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Monday, July 22, 2019

Q. What is optical compression?

By Paul White
Focusrite Trak Master.Focusrite Trak Master.
The Focusrite Trak Master and Behringer Composer Pro are two affordable compressors which use optical gain control elements.The Focusrite Trak Master and Behringer Composer Pro are two affordable compressors which use optical gain control elements.
The Samson S*Com, however, uses a VCA.The Samson S*Com, however, uses a VCA.
Lately, there seem to be numerous affordable hardware compressors on the market, and I've noticed that many of them (the Platinum Focusrites and the Joemeeks, for example) are described as optical compressors. What's the difference between optical compressors and other types of compressor, such as VCA, FET and valve compressors? Are there any relative merits to these different types of compressor and are they suited to any particular applications?

Luke Ritchie

Editor In Chief Paul White replies: After microphones, nothing stirs up a group of music professionals so much as a discussion about compressors. Essentially, compressors are gain-riding devices that monitor the level of the incoming signal and then apply gain reduction in accordance with the user's control settings. Given this simplistic explanation, shouldn't all compressors sound exactly the same, in the same way that faders tend to?

Clearly compressors don't all sound the same, and there are a few good technical reasons why. Perhaps of less importance than some people might imagine is the gain control element itself, which can be a tube, a FET (field effect transistor), a VCA (voltage-controlled amplifier), an optical photocell arrangement (a light source and a light detector) or even a digital processor. Certainly all these devices add their own colorations and distortions to a greater or lesser extent, but what influences the sound most is the way the ratio and envelope characteristics deviate from theoretically perfect behaviour.

In an imaginary, perfect compressor, nothing happens to the signal until it reaches a threshold set by the user, after which a fixed compression ratio is applied. For example, if the compression ratio is set at 4:1, for every 4dB the signal rises above the threshold, the output rises by only 1dB. A modification to this is the soft-knee compressor where the ratio increases progressively as the signal approaches the threshold, the end result being a less assertive, less obtrusive form of compression.

Many classic designs don't in practice act like this perfect compressor however, as their compression ratio may vary with the input signal level. For example, some compressors work like a perfect soft-knee device until the signal has risen some way above the threshold, then the compression ratio reduces so that those higher level signals are compressed to a lesser degree than signals just above the threshold. 

The reason for this change in ratio is simply that many early gain-reduction circuits don't behave linearly, especially those using optical circuitry as the variable gain element. The components themselves are non-linear so when, for example, you combine a non-linear light source with a non-linear light detector, the composite behaviour can be quite complex and unpredictable — however, history has buried those optical circuits that didn't sound good, so we're now left with those that happened to sound musical.

The other very important factor governing the sound of a compressor is the shape of the attack and release curves. While a modern VCA compressor can be made to behave in an almost theoretically perfect way with a constant ratio and predictable attack/release curves, many of the older designs had very strange attack and release characteristics, and, in the case of optical compressors, this was originally due to the relatively slow response of a light and photocell compared with a VCA.

For example, the now legendary Universal Audio 1176 combined a fairly fast attack time with a multi-stage release envelope. Conversely, the Teletronix's LA2A's rather primitive optical components resulted in a slower and quite non-linear attack combined with a release characteristic that slowed as the release progressed. Indeed, perhaps the reason the traditional opto compressor has so much character is that there are so many places in the circuitry that non-linearities can creep in.

Having said that, some modern optical compressors use specialised integrated circuits that incorporate the necessary LED light source (which has largely taken over from the filament lamps and electroluminescent devices used in early designs) and detector element in a single package that incorporates feedback circuitry to speed up the response time and to linearise the gain control performance. Indeed, some of these are so well behaved that they can sound almost like VCAs, but using clever design, it should be possible to recreate the old sounds as well as the new using contemporary electronic devices, or imaginative software design come to that.

