buffers, delay and volume control

buffers, delay and volume control

Postby Guest

I have a few questions:

First, I am using three monitors with the two outside screens with speakers built in. These are hooked up by audio cable w/splitter to my R/L audio jack. I am also using a USB speaker system. I am currently using "listen to this device" to the monitor speakers. I'm experiencing some lag. The FAQ says that's the trail version.

I was wondering if you guys have the best way to make this work?

Also what settings for Buffers, Sample Rate, Bits per sample, channel I should be using?

Also I noticed that if I just select my real microphone and my real audio card and just start the "Output" side it still works? Then what's the point of the left side?

Sorry if this is hard to understand!
Guest
 
Posts: 65
Joined: Mon Apr 15, 2013 8:30 am

Re: buffers, delay and volume control

Postby admin

I'm experiencing some lag.

The delay should be removed after you receive our key and activate VAS.

Also what settings for Buffers, Sample Rate, Bits per sample, channel I should be using?

You can use 44.1K sample rate and 16bits per sample. As to the buffer sizes, you need do some tests to find the best value on your computer. Click 'stop' button, then input a small buffer size, for example 1024 or 512 or 700 or even 256 to test. Small value indicates small delay, but requires more powerful CPU. If you have a slow computer, you should use high value such as 2048 or 4096.

An example:
Let's say, you use VAS with Skype and stream what you hear + microphone voice to Skype. You should configure VAS like below:
- VAS is set as the default playback device;
- Your VAS virtual microphone is set as the input device of Skype;
- VAS read your real microphone in. (left panel of VAS)
- VAS output audio to real sound card. (right panel of VAS)

There are several buffers in the above configurations that need to change/test:
1) on Real microphone input panel of VAS:
- reduce the buffer;
- change the priority to 'real-time'

2) on Output audio to real sound card panel of VAS:
- reduce the buffer;
- change the priority to 'real-time'

3) on driver configure panel (in the bottom of VAS UI)
- reduce the internal buffers.


Also I noticed that if I just select my real microphone and my real audio card and just start the "Output" side it still works? Then what's the point of the left side?

The left panel of VAS is for reading your real microphone. If you do not to use microphone, you can ignore it.
admin
Site Admin
 
Posts: 68
Joined: Mon Apr 15, 2013 8:08 am

Re: buffers, delay and volume control

Postby Guest

Thank you so much. Especially on the buffer part. I have a very powerful i7 and the 512 is working the best. I did have one more question though.

Since I'm using your program to link two audio devices (through recording-listen to this device) is there a way to control the volume? For some reason I can control the virtual volume, but not both audio devices through Windows. Is this where I should really be using the application feature? So Windows controls the application.

Sorry about all the questions.
Guest
 
Posts: 65
Joined: Mon Apr 15, 2013 8:30 am

Re: buffers, delay and volume control

Postby admin

You can control the volume via the real sound card's volume control.
admin
Site Admin
 
Posts: 68
Joined: Mon Apr 15, 2013 8:08 am


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