Universal Audio Powered Plug-Ins

[Siggidori]

Originally Posted by Anderton

Thanks for the suggestion! I've actually thought about that, but in some previous pro reviews, I made both downloadable audio examples and YouTube movies available. The number of YouTube views exceeded downloads by a huge factor - I think people liked the convenience of just being able to click on the video, and after watching it, didn't feel the need to listen to higher quality audio. I must say the YouTube audio is better than I expected, I upload a really high-resolution file so I guess the encoding doesn't beat it up too badly

But, I'll look into it because if one person is asking for something, that represents a lot of people who also want it but don't take the time to ask.

I think your right on all points
The option of being able to make timed comments on Soundcloud could be interesting though.. if people bother to use it that is

[UniversalAudio]

Hello Craig,

Thanks for the questions. It's great having a forum and format to answer these.

Regarding the Lexicon 224 Plug-In, you wrote:
There are four outputs, but I'm a little confused here. There's a Rear Outs control that swaps outs A and C (the "normal" stereo outs) and outs B and D, which can provide quadraphonic reverb. But, I have no idea if all four outs are available simultaneously, or how you'd use them...then again, I'm not particularly conversant in surround. Maybe someone from UA could describe a typical scenario for using the four outs?

All four outs are not available simultaneously per plug-in instance. However, users can use two instances of the Lexicon and set one to front and one to rear and successfully model the quad mode on the reverb. For this, the user would need to set up two busses, one for the front and rear separately. This would be helpful for a surround scenario.

Regarding 64-bit, you wrote:
Oh, and since the UA engineers seem pretty forthcoming about what's coming up in the future...are the rumors I'm hearing about 64-bit compatibility true? I've had no problems using the UA plug-ins with 64-bit Sonar using its built-in bit bridge, but wonder what kind of benefits native 64-bit compatibility would provide.

With over 50 plug-ins in the UAD catalog, 64-bit support is taking us a while to complete, but we know it is very important to our customers, so it's a very high priority for UA. So yes, while it's true that 64-bit compatibility is coming, a solid date is yet to be announced.

As you mentioned, there are no problems using UAD plug-ins with any 64-bit DAW, as long as the DAW supports a bit bridge. 64-bit drivers are already available for UAD-2 PCIe and Satellite cards for both Mac and PC.

With regard to the benefits of future UAD-2 64-bit plug-ins, they will primarily be related to improved workflow in the DAW itself, and no longer requiring a bit bridge. There will also be a slight improvement in CPU overhead as compared to 32-bit "bridged" plug-ins. Other companies with 32-bit plugs may have smaller memory addresses as compared to 64-bit, but that's not a problem with UAD plug-ins, since they run on our DSP Accelerator card.

As a workaround, many users use 32-bit mode, unless they are working on monster projects that require 64-bit capability.

So thank you for your question, and please know that we are definitely working hard on supporting customer needs, such as RTAS and 64-bit, while continuing to provide more top notch UAD Powered Plug-Ins, as well as future plug-ins from third parties.

Cheers,

Lev Perrey
Director of Product Management
Universal Audio

[Anderton]

Well I think we’ve pretty much covered the Lexicon 224, and I only have a few Summer NAMM videos left to upload to the HC Video Library (they’ve already all been uploaded to our YouTube channel), so let’s turn our attention to another powered plug-in—the Studer A800 tape emulator.

Everybody likes tape, right? Well, almost everybody. If I never have to record a classical guitar on analog tape again, that will be too soon. But, there are many types of music that tape flatters, as well as many individual instruments. I’ve talked to several top engineers who like to track to tape specifically for that reason, then bounce to digital to preserve the sound. It sure would be convenient, though, if you could simply take the tape recorder out of the equation, and get “that” sound in a totally digital environment. Apparently UA thought that was a good idea, too.

The Studer A800 plug-in is the result of a joint partnership between Studer and Universal Audio. UA was able to secure a “golden” unit from Ocean Way Studio, and set about doing the modeling. Studer provided the listening feedback—hey, if they don’t know what their units sound like, who does?—and gave it a final blessing.

Of course, if you’re going to model something, you’d better make sure it’s in good shape (just ask any company who does amp sims what they went through to get the best possible tubes for the amps they were modeling). As a result UA brought in consultant Jay McKnight, who’s an expert on tape machine setup and calibration, so that they’d be modeling a properly-tuned machine.

Here’s what part of the interface looks like. You can also open it up if you want to calibrate it yourself, as we’ll find out when we start getting into the audio examples and such.





