Universal Audio has long had a successful business selling hardware DSP effects, many of them carefully-modeling classic analog gear. These products use dedicated DSP hardware for number-crunching, requiring that you connect an extra box to your computer. UA has certainly had their loyalists, and for fans of the products, the dedicated gear is simply a convenient way to get all of these sound-processing goodies. But it’s fair to ask the question, as many producers have who read this site, what’s the advantage? Why not simply use native processing on your computer?

Apollo, UA’s new hardware, answers that question more emphatically. By integrating the processing prowess of the UA platform into a high-quality audio interface, you can now add UA effects live, as you record and mix, with extreme low latencies. UA reports latencies below a couple of milliseconds. That’s possible, theoretically, on a desktop computer, but not generally on a laptop and very often not with any real reliability. You can do it in a lab, but it’s not something typical users see.

So, in one box, you effectively get your whole studio: the audio interface, the DSP power, and real low-latency sound processing. It’s not the first audio interface with DSP, but it might be the most compelling case yet for why that combination make sense.

And here’s where things get interesting: via Thunderbolt, a single MacBook Air, costing just around $1000, could be your whole studio machine. And while Apollo runs a couple grand above that, that means the total price tag is stunningly low compared to what you’d pay just a short time ago.

UA briefed me earlier this week on the technology. Even as NAMM raves about iPads, you begin to see the real power of conventional computers. Steve Jobs once compared those computers to “trucks” – while quietly leading a company that profits on how cool trucks are, too. With an Air, adding only slightly to the weight of an iPad and at only twice the cost, you can connect to vastly greater native processing power, greater outboard processing power, and greater I/O. And now with Thunderbolt, you could connect a high-res display or two, a big, fast hard drive, and the audio interface, all without running out of power or impacting performance. (No, seriously – you can. The reason you haven’t seen this in action is that we haven’t had the hardware to show it off. Apollo will be a compelling case for that.)

All of this is academic until you actually have something to do with sound. So, UA is also expanding their developer platform to additional outside development; more on that soon.

Apollo isn’t for everyone; obviously, some people won’t like being tied to hardware, and native plug-ins do work for a lot of people. But it does solve problems for many potential producer customers by making something reliable, predictable, low-latency, extensible with lots of excellent processing tools, and all in one single-box solution.

Apollo will initially be Mac-only, but will come to Windows, too – and with more PCs supporting Thunderbolt in 2012, that means the MacBook is far from your only choice. So, you’ve got one add-on that’s your interface, your pres, and your mix/master/effect toolbox.

More specs:

• 18 x 24 FireWire/Thunderbolt-ready audio interface, 24-bit/192 kHz
• “Premium” mic pres – UA stresses that they’re also building on their mic pre reputation, and they claim the “lowest THD and highest dynamic range” in their class
• Dedicated front-panel controls: preamp gain, channel selection, mic pad, +48V phantom power, low cut, monitor level, and dual headphone controls.
• 4 digitally-controlled analog mic preamps, 8 balanced line inputs and outputs, dual front-panel JFET DIs, digitally-controlled analog monitor outputs, 8 channels of ADAT, 2 channels of S/PDIF, word clock I/O, FireWire 800 (standard), and a Thunderbolt expansion bay — making it a well-equipped centerpiece for the modern project studio.
• Core Audio drivers; ASIO coming, so you can use this with your DAW of choice
• Console application and plug-in for recalling all your interface and plug-in settings at once
• UAD-2 acceleration
• Analog emulation plug-ins from Ampex, Lexicon, Manley, Neve, Roland, SSL, Studer, etc.
• Thunderbolt will be available on a sold-separately Option Card; UA says it reduces latency and audio buffer sizes, improves high sample-rate performance, and allows greater UAD plug-in instances over FireWire.

Of course, because Thunderbolt also connects to FireWire devices, you don’t lose your FireWire investment. The only bad news is that you only get Thunderbolt here as an Option Card; I imagine we’ll eventually see UA ship Thunderbolt connections standard.

There are both two-core and four-core versions, powered by Analog Devices SHARC processors, running an estimated street of US$1999 and $2499, respectively. Apollo’s Thunderbolt Option Card will be shipping in the first half of 2012, with pricing TBD.

www.uaudio.com/apollo

Videos are available on the UA blog: http://www.uaudio.com/blog/apollo-intro-video

Windows 7 summer; 10.6 and 10.7 Mac OS X when it ships.

Software Images

Let’s get straight to it: Ozone has already established itself as a do-everything mastering tool. It’s a suite of interconnected modules handling frequency and dynamics, designed to work together in an integrated interface. It does so much, in fact, that it’s hard for an upgrade to do more, but Ozone 5 promises new sound and visual feedback that could further entrench this popular tool.

And that could explain how Ozone 5 stole the Audio Engineering Society trade show in New York. AES is a flurry of knobs, dials, and faders, but some of the major buzz we heard was just this single upgrade to the software. (CDM’s Marsha Vdovin was out on the floor, and the word “Ozone” kept cropping up.)

Ozone is eminently visual software, so a lot of what’s new you can glean just by looking through the screenshots. But there are sound improvements, as well, both in the standard Ozone and the spendier “Advanced” edition.

What’s new:

• Updated modules. iZotope says they’ve “refined” their DSP algorithms. (Let’s see, carry the one…) The idea is, existing modules should sound better. There’s a detailed list on the iZotope site – aside from more subtle changes, you’ll find very specific adjustments to how parameters are controlled and how they impact the sound. To give one example, there’s a …
• New Limiter. The latest version of iZotope’s “psychoacoustics-based” limiter in the Advanced edition has a new stereo link control for handling left and right separately or together, and new intelligent transient handling algorithms, among other improvements.
• Enhanced EQ. Analog-matching EQ models analog shelf modes and frequency response, matching is easier than before, as with other modules, you can use left/right separately, and now zoom and display stereo info in your spectrum. There’s also new variable-phase functionality.
• New Reverb. Yes, sometimes you use reverb when mastering. (A little light reverb can do wonders.) A new modeled reverb algorithm adds new models and spaces and gives you unique early reflection control, as well as “cross-mix” for stereo imaging.
• New UI, workflow. I’ll let you just see what this looks like, but suffice to say parameters and labels are better-organized to be friendlier to advanced and beginning users alike. Past versions of Ozone were sometimes pretty-but-counterintuitive; this looks a bit clearer. Of course, you might not notice while dazzled by the…
• Slick visual feedback. In the standard version, metering has been enhanced. In the Advanced version, you get slick 2D and 3D plots of your sound spectrum for the Meter Bridge and Meter Taps modules. They look awesome, yes, but I also think these kind of “alien world mountainscape” views can help you better visualize what’s happening in a sound, so there is a practical use, too.

And, of course, all of this means you can easily wow clients when mastering by showing them visualizations that look like Geordi LaForge is studying abnormal quasar activity from the deck of the Enterprise. Just try to avoid opening up a cosmic string-related time wrinkle while mastering.

(And yes, when you’re all alone and no one is looking over your shoulder, you can do something useful with it.)

Pricing: US$249 (€195); US$999 (€799) Advanced.

Why is Advanced so expensive? Well, each module is also an independent plug-in you can use in your host. With that in mind, this starts to look like a better deal – some terrific reverb, EQ, and dynamics you can use anywhere. You also get the Meter Bridge and Meter Tap for analysis, fancier 2D and 3D spectrographs, and more advanced loudness meters. On the other hand, the basic version will also work with your host and gives you the sound-processing functionality minus all those more sophisticated meters you might need.

This month, there’s also steeply discounted intro pricing: US$599 for Advanced, US$199 for the standard edition. Expires December 1.

Ozone 5 was announced last month, but is now shipping. An OpenGL 2-capable video card is required for the 3D visualizations, but nearly all machines now provide that (including most integrated chipsets, too).

