Old 07-26-2017, 07:36 AM   #1
Warrior
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Default Question Regarding Latency

I record my guitars direct to DAW via my Fractal Audio Axe-FX II connected via USB.

I find that after running a session for i.e. about an hour, I'll start noticing some latency and need to select the audio source and everything is fine for a while.

This can occur as well if I'm not actually recording but just playing guitar through Reaper and out to my studio monitors.

Is this caused by my interface?
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Old 07-26-2017, 08:00 AM   #2
serr
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Trying to play detective...

So maybe the driver crashes and defaults to a larger block size when that happens? Then re-selecting the device restores it by initializing things (until it crashes again)?

Reaper isn't set to disconnect the device when not active?
Drivers are compatible with OS install?
Swapped the USB cable to rule that out?
Ruled out other audio apps making a change request to core audio?

Are you controlling the sample rate and block size from Reaper or a control panel app for the interface? Did you try swapping that?

If all that checks out but it's still doing it, can you document the change? Do a loopback test when it crashes so you are armed with hard numbers and then email tech support for the interface.
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Old 07-26-2017, 12:24 PM   #3
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Been going on for years. Just finally considering trying to resolve it.

I've changed USB cables.
I'll verifyyhe drivers are compatible.

I don't know how to check if other apps are interfering.

I'll experiment with the interface software. Never touch it.

Loop back test?

I've got lots to learn.
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Old 07-27-2017, 09:39 PM   #4
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Your signature is correct? You're using OSX 10.10.1?

From what I read Yosemite was buggy as hell until 10.10.3, which is what I started with. Maybe update to 10.10.5 in this case, which was damn stable ... I currently use 10.11.6, which is also pretty stable.

I use an old Tascam interface together with BIAS FX on a Macbook Pro (mid 2014) and can't complain, but only with Tascam's "low latency" driver. With Core Audio I have slight latencies too.
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Old 07-28-2017, 08:52 AM   #5
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Quote:
Originally Posted by Warrior View Post
I don't know how to check if other apps are interfering.
If you have something that can output audio running (a media player app or internet browser for example), look at the audio settings in the apps preferences or some other menu.

Quote:
Originally Posted by Warrior View Post
I'll experiment with the interface software. Never touch it.
Using Reaper to control connected devices is always the first choice. Reaper lets you defer control to other apps for sample rate and block size. So you can try that if the first choice isn't behaving. Uncheck the boxes for those controls in Preferences/Audio/Device to disable them from Reaper.

Quote:
Originally Posted by Warrior View Post
Loop back test?

I've got lots to learn.
Measuring the lag. When you first tried dialing the system in for low latency live work, you would have been going for < 11ms round trip latency with stable operation. You use the loopback test (plug an output into an input and record a transient - easy to zoom in and see peak - to another track) so you can measure the lag for a sanity check.

If you contact the interface manufacturer and complain about variable latency or the unit losing communication and defaulting to some higher value, the first thing they are going to ask is what the numbers are of course. What did you set your block size to? What latency did you achieve with that (and at what sample rate)? OK, what does the latency change to when this event happens?

You'll want to respond with hard numbers (in samples).

Or... if you go through the exercise you may actually uncover the issue yourself.
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Old 07-28-2017, 01:10 PM   #6
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Quote:
Originally Posted by Aymara View Post
Your signature is correct? You're using OSX 10.10.1?

From what I read Yosemite was buggy as hell until 10.10.3, which is what I started with. Maybe update to 10.10.5 in this case, which was damn stable ... I currently use 10.11.6, which is also pretty stable.

I use an old Tascam interface together with BIAS FX on a Macbook Pro (mid 2014) and can't complain, but only with Tascam's "low latency" driver. With Core Audio I have slight latencies too.

Oops... need to change that. I'm running 10.10.5
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Old 07-28-2017, 01:11 PM   #7
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Quote:
Originally Posted by serr View Post
If you have something that can output audio running (a media player app or internet browser for example), look at the audio settings in the apps preferences or some other menu.


Using Reaper to control connected devices is always the first choice. Reaper lets you defer control to other apps for sample rate and block size. So you can try that if the first choice isn't behaving. Uncheck the boxes for those controls in Preferences/Audio/Device to disable them from Reaper.


Measuring the lag. When you first tried dialing the system in for low latency live work, you would have been going for < 11ms round trip latency with stable operation. You use the loopback test (plug an output into an input and record a transient - easy to zoom in and see peak - to another track) so you can measure the lag for a sanity check.

If you contact the interface manufacturer and complain about variable latency or the unit losing communication and defaulting to some higher value, the first thing they are going to ask is what the numbers are of course. What did you set your block size to? What latency did you achieve with that (and at what sample rate)? OK, what does the latency change to when this event happens?

You'll want to respond with hard numbers (in samples).

Or... if you go through the exercise you may actually uncover the issue yourself.
I'll work on this. Never done this before.

I'm considering giving this interface to my son and purchasing a new one for myself. Really leaning towards an RME Babyface Pro.
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Old 07-28-2017, 02:14 PM   #8
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Of course you can just take Reaper out of the box, open preferences/audio/device, check the box next to block size, enter 128 samples and go. If you don't perceive a lag and there are no pops and crackles then done and done.

If you're doing live sound or performance stuff though you kind of want to get on top of the system this way. It's really just alignment stuff. The point is not to struggle with a perceptible lag in a live system (a show stopper) on the one hand or make your CPU scream with a lower block size than you need to achieve the latency you need on the other.

You can work with hard numbers and zero mystery.

Record a sample with a loopback test. There's the system lag right there (for that sample rate and that block size). It should obviously be the same value every time or there would be mayhem. So run it 3 times. If this is a moving target, we have hardware failure or something seriously crashing!

(NOTE: Don't forget to turn off the 'use driver reported latency' in preferences/audio/recording when running a loopback test!!!!!! This is the autopilot that nudges your audio back into place. You don't want to go in circles and have it move your test recording when you aim to use it to note the uncorrected system lag! Really easy to space this one out!)

This tells you two things.
1. If you were hearing no lag and then the lag measures < 11ms, that verifies you were hearing what you thought you were. (ie You didn't have lazy perception today and miss something glaring. Which you then hear tomorrow and now you're wondering if it's a moving target or just you.)

2. Now you have the precise value for your system lag in samples. Take a look at where 'use default driver value' places recorded audio. Is it right? You can add an offset to hit perfect alignment (or uncheck the 'use driver value' box and take over completely with your offset).

TIP:
Using a combination of 'driver reported value' + an offset can be convenient depending on the system. Whereas there would be a different offset for every sample rate, using the 'driver reported value' may normalize that difference between sample rates and it may turn out that you can keep a single offset correction for all sample rates.
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Old 08-02-2017, 05:22 AM   #9
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I'm attempting to conduct a loopback test but i'm not clear on some things.

(My ignorance is about to become very apparent)

I found a thread where "Fabian" provided a project set up for a loopback test. I need to connect the left output of my interface into the right input (Line level). my interface doesn't have left/right outputs/inputs. Just numbered.

http://www.fullcompass.com/common/fi...UsersGuide.pdf
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