Changed it so that timestamps do not "reset" if under the expected
value. Seems that doing this prevent a lot of timestamp "jumps" and
allows data to flow far more smoothly, as long as it's within the
expected broundries.
Had to make a ridiculous hack to get more usable variables in the class
without breaking API. It's a totally disgusting hack, but it works.
Replaced the AudioSource::resampler variable to be a pointer to newer
variables. The reason this hack had to be done in the first place is
because I directly exposed data in the base class instead of using a
forward. Forward pointers are always far more safe and prevent API
breakage. This is one thing I really dislike about C++, you modify the
class what so ever and it will break API. I just don't like C++ when
used for libraries, but again I could have avoided it by simply handling
it better, so it was really my fault for having been so careless with it
in the first place.
- 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.
- Woops, don't use global audio time for sorting, use the latest audio
timestamp from that specific device.
- Also, made it so lastSentTimestamp is only done when data is actually
used, accidentally put it in the wrong scope.
- 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
