Previous 1 2 Next 23 Replies Latest reply: Jan 3, 2008 12:34 PM by Thomas O'Carroll
john sherlock Level 1 (25 points)
Can anybody tell me how they have their gain streuctures and levels set up in Logic?

I always strive to get everything coming out of logic at 0db. But recently did a project and decided to try something different: I made sure I had plenty of headroom on each logic track, the master and my MAudio audio interface software that feeds my speakers. The result was that I got a rally really loud sound ing track!

Can people explain how they moniter and record using Logic please. Doies everyone have everything set up so that its coming out the speakers at 0db or what?


1.6Ghz (single) G5, Mac OS X (10.2.x), m-audio 18/14 interface
  • Jim Frazier Level 5 (5,325 points)
    I know this is a long read, but please check out this thread from the REP forums.

    Basically, it's confirming what you have discovered on your own. There's a lot of technical info in there, so it's worth taking the time to look through it all.

    Congratulations though. You have just discovered one of the key secrets to getting a great sound from an "in the box" mix.
  • antenna45 Level 2 (155 points)
    I agree.

    Coming from analogue, I really confused myself about this when i first started using Logic and messed up loads of recordings by using 0dB as a target.

    Back in the day when I learnt analogue recording i was told the machine was calibrated to +3 so when I hit zero on the desk it was OK as zero on the desk showed up as zero on the tape machine but was really +3 and so the signal was going in hotter and that was good.

    Then when we replaced it with ADATs, I learned that zero on the ADAT was actually -18db so just hitting the red on the ADAT was OK as it was compensated for so again zero on the desk = zero on the ADAT meter and was OK as it was really -18 inside the ADAT, so loads of slack.

    For some reason I've always assumed that the meters on DAW's did the same so zero on the DAW was a target mark that lined up with a desk output as it was really compensated for inside the box.

    Just to make matters worse, page 549 of the Logic manual describes the clip detector and suggests if a signal clips +1.5 dB you can bring the level fader down by -1.2 as a +0.3 clip is nothing to worry about - and I always assumed that implied there was a safety margin built in to pad the input, since digital distortion is a must to avoid.

    But I discovered the thread mentioned above, which is basically a discussion about the merits of recording inputs at -18 dB rather than close to zero since the dynamic range available at 24 bits far exceeds actual need and 18 db is 3 bits worth of resolution and the benefit is better sound, more headroom for processing etc etc.

    And it works!!

    The only drag is you end up monitoring on the bottom 20% of a meter where there is hardly any resolution and so its a shame you can't calibrate the meter so zero is -18dB or make the meter bars change to red above -18dB so you get some use out of it.

    I use a Fireface 400 as input and double checked with a sine wave on a calibration CD that the level going in, through, and out to Logic shows up at the same level on all the meters throughout. So now I use the Fireface mixer meters as I would a desk just because they are easier to monitor than the Logic ones, and I stick to -18dB as an ideal input level.
  • The Innovator Level 4 (2,305 points)
    Interesting reading in that link. I'm going to have to try it. So basically, when recording a signal, we should make sure the levels are low before hitting the A/D converter of our audio interface. Then mix in Logic maintaining a low level through the final mix before taking it into a mastering setup. Then bring the volume up in the final bounce of the master. Correct?

    This leads to a question pertaining to the A/D and D/A conversions. Say I'm recording an analog signal. Since analog likes a hotter signal, I'll be recording a little more relative noise when I lower the level down before the audio interface. Then, when mixing, I need to use some external hardware effects. I'll be coming out of my audio interface at a low level, bringing the level up in the analog mixer, sending it to the effect units, and then lowering the level again to record the final mix digitally. Assuming my analog equipment is decent, how much extra noise is this going to add?
  • three_eyed_otter Level 1 (10 points)
    {quote:title=The Innovator wrote:}I'll be recording a little more relative noise when I lower the level down before the audio interface. Assuming my analog equipment is decent, how much extra noise is this going to add?{quote}

    I would think that your original signal would be as hot as possible, but your preamp signal would be lower therefore you really shouldn't be introducing any noise...That is to say, don't turn the source down, only the preamp.

    have a good one
  • Thomas O'Carroll Level 2 (300 points)
    This is one of the subjects that can really bend one's mind! The first important element to understand is that analogue metering can be very different to digital. 0dB digital is completely different and unrelated to 0dB analogue. In the analogue world, metering is related to things like current, volts or even Webers to name just three. Digital metering is simply a guide that tells you how close you are to the absolute maximum a D/A can handle (0dBfs).

    When an analogue device connects to a digital device we do need to know how they relate to eachother though, and this is where that -18dB number comes from. Its the level we commonly use as a reference (but it can vary depending on what "standards" are being employed). If you send a constant tone at -18dBfs, out of your digital box and measure the signal at the output with say a volt meter, you will find its the same reading as a tone coming from an analogue source at 0dBu. (In truth, this can be different depending on which country you are in! Here in the UK we say -18dB is -4dBu.)

    BUT... if you are "mixing in the box" this stuff really doesn't matter so much as we are purely in the "digital domain". Indeed if you are making an audio file for CD the "standard" is to get peaks close to 0dBfs.

