My last experiment with building a MIDI controller, about a year ago, was to see if there’s any benefit in using larger knobs for MIDI controllers.
I’ve always hated those tiny little plasticy cheap-feeling knobs packed close together that seem to be used on most MIDI controllers. I’ve found that depending on where you’re sitting or standing, it sometimes takes two grabs to make a complete revolution on them, with a pause in the middle, and it’s difficult not to bump the surrounding knobs as well. It can make for some frustrating moments when you’re trying to map each control to something in your DAW.
Plus smaller knobs give poor resolution and positional feedback to the user when turning, compared to larger ones. I also can’t help but feel that those huge vintage-looking knobs add part of the charm to vintage audio equipment.
It was pretty good actually – those huge knobs gave a silky-smooth and super-accurate response when controlling parameters in Logic Pro X.
So I worked with it for a while, but to be honest they were almost a little TOO large, and I wanted a couple more of them on the panel to get more convenient mapping to Logic’s Smart Controls. Additionally, I wanted some buttons on there for triggering drums and sampler sounds, plus a few old-school toggle switches just in case.
So – off to Jaycar once more to buy another load of second-to-largest knobs in stock and few other components that I needed for it. In the end I ordered a pre-assembled RGB Button-Pad from Livid – they were on sale that week and I couldn’t be bothered fiddling around too much constructing my own from scratch.
So my second version of the MIDI controller came together. I also decided to retain a single large knob on the right of the unit for Logic’s Automation Quick Access (AQA) function – basically for recording and editing automation in my tracks.
Here’s the second prototype:
I also put together a more vintage-looking wooden surround for it. As you can see I went for a vintage “distressed” look. Actually – this was only semi-deliberate, since I was missing some essential metal-and-woodworking tools in my workshop. Apparently the Renovation Tool is NOT perfect for everything.
The wiring was relatively simple, although it looks suitably impressive in the photos.
There was a bit of soldering involved of course, but each pot can be re-mapped to a different MIDI controller number via Livid’s Brain Configure tool if necessary. In my case it was, as I had plugged the ribbon cable headers in the wrong order in my excitement to try it out. It was much simpler and faster (a few seconds) to remap them in my studio than to remove the unit , open it up and then switch the plugs.
Despite the rough edges, the constructed unit is good enough to get an idea of how the final unit will be constructed, and everything works fine. It looks great in my home studio and it adds a nice degree of analogue vibe even though it’s digital.
My next goal is to get hold of some decent woodworking tools to make a tidier version of the case, and maybe brand my own logo on there somewhere.
One of the cool tricks you can do with Logic’s Drummer regions is to drag out an alias of the Drummer region to another software instrument track. (Drag a Drummer region with mouse while holding Shift-Option).
Aliases are virtual regions with no content of their own – they just follow another region’s content (although you can still do stuff to them like transpose them etc).
This new software instrument track can be another Drum Kit instrument, or a Drum Machine, sampler or even a third-party drum instrument like Slate’s SSD drum sampler.
Now your new instrument track (via the alias) will play exactly the same thing as the Drummer pattern. Even if you go back and tweak Drummer the alias will still follow it. And if you mute the Drummer region, the alias still works, and you will continue to hear the Drummer pattern through your new drum instrument.
That’s pretty cool, but what if you don’t want to layer the entire kit – perhaps just the kick or the snare?
That’s easily done too;
On the new alias instrument track, go to its Track Inspector pane. It’s the second box down in the Inspector window on the left. It’s usually hidden, so you might have to click the little disclosure triangle to pop it down.
Now you should see a “Key Limit” line, with something like “C-2 G8” in it. These are the low and high key limits, and it means that this track will currently accept MIDI notes over the full range of possible MIDI notes from C-2 up to G8.
If you only want to trigger the kick, double click the “C-2 G8” and type in “C1”. You should see two C1’s – meaning only this one MIDI note will now be accepted. You should only hear the kick drum.
And if you want a layered snare as well?
With the same instrument; Create another Virtual Track going to the same instrument; Menu: Track/Other/New with Same Channel
Drag a copy of the alias to this track. (Option-drag alias with mouse)
On this track, double click the Key Limit numbers and type in “D1”. This will now only accept the snare MIDI note.