But before we do, let’s touch on one more important subject: The A800 answers the question “so why do I need a DSP hardware card, given that native processing is so good?” The A800 is not shy about gorging itself on DSP power, and the whole point of a multitrack tape recorder is, well, being multitrack. If you try to put that many native processors into a virtual environment, you could get away with it up to some point; but then you’d need to bounce, or freeze virtual instruments, or make other compromises in order to accommodate processing on every track. By having dedicated DSP, you can run plenty of A800s (we’ll specify exactly how many shortly) without touching the power in your computer’s CPU.

Okay, enough background. Let’s load some tape, and fire this sucker up.

Oh, right...I don't need to load tape.

[Anderton]

Before getting into the individual controls, there are two “global” issues worth mentioning. The first is that the A800 doesn’t just give “tape sound” but also models the signal path and includes options for different tape formulations—a crucial aspect of emulation, as the tape you used, and how you calibrated it, had a major influence on the sound. The other is that as UA assumes you’ll be inserting the plug-in on multiple tracks, you can gang settings to maintain uniformity.

So how many instances can you insert, anyway? With a Quad UAD-2 card (the A800 won’t run on UAD-1 cards, sorry), what a coincidence—24 tracks! As you can see by the screen shot, each chip handles 6 instances.



Yes, the A800 was a 24-track machine, and now your DAW can do 24 tracks of A800 if you have a Quad card. Remember, all of this happens without loading down your computer’s CPU.

There are two control sections, the main front panel primary controls...



...and the secondary controls, which are basically the controls for which you normally needed a screwdriver and a steady hand.



Incidentally, when you’re showing the primary controls and you’ve selected to monitor the sync or repro heads, the tape reels rotate; it’s cute and all that, especially because the takeup reel changes based on the tape type you’ve selected,. But after a while, I started feeling like I was going to have a seizure or something so I’m very happy that if you click on the IPS label for the tape speed control, you can make the reels stop.

Let’s give an overview of the primary controls. Next post we’ll get into the secondary controls, and then present some audio examples.

• For the tape forumulations, you have a choice of 3M 250, Ampex 456 BASF Studio Master 900, or Quantegy GP9.
• Available speeds are 7.5, 15, and 30ips
• The four choices of tape calibration fluxivity are from +3dB to +9dB
• Input/output controls

On the secondary control page, there’s a button to calibrate these according to Studer’s original specs as you change tape speed, formulation, or EQ, but you can also tweak these trims yourself. The other front panel controls let you switch the virtual “input” among bypass, input electronics, sync head, and repro (playback) head.

[billybk1]

Originally Posted by Anderton

So how many instances can you insert, anyway? With a Quad UAD-2 card (the A800 won’t run on UAD-1 cards, sorry), what a coincidence—24 tracks! As you can see by the screen shot, each chip handles 6 instances.

Hey Craig, great pro review BTW! I just wanted to point out that the (6) instances per chip is for stereo applications. When used on mono material you can get (10) instances per chip and (40) on a QUAD @ 44.1k (with the Limit DSP Load option set to at least 99%). With many, if not most tracks using mono type sources (kick, snare, bass, vocal, etc....) you can get more total instances or at least have unused DSP left over for other plug-ins. One other thing, some DAW host apps (SONAR & REAPER come to mind) require that you install and use the specific mono (m) instances on mono tracks to get the reduced DSP benefit. While other DAW hosts (Cubase & Nuendo to name a few) will allow you to use the standard installed stereo version on a mono track and the reduced DSP happens automatically. Using your DAW, look at the DSP hit of a stereo instance, when used on a mono track and compare to the UAD Plug-in DSP usage charts to see what you should be seeing for mono. Something to be aware of when you have a finite amount of available DSP and you want to minimize the DSP hit so you can use even more plug-ins.

Cheers,

Billy Buck

[Anderton]

Great comments! I was aware that you save DSP with mono, but didn't know that you MUST install mono instances on mono tracks to get the DSP savings. I've instantiated stereo versions and they work fine on mono tracks, but now I know better

Any wonder why I like the pro review format? And I even get to play with the stuff and create audio examples!

[Phil O'Keefe]

 

Originally Posted by UniversalAudio

Hey Phil - The upcoming RTAS updates will be cross platform for both Mac & Windows.

 

That's GREAT news! My interest level just doubled.

[Anderton]

Now let’s look at the secondary controls, as they make a huge contribution to the plug-in’s authenticity. But to fully appreciate what’s going on, we need to explain a bit about tape bias.

