- I needed to change the parameter for my hack, so I'm just going to
create a new function instead, QueryAudio2. C++ mangling and API
breakage makes me most displeased.
- Fixed an issue where audio data that came in bursts would cause
desyncs. Basically, if data came in too late, it would continue to
buffer little by little, causing progressive desync with certain
microphones and capture devices (avermedia stream engine for the live
gamer portable especially). Also seemed to happen with HDMI data
from the newer game consoles, like xbox one apparently, though I
can't be too sure.
Now, it queries the mic and auxilary sound sources until sound
buffers are depleted. After doing so, it then "sorts" the audio
packets timestamps backwards from the most recent packet to the
oldest audio packet. By doing this, it compensates for burst, and
ensures that all audio data is seamless. New burst data coming in
will then line up properly with the older data via the sort function.
NOTE: This needs testing
Made is so you can adjust mic/device audio time while streaming
updated installer script
updated some locale
got rid of audio time calculation completely
