Review – the UAD Apollo Duo

UAD Apollo under laptop with heaps of UAD plug-ins showing

It’s no secret I’ve been a fan of UAD since I purchased the little UAD-2 Solo/Laptop card. Since the Solo has only one of the SHARC plug-in processing chips in it, my plug-in demands quickly overloaded it. This was usually accompanied by the fiddly and annoying juggling of said plug-ins and bouncing or freezing of tracks in Logic.

So when UAD’s Apollo emerged as their new flagship product, I was excited (and after 25-odd years in the music industry, I don’t get excited by much anymore). Not only did the Apollo come with some reputedly rather nice preamps, it contained either a Duo or Quad SHARC chipset for running UAD’s rather tasty plug-ins as well. Even better, you could have some of those UAD plug-ins in between those preamps and your DAW. This means you could have the virtual equivalent of a very expensive vintage tracking chain of effects with a usefully-low recording latency. That’s pretty damn cool.

So I managed to get my hands on an Apollo Duo for a try-out (Thanks Leon at NZ Rockshop!) since all the Quads sold out instantly in NZ on hitting the shores.

A big potential selling point for me was that the Apollo is supposed to be able to be fitted with Intel’s new Thunderbolt interface. (More here) This involves purchasing an extra add-on card which hasn’t been released yet at the time of publishing, but I really look forward to checking out the performance when these become available. The Thunderbolt transfer speeds are supposed to be blisteringly-fast – actually similar to connecting directly to the PCI-express port on your computer (and there’s two channels of that per Thunderbolt port). This means even lower latencies between the interface and the computer, with none of the Firewire-bus wrangling. This is really only an issue anyway if you have external Firewire drives daisy-chained to the Apollo.

UAD have a solid reputation in the audio industry – they make great “vintage”- quality hardware, and have also pretty much nailed the “accurately-modelled vintage studio hardware” DAW plug-in market. I have found the UAD version of classic vintage units such as the Pultec and LA-2A to be several steps above other versions I have tried. They model each component of prized representative units of vintage hardware to capture all the non-linearity that made the originals so musical and desirable. Then they add any extra handy functions to make them slightly more usable in modern DAW production.

As you can imagine, all this extremely detailed modelling takes a ridiculous amount of extra processing power – hence needing some heavy-duty plug-in engines to do all the hard work. This is why UAD plug-ins only run in UAD hardware (and no doubt also for copy-protection reasons). Although you can buy duo or quad-chipped UAD Firewire and internal PCIe units, the Apollo conveniently includes the chips in this tasty preamp/audio interface unit.

Apollo Front Panel

The build on these units is solid. They are only one rack unit high, but are quite long.  They are well-perforated for good ventilation, and the finish is impeccable. Simple front panel controls make operation easy, and probably help keep the cost down. Large gain and output volume knobs have an LED ring around them to indicate current level, and each has a push-switch included. For the gain knob, this selects between the four microphone preamps, and for the master volume knob, it is also a mute control.
Power is supplied through a fairly chunky external power pack that handles the various international voltages.

Apollo Rear Panel

Despite all the digital gear packed around it, the analogue side remains completely silent and clear.

The Burr-Brown preamps are very nice – high gain, quiet and very transparent. Luckily you can insert some UAD plug-ins between the preamps and the DAW inputs to colour this sound should you wish.

In fact, as I hinted at earlier, this is one of the other major selling points of this unit – the ability to put a chain of UAD plugins in between your microphone, guitar or bass and the input of your DAW. For those of us that like to commit immediately to a particular sound, and can’t afford the luxury of racks of vintage analog gear, this is a godsend. Of course this is only useful if the recording latency is low enough to be usable, and in this case it is – only a couple of milliseconds in total. Saying that – there was still a little wrangling of monitor sources in the DAW to avoid the ol’ comb-filtering effect of two sources with different latencies.

I look forward to seeing what the recording latency is like in the Thunderbolt-interface version of the unit when it becomes available. It means bringing it down to latencies that are very close to using the old Pro Tools HD hardware – less than 1 millisecond. That’s pretty darn good.

Okay, so far everything looks pretty good – there’s not much to fault in this unit really.

I would say that the only niggles are in the software/firmware.

My first issue was with the combination of the Apollo and my old UAD-2 Solo/Laptop card. Since my trial Apollo unit was only a Duo, I thought, “ah, I’ve still got my little Solo/Laptop card as well, so that’s three processors – not too far off four”.
While this was true in theory – it didn’t immediately work in practice. I discovered there are some “quirks” with laptop cards. Cardslots in general have always been a little quirky – although in theory it’s fine to hot-plug certain devices into the guts of a computer while it’s running (and the Solo/Laptop by itself usually seemed to handle it okay) – it’s really pushing the technology/software/OS towards their limits. I found that with the combination of the two devices, I was getting some UAD plug-in overload messages opening up old Logic sessions with UAD plugs inserted.