Ozone 5 Product Page @iZotope

For a look at what this tool can do, here’s our friend and experienced mastering and mix engineer Danny Wyatt, talking about how he works with limiting. The new UI and meters are actually a lot clearer than what you see in the video, and offer some nice, new functionality. I can tell you, Danny is a fully-converted Ozone lover, having worked with him in the studio as he mastered my own album. He’s got a big toolset of other stuff, but Ozone is very often what the real work comes down to, and — I think I can say this, Danny — he’ll be happy to evangelize the tool if you talk to him.

This isn’t a review, mind – in fact, my only significant reservation is that Ozone is so slick, it could distract from the reality that good mastering probably doesn’t need it. A great mastering engineer can do wonders with a fairly simple tool and their ear – no wild visualizations required. (“Great mastering engineer,” also known as, “not me.”) But that same person may well appreciate the level of precision iZotope, working with algorithms they’ve developed entirely in-house, can provide.

We want your feedback, as always. Ozone users – what do you think?

Users of rival products – what’s your all-in-one mastering tool of choice, and why?

At the AES show in New York, Avid as expected updated their flagship Pro Tools DAW to version 10, and unveiled a next-generation version of their higher-end HD line. Since it’s Friday, and perhaps not the best day for processing loads of information about new DAWs, let’s see if we can’t boil down the major points.

Pro Tools 10

Clip Gain helps you to set gain levels before you get to the mixer.
More flexible use of files: mix formats, and in a departure for Pro Tools, record and master natively in 32-bit floating points
Better performance on slow disk drives
Low-latency recording and direct monitoring at last added to third-party interfaces
New Avid Channel Strip plugs (based on the former Euphonix)
SoundCloud export (a bit surprised to see Avid beat some of its rivals to this… though, of course, you probably already know how to uplaod)

There’s also EUCON controller support, projects that now have a 24-hour timeline, and enhanced responsiveness.

This is really a Pro Tools upgrade for Pro Tools users – no splashy features, necessarily, so much as stuff their loyal user base is likely to appreciate. But I know Pro Tools users have a pretty long wish list, so PT die-hards, let us know what you think of the update, especially as you begin using it.

Pro Tools HDX

HDX is basically HD super-sized – and that makes sense, as it keeps pace with advances in technology (and particularly the vastly-expanded native processing capabilities we’ve seen over the past decade).

You get five times the DSP power per card versus the previous HD Accel, more tracks, and the new floating-point architecture.

It’s big news that Pro Tools has finally gone to a 32-bit floating-point architecture … well, anyway, if you’re into numbers. I’ll be curious to know if people can tell the difference. Sounds like we need a very controlled double-blind test, and comparing Pro Tools to Pro Tools would be perfect for the job.

Obligatory promo video. Hyperbole/marketing glasses at the ready!

And here’s a look at that new channel strip. Mmmm… channel strippy.

http://www.avid.com/US/products/Pro-Tools-Software

http://www.avid.com/US/products/pro-tools-hdx

With great power comes great learning curves – or maybe not. Csound for Live, just announced this weekend and shipping on Tuesday, brings one of the great sound design tools into the Ableton Live environment. You can use it without any actual knowledge of Csound, without a single line of code — or, for those with the skills, it could transform how you use Csound.

For anyone who thinks music creation software has to be disposable, you’ve never seen Csound. With a lineage going literally to the dawn of digital synthesis and Max Mathews, Csound has managed to stay compatible without being dated, host to a continuous stream of composition and sonic imagination that has kept it at the bleeding edge of what computers can do with audio.

Csound for Live does two things. First, it makes Csound run in real-time in ways that are more performative and, well, “live” than ever before, inside the Live environment. Second, its release marks a kind of “greatest hits” of Csound, pulling some of the platform’s best creators into building new and updated work that’s more usable.

If you’re not a Csound user, you just dial up their work and see what your music can do. If you are, of course, you can go deeper. And if you’re somewhere in between, you can dabble first before modifying, hacking, or making your own code. And that means for everybody, you get:

• Spectral processors
• Phase vocoders
• Granular processors
• Physical models
• Classic instruments

More description:

It looks great. It works great. It sounds… beyond great.

CsoundForLive is a collection of over 120 real time audio-plugins that brings the complexity and sound quality of Csound to the fingertips of ANY Ableton Live user – without ANY prior Csound knowledge.

Capitalizing on the design power of Max For Live, what once took pages of text in Csound can now be accomplished in a few clicks of your mouse.

Move a slider on your APC40 and deconstruct your audio through professional quality granular synthesis…

Touch a square of your Launchpad and warp pitch and time with real time FFT processing…

Press letters on your keyboard and create sonically intricate melodies through wave terrain synthesis…

And Dr. Richard Boulanger, unofficial Jedi Master of the Csound movement, instigator of this project, and Berklee School of Music sound and music wizard, posts a bit more:

With my former student, and now partner, Colman O’Reilly, I have been working around the clock for months to collect, adapt, create, wrap, and simplify a huge collection of Csound instruments and make them all work simultaneously and interchangeably in Ableton Live. In this guise, I am able to “hot-swap” the most complex Csound instruments in and out of an arrangement or composition – on the fly. This is something Csound could never do (and still can’t!), but CsoundForLive can, and it makes a huge difference in the playability and the usability of Csound.

Two weeks ago, I played a solo concert in Hanover Germany, at the first International Csound Conference. There, all of my compositions, from 20 years ago to 20 minutes ago, were performed in real-time using CsoundForLive. Tonight, at the Cycling ’74 Expo in Brooklyn, NY, I will be demonstrating the program; and next week, I will be releasing this huge collection (on Tuesday, October 17th, at 12:01am).

A huge part of the complete collection is FREE, and I hope it will make the creative difference in your (and your student’s) lives that it is making in mine. This is a serious game changer for Csound. Check it out. Dr. B.

If you’re at Expo ’74, do say hello to Dr. B for us (and I think you’ll get some nice surprises with this project).

I’ve got a copy in for testing, so stay tuned. And I’ll be doing some follow-ups with Dr. Boulanger and company.

The only bad news here, of course, is that both a supported version of Ableton Live and Max for Live are required to be able to run Csound in this way. In fact, sounds like we have a nice four-horse race going. Max 6 overhauls how multiple patches work (on top of Max for Live), SuperCollider has its own possibilities for multiple real-time patch loading, someone suggested in comments using pd~ inside Pd to manage multiple Pd creations (something fairly new even to most experienced Pd users), and now we have Csound in Live.

But overall, Csound for Live looks like a no-brainer for Max for Live owners, no question, and an exciting taste of the ongoing convergence of cutting-edge creative sound and code with live music making for everybody. As I hinted at in the Max 6 post, I think it’s suddenly a Renaissance for all these platforms.

http://www.csoundforlive.com/

Silly geeky footnote: With pd~ for Max, I know it’s possible to run Pd for Max. And via another external, Pd can also run Csound. So we could theoretically run Csound in Pd in Max in Live. But let’s not get carried away.

More Videos

Max 6 in Public Beta; For Home-brewing Music Tools Graphically, Perhaps the Biggest Single Update Yet

Just because a music tool fills your screen with tools and options doesn’t necessarily make it easier to realize your ideas. From the beginning, the appeal of Max – as with other tools that let you roll your own musical tools from a set of pre-built building blocks – has been the blank canvas.

Max 6 would appear to aim to make the gap between your ideas and those tools still narrower, and to make the results more sonically-pleasing. The reveal: it could also change how you work with patches in performance and production. I was surprised when early teasers failed to impress some users, perhaps owing to scant information. Now, Max 6 is available in public beta, and the details are far clearer. Even if Max 5 was the biggest user interface overhaul in many years, Max 6 appears to be the biggest leap in actual functionality.

