Previous 1 2 Next 17 Replies Latest reply: Jan 28, 2009 11:34 PM by AFS
AFS Level 1 Level 1 (0 points)
I could not find a way to contact iShwartz directly, so I hope you see my post here.

iShwartz said:

"But pretty much all other audio systems do not report driver latency to Logic, and therefore the recording delay value must be calculated (via the loopback test) and entered manually into the recording delay setting."

How do you conduct a loopback test please?

Also, do you or does anyone know if Pro Tools LE 7.4 Digi 002 A/D reports latency issues to logic?

To get the lowest latency while recording I use the 32 I/O buffer setting.
Also as instructed, I bypass all plug-ins and set "Compensation" to Audio & Software Instruments Tracks, with "Low Latency" mode checked.

I also make sure I'm running no other applications while recording with logic, including Airport.

Having said and done all this, could anyone please tell me WHY is my MacBook Pro Dual processor 2.5ghz constantly STILL displaying ..."Disk too slow, or System overload" when recording audio?

I have made an appt. with a mac genuis to see if something is wrong with my MacBook Pro.

Hope someone can advise or provide some answers.

One more thing...what is software monitoring?
should I have this not checked if I am monitoring from SPIDIF out from my Digi 002?

iShwartz you out there?

thanx~

AFS

MacBook Pro 2.5 Ghz, Mac OS X (10.5.6), 2 Gig RAM
  • Sampleconstruct Level 5 Level 5 (5,870 points)
    32 samples as a buffer could be too much for your Macbook, so try 64 instead, maybe that solves the overload issue whilst recording. When mixing raise the Buffer as latency is not a problem in the mixing process.
    To do a loopback test proceed as follows:
    Create an Audio file with a sound that has strong transients, like a clicktrack played with a rimshot sound. Route that sound to your Audio Interface (e.g. Outputs 1-2) and on a new Audio track underneath the original Audio track choose Inputs 1-2 and re-record the rimshot sound through your Interface into Logic, set the outputs of your re-recording track to "no Outputs".
    Now use the magnification tool to zoom in all the way on the first transient of your original file and compare it to the rerecorded file. The offset between the two transient is your recording latency. Then adjust this Latency by setting the recording delay slider in Logic's Audio prferences to that value.
    I recently did a loopback test with my RME FF 800 and it turned out that I had a negative latency of 54 samples (1,125 Milliseconds at 48 khz), so the RME actually recorded everything too early by 54 samples. Setting the slider to +54 samples gave me a perfect sync of the two transients.
    Also make sure that whilst recording Audio you set Logic's Plug In delay compensation to "Audio tracks and Instruments" and not to "All".
  • AFS Level 1 Level 1 (0 points)
    Sampleconstruct,

    Much thanks for your help I get what you are saying I think....so once U'r zoomed in U measure the Milliseconds & that's your Latency offset?

    You know, Logic and Audio I/O's should get together and you should be able to either push a button and have this latency issue job fixed on the fly somehow like they have with Appogee so I read here earlier.

    Also, can you explain how the software monitor button affects latency/CPU issues?


    It seems to me it's a big deal that Logic is such a CPU hog.
    Am I wrong about this? I would think that Mac owns them and writes code for Logic to run on their machines.... Pro Tools can do it? why can't Logic? I know Pro Tools has it's own system but it still has to run on a Mac just the same.

    Whatever the problem I hope they can solve some of these "disk too Slow, or System overload" error messages. Do you think once I get the latency value offset's right I will not see the "disk too Slow, or System overload" anymore or less so?

    Thanks for your help!
  • iSchwartz Level 5 Level 5 (4,315 points)
    Hi AFS,

    The procedure Sampleconstruct outlined is on the money. Just a few points of clarification, just in case they're not totally clear...

    • the loopback test is done by physically patching the outputs of your interface to the inputs using patch cords. In this case, you'd connect out 1/2 of your interface to input 1/2

    • you want to measure the delay in samples, not milliseconds.

    • Sampleconstruct suggested having PDC set to "audio tracks" but I prefer to have it turned off.

    • You want to do this test on a blank song with no plugins anywhere, especially on your outputs. I strongly suggest opening the Explore/Empty template, adding an audio file (as Sampleconstruct described), turning PDC off, and then doing the test.

    • If your looped back recording seems late with respect to the original, figure out the distance between them (in samples) and enter the inverse of that in the recording delay setting. So if you see that the looped back recording is late by 100 samples, enter -100 in the recording delay setting. But if for some reason the looped back recording is early (and it CAN happen), just enter the number as is.

    • Don't use the slider to enter the value for the recording delay -- enter numbers manually. It's much easier. You'll see.