Anyway – the friendly and patient support guy at Universal Audio helped me troubleshoot it on the phone, and it turns out that as long as the Solo/Laptop card is plugged-in before booting up the laptop (so it shows up first in the hardware list in the UAD Control Panel), and then the Apollo is plugged in/turned on afterwards, everything works just dandy. Any loaded UAD plug-ins in the session are now distributed over all the available SHARC chips. Sorted.

Another minor annoyance I found was with the implementation of the plug-ins within the Console app – you can’t drag and drop the plug-ins into a different order – you have to cut and paste etc. A bit fiddly when you’re wearing a guitar and trying to quickly get something tracked while the idea’s hot.

The Console plug-in is modelled on an analog console, with its own built-in effects sends and returns and headphone sends. It allows some nice low-latency patching of monitoring effects like reverb and delay, but it doesn’t include the more obvious ability to patch any input to any output like most audio interface mixer apps. This is offset somewhat by each of the various ins/outs within the Console app showing up in your DAW as ins/outs, but is not what you would call “conventional” or transparent operation. Especially if trying to set up multiple headphone sends, each with a different mix.

There’s no beat-clock for UAD effects that depend on session tempo either – I can see why this is difficult to include as the Console app sits external to your DAW session – but perhaps having the facility to accept MIDI clock via a virtual MIDI port would be valuable. (UAD boffins – maybe this could be implemented via the Console plug-in?) Having beat-synced delays is important whilst tracking to a song with delay-based effects.
I’d also like to see the ability to map MIDI controllers to various Console parameters/UAD plug-in parameters.

It’s not all niggles though – there’s a handy plug-in version of the Console for remembering the settings as part of your project – nice! And the Console app allows the easy copy/pasting of your channel mix settings to your effects/headphone sends – very handy!

Review of features

Sound quality; I couldn’t fault it. This is a very nice unit. Very low noise and no obvious colouration in the mic preamps. Plenty of gain. Full 48 V Phantom power for mics. Pad, low cut and phase switches. All controlled by a fancy rotating knob with surrounding lit ring showing current gain, and some selector buttons.

I tried recording and playback at various sample rates up to 192kHz. I didn’t find that much difference between them really. Just correspondingly lower latency and perhaps a touch more “silkiness” on some instruments in the higher frequencies. I’m guessing the Apollo oversamples for the lower sample rates anyway, so there’s not much in it. (And some of the UAD plug-ins themselves are upsampled as well).

I tried recording some acoustic guitar – first without any UAD plug-ins inserted – the preamps sounded open and clean-sounding. Very transparent and very low noise. About what I expected from UAD. Very nice.

I then tried inserting some plug-ins into the Console app that comes with the Apollo.

UAD Apollo Console App

Some gentle compression with the 1176SE, Pultec adding small amount of top, and LA3A as peak limiter. Beautiful – now the guitar was sounding shiny and firm, with no nasty artifacts, and no noticeable noise problems despite the compression.
I had to pull the fader down in Logic and just monitor through the UAD console app to avoid the latency-induced comb-filtering.

Next I thought I’d try one of the High-impedance instrument inputs on the Apollo’s front panel. The input automatically switches from Mic/Line to Instrument when you plug in. Handy. Oh – and it sounds really good. I’m not suprised – UA has an excellent reputation with not only audio quality, but also pro-level usability design.
As I’m guilty of pretty much always just plugging my Fender Jag-Stang straight into my interface to capture ideas as fast as possible – I could tell that this immediately sounded better than my old interface ever did. A LOT better. Winding the gain knob up gave a really nice creamy analogue distortion too. I’m really liking this unit so far. Clean on mic inputs, nicely coloured analogue on the Hi-Z inputs.

Let’s try the dodgy acoustic guitar piezo bridge pickup. Plugging straight into the Hi-Z jack. Wow – same thing again. A huge improvement – in fact the best I’ve heard it sound from the pickup. It’s normally quite clicky and overly percussive. This sounds much smoother and fatter.

And my Jaguar bass sounds great plugged straight into the Hi-Z input as well – as much as I love my old Drawmer 1960 tube preamp on bass, I have much more control with the Apollo.

So – it seems like we’re getting 4 very good quality mic preamps with this unit. The mic inputs can be switched to the line inputs on the back, with the first two inputs automatically switching to Instrument when something is plugged into the Hi-Z jacks in the front.
There’s also the usual S/PDIF I/O (with auto sample rate conversion if you like), and an S-Muxed dual ADAT I/O that can do either 8 channels of regular 44.1 or 48kHz sample rates or handle up to 4 channels of 192kHz audio.
There’s two headphone outputs that can be fed from various parts of the Console app.

The controls are simple and clear – each of the four preamp inputs can be selected by pressing the gain knob, and rotating to set amount of gain. A circular light ring surrounding the gain pot shows the amount of gain added. Switches apply 48v Phantom Power, Pad, high-pass filter etc to the selected channel.

