From 560f699dd856cea66a8777132c4e1a9f532f5638 Mon Sep 17 00:00:00 2001 From: Wojtek Figat Date: Fri, 6 Oct 2023 13:59:03 +0200 Subject: [PATCH] Fix various issues with OpenAL buffers playback (do proper bit convertion) --- Source/Engine/Audio/AudioClip.cpp | 6 +- .../Engine/Audio/OpenAL/AudioBackendOAL.cpp | 59 +++++++------------ Source/Engine/Tools/AudioTool/AudioTool.cpp | 13 ++-- 3 files changed, 28 insertions(+), 50 deletions(-) diff --git a/Source/Engine/Audio/AudioClip.cpp b/Source/Engine/Audio/AudioClip.cpp index 911a8be33..d04da7274 100644 --- a/Source/Engine/Audio/AudioClip.cpp +++ b/Source/Engine/Audio/AudioClip.cpp @@ -454,14 +454,12 @@ bool AudioClip::WriteBuffer(int32 chunkIndex) } break; case AudioFormat::Raw: - { data = Span(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)) diff --git a/Source/Engine/Audio/OpenAL/AudioBackendOAL.cpp b/Source/Engine/Audio/OpenAL/AudioBackendOAL.cpp index 84642ac1f..c905c7eef 100644 --- a/Source/Engine/Audio/OpenAL/AudioBackendOAL.cpp +++ b/Source/Engine/Audio/OpenAL/AudioBackendOAL.cpp @@ -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() diff --git a/Source/Engine/Tools/AudioTool/AudioTool.cpp b/Source/Engine/Tools/AudioTool/AudioTool.cpp index a5a5d4663..89d613116 100644 --- a/Source/Engine/Tools/AudioTool/AudioTool.cpp +++ b/Source/Engine/Tools/AudioTool/AudioTool.cpp @@ -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(sample * 2147483647.0f); - input++; } }