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Forum suitability: sure, this concerns music production much more than music playing, so it would seem more appropriate here.
Generic kernel will make dropouts ("xruns") more likely even when using jackd with realtime priority. The jitter introduced in Midi is not that much of an issue since signals are time-stamped in the kernel and traditional 31250bps ...
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If you are talking about recording audio, latency is not an issue in 2016. Today, audio interfaces that are designed for multitracking have zero-latency monitoring. The audio input is split and you monitor an analog loop back, while the other side of the split goes into the recorder. You don’t monitor what is coming out of the recorder.
Latency is not ...
3
I've managed to get consistent results now!
~7ms with 128 samples, ~4ms with 64 samples @ 44100 Hz.
I had Audio Out connected into Audio Mic Input port.
Now it is connected from Audio Out into Audio Stereo Input.
(i'm not sure why that made the difference, or if the measurements are reliable)
Update:
Out of curiosity, to make sure that CEntrace tool is ...
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There are all kinds of specific technical answers that you might receive. However, the one general rule that I have found to be true with audio workstation setup is this: some system configurations will work, and some will not work, period.
You are putting together a house of cards made of computer hardware, computer software, audio subsystem architecture, ...
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After having a decent hardware system, your bottleneck of latency will be the signal vector size and the IO-buffers of your software. The smaller you can set them, the less latency you will experience and the more cpu stress will be the result. If your media is not able to deliver in time, you will also experience signal drop-outs, but that's why you want to ...
2
I'd say "All of the above"
SSD/RAID for storage [with appropriate backups on & off-site]
I put my most-used sample sets on SSD & also record to it - with backups, of course, as I don't actually trust an SSD to live out the day ;-) So far, so good, I haven't lost one yet, but I like to be sure.
Fast architecture/CPU, high core count. [background was ...
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Well, I can't really say I am a hardware expert, but so far I have never met anyone having issues with inter-channel latency. I have worked on localization projects and everything seemed to work like a charm, latency-wise.
There are some things that, in my opinion, point to the fact that inter-channel latency will not be a problem to your application.
Most ...
2
(In my experience) Scarlett products can have some issues automatically switching sample rates, so either Audacity was recording a 48kHz signal at 96kHz or the Scarlett was playing back a 96kHz signal at 48kHz (which makes more sense if it was really choppy).
You can try Focusrite's beta drivers if you still have issues, they've improved the automatic ...
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The simplest way to record 4 channels with near zero latency is with a 4 channel recorder. Either a 4 channel USB interface or a 4 channel recorder like the Tascam DR-70 or Zoom H6.
Two other approaches you might take are Arduino with GPS or what Zylia calls "external synchronization".
A standard GPS receiver supporting the 1 second pulse code will give ...
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latency depends on a number of things, but primarily it is determined by your audio hardware driver buffer size.
As you increase your buffer size, you do a couple of things:
1 reduce overall CPU usage
2 increase latency.
As you reduce your buffer size, you...
1 increase overall CPU load
2 reduce latency.
So there is always a tradeoff between CPU load ...
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Just give a chance to Ubuntu Studio. All music stuff including low latency kernel are ready to use in this distribution and you can still enjoy all advantages of debian-based distros. I use it for everything.
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It's not the USB MIDI that is where the delay lies, it's the generation and digital to analog conversion of the sound. The amount of data that has to be processed to output a half second of audio is vastly greater than the amount of data processing needed to change a part of the display to show a character from the code page or show the mouse pointer in a ...
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It's a trade-off, so it's not a case of setting it up once and leaving it.
In general, lower buffer setting = lower latency = higher CPU load!
When recording a live singer or musician, you want your latency to be as low as possible without there being any audible glitches. Usually I do this by trial and error, working up from the lowest possible buffer ...
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I own the UR22 mk2 and had experienced similar problem before updating drivers (win7 64bit).
If it's correctly plugged into a USB 2 port (usb 3 is buggy on audio interfaces) It can really be only a driver issue or incompatibility and the best course of action would be to directly contact Steinberg or use their suport forums.
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What you interpret as inconsistency is just the normal system behaviour. I would say your results are pretty consistent, considering the normal variance that can occur in a multitasking system running multiple applications. Even if you have a system running only your audio applications as end user applications, normal system functioning (memory management, ...
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Most likely it is just your audio output device that is inducing lag. On playback, you wouldn't notice the lag as everything would have the same lag. People are far more sensitive to latency when they are expecting something in reaction to their action than when they are simply hearing something.
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