Need Limiter Settings for Live Recording

[bilco]

I successfully used a DIGI001 and a Behringer ADA8000 to record our band live Monday night. Unfortunately, the dynamics of the band range from one extreme to the other depending on what we're playing. I can't play bass and tweak levels at the same time.

My solution was to have all of the inputs come directly into Aux Channels with a limiter plugin inserted. Then each aux channel was bussed out to an audio track and went to hard disk from the audio track.

It did keep things from clipping, but the sound in some places gets a little...... squashed or something. I just used the generic "factory limiter" setting.

Can anyone recommend settings for the limiter plugin that would let things get right up to nearly clipping and then limit only just as it's ready to clip?


Thanks,
bilco

[benunh]

Brickwall limit.
Set your threshold really high (-.5 or so) and your ratio as high as it will go (at LEAST 20:1) and your output ceiling at -.3 to be safe.
That should do it for you.
No guarantees about it not being squashed (depends on the signal going into it), but that won't squash 'normal' levels.

Rock on.

Ben

[Kahli Burke]

If you are talking about a plugin and not an outboard hardware compressor/limiter, I don't think that running the tracks in through auxes and then recording the limited signal is the best technique. For one, it could make it harder to notice if your signal is clipping, for another it doesn't let you change the plugin settings after recording.

You aren't going to get any more dynamic range from the tracks by limiting in this way. The tracks (regardless of whether they are audio tracks or aux inputs) have a specific dynamic range that you must work within. If the signal is high enough to clip an audio track, it would be high enough to clip the aux track as well. Really what you're doing here is just committing to your limiting levels because you don't record the raw audio signal, you record the processed version only. If you are using an outboard compressor on the signals before ever running it into you 001, that's a different story, but you probably don't need to do that either.

So my recommendation is to record the raw audio and then experiment with limiting/compression after the fact. Then you can adjust the thresholds/ratios to the performance and even automate them for different songs if they have different dynamics.

Now the problem you will need to address is how to set your recording levels properly without clipping. As you may have already read in this forum, if you are recording at 24-bit resolution, you already have a very large dynamic range to work within (144dB), this means that you don't need to have your recording levels really high in order to get good results. Forget about the analog days and the idea of hitting the tape hard (so the meters peak in the red) in order to get a saturated sound. 24 bit recording means your noise floor is at -144dB, which is going to be much much better than the preamps, microphones and space you're recording in. You're basically going to get the same results whether your levels are at -3dB or at -20dB. Of course you'll want to bring the average levels up after you record when you are averaging a lower level, but by turning the gain down, you'll have less inherent noise from the preamps (cleaner sound) and you'll give yourself headroom for loud transients you'll get in that song where the drummer goes ape.

I record my band live using Pro Tools and an 002. I'm in the same situation with regards to tweaking levels while playing, so I need to set and forget. The best technique I've found is to do a short sound check to set the levels before we play. When I do this I have the other musicians play as hard as they think they're going to. I set the gain on preamps so that my levels are clearly in the green, maybe showing just a tiny bit of yellow. I then back off the levels just slightly to adjust for the fact that everyone ends up playing louder then their "loudest" when I do that sound check.

So this way, I get the recording at decent levels without having to worry about clipping. The waveforms look a little wimpy on the screen, that's true, but you've got a lot of headroom to work with and then you can compress after the fact when you can critically listen to avoid squashing or distorting the sound and compress/limit to taste.

Hope it helps,

Kahli

[JonnyGinese]

Bilco... if you are recording live.. and you arne't using the mix for audible monitoring... Just set the levels on the inputs so that they won't go into the red... even if they are only half signal... and then record them.. and use the plugins later to get them to the volumes you want. Aslongas you have a good clean signal it won't make a difference.. since its digital. I used to worry abotu getting my levels almost to peaking... i even had outboard limiters on all the inputs so taht it woudl never peak.. but i did get some squashing sound some times... Best way to do it is to make the level coming in at a point where it will never peak... and ten jsut raise it later with effects.

Jonny

[Carl Kolchak]

Another good reason for not recording through a plugin limiter is that it would be easy to clip the analogue to digital converters in your digi 001 or behringer, and not realise because the meters on your Aux/Audio channel strip are telling you there are no peaks. In essence you'd just be limiting an already knackered audio signal (in extreme cases).

It's standard practice to line up your recording levels so that 0VU = -18dBfs (if you don't come from a background of using VU meters and analogue tape that may mean nothing to you, but to get a simple referance you could insert the free bombfactory VU meter on one of your channels and see how it relates at different settings to the channel strips peak meters. Try using with the signal generator plugin and a 1K sine wave at -18dB) don't think of this as loosing 18dB of digital dynamic range, so much as gaining 18dB of headroom over an analogue recording medium.

As mentioned before, you have stacks of dynamic range at 24bit so why not keep your average signal floating around the -18dB area on the channels peak meters (ctrl click window at the bottom of the strip to cycle through level/peak/delay) then you have plenty of headroom and don't have to commit to printing FX to your audio. You're better off treating tracking and mixing as two different stages, rather than trying to do both at the same time.

Of course all this is just my opinion, hope it helps!

[bilco]

Thanks everyone! I'll try again tonight and I'll try 24 bit and try to get just decent levels without limiting. Sure wish I had another pair of hands........ I am pretty sure I WAS clipping the preamps and then smashing it back down again. Oh well, plenty more hard drive space to experiment with.