With another instrument; Create another software instrument track and dial up a drum patch.
Drag a copy of the alias to this track. (Option-drag with mouse)
On this track, double click the Key Limit numbers and type in “D1”. This will now only accept the snare MIDI note.
As you’ve probably figured out by now, you can carry on and layer as many extra kicks and snares on different drum instruments as you feel like.
Bonus tip for handy kick and snare layers:
Load up an instance of an EXS24 in a new instrument track. (In the Patch Library select Legacy/Logic/Logic Instruments/EXS24)
Click the EXS24 slot on the channel strip to open the EXS24 front panel up.
In the little panel above the Cutoff Knob, click and select Factory/Drums & Percussion/Single Drums/Kicks/Layer Kicks/Body Kick C1 1. If you click the little “+” symbol to the right of the panel you can step through each sample in turn.
As you can see there’s a whole bunch of “body” and “transient” kicks (and snares) that can be used to layer your existing kicks (and snares). Some of these sound great, although the “body” kicks sound unusual by themselves as they’ve had the transient part trimmed off the front.
Drag your Drummer alias onto one of these EXS24 tracks and set the Key Limits for the track as explained above and you’re away laughing.
My glory years of audio engineering were right at that transition between analogue and digital formats. My battles seemed to be between the harsh discipline of digital and the casual looseness of analogue. I became a bit of a cultist in league of getting that “analogue sound” on digital gear. And the trouble with cults, of course, is that you stop questioning things.
In my case I became so fixated upon just keeping my recording levels down (firm look/which is generally a good thing by the way! /firm look), that I forgot that with analogue-modeled plugins, overloading the input is the same as overloading the analogue unit itself. That means if you drive it hard, that instead of the usual shitty digital clipping events, you get a nice analogue harmonic-related distortion. Which means “musically-related” and pleasant. Anyone who’s ever seen a true analogue-based recording session (ie on actual tape) knows that those tape-recorder VU meters are usually pinned to the top. For some reason, our ears love the sound of harmonic distortion added to the original signal. It sounds more exciting and vibrant, while at the same time smooth and mellow. I think we will all agree that this can only be the perfect combination of anything.
So, as you’ve probably guessed, I was holding back from driving the inputs of my “analogue” digital plug-ins, as I had trained myself to avoid digital clipping. And this led to a pretty average impression of many of the cool UAD analogue-modeled plugins I had. Sure, they still sounded better than the average “digital” plug-in – but it was often subtle – just a hint of smooth silky velvet and random pleasant “character”.
Then I tried out the new Manley Variable Mu Limiter Compressor. I don’t know about you, but I can’t help but feel that there’s something inherently cool about the word “Mu”. Especially “Variable Mu”. Brrrr.
So I dialled up a few Manley (Variable Mu – sorry just had to say it again) patches here and there – “yeah I can hear it working – nice… nice”. Dialled up the patch “GG-The Fast Tickle”. Yeah beauty… nice. Then out of some crazy impulse (yeah I’m like that – completely unpredictable), I wind the HELL out of that input knob. I know – radical right? But, you know, I finally got the hug (well the song did) that EveAnna Manley promised us with this unit. It really does just grab everything and bring it together like a big group hug. I don’t know about you, but that sounds… awesome.
Izotope Ozone is a multi-tool mastering stand-alone application and multi-format plug-in. Version 6 has some significant changes over previous versions including a different interface. It comes in plain and Advanced versions. Ozone 6: $249 USD, Ozone 6 Advanced: $999 USD.
Good: Great sound, easier to use, better look and feel, some cool new features and more flexibility in multi-band modes. New Dynamic EQ in Advanced version. Bad: Removed “Amount” sliders. Advanced version is overpriced.
8 out of 10 – Worth buying/upgrading.
Izotope’s Ozone has become quite a stalwart of the low-to-mid-budget mastering scene over recent years. One of its key advantages is having an entire suite of mastering-ready tools condensed into a single mastering plug-in.
My own favourite reason for using Ozone over other mastering products was the cool trick of being able to hold down the Alt/Option key and then click in the EQ display with the mouse to solo a particular frequency. This made finding troublesome resonant frequencies ridiculously simple, fast and intuitive, and even when I moved into using other mastering tools, I still preferred Ozone’s EQ section for my precision-repair EQ’ing purposes. In the very-expensive Advanced version, each processing module also comes as a separate plug-in, so working with other mastering plug-ins is easy. I’m very glad to see this feature has not vanished in Ozone 6.
