Behringer UMC1820 And Ardour

I’m just getting the Behringer UMC 1820 to run in Ubuntu Studio 18.04 out of the box!

You can change WordClockSource and muting of ADAT-Channels in AlsaMixer.
You can watch this:

I was able to have 0.7ms latency with 96kHz.
You can hear (or not) here:
I was also able to use the 8 optical ADAT I/O without any problems.

The only thing was to get rid of puleaudio:

I did some tests and recordings to show setup, two time monitor-mixing, ADAT and latency demo on youtube: /channel/UCINH_Fh0Ego_IWIVed3RUmA

It’s in German but you’ll get the point.

I like to use an RME 96/8 PAD on my old PC. That PCIe Card was an expensive on!
The UMC 1820 was about 180,- Euro, and now I’m able to use USB an a newer Laptop
for recording.

I’m locking forward to test the Mic-Preamps against the ones build in my old Yamaha O3D.
You will hear that on my YouTube-Channel, next weeks.


I started with UMC404 and upgraded to Umc1820. Worked out of the box

As a question :How can I use both units ? in other words 4 input channels of the umc404 and 8 input channels of the umc1820 ? Qjackctl does not allow for it.


You have a few options. I do not think these boxes allow external sync (word clock for example) but if both of them do, then it is possible to create an alsa_multi pseudo device that looks to jack like one device with all io from both boxes.
Option 2) even if the two boxes may be synced together, it may be easier to use zita-ajbridge with the synced option (-S).
Option 3) unsynced use of zita-ajbridge. This uses software syncing also known as SRC (Sample rate conversion).
Option 4) use alsa_in and alsa_out which come with jack, they are similar to zita-ajbridge but I am told, do not do the SRC as well as zita-ajbridge does. “Not as well” means doesn’t sound as good and uses more cpu. (I have not tested if this is true) So far as I know, alsa_in/out does not have a synced option, it is SRC only.

With all of these options you obviously want to use the unit with the greatest i/o as the jack master. You do not want to have an stereo pair cross both units (if left is on unit (a) the the right should also be on unit (a)). The master unit should be used for things that are loudest and most important in your mix (lead vocals and lead instruments).

@lenovens , thanks for the reply !!! I am currently busy in other areas but will come back to it and keep you posted.

thanks and best regards


From Len’s comments I would suggest you try Zita first. I once did a few tests with a bass and a stereo pedal to split the signal and recorded both on Ardour as two different tracks using a Scarlet 2i2 and a Uphoria UMC with 2 inputs. For the test I did with alsa_in there was a small but noticeable delay from the second interface while using zita I could not notice it.

Hello, since some users of the umc1820 and ada8200 have the same problem (alsa is muting some channels)
i have found out a quick solution:

open a terminal window and type in the following:

press F6 to select sound card (in this case UMC1820)
The displayed sound-cards are starting with 0 (that means the first one has the value zero, the second one the value one and so on)
Find out the right value for your umc1820
Exit alsamixer by pressing ESC

type in the following lines: (the value after the “-c” parameter has to be changed to your sound-card number that you just discovered in alsamixer. In this case change the “2” to your own number)

amixer -c 2 cset numid=1 on,on,on,on,on,on,on,on,on,on,on,on,on,on,on,on
amixer -c 2 cset numid=3 127,127,127,127,127,127,127,127,127,127,127,127,127,127,127,127
amixer -c 2 cset numid=9 on,on,on,on,on,on,on,on,on,on,on,on,on,on,on,on
amixer -c 2 cset numid=11 127,127,127,127,127,127,127,127,127,127,127,127,127,127,127,127
amixer -c 2 cset numid=8 0
alsactl store

The lines will enable all outputs, enable all inputs, set all output volumes to max, set all input volumes to max and set the clock to internal .
that last command will store your configuration.

this did the job for me without reinstalling the whole os.

do it on your own risk!

Hello all,

Audio interface newbie here! I 'd like to ask the following questions regarding this Behringer unit:

  1. When used as a plain USB sound card (eg, YouTube, media player, etc) does the USB send audio to the main outputs as well as headphones? Follow up questions for this scenario:
    1a. WIll the audio coming from USB get mixed with the 1-4 inputs when the mix knob is set all the way to the left (IN position)?
    1b. Have you experienced any issues with the sound (when switching from the onboard sound card of your PC to the audio interface) getting either high or low pitched (as if you play vinyl records in higher or slower speeds)*
  2. When powered on does ALSA/alsamixer remember the volume levels for each input/output channel from the last time it was used? **
  3. Can it work on its own without turning on my PC (standalone mode, for example just hooking up the speaker cabinet simulation output that my guitar amp attenuator has for silent practising).

