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  • Wheresthedug - I am impressed! 20mS of latency is 40 ticks at 125bpm. Or to put this in real world terms, that's a perceptible snare flam or like someone clapping slightly out of time.

    I'm not saying that 20ms of latency is totally unnoticeable simply that it doesn't put me off enough to stop me playing in time.


    The Yamaha article seems to be referring specifically to live sound and singers with IEM where they can feel and hear the sound in their head disconnected from the IEM. They are referring to the comb filtering effect rather than the timing itself. The same article also suggested that in a studio environment that up to 20ms should be unnoticeable to most players.


    Focusrite say on their Q&A section that anything up to 11ms should make no practical difference to most people.


    Remember that even an acoustic piano has latency >5ms because of a) the hammer action and b) our position relative to the sound source which is at least 2ms with an upright and probably much greater on a grand piano.


    Obviously, you are very sensitive to latency and, if it bothers you then it bothers you. I have not sat next to you in the room and experienced your personal perception first hand but I suspect there is more at play here than 3ms of latency. I think there might be a healthy dose of placebo effect involved. This is something we are all affected by and is in no way intended as a slight on you personally.


    I will have some fun experimenting with the drum examples tonight but back in my day putting the snare late used to be called feel :). Some of my favourite drummers like Bernard Purdie were famous for laying so far back on the beat that they almost fell over ^^


    If you genuinely feel the Pod sounded better and had less latency issues you might be better served by its successors such as the Helix. I tried the Helix and KPA side by side for a full afternoon and they were beothe excellent. The Kemper sounded and felt more like my preference for real amps and the workflow was much better suited to my needs but they were both fantastic and it really was a matter of personal taste rather than better or worse.


    Burkhard ckemper can you guys through any light on the reason why the KPA supposedly has 10 times the latency of the humble Pod 2.0?

  • Michael_dk Thx for this.

    In my experience the opposite is true. It's often better NOT to hear the electronic version of the sound through headphone but instead ONLY hear the acoustic strum of the guitar strings. That way, what is - to me at least - what you're playing (and I guess after 40 years the sound of an acoustically strummed electric guitar is something I'm used to. Hence I often find myself taking one can off one ear and using this ear to listen to my acoustic strumming, with the other can listening to the backing track (with no electric version of what I'm playing).


    If I do listen to Kemper, the combined latency of Kemper + the ADC / DAC route of my audio interface (in direct monitoring) mode is about 5mS of latency - in other words what I hear is 5mS after I've played it. This - to me - drags and makes playing hard.


    This is just me - and it works for me.


    The strange thing about latency is - I am genuinely surprised that more musicians aren't worried by this sort of thing. For example, take a quantised midi drum track and randomly re-position the Kicks / Snare by +/- 5 ticks (=+/- 2.5mS @125bpm). Then try playing along accurately to it. To me, it's impossible.

  • Wheresthedug, if you can isolate the gene that lets you cope with this amount of latency, and develop a treatment for me - I am a buyer! :)


    And on the actual sounds, if by placebo effect you mean the "guitarist buys new gear, worries why it doesn't sound like their old gear, then tries to get it to sound like their old gear" effect, you may well be right. And it may well just be that POD v2.0 is familiar.


    What I'd like - I think - is just some simple, basic profiles of classic amps with an SM57 and / or MD421 a few inches back, gain settings that suit me and no room mics or ribbons. These seem to be hard to find. Perhaps I'm not looking in the right places. In fact, I may try and do the profiling myself with my remaining collection of amps - go to a local studio and spend a few hours doing this systematically. I have a feeling my wife would appreciate a profiled version of my Brian May AC30 rather than the real thing!!


    Thx for all the input.

  • Wheresthedug, if you can isolate the gene that lets you cope with this amount of latency, and develop a treatment for me - I am a buyer! :)

    :D Maybe that'll make me my fortune :)

    And on the actual sounds, if by placebo effect you mean the "guitarist buys new gear, worries why it doesn't sound like their old gear, then tries to get it to sound like their old gear" effect, you may well be right. And it may well just be that POD v2.0 is familiar.

    Yep, we are all guilty there.



    In fact, I may try and do the profiling myself with my remaining collection of amps - go to a local studio and spend a few hours doing this systematically. I have a feeling my wife would appreciate a profiled version of my Brian May AC30 rather than the real thing!!

    For my money that is one of the real major benefits of the Kemper. My wife has the ears of a bat and can hear an uplugged electric guitar from the other side of the house 3 floors up so not needing to crank my Mesa collection has been a revelation to me.