On the surface, not a lot different that it was before, but laid out a lot better. There’s eight EQ bands that can be switched between high pass/low pass/bell/high and low shelf and Baxandall (new) filter types, and as usual an excellent spectrum analyzer that lets you see a detailed view of what’s going on in the mix.
There’s a global button for Digital and Analogue EQ modes. Analogue mode includes the phase-shifts from real EQ types that add their own phase responses, and you can even select different types of filters depending on their type – including Analog, Vintage, Baxandall, Brickwall, Proportional-Q, Band-shelf and Resonant modes. If you want to see exactly what’s going on with phase, there’s also an optional display of Phase Delay, Phase Response and Group Delay.
If analogue colour is not your thing, then the Digital mode avoids it almost completely. You can switch into “Surgical” mode for super-accurate editing of EQ. But then you can still tweak the phase response for each band with a little “phase” slider.
And as with previous versions of Ozone, there is a “Matching” button for matching an EQ sourced from another track, or even against Pink Noise or 6dB Slopes.
One of the things I immediately noticed was missing in Ozone 6 was the “Amount” sliders. In Ozone 5, these handy sliders scaled the processing amount for each section (and globally). This was a convenient way to correct some heavy-handed tweaking after spending too long in the mastering zone, and provided much more detailed feedback on exactly how much processing was needed than the simpler “Bypass” button. I can’t believe this awesomely useful feature was removed – it seems like such a major step backwards. The Parallel and master Gain sliders in the dynamics section do go some way towards matching the features but nowhere near enough.
Finally, the multi-band dynamics, exciter and imager sections can each have an independent number of bands and different crossover frequencies. And even different types of crossover as well – analogue, digital or hybrid.
This is great news – it’s often been hard to find an ideal crossover point that suits all three processors at the same time.
Actually it’s often difficult to find the ideal crossover points at all – most people I know don’t even touch the preset ones.
Ozone 6 has a added a new “Learn” feature to each multi-band section that can automatically set the crossover points based on the audio itself. it’s certainly fun watching the crossovers whizzing all over the place as they detect what’s going on in the audio spectrum, and they seem to settle not far away from where I’d put them myself. Nice!
As with Ozone 5, the new version has M/S processing in its EQ, Dynamics, Exciter and also in the (Advanced-version-only) Dynamic EQ sections. Mid-side processing has become quite common in mastering workflows, and allows the center of the mix to be treated separately from the outsides. This gives increased control over things like vocals vs guitars.
I notice that the dithering algorithm choices have been reduced from three to just one; MBIT+. This makes sense, as most people probably just used the default setting (if at all!). One nice addition here is being able to see the dither noise-shaping curve itself.
The Mastering Reverb has gone in Ozone 6 as well. I’ve only ever used this twice in all my years of mastering with Ozone, and I’m guessing it caused problems with amateurs adding reverb to mixes they shouldn’t, so it’s probably safer to not have it.
Look and Feel
Looking at the overall look and feel of the new Ozone 6, it appears that they have adopted some of the design simplifications that have been applied to other Izotope products such as Alloy. I think these changes are generally an improvement – everything feels much more transparent and a bit simpler to use. There’s less visual confusion, and the simpler more 2-D look actually works well.
Processing Module Section and Browser
I really like the new modular tab along the bottom. Various processing modules can be added and removed and the order of them changed by simply dragging them around. This sort of functionality was present back in version 5 to some extent (it was called “Graph”), but it was significantly more limited than this new version. It’s simple to use. Being able to move the multi-band sections around as easily as the single-band sections is a big plus.
One of the cool new things with Ozone 6 is that it is not just a plug-in, it is also a standalone app that can import audio files directly.
(Although not mp4 MPEG2 file formats which is a shame – it will ironically open poorer-quality mp3s though). It would be good to see 32-bit float file formats supported as well.
