Need help changing sample rate

[OakHillChurch]

We upgraded our sound system at church where we now use OBS for streaming. 3 weeks in a row we have had audio issues with it crackling, popping then eventually cuts out all together. I have read that this is a common issue and we need to change the sample rate to 44.1. I have changed it in OBS but in the Windows Sound Settings I cannot change it. It seems to be locked at 48. I attached a screenshot. You can see it menu is not dark. Anyone have an idea on how to fix it? I have been doing sound for 30 years, video for 15 years. I have never been so close to quitting than I am now. We don't record our streams so I'm not sure if I can provide a stream log. I am hoping to fix it with the sample rate settings. If not I will find a way to provide a stream log on here. Thanks guys.

 

[AaronD]

in the Windows Sound Settings I cannot change it. It seems to be locked at 48. I attached a screenshot. You can see it menu is not dark. Anyone have an idea on how to fix it?

Wow! I've never seen that disabled. You must have something special.

I have been doing sound for 30 years

Analog?

Digital problems are different, because the underlying mechanism is different.

Signal processing problems are *exactly* the same; you don't gain any kind of "magic abilities" by switching to digital, and the "professional sound" of a modern concert is completely achievable on a pure-analog rig. The only real gain to be had from that switch is an amount of toys that if you were to have all of that in analog, you'd have an entire room with packed-full racks lining every wall. And the ability to save and recall settings.

We don't record our streams so I'm not sure if I can provide a stream log.

A recording would help a lot, even if it's only for support. Just like analog problems can often be diagnosed by ear, digital ones tend to have their own signatures as well.

It also helps to be able to see what the waveform actually was originally, before the encoder throws away what it thinks we can't hear in order to squash it into a small number of bits. For that, use a lossless audio format, like FLAC.

The logfile can be accessed easily from OBS's help menu. It's a text file with a bunch of "debug spew", as software engineers sometimes call it.

 

[Phandasm]

Wow! I've never seen that disabled. You must have something special.

This happens with devices that either only support one recording mode or need to be configured with a 3rd party utility.
For example I have a steinberg interface that appears to have the sample rate locked in the windows settings but can actually be changed with their special app.

I'd suggest checking with the manufacturer and see if they provide any special drivers/software.

 

[AaronD]

This happens with devices that either only support one recording mode or need to be configured with a 3rd party utility.
For example I have a steinberg interface that appears to have the sample rate locked in the windows settings but can actually be changed with their special app.

I'd suggest checking with the manufacturer and see if they provide any special drivers/software.

Ah! Okay, that makes sense. I've used some Behringer digital mixers that only had one possible configuration for their built-in USB "sound cards". Channel count to match the console, 32-bit integer, sample rate to match the console's setting, which itself requires rebooting the console to take effect. The host just takes that or leaves it. No options on that end.

That probably has the Windows setting locked too, but I've never had reason to check.

All of that said though, no specific sample rate is either a problem or okay. It's when *different* parts of the rig use *different* rates that things go awry. And that even applies to the same nominal setting, but derived from different clocks that are just slightly different from each other. DSP functions have no notion of time, only samples, so if the same number of samples is maybe 1/2 second different over an hour......

Thoroughly professional rigs make sure to have a single clock feed everything, so that drift doesn't happen. The master clock may not necessarily be dead-on either, but no one can tell because everything is sync'ed to the exact same thing and you're not going to notice that small of a "tape speed error".