It’s what I’d describe as a kitchen-sink approach, adding to every aspect of the tool, so there’s almost certain to be some things here you won’t use. What could appeal to new users, though, are I think two major changes.

More visual patching feedback and discoverability. First, building upon what we saw in Max 5, Max’s approach is to provide as much visual information as possible about what you’re doing. It’s probably the polar opposite of what we saw earlier this week in something like the live-coding environment Overtone: Max’s UI is actively involved with you as you patch. There are visual tools for finding the objects you want, then visual feedback to tell you what those objects do, plus an always-visible reference bar and rewritten help. This more-active UI should make Max more accessible to people who like this sort of visual reference as they work. No approach will appeal to everyone – some people will find all that UI a bit more than they like – but Max’s developers appear to be exploiting as much as they can with interactive visual patching.

Multiple patches at once. New objects for filters and data, a 64-bit audio engine, and low-level programming are all well and good. But the change that may more profoundly impact users and workflow is be the way Max 6 handles multiple patches. Max – and by extension Pd – have in the past made each patch operate independently. Sound may stop when you open a patch, and there’s no easy or fully reliable way to use multiple patches at once. (Compare, for example, SuperCollider, which uses a server/client model that lacks this limitation.) That changes with Max 6: you can now operate multiple patches at the same time, mix them together with independent volume, mute, and solo controls, and open and close them without interrupting your audio flow. (At least one reader notes via Twitter that you can open more than one patch at once – I’d just say this makes it better, with more reliable sound and essential mixing capabilities.) Update: since I mentioned Pd, Seppo notes that the pd~ object provides similar functionality in regards to multiple patches and multi-core operation. This has been an ongoing discussion in the libpd group, so I think we’ll revisit that separately!

One upshot of this change: some users have turned to Ableton Live just to host multiple patches. For users whose live performance set involves Ableton, that’s a good thing. But it could be overkill if all you want to do is bring up a few nifty patches and play with them. Now, I think we’ll start to see more people onstage with only Max again. (Check back in a few months to see if I’m right.)

Here’s an overview of what’s new:

• Discoverability: A “wheel” makes the mysterious functions of different objects immediately visible; Object Explorer makes them easier to find, and new help and reference sidebar keep documentation close at hand.

• 64-bit audio engine

• Open multiple patches, solo and mute them, open and close them without stopping audio, mix audio between them with independent volume, and take advantage of multiple processors with multiple patches.

• Low level building blocks: You don’t get new synth objects, but you could build them yourself. New low-level data-crunching goodness work with MSP audio, Jitter Matrix, and OpenGL textures

• More JavaScript: An overhauled JavaScript engine makes JS scripting faster and more flexible, and there’s a proper text editor with syntax highlighting (though, of course, you may still prefer your own).

• New visuals: Vector graphics and “HTML5 Canvas-like” UI scripting (though to me it’s a shame this isn’t just the HTML5 Canvas). There are also massively-expanded Jitter powers, but those are best left to our sister site Create Digital Motion.

• Filters: New filter-visualizing tools for audio filter construction and manipulation.

• Dictionary data type and associated objects let you describe information in a more structured way (all kinds of potential here from control to composition)

• Projects now let you organize data, media, and scripts in the manner more associated with conventional development environments

• What about Ableton? No news on that front, but I expect more soon. Max for Live users will at the very least get the advantages above, since Max for Live is really Max inside Live.

Looking over all that Max does, I have to say, I’m really amazed. I wonder if computer musicians ever pause to consider how fortunate we are. Even if this isn’t the tool for you, its availability – compounded by the availability of a range of other tools – is itself worth reflection.

Max is a program that shouldn’t exist, doing a number of things it shouldn’t do, for a user base that shouldn’t exist, doing things they shouldn’t be doing.

It doesn’t make sense that you could maintain a commercial project for this kind of audience, that you’d wind up with something this mature and powerful that had a continuous lineage stretching back to the 1980s. It doesn’t make sense that musicians would embrace such a tool and produce invention. The only explanation is sheer love.

Then, even as Max reaches new heights, some of the alternatives you have for making your own music tools are simultaneously growing by leaps and bounds. They provide very different approaches to music making (compare Overtone and SuperCollider, or Pd and libpd, or AudioMulch, or new Web audio tools). There really aren’t many fields that have this kind of choice, free and commercial, in their medium. In science and engineering, there’s private and public funding, producing some amazing tools but nothing with this kind of meeting of power and accesibility. There’s just something about music.

The fact that Cycling ‘74 can maintain a business model – just as open source projects maintain volunteer contributions – is a testament to sheer passion and love for music, and a commitment to perpetually re-imagining how that music is made from an atomic level up. There was a wonderful piece on C creator and UNIX co-creator Dennis Ritchie, whom I remembered yesterday, that observed that what he did was to do what others said couldn’t be done. From Max itself to what people make with it, I think that fits nicely.

So, have a look at the public beta, and let us know what you think. The release of Max 6 has caused more people to ask what this means for Pd and other tools, or even whether to patch things from scratch at all, but I’ll leave that question to a bit later. (I do have my own opinion about which tool fits which circumstance and user, but that’s best left to a separate discussion.) For now, you can try Max yourself and see what the fuss is about. If it doesn’t fit your means of music-making, know that you have a wide array of other options – pre-built to low-level code to old-fashioned tape-and-mic approaches, and everything in between. Go out and listen and see what you discover.

http://cycling74.com/downloads/max-6-public-beta/

Universal Audio has a big set of updates to their DSP software out today. It seems worth sharing in part because I find the (faked) look of that Ampex reel-to-reel does give me a warm and fuzzy feeling, even if it mainly makes me want to get in a studio and hear the whir of the actual gear.

Here’s what’s in the update:

For Pro Tools users, there’s vastly-expanded support, including native RTAS plug-ins (instead of the VST-to-RTAS adapter), control surface support, proper automatable parameters, and correct naming and sorting, plus a quicker installation and workflow. I’m embarrassed to say, not having tried UA’s stuff in Pro Tools, I was unaware they’re missing – that sounds like the set of functionality that would make mixing UA and Pro Tools genuinely workable.

Full Mac OS X Lion compatibility is a big deal. We’ve seen a lot of Lion support arriving lately, and I saw readers specifically complain about the lack of support in the update. Seeing a hardware DSP platform with support seems to me a good sign, because it means both plug-in and I/O compatibility have to be in place. It’s getting to be about time to do a new compatibility update, and it may be an OS update that’s possible to recommend shortly (though Snow Leopard is working just fine for now).

Then, there are the plug-ins:

Brainworx bx_digital V2 EQ: This popular mastering EQ now comes to the UA platform, with M/S mastering, M/S recording, and L/R stereo modes. It’s kind of an EQ-plus, with M/S de-esser built in and mastering-specific EQ options, plus a mono plug for mixing.

SPL Vitalizer MK2-T is back in the emulation territory, modeled on the tube-based gear. Separate bass and mid-high modes give you frequency-specific loudness controls.

Ampex ATR-102 models the legendary two-channel tape, and has Ampex’s seal of approval. That means UA now models the two studio workhorses, the Studer A800 (see our previous coverage and shameless gear pr0n gallery) and the Ampex. ATR users and engineering titans Chuck Ainlay, Richard Dodd, Buddy Miller, and Mike Poole contributed presets, in a nice touch. US$349 buys you the plug … and yeah, that’s likely to make a few studios nervous on their rental fees. Then again, look at the names on the presets, and remember just how valuable engineers are, whether they’re behind an analog deck or a computer with a UA plug. (Take it from me – someone who’s not really an engineer.)

The SPL and Brainworx plugs were each developed outside UA. My guess is that Pro Tools support is a factor in seeing more third-party development. And while readers correctly noted the demise of other DSP platforms, UA’s is looking very healthy from a business perspective.