It's harder when it comes to saying what type of compressor is best for which job, but in very general terms, a well-designed VCA compressor will provide the most transparent gain reduction, which is ideal for controlling levels without changing the character too much. However, a compressor that allows high-level transients to sneak through with less compression can also sound kinder to material than one that controls transients too assertively, which is why some of the older, less linear designs sound good. 

That's not to say modern designs can't sound good too though — Drawmer pioneered the trick of leaking high frequencies past the compressor to maintain transient clarity while other manufacturers, such as Behringer, use built-in transient enhancers or resort to equally ingenious design tricks.

Optical compressors, especially those that don't use super-well-behaved integrated optical circuits (or those that use them imaginatively) usually impose more of their own character on the material being treated, making it sound larger than life. In this context, the compressor is as much an effect as a gain-control device, and such compressors are popular for treating vocals, drums and basses. The Joemeek and TFPro compressors fit this 'compression as an effect' category as they use discrete LEDs and photocells in a deliberately non-linear topography that's really a refinement of that used in some vintage designs.

Digital compressors and plug-ins can reproduce the characteristics of vintage classics, but only if the designers successfully identify those technical aspects of the original design that make it sound unique. If they don't, you end up with an approximation or caricature rather than a true emulation.

Published September 2003

Friday, July 19, 2019

Q. Do I need balanced patchbays?

By Mike Senior
I am currently setting up a home studio, which I'm hoping to eventually turn into a professional facility, based around a Soundtracs Topaz desk, three Egosys Wamirack soundcards and a Pentium 4 PC, with numerous synths, samplers, effects and other outboard gear. I'm now looking to wire everything together using patchbays. Bearing in mind that my console does not accommodate balanced outputs and insert points (the only balanced connections on the console are at the input stages of all channels and the effects returns), can I use unbalanced patchbays, thereby simplifying the patch lead requirements? If you are going to suggest a balanced patchbay setup, could you describe where to connect and disconnect the ground/screen connections to avoid ground loops.

SOS Forum post
Installing balanced patchbays (as opposed to unbalanced ones) makes dealing with hum much, much easier.Installing balanced patchbays (as opposed to unbalanced ones) makes dealing with hum much, much easier.

Reviews Editor Mike Senior replies: It sounds like you've already invested a good deal of money in the gear, and there's certainly enough there to produce high quality audio. However, if you're going to retain audio fidelity with so many pieces of equipment working together, I would try to balance as many of your analogue audio cables as possible. Even in my more modest home setup mains hum and induced noise are problems (which have taken upgrading to balanced connections to sort out), so if you're ever hoping to use your studio professionally you don't really have a choice. Even in commercial studios a lot of time can be spent dealing with hum, so it's worth planning for it now, in my opinion. Unbalanced connections are fine for a smaller setup than yours, but, at the stage you're at, I reckon it's a recipe for disaster.

The great thing about balanced connections is that lifting the earth connections between equipment to break earth loops is comparatively easy — just disconnect the earth wire at one end of the signal cable — but with unbalanced gear the same trick very rarely works in practice and will often make things worse. If you're wondering how to decide where to make this disconnection in your system, Mallory Nicholls suggested that his preferred method was "to connect cable shields at equipment outputs and not at equipment inputs" in his Studio Installation Workshops in SOS September 2002and November 2002. So, disconnect the shield just before it reaches the equipment inputs. If you're using any moulded cables, then you might have to perform some modification on the patchbay, but this is not usually too difficult to work out — it's what I did, and it's worked very well so far!

To incorporate any unbalanced devices within the balanced system, you have two main choices: unbalance at the input to the unbalanced device — connect one of the balanced signal wires to the jack sleeve, along with the earth wire, and don't disconnect the earth wire elsewhere — or use a balancing transformer to do the interfacing. The second solution is more costly, but may be the only way to solve any hum problems which the first solution may create. Maybe you'll be lucky and not get any appreciable hum using the first system, but if you do get hum then have a look at the Ebtech Hum Eliminators — there's an eight-channel one for £295 which would probably isolate enough connections to sort remaining hum problems out. I've only needed to use a two-channel one to sort out a persistent hum in my system, but yours is much more complex, and all of it will be connecting to the central desk, which multiplies the potential for hum.

