- The entire editor can now be compiled using Clang on Windows. It also runs (as long as you use the MSVC linker.)
- Use UEBuildWindows.bCompileWithClang=true to help test Clang support
- Most UE4 programs can now be compiled using Clang on Windows, and linked using the Clang linker on Windows
- Many C++ syntax fixes to resolve Clang errors and warnings when on Windows
- Clang on Windows now supports PCH files properly (but not "shared" PCHs, yet.)
- Hacked the DirectX XAudio2 headers slightly to work around a macro pasting bug in Clang
[CL 2494439 by Mike Fricker in Main branch]
Consistent null checks, fixes for copy/pasted conditions, copy pasted LOCTEXT identifiers, comparisons against literals of the wrong type, missing return statements, dead code, and some system calls that were ignoring their return value
[CL 2494390 by Dan Oconnor in Main branch]
Original Notes (adapted)
Implementation and Integration of Oculus Audio SDK for VR Audio Spatialization
- Adding new audio extension which wraps the oculus audio sdk and the new XAPO plugin
- Adding new XAPO (Xaudio2 Audio Processing Object) effect for processing mono audio source streams inside the new module
- Added new enumeration which allows users to select which spatialization algorithm to use for spatialized mono sources
- Refactored the regular sound source spatialization/effect in XAudio2 device code to support the new HRTF mono-to-stereo effect
- Designed feature so that if the sound spatialization module isn't used, the spatialization will default to normal spatialization algorithm
Notes on implementation:
- Because the audio engine doesn't bifurcate spatialized vs. non-spatialized sound sources into separate source pools, I had to create up-front the effects for the full number of supported sounds 32). This is because the oculus sdk requires at initialization the total number of sound sources that will be used in the SDK.
- Because the oculus SDK refers to sound source instances by index (into the pre-allocated array of sources set at init), I had to save each sound source's index so that it can be used on the HRTF XAPO effect, and then piped to the oculus SDK inside the UE audio extension.
- The audio engine assumes that mono-sources will be treated a certain way (during send-routing and spatialization). Because this new HRTF effect is effectively turning mono-sources into stereo-sources, some code had to be changed to send/route these audio sources as if they were stereo.
- This implementation is slightly different than the original GDC implementation to better work with multiple audio devices. Each audio device creates an IAudioSpatializationAlgorithm object which contains the oculus HRTF processing contexts.
#codereview Nick.Whiting Marc.Audy
[CL 2488287 by Aaron McLeran in Main branch]
- Created audio device manager which manages weak ref handles to audio devices
- Audio device handles are created with WorldContexts and accessed through UWorld objects
- All access to audio devices are now through audio device handles
- There is a main/default audio device for GEngine for use with the Editor and single worldcontext games
#codereview marc.audy, matthew.griffin
[CL 2477046 by Aaron McLeran in Main branch]
Removed GVolumeMultiplier and added an equivalent to FApp to contain the code to load the unfocused volume multiplier from config.
UE-4449 - Allow users to choose whether audio can be heard when focus is lost
#codereview Robert.Manuszewski
[CL 2341022 by Matthew Griffin in Main branch]
Added support for ADPCM decoding.
Added 'sound type' information to USoundWave
- Android uses this to detect which type of asset it needs to deal with
Added support for selected encoding type via project settings
#codereview Daniel.Lamb, Chris.Babcock
[CL 2260750 by Robert Jones in Main branch]
- Inverse was one of the main reasons of crashes via lots of functions. Seems making default Inverse to be safer version seems better.
[CL 2255081 by Lina Halper in Main branch]
Implemented CookSurround function for AudioFormatOpus, refactoring as much common code from the standard Cook into re-usable functions.
Changed OpusAudioInfo class to use a multistream decoder for all sounds, as 1 and 2 channel sounds can still be decoded by it. Just had to create an array of mappings that would take the internal Opus format and give back PCM data in the order that Unreal expects.
Also made the OpusDecoderWrapper a true wrapper so that you don't have to use any opus functions outside of it and it will not function if WITH_OPUS is not defined.
Added a virtual function to the CompressedAudioInfo interface, to check whether its uncompressed data will be in the Vorbis format. This is to make sure that code currently just checking whether there is a DecompressionState can now be sure that the Vorbis format is being used before routing sound to different speakers.
Made sure that you can't cause any problems by adding or removing a SoundWave from the Audio Streaming Manager more than once. Also rearranged where sounds are added to the manager so that it is only done in one place and it's safe when changing quality triggers recompression/splitting.
Added a global multidimensional array to store all the Vorbis channel orderings as they needed to be used in multiple places.
Added Logging and Memory tracking for the streaming process.
[CL 2217839 by Matthew Griffin in Main branch]
New files added to contain FVorbisAudioInfo and FOpusAudioInfo, added all necessary includes
Added a few more checks for whether the target platform supports audio streaming where appropriate
Moved #define to .h file as it was being used in two .cpp files (working because of unity build)
[CL 2137973 by Matthew Griffin in Main branch]
Added platform feature/property for audio streaming so that it's not used on platforms that don't support it yet
Prevent full data from being cooked if sound is streaming AND streaming is supported on the cooked platform
Prevent streaming chunks from being cooked if cooked platform doesn't support streaming
Prevent the Resource data from being initialized and full data size being added to stats if sound is streaming
[CL 2130171 by Matthew Griffin in Main branch]
Audio Streaming Manager will keep track of SoundWaves that are marked for streaming and any sound sources that are using those waves.
Added functionality to all Sound Buffers to report which chunk of audio they are currently reading from, which the manager will use to request the current and next chunk be loaded in memory.
Added a struct to contain all of the additional information needed for a Streaming SoundWave and set off requests, which the manager will also keep track of to avoid increasing the memory size of all SoundWaves.
Tidied up the Opus Decompressor to reduce the amount of repeated code and share the basic functionality for streaming and non-streaming usage.
Set XAudio sounds up to use streaming data, with a function added to the audio device so that other platforms won't try to make use of it for the moment.
[CL 2123061 by Matthew Griffin in Main branch]
Moved most of the contents of FAudioDevice::FreeResource to USoundWave::FreeResources so that we can be sure that all memory is cleared up during teardown.
Removed FreeResource function from some subclasses of audio device, as they weren't doing anything different than the base one.
Also removed ListSounds function from these same classes as that seems to have fallen behind current base class version and isn't used anyway.
Removed platform specific audio buffer storage so that there isn't confusion about which is being used and ensured that casts are used when necessary.
Fixed compile issue with STAT_AudioResourceCreationTime for HTML5.
[CL 2109273 by Matthew Griffin in Main branch]
