Posts by EdwardArnold

    You could monitor the incoming USB audio using Quicktime (included with macOS) and 'New Audio Recording'. This lets you record to stereo and/or monitor. Presumably this is so you can play along to something in Apple Music or similar, otherwise I'm not sure what it nets you over monitoring from the Kemper direct. You can switch this around of course; computer out over USB to Kemper, to blend with your guitar signal. That might be preferable if you are experiencing latency monitoring through macOS.

    OP, Fault categories can include hardware, software/firmware and physical connections. I think all anyone is trying to do here is narrow it down to one of these seeing as you have asked the public community for an opinion. If you want to talk to Kemper Support directly, contact them appropriately if you have not already - it would make sense given they have the most insight to more serious issues. Remember that as disappointing as equipment failures may be, they are no-one's fault per se. Please let us know the outcome for future reference.

    I've only looked at the recording possibilities so far. I had an SSD error and lost multitracks coming off an A&H Qu mixer recently, (QuDrive) and it struck me that I could simultaneously record the USB out at the same time direct to Logic Pro for iPad as a redundant recorder next time. I have only tested it at home so far, but 24 tracks at 48 kHz ran for about an hour completely fine on a 2020 iPad Pro. There seems to be a slight bug if 'auto select audio devices' is enabled in terms of redetecting them after disconnection, but otherwise solid. Not sure I get the complaints about the subscription model; it is very affordable for what it is. I spent £75 for 1 month of disk recovery software to try and get my missing WAVs back and it failed, so 5 or 6 quid a month for Logic seems fine to me for the months I want it, and I'm not yet touching 90% of the feature set.

    That would be either Bluetooth or integrated proprietary hardware that supports Airplay or similar. I wouldn’t confidently bet against it in light of recent announcements, but it seems unlikely. You can pick up an Apple Airport 2 with Airplay cheap as chips, join the Kemper to that Wifi network and connect it via the returns in Aux mode.

    A mix engineer will want to see an average signal of -18dBS with peaks up to, say, -12 or -9dBFS for incoming digital signals.


    For analogue desks, average of 0dBu would be considered ‘line up’ level. Depends on how dynamic the signal is and whether it will be feeding a unity gain line input or a mic pre with a minimum gain value, as to whether it needs padding or not.

    I have some thoughts on this.
    A lot of stageboxes/wall boxes feeding FOH mixers are primarily XLR, and most of what comes from stage via XLR is mic level, with most XLR inputs on most consoles being mic level and not switchable (often a separate TRS input). Even at minimum gain values, mic inputs could be adding +8 dBu or as much as +28 dBu, with the only option to Pad down the input, or if no Pad, run it through a DI and take it down to mic level first at source. This is a good reason for the output Pad.


    Line level out of the Kemper into a mic pre at minimum gain, even with the -12dB output pad, could still be hot for a lot of preamps, but this is the same as signal from other line sources.


    TS outputs are commonly -6dB compared to balanced outputs, whereas specs say they are the same, so I can see that catching people out. In fairness, plenty of engineers would guess that a TS output from your *pedalboard* would be Instrument level without looking further into it, but wouldn’t choose to run that to FOH anyway.

    Lastly, given the noise floor of the Kemper D/A converters is going to be way below anything generated by a distortion effect, I wouldn’t have any worries about optimal levels when working below the 0 dBFS point in the output trim personally, with or without -12 dB engaged. (I also still have a sneaking suspicion that half of this pad function is in the digital domain -totally accept I could be wrong here.

    the -12 dB option attenuates the signal by 12 dB after it has passed the D/A converter. The volume level in the output menu attenuates the signal before the DA conversion. It is recommended to use the -12dB option first before attenuating the signal further in the digital domain.

    Running a test tone with a known signal level through, I found that the attenuation applied is either 6dB or 12dB depending on where in the chain your source your output. This suggests at least one of these is actually a -6dB digital trim before the amp stack and cab, so before the signal is back in the analogue domain. Unless there are two analogue pad circuits? Can you explain more about how this is actually operating? I have no issues with it, just interested/confused at some of the info posted. Thanks.

    Not sure of the context here but particularly if this is for live use, the Shure PSM-300 system I use is analogue wireless and therefore very low latency. The P3RA beltpack is nice and has a built-in limiter and EQ if needed. It's the first rung of the ladder for their stereo systems. Pair with some 215 or better IEMs and you have a good solution.

    None of this sounds as good as some Beyer Dynamic DT 770s connected direct though when at home.

    i guess the XLR outputs of the Kemper are electronically balanced rather than employing isolating transformers and would therfore have some protection to block DC entering given this is a "pro" bit of gear.