Published September 2003

Monday, July 15, 2019

Q. What's the right type of Rockwool?

I'd like to use some Rockwool in my studio to improve the acoustics, but this is the first time I'll have used it, so I could do with some pointers about how to work with it. What is the best density for a good, fairly wide‑spectrum absorber? I have found some quite cheap Rockwool that is 100kg/m3. Is that any good?

Via SOS web site
Remember to coat Rockwool with an acoustically transparent material to trap stray fibres, as shown above. Also, placing acoustic foam on top of Rockwool panels, as in the picture below, makes a far more effective acoustic absorber as the foam absorbs high frequencies that the Rockwool does not. Remember to coat Rockwool with an acoustically transparent material to trap stray fibres, as shown above. Also, placing acoustic foam on top of Rockwool panels, as in the picture below, makes a far more effective acoustic absorber as the foam absorbs high frequencies that the Rockwool does not.Q. What's the right type of Rockwool?

SOS Reviews Editor Matt Houghton replies: The denser the material, the more effective it will be at absorbing low frequencies, but the flip side of this is that it also becomes better at reflecting higher frequencies back into the room. The 100kg/m3 product that you've mentioned should do a decent job, but it's denser than I'd choose for a broadband absorber. In fact, in my home studio, I use 100mm-thick 100kg/m3 Rocksilk for bass trapping, with a decent gap behind it. 

However, if you then place some acoustic foam over the top of it you'll have a much more effective acoustic absorber, as the dense Rockwool will absorb lower frequencies, while the foam will absorb some of the highs that would otherwise be reflected, making a very effective broadband absorber. If you don't want the foam, try looking for mineral wool in the region of 45‑75kg/m3. Remember to cover these slabs in some acoustically transparent material that will trap any stray fibres. If you're in a commercial studio, this will need to meet fire safety regulations, but for a home studio you could get away with a cotton sheet (I've used tablecloths!).  

Friday, July 12, 2019

Q. What are the best freeware plug-ins?

There are loads of freeware plug‑ins floating around out there now, so I find I'm getting swamped by choices. One site I checked out listed 670 of them! I'd rather not slow down my sessions looking for the perfect delay when just sticking with a good one and working with it would be much more productive. I've checked out a few of the ones mentioned in Mix Rescue and have been quite impressed, so I was wondering whether you could give me some further suggestions for a couple for each basic category of plug‑in. In particular, I'd be interested in any 'go to' freeware choices. I'm on a PC, so VST would be best.

Eoghan Brady via emailSome good freeware and donationware VST equalisers: Cockos ReaEQ, Bootsy Nasty CS, Antress Modern Black Dragon, and DDMF LP10.Some good freeware and donationware VST equalisers: Cockos ReaEQ, Bootsy Nasty CS, Antress Modern Black Dragon, and DDMF LP10.Q. What are the best freeware plug-ins?Q. What are the best freeware plug-ins?Q. What are the best freeware plug-ins?

SOS contributor Mike Senior replies: First of all, you could do worse than just download the ReaPlugs VST suite, which is a big chunk of the Reaper plug‑in complement and includes everything you're after, in one form or another. I've done whole mixes with just Reaper's plug‑ins, so I can vouch for their effectiveness. Other particularly worthwhile sets I've found are those from Antress Modern (http://antress.er‑webs.com), Bootsy (http://varietyofsound.wordpress.com), GVST (www.gvst.co.uk), MDA (http://mda.smartelectronix.com) and Voxengo (www.voxengo.com), which cover a lot of bases between them.

