Budget Standalone "Toslink > DSP > Toslink" with Camilladsp. Set up instructions for newbies.

[melomane13]

Just recieved the U24 XL. Got it running on a windows laptop for now.

However the only way I can see to loopback the input signal to the output is to check the setting "listen to this device" in the windows recording audio device settings, which adds noticeable latency.

Do I need additional software like something from VB-Audio to route the signal via ASIO for the lowest latency? If so, is this also a problem I will run into on linux when eventually migrating to a Pi?

i not see particulary latency with RPI4 .

 

[mattzildjian]

Rather frustrating that the U24 XL offers direct monitoring of the analog inputs but not the digital inputs, so to monitor the digital inputs you have to go through the windows "listen to this device" setting which always adds latency. Maybe linux just has a better way of looping back audio?

 

[melomane13]

@mattzildjian , linux is free. so you can install on double boot on pc , when install camilladsp and try with alsa.

 

[mattzildjian]

@mattzildjian , linux is free. so you can install on double boot on pc , when install camilladsp and try with alsa.

Yeah I could do this, maybe make a live boot usb.

There has to be a way in windows though.
I am finding all sorts of applications / drivers that may work.
SAR / SynchronousAudioRouter
ASIO Link Pro
Voicemeeter
R3LAY VPB Virtual Patch Bay

Anyone familiar with these? I just want the digital input to route back to the speaker output with as little latency as possible.
Thanks.

EDIT: Could this function within EqualiserAPO work?

Equalizer APO / Documentation Wiki / Configuration reference

sourceforge.net

EDIT2: Solved it using Voicemeeter, it allows low latency monitoring of input devices via ASIO which the U24 XL officially supports.

EDIT3: Another issue, getting audio dropouts every 20 minutes or so. It's a similar sounding drop out that the CM6206 card was doing, though that was doing it constantly. The problem might be with my laptop and not the card perhaps. Not sure how to troubleshoot this.

EDIT4: Switched the input type from WASAPI to MME on voicemeeter and not had a dropout for 30mins, so hoping that solved it though I'm not holding my breath. I am able to set the asio buffer size as low as 32 samples and not have any glitching, which is quite remarkable. There is zero perceivable latency. Can linux do that? Tested for a further 30 mins or so without drop outs so seems good right now. I can turn the TV speakers on in combination with the DSP'd optical signal and they are practically bang on top of each other, the latency is so minimal all I hear is a little bit of phasing between them. Wondering now if windows is actually a best solution because of asio support.