You can open multiple files as a project and have different Ozone settings on each file, as well as reshuffle the track order, and do bulk exports. Although each track has handles for trim and fades, it doesn’t allow you a full CD-layout with associated song overlaps and index markers etc. Each file still remains individual, so if you’re doing CD production you’ll still need a specialist application that can create a CD playlist such as Apple WaveBurner, Sony Sound Forge, Steinberg Wavelab, or DSP-Quattro.
As a real bonus, you can also insert other plug-ins (VST or AU) within the standalone app.
Another new thing in Ozone 6 is the Dynamic EQ (in the Advanced version only). I’m somewhat familiar with the Brainworx Dynamic EQ and this works in similar fashion – although the Ozone one is a little simpler. You set an EQ and adjust the trigger threshold and it variably boosts or cuts that frequency band depending on its audio level and depending on whether you applied a boost or a cut. You can flip the response the other way as well (ie it will cut instead of boost or vice-versa). It’s pretty cool, and in many ways is a good substitute for a single-frequency-band treatment using Ozone’s multi-dynamics section. Great for rumbly sound-hole acoustic guitar tracks.
As with Ozone 5. the Advanced version comes with the Insight plug-in. This is an advanced metering plug-in that not only gives several more metering options, including LUFS and other TV and film-based loudness meters, but also allows collective metering of multiple tracks in your recording session.
Overall this is a good offering from Izotope. The sound quality is superb as expected, and the new and expanded features are excellent. Almost anyone should be able to pick up and use this plug-in, and the fact that it can be used as a standalone mastering processor will tilt the balance for many people.
I’m still a little gobsmacked about the lack of “Amount” sliders, but the other great features help balance this loss out. Maybe they’ll add them back in in a future release. We can only hope.
Regardless, upgrading to Izotope 6 would generally be a good value decision.
On the other hand the Advanced version would take more careful consideration.
It’s a shame the “Advanced” version is so ridiculously expensive. At $999 USD (that’s nearly $1,300 where I live) it’s FOUR TIMES the price of the vanilla version.
That puts it well out of the reach of most casual users and small studio owners, and although I agree most of the missing features would genuinely be considered “pro” it’s such an artificial and arbitrary method of getting extra money out of users, and consumers see this sort of thing as a bit of a rort – a poor pricing strategy that can eventually have fallout on a brand. Personally, I would be tempted to just invest the difference in price into other nice mastering plug-ins from other reputable suppliers like UAD or Slate. Izotope seriously needs to look at their pricing strategies in this regard.
I’m surprised that so many music sites insist on using mp3 format for their files despite the generally poor quality – even Soundcloud uses only 128kbs mp3 – when AAC is a patently superior file format at that same file size/bandwidth. Most noticeable is the retention of transients (eg drum beats) and much better top-end response. AAC sounds almost like CD-quality.
mp3 is generally MPEG 1 (layer 3). They did extend it slightly in MPEG 2 to add some lower sample rates and some different channel formats.
AAC (Advanced Audio Coding) is MPEG 2 and also in MPEG 4.
mp3 only uses 576 blocks to encode audio.
AAC uses 960 or 1024 blocks to encode audio.
Note: MPEG stand for Motion Picture Experts Group – the standard-setting body for this sort of stuff.
There’s a huge difference in quality between the formats – mainly because they tweaked the hell out of the encoding algorithm after the original mp3, so 128kbps AAC sounds almost like CD-quality, whereas mp3 128kbps sounds phasey and dull.
AAC is also what’s generally used nowadays for radio broadcast – they call it MP2 though, to confuse the production people.
So what about MPEG 3 and 4 then?
MPEG 3 only had a couple of tweaks to add to the standard, so rather than having a completely new standard, they just updated good ol’ MPEG 2 instead. They added things like more channels for surround-sound files and the like.
MPEG 4 was mainly about metadata (information embedded in the files) and even more channels – basically bundling a variety of video and audio formats (and other stuff like subtitles) into a handy container called “MP4”.
If there’s only audio inside it, it gets called M4A. Though some M4A files can also contain Apple Lossless format at master quality, but at a larger file size. Lossless formats are more like zip files – they’re compressed down to smaller size (not as small as mp3 or AAC) but can be expanded again without any loss.
What’s this bit-rate thing?
Since most of these lossy formats were designed to be streamed – either on the internet or off an optical disc, they’re measured in how many bits per second they need.