I used analog magnetic tape exclusively for over 20 years, and being a solderhead, got into the whole process of aligning and biasing tape. Recording to magnetic tape is anything but straightforward: you need to suspend a zillion little magnetic rust particles in plastic, tell them how to line up as they get dragged past an electromagnet, and feed them with a high-level supersonic signal just so the audio doesn’t sound like a buzzsaw (and you thought clock frequencies only related to digital...). Tape is temperamental, subject to environmental conditions, and expensive. So, I was very happy when digital recording became affordable—but the sound of tape is something else, and one of its cooler aspects is that the sound is customizable through a tape recorder's bias and EQ controls.

Like the plug-in itself, these controls were usually hidden behind a panel. You’d adjust them with a small screwdriver, and once set, you didn’t want prying hands to change them. Optimally, you’d check the bias and EQ for each reel of tape due to manufacturing differences, although some studios were lazier and readjusted only when changing brands or production runs.

Bias is the high-frequency signal that’s injected into the head to help overcome hysteresis (the tendency of tape to hold on to its magnetization until forcibly reset by the bias signal). The challenge is that bias influences frequency response, distortion, and noise level, but each requires different optimum amounts of bias. Therefore, you have to make tradeoffs.

Underbiasing gives the best high-frequency response, but is more prone to distortion. Slight underbiasing added an almost “exciter”-like effect if you got it right, as you had nice highs with a little crunch. Overbiasing reduced distortion, but resulted in less high-frequency response and if you went too far, the high frequencies would saturate sooner and you’d also compromise the signal-to-noise ratio.

All things considered, I generally went more toward over- than under-biasing to minimize distortion, and compensated for any lack of high frequencies with the EQ options that were also a part of tape calibration. Another option was to “cheat” and boost the highs going into the tape, then cut coming out, to reduce the amount of tape hiss.

Get the picture? How you calibrate tape is as much art as it is science, and one of the coolest aspects of the A800 is it brings that art to the plug-in world. Let’s look at the controls, going from left to right.



The HF control determines the high-frequency boost going into the tape, and below it, the bias control does the tricks alluded to above. Next you have separate high-frequency and low-frequency EQ for the sync and reproduction heads, and that requires a bit of explanation as well.

Early tape recorders couldn’t overdub because tape went past a record head first, then a playback head some amount of time later (the exact time depended on tape speed and physical construction; for example, the famous Sun Studios slapback echo sound was about 150ms due to picking up the echoed signal from the playback head). So if you wanted to overdub, when you heard the signal from the playback head you recorded at the record head, and playing them back at the same time caused a delay. DAW fans can think of this as the ultimate in lack of path delay compensation

Les Paul came up with a solution to this problem by creating a four-head deck with successive playback, erase, record, and playback heads. The signal from the playback head fed into Paul’s headphones, and was recorded into the record head at the same time Paul was playing—basically, sound-on-sound recording. Although Les Paul is often credited with the invention of sel-sync (the process of using part of the record head for playback, and a term trademarked by Ampex), that honor actually goes to Ross Snyder, an engineer at Ampex, who came up with the sel-sync head in the mid-50s; another engineer, Mort Fuji, did the circuit design. There’s no doubt, however, that Les Paul was an inspiration for sel-sync, and the first recorder with that technology was delivered to him.

Anyway, record heads have wider gaps than playback heads, making them less well-suited for high-frequency reproduction. Doing overdubs on early machines with sel-sync sounded like you were listening through tin cans, but the quality was good enough to use a reference for playing against. Over the years the technology improved to where the difference between the sync and repro heads was almost identical, but they always had a sonic difference to one degree or another, so both options are available in the A800.

Moving right along, we have hum and hiss controls, which of course were not variable on the original A800 but an inherent, uh, “feature.” These are for the purists who don’t think it sounds like tape unless there’s hum and hiss; I’m not a purist...next.

The NAB/CCIR buttons provide your choice of two different equalization curves. Another button turns the noise off or on (I have it taped to off, using virtual masking tape of course). Auto Cal is cool, because if you’re not into messing with the bias and EQ, you can just click “on” and you get what Studer considers the optimum calibration for whatever tape type you’ve selected. Click on “off,” and like yours truly, you too can obsess over details.

Finally, there’s the brilliant Gang Controls option, which as mentioned previously insures that if you’ve inserted a bunch of these into your tracks, tweaking one can tweak the others similarly so that you have a consistent tape sound across all tracks.

