Clicking/popping noises from the Kemper on Linux

    I'm experiencing intermittent soft clicking/popping noises when I try to use my Kemper Powerrack on Linux, but not on Windows, using the exact same hardware and connections. My physical setup is:


    1. Guitar plugs in to front input of Kemper,

    2. Kemper spdif output connects to Scarlett Focusrite 18i20 spdif input (Kemper is set to master in the Scarlett control software),

    3. Scarlett connects to the computer via USB.


    I'm using Bitwig for recording, which is set up on Windows using Asio4all, and on Linux using Pipewire. Bitwig, Kemper, and Scarlett are all configured with a 48kHz sample rate.


    I'm pretty sure the clicking noises are caused by the Kemper, because when I plug a different amplifier (Yamaha THR30II) into the Scarlett, there is no clicking on Linux or Windows. The clicking is making it into the actual waveforms recorded in Bitwig, and I can hear them when I play back the recorded tracks. The equivalent tracks recorded on Windows do not have the clicking sound. I've tried:


    - two different guitars and both experience the same issue

    - two different guitar cables and both experience the same issue

    - plugging the guitar cable into the alternative input instead of the front input; still experienced the same issue

    - routing the line outputs from my THR30II into the alternative input on the Kemper; still experienced the same issue


    I'm not quite sure how to troubleshoot this. Has anyone else experienced a similar issue? Any suggestions would be greatly appreciated!

    Troubleshooting:


    1. Is the audio interface compatible with Linux?

    https://support.focusrite.com/…ct-compatible-with-Linux-

    Ok, so it may only work in class compliant mode and there's no drivers and/or control software for Linux available.


    2. You get clicks and pops under Linux but not under Windows when using S/PDIF.

    You've set the audio interface to sync to S/PDIF under Windows, how do you do the same under Linux without a driver and/or control software? You probably don't because you can't without a driver / control software.


    3. You state that all works fine with a Yamaha THR30II connected to the audio interface.

    Well, the Yamaha THR30II doesn't have S/PDIF ... so you hooked it up through analog IO, e.g. TS cables. Try the same with your Kemper Profiler, just don't use S/PDIF when on Linux.


    This particular issue should be solved now. But here's a few more words:

    Why do you use ASIO4ALL on Windows when the audio interface comes with ASIO drivers?

    ASIO4ALL only makes sense for audio interfaces without ASIO support.

    Also, even if you use a 3rd party software like Pipewire for low latency recording and playback, you still can't use the zero latency DSP mixer(s) of your audio interface. So you're missing out on Linux, no matter what. If you can, stick with Windows for your audio related work ... saves you from a lot of headaches. :)

    Quote

    You've set the audio interface to sync to S/PDIF under Windows, how do you do the same under Linux without a driver and/or control software? You probably don't because you can't without a driver / control software.

    I'm making the (possibly incorrect?) assumption that the settings are saved to the interface when I set them in Windows. That appears to be true for some things -- e.g., if I set one of the inputs to "AIR" or "PAD" in the Windows control software, it remains set even when I switch over to Linux. (I have Linux and Windows computers both hooked up to the same equipment via a KVM switch, so it's pretty painless to go back and forth.)


    Quote

    Well, the Yamaha THR30II doesn't have S/PDIF ... so you hooked it up through analog IO, e.g. TS cables. Try the same with your Kemper Profiler, just don't use S/PDIF when on Linux.

    Good thinking, but I just tried it, and unfortunately still get the popping...


    Quote

    Why do you use ASIO4ALL on Windows when the audio interface comes with ASIO drivers?

    Tbh I don't know much about audio drivers; I did it this way because if I understand correctly that's what Bitwig recommends (https://www.bitwig.com/support…ing-wasapi-on-windows-36/).


