Q. Do I need a new soundcard to work with my 64-bit PC?

By Martin Walker

The Delta 1010LT features 10 inputs and outputs at a reasonable price.

I need a new soundcard, as I'm upgrading to a 64-bit system, and I've been looking at M-Audio's Delta 1010 and Delta 1010LT. What's the difference between them and will they work with my new machine?

SOS Forum Post

PC Music Specialist Martin Walker replies: Well, both are good products, but although they provide a similar number of inputs and outputs, they are very different beasts in reality, and you can't expect quite the same audio quality from the budget LT version as from the original rackmount model. This is evident when comparing M-Audio's published specifications. The dynamic range of the standard Delta 1010's D-A converter is 117dBA, while the Delta 1010LT only manages 101.5dBA; the standard 1010 features balanced inputs and outputs, whereas the 1010LT doesn't; and the rackmount version allows 24-bit/96kHz full-duplex playback and recording.

However, the 1010LT is great value considering its range of inputs and outputs, especially as it has two mic preamps that the standard 1010 lacks. It costs a mere £199, compared to the Delta 1010's list price of £399, but both products offer fantastic value for money. Remember that if you go for the LT you have to contend with the many cables that ooze from the back of the PCI card. Unless you have a patchbay, or are going to leave all connections in situ, re-plugging your soundcard will involve fumbling around behind your PC's tower. The standard 1010 has a rather more convenient breakout box.

 
The Delta 1010 remains M-Audio's flagship PCI soundcard after six years of production.

For more information on the Delta 1010, check out my review in SOS January 2000 (www.soundonsound.com/sos/jan00/articles/midiman1010.htm).

As for the 64-bit compatibility issues, you're in luck: M-Audio were the first manufacturer to announce 64-bit drivers for their Delta and Firewire interface series. However, it might be worth considering products from other manufacturers, including Edirol, who also stepped in fairly quickly with a raft of drivers for lots of their UA, UM, and PCR products.

Others offering 64-bit support include Emu, for their DAS (Digital Audio System) interfaces and their XBoard USB/MIDI controller keyboard; Lynx, for their Lynx Two/L22/AES16 range; RME (only their Fireface 800 at present); and Terratec, for most of their EWS, EWX, DMX and Phase products. I'm sure there are others on the market with 64-bit compatibility, but you can keep an eye on manufacturers' web sites for details of new updates.


Published August 2006

KROSS 2 Video Manual Part 3: Combining Sounds and Using Favorites

Q. What are the characteristics of vintage mics?

By Hugh Robjohns

I've been browsing a vintage microphone site and it got me thinking: what kind of characteristics are actually offered by vintage mics? Can the same sound be achieved with modern mics and EQ? Isn't most of the 'vintage sound' due to tape and valves rather than mics?

 
The sought-after sound of the classic vintage mics is partly down to the fact that microphones used in professional studios many years ago would have been of particularly high quality to start with — and quality tends to age well.

Via SOS web site

SOS Technical Editor Hugh Robjohns replies: A good vintage capacitor mic sounds much the same as a good modern equivalent, and the same goes for ribbons and moving coils. Having said that, there has been a tendency over the last decade or two to make modern mics sound brighter, partly because the technology has improved to allow that, and partly because of aural fashion.

Also, professional mics that are now considered vintage were usually pretty expensive in their day — studios and broadcasters bought very high‑quality products — and that high‑end quality generally persists despite the age of the microphones.

Most of the vintage mics you'll find on those kinds of sites, though, are either valve capacitor mics or ribbons, and they both have inherent characteristics of their own that a lot of people revere. Ribbons have a delightfully smooth and natural top end, while high‑quality valve capacitor mics often have mid‑range clarity and low‑end warmth. These qualities can still be found in some modern equivalents if you choose carefully.
Some of the vintage character is certainly attributable to recording on tape, replaying from vinyl, and the use of valves and transformers. But some is also down to the construction of the microphone capsules and the materials used, not all of which are still available in commercial products today.

