How to make latency compensation work? OR - suggestions for a small interface that DOES work?

[Guest]

I'm using a Roland UA-3FX interface with Audition 5.5 on windows 7.

If I load in a drum loop on one track and then record some bass on another track (monitoring direct though my interface), the recorded bass is slightly but noticably delayed from the drums.

As I understand it, the ASIO driver and Audition, between them, should be working out any latency and compensating for it.Any settings I can fiddle with? Can't find anything.

I did try using the ASIO4All driver which does have adjustable latency compensation in its settings, but this either wasn't enough or didn't do anything at all (not sure which!)

If it's an issue with the driver for the UA-3FX, can anyone else recommend a good small interface which can record in time?it's a shame 'cos the UA-3FX actually sounds great.

Ivan

[Guest]

I'm seeing similar results on my Win7x64 system as well

How is that system working with AA3?

[ryclark]

I have devised a possible work around to make the compensation more or less a one click operation. As you may have found out there is already a keyboard shortcut to Nudge a clip in a Multrack track backwards or forwards. Provided you have Snap to Frames enabled and no other snapping option you can Nudge using Alt< or Alt> to move the clip by one frame, whatever the zoom level.

For this to work for your latency compensation you have to calculate a Custom Frame Rate that reflects that latency and use that on the Timeline.  For instance, if your latency is 10ms then Custom Frame Rate will be 1000/10, ie. 100. Or in Samples, Frame Rate=Sample_Rate/No. of Samples. Thus you can shift your clip to the left or right by the correct amount. Unfortunately Custom Frame Rate won't accept decimal points so I hope you can calculate it near enough.

[Guest]

ryclark wrote:

I have devised a possible work around to make the compensation more or less a one click operation. As you may have found out there is already a keyboard shortcut to Nudge a clip in a Multrack track backwards or forwards. Provided you have Snap to Frames enabled and no other snapping option you can Nudge using Alt< or Alt> to move the clip by one frame, whatever the zoom level.

For this to work for your latency compensation you have to calculate a Custom Frame Rate that reflects that latency and use that on the Timeline.  For instance, if your latency is 10ms then Custom Frame Rate will be 1000/10, ie. 100. Or in Samples, Frame Rate=Sample_Rate/No. of Samples. Thus you can shift your clip to the left or right by the correct amount. Unfortunately Custom Frame Rate won't accept decimal points so I hope you can calculate it near enough.

That's pretty clever! I've never really worked in frames before so I wouldn't have thought of that.

Of course.not being able to use a bars/beats timeline is in itself a problem for musical applications.

[Guest]

So, any update?

[hawkenator24]

I have this same problem in cs6. Did they ever fix it? This issue makes the software useless IMO and it looks like I may be switching DAW's for good.

[Paul_Ferguson]

Two things:

* This is about laptops only, right?

* I haven't noticed anything like this using my desktop with CS6.

[hawkenator24]

No, not CPU specific. I have a quad core desktop with 8g of RAM. You don't have this problem?  What input method are you using?  I.e. USB interface etc, USB mic, line in, etc...  I have a USB mic and use ASIO4All drivers and have to manually realign vocals to match the timing of instrumentals for EVERY take. It's a royal pain. I'm going to upgrade to a USB interface to allow for zero latency monitoring as this is impossible with my current setup but the tracks will still have the same recorded latency so I'll be stuck with the same problem.

[ryclark]

You may find that if you get a USB audio interface with good ASIO drivers the problem may go away. ASIO4All only converts standard OS sound drivers to ASIO for Audition to use, so you will still suffer from the latency of the audio going via the OS.

[hawkenator24]

Thanks for the tip. That answers some questions of mine. I'm looking at the focusrite Scarlett 2i2 because I'm only going to really need 1 input for the foreseeable future. Do you use audition and, if so, have any interface recommendations. The focusrite website doesn't list audition as a compatible DAW, but that doesn't mean it's incompatible. I'm just a but weary of investing in any equipment that doesn't do what I need it to.

[Guest]

ryclark wrote:

You may find that if you get a USB audio interface with good ASIO drivers the problem may go away. ASIO4All only converts standard OS sound drivers to ASIO for Audition to use, so you will still suffer from the latency of the audio going via the OS.

It might make the problem less noticeable, but It wouldn't make it go away. In a way that's even worse - you're recording things out of time, but not noticing it!

[Paul_Ferguson]

hawkenator24 wrote:

No, not CPU specific. I have a quad core desktop with 8g of RAM. You don't have this problem?  What input method are you using? 

I didn't say I don't have this problem; I said I haven't noticed it.  I'm using Firewire 400 - Presonus Firestudio Mobile - but my next interface (which I'd like to get this year) will be a USB interface, so I'm interested in all this.

