Digital distortion not due to high input levels

[Bflobassist]

I've see a few threads talking about distortion, and it seems the replies are implying a problem with audio levels. So I thought a thread for ideas on this problem and how to correct it was appropriate.

Some background; I've been involved with audio engineering and recording (analog and hardware digital recorders) for some time and have a few digital consoles but I have not really entered the full on ProTools world of complete computer based recording. Hopefully that gives a basis for the accuracy of my descriptions of this problem, especially in reference to typical 'digital distortion' due to improperly set gain structure or not understanding How 0 dbu differs with analog vs digital audio signals.

We run band JamKazam rehearsal sessions between 5 people sometimes for few hours, each using an audio interface for vocals and their instrument, and have been doing this for quite a few months. Occasionally, one person will complaint of 'static' or 'distortion', including me. By 'occasionally' I mean the signal between us is good, clean, and low latency but then the 'static' will happen for anywhere from a few seconds to someone needing to leave the session, reboot their setup and rejoin the session. 

What we noticed was people using Behringer interfaces had no problems. We tried an older Focusrite Scarlet Solo, Focusrite 2i2 Gen3, and both and Allen & Heath QU16 and SQ5 digital mixers' interface functions. All except the Behringer hardware interfaces give that occasional distorted 'static'. Further details of out setup is hardwired Ethernet connections (of course), Windows computers all running 10, but some desktops and some laptops, typical total latency between members is 30ms or less, though we see 'red' indications for internet and I/O jitter occasionally. We are all using The manufacturers' ASIO drivers, and the sample buffer is forced by JamKazam to the lowest possible value available, which is 6ms I think. When we try to change it to, say, 32ms, JamKazam changes it back to the lowest setting within seconds.

We all now use Behringer interfaces, from the cheapest UM2 to the UMC1820 and we have not had any 'static' issues since. But I would rather use the QU16 as my interface if not for the 'static' problem.I've investigated USB signal jitter reducers and re-clockers, but I am not sure that is the problem. 

Has anyone has similar experiences and, if so, how did you solve it? Would LOVE if a JamKazam programmer could chime in with the technical perspective of the cause. 

Thanks!

[StuartR]

I've see a few threads talking about distortion, and it seems the replies are implying a problem with audio levels. So I thought a thread for ideas on this problem and how to correct it was appropriate.

Some background; I've been involved with audio engineering and recording (analog and hardware digital recorders) for some time and have a few digital consoles but I have not really entered the full on ProTools world of complete computer based recording. Hopefully that gives a basis for the accuracy of my descriptions of this problem, especially in reference to typical 'digital distortion' due to improperly set gain structure or not understanding How 0 dbu differs with analog vs digital audio signals.

We run band JamKazam rehearsal sessions between 5 people sometimes for few hours, each using an audio interface for vocals and their instrument, and have been doing this for quite a few months. Occasionally, one person will complaint of 'static' or 'distortion', including me. By 'occasionally' I mean the signal between us is good, clean, and low latency but then the 'static' will happen for anywhere from a few seconds to someone needing to leave the session, reboot their setup and rejoin the session. 

What we noticed was people using Behringer interfaces had no problems. We tried an older Focusrite Scarlet Solo, Focusrite 2i2 Gen3, and both and Allen & Heath QU16 and SQ5 digital mixers' interface functions. All except the Behringer hardware interfaces give that occasional distorted 'static'. Further details of out setup is hardwired Ethernet connections (of course), Windows computers all running 10, but some desktops and some laptops, typical total latency between members is 30ms or less, though we see 'red' indications for internet and I/O jitter occasionally. We are all using The manufacturers' ASIO drivers, and the sample buffer is forced by JamKazam to the lowest possible value available, which is 6ms I think. When we try to change it to, say, 32ms, JamKazam changes it back to the lowest setting within seconds.

We all now use Behringer interfaces, from the cheapest UM2 to the UMC1820 and we have not had any 'static' issues since. But I would rather use the QU16 as my interface if not for the 'static' problem.I've investigated USB signal jitter reducers and re-clockers, but I am not sure that is the problem. 

Has anyone has similar experiences and, if so, how did you solve it? Would LOVE if a JamKazam programmer could chime in with the technical perspective of the cause. 

Thanks!

Since you're all using Windows machines, have you applied the appropriate tuning steps to ensure proper real-time audio handling? As a starting point be sure all these machines are NOT running on battery power and must be plugged into AC power to be sure their performance is maximized. But there are many additional steps required... See below for a great reference guide:

https://www.cantabilesoftware.com/glitchfree/

[Tobivi]

@StuartR

Thanks for this info. Yes, all machines are running via AC power with the power profile set to max performance.

I'll check out the publication to see if I can get better results, and will post what helped if it does.

[Kannon82]

This debate, which is based on digital distortion might helpful for IT worker to understand this concept of the digital world. The college essay writing service can help you to cope up with the burden of assignment work and other academic help.

[Chris Frost]

@StuartR

Thanks for this info. Yes, all machines are running via AC power with the power profile set to max performance.

I'll check out the publication to see if I can get better results, and will post what helped if it does.

I remember reading somewhere that the sample rate needs to be consistent or an exact multiple .... for example: ‘All on 48k’ or ‘mixture of 48k & 96k’ .... but not with one person on 41k.  JK has a ‘sync’ option to deal with this (at least it did when I set my Audio up - but I know that controls sometimes disappear)

I believe that the logic behind this is that the sample rate is used for digitisation of the Audio .... and if you are not in sync with everyone, you get gaps - which then sound like a click. .... unfortunately you would get the same effect if the A to D converter part of your Audio i/f had a fault..... and I wonder if those are actually designed to be used just between the musician and the DAW (same unit used for recording as playback) ..... and therefore if this is a bit like analogue audio cassettes being recorded on on machine and played back on another .... and souring.