertToFloat(const byte* input, uint32 inBitDepth, float* outp
for (uint32 i = 0; i < numSamples; i++)
{
const int8 sample = *(int8*)input;
output[i] = sample / 127.0f;
output[i] = sample * (1.0f / 127.0f);
input++;
}
}
@@ -241,8 +240,7 @@ void AudioTool::ConvertToFloat(const byte* input, uint32 inBitDepth, float* outp
for (uint32 i = 0; i < numSamples; i++)
{
const int16 sample = *(int16*)input;
output[i] = sample / 32767.0f;
output[i] = sample * (1.0f / 32767.0f);
input += 2;
}
}
@@ -251,8 +249,7 @@ void AudioTool::ConvertToFloat(const byte* input, uint32 inBitDepth, float* outp
for (uint32 i = 0; i < numSamples; i++)
{
const int32 sample = Convert24To32Bits(input);
output[i] = sample / 2147483647.0f;
output[i] = sample * (1.0f / 2147483647.0f);
input += 3;
}
}
@@ -261,8 +258,7 @@ void AudioTool::ConvertToFloat(const byte* input, uint32 inBitDepth, float* outp
for (uint32 i = 0; i < numSamples; i++)
{
const int32 sample = *(int32*)input;
output[i] = sample / 2147483647.0f;
output[i] = sample * (1.0f / 2147483647.0f);
input += 4;
}
}
@@ -278,7 +274,6 @@ void AudioTool::ConvertFromFloat(const float* input, int32* output, uint32 numSa
{
const float sample = *(float*)input;
output[i] = static_cast<int32>(sample * 2147483647.0f);
input++;
}
}