I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
- Probable fix for a mysterious crash in WASAPI
- Hopefully fixes telegramdesktop/tdesktop/4219 by setting PA stream role and bypassing filters
- Outgoing packet queue now uses Buffer instead of BufferPool
- Refactored audio I/O to allow sharing a common context between input and output, for those OSes that require this
- Rewritten periodic operation handling to use a "run loop" thingy instead of an ugly loop formerly known as tick thread
- Fixed a bunch of compiler warnings (closes#13)
- Added automake so you no longer need to use the GYP file for standalone builds (closes#43)
- Enabled delay-agnostic AEC on Windows & Linux, it seems to make a difference after all (telegramdesktop/tdesktop#4881)
- Fixed PulseAudio crashes, at least I hope so (closes#42)
- Fixed parsing of floating-point server config values in some locales in Linux
