The Apogee GIO and Mainstage Experiment Part 2

Well, I got through my solo gig in one piece and with reasonable success, but some things became immediately apparent that I will definitely change for next time.


My Setup:

17" Apple MacBook Pro (mine's an older one) running MainStage 2

PreSonus Firestudio audio interface. It uses a FireWire connection, so has lower latency than most USB-based interfaces.

Fender guitar and for vocals a Shure Green Bullet microphone plugged into the PreSonus
(I usually have another Shure Beta58 mic set up for percussion loops, but I didn't bother for this gig).

Novation 49SLII keyboard controller connected (and powered) via USB to the laptop - for playing the occasional keyboard line and controlling levels etc

The Apogee GIO connected (and powered) via USB to the laptop for playing backing and loops, with my expression pedal connected to it for guitar bits.



Come time to perform, the laptop conformed to Murphy's law relating to gigs and played-up despite being solid on every rehearsal, and I had to boot it three times before it played nice - including a forced-shutdown once as it froze up.

The Novation keyboard comes with its own Automap software, and the software runs automatically when you start up a MIDI-compatible application so it can act as an intermediary between the application and the keyboard, but it in this case it locked-up searching for the Novation (which was plugged-in with all its lights going) - forcing the restart.
Of course it goes without saying that this was an agonizingly long time while standing on stage with my guitar waiting to play.

Also - for some reason the GIO didn't recognise my expression pedal - a bit of a major since I need it to cross-fade between some of my guitar tones. I have it set up so it either cross-fades between two separate channel strips with, for example, verse and chorus guitar patches (rather than a complete patch switch I often like to mix in a bit of "clean" guitar with the "distorted" guitar as it adds clarity, or sometimes I set it up so the pedal turns up a second "layering" channel strip with some pad-like or weird character guitar effects at appropriate times in the song.
I suspect maybe the GIO likes to see the expression pedal plugged-in as it fires up, and on the third laptop reboot it finally discovered it (after I had decided it must be the cable!). The GIO doesn't have a power switch, it just turns on when you plug it in.

Both the Novation and the GIO both get their power off their USB connections, and although it normally doesn't seem to make any difference, I made sure to turn on the Novation well after the laptop booted on that third attempt. At home I also usually have a computer keyboard, wireless bluetooth mouse dongle and external hard drive all running happily off the USB power as well, so the lappie should be able to run just the Novation and GIO.

Mix Issues

Once it was all up and going, the issues were mainly mix-based.

The trick, of course, is getting something that works out front as well as in the foldback monitors, and although it actually sounded fine in the foldback, the vocals were apparently too quiet out front.
Trying to turn them up got the mic a bit too close to feedback, which meant turning down the backing instead, which meant some of the backing became just a bit TOO quiet to be able to hear. One song had a triangle rhythm intro that ended up being way too quiet and I got out of sync - needing a restart of the song. A wee bit embarrassing.

So - before the next gig the main thing I will do is;

Create separate audio outputs to the PA system for the different mix elements.

Or at the very least create a separate physical output for the vocals, since they're one of the most critical things to get happening properly in both monitors and out front.


For the gig I did actually create separate subgroups for each type of sound:
(Vocals, Guitars, Drums, Backing, Keys, FX) so I could use the nifty little faders on the Novation to balance the overall mix, but it wasn't enough. It has to be a separate output from the audio interface into its own channel on the PA mixer.

Backing Tracks

Apart from that, the only other niggles I had were with the backing tracks - they were a little inconsistent with their start times due to the too-quiet monitoring.

I have it set up so I can switch between sections of a song with the GIO with the "wait for next bar" setting - meaning you have to hit the foot-switch within the last bar before you want the next section to start. If you're a fraction too early or late, the whole backing is out by a bar.

I'm still not sure of the ideal way to set these backing tracks up. I've tried having the entire backing for the song as one track, but it leaves no flexibility for jamming out on sections or padding it out a bit if you stuff up or something.

I've also tried having just the one backing track with some song section markers that you can cycle within when necessary, and to be honest that wasn't too bad, so I may go back to that method.

The beauty of the way I was doing it this time though is that you can jump to any section of the song if you feel like it, but that flexibility comes with its own risks and problems.

The thing is to try to keep it all as simple as possible for the performance itself, so I'll need to experiment a bit more with the ideal method.


Finally, I'd like to come up with a better system for using Ultrabeat drum machines in my setup and find a way to simply switch between patterns - I might map the bottom few keys on the Novation for that purpose or perhaps assign some of the many buttons on it.

Overall, I'm pretty pleased with the whole setup apart from those few tweaks I'll need to make.
I really like MainStage 2 - it's an incredibly powerful live performance program with only a few minor bugs that will hopefully be sorted soon.

