FM3 recording latency offset

Yes, the one I'm using is very short. But I don't think that's the reason anyway, considering I'm not alone with the issue.
Even if the USB cable you are using is short, it can be defective.
If you have not tried other USB cables, try them.

There is also a buffer size parameter in the FM3 unit (Setup). Have you set it to the minimum?
 
Even if the USB cable you are using is short, it can be defective.
If you have not tried other USB cables, try them.

There is also a buffer size parameter in the FM3 unit (Setup). Have you set it to the minimum?
I'm so surprised.

First of all - it worked. I've lowered the buffer size setting to the lowest one, which is 48 (samples, I guess?), and my latency went down to 9 ms, which is within the range of other audio interfaces I've been using. I've only run my test and haven't tried to actually record any music, so I have no idea if it affects any real life application of FM3 in any way. In case this is the solution I've been looking for all along - thank you very much!

But secondly, it's either I'm too stupid to find it, or there is no mentioning of this parameter in the manual. I think I'm using the latest version, but it's just not there. Am I missing something?
 
I'm so surprised.

First of all - it worked. I've lowered the buffer size setting to the lowest one, which is 48 (samples, I guess?), and my latency went down to 9 ms, which is within the range of other audio interfaces I've been using. I've only run my test and haven't tried to actually record any music, so I have no idea if it affects any real life application of FM3 in any way. In case this is the solution I've been looking for all along - thank you very much!

But secondly, it's either I'm too stupid to find it, or there is no mentioning of this parameter in the manual. I think I'm using the latest version, but it's just not there. Am I missing something?
That's why I mentioned that parameter at the top of this thread :). It lowers latency, but increases the cpu usage on your FM3, so using the minimum value may not be practical. It was added in a recent firmware update.

It reduces latency, but does not solve the problem of misalignment. But hopefully you're now in a range where you can set the preference in Bitwig to compensate for the error.
 
That's why I mentioned that parameter at the top of this thread :). It lowers latency, but increases the cpu usage on your FM3, so using the minimum value may not be practical. It was added in a recent firmware update.

It reduces latency, but does not solve the problem of misalignment. But hopefully you're now in a range where you can set the preference in Bitwig to compensate for the error.
Yep, you did. I've assumed you are talking about the ASIO buffer size in my DAW's settings, and it was the only thing I've actually tried. Apologies, my bad.

Big thanks to you and jordikt, and everyone else for the contributions!
I hope we see the issue truly fixed at some point :)
 
I'm so surprised.

First of all - it worked. I've lowered the buffer size setting to the lowest one, which is 48 (samples, I guess?), and my latency went down to 9 ms, which is within the range of other audio interfaces I've been using. I've only run my test and haven't tried to actually record any music, so I have no idea if it affects any real life application of FM3 in any way. In case this is the solution I've been looking for all along - thank you very much!

But secondly, it's either I'm too stupid to find it, or there is no mentioning of this parameter in the manual. I think I'm using the latest version, but it's just not there. Am I missing something?

Happy to hear that you can manage a lower latency now!! :D
58 ms was a crazy value...:eek:

The buffer size parameter has been added in the last firmware 4.0 (Cygnus). It is explained in the release notes of the firmware, but it's not explained in the manual.

The manual is not updated frequently, so it can happen that you don't find something there. In that cases, it's better to take a look to the wiki or to search here in the forum.

Regarding the Buffer Size of the unit, the minimum is 48 samples. In the case you experiment clipping, you should change the value.

Here is the explanation of USB Buffer Size in the release notes:
Added: “USB Buffer Size” parameter. The USB buffer size determines the number of samples and thereby the latency between the FM3 and USB host. Lower settings result in better latency (recording and playback) but smaller sizes may not work well with all hosts. A good rule of thumb is to set the buffer to the smallest possible size, increasing if you encounter any USB audio performance issues. Find USB Buffer Size in the Digital I/O Configuration section of SETUP: I/O: Audio.
 
Yes, I know that. I have the same feature it in Cubase.

I was just curious to know how many samples or ms you are correcting your recording tracks.
I have it set to about 200 samples in Cubase.

The value you should set it to will depend on a number of factors that are particular to your rig, so there is no single correct answer. You'll need to do a loopback test to measure, then confirm, the correct value.

People often attach too much importance to latency when recording with an outboard modeler like an FM3 or AxeFx. If you're monitoring direct (and you should be) latency is almost completely irrelevant. All that matters is that the daw correctly compensates for it when aligning audio when you record.
 
I have it set to about 200 samples in Cubase.
Thanks for the info. I was curious to know that for comparing a "normal" correction (yours is around 4,16 ms) with the 58 ms of the OP.

People often attach too much importance to latency when recording with an outboard modeler like an FM3 or AxeFx. If you're monitoring direct (and you should be) latency is almost completely irrelevant. All that matters is that the daw correctly compensates for it when aligning audio when you record.
Fully agree. 👍
 
Maybe you're just not hearing it? How many samples do you measure for your misalignment?
Oh I can hear latency when its there, I cant play at all when its present. I am not sure about the samples but guessing 98?? Does it help me bcuz I set Logic buffer to 128 and use Input FM3 and output my Apollo?
 
I thought I remembered reading somewhere on here that using a dedicated interface and/or using SPDIF fixes this latency problem is that correct? I can’t seem to find what I thought I had read somewhere.
I’m fixing to be doing some projects and wanted to hit the ground running without too much fiddling if at all possible, this would be nice to know in advance to save time. I will be using a Focusrite 18i20 as the interface and connecting my FM3 to it via SPDIF, oh and Ableton is my DAW.
Am I going to be good to go or might I still encounter recorded audio latency that’ll need fixed?
 
I thought I remembered reading somewhere on here that using a dedicated interface and/or using SPDIF fixes this latency problem is that correct? I can’t seem to find what I thought I had read somewhere.
I’m fixing to be doing some projects and wanted to hit the ground running without too much fiddling if at all possible, this would be nice to know in advance to save time. I will be using a Focusrite 18i20 as the interface and connecting my FM3 to it via SPDIF, oh and Ableton is my DAW.
Am I going to be good to go or might I still encounter recorded audio latency that’ll need fixed?
the latency is related ONLY to using the FM3 as audio device in your DAW through usb.
Out 1/2 (and obviously spdif) cannot be affected
 
Back
Top Bottom