Posts by EdwardArnold

    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.

    Thanks for finding the thread - having looked through again, it was 44.1 kHz operation that produced the extra latency, but this was solved in subsequent releases as quoted.

    Yep, give it a go. It’s likely that the checkbox will stop the software from pulling up/down the sample rate on load. Go for whichever is the easiest workflow option.

    I don't think there are any compelling reasons to go back to 16bit for recording purposes; the 144dB of dynamic range that 24bit recording offers, vs 96dB for 16bit dramatically improves the signal to noise ratio and lessens the requirement to drive inputs near the maximum threshold and risk clipping a good take. The cumulative reduction in noise over multiple tracks is beneficial across budget and pro audio interfaces alike and presents more noticeable, audible improvements than say, doubling sample rate.

    Regarding the extra latency at sample rates other than 44.1 kHz, can you remember what the values creep up to? I remember there was a discussion on it but can't locate the thread now. I thought it was addressed in one of the beta releases but looking back through the change log I can only spot the 'reduced latency at 44.1kHz' note from v7.5. Is it possible this was the other way around, as in, the latency was high when operating at 44.1kHz?

    Under normal operating conditions (i.e. no S/PDIF) involved I measured latency at just over 2.5 ms to main outs, which is a good result considering the A/D D/A stages. When recording I monitor via an analogue desk, so direct from Kemper and mixed with the DAW playback, then shuffle the audio regions earlier in Pro Tools as required. If you have a means of reliably measuring your input latency, I guess you can apply this numerically rather than visually.


    As for which clock should be master, unless your interface lifts its output relays when re-clocking, I'd keep it on internal and avoid it blatting through your speakers/headphones. Some software also tries to control the audio interface settings (Pro Tools, for example), so loading up a session can change sample rate to match the previous settings. If the interface is the master, this can happen more elegantly without you needing to manually change the kemper, then re-clock the interface etc; the Kemper should just follow.

    Hi, is the noise the same no matter where you plug in to mains in your house? For example, does the upstairs mains ring sound any different to downstairs? It seems you've already identified that some equipment is dumping noise onto your mains ground signal or similar so if audio equipment is using that for a reference it seems it could become audible. An electrician is likely to compare the earth connection once mains to a copper stake in the ground outside (or at least they did at a studio install I was working on) for a differential but not likely to put an analyser on the mains to look for noise. There might be someone more mains savvy one here who can advise.
    A DI box can lift the ground connection just like the Kemper does, but this is for prevention of ground loops where errant 50/60Hz signal flows partially through mains socket to socket, then via audio cabling between the bits of equipment powered for those sockets (I'm perhaps telling you what you already know here). If the noise is part of the audio signal though, ground lift won't touch it as you've found. A passive DI box is transformer-based and will provide physical de-coupling in addition to ground lift, but again if the noise is already part of the audio signal, it will get transformed too. Still worth a try if someone can lend you one.

    It might be possible that a mains RF filtered socket extension would help. Quite a few affordable mains extension leads feature this, filtering errant high frequencies that can sometimes cause issues (I have some D-Link powerline network equipment that operates by dumping 2.5MHz signals over the mains in my house, which is great for networking but not so much for some of my audio equipment).
    Best of luck with it.

    No gigs for me, but I’ve thought about just busking around my local town to get my kicks. Started travelling again with work (technical rather than musical) to some theatres and concert halls in Vienna and they intend to open at 50% capacity next month.
    Back in the UK, there are rumours that the live hire places I deal with are dusting off their consoles and wheeling them out in anticipation of things starting to pick up a little bit. I really hope so, but it’s difficult to imagine still.

    It’s highly likely that the stagebox inputs onstage are feeding mic preamps not line ins as Ingolf has suggested. Many desks don’t have mic/line switching for their XLR inputs, and sometimes also no input PAD to drop it signal down. Plenty of mic pres have a minimum gain value of 15-25 dB or so though. What desks are you typically running into at different venues, if you recall?


    If the Kemper’s -12dB isn’t enough attenuation, what you require is a DI box that can step down the signal level further e.g. -20dB and -40dB steps are common. I’ve said it before on these forums and been challenged, but a DI box is not completely replaced by the output features of the Kemper, nor the input features of smarter consoles, and this is one of the examples.
    If you use a passive, transformer-based DI, you’ll have the added benefit of being decoupled from the FOH rig as well, providing some level of protection for your kit if there is a floating earth at a venue or a mixer PSU that decides to let go.