FYI, I 'm on KDE Neon (Ubuntu LTS based distribution) using Pulseaudio. Thanks for any response :slight_smile:

  • I get this behaviour sometimes with a Scarlett 18i8 1st Gen with or without JACK being involved
    ** With the Scarlett 18i8 1st Gen this is not the case, but I can workaround it with alsactl
  1. Yes.

1a. No, that is the point of setting it to the “IN” position, to only hear the inputs. Conversely, if you set it fully to the “PB” position you will only hear the USB playback. Setting the dial at twelve o’clock gives a 50/50 blend of the inputs and USB playback.

1b. No, but this likely due to a sampling rate setting difference between the two interfaces you are using. One probably defaults to 44.1k and the other 48k.

  1. Unlike the 18i8, the UMC interfaces have no internal routing software. Powering on the interface will have no effect on the ALSA settings you have stored.

  2. I am not entirely sure what you mean by standalone mode. It is a computer audio interface. What would you be accomplishing by using it without a computer? Are you wanting to use it as a headphone amp? I have not tried this.

Hi Gunther,

Thanks for your response:

  1. Can you set the Mix knob in between positions or is it only IN, 50/50, PB?
  2. With regards to sampling rates, doing some more reading my understanding is that Pulseaudio handles resampling. You can configure it so that the sound card follows the source’s sampling rate, but I haven’t tried that and it is suggested to better avoid it. I need to investigate this a bit more.
  3. You can save Scarlett 18i8’s routing/mixing in it’s internal memory using Focusrite’s software and use it as a mixer without connecting it to a PC. I appreciate that the UMC1820 has a default routing which you can’t change, but the question is can it be used without turning on my laptop, eg, just connect a source in one of its inputs (in this case speaker cabinet simulation) and hear the audio from its main outputs and/or through heaphones?

It is a continous knob, so yes, you can set in-between positions.

With an audio player like DeaDBeeF, you can set it up for audiophile playback with no resampling: You select your interface output analog or digital but the important thing is to select an ALSA option with “direct hardware device without any conversions”. You also disable ALSA resampling. FYI, this all works seamlessly without needing to disable pulse audio on your system.

Yes, AFAIK. The UMC1820 (and UMC404HD) comes with its own power supply so it would work without needing power via USB.

Thanks bachstudies. I think I get the whole idea now with the audio interfaces. I have the aforementioned Scarlett 18i8 1st Gen for a long time now, but never had the time to properly try it out while now with all those lockdowns it’s easier.

I 've done a bit more testing today with some bash scripts that I 'm working on and looks better than I thought, although all the mixing/routing options which the Scarlet has still confuse the hell out of me (and imagine I 'm an IT/Linux engineer by profession :stuck_out_tongue:)

I 'm not sure if I want to keep it though or go for the Behringer. I don’t need all those inputs/outputs, but the fact that the majority of the XLRs are on the rear panel makes things easier for my space. Having said that there is no availability at the moment in all the big UK/European music shops for most of the UMC units.

Thanks very much all for the feedback you provided so far :+1:

The recent politics aside and nature of “cloning” other companies’ designs, the UMC series are fantastic devices for the price. As far as I’m concerned they knocked it out of the park with the UMC range. The fact that they work out of the box in Linux is the icing on the cake.

So, at the moment I 'm fine after all the work I did today tweaking my script, systemd and pulseaudio. They all seem to play nicely together with/without JACK. I just need a pair of small monitor speakers (I 'm leaning towards the Fluid Audio F5) and I 'll be sorted. I 'll be keeping an eye though on those Behringer interfaces and if I 'm not happy with the Scarlett I may give them a shot.

Thanks for all your help people! :+1:

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Was this related to the incorrect SPDIF clock sync, or was this some other problem with your original unit?

Anyway, thank you for all the help with this post. Very informative.

It was not related to a clock. Without any audio cables connected the device just clicked continuously. The second device I got had not problems at all.

Thank you [Mikael Hartzell] ( The OP )

This has been a very helpful thread,.

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After reading this thread I decided it is time to try the UMC1820. Ordered from UK new (price in US was too steep) I received it this morning. Less than 3 days shipped from the UK which is faster than buying here in the USA.

