Jump to content

Record Delay is now in MS?


majool

Recommended Posts

Perhaps I'm late to the conversation and it has already been discussed here, but I couldn't find anything using 'search'.

 

How can it be that in Logic 8, the Record Delay is now set in ms instead of samples?!?!

 

Totally messed up...

 

Anyone have any thoughts, advice and/or workarounds?

 

Thanks!

Link to comment
Share on other sites

hi

 

while you may not realize that your recordings are late, chances are they are

unless you've made adjustments to your record delay setting which used to be set in samples.

 

there are many posts discussing record latency or record offset

(check here for instructions on how to fix...a bit dated, but still holds true: http://www.opuslocus.com/logic/record_offset.php)

 

in LP7, you'd calculate the latency in samples and compensate nearly perfectly. (since logic can't handle sub-sample level changes, it was hard to get it 100% phase cancellation).

 

now, with the 'ms' option it is much more of an approximation.

 

anyhow, nitpicking...but it can be a pain.

Link to comment
Share on other sites

I noticed this too when I tried to do the process today to figure out my recording latency. My latency seems to be 1 ms off (though I cannot get the sound file to cancel completely, it is most thin at one ms to the right of the anchor point). I guess I should set my delay to 1 ms
Link to comment
Share on other sites

I don't have L8 open right now, but if I'm not mistaken, this is a labeling error. It's actually "samples", but it says "milliseconds".

 

Check out this post:

 

http://www.logicprohelp.com/viewtopic.php?t=22161&postdays=0&postorder=asc&start=0

 

Guys, do the loopback test. At the end of Step #6, stop following the procedure and simply hit PLAY. If you hear nothing (perfect cancellation) or even near-perfect cancellation (a very thin sound), then your recording delay setting is already correct or very close to it.

 

But... if you hear a relatively loud, full-bodied signal, or discernable flamming or comb filtering, then you absolutely need to follow the entire procedure to set your recording delay setting. Otherwise, it's guaranteed that any audio you record will be out of time upon playback.

Link to comment
Share on other sites

hi fellas, thanks for the replies

as for the sample delay suggestion, it simply won't work as it delays audio by samples it doesn't place it earlier.

 

and ski, i don't think its a graphic error but i'll check again. i seemed to be getting all sorts of weirdness happening with my old setting.

 

i hope you're right though, but what a huge typo that would be!

 

also, as an aside...i was running some tests bouncing live audio into logic using the i/o plugin and ran into the strangest thing:

 

with latency fixer on the track set to 120samples, everything bounces accurately (regardless and independent of record delay setting).

 

without latency fixer on the channel, the amount my audio came in late was completely unrelated to and much greater than the 120 samples!

 

in fact, it was the same figure i came up with when testing realtime bounce of live audio via input monitoring and auxes (1144 @ 512 buffer size) plus 120 samples.

 

how in the world could that be unless logic was aware of my buffer size induced latency figure?

 

why would it compensate for it with latency fixer on the track to 120 samples?

 

i will test this again to make sure i wasn't drinking heavily at the time i ran the tests and report back. :)

Link to comment
Share on other sites

I meant like if you were sending a signal out into the analog world and through a compressor or some signal chain and back into the computer, that will introduce delay.

 

but do you need to have delay compensation if you are just recording directly into the computer from a guitar or something and not playing/effecting/re-recording a signal by sending it from ITB to OTB back to ITB..

 

or is delay introduced no matter what when recording from the analogue realm?

Link to comment
Share on other sites

no matter what.

if nothing else its 4 samples (or more) because of A/D conversion.

 

if you are sending it out it goes D/A (some delay) through the unit (depends on unit if it delays or not, analog compressor shouldnt delay), and back through the A/D (some delay).

then you have the delay that happens because of your I/O buffer setting :)

 

anyway, why does the manual say you should not need to touch the Delay Compensation settings?!

Link to comment
Share on other sites

anyway, why does the manual say you should not need to touch the Delay Compensation settings?!

 

I don't think anyone can explain why it says this. But it doesn't really matter much, because the manual is wrong. What's been discussed in this thread kinda proves that, don't you think? ;)

 

http://www.logicprohelp.com/viewtopic.php?t=22161&postdays=0&postorder=asc&start=0

Link to comment
Share on other sites

Join the conversation

You can post now and register later. If you have an account, sign in now to post with your account.
Note: Your post will require moderator approval before it will be visible.

Guest
Reply to this topic...

×   Pasted as rich text.   Restore formatting

  Only 75 emoji are allowed.

×   Your link has been automatically embedded.   Display as a link instead

×   Your previous content has been restored.   Clear editor

×   You cannot paste images directly. Upload or insert images from URL.

×
×
  • Create New...