Updated android audio device/buffers/souce to support creation of procedural data from a sound source. Tested with MOD replay in QA test level.
#codereview daniel.lamb, chris.babcock, dmitry.rekman
[CL 2276233 by Robert Jones 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]
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]
Followed pattern set out by texture streaming so that each chunk of audio data resides in its own bulkdata struct. It is currently possible to split audio into chunks when SoundWaves are marked for streaming but there is no way of doing this exposed at present.
Changed the parameters of FAudioDevice::GetRuntimeFormat so that the relevant SoundWave must be passed in, to allow for different formats for individual sounds/streaming options.
USoundWave::FreeResources no longer resets the NumChannels as it is unnecessary and causes sounds to be unable to play after the OGG data is flushed when attempting to switch to OPUS.
[CL 2099012 by Matthew Griffin in Main branch]
Moved functions to check for and select Compressed Audio Info class to the audio device as the audio formats are not accessible in cooked builds (and shouldn't really need to be either).
[CL 2057758 by Matthew Griffin in Main branch]
Added an audio format for Opus, which also required a resampler to make sure that all imported sounds are converted to a compatible sample rate.
Added the speex resampler from the opus tools package to the third party source, including built libraries for windows and mac.
Changed FVorbisAudioInfo so that it inherits from an interface for any kind of compressed audio, which can be used everywhere instead of being wrapped in #WITH_VORBIS.
Added FOpusAudioInfo to decompress Opus data, not sure at this point whether it's only going to be used for streaming audio but works for non-streamed playback.
[CL 2056352 by Matthew Griffin in Main branch]