    I'm also interested in whether the line output drivers are happy to repeatedly take phantom power being applied. Seems fine but I don't want to test the theory more than once. A line isolating transformer rather than step-down D.I. box is what I've used. I recall a gig where an analogue console power supply let go spectacularly mid-soundcheck, taking out some connected monitor wedges and an outboard effects unit. I had an amp miked up then, so a suitable air gap, but it made me keep a Radial Stagebug in my bag for the Kemper since.


    Also, have just visited a BBC studio and any stage box inputs around the facility have 48V on as standard. As long as it's not a ribbon mic or an iPhone, it seems the approach is that it needs to be able to take it.

    I think tenderboy makes a really good point; placement in your studio space and even modest acoustic treatment can have a significant impact - one that is worth exploring/experimenting with if you haven't already. In terms of those published specs, they show they are not the flattest response, varying more than say a standard Genelec 8030 or similar. In particular, you can see the effects of the ported design on the low-end response in that plot, which has a peak below 50Hz that you should just be mindful of in case it impacts upon some of your decisions with the low end of your profiles or kicks off room modes, especially if you are playing bass guitar through your Kemper as well. In context though, if you like them and they sound decent, it doesn't matter much what the specs say. You can always check custom profiles or mixes with a pair of good headphones as well to ensure you're not cutting low end too much to compensate, for example. For comparison, I took some measurements in my room for my Genelec 8030As and the response was 'dramatic' in the mid-to-low end despite some efforts with treatment. I can still mix in that room with them though, and it never comes to mind when I play guitar...

    Awesome read, thank you. I've gotten as far as racking up a Qu-SB, DBX driverack, wireless router and Shure PSM system into an 8u case with power distribution and a drawer for the iPad etc. Seeing your videos I'm back to wondering if I can squeeze in an ARPnet controller for lights as well... Cool stuff, well done.

    Hmmm. The spec page quoting +4dBu implies to me that this is stating the operating level, i.e. both units use the -20dBFS = +4dBu line-up standard, which defines the voltage output would be 1.228 V at +4dBu.


    The specs for maximum output level are, as far as I'm understanding this, the maximum output level that can be driven prior to the onset of distortion. So, the Stage's output drivers are just a touch over 6dB lower (7dB according to the published figures but I'd put that down to rounding discrepancies between the two). 6dB is the difference between a balanced and unbalanced line, so I think the output driver has been redesigned on the stage to deliberately align the levels for the balanced and unbalanced outputs, thus avoiding any unexpected leaps in level when different interconnections are used. Otherwise, I guess it would drive +21/22 dBu cleanly as well.

    Not sure about the board swap - seems highly likely that the SPDIF circuit is integrated into the main PCB. That aside for a moment, I wouldn't get too hung up on master clocks. It's common for manufacturers to infer that the clock quality/PLL chip they've used is going to create some sort of leap in sound quality, when the reality is clocking chips need to be reliable and not drift away from a known sample rate and provide steady modulation without jitter. SPDIF embeds clock signal this way, the same as AES. When that embedded clock signal is received it will be regenerated. So, if your clock in your interface would be slaved to an external device (Kemper), but it would still be regenerating its own clock signal as far as I understand it.

    Instead, I would always think about the workflows that your chosen clocking scheme dictates. It's unfortunate that you will need to slave to the Kemper in this case whenever you power it/attach it and want to use it. If you use your interface frequently without it, then you'll have to keep in mind any blats that can occur as the interface re-clocks, reaching your speakers or headphones. Ideally, the interface would lift its output relays when reclocking to avoid this, but I wouldn't know for the Apollo - worth finding out though as it makes much of the issue go away.

    This sort of stuff is going to bug you much more than the perceived sound of one clocking chip vs another.

    I agree that I prefer to work in samples, ms and dB, but then I suppose not everyone does.

    On the subject of delay for widening purposes though, I find just delaying a leg isn’t ideal. The Haas effect means that the earlier signal is prioritised and the stereo image subjectively shifts, so you then either try and compensate by level or by filtering off lower frequencies from the earlier signal to try and take the energy out of it. This also helps reduce the loss of low end due to phase cancellation for anyone listening in mono (front fills, repeaters, recording or livestream in mono etc.).
    If I’m ‘manually’ widening sources when mixing, I’ll create mid+sides channels so that the centre signal can be the earliest signal and provide mono compatibility. And on a digital mixer, you can dea exclusively in Hz, ms, samples and dBFS ?

    There is a delay widener available that you can put after the stack (in one of the stereo slots). If the PA destinations have any mono fills, just get the engineer to check mono compatibility is passable by folding down during sound check. Alternatively, you may have more success with the two other widening effects available. Check out page 167 of the manual for more. The wideners are under the 'EQ' section, which isn't immediately obvious.