    I made some simple profiles at home and they work for me but going to a studio would be a better option if you can. Get a sound in the studio that rocks YOUR world (nobody else's world matters) and capture it in a profile for ever. Honestly I've done profiles with a 57 that are indistinguishable in a recording from the same amp recorded with the mic. Whether they are good, bad or indifferent sounds to someone else is a matter of taste but the Kemper can definitely capture them for you.


    Good luck.

  • I think what so many of us are confused about in this thread is Mark1964 is not talking about just flat out across the board latency. 3-4 ms absolute delay is really not perceptible, that is the above mentioned standing 3 feet / 1 m from the amp.


    The issue - I think - Mark is mentioning, and the Yamaha article talks about is the difference of delay between one "track" of music, the one you play vs the rest of the music.


    As you said, it is more of a curse than blessing if that 3.5 ms really is that perceptible to you. Maybe you could try to get a mix from the DAW into the Kemper Aux in and that way everything would be delayed (almost) the same. Using the Kemper headphone like this may help.


    Also the Yamaha article says that using speakers (instead of in-ear) the latency can go up and things are still playable. They say 5-10 is still playable with floor monitors.


    Maybe worth trying to use speakers instead of headphones. Just crank them up enough to not hear the acoustic sound of the guitar.

  • Hello fellow Kemperers... been away for a few days helping my mum with my Dad, the latter having just had a knee replacement. Both in good shape, which is good! Had a bit of time to do some thinking and analysis on the above thread.


    Here's what will probably win the award for the longest, most boring post by some random middle-aged bloke (if such an award exists...).


    First, my current project is a "live" version of ZZTop's Gimme All Your Lovin' - recorded very badly at a Christmas gig on a multitrack back in 2010. Among other things, I'm redoing the guitar part - so the "guitar" context is a part that's mainly (but not always) 2 strings, 4ths and 5ths and sometimes 3rds (always a good test of whether on overdrive amp sounds "good"), quite rhythmic etc. with an overdriven sound.


    When I play little chords like this on the Kemper and look at what a frequency analyser shows, typically it's a sort of skewed hump (skewed to the right i.e. the higher frequencies) BUT as the chord decays, you're left with about 3 quite sharp peaks which are the lower harmonics of the chord i.e. there's a burst of high frequency distortion at the start of the sound, but this quickly abates leaving, well, essentially a "clean" sound (and a fairly dull one at that). I know, as someone once said, you "engin-ear" not "engin-eye" but this is interesting, because a real overdriven valve amp seems not to do this. It somehow "smudges" those first 3 or so lower harmonics so they are much less prominent as the note decays, and the upper harmonics persist for much longer.


    This, I think, is what's at the root of the "flat" Kemper sound. To my ears at least.


    So, I get back home and decide to have a go at re-creating Billy Gibbons' finest. Thus, Gibson Explorer into my old Marshall 4210 combo used as head, into my Marshall 1936 2x12 mic'd with an SM57. And on the mic-ing I did something slightly different to normal i.e. mic more towards the edge of the 'speaker and mic angled so it's at about 90 degrees to the plane of the 'speaker cone i.e. off axis. This, predictably, sounds nothing like Billy Gibbons, and neither does my playing! But it is a nice warm sound, with enjoyable sustaining harmonics in the tone. And btw I did this two ways: one using the boost channel on the Marshall, and the other using the clean channel on a, well, clean-ish setting (everything on about 1/2) but boosted with a Maxon TS808 (drive=off, balance=max). Oh, and all this went into my old Roland VS2480 to make sure there were no latency issues.


    And to me the above sounds great!


    Given that I bought the Marshall combo in 1987, a fair criticism would be why I didn't figure this procedure out many years ago! The answer is: I did, but it's a bit of a hassle (see below). Plus - all the recent technology - including Kemper - maybe makes one not just more discerning about guitar sound, but actually more motivated (and able) to do something about it.


    And on latency. This set up - recording onto the VS2480 - is almost latency free. And it feels it. The VS2480 has been criticised for it's pre-amps and converters - and this may or may not be an issue - but it is pretty darn good when it comes to latency: no adjusting the I/O buffer or setting plug-in latency compensation and so on. Kemper in contrast, to me, has latency: when I hit a string the "thump" of that string hit from the body of the guitar in my chest is in time with my playing: the thump from the Kemper profile is late. I know I'm sensitive to this, and it's worse if I record through Apogee into Logic on my Mac. Most people don't notice latency, but I do.


    Now, here's the thing.


