Hi,
I know this is probably the oldest question in the book, but I still can't seem to find a conclusive answer to it.
So, in the last session where I was recording vocals, there was a significant delay when the vocalist sang her lines resulting in a horrible sort of echo effect. We managed to get round it by her basically not hearing herself in the headphones and just relying on performance, but it was far from ideal.
Just to give some details of my set-up:
Audient ID14
Mac mini 2012- 2.3 GHz Intel Core i7, 16 GB ram
Recording onto an external drive.
The tracks are admittedly quite developed and running quite a few plug-ins, which presumably is slowing things down. Generally about 25-30 tracks, with compressor vst's etc. It's a pretty powerful Mac though, so I would have thought it could handle such things.
I tried turning the buffer size down and this did help somewhat, but didn't entirely resolve the issue.
Any suggestions?
I have another session on Monday and would like to have this sorted before we start if possible.
Many thanks,
Dom
Comments
There's one, which is number of samples in the buffer (32,64,128,256 etc...) where obviously the higher number = longer delay.
The other one is fast, normal, safe, ultra safe etc and this is to do with how often the ID14 interrupts the computer's processor to send and receive packets of data; the audio going in and out isn't a constant stream, it's sent in bursts and this setting determines how often those bursts are sent. With slower settings, you'll find that some of the lower buffer settings get greyed out because the gap between the packets of data being sent would be longer than the size of the audio buffer.
So for the lowest latence settings, you want to basically set both those as low as possible. If you get crackling and dropouts, start increasing them. The lowest buffer size you can go to will purely be a factor of whether or not your computer can keep up with the realtime processing. The lowest latency setting you can get will be affected by how good your USB busses are - if there's another USB device trying to interrupt the same USB controller in your computer, they might clash and that'll affect how low that setting can reliably go.
Then you've got latency introduced by effects. Some plugins are very low or almost no latency, and some have huge amounts of latency and hence just aren't suitable for realtime monitoring.
The Audient does have zero latency monitoring, controlled by the same mixer you set the latency settings with. Obviously you can't add effects to that monitoring, but if you want to add reverb or echo or something you can set it up so that you monitor the direct signal with zero latency, then also monitor a reverb effect 100% wet where a few tens of millisecond latency just don't matter - it's basically free pre-delay.
Worth pointing out that the Audient drivers are famous for not being great at low latency in contrast to, say, RME that's bloody amazing at it, though the Audient is a great piece of hardware and Audient have just released new drivers which may or may not improve things depending on your system.
And I'd be remiss if I didn't close by suggesting not using headphones when recording vocals, unless there's an overwhelming practical reason for doing so. It's just more fun, and there are no monitoring issues to contend with at all!
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A few things that I'm not clear on though:
1) Can you set the buffer size on the audient, or is it just through Reaper? I can't see any settings for buffer size on the ID software.
2) Do you know if there's any way of muting plugins on a mix in Reaper to see if these might potentially cause problems?
3) tHere's a setting in Reaper under 'Preferences - Recording' that says 'Use audio driver reported latency. You can change the output and input manual settings here. Is that of any use?
4) Not sure what you mean by recording vocals without headphones? Sounds great in theory, but how on earth would you stop bleed from the track being recorded through the mic?
Thanks again,
Dom
2.) It depends how latency compensation works in reaper - probably hard-bypassing the plugin using reaper's own bypass option, rather than any bypass button build into the plugin's GUI, will also stop it introducing its latency. You might need to stop and play to get Reaper to re-calculate all its latency compensations.
3.) That's only for if the driver is reporting the *incorrect* latency - then reaper will think the latency is different to what it actually is, and might place recordings at slightly ahead or behind the beat because it thinks you were hearing the playback a few milliseconds away from where you actually were.
4.) You minimise it using sensible microphone placement, using the mic polar pattern to your advantage, and playing back the track just loud enough for the vocalist to feel comfortable relative to how loudly they sing. What bleed there is is *never* an issue, unless you've done something daft like blast out bass notes that are subsequently changed or loads of tambourine or something. You might be surprised how many successful records were made this way.
Bandcamp
Spotify, Apple et al
Bandcamp
Spotify, Apple et al
Bandcamp
Spotify, Apple et al
The amount of latency depends on the AI drivers and how fast your computer is - faster computer = lower latency. Also the more plug-ins your computer has to deal with, the higher the latency, so it's a good idea to do your tracking first then mix later. If your DAW has a track freeze facility, you can use this to reduce the processing power used, and thus the latency.
I would not recommend recording at 96K, this doubles your file size thus resulting in a halving of the number of tracks your system can handle, plus any plugins will use twice the processing power. Judging from the buffer sizes you describe I would guess you are not using the most powerful computer around. Another thing to check (assuming you are on Windows) is that you are using the ASIO driver, selected from within Reaper. With a Windows machine, it can often be useful to optimise your computer for audio recording - excellently described by Robin Vincent in his Molten Music channel on the tube.
In my experience, USB interfaces all have fairly high latency. RME interfaces have a reputation for lower latency, but do not come cheap.
I would have thought that would be more than enough for 30 odd tracks, but it has begun to struggle when loading up and the number of plug-ins seems to be taking its toll. This could be a problem when doing final mixes with everything running. Is the idea to run at a higher buffer when mixing as latency isn't really a big issue then?
Honestly, I've pretty much always just stuck with 1024 buffer size unless I've had a reason to lower it, whether I'm recording or mixing. But if low latency is important to you while recording, yes increase the buffer size to mix.
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