Not sure I understand Buffer Size. Can anyone explain?

Not sure I understand Buffer Size. Can anyone explain?

Posted on Apr 7, 2021 11:06 AM

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Posted on Apr 7, 2021 11:20 AM

Hi there,

Buffee Size has to do with your computer and how it processes the information in your track in real time. The easiest way to remember it is...


"Low When Recording, High When Mixing/Listening"


When recording Vocals or Guitar, you want to settle buffer size low (64/128) so there is no latency/delay on the signal. It is almost impossible to record when there is even a modicum of delay.


When you set the buffer size to HIGH (512/1024), it is easier for the computer to handle processing but there might be a slight delay if you try to record.


Be sure to use this in conjunction with LOW LATENCY MODE.


"Low When Recording, High When Mixing/Listening"


Hope this helps and solves your query,

Eddie G.

9 replies
Question marked as Top-ranking reply

Apr 7, 2021 11:20 AM in response to soulmuse

Hi there,

Buffee Size has to do with your computer and how it processes the information in your track in real time. The easiest way to remember it is...


"Low When Recording, High When Mixing/Listening"


When recording Vocals or Guitar, you want to settle buffer size low (64/128) so there is no latency/delay on the signal. It is almost impossible to record when there is even a modicum of delay.


When you set the buffer size to HIGH (512/1024), it is easier for the computer to handle processing but there might be a slight delay if you try to record.


Be sure to use this in conjunction with LOW LATENCY MODE.


"Low When Recording, High When Mixing/Listening"


Hope this helps and solves your query,

Eddie G.

Apr 7, 2021 1:41 PM in response to soulmuse

Simple: the bigger the buffer, the more time Logic and its plugins have to process all the audioprocessing. There is a lot of computation going in on in some plugins, and they need time (in milliseconds) to process. So the larger the buffer, the more time there is for computation, the more computation can be done, the more intricate and convoluted the audio can be processed. However, the more delay there will be between notes being played or sung, and the notes being heard on whatever you are monitoring with (speakers, headphones). With a buffer of 512 or 1024 samples (@44.1 kHz) the time (latency) will be too much for your (or anyone's) brain to be able to process that delay: when you record at such buffers, the performace will be so laggy that your brain gets confused and you will stumble. 256 is the biggest buffer you can still manage. Just try it. Buffers 512 and 1024 are only usefull when mixing; they can't be used when tracking.

Apr 12, 2021 11:13 AM in response to soulmuse

I wouldn't worry about it too much soulmuse,

Theres a million things that you don't really need to understand, I would stick it in the middle and get on with the recording and enjoy, I have never touched mine and have had no problems, if its working fine, then no problem.

But as EddieGrey says, if you really need to change it, low for recording and high for mixing.


  • I will point out though, it will depend on what audio interface you are using, I use a Universal Audio one, with this theres no need to go changing settings in Logic. It may be different with your set up. Good luck




Apr 13, 2021 1:57 AM in response to Want_2_Know_More

Most new Logic users think it's some magic tool, that lowers latency to a minimum, that's completely off the mark, if left on it can destroy a mix. It has a very specific function. You should spend time learning Logic's Preferences, you will never regret having a working knowledge of Logic's Preferences (global) and Logic's Project Settings (project specific).

Apr 7, 2021 3:51 PM in response to soulmuse

Also, regarding Low Latency mode, it has a very specific purpose.


Low Latency Mode checkbox and Limit slider: You need to select the Low Latency Mode checkbox in order to activate Low Latency mode and use the Limit slider. The Limit slider lets you determine a maximum amount of allowable delay that can be caused by plug-ins when Low Latency mode is enabled (by clicking the Low Latency Mode button on the Transport bar). In Low Latency mode, plug-ins are bypassed to ensure that all delays (across the entire signal flow of the current track) remain under the Limit slider value. This is useful when you want to play a software instrument with several latency-inducing plug-ins inserted in the channel.


Low Latency mode allows you to limit the maximum delay time caused by plug-ins. Plug-ins will be bypassed to ensure that the maximum delay that can occur across the entire signal flow (of the current track) remains under the chosen value. The Low Latency mode is extremely useful when you need to play a software instrument (or to monitor through an audio channel) when plug-ins with high latencies are already in use—at any point in the signal flow for the selected track/channel.



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Not sure I understand Buffer Size. Can anyone explain?

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