Latency reamping with my KPA into Logic

  • I am getting latency when reamping. I think I am getting some too when tracking, which is arguably more worrying since you need low latency to really nail your performance and have it on grid. I will focus here only on the latency I get when I reamp, since I still need to double check that I am indeed getting latency too when tracking.


    Hardware & software I am using:

    KPA -> Focusrite Scarlett 6i8 -> Logic Pro on an 8 months old MacBook Air


    I have set every setting I know to clamp down on latency as best as possible. In Logic:

    Low Latency Monitoring Mode is checked.

    I/O Buffer Size set to either 128, 32 or 64 Samples

    I even downloaded this Focusrite Low Latency widget.


    And still this is what I get when I reamp: Lag between DI track and Reamped track (click on to see image)


    Notice how the waveform of the Reamped starts later than the DI track. That lag amounts to 25 milliseconds. It's noticeable to the ear when you listen to the song. The reamped track doesn't quite sound on time.


    Interestingly enough, lowering the I/O Buffer Size from 128 Samples to 64 Samples, and further down to 32 did not make any difference. The lag between the DI track and the reamped track remained at 25 milliseconds for every buffer size. :rolleyes:


    Lastly, and given the fact that none of the above seemed to be reducing the reamp latency in the slightest, I set the Recording Delay to -528 Samples as per here to make up for the lag. Now there is no longer a lag between the DI track and the Reamped track. :thumbup:


    I feel disappointed that none of the settings aimed at reducing latency mentioned above did much (if anything) and that I had to resort to tweak the recording delay to clamp down on the latency for good. I feel it's just a patch-up to work around an issue which is more fundamental and that in this time an age should be long dealt with by our modern hardware and software.


    I don't rule out having omitted or done anything wrong in the process of trying to reduce latency that I describe above. If you have anything to comment to this regard, could you please enlighten me? It will be forever appreciated.


    All the best




  • That latency is way too high. You should be somewhere around 3-5ms. Something maybe wrong with audio driver or the interface. Have you ever updated the Focusrite drivers? I personally use Windows and Reaper with a Motu. I remember checking my offset once but found I was only 1/2 a sample off so I left it alone since the minimum adjustment was 1 whole sample.


    Check this out...


    External Content www.youtube.com
    Content embedded from external sources will not be displayed without your consent.
    Through the activation of external content, you agree that personal data may be transferred to third party platforms. We have provided more information on this in our privacy policy.

    Larry Mar @ Lonegun Studios. Neither one famous yet.

    Edited 2 times, last by BayouTexan ().

  • Thanks for your input BayouTexan!
    The video basically talks about adjusting the recording delay, as I mentioned in the initial post. For now I will work with this, but to put it mildly I am surprised that for a project that's not overloaded at all like mine is, and with devices that are relatively new, I am getting such amount of latency.

  • Thanks for your input BayouTexan!
    The video basically talks about adjusting the recording delay, as I mentioned in the initial post. For now I will work with this, but to put it mildly I am surprised that for a project that's not overloaded at all like mine is, and with devices that are relatively new, I am getting such amount of latency.

    I agree. Something is not right. I've never heard of that much latency before. I don't think I've ever read someone having more than 6ms. Makes you kind of wonder if the computer is have a memory or processor issue. Can you try on other peripherals to test?

    Larry Mar @ Lonegun Studios. Neither one famous yet.

  • Sorry for chiming in a bit late but here's my thoughts:


    1. At the lowest commonly used sample rate of 44.1kHz, the 528 samples you mentioned would equal 12ms. You would get 25ms compensation with 528 samples only at a sample rate of 21.12kHz (which I doubt you're using). :)


    This short episode of nitpicking aside, here comes the important part. Be aware, it might get a bit technical from here:


    2. An audio interface typically reports its latency at any given buffer setting to the DAW so the DAW can compensate for that. What isn't actually reported to the DAW is the gear you have hooked up to your audio interface ... e.g. the Kemper Profiler's own latency. Even if you have hooked up e.g. a AD/DA converter via ADAT or MADI, the DAW will not know the added latency introduced by these devices. So the DAW doesn't know what it should compensate if you use the Kemper Profiler like a regular "input" source.


    3. This changes if you setup an "External Effect" and use it as an insert effect in your DAW. Now you have basically created a complete signal chain that your DAW (and the "External Effect" feature) can measure and compensate for, typically by a short ping. I do this all the time, especially for reamping. It helps to set the Profiler to "Constant Latency" to get consistent ping latency values most of the time. This part might be tricky if you're not that much into audio engineering but I'm pretty sure Logic Pro can do this ... personally I use Cubase Pro or Nuendo and it works like a charm.