The Console app shows much more detail, reading out in decibels and showing the state of everything all at once. It has its own built-in headphone busses and effects monitoring setup.
Because this is a digitally-controlled preamp, all settings may be saved and recalled – as I mentioned, there is a Console plug-in (in all the typical plug-in formats) that can be inserted into the session that can recall all these settings automatically if you like.

Conclusion

A great unit overall. Sounds fantastic and is, without doubt, value for money – especially the Quad version.
Not quite perfect yet – there’s still a few very minor things to iron out with the interface software, and the Thunderbolt adaptor is still absent (as at July 2012).

For this price (about USD$2,500 or just under NZD$4,000), the question is why woudn’t you buy one.
I did.

ps. I forgot to mention, for those of you who know little about UAD yet; like most of their plug-in based devices, the Apollo comes standard with the UAD-2 Analog Classics plugins collection (the 1176LN, LA-2A and PultecEQP-1A) and a voucher for $100 worth of plugins.
That’s pretty cool value – especially if you wait until one of their fairly regular online sales to use it.

Plus – when you install your plug-ins, it installs every UAD plug-in, and the others (that you haven’t purchased yet) can be demoed for two weeks from whenever you click the little “demo” button on the plug-in. I’ve also noticed that all the demos seem to be reset every time you purchase a new plug-in. These guys are very canny yet professional with their marketing, and they certainly know how to look after their customers.

8 Top Logic Pro 9 Features

Bounce Track in Place

Here’s 8 of my top features in Logic pro 9. If you have any others – feel free to comment!
Click on the pictures for larger versions.

 
8 Bounce-in-place, for either track or region. Bounces down either your audio instrument MIDI regions or your audio regions – with or without plugins – to new audio regions and then mutes the original track or region. Great for rapidly “printing” any special processing or pitch-fixing plug-ins like Melodyne into a solid file. Note you can also do this for EVERY track simultaneously should you wish to export all your tracks into a different DAW program for mixing or something. Closely related to…

Freeze Track – the Two Modes

7 Freeze Track – you can freeze a track either just after the instrument (or less-usefully after an audio file I suppose), or after all the plug-ins and the fader, depending on how much load you want to take off the processor and how much control you want over tweaking plug-ins. Basically it does a cunning invisible 32-bit float bounce of the track (which you can go and copy from its folder if necessary) and then disables plug-ins/instruments on that track – in other words swapping processor power for hard-disk speed. It still sounds exactly the same but now you have more CPU power to do stuff. You can still un-tick the freeze button at any time to do some editing etc. Essential when you inevitably have so many plug-ins and instruments in your project that it won’t play!

Replacing or Layering Drum Track

6 Replace or layer individual drum tracks – need to fix up or bolster those poorly-recorded kicks, snares or toms? Logic can automatically detect the drum hits on an audio track (you can adjust sensitivity), then you can choose a replacement (or layered) drum sample which is then automatically imported into a new EXS24 sampler which is placed underneath the original track with a handy MIDI trigger region to play it.

Convert Audio Region to Sampler

5 Convert to sampler – this is cool and ridiculously easy. You can either select a strip-silenced bunch of audio regions on a track, or you can let Logic identify the transients in the audio region/s, then it will automatically pack all those audio chunks into a sampler assigned to MIDI notes ready to play. And then mute the original audio regions. Oh, and it conveniently creates a MIDI region that plays those samples exactly as if they were still the original audio files – so it recreates the source audio exactly. You can delete this if you want to do something different.

Select Sampler Options

Final Converted Sampler Track with MIDI region

Making Groove Template from MIDI Region

 4 Create your own grooves, or match your MIDI stuff to live drums. You can do this in a few ways – using the “Audio-to-MIDI” groove template under “Factory” in the Sample Editor window, or recording a MIDI track while you tap along with every 8th or 16th note on a MIDI keyboard, or use the drum/replacement doubling trick above. Once you have a MIDI file, select it and go to the region inspector pane, click on the “Quantize” drop-down menu and select “Make Groove Template”. This can then be applied to any other MIDI or Audio regions.

Audio to MIDI Groove Template

 

Region Gain

3 Region Gain – many users of Logic still don’t know you can use the little region inspector panel top left of the Arrange window to adjust the individual gain for each audio region.  It can replace the use of automation in some cases – just cut up your regions and set the gain for each one. It handily does this before it goes through any plug-ins in the channel strip (although this can be a problem if you have compressors inserted and you’re trying to use it instead of automation – the compressor may “fight” any gain changes).

 

Capture Record Button

2 Capture Record. This mysterious little button is like magic. Say you’re just been jamming along on your MIDI keyboard or controller in Logic, then you realise you’ve just played something absolutely amazing, but “oh no!” you haven’t been recording!

Fear not, just press this little button and what you just played will magically appear as recorded. If the button doesn’t show up for you – Ctrl-Click on your Transport bar and add it.