Did I hear someone say "Cool, now how about some audio examples?" Okay, we'll do that next. But first, any questions?

[MikeRivers]

Since you prodded me into being picky . . .

Originally Posted by Anderton

Bias is the high-frequency signal that’s injected into the head to help overcome hysteresis (the tendency of tape to hold on to its magnetization until forcibly reset by the bias signal).

Bias and hysteresis are related, but not directly. The relation between the magnetic field strength (which is directly related to electrical current in the record head) that's trying to magnetize the tape and the magnetization of the tape has, like a vacuum tube, a range over which the relationship is fairly linear. Below the linear range, increasing the magnetic field strength makes very little change in the tape magnetization until you get to the point where it sort of "kicks into gear." When you reach the top of the linear range, the tape is saturated, meaning that no matter how great the field strength, the tape won't magnetize any further.

What bias does is centers the audio current in the linear range so that the changes in current going into the record head will result in the same changes in current coming off the play head. Bias was originally DC, a constant magnetic field which added to the alternating audio signal. The thing that made magnetic recording worth using for audio was the discovery of AC bias.

AC bias offers the improvement in linearity but with lower noise than DC bias. The reason why AC bias works is related to hysteresis. There's some "stickiness" as you move along the current vs. magnetization curve. When you're going up along the curve and change direction, the magnetization doesn't go back down along the same curve, but turns around slowly and lags behind the upgoing curve.



The combination of the upgoing and downgoing curves is called a "hysteresis loop" (unrelated to a ground loop or Fruity Loop). By using AC bias, you're constantly moving between the two curves and the average stays pretty close to a curve that follows the center of the loop.

The challenge is that bias influences frequency response, distortion, and noise level, but each requires different optimum amounts of bias. Therefore, you have to make tradeoffs.

Underbiasing gives the best high-frequency response, but is more prone to distortion.

Overbiasing reduced distortion, but resulted in less high-frequency response and if you went too far, the high frequencies would saturate sooner and you’d also compromise the signal-to-noise ratio.

This effect on high frequency response is known as "self erasure." By adding too much bias, you randomize too many magnetic poles, effectively erasing the tape. It takes a lot more field strength than what bias provides to effectively erase the tape, but the highs go first, and the more bias, the more high frequency erasure. You can't apply enough bias (at least not with conventional recording heads) to push the current into the upper non-linear region.

If the tape saturates, it's because of the audio, not the bias. Though you can get it closer to the saturation with more bias, the self-erasure usually takes over before you get very far into the non-linear region.

Too little bias and you don't get the audio up into the linear region of the curve. In this region, high frequencies are more effective in magnetizing tape than low frequencies, hence the appearance of increased high frequency response. The "crunch" is a result of the non-linearity at the bottom of the curve.

All things considered, I generally went more toward over- than under-biasing to minimize distortion, and compensated for any lack of high frequencies with the EQ options that were also a part of tape calibration.

There's another effect of bias that's known as "modulation noise" or "bias rocks." I don't have a good explanation of what causes this, but it adds a sort of gurgling sound to the recording. It's another of the trade-offs when adjusting a tape deck.

The formulation of the tape oxide material is where the trade secrets lie. The goal, it would seem, would be to find a formula that gave the best linearity, lowest noise, and lowest distortion. This is one of the things that distinguished Agfa 468 series tape and made it popular particularly for recording acoustic music where you want the maximum dynamic range without driving the tape into saturation. Most other tapes have each of these characteristics optimized at a different bias level.

One of the things that Mike Spitz of ATR Service emphasizes and teaches in his tape recording alignment class is that tape doesn't have one particular sound, but there's a range of sounds that you can get from a particular piece of tape depending on how you adjust the recorder.

Next you have separate high-frequency and low-frequency EQ for the sync and reproduction heads, and that requires a bit of explanation as well.

Anyway, record heads have wider gaps than playback heads, making them less well-suited for high-frequency reproduction. Doing overdubs on early machines with sel-sync sounded like you were listening through tin cans, but the quality was good enough to use a reference for playing against. Over the years the technology improved to where the difference between the sync and repro heads was almost identical, but they always had a sonic difference to one degree or another, so both options are available in the A800.

Tape equalization is a pretty deep subject. Since the current produced when you throw a magnet through a coil of wire is related to the speed that the magnetic field cuts across the coil, because high frequencies change 'direction' faster than low frequencies, assuming that recording current remains uniform over frequency, there is a natural rise in playback frequency response.