But on to some specific things I like, all of which have proved their worth in the heat of Mix Rescue! For general‑purpose EQ'ing, I do like Reaper's ReaEQ a lot, but for extra colour, try Bootsy's Nasty series and the Antress Modern emulations. DDMF (www.ddmf.eu) have a great donationware linear‑phase EQ called LP10, too. For synth‑style filtering, I usually just tend to automate ReaEQ, but Camel Audio's Camel Crusher (www.camelaudio.com) and Ohm Force's Frohmage (www.ohmforce.com) have more obvious attitude, if required. As far as dynamics are concerned, ReaComp and ReaXcomp in the ReaPlugs set are, again, good all‑round workhorses, but things like Georg Yohng's W1 (www.yohng.com), Buzzroom's BuzMaxi 3 (www.x-buz.com), Bootsy's Density, Jeroen Breebaart's PC2 (www.jeroenbreebaart.com) and the Antress Modern vintage emulations all get regular use on my projects. ReaGate and ReaFIR are a solid bet for most expansion and noise‑reduction tasks, so I've never really bothered looking elsewhere.

My freeware fallback for chorus, phaser, and flanger effects is Kjaerhus Audio's Classic series, and although I could no longer find a web presence for them at the time of writing, it's still possible to find the plug‑ins hosted on other sites via Google. MDA's Leslie and The Interruptor's Wow & Flutter (www.interruptor.ch) are cool for general modulation grunginess and I use those a lot. For tremolo/chopper effects, try Tweakbench's Cairo (www.tweakbench.com) or Oli Larkin's Autopan and LFO Chopper (www.olilarkin.co.uk). When it comes to distortion/saturation, there's lots of good stuff and I admit to being a bit of a collector in this respect. Some of my favourites are Bootsy's Ferric, GVST's GClip and GRecti, Jeroen Breebaart's Ferox, MDA's Combo and Bandisto, Mokafix Noamp (www.mokafix.com), Silverspike's Rubytube (www.silverspike.com), and Voxengo's Tubeamp: so much dirt, so little time! For more outrageous grainy and grungy effects, DBlue's Glitch (http://illformed.org) is a good bet, as are Jack Dark's outrageous Darkware series (www.gersic.com/plugins/hosted/darkware/darkware.html) and Tweakbench's Pudding and Sideslip.

The Interruptor's delay plug‑ins are good, as are GSi's WatKat (www.genuinesoundware.com), Tweakbench's Maelcum and GVST's GDuckDelay. That said, I tend to use ReaDelay for basic delay requirements most of the time. Smart Ambience is a great functional reverb demo, but Christian Knufinke's SIR (www.knufinke.de/sir/sir1.html) with impulses from Echo Chamber (www.memi.com/echochamber/responses/index.html) takes the cake for me in the freeware reverb department. For stereo image adjustment and M/S processing, my clear favourites are Voxengo's MSED and Flux's Stereo Tool (www.fluxhome.com). The latter has one of the best stereo vectorscope displays I've encountered anywhere. Speaking of displays, Roger Nichols' Inspector (www.rndigital.com) was my metering and spectrum-analysis plug‑in of choice for a long time, although Voxengo's SPAN is also good. I tend to use Schwa's payware Schope instead for most things these days, however. And speaking of Schwa (www.stillwellaudio.com), they have a great freeware bitscope plug‑in called Bitter that can be handy for digital troubleshooting. The TT Dynamic Range Meter is great if you're interested in the mastering 'loudness wars'; you can get it free on request via the Brainworx site (www.brainworx‑music.de).

Finally, here's a couple of odds and ends. Although I've yet to come across a decent, simple, freeware pitch‑shifter, if you're after freeware pitch correction, look no further than GVST's GSnap, which is pretty effective and has seen use in a number of Mix Rescues before now. If you're a fan of Aphex‑style psychoacoustic enhancement, also be sure to fire up Stillwell Audio's exciter, one of the plug‑ins available within the ReaPlugs ReaJS host, which does the same kind of thing.