The only bad news: I’m confirming this, but there appears not to be support for 64-bit operation on Windows, which remains a significant omission in UA’s software.

Now, we need some double-blind A/B tests with both the Studer and Ampex (and maybe the real gear). Wonder if someone could help us set that up.

I really enjoyed some of the skepticism in comments on the last article. I think the questions “do I really need this?” and “will this give me the most value over time?” and “what alternatives should I consider?” are absolutely the questions you should ask about any music tech products. This is a news item, so I’m not anywhere near knowledgeable enough to comment in this case. I think the best thing to do is, beyond just reviews, continue to talk to producers and engineers about how they use software, and to developers about how it’s made. If you have specific questions, let us know.

And let’s have another look at the original. Simon Campbell on Flickr has this shot of one with the spools spinning:

We mastered onto this rascal at 30 ips using Dolby SR. Nice.

Recording the album ThirtySix at Gracieland Studios [UK], Rochdale

Photo is (CC-BY-ND) Simon Campbell, whose music you can go hear (linked via his name).

Universal Audio Downloads for the latest updates | Universal Audio Site

With all the horsepower computers are now packing, you might be surprised at the idea of adding on dedicated hardware for sound processing. Or, you can look at it another way: with computers more powerful than ever, with digital processing sounding more convincing both as emulation of traditional gear and in imagining never-before-possible sounds, the digital studio in a backpack is even closer.

Into that picture, enter the Universal Audio UAD-2 Satellite. Enclosed in a metal housing about the size of a large-ish external hard drive, the Satellite could absolutely fit into the side pocket of a computer backpack or messenger bag. Coupled with a MacBook Pro laptop, you could very easily carry your entire studio on a bicycle. That’s not to take away from the joys of outboard gear, but if you’ve got some decent engineering chops, such a rig could really be a studio that can live anywhere.

I’ve had a chance to begin working with the Satellite as my main platform for UAD’s plug-ins for some months now, and it’s an extraordinary box. The most important thing to know about it is that it’s intended for select Intel Macs, and it’s equipped with FireWire 800. Dedicated DSP processing goes back to the very first days of digital audio on computers. (Early Digidesign products and even the IRCAM-developed predecessor of Max/MSP all used DSP hardware.) DSP also naturally appeals to sound engineers: it’s hardware built for the sole purpose of doing the kinds of number crunching in audio, as opposed to the general-purpose architecture of a computer CPU.

The challenge has always been how to get data between the computer and the DSP device. That’s led to an array of buses, like PCI Express slots (which requires jamming a card into a desktop computer) or ExpressCard. As Apple have largely phased out ExpressCard expansion, Mac users have found themselves without a solution.

The UAD-2 Satellite makes use of FireWire 800 instead, thus opening up compatibility with recent Intel-based MacBook Pro laptops as well as the iMac and Mac mini. The 17″ MacBook Pro continues to support ExpressCard, but FireWire 800 offers greater processing power (and is less prone to popping out, as cards in the ExpressCard slot have a nasty tendency to do). The Satellite is available as a “DUO” and “QUAD,” indicating the number of internal processors. (You do the math to work out how much more you get from a Quad than a Duo. I’ll wait…)

As an aside, readers routinely ask if something like the mini or, especially, the iMac could work for audio production. Resounding answer: yes, absolutely. The iMac in particular has a pretty compelling price/performance ratio if you want a compact machine to drop on a desk and don’t have a spare monitor. The mini’s no slouch, either, and seems a logical addition to, say, a project studio.

And that brings us back, full circle, to the reason the Satellite is compelling. It unlocks processing power exclusively dedicated to some tasty and useful processing, all emulating classic gear, while freeing up your computer to do other things. You might, for instance, focus on native processing for a software synth and some creative effects, then bring in the Satellite’s UAD-platform effects to add some historically-accurate compression. And even an entry-level, lowly Mac mini, coupled with the Satellite, is perfectly capable of handling typical compositional and mixing environments without bouncing to audio or freezing tracks.

You certainly need to be interested in the UAD catalog of audio processing tools before this really becomes relevant. For some insight into how Universal Audio does their development and conceives what they do, with the obligatory drool-inducing photos of some retro hardware, see our interview with Dr. David Berners:

Modeling Analog in a Digital Age: A Conversation with Universal Audio’s Chief Scientist; Gallery

But if you are looking for a platform on which you can run these effects – or if you’re ready to upgrade from a previous UAD system (your existing plug-in registrations will port right over) – read on.

What UA Says About the Satellite

Looking at a Satellite, you can’t really see much – it’s a magical, mystery box that processes sounds. It’s what’s under the hood that matters. So I talked with Amanda Whiting of Universal Audio about some of the technical details of the Satellite.

CDM: How has the Satellite evolved from previous UA gear?
UA: The UAD-2 Satellite marks the first Firewire-based external DSP unit offered by UA, for Intel-based Mac laptops and desktops. The Satellite provides up to a four-processor UAD-2 DSP Accelerator for the many people mixing and mastering on the road with their laptops, and for those who don’t have a desktop system that includes PCIe slots. It also allows for easy session compatibility — so you can take your UAD-2 plug-ins with you, and mix on another Intel-based Mac system that may not have a UAD-2 card installed. The FireWire 800 onboard provides twice the power of FireWire 400. This allows you to connect a UAD-2 Satellite and a FireWire audio interface together on the same FireWire bus, and still have enough bandwidth for lots of plug-ins.

Ed.: That’s an interesting detail, in fact – hard disks and most other accessories take advantage of only a fraction of the added bandwidth of FireWire 800, meaning for many applications, the “800″ is a bit of a misnomer. Us audio folk are different – we really are talking about maxing out that additional bandwidth for the UAD-2.

Any rough practical info on the DSP horsepower? I’ve done some tests as far as what I can run simultaneously and it’s a great step from ExpressCard, but curious how best to quantify it.

The UAD-2 Satellite QUAD is four times as powerful as the ExpressCard-based UAD-2 SOLO/Laptop card, which has a single chip. For your reference, here’s a couple of links:
http://www.uaudio.com/blog/uad-2-satellite-basics-faq
http://www.uaudio.com/support/uad/satellite-support

Also here is a link to the instance chart. The UAD-2 Satellite has the same theoretical instance counts as the PCIe cards, except at very high instance counts where the FireWire bandwidth is exceeded. Still you can run 152 mono plug-ins or 77 Stereo plug-ins with UAD-2 Satellite, so the FireWire bus is not typically a factor.
“http://www.uaudio.com/support/uad/compatibility/instance-chart.html

What real world uses are you seeing so far from your users? To me, it seems really practical for even live laptop performance, and of course mobile production … not to mention the ability to take your UA faves to a studio.

The UAD-2 Satellite definitely provides a level of portability that we haven’t been able to offer previously. It’s great to take into any studio and pull up your UAD-powered sessions, and it’s absolutely ideal for mobile production. As far as live use, latency is always an issue with running audio over Firewire, but with certain effects — particularly reverbs and delays that lend themselves to live tweaking — the latency may come across as a pre-delay and sound just fine. We’ve heard this more than once from our users. We’ve also seen a lot of adoption with Pro Tools HD Customers – these customers typically have all three PCIe slots taken up on a Mac Pro so the UAD-2 PCIe version may have been a non-starter, but with Satellite, they can keep their cards in the machine and simply connect UAD-2 Satellite via FireWire with great results.

Ed.: One note on latency: with a recent update, you can set minimum latency to an impressive 256 samples. So, at the very least, the software itself is not a limiting factor.

Setup and Use

Setting up the Satellite is fairly easy. I’d actually struggled a bit with an ExpressCard UAD-2 – firmware updates, a card that initially didn’t work, and difficulty, as with all ExpressCards, with the card popping out of the slot. (That’s not UA’s fault: it’s the result of overly springy slots on typical laptops, and the fact that the spring-loaded eject is itself a really terrible idea for something you want to stay connected.) The Satellite was much easier: plug in power, plug in a cable, boot up the machine, and go.