There's a gap in the head pole piece which is what the magnetization bridges, and that's where the playback action takes place. But when the wavelength of the recorded signal becomes as short as the gap is wide, the magnet no longer bridges the gap and the high frequency response drops like a rock. The head is designed so that at the highest tape speed (which, along with the frequency, determines the wavelength that's recorded on tape) the highest frequency we want to record will still bridge the gap. This is why increasing the tape speed extends the high frequency response of the recorder. High frequency EQ can't compensate for the gap loss, but it compensates for the high frequency rise with wavelength.

Low frequency EQ is necessary for a different reason. At low frequencies, the recorded wavelength approaches the length of the part of tape in contact with the head. The magnetic field takes a "short cut" through the head pole pieces which adds to the current induced when it passes through the gap in the head, resulting in some low frequency irregularities usually called "head bumps." Bass players really like this because the LF playback EQ is at a single frequency so the low frequency response is always a little irregular, and some tape machines are particularly favorable to the low end of a bass. Dan Dugan made an 8-channel parametric low frequency equallizer designed to be used to smooth out the head bump on the narrow gauge tape decks but they didn't sell very well.

Studer has always used a different head design for the record and play heads, hence needed to help out the record head with EQ when it's used for sync playback. Ampex discovered that at 15 ips, they could design a head that worked equally well for recording and playback, so the MM1000, 1100, and 1200 multitrack Ampex recorders used the same head in both positions, allowing them to use the same electronic design for both "normal" and sync playback.

TASCAM used the same design for their narrow gauge multitrack recorders, and Fostex took it a step further. They use a single head for recording and playback, saving the cost of one head, assuring that there will be no difference between sync and playback, but making adjustment difficult because you can't "see" your record adjustments in real time.

Personally, I think trying to simulate the deficiencies of a tape deck is pretty silly. There are better ways of being creative, but nostalgia has a big influence on how we think things should sound.

[Anderton]

Originally Posted by MikeRivers

Personally, I think trying to simulate the deficiencies of a tape deck is pretty silly. There are better ways of being creative, but nostalgia has a big influence on how we think things should sound.

In one sense I agree, but those deficiencies are part of the "sound" that people associate witih tape. When I post the audio examples, you'll hear the distinctive "tape crunch" that's different from electronic saturation, and which some people really like, particularly on drums. Another issues is that when a company decides to model something, they can make a decision to model selected aspects, or go all the way.

You may have missed a comment about the 224 toward the beginning of this thread:

I had reviewed the Waves Aphex emulation and made a snarky comment about why anyone would want to include noise in a plug-in. I mean, it's a no-brainer, right? Why model noise when you don't have to?

But he pointed out two things. First, for some people, that WAS a part of the sound and therefore, for the emulation to be accurate, it needed to be able to offer that sound. Besides, as with the 224, you could disable it. Second, and more intriguingly, some listeners preferred the sound with the noise and not for nostalgic reasons - they thought it added a useful sonic character that was more interesting than a sound without noise.

Prior to that conversation, I would have used this post to say "Hey UA, why do you emulate noise when you don't have to?" But now I know better. I presume UA's motivations for being able to add system noise are similar.

So I assume that from UA's standpoint, they decided "Well, if we're going to model a tape recorder, we're going to model a tape recorder." Somewhere, there's probably some customer who's using some esoteric feature of the plug-in that I'd never use. Granted, I'm sure most users will just click on auto cal, hit the "tape" hard, and that will do what they want. On the other hand, being able to re-visit the sounds you can get from over-biasing and under-biasing tape is useful. I definitely used those options with real tape (or is that "reel tape"?) to alter the sound, so there's no reason I wouldn't want to do that with a plug-in that's intended to duplicate tape.

[MikeRivers]

Well, there's no accounting for taste. I suppose that a noiseless tape recorder would make one wonder. But I guess I don't really get the nostalgia for the "sound of analog tape" since it's something that I've always tried to eliminate as much as possible. I don't think my digital recordings sound worse than my analog recordings because I'm not using tape. It has more to do with the way we record today, the workflow, the gear, the emphasis on each track being as transparent and detailed as possible (words we didn't use very often 35 years ago), and the unlimited number of do-overs.

If UA did this right, they'd make the plug-in so that restricted the DAW to 24 or 16 or 8 tracks and made it really easy to bounce tracks and punch in without fiddling with the DAW's busier user interface.

But I digress.