Tip – use “Take Folder” mode for your MIDI cycle recording – even more useful.

NB: This doesn’t work with audio unfortunately – unless you use the sneaky “Punch-on-the-fly” mode trick. When “Punch on the Fly” mode is selected under the Audio menu, Logic is ALWAYS recording any armed audio tracks whilst playing-back. You just need to actually hit “record” for a second while it’s still playing to let Logic know you want to capture and hence retrospectively grab what you just played.

Take Folder

1 Take Folders. Most people know about this one but it still bears mentioning. Record while in cycle mode, then just swipe the bits you want to keep from each take. It’s a fantastic way to get great vocal takes and Apple really has this one nailed better than the competitors. Many people aren’t aware of all of its cool features though.

Some additional tricks are; creating alternate versions of your comp (good for backing vocals or doubling lead vocals), exporting particular comps to duplicate tracks, editing the audio itself (eg trimming or moving parts) while comp’ing a good take, tagging the best bits as you go, or editing the size of the looped section (and folder) if it chops off the beginning of the loop each time.
You can also manually create a take folder out of various selected audio regions for creative purposes and you can also cut the take folder into chunks if needed.

What are your favourites?

Zed’s Digital Audio Top 10 of 2011

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This list is made up of those plug-ins and digital apps I used the most this year and/or made me the most excited. Most are available for both PC and Mac, and only a few are still stuck in 32-bit mode – hopefully that will change soon.

10) Izotope Ozone 5. Link This is the Mastering app of choice for most semi-pro engineer/producers and a great update to the ubiquitous Ozone 4 multi-processor – albeit slightly expensive for the “Pro” version without the introductory discount. The Pro version allows you to insert each module as a separate plug-in if you so desire, and has awesome audio visualisation options (plus a few extra features per module). Within this plug-in you get all the essential tools to repair a finished mix – a mastering limiter (Maximizer),  multi-band compression, multi-band exciter, multi-band width controls (Imager), mastering reverb, and an awesome EQ with handy frequency “solo” for ease of locating those crazy out of control frequencies. Oh, and you can go stereo or mid-side depending on your needs, and you also get all sorts of dithering and metering options.

9) SoundIron Emotional Piano 2Link It’s amazing how often you need a piano in a mix, and because I don’t have access to real one, I’m always struggling to find one that sits nicely in a song, especially as the ubiquitous grand pianos that seem to come with various packages don’t always work with the track. This piano is meant to be more “soundtrack-ey”, it’s warm, has character, and seems to sit much better than any of the others. If I want a clean sound I use Modart’s Pianoteq Link – a very nice modelled piano.

8) Avid Pro Tools 10Link Just so you know, although I’m a Logic Pro afficianado, I’m also a trained Pro Tools user and it’s good to see Pro Tools coming along so well and, although they’ve had the studio-recording side totally nailed for so long (and are the industry-standard for recording in the studio), they are still catching up somewhat with everyone else in the compositional features stakes. Also – now you don’t need to have a piece of Avid hardware to run it, it simplifies (and cheapens) your setup. Still a bit overpriced (especially as you have to pay quite a bit extra for much of the really cool stuff), but if you want to work with a variety of studios, you will probably need to use it at some point.

7) Arts Acoustic ReverbLink This algorithmic reverb is not only easy on the CPU, it sounds fantastic. I think our love-affair with the impulse reverb is fading, because as good as they initially sound, they are inherently linear – the sound doesn’t change based on level going in, so they can end up being a bit sterile. I think of them as “precision reverbs”. The Arts Acoustic can still sound clean, but you can get some pretty twisted sounds out of it if you need to, or some gorgeous Lexicon-like warmth. I use it a lot for dark and twisted drum reverbs, and for clean and open vocal reverb.

6) Logic Pro 9Link Notice it’s not at number 1, because as much as it’s my main tool in the studio, and it IS pretty damn awesome, (and ridiculously good value for money BTW – especially as you can now buy it on App Store for $200 USD) Apple have let it sit in the background for a while now, with very few updates, and some bugs that have been there for several years. I’m hoping that they release version 10 soon, without destroying what makes it so good – like they almost did with Final Cut Pro X. Runners up – Ableton Live Link – you’d have to have your head in the sand not to notice this DAW (Digital Audio Workstation) app increasingly dominating the market – although mostly in DJ, electronica and live performance realms, and Reaper Link – an inexpensive and increasingly fully-featured application that’s really taking off.

5) SoundToys Devil-Loc DeluxeLink Actually the entire Sound Toys native bundle is also fantastic (and pretty well priced if you’re a student and can get the Academic pricing), but I’ve found this particular compression-with-distortion plug-in an essential for big fat drum sounds. You can get it pumping in a really good way, and it sure adds instant excitement to the drum mix.

