In our recent discussion about audio interfaces and their latencies as influenced by buffer size I remember @miked wondering why 128 samples was still the default size.
I have been experimenting with smaller buffer sizes both at 48KHz and 96KHz and it turns out the 128 samples seems to be the minimum buffer size required for acceptable audio quality. There seems, nevertheless, a tendency that smaller buffer sizes may kind of work for locations nearer to the server location where the virtual studio is running.
For the distances I am working with (600 up to 1000km) anything less than 128 samples produces sound artefacts. Reaching out from France over the Atlantic to NYC 128 samples works OK without being perfect.
Thank you for sharing your experience, @wollethom!
A few of the biggest factors that seem to impact this are:
Audio interfaces: Many audio interfaces claim (to your operating system and apps like JackTrip) to support smaller buffer sizes, but fail when you try to use them due to using slower chips. This is especially a problem when using generic drivers (Mac or Linux) because they don’t actually know anything about the device’s true capabilities.
Computer performance: smaller buffers sizes demand more CPU power from your computer, so slower computers are unlikely to handle them well. This can also be a problem if you have a lot of other applications open at the same time, or are using a laptop on battery power.
Internet router: smaller buffer sizes require more packets per second to be sent through your Internet router. I’ve found many routers out there than can handle lots of bandwidth, but not very many packets per second (also a CPU bottleneck).
I have to admit that one nice thing of the Analog bridge devices is that they check the first two boxes for you and perform great at a buffer size of 64…
It’s a frustrating state of affairs for the industry because JackTrip is stellar with low buffer sizes (I know of people even using 16 successfully), and it does seem to yield big latency reductions. But the rest of the technology world still needs to catch up.
@wollethom one thing I would love to know is if changing buffer size to 64 in your studio’s advanced audio settings, but leaving your desktop app at 128 works ok for you? This is a default setting change I’ve been thinking about for a while, which seems safer (we size studios appropriately) but does still ask a little more of routers and computers.
Interesting topic.
This implies it would be worthwhile to review audio interfaces with regard to whether or not they have specific drivers and how these perform.
Also for computer performance, it would be good to know how much CPU power we need for optimal performance with different buffer sizes.
The same applies to internet routers. What is the available knowledge base for deciding whether it is worthwhile to look for another router ?
Of course, all these question are only as interesting as they help musicians to play in sync. My current experience with people mostly beyond the recommended distances from the studio server still leaves room for improvement.
I tried tried 64 for the studio with 128 for the app as well but I do not get a clean and stable sound regardless whether it is on windows or Mac OS.
My PC has these specs :
and the other machine is a MacBook pro of 2024 with a M3 chip.
Thanks @wollethom. Your computers are definitely fast enough, so it seems like my concerns about router/connection performance are warranted and sticking with 128 is best for now.
