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Logic 9 Best/Optimal I/O buffer size

Discussion in 'Logic 9' started by satellite maps, Dec 27, 2011.

  1. satellite maps

    satellite maps New Member

    Hi good folks,

    another Logic Express 9 newbie question. I currently have the IO Buffer size set to 128 which gives a resulting roundtrip latency of 8.7ms. I'm using the new Duet 2 audio interface. I'd like to know what the best or optimal setting should be in order to really take advantage of the AD/DA converters. I must admit to still being mystified about latency vs. sample rate.

    thanks much in advance

  3. JuanTahnahmahrah

    JuanTahnahmahrah Senior member

    Welcome to the forum!

    Some hardware features onboard processing or DSP that gives very low latency.

    Otherwise, you may find that different buffer sizes work better depending on specific situations. If you want to record audio, you will generally pick as small a buffer as possible that will work without "clicks and pops," and avoid plug-ins to minimize latency. If you are mixing, and adding reverb or other plug-ins, you will typically increase the buffer size, also to get rid of distortion and clicks and pops. Ideal buffer sizes vary with the power of the CPU, speed of the memory and hard drives, etc. Optimizing your system for audio is also essential.

    Best thing to do is try different buffer sizes and pick what works best for whatever stage of production you are in.
  4. satellite maps

    satellite maps New Member

    thanks for the quick response Juan. So if I'm understanding correctly when i'm preparing to mix I could/should theoretically increase the I/O buffer and sample rate to 1024 to get the best quality for my mix, correct? Conversely If i'm recording a vocal or acoustic guitar I should lower the buffer size to maybe 128. Also looking quickly at my Duet 2 I don't see any onboard processing or DSP. Is that right? Am I missing this setting somewhere?

  5. JuanTahnahmahrah

    JuanTahnahmahrah Senior member

    Those buffer sizes should be fine.

    8ms is probably ok for tracking, depending on the musicianship of the players/vocalists. The initial tracks should not be a problem, it's noticeable when you start overdubbing.

    I don't have a Duet, but I have a preamp that lets me loop the active input mixed in with the playback track, without using software monitoring.

    When you start having issues with tracking, you will start to figure out workarounds.
  6. CSeye

    CSeye Senior member

    Satellite Maps,

    You're thinking is correct: Low I/O buffer e.g. 128 for tracking, and a high I/O (512 or 1024) for mixing.

    If you're mostly recording voice, electric guitar, miked acoustic instruments, etc, you can probably keep the I/O lower until you start to add effects and processing. If you' re recording MIDI triggering software synths, you'll know when it's time to increase the I/O as you'll hear artifacts indicating cpu stress in addition to the visuals provided in the System Performance meter in the Transport.

    Don'y worry about onboard DSP processing with your Duet 2 because it doesn't have any.

    Set the bit depth to 24 in Logic Preferences>Audio> Devices tab.

    The sample rate is set in the Project Settings>Audio. Set it to 44.1 if you're recording music for CDs. If you're creating soundtracks for film, then use a sample rate of 48. Since this is a project specific setting, create a Logic template with the sample you routinely use so you don't have to worry about this setting.
  7. georgelegeriii

    georgelegeriii Senior member

    One thing about the Duet: it does have an internal mixer that allows you to monitor directly from the inputs, giving you no latency. Read the Maestro manual or check the Apogee web site for a little tutorial on how to use it.

    The bad thing is that if you are using any of logics internal effects for your track (either as an insert effect on your input, or as one one your track insert) you won't be able to hear them back without hearing both your input from Logic with it's latency as well as the Duet's direct input motor, that means your sound will be flanged or phased (technically not really, to do either the delay tome has to modulate), basically thin and strange.

    So, you will need to turn down your record track when recording, and then turn it up when playing back. A bit of a pain, but you get NO latency at all.

    I would really suggest you use an outboard mic pre/compressor combo, that way to can make your levels a bit more consistent with the compression. Otherwise your vocalists dynamics will make monitoring for both of you an issue: going from very soft to very loud for example.

    I hope this isn't too confusing. I am assuming you have a bit of knowledge regarding signal paths and recording technique. If you don't, you might want to check out one of "Sound On Sound" magazines older tutorials own this. Any articles older than 6 months can be looked at, and frankly it is one of the very best resources of recording and technology out there in my opinion.

    You can find SOS here:
    it is a great resource regardless of your experience. I have been in this industry for 28 years now, and do high end tech support for composers and writers here in La, and I still learn new things every month from SOS.

    Good luck, and welcome to this new addiction ;-)

  8. satellite maps

    satellite maps New Member

    Thank you all very much for the excellent advice. This is one of the most supportive forums i've ever encountered.

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