4) D16 Group ToraverbLink This plug-in reverb can get the biggest, widest, lushest chorused reverb sounds ever. It’s very impressive, and once you hear it, you’ll want it. I use it every time I need a huge sense of space and distance on something in my mix. Actually the D-16 Group do some fantastic plug-ins – I have the rest of the Silverline collection and also use the Decimort (which can emulate the colouration from various older samplers) and Devastator (a multi-band distortion unit) plug-ins a lot.

3) Slate Digital VCC (Virtual Console Collection)Link The idea with this plugin is that you put it on every channel strip and/or the busses to simulate one of four (now five!) analogue mixing consoles. It’s very subtle per channel strip, but somehow adds up to making a mix sound great and just “gel”. Runner-up to this is the very affordable Sonimus Satson Buss Link.

2) Celemony Melodyne Editor 2Link When it comes time to transparently fix poor intonation in vocals, without the obvious side-effects that you might want for some styles of music, then Melodyne is the one. It retains the nuances of phrasing and vibrato, and allows you to just fix the gross pitch errors if you like, or you can still go more extreme if you really want to. Also great for matching and creating backing vocal lines, repairing guitar tracks (got one string out of tune?) as it can now do polyphonic tracks, and my favourite; fixing poorly-played bass lines, because you can quantize to a time grid as well as fixing poor intonation on cheap basses. There are a bunch of products that Celemony put out, including the multi-track Melodyne Studio, but I like this one as it’s a pretty full-featured plug-in that can also do Rewire. An absolute essential!

1) Anything by UADLink This was my big “Eureka” moment this year. I decided to buy the UAD-2 Solo Laptop card to get some more processing into my overstressed Mac Book Pro laptop. Here’s what I found – the UAD plug-ins sound so much better than any other versions of the same plug-in, and sound so very close to the real hardware units that they’re modelling. You don’t need “golden ears” to tell the difference either. It might have something to do with the way the plug-ins are up-sampled for processing, or it might be the ridiculous huge amount of detailed modelling that they’ve done to recreate the vintage equipment so realistically. My favourites so far are the good old Pultec EQ – it really does just make things sound better – even without adding any EQ (although you probably will), the Ampex ATR-102 reel-to-reel, the Fatso Jr/Sr for, well, fatness, and the SPL Vitalizer for adding character to synths. My credit card is still hurting from going a wee bit crazy on these plug-ins this year, but I don’t regret it.

Notable mentions: the free Michael Norris effects collection Link for some quite radical granular processing options – especially useful for sound design. Some of the cool Waves plugins Link; for example the Kramer MPX reel-to-reel tape recorder and the Vocal Rider Not cheap, but good. Xfer Records’ LFOTool Link – adds tweakable sync’ed modulation to just about anything. Great for locking-in, enhancing or creating grooves in any track. Izotope’s Stutter Edit Link – awesome for adding those extra crazy head-sounds to your mix and for creating some extra action when it gets too boring – and you can play it in from a MIDI keyboard. The Sonnox collection Link – every single plug-in is useful and just sound awesome. And they’ve dropped the prices so real people can now almost afford them. Cytomic’s “The Glue” Link – a really excellent analogue-modelled master bus processor that you just set and forget.

The Lucky 13 Song Mixing Tips

Before I get started I just want to reinforce something I’ve mentioned in earlier posts – sometimes a reduction in parameters actually generates more creativity. Being aware of a set of limitations, or guidelines, can actually allow you much more creative control over your final mix.This could mean limiting the amount of effects that you allow yourself to use, or a more obvious one is to only use a particular set of effects that suits the genre or style. If you have the permission to do it, perhaps editing tracks or even removing “surplus” instrumentation or vocal is the first step.

Approach-wise, ideally you want all aspects of a song to reinforce together and create a stronger impact, and if you aren’t aware of what you’re doing, it’s very possible (in fact more common than you think) to get a generally nice balance of instruments that somehow doesn’t “gel”. You can hear everything, but it lacks emotional impact.

So here’s a bunch of ideas to think about next time you’re mixing a song – there are many more ideas and concepts to experiment with than these, but I stopped myself before the post became a novel.

1  Know what the song’s about. Clues are in the lyrics. Knowing what it’s about gives you the opportunity to amplify the concept rather than inadvertently fighting it. That doesn’t mean you have to “follow” the lyrics with the mix in a literal sense – you might do nothing at all in that regard, but at least you won’t be fighting the meaning of the song without even realising it, and when it comes to trying to think of creative mix directions, it’s yet another clue to help you.

2  Know the context of the music. What’s the genre or style of the artist. How does it relate to the artist’s identity? Being aware of this really makes it much more likely that you’ll promote that artist’s identity and overall concept, plus the artist will be more likely to appreciate what you do with the mix. For example does the artist exemplify “authenticity” where a raw, “character” sound with any intonation problems remaining unfixed is most desirable? Or is it about slick and smooth production?

