This sounds like it's being recorded on Windows + USB, with not-fast-enough ports or processing/DSP + buffer size issues. Different generations of the Scarlet 212 have different USB connections, and without knowing which generation Scarlet, can't really say for certain.
He needs to:
- Make sure he's using a good quality USB cable that's capable of the speeds required for audio. Surprisingly, not all USB cables are made the same. I've even bought some that didn't have the power terminals connected to anything and wouldn't deliver power to my Arduino.
- Use the fastest ports on the computer. If there's USB-C, use that. If there's USB-A, make sure to use the USB 3.0 port, not the 2.0 Port.
- Make sure the Cable itself is capable of USB 3.0/Superspeed.
- Make sure the USB cable is of the right type. Each end should be the type he's plugging into; don't use adapters.
- If the computer only has USB 2.0 ports, and it's capable of adding peripherals, add a USB 3.0 card or it's time for a new computer. Otherwise, USB 2.0 will be giving him this headache no matter what he does. USB 2.0 is kinda terrible for audio. USB does data bursts (as opposed to other standards that use a more constant data flow.) USB 2.0 is too slow and there will be times between data bursts that are too long, not delivering audio information fast enough, leaving the gaps you're hearing in the recording. USB 3.0 does similar things but is so much faster it's a negligible. Even an old, cheap Mac might be better because even old standards of Thunderbolt or even Firewire don't have those bursting issues. (Whether or not it's compatible with his Scarlet might be another issue.)
- If the Scarlet is an older Generation that only outputs USB 2.0, get a new interface with a more modern USB standard. He'll be battling digital artifacts and buffer sizes forever.
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Fixing the Cable/Port situation might solve the problem. If not, keep going to the software side of things:
- Adjust the Scarlet's device settings to have a larger buffer. Start at 32 and go up. Make sure Ableton's buffer size matches the interface's buffer size. That shouldn't really matter, but, might as well until everything is done being troubleshot. Even if his computer's a little old, he should be able to attain 128 samples. I've been down that low for about 15 years, but i've been using mac. 512 should be the max before the latency gets atrocious. If it can't handle 512, time to start thinking about faster/more powerful computer.
- If it's an older/slower computer, set the Sample Rate to 44100Hz. Ideally you want to run at 48KHz but if the computer can't handle it, 44.1KHz is fine.
- Ableton provides a Test Tone that emulates running under different CPU loads. It's in the Preferences Audio Tab. Set the buffer sizes, and run the test tone with the CPU usage cranked up to 80%. If the audio has dropouts, bump up the buffer sizes of the Scarlet and Live until there's no more dropouts.
- Test it actually playing bass. On the one hand, he's gonna have to find a balance between big enough buffer size and latency. On the other hand, latency doesn't really matter if there are digital dropouts. If there are dropouts, it's unusable.
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And back to the hardware side:
- Don't record to a spinning disk hard drive. Record to SSD. If his hard drive is a disk type, and it doesn't have fast enough RPMs and read/write caching, this could be the chokepoint losing data.
- If he has a bunch of devices attached to his computer on a USB Hub, make sure the Interface gets its own dedicated USB port on the computer. Don't run the audio through a hub that's then distributing data between multiple devices and the single port the hub is plugged into.
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For reference, on my MacBook Pro with the M2 Pro chip recording to an external SSD, i'm running Ableton Live at 48KHz with 32 sample buffer. I'm playing bass live, real-time, along with other audio, and a bunch of VST3s and native FX running and my computer never really goes above 18% and there are no dropouts. With modern computers and the right cables, there's should be no reason for dropouts anymore. (Although i haven't used Windows for audio in years, and that's mainly because no matter how good everything is in Windows, something always doesn't work. Windows has come a long way and hardware makes up a lot of the difference from the bad-old-days, but i'm still suspicious of doing audio work on windows. -- *my personal experience + constant complaining from my Windows using music producer friends for the past 20 years or so.)