[billybk1]

I just came across this illuminating article "Modeling Analog in a Digital Age: a Conversation with Universal Audio's Chief Scientist":

http://createdigitalmusic.com/2011/0...ief-scientist/


In the latter part of the article, Dr. Dave Berners goes into quite some detail about some of the methods and techniques UA used to accurately model the Studer A800.

Cheers,

Billy Buck

[Anderton]

Thanks for the link, good stuff.

[Anderton]

I was wondering how I was going to get this across in an audio example, but actually, I think this one really gives a good idea of what's going on.

I used a drum loop from the Discrete Drums series, but chose one with a tom part and no cymbals or snares. The "sound of tape" is most obvious with percussive sound sources that have low harmonic content, and a tom loop works very well for that.

I used the 456 tape option at 15 IPS and set the calibraton to +7.5 because, well, I'm like that

After recording each of the four examples, I adjusted their levels for the same peak level. Therefore, if one of them appears louder, it's due to a higher average level because of the tape compression, not louder peaks.

The first four measures are the loop by itself, with no processing.

The next four measures are the loop with "neutral" tape processing - not much crunch, just a touch of "tapeness." Nonetheless, it gives a nice "lift" to the sound that will really be obvious if you listen on headphones.

The next four measures have heavy tape crunching. This is one of the tape sounds that's burned into my brain from pushing analog drum sounds hard on tape so they could cut through better on a mix.

The final four measures are at least to me, the most interesting of them all because I wanted to see if I could duplicate the sound that results from a slight amount of underbiasing, which is a technique I used quite a bit (particularly for acoustic guitar). This is another tape sound that's burned into my brain, because I always made this adjustment by ear. And wouldn't you know, I could make that same adjustment, with the same sonic results, using the A800 plug-in. I thought that was pretty effing impressive.

[Anderton]

Originally Posted by MikeRivers

I guess I don't really get the nostalgia for the "sound of analog tape" since it's something that I've always tried to eliminate as much as possible.

Same here...mostly...but...

What I disliked the most about tape was hiss, followed by modulation noise and flutter/wow. Take those away, though, and tape becomes a signal processor with a very complex set of interactions among level, distortion, and frequency response.

I'm a "don't look back" kinda guy, and after having digital and analog recorders set up side-by-side for a while, came to the conclusion that I liked digital better. The problem was that analog had characteristics I could not remove, whereas I could add characteristics to digital if I wanted.

However, playing with the A800 plug-in is making me re-consider the value of tape as a processor. If UA had called this plug-in a Dynamic Distortion Response Processor, never mentioned tape, and included the same complement of controls but didn't include references to tape or tape machines, I'd think it was a pretty cool processor. Now, I wouldn't use it on EVERYTHING, because that was the problem I had with tape - I HAD to use its "processor" on everything. But the ability to use the processing as I did in the 2nd and 4th group of four measures in the previous post's audio example is pretty cool; it would be extremely difficult to re-create those sounds with conventional EQ or saturation.

So even though we haven't covered every permutation and combination of A800 controls, I've pretty much come to a conclusion: Forget about tape, nostalgia, people who romanticize tape because they didn't have to demagnetize/head lap/align/calibrate/lubricate/etc. etc. the $%^&* things, and forget about modeling. Does this plug-in make useful, unique sounds? The answer AFAIC is definitely yes. I can see myself using this judiciously on multiple tracks in a mix.

[MikeRivers]

Originally Posted by Anderton

So even though we haven't covered every permutation and combination of A800 controls, I've pretty much come to a conclusion: Forget about tape, nostalgia, people who romanticize tape because they didn't have to demagnetize/head lap/align/calibrate/lubricate/etc. etc. the $%^&* things, and forget about modeling. Does this plug-in make useful, unique sounds? The answer AFAIC is definitely yes. I can see myself using this judiciously on multiple tracks in a mix.

I surely won't disagree with that. A signal processors is a signal processor. You use it when you have an accurate recording and you want to change it to something other than accurate. "It makes that track stand out (or blend in) better" is, to me anyway, a better loosey-goosey description than "it makes it sound like tape." At least it would, coming from someone who has plenty of tape experience like, well, you.

You work on a wider variety of projects than I do. I think I can safely say that if I had that plug-in, I'd never use it. I use EQ judiciously, occasionally a bit of compression, and some reverb. That's about all.

