mirror of
https://github.com/danog/libtgvoip.git
synced 2024-11-26 12:14:39 +01:00
5caaaafa42
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. |
||
---|---|---|
.. | ||
Info.plist | ||
libtgvoipTests.mm | ||
MockReflector.cpp | ||
MockReflector.h |