Published November 2010

Wednesday, July 10, 2019

Q. Where's the best place to mount a large monitor screen?

I'm using a big desk with a shelf on the back as my studio workstation. Being partially sighted, I need my screen fairly close in order to see the details. I'm thinking about buying some studio monitors to put up on the back shelf of the desk, but will the fact that my screen is in front, albeit in the centre of the speakers, be a problem? Would the problem concern stereo imaging more so than frequency response? The screen is a 24‑inch model that is mounted on an arm for maximum flexibility.
It's better for acoustics if everything, where possible, is placed symetrically in a room. If you require a large screen in a studio, for any reason, it's a good idea to place it between and behind your monitors.It's better for acoustics if everything, where possible, is placed symetrically in a room. If you require a large screen in a studio, for any reason, it's a good idea to place it between and behind your monitors.

Via SOS web site

SOS contributor Martin Walker replies: To get the flattest frequency response from your loudspeakers, you need to install some acoustic treatment to damp down the room 'modes' that make each room resonate at certain frequencies, depending on its dimensions.

On the other hand, to get the best stereo imaging, the left and right halves of your studio should, if possible, be a mirror image of each other, and you should place the loudspeakers symmetrically with respect to the walls and fit acoustic absorption at the 'mirror points' on both side walls and the ceiling. 'Early reflections' bouncing off these points will obscure the details in your mixes and make it more difficult to pinpoint where each sound is panned.

Even your gear should, ideally, be installed in a symmetrical fashion. For instance, avoid placing cupboards, shelves, desks or keyboards on one side of the room only, since the sound bouncing off them will result in an unbalanced stereo image that will muddle your imagery.

Moreover (and here's where we get to your specific query), you can get troublesome reflections from audio bouncing off other objects between your ears and the loudspeakers, such as mixing desks and forward‑mounted monitor screens. Maintaining a clear area in front of your loudspeakers is the secret of good stereo imaging, although, thankfully, most modern flat-screen monitors will result in far smaller acoustic problems than the old (and relatively massive) CRT monitors.

A quick way to hear what difference any object is having on your stereo image is to temporarily drape a duvet, or similar, over it while listening to a mono signal being played through both loudspeakers (solo acoustic guitar might be a good one to try). If, with the duvet in place, the phantom central image between your loudspeakers becomes better focused and more concrete, as if a physical player is sat in front of you, then that object is interfering with your imaging.

There are several possible ways to avoid such audio compromises. The easiest is to place your monitor screen further away, either between the loudspeakers or behind them. This generally means you need a larger screen, and some studios hang huge monitor screens on the wall behind the speakers so their clients get a good overview of what's going on. However, even with 20/20 vision, this approach is often not good enough for detailed editing, so a second, smaller, screen is generally mounted much closer to the operator. Keep this as low as possible so that it's out of the speaker's line of fire.
Another more specialised, but elegant, alternative that I've spotted in various studios is a monitor screen recessed into a hole in the desktop. An easy version of this is to remove your monitor stand and lay your screen at an appropriate angle on your desktop, well below the critical area in front of the loudspeakers.

Yet another approach is the one you've already adopted. Since your screen is "mounted on an arm for maximum flexibility”, you can simply push it back out of the way of the loudspeakers for critical listening. This is a great idea for any musician; if only we had a similar option for mixing desks!

Published October 2010

Q. What is that Jimi Hendrix effect?

I'm trying to do something psychedelic with guitars — a bit like the song 'NY' by Doves — and I think the same effect was previously used on 'Voodoo Child (Slight Return)' by Jimi Hendrix. I have tried messing around with the Leslie and delay effects that you get with Logic 9, but have not even come close. What is that effect?

Via SOS web site

SOS Editor In Chief Paul White replies: The sound on that record was almost certainly produced by flanging the whole track. You can get close using a flanger plug‑in, though the original effect was created by running two tape recorders carrying copies of the same tape, then adjusting the speed of one of them so that one machine overtakes first of all, then falls behind the other. As the machines weren't perfectly in sync, the small delays caused phase cancellation of specific frequencies, and these varied as the relative timing between the two delays varied. That's what produces the familiar 'whooshing' sound.The recognisable sound of Hendrix's 'Voodoo Child (Slight Return)' was created by flanging the whole track. This was achieved with two tape machines carrying the same recording, with the speed of one or both being adjusted throughout.The recognisable sound of Hendrix's 'Voodoo Child (Slight Return)' was created by flanging the whole track. This was achieved with two tape machines carrying the same recording, with the speed of one or both being adjusted throughout.