    Hassle.


    Doing all this reminded me of what a hassle setting up a real recording rig is. Some really enjoy setting gear up: in fact, so do I. But even so, all those wires, messing around with the amp controls, mic postioning, making sure the wife is out etc. And usually, by the time it's sounding "good" it's time to put all the toys away and go and feed the dogs. This to me, is one area where Kemper absolutely excels. It's just so easy: plug-in and play, and if you want a clean AC30 rather than an ODd Marshall it's there at the press of a few buttons. No wires, no noise, no hassle. It's a really great tool therefore for figuring out what guitar sounds / parts you want on a project, and "roughing them out" as it were. However, I haven't yet been able to make the jump to using Kemper as a substitute for real amps. Maybe I will get there in the end, and maybe I won't.

  • Question: is it possible to improve Kemper's latency performance?


    And here's some data as background.


    I did some tests, using a Logic Klopfgeist click recorded onto my old Roland VS2480. This, if you like, was the "Control Click". I then bounced the CC using various routings to test the latency thereof. Results obtained by really zooming in on the waveform and measuring are as follows:


    1. Out of VS2480 headphone output, back into VS2480 input. Result: 1.67mS delay (this is the "inherent" delay of one loop through the VS2480, obviously including it's DAC /ADC converter cycle, which is what you get when you record and monitor with this device).


    2. Out of VS2480 master output, back into VS2480 input. Result: 1.67mS delay i.e. the same as 1, above (this was just to test that there's no difference between using the headphones out and the master out of the VS2480).


    The following results are all stated both "gross" i.e. the latency of the whole chain, and "adjusted" i.e. the latency of just the component under test, after deducting the VS2480's "inherent" latency.


    3. Out of VS2480 master output, into Apogee Duet connected to MacBook pro and with Logic booted up @44.1kHz sample rate and 64 samples I/O buffer. Then straight back out of the Apogee's headphone out i.e. not using software monitoring. The idea here is to test how good Apogee's "direct monitoring" is. Result: 3mS delay. Adjusting for the VS2480's 1.67mS inherent delay, this means Apogee - even in direct monitoring mode - is adding another ~1.3mS of latency.


    4. Out of VS2480 master output, into 1998 POD on "POD Clean" patch. Result: 3.3mS delay (~1.6mS adjusted).


    5. As for 4. but using a different POD patch: this time the "Brit Hi Gain" patch. Result: identical to 5. above, so it seems POD patch has little effect on latency.


    6. Out of VS2480 master output, into Kemper with Stomps, Stack and FX all disabled. Result: 4.3mS latency (~2.6mS adjusted).


    7. Out of VS2480 master output, into Kemper on "AC30 Clean SM57" patch (I think the 3rd patch that comes up on the factory default list). Result: 5mS latency (~3.3mS adjusted).


    8. As for 7. above, but using AS Mars MP Gain 8 patch. Results: identical to 7. above.

    So what?


    Well, first, this means that in a "real world" recording scanario (recording using Kemper into an Apogee Duet using Apogee's "direct monitoring") total latency is ~3.3mS for Kemper (the manual I think says 3.5mS) plus a further ~1.3mS of latency for the Apogee Duet's "direct monitoring" i.e. a total of 4.9mS.


    Latency seems to be a bit like having a drink. You don't notice the first one, or even the second. But after a few drinks you definitely notice the cumulative effect.


    4.9mS is right on the cusp of when no less an authority than Yamaha say "playing starts to become difficult" due to latency (i.e. hearing what you're playing fractionally late verses your actual playing). Here's a link to that Yamaha article:


    http://www.yamahaproaudio.com/…ter5/05_absolute_latency/


    and, in summary, this is what it says.


    signal path latency for in-ear monitor systems

    1.15 - 2 ms Playable without any big problem.

    2 - 5 ms Playable, however tone colour is changed.

    5 - 10 ms Playing starts to become difficult. Latency is noticeable.

    >10 ms Impossible to play, the delay is too obvious.


    So, in summary, with a real world recording chain, the latency of playing using Kemper is such that "playing becomes difficult". This is my experience.


    Back to my original question: is there any way of improving Kemper latency? I ask because my 20 year old POD has about HALF the latency of Kemper (~1.6mS vs ~3.3mS). And POD plus Apogee Duet comes in at ~2.9mS which is less than listening to Kemper with headphones plugged directly into Kemper.