Only once have I ever used pitch correction. That was on an old time fiddler, and that was just two notes in a tune, nearly 20 years ago. I rented an Eventide Harmonizer, set up a power supply as the external control voltage input, calibrated the pitch shift to get that note right, and did a tape-to-tape copy with the Harmonizer in between. When the note came along, I switched on the power supply, the Harmonizer adjusted the pitch, and that was it. I took a shower afterward.

Nothing sounds worse than a banjo with flutter, and the kind of music that I record has always sounded better when recorded digitally, as far back as the Videotape+PCM adapter, the first digital recording system that I owned (and still have, in working condition). I don't think anything I've recorded would sound better if it was run through an analog recorder, a good one or a crummy one. But I understand that there are some things on which such abuse is effective.

I suppose that there are enough people who are either nostalgic for tape (really) or nostalgic for the period when tape was the only thing we had, that calling it an A800 simulator and putting that set of controls on the user interface is a good marketing move.

[Anderton]

Well, quite a few people have downloaded the A800 clip. I've been trying to think if there was anything else I should post in terms of the A800, but really, I think that one clip kinda says it all: You can indeed get "the tape sound," from subtle to nasty, and you can mess around with the bias and EQ if you're so inclined. We've covered the ganging aspect, which is pretty important, and had a pretty decent discussion about the gestalt of tape...so unless anyone has any more questions or comments, how about we move on to another of the powered plug-ins and dissect that?

[Siggidori]

No Soundcloud yet? .... I'll download the sample now ... but I would like to encourage you to try/use Soundcloud to post samples. I mean...even Universal Audio uses Soundcloud
http://soundcloud.com/universal-audio/

I even posted some samples myself when I was first trying out the Studer A800 plugin. Here you can check out the samples if you like: http://soundcloud.com/siggidori/sets...n-test/s-8HZCp

hmmm doesn't seem like it's possible to post embedded media in this forum.

Thanks for all the tests and the writings!

[Anderton]

I'm baaaack...as if forest fires weren't bad enough, I was without internet for a week because a series of lightning storms blew out four junction boxes between the main phone line and my house. Trying to maintain a web site from an Android is, shall we say, "challenging."

Anyway, it's kid in a candy store time, because it's time to pick the next plug-in to cover. I was thinking of doing the Manley Massive Passive next just because it's so effing impressive, but I've read some comments online that basically said "Yeah, the UA stuff is good, but you really have to buy extra plugs, you don't get much with the board itself."

Well, let's briefly put that one to rest before moving on to the Massive Passive. (To refresh your memory, post #4 lists the included plug-ins.)

The audio example is from a remix I'm doing of a song by Alyssa Atherton, a singer/actress. I asked for tracks that were relatively unprocessed; the first half of the example is the original vocal track for the first verse, while the second half is after processing by three of the UA's bundled processors.

I felt the vocal needed three elements, shown in the following image.



I added the LA-2A compressor to make the vocal a little more intimate and bring the voice in closer to the listener. Also, it seemed to me the voice was somewhat harsh--a perfect candidate for the Pultec, which I've always felt was a great EQ for general tone-shaping. Finally, some ambience seemed in order; I wanted a warmer kind of hall sound, and the RealVerb Pro should not be underestimated. It gives a solid, realistic sound quality without fluttering or periodicity...listen, and judge for yourself.

Remember, none of these are hitting your CPU. With stereo instances, the LA2A takes about 4% of one chip's DSP, the Pultec about 6.2%, and the RealVerb Pro about 10.4%.

Also, something weird happened in the process of bouncing the track down through the effects to make the audio example: I had fast bounce on by accident (force of habit and all that) instead of real-time bounce, and it still worked fine. Maybe it's because the example was short, but in any event, it worked. I thought that wasn't supposed to be possible

[Anderton]

So how are the LA-2A and Pultec as bus processors? Well, as Alyssa isn't local (and there are time-shift issues when she spends long days on movie sets), I've been sending rough mixes-in-progress to give her an idea of where the song is going. As these are just temporary, I don't want to spend time "mastering" them but I do want to add a little bus processing just to give a little more of a commercial veneer. I also live with these roughs for a day or two myself to listen for issues (like, where is that weird frame drum sound coming from? I don't remember hearing it on any of the tracks...).

The Pultec has been great for adding a little upper mid clarity, and trimming the lows a bit to tighten up the low end. The LA-2A is traditionally thought of a compressor for vocals, bass, etc., but its warm, "analog" characteristic flatters some types of program material. As the track combines synth sounds with acoustic/electric instruments, the LA-2A seems to tone down the synths a bit so they match better with the other instruments.