The Apogee Gio and Mainstage Experiment

I have a solo gig coming up and have decided that being yet another singer-songwriter is boring as hell. Especially as I haven't been blessed with one of those voices that could make singing the shopping list sound awesome.

So I need to use everything in my power to add value and variety to the gig - hence the MainStage experiment.

I wanted to be able to go from simple vocal and guitar to full-on backing based on my recorded songs. While keeping it all "live" and interactive so I can jam it out a bit if the opportunity arises.

The beauty of MainStage 2 is that it's basically the guts of Logic Pro bundled into an application for performing live. That means you get the same instruments and effects, plus any of your third-party plug-ins as well.


It means you can also add bounced backing tracks for your songs - with markers that you can loop around or jump to. The markers allow you to see what song section's coming up next in case you forgot.

And there's a cool Looper plug-in that allows you to recreate the current trend of having those dinky guitar pedals that allow you to build up your own musical or percussive layers during a live set. You just play something in, hit the pedal and it loops around while you play something over the top, or you can just keep recording more layers, undo the last one, or clear it all and start fresh.


MainStage allows you to create your own user-interface - you can customise what you are looking at on the computer screen, and also create objects that will be controlled by whatever pedals, buttons, knobs, faders or keyboards you have connected to it in the real world.

Hence me also getting an Apogee Gio - this allows me to have 12 buttons on the foot controller that I can assign to whatever I need to per song, and I can also plug in my expression pedal to do my chucka-chucka-wah-wah thing.


The Gio also has a built-in audio input for guitar or bass, which actually sounds great. Apogee are renowned for their great-sounding converters and it's nice to find even their cheap-ish ones are good. Definitely a good way of getting your instrument into MainStage.

The only hassles I had were when I wanted to plug in a microphone as well as my guitar - meaning I had to use another audio interface as well - in this case an M-Box Pro.

Apple's OSX allows you to combine two separate interfaces together as an aggregate device so they appear as one source to the audio application, but no matter which way I did it, they didn't play nice with each other, eventually degrading the audio quality.

So I had to ditch the awesome sound of the Apogee for the more average M-Box one.
Oh well - at least the Gio buttons still worked and looked pretty.
The little LED indicators change color to suit what the pedals are mapped to in MainStage - ooooh aaaaah....

When you use the Gio with Logic, and apparently GarageBand as well, the foot controls are automatically mapped to Record, Play, Rewind, Fast Forward etc for hands-free recording which is a bonus.

Build quality of the Gio is great by the way - it's a solid little unit - quite heavy in fact, so it's going to stay put on stage, and feels fairly indestructible.

So, for the moment I'm still wrestling my way through customizing MainStage for the upcoming gig - there's still a trick or two I need to learn. There's a Concert/Set/Patch hierarchy that is important to get your head around otherwise the backing stops when you change guitar patches for example - and the synchronisation options with backing tracks and loops has some quirks.

But I'm getting there bit by bit, so I'll let you know how it goes...

Links:
The Gio
MainStage

Digital Recording Levels - a rule of thumb

Okay, I mentioned this as one of my tips in a previous post, but there's confusion and many heated debates out there about the ideal level to record into your digital audio workstation.

I'm just summing up the information readily available elsewhere (if you are willing to wade through endless online debates and the numerous in-depth articles), for people who just want to know right here and now what the best level is to record into their digital audio systems.

So I'm going to start with just a quick easy rule of thumb for these people, followed with a little bit more detail after that to explain why I'm recommending these numbers.

I apologize for simplifying some of the math - but if you're really interested there are plenty of texts and in-depth articles available with a bit of searching. I've included a few references and links at the end of the article.



The rule of digital thumb

1. Record at 24-bit rather than 16-bit.
2. Aim to get your recording levels on a track averaging about -18dBFS. It doesn't really matter if this average floats down as low as, for example -21dBFS or up to -15dBFS.
3. Avoid any peaks going higher than -6dBFS.

That's it. Your mixes will sound fuller, fatter, more dynamic, and punchier than if you follow the "as loud as possible without clipping" rule.

For newbies - dBFS means "deciBels Full Scale". The maximum digital level is 0dBFS over which you get nasty digital clipping, and levels are stated in how many dB below that maximum level you are.

Average level is very important - people hear volume based on the average level rather than peak. Use a level meter that shows both peak and average/RMS levels. Even better if you can find a meter that uses the K-system scale.


Some common questions:

Q: Why do we avoid going higher than -6dB on peaks? Surely we can go right up to 0dBFS?

Answer 1 - the analogue side.
Part of the problem is getting a clean signal out of your analogue-to-digital converter. Unless you have a very expensive professional audio interface, or you like the sound of the distortion that it makes when you drive it hard, then you're going to get some non-linearities (ie distortion) happening at higher levels, often relating to power supply limitations and slew rates.