    4. To make things even more complicated ... the above only makes sense for reamping. But most of the time you probably just want to record your playing directly. So it would be very helpful if you could setup your DAW I/O in a way that you can use the Kemper Profiler as a regular input source for audio tracks AND as an "External Effect" on an insert slot without changing your DAW I/O configuration all the time. This is possible if your audio interface has a decent DSP mixer software (like the RME interfaces I keep recommending). But it would become a novel if I would try to explain this step by step here and now.


    Hope this helps at least to understand the cause for the issues you (and likely many others) have. :)


    Cheers

    Martin

  • so the DAW can compensate for that.

    personally I use Cubase Pro [...] and it works like a charm.

    This! I can only confirm that. Works like a charm here as well once the routing and input/output options of the Profiler, the audio interface and the DAW including all their mixers are set up well - which can be a bit tricky indeed depending on the audio interface and related software. Thanks lightbox for the great explanation :thumbup:8)


    Not using the Profiler in the above described way as an external effect to the DAW will most likely always create the latency topic with reamped tracks. Potentially resulting in phase issues and so on. Some semi-automated alignment of the waveforms could potentially heal that as well. But I assume never as accurate the above described method.

  • This part might be tricky if you're not that much into audio engineering but I'm pretty sure Logic Pro can do this ... personally I use Cubase Pro or Nuendo and it works like a charm.

    I've never done it (but will be trying now 😎) but as a Logic user I can confirm it does have the option of setting up external devices as an effect insert.

  • Not sure it matters, but do you have a lot of processing on your master bus?

    Kemper PowerRack |Kemper Stage| Rivera 4x12 V30 cab | Yamaha DXR10 pair | UA Apollo Twin Duo | Adam A7X | Cubase DAW
    Fender Telecaster 62 re-issue chambered mahogany | Kramer! (1988 or so...) | Gibson Les Paul R7 | Fender Stratocaster HBS-1 Classic Relic Custom Shop | LTD EC-1000 Evertune | 1988 Desert Yellow JEM

  • Dear lightbox, you're right in time. Thanks for chiming in!

    I am on the go so cannot check what you wrote in detail until I get back home.
    For the time being though, on this you said:

    1. At the lowest commonly used sample rate of 44.1kHz, the 528 samples you mentioned would equal 12ms

    I first tried with 1056 samples, but then I saw that the waveform for the reamped track was ahead exactly by the same lag as the DI was before applying any recording delay. So I set the delay to exactly half of it, that is 528 samples, and it worked just fine.

    I may have gauged the wrong lag amount to begin with, and the lag might be 12ms instead of 24-25ms. I did it using the marquee tool in the track window that opens at the bottom of the Logic project screen when you double click on a track stem. I could send snapshots when I am back home, but in any case the 528 samples adjustment seemed to do the trick. Visually both the DI and Reamped waveforms start now at the same point on the grid line, and to my ear they sound aligned with the tempo.

    I'll come back to the rest of your points above later when I'm back in my studio.

    Thanks a bunch for your input.

  • This! I can only confirm that. ... Thanks lightbox for the great explanation :thumbup:8)


    Some semi-automated alignment of the waveforms could potentially heal that as well.

    Thanks :)


    Regarding the semi-automated alignment would heal that after you have reamped, correct. But only as long as your Profiler effects chain stays within the boundaries of "Constant Latency". If you use e.g the Transpose effect, you would have to check and measure manually again . That's why I wrote "most of the time" in my previous post.

    By using the "External Effects" insert, the re-adjustment is only one quick mouse click to execute the ping measurement and the corresponding compensation. :)


    Cheers

    Martin

  • ... I could send snapshots when I am back home, but in any case the 528 samples adjustment seemed to do the trick.

    Thanks a bunch for your input.

    Don't worry, I totally believe you that the 528 samples fixed it for you. There was just something off with the milliseconds you mentioned, hehe. :) Typical german nitpicking, nothing more, haha.


    The important part is after that and once someone gets his head around that (and is the proud owner of a capable audio interface), it can make life so much easier in the long run.

  • Would you mind telling us how you do that?

    Got it. I wasn't worried about CPU usage. The latency does add up though and might result in some weird results if the DAW is not properly latency compensated.

    Kemper PowerRack |Kemper Stage| Rivera 4x12 V30 cab | Yamaha DXR10 pair | UA Apollo Twin Duo | Adam A7X | Cubase DAW
    Fender Telecaster 62 re-issue chambered mahogany | Kramer! (1988 or so...) | Gibson Les Paul R7 | Fender Stratocaster HBS-1 Classic Relic Custom Shop | LTD EC-1000 Evertune | 1988 Desert Yellow JEM