    DI boxes are also ideal for wedging venue doors open, tilting back real amp cabs, tripping up people who shouldn’t be wandering the stage, throwing at drummers and muting banjos etc. so they really are the musician’s Swiss Army knife that should be taken to all gigs.

    Low cut = High Pass filter

    High cut = Low pass filter


    This is referring to the high and low frequency content of the signal. As you move the crossover frequency point around (the only control in this case), you will hear all frequencies below/above being attenuated for the HPF/LPF respectively. Give it a try and you’ll hear what it’s doing right away.

    What’s the workflow challenge that you’re trying to solve with more metering? E.g. preventing overload of analogue inputs on a mixing desk or audio interface?


    Is the metering you imagine a bar graph-style meter with a dBFS, or a dBu scale? The former would be common to most digital desks and DAW metering, where analogue equipment would feature the latter.

    Looks like Lindy make a TOSLINK to Coaxial SPDIF converter, available from Thomann and others. Also cheaper (possibly lower quality) items on Amazon.

    If you’re new to re-amping, have you already explored the analogue method and ruled it out?

    Thanks very much for these videos Martin. I learned a lot from the second one especially, as I was blissfully unaware bootleg grounding was a thing. Sketchy! I take a basic socket tester with me to venues that checks for earth and whether the mains pin-out is correct. My question is, would this be able to tell the difference between a good earth and a bootleg earth connection? Presumably it’s lighting up good when there is no resistance, so a bootleg connection to neutral would fail this?

    I’ve checked my (UK) home fuse box has RCDs as well as resettable fuses, as I’d not thought to use an RCD at home otherwise. I have a couple of in-line RCDs that I usually take for gigs and work trips. I’ll change this to ‘always’. I appreciate your advice.

    Yep, a guitarist as a hobby - I don't play on a scale that would warrant a large format digital console capable of these processes, unless I steal one from work and hire a van to shift it (it has happened). For a day job I'm a Systems & Support Engineer for a console manufacturer. I count myself fortunate to work with a lot of live, broadcast and studio engineers internationally on tours, festivals, concert series or other live-to-air events. Primarily I work through the preparatory stages of facility commissioning, tour rehearsals/first gigs, provide operator training, often through first shows or station broadcasts as well as offline events. It's the broadcast work that has drilled mono-compatibility into me, but live pop/rock/world music tours where stereo widening techniques seem to be deployed readily to, at least in part, clear up busy centre-focused mixes and make a narrow stereo mix seem wider than it is without hard panning.


    I don't see the workflow as just for guitar; it's usable for any single point source sound, and yes the context of an overall mix would be considered. It's also born out of practical use and experimentation rather than just theoretical; I've either tinkered with it myself when mixing or worked through it with others who are far more talented than I, putting it to use it on backing vocals, small brass sections, widening mono effects returns, a host of world music instruments or in some cases anything in a busy centre-focused mix that can afford to be lost in the mono down mix due to the cancellation. This is where the centre channel can help - there aren't many guitarists that would agree they are sonically dispensable. Typically, it's the sort of workflow question that emerges when a tour has gotten underway and engineers look for subtle refinements they can add in now that they are recalling the mix settings from previous nights and not having to start from scratch each time.


    Having settled on this mono+sides workflow when required, I'm still trying to work out the best way to combat the more obvious remnants of the Haas effect, and have been trying out all-pass filter effects to rotate phase of particular frequency bands on just the early channel. I got poor results when trialled at a tour rehearsal though as it seemed to exacerbate comb-filtering effects. An engineer at a theatre I regularly visit mentioned he used the HPF trick on the early signal to try and improve things. This is different to 'usual' widening workflows I've seen, where a low pass filter is used on the delayed signal to keep the effect without adding to the 3k and above busy part of the mix. I'm not convinced of that method, but it seems reasonable though that if you high pass the earlier of the two delayed components (left through this example) you can reduce the chance of comb filtering with the centre component or the right channel, and the inter-aural amplitude difference favouring the R component will slightly offset the effects of inter-aural time difference favouring the left. To my ears, this is what seems to happen. These later conclusions are indeed more 'theoretical' explanations to what I perceive, so it would be good to get someone else's opinion. Worth experimenting with though. I'm by no means an authority on the subject of widening though, having 'found my way in the dark' to get to this point, with some experiments and collaboration. This is why I mentioned a PA systems tour tech or a seasoned FoH engineer would be a good person to comment, given that they are either having to solve phase incoherency or deliberately introduce it for creative effect on a more regular basis than myself.


