AFIII What buffer, bit, latency are you recording/mixing at?

V

vejichan

Inspired
I am using 48000 24bit, 128 tracking/recording and 1024 mixing. What about you?

getting 5-6ms when tracking at 128.
 
It depends on how many tracks and plug-ins I have in my DAW project. When tracking, or if I need to trigger a VST instrument from a keyboard or electronic drum kit, I'll mute or freeze tracks until I can use an 8 sample buffer. When mixing, I use the smallest buffer that works without glitches.
 
If I’m doing a full band session, I set the session at 48k 24 bit. While tracking, use the lowest sample rate you can without digital pops, some computers can handle 64 samples and some can’t go below 128. When you’re mixing, use the highest sample you have, which might be 1024 in your case.

If you try tracking at 64 samples and get pops, try deactivating (not bypassing, as that still utilizes CPU) unnecessary plugins. Reverbs and Delays use a lot of CPU, so deactivate those first. Once you’ve tracked your song, set your sample back up to 1024, reactivate your plugins, and enjoy mixing. :)
 
York Audio said:
If I’m doing a full band session, I set the session at 48k 24 bit. While tracking, use the lowest sample rate you can without digital pops, some computers can handle 64 samples and some can’t go below 128. When you’re mixing, use the highest sample you have, which might be 1024 in your case.

If you try tracking at 64 samples and get pops, try deactivating (not bypassing, as that still utilizes CPU) unnecessary plugins. Reverbs and Delays use a lot of CPU, so deactivate those first. Once you’ve tracked your song, set your sample back up to 1024, reactivate your plugins, and enjoy mixing. :)
Click to expand...

fyi, when recording audio with direct monitoring, the buffer size is almost irrelevant. Your DAW needs to compensate for the latency no matter what your buffer size is, and align the audio to ensure proper playback. So, there's usually no point to turning down your buffer size during audio recording when monitoring direct.

When recording virtual instruments, it's a whole other matter. That's when you need low latency. The same goes for monitoring through your DAW when recording audio (but that's an unusual case). Most DAWs have a "low latency mode" that you can enable in these cases to automatically disable all plugins which introduce latency. And you may have to resort to adjusting your audio buffer size to get the desired latency without overloading your cpu.

When mixing, almost all DAWs use hybrid buffers these days to maximize cpu headroom, so there's usually no point to turning up your buffer size during mixing.
 
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