As with all UA products, the Satellite relies on a single, unified installer that gives you all of the plug-ins ready-to-use as VSTs, Audio Units, or, for Pro Tools, RTAS. (On the Mac, I’d recommend defaulting to the VST in hosts that support it.) You get a 14-day trial of everything; for longer use, you’ll need to purchase and authorize the plug-ins. (Various bundles with the hardware get you started with licenses for a range of tools.)

Download and open an authorization file, and you get access to the plug-ins you need.

On the hardware side, you need two cables to connect the Satellite: power and FireWire 800. (Bus power over FireWire 800 is insufficient to drive the Satellite.)

Stop and consider this for a moment: you get exactly the same power out of a Satellite that you would out of the equivalent internal card. That means the argument for a desktop chassis is greatly reduced versus a more convenient iMac, mini, or MacBook Pro.

Because the authorization is associated with the hardware, you can also move the Satellite between multiple locations. These days, a lot of us do production and mastering and such against tight deadlines or in time on the road. Now, you can do that, but still bring your arsenal of effects into a physical studio environment when you’ve got a couple of days booked for recording.

The hardware itself is really lovely; it’s definitely been rugged enough to hold up to all that travel. For anyone considering this for an institution or studio environment, there’s also a Kensington Lock so someone doesn’t walk off with your valuable gear and authorizations. There’s also a pretty, glowing UA logo that shows you power is provided. Unfortunately, the LED that shows you if you’re properly connected to the computer is hidden away on the back; it would have been nice if UA had associated that to the giant herald on the front, instead, so you could actually see it.

Also, I was surprised to learn that hot-swapping is okay. So long as you shut down your UAD software first, UA says you can feel perfectly safe disconnecting and reconnecting the hardware to an active or sleeping computer. That’s a nice boon to us laptop users.

“Ah,” you say, “this FireWire business is great, but I also use FireWire 800 for my project drive. And an audio interface.” The Satellite gives you a couple of options here. For one, it has a pass-through port, so you can daisy-chain additional hardware without a hub. Again, power becomes an issue. Most hard drives I’ve found have power ports, so you can simply bring along their power wall wart and power them separately. (You’ll want to bring along a little power strip.) UA suggests that for gear that lacks that – like certain Apogee audio interfaces – you can use a FireWire repeater, an affordable accessory which injects power externally.

The more significant issue is bandwidth and speed mixing. Any additional device will take up some bandwidth, because they’re sharing the same bus. In my use, I chose to simply run audio over USB2 and connected, with the addition of my wall wart, a portable Glyph project drive to the FireWire 800 port on the UAD-2. If you’re doing something fancier than that, you can allocate bandwidth in the UAD software.

The trick is if you add a FireWire 400 device to the mix. If chained in the wrong order, any single FW400 gadget will cause the FW800 gear to slow to 400 speeds. The solution: just connect that 400 device (like, say, an old MOTU audio interface) last in your chain.

This, to me, brings up an unexpected reason the new Thunderbolt port on new-model Macs becomes useful. The UAD-2 Satellite, for its part, gets plenty of bandwidth from FireWire 800. Universal Audio has expressed an interest in supporting Thunderbolt in the future, but for now, FW800 works just fine. If you invest in a Satellite now, but you’ve got a Mac with Thunderbolt, you could in future connect a different accessory to that Thunderbolt port rather than the FW800 port, thus leaving the Satellite its own bus. (Got that?)

In the meantime, I’m absolutely, positively happy with my 2010-model MacBook Pro, which I picked up steeply discounted when the new Thunderbolt models came out. I maxed out the RAM and saved hundred of bucks, and the combination of 2x USB2 and FW800 more than suits my needs. No complaints here.

As Amanda indicates, the Satellite gives you a significant amount of processing power. For my use, this was perfect for experimenting with creative effects and adding UA’s excellent compression and channel processing tools. I’ve got some mixing and mastering projects coming up, and can’t wait to bring the UA stuff into the workflow.

Recommendations

Whether you want access to the UAD platform is really dependent on your needs and tastes. Certainly, there’s a wide variety of native processing tools that don’t rely on external DSP hardware. The main appeal, as I’ve said in the past, is the unique, historically-informed modeling approach that Universal Audio take to their work. Their catalog is certainly extensive, and I’m especially happy with the quality of the recent additions, like the Studer multitrack tape emulation and some superb reverbs, compressions, and the like. (One new entry: the Lexicon 224.)

The question is really whether the FireWire 800 bus is big news for UA, and there, it’s tough to overstate how much this changes working with UA’s stuff on a variety of Macs and on the road. The SOLO I’d tested previously is nice enough, but the DUO and QUAD really give you the amount of processing power you’d want to do some real work, to experiment live across a number of tracks without running out of horsepower – and that’s, after all, the point of using a DSP platform.

A new Mac and a Satellite are really all you need to build an impressive digital studio. They now give you the freedom to make that studio exist anywhere, and with almost any set of tools. We’ve seen that kind of liberation with native processing, but to get that native power and DSP power at once is really a dream. For existing users, moving over is a no-brainer, since sharing authorizations is a cinch. For newcomers who’ve been waiting for the optimal hardware choice to unlock the UA catalog, this is it. (It’s worth looking into bundles to try to get your collection of effects rolling.)

Even with Thunderbolt on the horizon, external DSP on a MacBook Pro or mini is now finally accessible. UA’s stuff isn’t cheap, but if the value proposition makes sense to you, and you’re a Mac owner, you now have the combination you’ve been waiting for. The only bad news, really, is for PC users left in the cold – and there, we may just have to wait and see what direction laptop buses take in the wider market.

Universal Audio UAD-2 Satellite DUO and…
Satellite QUAD

Got questions for me, or for UA? Feedback on this gear or this review? Fire away in comments, as always, folks.

Comfort and creativity – the mystery of what makes certain vintage gear so appealing remains. There are few people closer to the meeting place of digital and analog, reason and sentiment, than Dr. David Berners. He’s the chief scientist for Universal Audio, responsible for modeling in digital software form the characteristics of sought-after, beloved analog gear. It’s science: Berners cut his teeth as an engineer working on the physics of nuclear fusion, going on to pursue a love of music and sound. Now he uses knowledge of physics and the characteristics of sound equipment to model computationally what makes this gear sound the way it does. But it’s also commerce: UA’s DSP platforms unlock access to a range of a la carte plug-ins, bringing a menu of sounds from the past to modern engineers without the associated bulk, inconvenience, and cost of the real thing.

So, if you’re curious to know a bit about what makes analog and digital gear tick, what that analog gear means in a digital age, Dave’s a good place to start. The timing’s good: UA’s on a bit of a roll. The company’s heritage begins entirely in the analog domain, founded in 1958 by Bill Putnam, Senior and resurrected in 1999 by his sons, James and Bill, to make new tools in both hardware and software. UA has recently introduced an elaborate software model of the Studer A800 tape recorder, one that seeks to make a digital workstation sound like a beloved, high-fidelity multitrack tape setup. There are also new models of the SSL console, authorized by manufacturer Solid State Logic, providing the channel strip and bus compressor; the real thing earned more Platinum records than any other gear, so it’s more or less guaranteed you’ve heard it unless you’ve been holed up on a farm listening to old-timey AM for the past few decades. And they’re expanding compatibility, with new support for Pro Tools and, via FireWire, all those Mac laptops that lack ExpressCard slots.

None of that, though, really winds up being the focus of our conversation. Dr. Berners is also Professor Berners, teaching the elements of DSP to students at Stanford with another UA alum and former CTO. Here, class is in session, as he talks about his laboratory-style approach to understanding how equipment works, and why having a theoretical model is so essential. He hedges on the question of why analog gear is appealing, leaving that to others, but opens up when explaining why he fell in love with engineering.