[hawkenator24]

Realdobro; sorry for the misinterpretation. I imagine I you haven't noticed it then it probably works on your system. Are you recording multiple tracks where timing is relevant. i.e. do you record vocals over music that's already in the multitrack? Also, what OS do you have on your PC? 

[Guest]

hawkenator24 wrote:

Realdobro; sorry for the misinterpretation. I imagine I you haven't noticed it then it probably works on your system. Are you recording multiple tracks where timing is relevant. i.e. do you record vocals over music that's already in the multitrack? Also, what OS do you have on your PC?

That's the thing - it was only really an issue when overdubbing timing sensitive material. You wouldn't necessarily notice a few ms delay on a spoken word overdub.

[Guest]

It would be nice to have a definitive statement from Adobe, though, about whether this bug has been fixed.

This was one of the reasons why I decided not to upgrade to 5.5/6 after doing the trial.

At the time my trial period ended, AA3 was giving near perfect asio sync, but later versions were not.

[Guest]

hawkenator24 wrote:

I have this same problem in cs6. Did they ever fix it? This issue makes the software useless IMO and it looks like I may be switching DAW's for good.

I believe a bug fix was in the offing around the time of my comment above.  IIRC it wasn't at the time possible to confirm when it would be ready as an update to 5.5, so I'm afraid I did have to jump ship because of this issue. I would be surprised if the fix wasn't in 6.

[hawkenator24]

I just purchased the Focusrite Scarlett 2i2 with a Rode NT1A mic and tested it on Audition cs6.  I tested the timing and latency and saw a MAJOR improvement.  I was actually able to monitor my speech without using the direct monitor function on the interface with no noticeable latency.  Yes, I know there is some latency but not enough for me to hear myself as an echo or for recorded vocals to be out of sync with an instrumental.  I was previously using a blue snowball usb mic and the latency was absolutely unusable.  There was NO way to monitor yourself while recording without a ridiculous echo.  Also, the timing was so off that I had to visually align every track after it was recorded which took FOREVER.  In summary, my current interface seems to have solved these problems.  I just got it and haven't had much time to use it but I will post an update with any new info. 

[Guest]

hawkenator24 wrote:

I just purchased the Focusrite Scarlett 2i2 with a Rode NT1A mic and tested it on Audition cs6.  I tested the timing and latency and saw a MAJOR improvement.  I was actually able to monitor my speech without using the direct monitor function on the interface with no noticeable latency.  Yes, I know there is some latency but not enough for me to hear myself as an echo or for recorded vocals to be out of sync with an instrumental.  I was previously using a blue snowball usb mic and the latency was absolutely unusable.  There was NO way to monitor yourself while recording without a ridiculous echo.  Also, the timing was so off that I had to visually align every track after it was recorded which took FOREVER.  In summary, my current interface seems to have solved these problems.  I just got it and haven't had much time to use it but I will post an update with any new info.

Note that this topic isn't about the amount of latency - it's about Audition's ability to compensate for the latency reported by an ASIO driver, however small or large. Assuming you're on PC, were you using ASIO drivers with the Snowball? If so, that suggests that either there was something wrong with the drivers or their setup, OR that there is still, in fact, a problem with the latency compensation in CS6.

It may be that you are still experiencing the problem with the Focusrite, just not so noticeably due to the lower latency. Maybe try the loopback test I and others have mentioned earlier in the thread?

[vnhllsng]

Uhm. Sorry for necroposting, but I think I might have a solution.

I was having exactly the same problem on Audition CS6, and probably dislocated my brain trying to solve it. I even read this whole thread in in hope to find a solution. So this is for those poor souls who come after me:

My problem was in mismatching sample rates of my session and my interface. That's it.

What I noticed is that I was trying to write in a session I made a while ago. Actually, it was a copy. I was recording several instruments, and decided to make one template session with a click track, guide tracks, markers for song structure and stuff. And then copy this template instead of configuring a new session all the time. And that template was in 44100 Hz. About six months later I decided to put a new bass track, but I've moved to a new audio interface which allows 88100 Hz. So I had my session copied from my template, with 44100 Hz sample rate and my interface set to 88100 Hz.

Then when I was trying to set some experiments, I made a new session, and recorded click using the simplest loop-back: i just put a mic to my monitor. And everything worked. That puzzled me even more. And then when I turned back to the problematic session thinking that what I've experienced earlier is just a glitch, and maybe rebooting my system helped. But the latency compensation didn't work again and only then I started to think how these two sessions could be different and remembered that the one I made earlier had a different sample rate. I just didn't think about it before, because I'm used to work in one sample rate all the time.

So yeah. Sorry if my response is too long and please forgive me my English.