3  Be adventurous. A mix is not a simple balance of levels of the instruments in a mix, it’s about featuring various aspects that you think the listener would like to hear, or more accurately needs to hear at any given section of the song. Pretend it’s a movie – how do you present each section of the song? Don’t be scared to go “over the top” with effects, fader moves and featuring of mix aspects – you can always tone it back if need be. Don’t be scared to turn vocal up loud – trying to hide weak vocals makes it even worse. Even ugly actors have to have close-ups in a movie to make it all work.

4  Think about texture and tone. It’s partly tone, partly level, partly how dominant something is in the mix. If you compress something – its texture changes. Listen out for it tonally as a sound rather than just checking it’s variation in level. How pervasive is it compared to everything else, despite its volume in the mix?
How does it link into the overall texture of the song? Textures are like a tonal colour palette – you probably don’t want to mix a neon green element in with some nice earth tones (remember there are no rules!), but then again you don’t want everything the same shade of beige.

5 It’s about melody In even the most distortion-fest mixes, our human nature will use our built-in pattern-detecting algorithms to extract a melody out of it somewhere, whether it be in the movement of the harmonics in the wall of guitar noise or in the groovy bassline. Make sure there’s one dominant melody at any given instant, or if there’s more than one, that they aren’t fighting each other and canceling out.

6 The pocket. It’s more than something to put your wallet in. It’s that magic interaction of instruments when it all suddenly locks into a groove. Spend some time adjusting relative timing of instruments to see if you can help the groove “gel”. You’ll know when it happens because it’s magic and you’ll start moving with the music whether you want to or not. Note that Beat Detective and other forms of quantization can fight this effect – it’s “felt” rather than being on an exact grid. Saying that, if the playing is too loose then a timing grid is definitely a step up.

7  Keep it simple stupid. Less is more. These things are fundamental truths, despite our over-familiarity with them often leaving them as meaningless statements in our minds. Think about the mix as a photo – the more people you want to appear in the photo, the smaller they’ll have to be. Don’t be scared to bring the main things to the foreground, and push other things back to the point of blurriness or being hidden behind the main elements. A good mix is not about individual band members’ egos, it’s about the overall blend. When you think about it, the individual band members have the least idea about what the mix should sound like – they all hear completely different versions of a mix depending on where they stand/sit when they perform.

8  Three “Tracks”. Back in the olden days, after mono and stereo, there were three tracks. One was for “Rhythm” (and could include drums, bass, percussion and rhythm guitar for example), one for Vocals and one for “Sweetening” which might be things like brass, strings, lead instruments etc. This strategy is still a great one to keep in mind for mixing. It forces you to think about your rhythm section as one single thing, and you need to make it all gel. Bass needs to lock in the pocket with the kick drum. Sweetening nowadays is whatever else you need outside rhythm and vocals. Think carefully about which mix elements fit into each of these three roles, and if all three are already populated – maybe it’s time to do some cutting. Note that some instruments such as guitars might switch between modes depending on what they’re playing at the time – rhythm, fills or lead.

9 One thing at a time. Rather than thinking of one of the aforementioned three tracks as just “Vocals” perhaps it’s better to look on it as “Melody”. The melody line often chops and changes between vocal, instrumental fills and solos. If you think of these three elements as playing a similar role at different times in the song, it makes it easy when trying to decide on levels/sounds between the three. It also highlights that you shouldn’t have any of those melodies crossing over each other and fighting at any point – keep ’em separated!

10 Getting the bass sitting right is tricky – especially when it needs to work on both large and small speaker systems. Try mixing the bass while listening on the smallest speakers that you have, to get it sitting at the right level. Then adjust the tonal balance while listening on bigger speakers to reign any extreme frequencies back in. Sometimes you might need to layer the bass sound to get this to happen effectively.

11 Don’t over-compress everything. Listen to the TONE while compressing each instrument and keep it sounding natural if possible. Pay close attention to the start and end (attack and release) of the notes of each instrument you compress. Your final mix should be sitting at an average RMS level of about -12 to -18dBFS with peaks no higher than around -3dBFS. Leave the mastering engineer to do the final compression and limiting. Remember to leave dynamic range in the mix – contrast! Our ears need some sort of contrast to determine what’s loud and soft. If you hammer all the levels to the max you may as well just record the vacuum cleaner at close range and overdrive the mic/preamp. Hmmm. Might have to try that.

12 Easier than Automation. In these days of automation, it’s easy to spend inordinate amounts of time tweaking automation changes on instruments or vocals between different sections of a song (eg adding more reverb to the vocals in the chorus or adjusting rhythm gtr levels in the bridge). With today’s digital audio workstations, extra tracks are usually in ready supply, so rather than fluffing about with automation for a specific section of the song, why not just move that part over onto another duplicated track instead, then just make whatever changes you need to suit that section. Much quicker than continually mucking around with automation on the same track. By the way – make sure your mix is dynamic. A mix is a performance in itself, not a static set of levels.

