Reply To: Audient iD4 MKII

Thanks a lot for responding and explaining. I like to learn something new, so I can help in similar situations. And it would be helpful for others reading this which is the basic idea of this forum.

The latency measurement shows a rather large number, but since I don’t know what this test measures, it might be OK.
If you can adjust the sample buffer size (audio buffer) to a value of 64 or 128 and do the test again you might get smaller numbers … or not.

If I measure interface latency, I loop the output of the interface back to the input and let a program fire an impulse out and let the program measure the time until the input delivers the impulse back to the software. So no network traffic involved there.
I use a software named “pure data” (pd) together with a patch (a little program written in pure data) that can measure the latency (precision is not that high, but gives a hint, depends on the settings of the software). It’s a programming software for musicians (mostly used by classical electronic composers, but also improvising musicians) that I have used some years ago and someone else wrote that latency-test that I use. I think pd is also available for the raspberry pi, but if you are not used to it, setting the audio interface connection values might be a bit hidden.

I have measure several of my audio interfaces on my Mac and Windows computer (both older machines). A value of 24 to 36 is typical for an audio buffer size of 256. For an audio buffer size of 128 I get values of 10 (12) to 23 ms and for an audio buffer size of 64 values between 4 (8) and 18 ms. But an audio buffer size of 64 puts a high processor load on my old machines.

So you might check if you can set the audio buffer size in the latency test program and check again with 128 and 64. BTW, there is a list of tested interfaces and their latencies on the FMB website. It is a bit hidden. Go to the store, then to the FMB software. There you find a link to build instructions for your own FMB box and there you find the link to the tested interfaces. They are all run with an audio buffer size of 64, so don’t wonder about the low values. Check yours with an audio buffer size of 64 and only then you can compare the latencies.

At the end the effective sound latency collaborating with others is the most important. I always recommend 64/256/1 (sample buffer/network buffer/jitter buffers) or 64/256/2 if the first does not work well. Sometimes 64/128/1 or 64/128/2 works, that would be faster, but often results in a lot of audio artifacts.