Record Latency Adjustment (samples) setting in X2

I understand this setting adjusts newly overdubbed tracks back a certain amount in ms to keep them in time with previously recorded tracks (due to audio interface latency) But what happens with wav files that I import into Sonar if this setting is on? Let's assume these wav files I import into Sonar are perfectly in time and line up perfectly on the timeline. If I have the 'Record Latency Adjustment' setting on, does it automatically move these tracks backwards by that amount in the setting (and thus now make them offtime)?

When running ASIO drivers just let Sonar mange the offset.
You don't need to worry about it.
DON'T turn it "OFF".

T

Thanks guys. I understand I shouldn't turn this off. But I'm trying to understand how this setting will affect wav files imported into Sonar (not actually recorded into Sonar). I'm very picky about timing and want to make sure things always line up correctly.

I understand it will import it where I want it to. But how does the record latency setting affect those imported files after importing? After all it's a wav file, like all wav files created after recording. Or does the setting only affect wav files that were recorded using Sonar?

Let me sum this up for you.

Please get it out of your head that you need to deal with using this parameter.  It is NOT needed for what you are doing.

Every single one of us needs our tracks lined up properly, and guess what - they BUILT Sonar to handle this automatically, by using the automatic setting you see there.  Sonar queries your interface to ask about latency, and your interface tells Sonar whatever that value is, and then Sonar factors that (properly, I might add) to the recording process, which is the only place this parameter has any meaning anyways.

MULTIPLE posters here have told you to not make manual adjustments to this, and yet you persist down the path of trying to use the parameter.

You asked for advice, then a bunch of folks all independently told you the same thing, and yet you don't accept it?  Really?

It does not matter if it is a wav file, an MP3 file, or any other kind of file, nor whether or not you recorded it or imported it.  It just WORKS.

Bob Bone

WAIT!
This is NOT what you asked about originally!
If I understand you correctly each "take" is being offset more and more?

I'll have to test and get back.

You need to put some system specs in your signature or at least post your specs.

T

Okay, now this sounds like the exact same problem I'm having with X1.
DZarob - Take a look at my post, pull up those same screens and let us know what you're seeing for settings (ASIO Reported Latency and then from the other screen Input, Output and Total Roundtrip Latency).  

Unfortunately, I haven't figured out how to fix this problem (thinking about re-installing X1 in Safe Mode).

http://forum.cakewalk.com/fb.ashx?m=2744100


{edited to be more specific about the parameters}

Tom and All - Thanks for your patience on this.  What DZarob and I are experiencing is an anomaly.   (Although I can't claim mastery, I've worked with Cakewalk software since the Home Studio 2 days and have never had this problem.)

Tom - Thanks for your excellent suggestion on how to show the problem.  I set up a drum track with a snare hit at every measure, left the metronome on, and then patched a loop-around on the sound card to feed that back into an audio channel.

For this first demo, I left the "Use ASIO Reported Latency" checked (so 'on'):


Here's the result.  You can see the audio drifting later and later with each take.  Also, a short "tail" track is stuck onto the end (I stopped recording before starting another loop).



I ran the same test with a second track and Use Reported ASIO Latency unchecked and then the audio lines up fine.

(So in this shot "Latency On" track = "Use Reported ASIO Latency" is checked; "Latency Off" = unchecked.)

I admit this seems counter-intuitive, but there it is.


Lacking a better idea, my next move is to re-install X1, but since I have a work-around and am in the middle of a project I'll wait until later.


I'm still open to suggestions, but meanwhile, DZarob, at least you know you're not alone with this issue.

j,
I just looked at your referenced thread.
Try setting the TonePort, Playback and Record to the "same" channel.
You've got some kind of external hardware compensation going on me thinks.
You've got Playback set to channel 1-2 and Record set to the Send.
Set those the same and report back...

T

j,
Are those your only choices because Windows has grabbed something for itself?
You maybe should go into Control Panel>Sounds and make sure the greedy b****h,
excuse me, Windows hasn't misappropriated something.

Worse case scenario, uncheck Use ASIO reported latency.
It looks "close" on your screen shots, is it "spot on"?
You can always use "Manual Offset", the problem there, if you change latency your "offset" is wrong.