13 Use submix busses for each element of the mix. Eg drum subgroup,  guitar subgroup, vocal subgroup etc. Rather than send all your drums straight to the L/R or Stereo mix, first send them all to an Aux return channel instead. Then send that Aux to the LR/Stereo mix. (Tip: disable solo on the Auxes) This makes it simple to do overall tweaks to your mix even after you’ve automated levels on individual tracks.
You need to be careful about aux effects returns and where they come back though, as their balance might change slightly if you adjust the instrument subgroups.
And hey, what about creating just three subgroups – Rhythm, Melody, Sweetening? Let me know if it works ;o)

Sources: Stephen Webber, Bob Katz, Mixerman, Mike Senior.

Relieving Threshold Shift (Temporary Hearing Loss) with Acupressure

This is a handy tip for those moments when you’ve gone to see a loud band and forgotten to take earplugs, and one that I’ve used numerous times to “reset” my ears after a gig. I was shown this trick about 20 years ago by a friend and have been using it since then, but in preparing this blog I’ve also found lots of supporting evidence on the web that reinforces the basic concept. It has definitely and audibly worked for me and others that I’ve shown it to and it really can’t hurt to try it. Actually it does hurt a bit when you find the right spot to press, and I also have to admit it looks a bit stupid when you’re doing it, so best not do it when actually walking out of the gig – at least wait till you’re in the car when nobody can see you.

Press and hold the area shown in the diagram – it’s in the hollow just in front of the ear lobe. If you press the right spot it will feel tender, and after a few minutes you should feel the “cotton-wool” feeling diminish and your hearing begin to return.

Threshold Shift, for those who don’t know, is muffled high-frequencies or pressure or ringing in your ears that you can feel as you’re walking out of a loud gig. This is extremely dangerous in the long-term, and has even more significance nowadays for long-term headphone or ear-bud use.

Long-term exposure to loud sounds:

What happens is that when loud noise is perceived by the brain, it attempts to protect your hearing by tightening the muscles inside the ear in order to reduce the amount of noise passing through the ear mechanism. A fantastic system really, but not designed for a lifetime of loud music or industrial noise.

This muscle constriction can also restrict blood flow to the inner ear, and if it happens repeatedly it can cause long-term damage to the nerve cells in the inner ear, which eventually end up dying. Fatally. As Motorhead almost said – “Killed by Deaf”.

Seriously – long-term noise exposure can cause permanent hearing damage.

This acupressure trick relieves the constriction of the muscles around and in your ear, and hence allows full blood flow again to the nerves in the ear, hopefully extending the life of your hearing a bit longer. Obviously it won’t suddenly reincarnate the dead nerve cells in your inner ear, but if used early and often enough it will hopefully at least minimise the damage somewhat. No guarantees of course.

* Threshold shift and associated long-term hearing damage are not the only cause of hearing damage. I have met people who have lost hearing with a single exposure to a loud impulse sound (someone pressing the wrong button on the mixing console and blasting maximum volume through the headphones, or a massively loud click through a PA system at a gig) as well as others who have ended up with tinnitus (ringing in the ears) which can last FOR THE REST OF YOUR LIFE. Apart from these problems there are other odd things that can happen related to your inner ear – for example upsetting your sense of balance. Not much fun – I had continuous vertigo for a few days when I had the nasty flu earlier last year – no laughing matter as I ride a motorcycle to work.

Noise vs Music – I’ve often pondered this as I’ve been assaulted by a band that sounds like crap – as long as you perceive the music as, well music, your brain isn’t trying to shut your ears down, but if the band sucks and it sounds like obnoxious noise they’re effectively killing your ears! Obvious solutions – drink more alcohol to thin the blood and keep that oxygen getting to the ear cells, or try to psych yourself into believing the band is awesome, thereby fooling your own brain.
My wife says “why don’t you just leave?”, but I view that as defeatist.

Factoid 1: Research disputes what I just said before. Studies have shown that musicians suffer as much hearing damage as those exposed to industrial noise of equivalent level. I argue that musicians aren’t ever just exposed to music they like (we usually all have to share gigs with other bands), or other loud noise, so it’s hard to prove this either way without adequate methods or controls.

Factoid 2: Published Acceptable Exposure Time vs Sound level graphs are based on industrial noise, not music. At 110dBA your acceptable daily exposure time is 1 min 30 seconds!

Other (more serious) solutions:
Obviously, considering all this, the best solution is to avoid loud sound or wear appropriate hearing protection. Go get some proper earplugs -the custom-moulded “musician’s” earplugs are pretty darn good- they’re relatively “flat” and uncolored but there are other slightly cheaper options as well (custom-fitted plugs can be quite expensive, but they last for years with careful use). The problem I’ve found with them is you can truly hear how out-of-tune the singer is when watching a live band which might ruin your enjoyment or “perception of talent” slightly, but I have to say I’ve been to gigs and wearing some -15dB custom plugs and my eardrums have still been distorting painfully at times. You can get -25dB or more reduction plugs, and some come with both inserts as options so you can swap them.