Is that the same as sample-rate?
No – mp3s and AAC files still retain the original sample-rate and bit-depth as the original file. At lower bitrates, obviously more information needs to be trimmed out of the audio to make it smaller – hence the quality loss.
So why do we still use mp3 rather than AAC?
Well, if you’re using Apple products, you’re likely using AAC way more than you know. Otherwise, mp3 has just been around longer and more software can be guaranteed to play mp3s than AAC (although it’s pretty rare that something can’t play both now). It’s like the old VHS vs Betamax video tape format thing all over again – the lowest common denominator (and cheapest format) usually wins against quality.
So what should you use?
If there’s a choice – go with AAC. If not, go with mp3. Simple.
One of the cool things you can do so easily in a DAW (Digital Audio Workstation) is slip entire tracks, or actually the regions within it, to the left (earlier in time) or to the right (later in time).
What’s so cool about this?
There’s a couple of things you can do with it.
1. Correcting Microphone Delays
One fairly obvious example is tightening up drum kits. It’s pretty common to use room mics when recording a drum kit. Let’s say your room mics are 3m (9.8feet) from the snare mic on the kit. That’s about an extra 8 milliseconds, or (@44.1kHz sample rate) 353 samples.
If you slid your room mic regions 8ms/353 samples to the left, the recorded signal would coincide perfectly with the snare mic – ie there would be no delay between the mics.* Note that you’ll need to have all your tracks starting a little bit later than 0′:00″ or Bar 1 so you have some “left” to go to.
You don’t have to pull the room mics all the way back though – you might just want to pull them a bit closer in time to tighten up the room sound.
2. Creating a Pocket/Helping the groove
This is where you can subtly shift the timing of regions to help the overall groove of the track.
For example, many bass players get a little excited when recording, and can play a little ahead of, or right on top of the drum beat. Although it’s still “in time”, sometimes delaying the bass very slightly can make it “groove” more with the drums.
In this case you would incrementally slip the bass region/s to the right until the groove feels better. The kick drum often masks the attack of the bass to a certain extent, so it can also clean up the kick/bass combination.
It can help to think of being “in-time” as a window rather than a vertical line, and you can be at one edge or the other of that window and still be in time, but get some huge changes in “feel”.
How to do it
There’s a couple of methods.
Shifting the actual region itself using “nudge”. In Logic or Pro Tools you can incrementally bump something in either direction by using the nudge keys. You can set the nudge value to pretty much anything -samples, beats etc. The dangers of this method are if you give the files to someone else to mix they may not realise you’ve moved track regions away from the same start position. Keep good notes!
Region Delay in Logic Pro
In the region inspector (under “more”) there is a parameter for Delay. You can set it earlier (- values) or later (+ values). The handy thing about this way of doing it is that you can split each track into different regions for chorus/verse etc, which can each have a different delay value.
Using delays. This is the old-school way of doing it. Insert a delay plug-in (with the same delay value eg 500ms) on every track. If you want to make a track play earlier – lower the delay for that track. If you want it later – make the delay longer. The beauty of this method is that it automatically keeps a record of what you’ve done by saving the plugin settings.
*Phase problems. Watch out for this. Once you get down to a few samples difference between different microphones on the same instrument (eg the drum kit), you’re potentially going to get phase issues. Sometimes even a few samples can make a difference. The wavelength of around 2cm (almost an inch) is 13.5kHz, so that means moving the microphone “virtually” even that much can make a big difference. This is where phase “rotation” plugins can be handy – such as UAD’s Little Labs IBP Phase Alignment Tool.
It’s no secret I’m a big fan of Universal Audio and their products – I have my Apollo Quad and a selection of my favourite UAD go-to plugins.
Coming from a lifetime of mainly indie recording and freelance work, I’ve managed to avoid working more than occasionally in expensive studios with swags of outboard equipment. So I’ve always had expensive tastes but only depressingly-low indie budgets, making the availability of rare and expensive vintage equipment well out of my price range.
