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Latency. Is it using software, does hardware fix?


Thunderpaws
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Hiya. 
 

I’m using Torpedo Wall of Sound in Cubase LE 10. Usually when I’m recording Bass (amp is di in to my Audient interface) I get latency that means I’m struggling to record in time. 
 

would buying an ir loader pedal before my interface help reduce latency and solve all of my worldly problems?

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35 minutes ago, Thunderpaws said:

Hiya. 
 

I’m using Torpedo Wall of Sound in Cubase LE 10. Usually when I’m recording Bass (amp is di in to my Audient interface) I get latency that means I’m struggling to record in time. 
 

would buying an ir loader pedal before my interface help reduce latency and solve all of my worldly problems?

 

 You will probably need to check and experiment with the buffer sizes in your Audio Interface settings before you do anything else. It's probably set quite high for recording. High is okay for mixing:

 

"When recording, use the lowest buffer size as possible. When mixing, change it to a higher setting like 1024 or 2048.

The standard buffer sizes are 32, 64, 128, 256, 512, 1024, 2048, 4096 and 8192.

 

Higher sample rates = lower latency. 128 samples at 44.1 kHz is slower than 128 samples at 96 kHz, for example".

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IME the best hardware fix for recording latency when using a DAW is to use a small multi-bus mixer. You only need 2 more channels on it then the number of sources you intend to record in a single pass.

 

Run each source you are recording through it's own channel and output on a separate bus going to your interface, at the same time also routing them to the main stereo bus on the mixer. Then have two channels (not routed to any bus other than main stereo mixer outputs) taking the main stereo output from your DAW. Then turn off software monitoring, so that only the channels being played back by the DAW are going to the stereo output (and not any that you are recording). You monitor from the mixer. You here a stereo mix of the tracks already recorded on the DAW plus the instruments you are recording just through the mixer. As the new sources you are recording don't have to do the round trip from interface to DAW and back to the interface there is no latency at all.

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I own an Audient. It's wonderful. Like @lowdown says you need to set the buffer size to as small a number as you can get away with. Too low however and you will get skipping as your computer's CPU runs out of puff. It'll run out sooner the more plugins you use and cab simulators can be quite high cpu usage. To get you some idea of numbers, at a sample rate of 48k I have a block size of 128. I don't even notice the lag (which according to my DAW should be ~5ms) and I can run all my plugins except an octaver on my 2019 laptop.

 

Specific to the Audient, you can set this either in your DAW (probably), or the iD program that manages the various levels of the different inputs

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