For those that uses Jack here is a thread where I describe a HOWTO for getting the UMC to work
jack UMC1820 howto

I can only say one thing… I am completely blown away. Not a Behringer fan due to all the previous low quality crap, but the UMC blows my Presonus so far out of the park it is not funny.

I just happened to have a keyboard player in this afternoon and she commented how much better everything sounds. didnt mention any equipment changes to her. She just said it right of the bat after a minute of playing. She mentions how previously unusable crap sounds (like the horrible unusable guitar patches) suddenly became utterly useful. I am a bit baffled. I cannot believe that there can be that difference between comparably priced interfaces. There is something with the Midas preamps… I am totally sold - period. Nothing changed except swapping the Presonus with the UMC. Same mixbus same eq, same jack, same alsa, same server/Linux same cables. The fact that completely unusable programmed keyboard sounds on the Motif XS8 suddenly now sound useful is beyond comprehension. Those previous crap sounds used to sound flattish and now they have so much depth. Completetely and utterly useful and inspirational. It must be the combination of Midas and an exceptional A/D converter.

She played a few notesm then asked me why this is suddenly so nice what did you do ? Completely independent verification as she was used to the Presonus. I am blown away literally. Some irritating sounding patches now became quite beautiful and useful.

Either Presonus is very bad which it is not really, (I have two and they both sound the same, so no chance of a defect really) or these Midas preamps has some amazing magic to them. It must be the latter. There is definitely a huge improvement in intermodulation distortion. it is my guess that the Midas preamps and A/D is absolutely linear, whereas the Presonus input amplifiers are more non-linear. Low intermodulation distortion (due to ultra l high linearity, will always give a marked improvement in separation of sounds and therefore increase in depth markedly, and that is what I am hearing.

I bought the UMC as a placeholder until I get my Motu B16 and LP32, but I am starting to wonder if the Motu could be better than this. I dont think it can be. The B16 is indispensable to my needs and I need it for several reasons, but I am going to load the LP32 with Midas preamps for sure. Too good sorry.

I guess if you didnt have another interface before you bought a UMC it is not going to be obvious what my experience is. I would like to try an Apogee in future though.

All I still need to get working is SPDIF.

I’m glad you had a good experience with the device. And you probably have better ears than me because I don’t hear that much difference between the audio devices I’ve had (Presonus VSL1818, Zoom R24, Alesis IO2, Alesis IO4, UMC1820). Part of that might be that I’m always recording rock with distorted guitars.

The price has come down since I bought the device and I think it’s amazing value for the 205 euros the device goes for at the moment.

One user had problems with spdif due to the expectation that it would always sync to incoming spdif when it appears on the cable. But the device switches sync to internal and stays there whenever spdif signal disappears. One needs to then change the setting in alsamixer manually when wanting to use spdif input again.

Thanks a lot. I will change it in alsa. Didnt get round to it yet.

I dont know why everybody here hears such a huge difference. It is MASSIVE.

You have to do the test with a very good Production Keyboard like Motif XS8 and not just single staff instruments. It must have the extreme wide spectrum of an electronic production keyboard. As an example a Wurli sounds the same on both interfaces -to be expected. It really happens to change and diverge once you have lots of voices and pads that needs to be separated properly. There are lots such keyboard emulations and patches. The difference is enormous. If a preamp and ADC has linearity problems you will immediately hear the difference as the non-linearity introduces distortion that will remove the depth of the sounds and you will hear that several pure voices played together doesnt separate properly and extra harmonics is created. This makes the sound flatter and voices less separated. Arguably, if you record distorted instruments, the intermodulation distortion of an interface preamp will be masked as the same intermodulation frequencies are likely already created by the distorted source. That is what makes heavy rock great and pleasant, it is the flatness and ball to the wall sound, which will most likely mask a nonlinear preamplifier as the source is already filtered through non-linerarity, the overdriven guitar etc. Therefore under that scenario preamplifier linearity has less of an audible effect as it is another subtle distortion effect way less than the source…

That is exactly what I hear.\ and others hear the same.

Absolutely enormous difference. No turning back for me. And that is taken into account that I really had no interest in Behringer. What I hear must definitely be the linearity of the Midas preamplifiers compared to the presonus. Midas seemingly knows something presonus dont.

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SPDIF used to work out of the box (DAW Channels 9&10), but now stopped working. Where in alsa do you set the spdif clock?
There is no SPDIF clock source setting in alsamixer I could find. There is only Clock Source (internal,optical) but no spdif.