USB vs SPDIF observations

gdgross

Experienced
Morning all -

On the ax8 we didn't have USB audio, so I got used to using SPDIF only for getting sound into my DAW, and that's how I've been operating for years. Now with the FM3, we have the option of USB audio, so I figured I'd give it a try this morning. I have a few observations:

1. There is definitely a noticeable latency with USB vs SPDIF. It's not terrible, but now that I know it's there I'm gonna stick w spdif for audio, even though it means an additional connector.
2. There was some occasional frying bacon kinda sound , which I assume was some kind of digital clipping, that lasted maybe just half a second or so. I tried clocking my interface with the SPDIF clock, and still getting audio from USB, and the noise persisted.
3. I notice that the USB audio also bypasses the volume knobs on the top of the unit (above the screen). Is there a way to do the same for SPDIF? I know we can select what audio goes to the SPDIF output, but bypassing these knobs would be easier for me to maintain a consistent recording level across different projects and over time (since I sometimes adjust the knobs per gig, and then forget about them later!)

happy Friday :cool:
 
Strictly speaking, you're not measuring a latency difference between USB and SPDIF. You're measuring a latency difference between two audio interfaces. How much of a difference did you measure?
 
Noticed the same in regards of USB vs SPDIF latency with the Axe FX III too...

I was under the impression that the Digital Output wasn't affected by the volume knob, at least it isn't on the Axe FX (I control my mix via Focusrite Control app), haven't had a chance to check that out on my FM3, you might be right.
 
Strictly speaking, you're not measuring a latency difference between USB and SPDIF. You're measuring a latency difference between two audio interfaces. How much of a difference did you measure?
Good point haha, i forgot that technically that's two interfaces, one on the way in and one on the way out! I didn't measure latency, but based on how it felt I'm gonna guess it was less than 50ms going in through USB? The latency in the SPDIF connection was not noticeable (to me at least) My interface itself plus the buffer setting I was using should have been <10ms.
 
50ms of latency makes it sound like the problem is your configuration. For example, the latency difference AJ Vargas measured was only about 1ms.
 
50ms of latency makes it sound like the problem is your configuration. For example, the latency difference AJ Vargas measured was only about 1ms.
Could be! I'm probably overestimating it too; I didn't actually measure it. But at this point, and esp given the strange bacon frying sound I hear, it's a bit of a don't care for me: I'll use SPDIF for audio and USB for connecting to FM3 edit. (although as it would be nice to bypass the volume knobs in SPDIF :) )
 
The clock drift (the frying sound) might be caused by the way you set up your multiple interfaces to be used simultaneously. Without knowing the details of what you did, it's hard to say. However, with all due respect, the observations 1&2 in your OP seem more likely to be due to something you did rather than something in the FM3 :).
 
The clock drift (the frying sound) might be caused by the way you set up your multiple interfaces to be used simultaneously. Without knowing the details of what you did, it's hard to say. However, with all due respect, the observations 1&2 in your OP seem more likely to be due to something you did rather than something in the FM3 :).
Right, that's why i tried clocking my interface from the FM3 via spdif, when i was using USB as an input, but that didn't seem to clean it up,
 
Right, that's why i tried clocking my interface from the FM3 via spdif, when i was using USB as an input, but that didn't seem to clean it up,
The problem is not the clock on either of the interfaces. The problem is the clocking of the USB stream on the computer.
 
Try adjusting your USB buffer size. You can check the buffer status in the Utilities menu while it's streaming audio. Should stay roughly in the middle. If it goes too high or too low, you can adjust the buffer size to keep it in the middle.
 
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