And finally an observation – isn’t it weird how society is au-fait with people wearing glasses to correct their vision, but wearing a hearing aid has a stigma attached to it. You see graphic artists, photographers, directors and numerous other industry professionals (who rely on visual acuity!) wearing glasses, but would you trust an audio engineer with a hearing aid? Hmmmmm.
Not that I need one YET, just paving the way for the future.

References:
Acupressure Points

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-to-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 (ie on your Master fader) between about -12 to -18dBFS, 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 – say “no” to normalizing. That’s almost 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 )

Transferring MIDI and Audio sessions from Logic to Pro Tools in about 5 minutes.


It’s pretty common to have to transfer a song written in Logic into Pro Tools for a client to mix (or remix). Here’s how to do it as fast as possible with the least amount of hassle.

Audio Files Only

If all you need to supply is audio files for transferring to Pro Tools (usually the most common requirement), it’s a very easy 5 steps (MIDI files are trickier – we’ll get to those later).
All files will start at the same point and be as long as they need to be.
Files won’t include any Bus/Aux effects, only what’s on each Channel Strip.
Files are PRE-fade (ie the equivalent of the fader being at 0.0), so they may be quite loud.

1. Name your Logic tracks intelligently (double click on the track header to give it a useful name – this is what your file will be named)

2. Make sure the length of your song is set to about the right length -ie not 200 bars if it’s only 20 bars long. It’s no biggie if you forget this one, but you’ll be sitting waiting for longer than you need to while waiting for the files to bounce.

3. Delete any unused tracks and/or mute unwanted regions.

4. Select menu File-Export-“All Tracks as Audio Files”.


5. Select Wave and 24 bit (unless something else is desired). Select Normalize “Overload Protection Only” (this is not your typical “normalize” function and will just make sure your Channel Strip level will never overload). Make sure you know where you’re bouncing to. The default is the “bounce” folder within same session. (You don’t have to enter any file name/s). Hit “Save”. All done.


Easy huh?

MIDI File Export

Exporting MIDI tracks as MIDI files is a bit fiddlier than creating audio bounces, as many of the processes in Logic such as region Quantise and Transpose are “real-time” and need to be rendered into the MIDI track itself before exporting as a Standard MIDI File.

Do this (assumes standard Logic key commands):

1. Select all MIDI regions you’re going to export as a file.

2. Press “Control N” (normalises any region parameters for the selected regions – eg Transpose).

3. Press “Control Q” (normalises any Quantize parameters for the selected regions).

4. Press “Control L” (turns any loops into copies).

5. Press “Shift =” (merges the copies and other regions into a solid file on each track).

6. Name each region with the text tool (you’ll thank me later).

7. Select menu; File-Export-“Selection as MIDI file”. Name your file (eg blah.mid), hit Save and you’re done.


Importing into Pro Tools

Now to bring these shiny new audio or MIDI files into Pro Tools.

The easiest way is to create a new, empty Pro Tools session, then drag your files directly from the “bounce” folder in Finder and drop them into the empty Edit window in Pro Tools. PT will now import the files and automatically create the appropriate track for each file.

Logic 9 – using Pedalboard in parallel mode for fat Bass and Guitar sounds

Click on the photo to enlarge.

A little while back I wrote a blog article about cool things to do with multi-band compressors

One of the things I discussed was how to use the crossovers built into one of these plug-ins to separate the lows and high frequencies of, for example, a Bass track, so that distortion could be added to the top-end of the Bass without robbing the fat bottom end.

Well now with Logic 9’s new Pedalboard, you can easily add some grainy distortion to the Bass track without thinning the sound by using the distortion pedals inserted in parallel mode.

Pedalboard is a great new plug-in that has been added to the latest version of Logic, and includes some great-sounding pedals that can be custom-assembled into complete pedalboards. (You can even map individual pedals to controllers with built-in macros, but we won’t cover that in this article)

By dragging, for example, a Distortion pedal from the selection box on the right into the main pedalboard, then adding a Splitter pedal, you can then click on the name above the Distortion pedal to toggle it between series and parallel modes.

Series means the whole Bass sound goes through the distortion pedal, parallel means the distortion pedal is blended with the original dry Bass sound.

What’s even better is you can switch the Splitter pedal into “Freq” (Frequency) mode. This allows you to select what range of frequencies goes into to the parallel chain. In my example, I’ve set it to send from 1.5kHz upwards. (Hint: to see this exact value, I temporarily switched the plug-in “View” from “Editor” to “Controls”).

When you insert a Splitter pedal, it automatically inserts another Mixer pedal at the end of the chain so you can blend the two parallel paths back together again, in whatever proportion you desire.

Here’s another tip – if you’ve recorded your electric guitar straight into Logic via your audio interface and are then adding effects in Logic – try using the parallel mode to blend your clean electric guitar with the distorted version on the other side of the parallel chain. This can give your wall of distorted guitars some extra clarity.