The tape speed was adjusted either by using the varispeed control on one machine, or by slowing one, then the other machine slightly, by dragging the hand on the supply tape-spool flange. The most impressive effect occurred when one machine caught up with, then overtook the other. As you can imagine, the process was a bit hit‑and‑miss, as you had to line up both machines so that they'd start at the same time, but it certainly produced a trippy sound.

Flanger plug‑ins can process both mono and stereo mixes, but most tend to operate from an LFO and so can sound rather too regular. But if you automate the speed and depth controls to create a pseudo‑random effect, it can add an authentic feel. Most flanger plug‑ins are also limited in the minimum delay time they can apply, so can't quite recreate the 'through zero' effect of tape where one machine passes the other, though some of the more advanced plug‑ins use an additional delay in one side of the signal path to fake this effect. 

Published November 2010

Monday, July 8, 2019

Q. How do I record drums in a church?

I have the opportunity to record in a church that is of modest size with high ceilings and a lively acoustic. I'm thinking about using this room to track drums for my new album. Would you advise recording there, or would the acoustic just be too overbearing? I want a clean, natural‑sounding recording for a folk band, so I was thinking this room might offer the right sort of character.

Via SOS web site

SOS Technical Editor Hugh Robjohns replies: The limiting factor with most home project‑studio drum recordings is a combination of a generally dead‑sounding space, with little in the way of diffuse reflections, and a serious lack of height, which prevents the overhead mics being placed optimally. What you tend to get is a strong (but not diffuse) early reflection from the ceiling, which tends to degrade the quality of the sound captured — not only of the overheads, but sometimes the close mics too — and doesn't contribute in a nice way to the drum sound. The only solution is to make the recording space as dead as possible and then add artificial reverb to inject some life back into the drum sound, but the source often still sounds coloured and unrealistic.

A reverberant space often works well with drums because the diffuse early reflections add a welcome presence and scale to the drums without overpowering the direct sound. But typically we're talking about a modestly sized reverberant space here — a stairwell, for example — so that the early reflections are strong and the reverb time not excessive.
Recording in a typical parish church will provide the benefit of a very high ceiling, but most also have pretty lively acoustics with relatively long reverb times. This kind of acoustic character works well with some kinds of instruments and genres, but it's not a sound character I generally associate with folk music, which usually requires a more intimate treatment. All is not lost, though, and the benefits of the large space can still be used to your advantage if you are prepared to spend some time and effort controlling the acoustics and experimenting to optimise the arrangement.If you've got the opportunity to record drums in a church, it may be tempting to just go for it and set up right in the middle of the nave, to take advantage of the large-space reverberation. However, this kind of natural reverb can be hard to tame, and choosing one of the smaller spaces a church has to offer, such as a porch or vestry, may well produce a more usable sound. If you've got the opportunity to record drums in a church, it may be tempting to just go for it and set up right in the middle of the nave, to take advantage of the large-space reverberation. However, this kind of natural reverb can be hard to tame, and choosing one of the smaller spaces a church has to offer, such as a porch or vestry, may well produce a more usable sound.

The first thing to do is identify a suitable recording location within the church. Setting the kit up in the middle of the nave is probably not going to work: you'll need somewhere more enclosed to generate some strong early reflections. An entrance porch or a vestry might provide a more appropriate acoustic, or perhaps setting up in the aisle between the choir stalls.