T

Just so you see it for yourselves and know what I was talking about above, here's a little video to show you exactly what I meant.

https://dl.dropbox.com/u/4909348/SonarCardSync.rar

Hope this helps. :)

-Danny

You're quite welcome Jeffrey. I hope it helps you out. That whole drifting thing can really be annoying and when you've been a drummer first as your weapon of choice and have paid close attention to your meter, this can really drive you crazy. I've honestly never considered myself anything special as a musician and I mean that...but man, one thing I got is timing. Many times I found myself beating myself senseless with Sonar while talking to myself (that's just only child syndrome lol goes with the territory) really wondering if it was me.

I'd sit there with my guitar just strumming in time to my click to see if the strums were landing right on the beat...and I was positive they were, yet the grid showed me different. That's when I knew something wasn't quite right. When doing this with other DAWs, it showed me that I was actually as close to spot on as a human could be with the majority of my strums being dead on while the ones that weren't were so close to on, they weren't worth fixing as no one can really be a metronome. But the results I was seeing in Sonar were scaring me.

I bought a new soundcard one day and it helped leaps and bounds of course, but there were still a few things that just never made sense to me. I either left them alone or tried to compensate if something bothered me by purposely playing something a little early. Or sometimes, and I know some of us have HAD to experience this....but I'd record something in Sonar and that particular take would be WAY off. I'm talking like so off you had no choice but to redo the whole track. This anomaly would happen to me about every 100 times I'd press record and was in every version of Sonar I have ever used. This would happen if you pressed play and then pressed record while playback was already rolling. Certain times when doing this, it would be so out of time it was like I was recording from another country through a 56k modem. LOL!

But upon learning of this clock sync thing from Eric, none of that happens to me anymore and all my tracks are now spot on when I record them. The only thing that sucks about this method is I must use a few different clock sync numbers per situation. I'll explain as it may be helpful/useful t others that may be in my situation.

I use consoles here for all my stuff. So, this means that input monitoring is not needed on my behalf since everything I do plays in real time into my mixer and my soundcard input monitoring is disabled as it's just not needed. This in turn allows me to not have to touch my ASIO buffers and get down super low like everyone else has to when they record. I can leave my buffers set to 2048 or 4096 while recording. The problem here is, if you do the sync test at THOSE buffer settings, you get a different "nudge number".

So I have sync numbers for 32, 64, 128, 2048 and on one of my older cards, 4096. Sometimes due to the project we may be working on, 32 or 64 buffers can't be used on my end if I've already loaded up a bunch of things that are cpu intensive and I find out I need to re-record something that forces me to lower my ASIO buffers. I normally don't have to do this for audio, but if I'm recording using a guitar sim, I can't leave my buffers at 2048 or the latency is too insane as you know. So for that, I do a buffer change and may end up at 128 for a loaded up project. If the project isn't loaded up with stuff, I'll use 32 or 64. Each one of those buffer settings will give you new nudge numbers, so I have them all written down and just change when I need to. It's rare for me to use anything other than 64 or 2048 though, but I have the others covered should I need them.

So if you (or anyone else) finds themselves in this situation, it's not a bad idea to do a few sync procedures using different buffer sizes if you ever need them. I hope this stuff helps you out...make sure you post a message back and let us know if this helped you or not. Good luck with the Sonar re-install too! :)

-Danny

First, Dennis, sorry I hi-jacked your thread.
I did a full uninstall and re-install of X1.
Everything looked great since I was checking the Reported ASIO Latency after the initial installation, then after each of the patch files.  

I was looking forward to coming in here and reporting success when I noticed that  the Reported ASIO Latency had jumped back to being the (incorrect) full roundtrip latency that was causing delays when recording including a cumulative delay when recording loops.

Sonar 8.5 still shows the correct value, so I compared config files but no luck (they match exactly).

May be time to try tech support.....

Guys sorry for the late reply, been off the forums for awhile. Jhughs this is the EXACT problem I was having. Never came up with a solution. Went to Cubase for now until Sonar gets their bugs out. Seems to be a bug to me.. Unless you came up with some solution?

Here's the response I got from Tech Support.  Just shutting off "Use ASIO Reported Latency" pretty much solved it, but I never had to do that before (going all the way back Home Studio XL).

"Please make sure to shut off your "Use ASIO Reported Latency" is turned off.
After this make sure that your Playback I/O Buffer Size(KB) is set to 2048(Maxed Out) and go to Edit|Preferences(advanced tab)|MIDI|Playback and Recording and make sure that under "Prepare Using ____ Millisecond Buffers" is set to 1000."