    Quote

    Also, even if you use a 3rd party software like Pipewire for low latency recording and playback, you still can't use the zero latency DSP mixer(s) of your audio interface. So you're missing out on Linux, no matter what. If you can, stick with Windows for your audio related work ... saves you from a lot of headaches. :)

    I used Pipewire because it was the only way I could get Bitwig working without just plugging it directly into alsa (and thereby monopolizing the sound card). I'm not sure what zero latency DSP mixer(s) you're referring to -- do you mean the Focusrite control software? How would I use that to hook up to Bitwig (or any other DAW)?


    Working in Windows would definitely be one way to solve the problem! I'm still hoping to get everything working smoothly on my Linux box though -- I feel like it's quite close now, everything works except for this pesky popping issue...

    Quote from tinybike

    Tbh I don't know much about audio drivers; I did it this way because if I understand correctly that's what Bitwig recommends

    Quote from the page you linked:

    "To record audio in Bitwig Studio on Windows, you will need to use another driver called ASIO. Most audio interface manufacturers release their own ASIO drivers for Windows. If your audio interface does not have a dedicated ASIO driver, we recommend that you use ASIO4ALL:"


    Focusrite does provide ASIO drivers, so it's recommended to use that driver instead of ASIO4ALL.

    Quote from tinybike

    I'm making the (possibly incorrect?) assumption that the settings are saved to the interface when I set them in Windows.

    I can't tell you whether the Focusrite 18i20 automatically saves all settings made in Windows. But what I can tell you for sure is that you don't have access to 'internal' settings unless you have driver and control software. Only settings with dedicated 'hardware' access (knobs, buttons) on the interface will be accessible in 'class compliant' mode.

    On Linux you're forced to run the audio interface in 'class compliant' mode ... which is basically the smallest common denominator for using an audio interface without dedicated driver. Very rudimentary implementation of audio provided by the OS manufacturer, not the hardware manufacturer. If you want to leverage all features and possibilities of your hardware, you need to use it on a supported platform with the software provided by the manufacturer. No way around that, sorry.

    Quote from tinybike

    I'm not sure what zero latency DSP mixer(s) you're referring to -- do you mean the Focusrite control software

    Your audio interface has built-in 'mixers' you can control with Focusrite Control. This way you can send different mixes to different hardware outputs with (almost) zero latency input monitoring.

    One basic example:

    You have a singer and a guitarist in your home studio. They want to jam (and record) over a backing track you've loaded or arranged in your DAW.

    1. You're the recording engineer and you want to hear the DAW playback and the singer and the guitarist on your studio monitors in a nicely balanced raw mix. So you create a dedicated monitor mix for your monitor outputs.

    2. The singer wants to hear a bit of the backing track coming from the DAW and just a little bit of the guitarist. But he needs to hear his own voice (microphone) loud enough and without any noticable latency to feel comfortable to control his performance and pitch. So you create a separate mix for e.g. headphone output 1.

    3. The guitarist obviously needs to hear his guitar the most and e.g. snare, kick and bass. He doesn't mind to also hear the vocals and other stuff ... but he wants these other parts to be lower in volume. So you create a separate monitoring mix for headphone output 2.


    Your audio interface provides all these things/features/options with pretty much zero latency .... IF you can control it and setup these mixes through Focusrite Control.

    If you setup monitoring mixes in your DAW (because you can't use Focusrite Control), it will inevitably come with quite noticable latency cause the input audio has to run through USB->DAW->USB.

    In case anyone else has this problem, here's the solution. It turns out that ALSA has its own built-in word clock settings that you have to configure in Linux. So what you need to do is, open alsamixer, hit F6 and select your interface (the Scarlett), then use the up/down arrows to select SPDIF for the clock source. (Run sudo alsactl store to make the change permanent.) This caused the clicking to stop for me.


    Re: using the Focusrite Control for mixing, that's a good point. Actually I do this already! I have both my Linux and Windows boxes attached to the Scarlett via KVM switch, so it's a single button press to switch over to Windows, adjust anything that needs to be adjusted in Focusrite Control, and then click back over to Linux for the actual recording. (If I was starting completely from scratch, maybe just doing everything in Windows would be smart, but -- like many programmers -- I have a strong preference for Linux as a computing environment.)