    To those that say "but 5mS of latency is like standing 5 feet away from a real amp / cab" I say: take a simple midi drum loop and shift e.g. the Kick backwards (or forwards) 5mS and then then try playing along to it. Or leave the Kick and instead move the Snare by 5mS. Or leave the entire loop as is, but add a click track mis-aligned by 5mS and try playing along listening to both. Then repeat the whole thing using 10mS instead of 5mS to see what Yamaha mean.


    To me, Kemper drags. It just does. And I think latency is part of, but not all of, the problem. But it would be nice at least to try to improve the latency bit.


    Thx!


    :)

  • Thx christianbad for your very helpful response.


    I'm not using Logic "software monitoring". In this test, the CC went directly from the VS2480 into an Apogee input and I recorded what came out of the Apogee's headphone output back onto the VS2480. This produced a latency for the Apogee (adjusting for the VS2480's inherent latency) of ~1.3mS. By comparison, at a 44.1kHz sample rate and 64 samples I/O buffer, Logic tells me its round trip and output latency are 5.8mS and 2.5mS respectively.


    If I go down to 32 samples I/O buffer Logic RTL becomes 4.3mS (output latency=1.8ms). And if I go up to a 192kHz sample rate AND 32 samples buffer, Logic says RTL is still 4.3mS but output latency decreases a tiny bit to 1.5mS.


    And in the real world, tracking at 32 samples I/O buffer and 192kHZ doesn't work due to glitching, even if - as I do - I record into a very simple tracking project with just a handful of basic audio tracks and no plug-ins whatsoever.

    I think Logic latency is therefore not relevant here, and all that matters is the combination of a) Kemper's latency and b) the latency of "direct monitoring" using my Apogee interface. On b), the issue is - I suspect - that "interface latency" in this scenario is almost (?all?) due to the Apogee's ADC / DAC convertors. Therefore, increasing the sample rate and reducing the buffer size in Logic has a no impact. And perhaps to prove this the Apogee "direct monitoring" latency I'm measuring of ~1.3mS is less than any of the numbers quoted by Logic (even at 32 samples / 192kHz).


    The problem with latency seems to be that cumulative effect is what matters. In total (Kemper + direct monitoring through Apogee) monitoring latency is about 5mS. According to Yamaha, 5mS is on the cusp of where "playing becomes difficult" and this is my experience. I'm not a virtuoso musician by any means, but I did all the piano grades as a kid and I've been playing the guitar for a long time so I'm not completely unmusical and - to me - Kemper drags: it just does.


    The real issue is that Kemper latency is most of the problem: 3.5mS out of 5ms = 70%.


    And if there's no solution for this, well... ...oh dear...


    Anyway, thx again, christianbad, for your very helpful response.


    :)

  • And just to add a few things:


    1. I think Kemper is really great idea i.e. modelling (sorry, "profiling") the entire signal chain, rather than components (pre-amp, power amp, cab, speakers, mic etc.) like other software guitar rigs do. And the convenience and simplicity vs. a real rig are great too. Especially for home recording. In principle at least...

    2. The relative lack of tweakability of Kemper vs. others is, IMO, a good thing because it's possible to waste a lot of one's life tweaking. With Kemper, if you like it great, if not try another profile. And it's not like there's a shortage of profiles to choose from.


    3. But WHY is the latency so high? Even with EVERYTHING (stomps, stack, FX) DISABLED? My experience with Kemper - and I've had mine for a couple of months now - is it's unusable as a direct recording tool for anything remotely precise. This is due to the cumulative latency of Kemper + "direct monitoring" with a relatively high-end interface. 5mS of latency (of which 3.5mS from Kemper) is an issue. At least it is to me. Everyone will tell me "What's wrong with you? 5mS latency?! You've gotta be kidding!" Everyone that is, except my ears and the "feel" when I play the guitar. Oh and Yamaha as well...

  • 5 msec of latency is what you hear with your ear five feet from your speakerbox with an amp. Unplayable? I think not.


    Yamaha, I suspect, are talking about the situation where you are hearing two sounds - eg your own vox in your head + the sound of your voice in the cans from direct monitoring or your DAW mixer. In that case you definitely get an interaction between the two versions of the same sound. Same goes for a horn player - eg sax - the horn sound is in your head acoustically as well as in the cans.


    Playing your electric guitar through the KPA and monitoring on cans - not so much. The direct sound is more or less in audible and the process sound totally dominates.

  • Yamaha, I suspect, are talking about the situation where you are hearing two sounds - eg your own vox in your head + the sound of your voice in the cans from direct monitoring or your DAW mixer

    I did read the article linked and this is exactly what Yamaha is talking about.