Once you've found somewhere that produces the right kind of early reflections, you'll probably have to try to tame the long reverb tails, and that will come down to the use of carpet and drapes. Suspending large duvets from hired‑in lighting stands will make a worthwhile difference if you experiment with their placement around the kit, to help shield the overhead mics, in particular, from the worst of the long reverbs. Remember that a cardioid mic is most sensitive to sounds from in front and around the sides, so rig the duvets (or whatever) in front and to the sides of these mics to try to stop reflected sounds from being picked up.

The low-frequency end of the reverb will be the most difficult to control — drapes and duvets won't do much at all — so be prepared to high‑pass filter the mics (especially the overheads) to reduce any tendency to boominess and muddiness!

Close‑miking will obviously provide the best direct/reverberant sound balance, and you can mix in one or more 'space' mics set up further down the church to bring a more reverberant quality to the balance. Compressing the 'space' mics quite heavily and mixing them in at a low level often works well.
Judging the subtleties of a miking situation like this on headphones is very tricky, so my approach would be to do an experimental setup and recording: multitrack each mic and try out several alternative techniques. Document everything carefully with measurements, photos and whatever else you need to be able to recreate the setups later. Then take the recordings back to your studio, experiment with balancing the mics and listen critically to see what works and what doesn't.

You can then either go back and experiment more to further optimise the setup, or if you find one of the arrangements delivers the sound you want, simply go back and record the tracks using the best rig from your experimental recordings.

Published November 2010

Friday, July 5, 2019

Q. Can I use a Mac Mini for music?

I always hear people saying that the Mac Pro is the Mac of choice for musicians but, as a hobbyist, I simply can't justify the expense. I'm tempted by a Mac Mini, as I already have a decent screen, but am concerned that it won't be able to cope with the requirements of audio recording. What are the pros and cons?

Petra Smith via email

SOS contributor Mark Wherry replies: While it used to be the case that a high-end computer like the Mac Pro was essential for running music and audio applications, these days it's really hard to purchase a system that will be incapable of such tasks. It's all a matter of how many audio tracks, instruments and effects you need the computer to handle. Among the most important factors to consider in determining such handling are the type and speed of the processor, the amount of memory and the speed of the hard disk.Q. Can I use a Mac Mini for music?The updated Mac Mini comes with a 2.4GHz processor, 2GB RAM and a 320GB hard drive as standard, making it perfectly capable of running decent numbers of tracks.The updated Mac Mini comes with a 2.4GHz processor, 2GB RAM and a 320GB hard drive as standard, making it perfectly capable of running decent numbers of tracks.Photo: Apple
Since the first Power PC-based model was introduced (see the full review at/sos/may05/articles/applemacmini.htm), the Mac Mini has established itself as a basic-yet-capable studio computer. The current range features Intel Core 2 Duo processors, and the 2007 MacBook Pro (which, with a 2.4GHz processor, had similar performance capabilities) gives us a rough guide of the performance you can expect: using Logic Pro 7, this was capable of running 150 PlatinumVerb instances, 54 Space Designers and 512 EXS24 voices (with the filter enabled). Today's baseline Mac Mini also has a 2.4GHz processor, so those figures should be roughly comparable.

When it comes to memory, the 2GB supplied in the entry-level Mac Mini should be just enough to get you started. But you'll find life rather more comfortable with 4GB, especially if you want to work with sample‑based instruments. It's worth bearing in mind that 8GB is the maximum amount of memory supported by the Mac Mini.

In terms of storage, the basic Mac Mini comes with a 360GB drive. But, perhaps more crucially, this internal drive runs at 5400rpm — slower than those used in most other Macs — which will limit the number of audio tracks you can play back simultaneously. As a guide, you should expect to be able to handle approximately 50 to 60 mono 16-bit tracks at 44.1kHz. However, it is possible to connect a faster drive for audio, thanks to the Mac Mini's built-in FireWire 800 port — assuming you're not already planning to use this port for an audio interface, of course, since daisy‑chaining devices isn't always possible.