And, in the process, we get some serious gear porn courtesy of photography (and UA PR rep) Marsha Vdovin. She takes us inside the UA studio for a glimpse of a treasure trove of drool-worthy vintage gear and modern test equipment.

Deafening us with science, here’s Dr. Berners, proudly sponsored by our favorite advertiser, The Field of Mathematics. (They’ve been working on improving their PR lately. I hear they’re on Twitter.)

But deep beneath all that science, all the most empirical techniques for modeling, you might just discover how and why digital audio today could find its connection to the past.

CDM: Can you tell us a bit about how you wound up in this field? What led you to working in the science of DSP?

Dave: My parents told me that I wanted to be an engineer, ever since I was about five years old. I described to them the job that I wanted to do, and I asked them what it was called – and they said, that’s called an engineer. As far as I remember, that’s always what I figured I would end up doing.

After finishing a Masters Degree in power supply stuff, I worked at NASA a while on some design stuff for a couple of different projects, and then after that I worked at the Lawrence National Lab in Berkeley. It was some physics projects related to fusion power plants, so that was very different from audio. While I was doing all those things I didn’t realize I could find work in audio. I always liked the idea of doing audio-related stuff, but I didn’t know there would be any way I’d be realistically able to do it. While I was working at those two places, I found out about the CCRMA center at Stanford and decided to apply there for a graduate program [in the early 90s.] That was when I met Bill [Putnam, Jr. founder of the modern UA], because he was also a student there. I had done a little bit of DSP before that, but that was where I learned most of what I know.

And you’re teaching now at CCRMA.

Yes, along with Jonathan Abel. There’s one course in the fall that’s an introductory DSP, Discrete Time Filtering class. That was a course that was created by Julius Smith. He’s written the textbook for it. It’s meant for people who are just getting their feet wet … with no prerequisite other than high school math. The other class is the one Jonathan and I created, and that one’s more related to audio effects processing — tricks, I guess, on how to define DSP effects.

How did your background apply to coming the Universal Audio? Was there an additional learning curve, getting into work in audio?

For me personally, it was pretty smooth because I had real strong musical interests, the whole time I grew up. I had been an amateur musician my whole life, and spent a lot of hours playing music and working on music. Somehow that gave me an advantage – if I discovered I had made a bug or done something wrong, it gave me a good intuition — if something isn’t sounding right, what is it likely to be?

How much of modeling is intuition? I’d imagine that, with audio, the ultimate test is sort of if it sounds right, it is right?

The feeling thing is important. The way I like to think about it is that it’s not really among the design criteria. It’s more of a check. Ideally, what I would like is to be able to get a bunch of information about a product — schematics, info about the physics of how it works, whatever I need to understand the processes by which it operates — make a model, and implement the model. Human hearing comes into play, psychoacoustics, to determine what may or may not be important perceptually. But what I always hope is that by the time we get to the listening phase, there’s not anything left that we’re trying to tune, so to speak. It’d be more like catching bugs.

I do rely on our listening team — Will Shanks, in particular. He’s in charge of the qualification of our products in terms of our sound. I rely on him a lot, but it’s more that he’s finding little mistakes and errors. I don’t ever want to get into a situation where we listen to something and compare it to the original and say, well, I wish this sounded a little brighter. I would be very unhappy if I got in that position. I would much rather be able to have a complete understanding of how the original equipment works, and match that.

There’s a particular reason why I developed that opinion, and it has to do with the non-linearities that exist in a lot of this gear. If we had a totally linear system, like an equalizer that has no measurable distortion, that kind of system can be characterized fully just with one measurement. If I make a very good measurement in a careful way, for whatever setting the controls on the equipment is at, I can know everything there is to know about that piece of equipment. I can be totally confident that no matter what signal somebody puts through it, I can predict the behavior, just on the basis of my one measurement. What happens is if there’s any sort of non-linearity at all, unless we can characterize the non-linearity in a very specific way … it becomes absolutely impossible to characterize by measurement. There would always be the fear that even though you’d listen to a thousand audio snippets and they’d all sound identical, the next one that you try could sound different. It’s very difficult to have confidence in a model of a non-linear system, unless you know how that system works.

That goes hand in hand with how we do our measurements. We do use specific signals to cross-check our model — I’ll take a piece of gear, and start with the schematic, and write out with a pencil how it works. And then it’ll turn out that there are certain things in that circuit that aren’t really specified by that schematic. There’s a lot of behavior of different components – say you have a transistor or a tube or something – [where] you can write the part number on the schematic, but that doesn’t fully specify what that part does. We do have to do some measurements, but the only way we can trust the results of the measurements is if they’re informed. If I try to just take a piece of equipment as a black box, if I didn’t already know what was inside the box, it’d be impossible to make a good model.

You can do signal modeling, where you have some array of test signals that maybe you’ve developed and see what happens to them when they go through the equipment — and that to me is the risky way to do it. And then the other method is called physical modeling, where you try to understand all the processes that are happening inside the box. With that type of a model, you have to build the behavior from the bottom up, and then once you’re done, you need to verify that you’ve got all your parameters right. So instead of unknown types of behavior you just have unknown parameters. So you might say, I know there’s a capacitor inside here, and it probably has some resistance associated with it, and that resistance doesn’t appear on the schematic, because nobody knows what it is. But I can find out what it is by doing a particular measurement.

So then what happens is we’ll build up a behavioral model based on the physics of all the parts. And then only after we make the model can we decide what test signals are appropriate to expose all the unknown parameters. Every model that we make of a different piece of gear, we’ll have to invent a completely different set of test signals to find out the parameters of all the different components. Hopefully we’ll be able to do something without taking everything apart. In some cases, there are behaviors that are unobservable directly. Sometimes we’ll have to unsolder all the components and measure them separately and then put it back together. In general, we’re more comfortable trying to understand the real processes that are happening inside a box.

I’d assume you get better at doing this over time. Does what you learn in one place carry over to another?

Yeah, that’s half of it, and then the other half is, as time goes on, we have more and more processing power available to us. There are certain things that we would have liked to do in 1999 but couldn’t. We already foresaw that we might eventually be able to add some of these effects in. Every once in a long while, it’s worth revisiting some of these things and saying, well, now I have one hundred times more computational power available to me, so now I can start putting in more effects that are less noticeable than the ones we put in already, but maybe above the threshold of being able to be perceived.

So not only do we learn more as we do more projects, but we also have more opportunity to include effects that would have been too expensive ten years ago.

That seems to be a story that’s largely untold. People are aware of the trajectory of CPU power over the years. People are now looking at the area of the GPU and low-power CPUs. But people seem unaware that DSP chips has grown, too. It seems the bang for your buck is better today than it was even recently.

Oh, yeah, I’d say definitely. I think the tangible manifestation of that idea can be seen by the difference between our original UAD-1 card and the UAD-2. Over the last couple of years since the UAD-2 came out, we’ve had increasingly power-hungry processors that we’ve released. Now we’re to the point that a lot of this stuff would not have run even one instance on the old card. But we can still do it. We always feel like if there’s a question whether to include a part of the model that’s a little bit expensive, so far, we always put it in. The amount of processing power is never going to be reduced. We’d rather include more right now, because then we’re ahead of the curve.

Sometimes, it’ll turn out that we’re at a certain point in complexity, and in order to gain a tiny bit of perceptual improvement, it’d take a huge computational cost. And so then we figure we’re at a sweet spot, and so that’s good. Other times, we’ll look at something, and maybe by increasing computational cost a tiny bit, we could get a significant perceptual improvement. Then we may be inclined to put it in even if it stretches the current capacity of our hardware. There’s also cases where, if we feel like something’s gotten really expensive, sometimes we’ll make two versions of a plug-in. We always try to order everything so we take care of the major artifacts first.