I have to say, for a first time out using PTLE to record a band live, it's really awesome. Way cooler than my vintage 1978 Dokorder 4 track (copy of a Teac 334 ) used to be.

I love Pro Tools!!

bilco

[JonnyGinese]

I agree with Carl... good advice... ANd as far as it being common practice of 0 analog to -18 digital... That IS what the convertion is. lol.. its not just a practice... Its what its supposed to be. ON my analog board I have my inputs with the trim almost wide open and i have the inputs on my Digi002r at -10... it works out really well.. My board is really quiet and it never clips unless i push it to about +30(my overload indicators).. and an input signal @ +3(my peak indicators) on my board will peak my PTLE meter.. I prefer to set it up this way.. I could run my inputs on PTLE to be 0 = -18.. but This way i think i get a better signal to noise ratio. On my 4 volume pods on my 002r i have each one only up 1/3 of the way.. so i really have ALOT more space to push my signal.

As for your Live Recording Bilco.. Just record it at an average of being in the middle of the meter on PTLE and you won't have a problem... and if you are using 24/48 and it happens to peak slightly... it won't break up or clip.. unless its really in there.. Like a SNare drum.. if its cracking around -4 and the drummer hits a rimshot and it cracks to +2 or +3 it shoudln't clip. I used to worry about it alot.. but not anymore.. I just slap on the plugins and put the output... anyways.. have fun

jonny

[Carl Kolchak]

" ANd as far as it being common practice of 0 analog to -18 digital... That IS what the convertion is."

You're right Jonny, I just meant in terms of line-up standards it's not the only one, some studios use -20dBfs.

It's also worth noting that the digi 002/rack i/o are pre configured to give you 14dB headroom, meaning if you're matching a signal at +4dBu or -10dBv throughout your audio path, the peak meters in pro tools should be reading -14dBfs (I don't know if that's true of the digi 001, and I very much doubt it of the behringer unless digi have put 14dB headroom on the light-pipe i/o which seems unlikely).

cheers!

[bilco]

I ran out of time to get everything setup last night; I'll try next week. I moved the DIGI001 card to my Dell Laptop's C-Dock II Docking Station, but I was getting some glitchy behavior and I need to see what's up with that. The Dell does not have as much firepower as my HP tower, slower hard drives too. But I have gotten it to work before with an Audiomedia III and lately with the DIGI001. I'll have to run the Davec to see how much torture it can take.

I will have to try the 24 bit thing I guess. I have always done 44,100 @ 16 bit to avoid the whole having to dither thing. The one time I did record some session in Cakewalk @ 48,000 before I got Pro Tools, I ended up with some strange sounding harmonics on the lead guitar when I converted to 44,100 for CD and so I have always stayed at 44,100 since then.

The light bulb finally went off about having the limiter inserted AFTER the preamps and converters in the ADA8000. I could be limiting a signal that is already distorted by the preamp/converters in the Behringer, although I had tried to set the levels initially in each AUX input with the plugin set to the Bypass mode.

About calibrating 0 on a mixer to be -18 in PT: When I was coming into my AMIII card from a Behringer mixer a few years ago, I did the whole sine wave in thing to calibrate, except I think I was going for -12 instead of -18. So I get the concept. But how does that translate to the Behringer ADA8000 where I am plugging mics into the XLR inputs, setting levels by watching the clip LEDs (which I don't trust to be all that accurate) and the PT meters, AD converting and then sending it through the optical ADAT lines to PTLE? How do I know if the Behri is calibrated? I guess I could run a sine wave into the line inputs of the Behringer and then calibrate that in the hardware setup of the 001. Please help me get this. I really want to get the gain staging thing right.

PS Are any of you living in the Austin, Texas area? I would love to get together with other PT-Heads. Maybe we could help each other out in these live recording situations. It is REALLY hard to concentrate on playing and keep from looking over at the monitor to see what PT is up to. I am willing to tweak knobs and run mic cables etc. I am the bass player in a 4 piece Piano Bar cover band that plays everything from '30s and '40s swing era songs to Sinatra, C&W and some rock standards.(THAT's why it's so hard to set the meters and forget them, one minute some torch song with brushes on the snare drum, one request later, the drummer is slamming out "Born to Be Wild") We're a bunch of old guys having a great time and getting paid for it! I am also one of about a million singer/songwriters living in Austin, but alas, those annual royalty checks for $2.34 from BMI pay NO bills, so the cover band thing helps keep the kids in shoes, cars,etc.

This is a great forum and yes, I really, really love Pro Tools!
Thanks,
bilco

[JMS40]

I will have to try the 24 bit thing I guess. I have always done 44,100 @ 16 bit to avoid the whole having to dither thing.

You should dither even when working with 16bit sessions.
From the reference guide:

You should use the Dither Plug-In when mastering to a 16-bit file with the Bounce To Disk command, or when mastering to an external device that records at 16-bit. This is even when using 16-bit sessions.
Even though 16-bit sessions use 16-bit files, they are still being processed internally at a higher bit rate:
-32-bit floating for Pro Tools LE systems.
For this reason, whether you are using a 16-bit session or a 24-bit session, it is reccomended that you use the Dither Plug-In when mastering to 16-bits