But the alternative – virtually-modeled vintage hardware – has always lacked some part of the magic of those original analogue versions. Until lately. Now increased computer processing power and more detailed and accurate non-linear modelling processes have allowed more of that magic to be emulated in software, and also processed in real-time. Waves, Slate Digital and others have done wonders with the quality of the software models. Universal Audio started off way back in the day making analogue gear, and then relatively recently moved into the digital world with their UAD side of things. Despite the move into digital, their ethos has remained based on professional-audio quality. And by hosting those resource-hungry plugins on some external processor chips, it also allowed for higher-quality modeling and/or more plugins to be used.
When I first heard the UAD plugins, I was sold. It was the first time I had heard plugins approach the original sound I was used to. In my case it was the Studer A800 plugin, which felt remarkably close to what I’d been using for years. Was it exactly the same? I didn’t have the opportunity to A/B between them, but it “felt” and sounded the same as I remembered. It did the same things to the recorded tracks that I used the original for – the same colouration and compression I wanted for certain instruments. Except I could also turn off the tape hiss if I wished, and eliminate the wow and flutter. So the models not only recreate an original unit pretty closely, “warts and all”, but also allow you to disable some of the inevitable (and at the time really annoying) down-sides of vintage analogue technology. Or they might include some extra features that enable some easier practical uses, like having a mix control. By the way, they also allow a budding engineer/producer to get a feel for the vintage interface before they ever run into a real one in the studio. So – I have been relatively spoiled by my UAD plugin collection.
When I had to upgrade my cheapo portable (OSX Mountain-Lion unsupported) Presonus Firestudio a few years back, I sprung for a UAD Apollo Quad. It’s been great. I bought the Thunderbolt option and never looked back.
So it was quite a pleasant surprise to have this v7.5 update not only add a few extra plugins to the collection, but also to add extra functionality to my Apollo hardware. It’s a bit of an “Easter-Egg” really – apparently the functionality has always been there. The new preamp plugins directly access the preamp chips on the Apollo board and can adjust not only the gain, but also the impedance. This means that you get that much closer to the authentic sound of the real preamps.
You get a choice of three preamps – the UA610-A, UA610-B, and the API Vision channel strip. These slot into a new “Preamp” slot in the Apollo Virtual Console. The beauty of this is that you can then record through them as if it is real hardware.
How do they sound? Well – remarkably good. I tried various mics (Ribbon/Condenser/Dynamic) through each preamp type (even my gritty little Shure Green Bullet), and the difference in character and smoothness of each preamp was quite noticeable. The quite neutral preamps built into the Apollo itself allow the plugin character to feel pretty genuine, and the impedance selection just nails it. The harmonic overtones when driving them hard felt completely analogue.
I did have some initial issues figuring out how to match gains when switching between the built-in Apollo preamps and the plugins (they have their own gain and level controls) for comparison, but once I sorted that out, there’s no way I’d ever go back to NOT using one of the new preamps. They all added some level of subtle musical character to the microphone, and the hardest bit was choosing which one was the most suitable for each mic or thing I was recording.
My favourite is the 610-A at the moment, but that’ll probably change for my next recording.
For those that recently jumped into purchasing Apple’s latest version of Logic Pro X, there may have been a few nasty surprises with older 32-bit plug-ins not working.
That’s because Apple finally dropped support (as it seems to do on a fairly regular basis) for aging 32-bit plugins and instruments. Actually, they just removed the 32-bit plugin wrapper which Logic 9 had. The wrapper was pretty clunky and annoying anyway – it continually stole focus from Logic when you needed to open a 32-bit plugin’s graphical interface.
The lack of Logic’s 32-bit wrapper means that third-party plugin providers would now need to update all their plug-ins to 64-bit, or they simply wouldn’t show up in Logic. Unfortunately, the outcome of this strategy is that it can take ages for manufacturers to get around to updating all their products, especially the older ones that may not have been coded very tidily in the first place. In fact some manufacturers had already dropped support for older plugs and instruments, so the likelihood of getting a shiny new 64-bit version of some products is pretty much nil.
That’s a bit sad when you want to open up a song you created only a couple of years ago (because you might not have finished it yet), and instruments and/or plugins are missing. It can completely change the track – it may even remove the main hook sound that the track was built around.
