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Re:Lessons learnt building a real-time audio application in ... by camtarn on Monday September 09, 2024 @12:38PM (#41490309)
Yeah, your Scarlett should be capable of single-digit ms latency. If you're on Windows, you need to install its ASIO drivers and figure out how to use them from Python. Then, yes, use tiny buffers and run your audio processing very fast - which is where Python's slowness will probably become a real problem.

10ms latency is how long sound takes to travel 3-and-a-bit metres. So if your amp is a few metres from you, you would experience that delay between hitting the guitar strings and hearing the amplified sound. This should barely be noticeable. If you were noticing a delay greater than that in your Ableton effects setup, your settings needed tweaked. All of this is completely possible - I had a PC-based electronic drum setup in 2006, running through the Reason DAW, which had 8ms latency between hitting a pad and hearing the result.

Hmm, I wonder if Cython (static Python-to-C compiler) would make writing audio code easier/more possible?

Re:Lessons learnt building a real-time audio application in ... by spmvg on Monday September 09, 2024 @01:05PM (#41490696)
With Ableton and the default ASIO configuration on my Scarlett I get 96 ms combined input+output latency without any processing in between, so that's probably what made my ears bleed before. Tweaking the sample rate and buffer size gets me indeed single digit latencies in Ableton. So I'm definitely going to adjust the section about latency, thanks for this!

I'm a bit on the fence about what this means for the difficult latency calibration routine in the application. Ideally I could throw the calibration routine away, but then I require that users have ASIO installed, while the app now also works with non-ASIO drivers. And indeed Python itself might become a bottleneck (making this work in Python has been half the fun).

Re:Lessons learnt building a real-time audio application in ... by sim7c00 on Monday September 09, 2024 @01:35PM (#41491107)
try clarett interface. it also comes with pre amps which will make your sound less noisy , scarlet preamps are just absolutely terrible. you can debug your daw to see how it uses drivers and make a python module which exposes similar functions to python. you will likely still want a delay compensation to make things seem free of any latency, but it will be doing _much_ less compensating. maybe theres an opensource daw if you want to skip reversing driver calls from a debugger.
Re:Lessons learnt building a real-time audio application in ... by spmvg on Monday September 09, 2024 @01:53PM (#41491327)
Debugging an existing DAW to see how they do it under the hood is an interesting idea. Haven't done that yet.

About another interface: I do want to keep the application supporting cheaper interfaces such as the Scarlett, because the target audience (hobby musicians) will be using those. Still would be a nice upgrade for me!

Re:Lessons learnt building a real-time audio application in ... by kibibu on Monday September 09, 2024 @06:35PM (#41494919)
Can take a peek at how Tracktion engine does it too
Re:Lessons learnt building a real-time audio application in ... by spmvg on Tuesday September 10, 2024 @10:41AM (#41501279)
Tracktion looks like an interesting project, thanks for pointing me to this!
Re:Lessons learnt building a real-time audio application in ... by ubercore on Monday September 09, 2024 @02:41PM (#41491916)
I don't know windows audio, but on mac audio that's wildly high latency for a scarlett interface.
Re:Lessons learnt building a real-time audio application in ... by dist-epoch on Monday September 09, 2024 @01:43PM (#41491204)
Even without ASIO you should be able to hit 40 ms latency on pretty much any Windows audio hardware, including motherboard built-in.

If you get 300 ms you're doing something wrong. Note that Windows has multiple audio APIs, 300 ms is about the latency of the old MME api, you need to use the newer one, WASAPI.

Re:Lessons learnt building a real-time audio application in ... by spmvg on Monday September 09, 2024 @01:56PM (#41491380)
I apparently only have the old Windows MME drivers indeed (and ASIO, on Win10). Need to look into why I can't find WASAPI and if I can assume other Windows users have those by default.
Re:Lessons learnt building a real-time audio application in ... by michaelrmmiller on Monday September 09, 2024 @03:12PM (#41492387)
WASAPI has been available since Windows Vista. It isn’t its own set of drivers but rather a unifying layer for the WDM driver and the preceding mishmash of Windows audio APIs (MME, DirectAudio, etc). WASAPI supports low ish latencies with Exclusive Mode and then something like 10ms buffering in Shared Mode through the Windows audio server, I recall.

Put another way: any Windows audio device supports WASAPI unless it only ships with an ASIO driver which is unlikely, even in the pro audio space.

Re:Lessons learnt building a real-time audio application in ... by bongodongobob on Monday September 09, 2024 @01:38PM (#41491144)
I would disable any services and programs running in the background as well. Years ago I disabled the Windows print spooler and it greatly improved jitter. Not sure if that's still the case these days though, that was probably 10 years ago.
Re:Lessons learnt building a real-time audio application in ... by spmvg on Monday September 09, 2024 @02:02PM (#41491442)
So far CPU usage hasn't been an issue at all (<1% usually on my not-very-impressive laptop), which surprised me as well