Most interfaces are calibrated to give around -18dBFS/-20dBFS when you send 0VU from a mixing desk to their line-ins. This is the optimum level!
-18dBFS is the standard European (EBU) reference level for 24-bit audio and it's -20dBFS in the States (SMPTE).




Answer 2 - the digital side.
Inter-sample and conversion errors. If all we were ever doing is mixing levels of digital signals, we would probably be fine most of the time going up close to 0dBFS, as most DAWs can easily and cleanly mix umpteen tracks at 0dBFS.

EXCEPT there are some odd things that happen;

• Inter-sample errors can create a "phantom" peak that exceeds 0dBFS on analogue playback.

• When plug-ins are inserted they can potentially cause internal bus overloads. These can build-up some unpleasant artifacts to the audio as you add more plug-ins as your mix progresses. They can also potentially generate internal peaks of up to 6dB - even if you're CUTTING frequencies with an EQ, for example.

• Digital level meters on channel strips seldom show the true level - they don't usually look at every single sample that comes through. It's possible to have levels up to 3dB higher than are displayed on the meters.
  

Keeping your individual track levels a bit lower avoids most of these issues. If your track levels are high, inserting trim or gain plug-ins at the start of the plug-in chain can help remove or reduce these problems. Use your ears!

Q: Aren't we losing some of our dynamic range if we record lower? Aren't we getting more digital quantization distortion because we're closer to the noise floor?

Short answer. No.

Really, both of these questions sort of miss the point, as we shouldn't be boosting our audio up to higher levels and then turning it down again. So there's nothing to be "lost".

It's the equivalent of boosting the gain right up on a mixing desk while having the fader down really low, giving you extra noise and distortion that you didn't even need. You should leave the fader at it's reference point and add just enough gain to give you the correct audio level. This is what we're trying to do when recording our digital audio as well - nicely optimizing our "gain chain".

The best way to illustrate this is to throw a few numbers up;

Each bit in digital audio equates to approximately 6dB.
So 16-bit audio has a dynamic range of 96dB.
24-bit audio has a range of 144dB.

With me so far? Probably doesn't mean a lot just yet.

Now, let's look at the analogue side where it becomes slightly more interesting.

The theoretical maximum signal-to-noise ratio in an analogue system is around 130dB.
Being awesomely observant, you picked up immediately that this is a lot less than 24-bit's 144dB range!

In fact, the best analogue-to-digital converters you can buy are lucky to even approach 118dB signal-to-noise ratio never mind 144dB.

So - let's think about this.
If we aim to record at -18dBFS, how many bits does that give us?

24 bits minus 3 (each bit is 6dB remember). That's 21 bits left.
What's the dynamic range of 21 bits? 126dB
What's the dynamic range of your analogue-t0-digital converter again? 120dB-ish.
Less than 20 bits.
One bit less than our 21-bit -18dBFS level.

The conclusion is that when recording at -18dBFS you are already recording at least one bit's worth of the noise floor/quantization error, and if you actually turn your recording levels up towards 0dBFS, all you're really doing is turning up the noise with your signal.

And most likely getting unnecessary distortion and quantisation artifacts.

Apart from liking the sound of your converter clipping, there's NO technical or aesthetic advantage to recording any louder than about -18 or -20dBFS. Ta-Da!

Mix Levels

If you've been good and recorded all your tracks at the levels I recommended, you probably won't have any issues at all with mix levels.

The main thing is to make sure your mix bus isn't clipping when you bounce it down.

Most DAW's can easily handle the summing of all the levels involved, even if channels are peaking above 0dBFS. In fact even if the master fader is going over 0dBFS, there's generally not a problem until it reaches the analogue world again, or when the mix is being bounced down.

Most DAWs have headroom in the order of 1500-2500dB "inside the box". You can usually just pull the master fader down to stop the master bus clipping.

Saying that, it's still safer if you keep your levels under control.
Like I mentioned before - a key problem is overloads before and between plug-ins. If your channel or master level is running hot and you insert a plug-in, it could be instantly overloading the input of the plug-in depending on whether the plug-in is pre-or-post the fader. So use your ears and make sure you're not getting distortion or weird things happening on a track when you insert and tweak plug-ins.

Try to use some sort of average/RMS metering, and try to keep your average mix level between about -12 to -20dBFS, with peaks under -3dBFS.

Mastering will easily take care of the final level tweaks.

To conclude - when recording at 24-bit, there is a much higher possibility of ruining a mix through running levels too high than having your levels too low and noisy.

As Bob Katz says, if your mix isn't loud enough - just turn the monitor level up!

PS - just say "no" to normalizing. That's almot as bad as recording too loud.

References:
Bob Katz' web site.
Plus Bob's excellent book "Mastering Audio - the Art and the Science".
Paul Frindle et al on GearSlutz.com
A nice paper on inter-sample errors

Download a free SSL inter-sample meter (includes a nice diagram of inter-sample error )