    Just straight up 2-3 ms latency really cannot have that much effect on humans, our plucking hand is 2 ms (feet) away from our ears. :)


    Mark1964 Did you have a chance to see if with speakers it feels (sounds) better for you while recording? Not quite sure why the same article says it would be but probably worth the try.

  • I play live, maybe 10 feet from the speaker, using a wireless.


    So I have 3.3 ms for Kemper, 4ms for the wireless and 10ms from the speaker....its a wonder I can play anything!


    Seriously I just don't notice this ( and I think I'm fairly sensitive to latency)...

  • The start of this thread was a cry for help, not an invitation for criticism. And the OP was not only on latency.


    But on latency specifically: it is a problem if you're trying to play - and record - accurately, and what you're hearing in the cans is 5mS late vs. what you're actually playing and feeling. It just is. Kemper feels like it's dragging. It just does.


    I only quote the Yamaha article as an independent source which disagrees with "the crowd" - " the crowd" as exampled on this forum.


    This "crowd" says things like I play "standing 10 feet away from my cab..." and "1 foot=1 mS..." and uses such arguments to make the case that latency doesn't matter and you can't hear it. Well, I - and Yamaha - say it does matter. And I can hear / feel it - and I think any half decent musician can too.


    Trying to track accurately in a recording context with Kemper and its inherent latency is not easy. For me, it's too much trouble - but I grew up in an analogue world where nobody had even heard of latency.


    These latency problems don't occur DI-ing a clean guitar, or indeed using my 1998 POD, or a real amp for that matter (which mic'd up and monitored through headphones doesn't have this sort of latency).


    Also, I don't think it's just latency. The attack of the Kemper isn't like a real amp and the sound lacks sustaining upper harmonics - please see the start of this thread.


    Anyway, I'm a bit disappointed that there doesn't seem to be a solution from Kemper to the latency issue. It feels like I've inflamed a sensitive spot (that spot being Kemper latency), because nobody has made any suggestions on the other issues I have with Kemper in my OP. I thought forums such as these were supposed to be helpful - rather than scoffing.


    Never mind.


    This will be my last post.


    I thank the minority of forum relpy posts that were helpful / understanding on these important topics.


    And I wish everybody the very best of luck and happiness with their musical endeavours.


    Best regards,



    Mark 1964

  • I thank the minority of forum relpy posts that were helpful / understanding on these important topics.

    Important topics for a majority of people have thousands of posts. The main reason you get no answer is because this is not an important topic for the majority of people. I understand that it is for you. And I am so sorry that you can't enjoy all the other benefits the Kemper offers. But Kemper is not the only option out there, so in your case I would go for one of those better options for you instead of wasting so much time trying to fix something that apparently has no fix.


    And you should not be upset with other users just because they don't find this a problem for them.


    The Kemper is not perfect. We all agree in that. We all have to choose the option that is the better one for us. For many it is the Kemper, for others are tubes, or other modelling amps... We are lucky to have so many options today.

  • Also remember that no technology exists that can eliminate latency. A signal path will always result in analog to digital conversion, processing and then back again if you do not use SPDIF. What helps to keep latency low on the Kemper is that latency isn't constant and does change based on how taxing the processing on the CPU. That's why there's a "constant latency" mode in the output section so that if you do reamp, it ensures the latency is kept at a particular constant setting. I don't notice any latency and I am going into a line 6 interface via SPDIF. It really does come down to personal preference. I hope you find a device that will eliminate your issue.

  • Yamaha, I suspect, are talking about the situation where you are hearing two sounds - eg your own vox in your head + the sound of your voice in the cans from direct monitoring or your DAW mixer. In that case you definitely get an interaction between the two versions of the same sound. Same goes for a horn player - eg sax - the horn sound is in your head acoustically as well as in the cans.

    They are indeed. In that article they actually say that around 20ms of latency should be virtually unnoticeable in situations where you can't hear the two sounds together.


    "The latency of a signal chain in a typical networked audio system at a sampling frequency of 48 kHz is around 4 milliseconds. The main factors are distribution latency, DSP latency and AD/DA latency. In recording studio’s with isolated control rooms - where the listener never hears the sound source directly - latency is not a problem at all. If the listener can see the sound source (but not hear it), a latency of up to 20 milliseconds (corresponding with the PAL video frame rate of 50 Hz) is allowed before video and audio synchronisation mismatch can be detected - mainly due to the slow reaction time of the eyes and the visual cortex in the human brain."


    Focusrite quote a figure of around 11ms before it is noticeable.