    @ Sampleconstruct... you said that you had to set your recording delay to a positive value. Curious about something... Did you arrive at that value by physically patching outputs to inputs on your interface and doing the test? Or did you get that value by looping back "internally" via TotalMix?
  • Sampleconstruct Level 5 Level 5 (5,870 points)
    Whatever the problem I hope they can solve some of these "disk too Slow, or System overload" error messages. Do you think once I get the latency value offset's right I will not see the "disk too Slow, or System overload" anymore or less so?


    No, the latency offset is not related to the system overload messages at all.
    I would skip the software monitoring alltogether and get an Audio Interface that enables you to directly monitor your recordings through the Interface and not through Logic at all. In my setup I use my Tascam DM 24 Mixer for monitoring linked to the RME via a tosslink cable. To add reverb in the monitoring I have a dedicated Hardware reverb that is not linked to Logic at all, this gives me and the musicians I record the best results and the highest flexibility with no audible latency at all.
  • Sampleconstruct Level 5 Level 5 (5,870 points)
    Sampleconstruct... you said that you had to set your recording delay to a positive value. Curious about something... Did you arrive at that value by physically patching outputs to inputs on your interface and doing the test? Or did you get that value by looping back "internally" via TotalMix?

    I actually tried both, patching and routing via Total Mix yielding the same results, 54 samples negative offset. I thought it couldn't be true but it was, the RME seems to be a timewarp machine
  • Eriksimon Level 6 Level 6 (11,550 points)
    Sampleconstruct wrote:
    Sampleconstruct... you said that you had to set your recording delay to a positive value. Curious about something... Did you arrive at that value by physically patching outputs to inputs on your interface and doing the test? Or did you get that value by looping back "internally" via TotalMix?

    I actually tried both, patching and routing via Total Mix yielding the same results, 54 samples negative offset. I thought it couldn't be true but it was, the RME seems to be a timewarp machine


    So, you can actually hear what people are saying before they are saying it? (If they speak really fast) (flabbergasted emoticon needed)
  • Sampleconstruct Level 5 Level 5 (5,870 points)
    Yes, if I seriously mess aorund with some stuff I can actually record a sax solo before the musician has even arrived in my studio, this saves money, time and nerves!
    I can then put the saved time on my time account and trade it for more lifetime which will prolong my actual lifetime for x-times 54 samples = 1.125 milliseconds. So if I record for long enough I might actually live forever.
  • Eriksimon Level 6 Level 6 (11,550 points)
    So what's keeping you? Afraid to live forever?
    And: do need any extra plugins for it?

    Though it also shows that seriously messing with stuff can get you anywhere - even out of now! Wow!
  • iSchwartz Level 5 Level 5 (4,315 points)
    Sampleconstruct wrote:
    I actually tried both, patching and routing via Total Mix yielding the same results, 54 samples negative offset. I thought it couldn't be true but it was, the RME seems to be a timewarp machine


    Interesting result. Just for comparison, I've found that RME reports the latency to Logic and I can leave the recording delay set to 0. But when I use TotalMix to bounce audio back into Logic, I have to set it to +75 samples.
  • Sampleconstruct Level 5 Level 5 (5,870 points)
    Hmmmmmmmm.....
  • iSchwartz Level 5 Level 5 (4,315 points)
    Sampleconstruct wrote:
    Hmmmmmmmm.....


    Ah, you may "hmmmm" but it makes sense. (Actually, the fact that you're reporting the same latency via either method has me rather puzzled...) See pages 96 - 97 of the TotalMix manual (the version I have is fface800_e.pdf) and then let's discuss!
  • Sampleconstruct Level 5 Level 5 (5,870 points)
    I'll do another test in the next days and report back, maybe something was wrong in my Total Mix routing when doing the patched test because it really makes no sense at all....
  • AFS Level 1 Level 1 (0 points)
    Hi iShwartz, you found me!

    haha

    Thanks a lot for clarifying how to do the Loop test.

    Now I have to try it out and get back to you here.
    Are you saying to patch actual direct output (1 & 2) from my Digi 002 back into
    Direct inputs 1 & 2?

    or does it matter if you go master output 1 & 2 back into Direct inputs 1 & 2 on my Digi 002?


    One question though, you mentioned "Total mix" what is total mix vs outputs 1 & 2?
    Do you mean master fader output?

    Also, what is PDC to be sure please? is it Plug in Delay Compensation?
    you say have this turned off?

    for now, much thanks I'll respond after the tests.

    AFS
  • AFS Level 1 Level 1 (0 points)
    Hi Sampleconstruct,

    thank you again for your help!

    I monitor via SPIDIF out into my Masterlink and back into a Presonus "Central Station".
    So, my signal flow chain is as follows:

    Firewire out of MacBook Pro
    Into Digi 002

    SPDIF out of Digi 002 into SPDIF in of Masterlink

    SPIDIF Masterlink out into Presonus SPDIF in Central Station

    Central Station to 3 sets of monitor speakers, Genelec, Tannoy & KRK.

    So, this is essentially the same setup you have right?

    Peace and thanks!

    AFS
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