Another important factor when considering the Mac Mini, and one that might initially sound a little bizarre, is price. Although the Mac Mini is the cheapest Mac that Apple sell, its£649starting price can be deceptive in terms of value, even though, on paper, it's several hundred dollars cheaper than the cheapest iMac. If you already have a suitable monitor, keyboard and mouse, that's fine. But if you factor in the cost of these required devices to even the cheapest Mac Mini, the price difference between that and the low-end iMac starts to narrow considerably.

In a nutshell, the Mac Mini remains a basic, yet capable machine that provides a good starting point. However, in many ways, the entry-level iMac represents better value for those on a budget, especially if you see yourself quickly outgrowing the Mini's capabilities.

Published November 2010

Wednesday, July 3, 2019

Q. Can I make my computer perform better with Reason?

By Debbie Poyser
Reason v2.5.I have an oldish Mac with only 128MB of RAM, and would like to know how I can squeeze the most out of it with Propellerhead's Reason.

Joe McConnell

Columns Editor Debbie Poyserreplies: Reason runs very efficiently, even on what we might now call low-powered computers, but there are still things you can do to optimise its performance. Firstly, with not much memory, a common-sense measure is not to have other applications open at the same time unless you really have to. In addition, don't have more than one Song at a time loaded. Devices used in a Song can also affect the Song's RAM requirement: Songs using a lot of samples will need more RAM to load, while ones using, for example, mainly Subtractor modules as sound sources won't need as much. Another option, of course, is to buy more RAM! 

With lower-powered computers, think about the choice of devices in your rack, as some are definitely more CPU-intensive than others: using lots of Malström synths, for example, will have an effect on performance. Likewise, samples layered and tweaked to death in NNXT. The newer devices added to Reason in v2 and 2.5 tend to be the more power hungry. Watch out with the powerful new RV7000 reverb, and if your computer is struggling, confine yourself to the older effects devices. If you need to use the RV7000, make sure to use it in a send-return configuration so you can effect lots of different signals with it, rather than 'inserting' a reverb (actually, in Reason, using one in-line) for each device you want to process. In fact, generally try to use the send-return system where possible. If you're using samples, employing mono samples where you don't actually need stereo ones may help too. 

Don't use the High Quality Interpolation button you'll find on the sample-based devices, as this activates a more complex interpolation algorithm and needs some extra processing power (except, apparently, in the case of Apple G4s). Conversely, do try using the 'Lo BW' (low-bandwidth) switch, which reduces the HF content of sounds/samples and cuts overhead a little. You might find with some sounds (those at the lower end of the frequency spectrum or those which have been heavily low-pass filtered) that you don't hear the difference anyway. It probably goes without saying that you shouldn't use 24-bit samples if you don't have to; un-checking 'Use High Resolution Samples' in the general Preferences menu makes Reason convert any 24-bit samples to 16-bit. Other little tweaks you can make include deleting unused devices from your racks, and folding up ones that the Song is using but whose settings you don't need to change. Disabling cable animation, in the Edit menu, won't do any harm either. 

Be efficient with your sound programming and don't use resources you don't really need. For example, bear in mind that single-oscillator synth patches (in both the synths) use less power than dual-oscillator ones, and that filtering demands some juice too. Don't activate filters you're not going to use, and disable noise in Subtractor or the Shaper in Malström if you don't need it. 

If you've really gone to town with a track, using lots of sound sources and complex processing, and as a result your computer is struggling to play the Song back reliably, there may be something you can do to save power but still retain the work you've done. If the song is repetitive, consider using the 'Export Loop as Audio File' option in the File menu to save a few bars as an audio loop, then load it back in to an NN19 or NNXT, or a Dr Rex if you can first process the audio using Recycle or similar software. Then delete the original devices used to produce the loop (after saving the Song first under a different name in case you want to go back to it). You won't be able to keep tweaking the now-fixed loop but you'll save on the devices and effects used to produce it. Look in the 'Optimising Songs' section of the Reason user manual for more power-saving suggestions.

Published October 2003