This example shows the bus processing in action. The first part is unprocessed, the second part is the same material but with the Pultec and LA-2A. The difference is fairly subtle - it doesn't hit you over the head - but it definitely gives the song a "lift."

[Anderton]

Hey everybody! New stuff from UA. I'm particularly interested in the ATR-102 and Vitalizer, and also, how the plugs work with Pro Tools now that the wrapper is gone. (Unfortunately I won't be testing this with Lion because I haven't upgraded my Mac to Lion yet, and won't be until some of the incompatibility issues are resolved.)

Anyway, following is the official press release but note that there's also an introductory special for the month of September on the UA site.

The long-awaited UAD Powered Plug-Ins v6.0 has landed, with UA's stunning emulation of the Ampex® ATR-102 Mastering Tape Recorder, Direct Developer plug-ins from Brainworx® and SPL®, significantly enhanced Pro Tools® integration, and support for Mac® OS X 10.7 Lion.

New v6.0 Software Features:

Ampex® ATR-102 Mastering Tape Recorder Plug-In
The ATR-102 plug-in can turn music recordings into records, faithfully replicating the unique dynamics, saturation, head configurations and tape formulas used on the best-sounding mastering tape machine in history.

Brainworx® bx_digital V2 EQ Plug-In
This integrated audio toolkit provides precise 11-band equalization and M/S (mid-side) processing — letting you add presence and transparency to mixes.

SPL® Vitalizer MK2-T Plug-In
A spot-on emulation of the popular tube-based hardware unit, this longtime staple works in both the time and frequency domains to effectively unmask overlapping sounds and bring life to mixes.

Significant Pro Tools enhancements
UAD v6.0 streamlines the UAD plug-ins workflow in Pro Tools, via removal of the VST-to-RTAS wrapper, and numerous control and automation improvements.

Mac® OS X 10.7 Lion compatibility
UAD v6.0 software ensures that customers running Mac OS X 10.7 Lion will have a seamless experience with their UAD Powered Plug-Ins.

[Anderton]

Here's some more info from a separate release about the Pro Tools enhancements.

• UA-developed RTAS plug-ins, replacing current VST-to-RTAS Adapter
  
• Full support of all Pro Tools control surfaces, including customized control layouts
  
• Fully automatable plug-in parameters, with values and ranges displayed in automation lanes and on control surfaces
  
• Plug-in names clearly displayed in the Pro Tools Mix and Edit windows
  
• Plug-ins sorted by category in the Pro Tools plug-ins menu
  
• Quick installation process
  
• Many other workflow improvements

I don't know about you, but I think that's pretty significant stuff. I did a video at NAMM awhile back with a UA representative about how they were planning more collaboration with Avid, and it seems that really has come to fruition.

[Siggidori]

This is good stuff. I'm looking forward to trying these new features and plugins.

[Anderton]

There was a REALLY interesting comment on how the whole "sounds like a record" could be taken one step further, posted by bookumdano2 in my Sound, Studio, and Stage forum (I've included excerpts here). But be careful what you wish for...seems like the folks at UA like challenges...

Model a vinyl cutter lathe. And I'm not even too sure it can be done.

For my interest.. not for that cliche vinyl sound that kids can not fully understand... but with real controls modeled so that I can .. cut hot 45s. In the virtual world.

It's so dang expensive to go over to Bernie Grundman, but there is such an art to standing there, experimenting with pushing the process just far enough go get that cool, over the top, magic of old 45s. Not the scractches (think we have enough plugins for that), but the vibe. Something even 33 1/3rds don't have when stacked up against 45 mastering.

So far in 2011, tape doesn't get you that last half mile, plug ins don't get you there. The only way you get there is if you stand there with the cutting engineer, and push the cutter until "I'm givin' it all she's got Captain". And then throw out the idea and start over for another couple of hours until you hit the magic zone.

The way it is now, you get a cut you like the sound of, make a lacquer, and you're already in the $ past the point of no return. You either play the lacquer to capture it back to digital archiving (ruining the lacquer), or high tail it over to a pressing plant to spend even more $ making 10,000 records. And that's for one song. Or two in the case of 45s.

And I don't even think most artists even spend time with the cutting engineer to participate in that vital part of the process. At least for those who still do vinyl.

Maybe a software version of this will be something out of Celemony's mind at some point rather than a company like UAD. But this is something I would love to dive into. It's one of the few secret-guru areas remaining.

[Siggidori]

:-D now that would be interesting! Haha I'm convinced they could do it ... be it UA or Celemony