Let’s talk about the Studer. First, to revisit this idea of process, where do you start modeling something like this? In some ways, it’s not the most non-linear of the things you’ve had to model. It does seem like it’s a complex system. There was a lot there to take into account in the design.

Yeah — the signal path is long, and there’s a lot of things happening in there. Also, the non-linearities, while they may not be as dramatic as, say, a guitar stompbox or something, they’re considerably complex. There’s a lot of behavior that has a fair amount of subtlety. I think that just about any magnetic mechanism is going to be complicated, because of the hysteresis that you get in magnetic processes. Not only is the tape deck magnetic, but it has a spatial extent. So whereas, if you have say a transformer or an inductor with a magnetic core, unless you’re being very picky, the coils of the wire don’t really move. They might deflect a tiny bit when current goes through them, but for the most part they stay put. And if you imagine the coils of wire are actually fixed on a transformer, the fields that are created don’t change their shape that much, unless you have a material that’s really saturating a lot. Basically, you have a one-dimensional system.

Whereas with the tape, there’s the thickness of the tape and then the width of the tape, and then there’s the length of the tape on which you’re making the recording. That’s all going by the heads, the record and the playback heads, and so the geometries become really important. Any time you have a system that’s got a spatial extent, and especially one that’s got moving parts like that, the computational complexity can go way up. Let’s say you have a tape that’s magnetized, it’s not going to be uniformly magnetized. The magnetization will be a function of the depth of the tape and the width of the tape and of course the length. If you wanted to keep track of all of that stuff, you have this sort of geometric explosion of complexity. It was really necessary to think very hard about how we could have some kind of a model that would be practical to implement – keep all the subtlety that we wanted to have.

Even though the original intent of the [Studer] deck was to be as linear as possible, to be a transparent recording medium, all those different factors made it one of the longer-term projects that we’ve ever done – just trying to figure out how to do the simplifications that we were going to have to do in a way that wouldn’t really detract from the fidelity of the model.

So, how much of this was theoretical? At what point do you have to look at the actual hardware?

We knew that we’d have to have a machine. Just distinguishing between the different tape formulations, it would be very difficult to be confident in those models done all in the abstract. This is one of those cases where we like to have a model, but it’s very important to be able to cross-check the results with the real thing. We had a Studer deck that we got from Ocean Way [Recording, the legendary Hollywood studio] and brought it into our studio. It’s been here for the last year and a half or so, and we’ve used it heavily. It’s really tough to take something like that apart; the cards plug into the interior of the machine. So we’d take the cards out and work on them, and I soldered a bunch of leads on different parts of the circuit that I wanted to look at, and then we could temporarily just lift a component if we really needed it to be disconnected.

For this, we ended up bringing in a bunch of scopes and other test equipment into the control room. I soldered flying leads onto the cards. It really turned out to be critical that we could look at different points inside the circuitry while we were using the deck.

It’s kind of related to the stuff I was trying to describe before. Even if we know a model for the whole process, if we want to expose a particular non-linearity or behavior at a certain point in the circuit, it’s a lot easier if we can look at data right from some internal circuit node rather than the output. So that’s how we did our verification – and obviously, listening, too.

It’s well known that the Studer is really carefully designed to have high fidelity and be well-behaved. But in spite of that, it turns out that there’s a little bit of non-linearity on the record amplifier, so the signal’s [got] some artifacts associated with it on the way to the record head. So that’s why we felt we had to monitor all these points on the circuit.

On the other hand, if we would have just started hooking up wires at different places and blindly tried to figure out what’s going on without knowing anything about how this stuff works, there’d be no way to work out a workable model. If we put out some signal that we just made up out of thin air, it would be overwhelming.

Having gotten intimate with this equipment, can you comment on what makes this gear so desirable in the first place, aside from pleasant associations with it historically?

I’m comfortable with audio and music, but I don’t want to decide upfront that something will be unpleasant or undesirable and leave it out. I’d rather put everything in. It’s never obvious, really, which artifacts are the desirable ones.

It seems really the opposite from what we’re seeing in consumer photography. There, when you see iPhone apps like Hipstamatic and Instamatic, the idea is to apply very specific, desirable qualities from a camera intentionally, rather than to model the whole camera. So they really have decided what’s desirable.

[laughs] If we had as many customers as the iPhone, maybe we’d charge $2 for an app.

Hey, in that case, maybe you’d just listen to your entire song library as if it were coming through the Studer.

That’s right. That could actually be great.

There is a place for that line of thought. And to me, that place is to make forward progress. Let’s say that we’ve analyzed a hundred highly-prized pieces of vintage gear, and tried to understand what makes them all special. Now, it gives us hopefully a good information base, and maybe a little intuition ourselves of how we’d design a new piece of equipment if we wanted it to have a specific sound. If we were going to design something like that, then we’d have a lot of freedom that we wouldn’t [otherwise], if we’re not claiming it’s identical to something.

For us, it’s worth it to do the modeling just to achieve the models ourselves. And when we are doing a model, we don’t want to interpose our own ideas about what’s important. The one case where we do is if there’s really solid evidence from psychoacoustic experiments that people will not be able to perceive something, then we will neglect that if it turns out to be expensive to put things in.

We’re willing to accept the fact that people will be unable to perceive certain things. But what we’re not willing to do is to decide whether something will be pleasant or unpleasant.

What were some of your favorite projects from UA’s now fairly large back catalog – or what were the toughest models you worked on?

Almost the first two models were the 1176 and the LA2A. And it’s kind of interesting to think about those, because they both were difficult, but for different reasons. The LA2A has this little electro-luminescent panel in there that lights up and shines on a light-sensitive resistor, and that’s how the compression happens. And it turned out that the physics of that panel were very difficult for us to understand. And so we spent a long time trying to figure out how in the world we would even understand the mechanism of how that worked but then characterize them somehow. The behavior was just very, very complex and multi-dimensional. It just was very difficult. It really was satisfying to finally get a model that had the right behavior.

The thing that made the 1176 very hard was that the attack is very fast. It’s actually faster than one sampling interval if you’re at 44k. The attack is pretty much just about complete by the time you advance one sample forward in time. Even though we could characterize the behavior of the 1176 more easily than the LA2A, implementing the plug-in became very, very tough, because we had to make this feedback loop. It’s a feedback compressor, and we had to make the loop behave properly, even though these processes were happening much faster than one sample period. So we had to think really hard in terms of how to implement the thing. There’s a lot of different ways to get stuck — you could get stuck trying to understand the actual process, or you might understand the process but then think, “How can I implement this as a digital system?” So at different times, we’ve had different things that stuck out as the tough part of a project.

It does sound like you have a strong philosophy.

When I first started working at UA, Bill met with me and said he had the idea and the vision to do these models, based on physical process. It’s been a company point of view, irrespective of who does the work. There’s actually three or four of us now that do algorithms here, all working with the same ideas and the same ways of going about things. It’s a broad angle of attack that we as a company decided to do, not something that any one person developed.

Bill’s been really great to work with from the days when we were in school together up until now. One thing I really admire about Bill is that he can look at a problem and reduce it to the important components immediately. He can look at something that’s really complex and has a lot of different factors that are difficult to dis-entangle for someone else and get right down to what the important behaviors are going to be. It’s just a really nice, organized way of thinking that he has.

I should also mention Jonathan Abel. [founder of Kind of Loud before it merged with UA]. He had a lot of input – a huge effect into the shape that the first batch of plugins evolved. He worked a lot in the trenches on the algorithms with me, and had a huge impact on how that stuff came out.

So why model historic gear in the first place? And once you are done with the process, what does that tell you about why people value these tools?

That’s a tough question to answer definitively. It’d be very hard for me to make a convincing argument that someone should want to have those models and use them all the time. But if you just want to answer the question, why would someone ever want them, then it’s easier to answer that question.