    The subject can go on and on but as a summary I'd say record your sound between say -18dB and -10dB: its "safe" but also "readable" on the scale, and should work well if going to analogue stuff. This applies to both recording and monitoring.

    Using 24bit recording means you really don't need to worry about having too low a signal. the dynamic range is so huge that you'd have to be incredibly quiet before degredation of the signal occures so why not provide a good safety margin below 0dBfs.

    Everyone was so conditioned to "get levels right" with analogue its hard to accept how forgiving digital mixing is, so long as we don't go passed 0dBfs (and we can even do that in many situations!)

    Below is a list of how the digital dB scale relates to analogue ref. standards...

    0 Digital Clip
    -10 PPM6/+4VU/510nW/100% Recording (analogue entering 1% distortion zone)
    -14 PPM5/0VU/summed Stereo Nagra Tone/UK Dolby A Tone/320nW/(Betacam 2 tracks played in mono)
    -17 Betacam tone -3VU
    -18 PPM4/-4VU/Mono Nagra Tone/UK DAT Tone/UK SQN Tone/US Dolby A Tone/185nW
    -20 US DAT Tone
  • Paul Bissell Level 1 (45 points)
    Good post Thomas.
    I would just like to add that it is pretty much impossible to actually clip once INSIDE of Logic due to the 24bit floating point. This will give you something crazy like 1000dB(fs) theoretically. I didn't believe it so I put a sine wave on a channel and then started adding Gain plug-ins. The channel is waaaay into the red (obviously) but as long as you compensate with the master fader (main outs) and don't allow that to clip, you will have a perfect sine wave in the end. I hit +120dBfs on the channel before I stopped my experiment.

    That being said slipping during recording is not a Logic clip, but a converter clip. Of course, when the main outs clip, this too is a converter clip as it runs out of voltage to represent the digital signal.

    I agree that the standard meters in Logic could be better although I have kind of adjusted my 'sight' when I use them. You can use one of the Logic Pro metering plug-ins that show Peak and RMS values although these actually have +10dBfs at the top which I think is silly as it makes them very deceiving for recording and on the main outs. I personally use the UAD-1 Precision Limiter and bypass the limiter part - I just use it for the metering, both standard and K-System when recording.
    My 2 cents...

  • Bee Jay Level 6 (10,895 points)
    I would just like to add that it is pretty much impossible to actually clip once
    INSIDE of Logic due to the 24bit floating point.

    That's 32-bit float internally. And yes, that's broadly true, subject to the proviso that is is possible to clip some plugins when using high (over 0dBFS) internal levels, as some plugins, notably some Waves stuff, use a fixed 24-bit implementation for input and output, so they do have a fixed ceiling.

    If you really really want to get into the details... deep breath... and...
  • byrd62au Level 1 (0 points)
    If you want better metering (for free) look no further
  • The Innovator Level 4 (2,305 points)
    Yeah, Once into Logic, it's going to be very hard to clip. I've experienced the same thing as Paul...I can peg a track in the mixer and not have any distortion issues but the main outs will begin to distort when pushed too high.

    The 'low level' discussion seems to pertain mainly to the A/D converters. Keeping the peaks low when recording allows you to get all the transients and quick peaks that don't necessarily show up on a meter as well as prevent any possible plug-in issues. As stated in several places, 24 bit recording allows this extra headroom.
  • Paul Bissell Level 1 (45 points)
    Sorry, yes 32bit.
    Thanks for the link - and to think this level of discussion can happen while people are still throwing chairs in the Logic versus Pro Tools stage fight...

    One final thing, after I couldn't make Logic clip internally, I wrote to Hugh Robjohns at Sound on Sound. If yall really, really want to know how to get 1000dBfs from 32bit floating point, it is in this month's (piano on the cover) edition of Sound on Sound. (

  • Bee Jay Level 6 (10,895 points)
    Yep - all this info is in the thread I quoted. In fact, we actually go into more detail than this - it really is quite a mammoth thread, should you choose to wade through it...

    (And I wouldn't want to contradict Hugh, anyway...
  • xs4is Level 4 (2,800 points)
    Bee Jay wrote:
    If you really really want to get into the details... deep breath... and...

    AAAAaaaaaaah. Good times.

  • Thomas O'Carroll Level 2 (300 points)
    byrd62au wrote:
    If you want better metering (for free) look no further

    Nice one

    while we're at it, does anyone know if there is anything similar for a PPM type IIa style meter? I've found one here...
    ...but I'd rather something cheaper because I'm so tight fisted
  • Randall Thomas Level 3 (555 points)
    Thomas O'Carroll wrote:
    while we're at it, does anyone know if there is anything similar for a PPM type IIa style meter? I've found one here...

    Hi Thomas,
    I've been interested in this too, but I've not been able to get the AU version of the demo to be recognized in any host yet. Have you? The VST version loads up in Plogue just fine though.

    No, my search has not turned up anything else either, although the price of this one isn't "too" bad. I'd probably pick it up, if/when it actually works!
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