Even with instruments and plugins that have actually been updated to 64-bit, there are issues. Some newly- ported 64-bit plug-ins do not open up their original patches anymore. This can be phenomenally annoying. Imagine your joy at finally being able to open up your old song in Logic to do some more work on it, only to find that the instruments or plugins you used have been restored BUT they are restored to completely different sounds or settings. It’s tedious enough trawling through hundreds of presets trying to locate the one you used for your song, but it’s way worse if not impossible to recall a patch if you actually edited it as well – it means you may never get that exact same sound back.
Of course you could simply keep an older version of Logic on your computer to open these older songs, and sometimes that works, but even then it’s not always that simple. Installing later versions of some products pretty much destroys the previous version’s patch library, so even opening in Logic 9 doesn’t get them back. Can you feel my simmering anger?
But look – there’s a shining star on the horizon! Sound Radix – a company with some quite interesting plugins themselves, observed the problem everyone was having with lack of 32-bit support in Logic Pro X and came up with a great solution. What they did was create a very tidy wrapper for your 32-bit plugins. Due to the quirks of some older products, not everything was supported at first, and 32 Lives is ostensibly a “beta” product (it’s just reached release candidate recently). Regular updates were released to make even more 32-bit plugins compatible. I found that even some of the plugins that weren’t on their compatibility list seemed to convert and operate just fine.
So how does it work? The application scans all your installed AU plugins and comes up with a list of the 32-bit versions. You can select which ones you wish to wrap, and away it goes wrapping them. It creates 64-bit versions of each of these plugins. Then when you open Logic Pro X (or Logic 9 in 64-bit mode), Logic scans the “new” plugins. I found this to be the most tedious part of the process, and Logic’s scanning process actually seemed to reject a bunch of the wrapped 32-bit plugins when it completed. I think there are issues with the way Logic scans multiple plugins at the same time on different CPU threads, so if one instance of Auval (Logic’s AU plugin-scanner) or 32 Lives crashes during the scanning process, it can affect other threads. I found that selecting each rejected plugin/instrument manually in AU Inspector in Logic seemed to add the rejected ones just fine in most cases. (Note – this scanning issue has all been sorted out in subsequent releases of 32 Lives)
The wrapped plugins and instruments now appear just like 64-bit versions – but there’s no annoying visible wrapper like in Logic 9. The patches you originally used are also restored. It just works. Transparently.
I spent several hours opening up old Logic sessions from years ago and had no problems opening them up and playing them. Actually my only issues were with really old versions of Native Instruments products, but that’s another story.
So – I have to say I was pretty darn happy with this purchase. Good on you, Sound Radix!
It continually surprises me (yes I’m in a perpetual state of surprise) how many Logic users don’t use (or even know about) the Marquee tool. Then again, I remember the first time I tried using the Marquee tool, and all it did was annoy me since I didn’t really know how to use it, so perhaps that’s what’s happened to everyone else as well.
Well, fear no longer people, here’s the tricks to some happy Marquee Tool use in Logic.
Also – one of the common whinges from Pro Tools users is Logic’s lack of a “Smart Tool” mode – where various tools are automatically selected by their position on a region. This function is always partly-available in Logic anyway (just trim & loop), but you can easily add a couple more tools to the palette by following the next screen shot. This works in Logic 9 as well.
As you can see, I prefer to assign my right mouse button as a tool, and just use Ctrl-left mouse click to bring up any menu items I need. That way I have even more edit tools rapidly available.
By selecting these options above, the good ol’ Pointer tool will now automatically become the Fade tool on the TOP right and left of a region (click again and drag up and down to change the shape), and the Marquee tool when in the BOTTOM-half of a region. This is in addition to the traditional Pointer tool also being the Trim tool on BOTTOM left/right of a region, and Loop tool on MID-right of a region.
I recommend that you try these settings and persevere for long enough to get used to them, as it will speed up your editing, and it helps when jumping between Logic and Pro Tools. It does take a little more precision and decent zoom settings but the pay-offs are worth it.
It’s at this point in the article that I rediscovered the annoyances of trying to get decent screenshots of Logic’s tool cursors. You can’t do it with the usual Apple shortcuts or even the Grab utility. The cursor doesn’t show up as other than the usual boring cursor (Grab lets you choose a few other cursor options, but certainly not any of the Logic tool cursors).
SO – I made you a short video instead, using Screenflow. I show the “smart-tool” setup and use, then how to use the Marquee tool.