There are thousands and thousands of vintage projects that have been designed. The ones we focus on are the ones that for some reason have become highly coveted. In a lot of cases, those ones are the ones that were the most carefully designed or the most expensive things available at the time. Not all of them – some of the stuff that turns out to be really popular and sound great, some of those things have a lot of their good characteristics almost accidentally. I’m absolutely sure when we do a lot of these models, we know things about the circuit that the engineers didn’t. People would design something and then just put up with a little bit of an artifact without understanding it or caring about it, whereas we, if we want to recreate that, have to go down into the weeds and really understand it to a higher degree than sometimes the people who designed the gear.

For whatever reason, there are certain pieces of gear that have become super-popular. Whether it’s an intrinsic human characteristic to like those things, or whether it’s cultural weight, or familiarity, for whatever reason, they’re pleasing. And so, for people to have those sounds available to them I think is always going to be beneficial, until people just forget about those sounds, if that ever happens.

I think familiarity definitely leads to comfort, and comfort can lead to creativity just as well as being off-balance can. They’re two different kinds of things. It seems you were making the point that there’s a whole world of new stuff out there where you could make new sounds, and that’s probably true. I hope that people – even us – continue to do that kind of work, too. On the other hand, there are certain sounds people are used to and enjoy, and I think it’s good to have those sounds at their disposal, too.

These tools allow someone to make a recording with a grounding, that gives it a pleasant, familiar, comfortable sound. And then you still have the freedom to add your own novel ideas to the music. Maybe someone’s never used that piece of equipment before, but they’ve probably heard records that were made with that equipment.

It’s happened to me in development. For example, when we did our very first Neve EQ model, I worked out all of the math, designed all of the filters and everything. And then I got hold of the hardware, and started cross-checking the results of my design with the hardware. And I started playing music through both. It was really eye-opening to me. I made some adjustments, and thought – oh, it’s that sound. I know what that sound is; I’ve heard a lot of records that sound like that. But I never knew that that was a 1073 making that sound. But now I do. And it’s the same thing with the 1176 – you know, like I said, if you put that on a drum kit, you think, oh, it’s that, it’s this record and that record, and it’s a beautiful sound, and I always wondered how people got that sound. To me, it’s kind of exciting to have that comfortable feeling of thinking, I love that sound, and now I can do it.

By the way, I’m a design engineer, not a professional musician or a recording engineer, so these perspectives should probably be given very little weight. But I’m just telling you my personal opinion. There’s other people even within UA that probably should carry a lot more weight. I don’t know — I like creativity in music, but also a grounding in some aspect of it that sound comfortable and familiar.

Thanks, Dave. We’ll be taking a closer look at the Universal Audio solutions and where to begin using their stuff in your music, as well as their new FireWire-based Satellite for you Mac users. And in the interest of balance, I also have a very different take on modeling analog, from guy named Dave. I spoke with Dave Hill about HEAT, the Avid product; watch for that interview soon. HEAT is quite different from the UA stuff, but you’ll hear some familiar themes about the larger picture. Got questions for this Dave and UA? Thoughts on your own experiences with hardware and software? Let us know in comments.

http://www.uaudio.com/

TC Electronic’s PowerCore, a hardware DSP platform for audio processing is officially discontinued. You could easily continue working with PowerCore for some time, however. A recently-released 4.0 update provides full support for current OSes and hosts, even 64-bit ones, says the manufacturer. Nor is TC going anywhere: the company says it will continue to pursue audio interfaces and software (non-hardware-DSP) plug-ins. In fact, it’s a safe bet TC simply wasn’t getting enough business out of PowerCore to justify it, and will devote its resources elsewhere.

One manufacturer that is making it in hardware DSP that’s not Avid Pro Tools is Universal Audio. Their UAD-2 system remains a popular choice, and may become more so now that it offers FireWire support on Macs (something previously offered only by the PowerCore). Updated: James Grahame reminds me that it’s worth mentioning Sonic Core, who have their own Scope platform. As with UA, they’re basically a specialty shop, whereas TC has a number of other businesses. With S|C covering new-fangled digital ground like graintable and physical modeling, and UA focusing on first-party development of analog modeling and vintage emulation, you’ve got a good range of choice for DSP.

The crossgrade doesn’t give you a break on new UAD hardware, but it does give you significant bundles of free plug-ins for the platform. The breakdown goes something like this:

Solo line: Fairchild + Cambridge + Puletc-Pro ($377 value)
Duo: add LA-3A, DreamVerb ($675 value)
Quad: add Precision Limiter, EQ, multiband ($1322 value)

That’s on top of coupons for plug-ins of your choice included with each of those packages. And it includes the new FireWire Satellite systems.

None of this will entirely comfort existing users, though, I’m guessing. Our friend Oliver at Wire to the Ear writes a nice obituary – and I agree that I hope we see native ports of some of these plugs. (That or else maybe we’ll see TC introduce them on the iPad, the way the industry is going.)
TC Electronic Discontinues the PowerCore

The ongoing success of the UAD-2, even if for a specific niche, demonstrates that hardware DSP platforms aren’t necessarily going away. At the same time, it’s easy to understand that TC might focus instead on native software and specialized hardware (like stompboxes).

It’s difficult to describe Universal Audio’s plug-ins until you’ve tried them. It’s a bit like having chocolate sauce at your disposal, sonically speaking. Whatever your higher-level brain may have to say, somewhere deep in your mammalian brain, you hear only … mmmmmm. Chocolate. It’s the word I get from UA users, and I’ll also have an interview with UA to post next week in which we get deep into the philosophy of sound, software design, and modeling, a conversation that transcends any one product. They’re not for everyone – they demand a price premium, to be sure, versus rival CPU-native options, and those with access to a studio might just use the real thing. But for enthusiasts, they can be a delight.

The trick is actually having access to UA’s extensive library of analog-modeling effects. Even as CPUs have marched forward in performance, dedicated DSP chips haven’t stood still. That makes these number-crunching brains a convenient platform for sound, as they are on UA’s effects. If you’re not using a desktop PC with a PCI slot, that means you need an external bus through which the hardware and your computer can connect. ExpressCard is a big boon, but especially for Mac users, it’s been an obstacle. A powerful MacBook Pro is now available for around a grand, but to get ExpressCard, you now need the wallet-busting, messenger bag-cramming 17″ model. It’s clearly a deal breaker.

The UAD-2 Satellite DUO and QUAD should greatly broaden the appeal of the platform, making it accessible to inexpensive, recent iMacs and MacBook Pros. I’m sure Universal Audio is smiling at the idea that you can take that $1000-2000 premium and spend it on UA instead of Apple.

By connecting via Firewire 400 or 800 on a “select” (recent) Intel Mac, the Satellite gives you access to models of vintage hardware UA has made with the likes of Roland, Neve, Moog, Studer, dbx, Solid State Logic, Lexicon, Manley, Empirical Labs, Trident, SPL, and EMT.

On the PC, I’d still recommend the inexpensive ExpressCard option to get started. It’s cheaper, and won’t have the strict compatibility requirements. On the Mac, though, with $500 plug-in vouchers or 50 plug-ins included, the US$899 starting point on these models doesn’t look bad at all. In fact, someone out there may be over at the Apple Store already pricing out one of the “low-end” iMacs or MacBook Pros, as they represent desktop-class performance from just a couple of years ago, particularly with what you can do now with USB2 and Firewire.

UA also recently added Pro Tools compatibility, so that combined with FireWire options means just about anyone can now consider their platform. If it’s something that interests you and you’ve got questions, fire away in comments and I’ll pass them on to the engineers at UA.

http://www.uaudio.com/

UAD-2 Satellite DUO Product Page (the entry-level model … more specs on compatibility forthcoming)

Darnit. Now I mentioned chocolate, so I have to go eat a chocolate.