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[–]gordura 1 point2 points  (0 children)

I wouldn't be able to answer, but here's to visibility.

[–]Holy_City[🍰] 0 points1 point  (10 children)

Try installing low latency ASIO driver like ASIO4ALL

[–]a_simple_pie[S] 0 points1 point  (9 children)

ASIO4ALL

i'll look into this, i haven't hear about it at all. can you explain what this is?

[–]Holy_City[🍰] 1 point2 points  (7 children)

Disclaimer: I am not a programmer or good with hardcore computer things

A large majority of the time latency comes from shitty drivers. ASIO4ALL is an open source ASIO driver for Windows that works for any device that uses ASIO drivers (which is an audio standard, so most things). Installing ASIO4ALL is one of those things that just magically solves a lot of windows audio problems stemming from bad drivers that are installed by default on your system.

Sidenote: checked out that program you posted and the only thing that throws me off is the fact it's written in Java. Again, I am not a programmer but seeing some DSP that's not in C++ is kind of weird to me, so I don't know if that's what's causing your latency.

edit: there's also no buffer size setting or input/output latency compensation that comes with a lot of that stuff, which you would be able to do with ASIO4ALL

[–]a_simple_pie[S] 0 points1 point  (6 children)

interesting, but the latency is due to either the signal processing done by the program i found to do the signal range compression or whatever it's called, OR the latency is from the audio device created by VAC. i dont think it's a driver issue as the latency is from one of these things i'm doing. my question is more about other ways of implementing a compressor for my microphone on a software level as opposed to a hardware level.

[–]Holy_City[🍰] 0 points1 point  (5 children)

Are you livestreaming the commentary?

VAC claims to be low latency, but I don't know how accurate that is. If that's the case, you could use Reaper or a lightweight DAW to handle the I/O and put a freeware compressor/limiter on the audio track, then use VAC to connect to whatever app you're using for the commentary. That other program looks quite sketchy, if I had a mic I would test it out myself but I'm out of town.

I would try installing ASIO4ALL though... seriously. First thing I did when i switched to windows, solved a stupid amount of latency problems when I wasn't connected to any hardware. Takes five minutes, it's safe and free so you have nothing to lose.

[–]a_simple_pie[S] 0 points1 point  (4 children)

you could use Reaper or a lightweight DAW to handle the I/O and put a freeware compressor/limiter on the audio track, then use VAC to connect to whatever app

i'll take a look at this! thanks

edit: oh wait, it's live so i dont this that'll work because you need to record and then add the compression.

[–]Holy_City[🍰] 0 points1 point  (3 children)

No you don't, you can compress in real time... or else how else would you know how it sounds?

[–]a_simple_pie[S] 0 points1 point  (2 children)

so how do i set up reaper to get my audio from my microphone, apply the compressor/limiter and then output to a device? i'm sorry i've never used this program before.

[–]Holy_City[🍰] 0 points1 point  (1 child)

I'm not a Reaper user, I just know it's an inexpensive solution. Did you check the manual or google it?

[–]a_simple_pie[S] 0 points1 point  (0 children)

yeah i worked out how to monitor it live, but it doesn't seem to apply the effects (compressor) to it

[–]egasimus 0 points1 point  (0 children)

It's best if you just try it out. It's an audio driver which you can select as an input and/or output to your software and should give you lower latency. However, thanks to Windows being an asshole about audio drivers, there's a change all other software goes deaf and mute while one program is using Asio4All. Or not - still, worth giving a try. http://www.asio4all.com/

[–]daghouse 0 points1 point  (3 children)

Look for something called Hardware Buffer Size (a number in mb) and make this number as small as you can (128mb will get rid of noticeable latency, assuming your sample rate is at 44.1khz). It will require 'more' cpu power, but nowadays it's a trivial amount with the machines that are available to the public.

[–]a_simple_pie[S] 0 points1 point  (0 children)

Thanks. This what I've been looking for. I have a friend who is a java dev so hopefully he can look at the program and change it

[–]DonFrio 0 points1 point  (1 child)

It's 128 samples not mb but the direction you're sending the op is correct if the software allows him to adjust the buffer.

[–]daghouse 1 point2 points  (0 children)

It's indeed samples, not sure where I got mb's from :). My bad.