    It's quite amazing to me what the Kemper can already do. It's this ability to break new ground that makes me keen to throw further ideas into the ring - I shan't be offended if they are deemed as not worth pursuing, just trying to offer something that may be of some use.

    I'm aware my suggested method is a bit of a hack as it is for a single audio source rather than a true mid/side mic technique using a fig-8, and side channels need to be used sparingly as a result. It's a shared workflow with some other live engineers though, so I'll try and explain as best I can.

    How would you benefit from a non-delayed center channel, making the signal drifting back to the middle (Haas effect) for the audience positioned around the center line?

    I've tried to retain the mid/side workflow at least by using the centre 'dry' channel for the following reasons:

    • To provide the full frequency range of the source without HPF - HPF is used to reduce the Haas effect for side channels.
    • For use as the primary point source in the mix (so some pan could apply within the stereo mix), then act as a reference around which the sides are constructed/faded in, always retaining a point of reference. The idea is that the Haas effect of the L channel arriving first will have been reduced enough by the HPF that it won't 'disturb' the pan position of the main component as such. (The placement of L and R components can also be done relatively if the main component is panned, as you would a stereo reverb, if you use a stereo bus send set to 'follow channel pan').
    • To fall back on exclusively if the mix destination is purely mono (PA or broadcast), or if comb-filtering effects in a particular venue are noticeable enough to render the effect unsuccessful. You can just mute the sides channel rather than having to change delay values.
    • If there is a guitar solo, the centre channel can be pushed to bring focus to the centre, just as the guitarist struts to the middle of the stage for a while. This is easier than boost + pan operations, requiring one fader only, and you know it'll work for the mono destinations too.
    • For compatibility with, centre fills, mono repeaters (possibly LCR PA systems), which are likely to use a mono downmix of a stereo master via a matrix rather than multiple masters with 2.0, 3.0 and 1.0 formats. Although the HPF reduces the noticeable effects of comb filtering in low frequencies, a mono downmix of the sides alone will have lost those low frequencies entirely; the dry channel as the main component provides them.

    And how would the audience positioned to the left and right benefit from mostly listening to the delayed side channels?

    The dry channel is still the most prominent component of the sound that they hear; the delayed channel is faded in only enough to generate the effect. I referred to the dry signal as 'centre' but if it needed to itself be panned it could, as described above. For the moment, let's say it is still located centre. Audience members left of centre would therefore perceive the dry signal to their left, as the amplitude is greater from that speaker and it arrives first. The L delay signal would come exclusively from the L PA out, with the intention of expanding the image between C and L, so that wide image of the guitar moves with them relatively the more left of centre they are. There would still be some pickup still from the R PA side and mono front fills that contain the other stereo components, downmixed. The interactions are surely only as complex as if you try to mix mono through multiple PA zones, but again if it's not working the sides channel can just be discounted. I think the point at which this method comes unstuck is when audience members are very close to a singular PA point source. If it's a centre fill it's ok as the dry component takes precedent and mono compatibility is there for this purpose. If it is either the L or R speakers exclusively, then there's not much helping them any case, as they can only ever be listening in mono, but at worst they will be listening to C + delayed L, which I suspect is only as complex in terms of phase interaction as the reflections coming acoustically from the venue.


    Now, what we really need is a PA tour technician that has to figure out these interactions with a SMART meter all the time, as I could be way off the marque in some of my claims without realising it.


    If were to simplify things for a moment though, do you think just a subtle HPF on the left channel of the Stereo Widener might reduce the Haas effect enough to please those that notice it?






    I’m fairly convinced there is one digital trim element before the stack that is shared/affected by rig volume, pure boost, EQ volume etc. Once you drive to +12dB subsequent increases have no effect. Is this what others have found/is this already common knowledge? The amp, if you dare switch it in, has its own gain, and also input gain becomes active, so perhaps the AcSim needs to move to the amp block in future to make use of another gain stage? If the existing one was retained in its present state as an EQ only version as well, people could still combine with preamp profiles if they want.