Published January 2011

Korg Pa-series Songbook and Songbook Editor Software

Q. Why does using a high-pass filter make things seem louder?

By Hugh Robjohns

Why does applying a high‑pass filter to a sound sometimes result in the output being noticeably higher than it was before? Today I have been working on a sound that peaks at 0dBFS. It has a lot of low‑frequency content. I am applying a high‑pass filter at around 100Hz and the output from the EQ is peaking at around +4dBFS. Why should this happen? Most of the power in this sound is in the low frequencies, and it has little going on above 2kHz, so surely with the high‑pass filter most of the energy from the sound has gone!

Via SOS web site

SOS Technical Editor Hugh Robjohns replies: This is a very common effect and there are several possible reasons for it. Fundamentally, the filtering process changes the shape of the waveform, so although there may be less total energy in the signal, the peak amplitude may well increase.

If you think about a bunch of different‑frequency tones all playing at the same time, their phase relationships vary continuously and add to or cancel each other to create the total waveform. Remove some of those tones and some of those cancellations won't occur. That can result in the waveform becoming bigger.

 
There are various reasons why using a high-pass filter on a signal can make it sound louder. Some equalisers, for example, actually boost the region just above the turnover frequency, which can produce an increase in peak level.

Most equalisers also introduce significant phase shifts and that, again, will change the way different frequencies combine and cancel. It can also happen because some equalisers actually boost the region just above the turnover point below which they are attenuating, potentially increasing peak level.

Published December 2010

Korg All Access Tuomas Holopainen on tour with the new Korg Kronos

Q. How can I make my masters louder?

By Mike Senior, Matt Houghton & Hugh Robjohns

I'm really new to recording, but I've been getting on well using the Tascam 2488 Neo 24-track digital recorder. However, when I create a master and then burn to CD, the overall volume is low. I record at about ‑10dBFS to avoid clipping and then use the compressor at mixdown to boost levels and even things out. This does raise the volume a tad, but nowhere near to that of commercial CDs. Am I correct in thinking that I'll have to use a lot of compression and limiting to get the levels to where I want them?

 
Low‑threshold, low‑ratio compression can be used to increase the subjective loudness of your mixes without excessively compromising dynamic range.

Via SOS web site

SOS Technical Editor Hugh Robjohns replies: Indeed, you will find it very hard to match the insane levels of some commercial CDs, but with a little compression and limiting you should be able to produce something that doesn't sound excessively quiet in comparison.

There are lots of ways of approaching this but, in general, when you're working on a track that is fundamentally well balanced but lacking in overall volume, I would start with some wide‑range and gentle compression. Typically, I would use a very low ratio — say 1.5:1 or even lower — and set it up with a very low threshold of around ‑40dBFS, so that it is slightly squashing everything in the mix, from the loudest to the softest instrument, all the time. This gives a very subtle and homogenous sound and is very different to the more typical use of compression with higher ratios and higher thresholds, which only affects the loud bits and for only some of the time.

Using this low‑threshold level, very gentle compression technique you can often squeeze as much as 8dB of gain reduction without the material sounding squashed at all. Adjust the attack and release times to suit the track — slower rather than faster is usually the right way to go for smooth level control — and then crank up the make‑up gain to raise the level close to 0dBFS.

The track will now sound significantly louder than it did, but there will still be spiky transients poking up above the main body of the waveform, and these are now restricting the total volume you can achieve. So the next process is to shave off those brief transients with a fast‑acting limiter and then wind up the make‑up gain again (unless your limiter does that automatically, as many do) to give another 2‑4dB of level increase.

By using this simple approach of a low‑threshold, low‑ratio compressor followed by a good limiter, you should find the material is substantially louder than the original mix track, but still without sounding overly compressed. However, this is obviously an artistic judgement call that only you can make: how much 'squash' will you accept for a louder‑sounding track? Sometimes you can only squeeze a few decibels before it starts to sound damaged, and sometimes you can manage 10dB without obvious problems.

Korg 40th Anniversary of Tuners

Q. Should I apply bus processing while I am mixing?

By Mike Senior

Amongst other things, the Drawmer DC2476 provides high-quality compression and EQ.

I've seen it suggested that compressing the stereo bus is the key to getting a mix to come together and sound 'like a record'. Is this really the case, and if so, at what point in the mixing process should I be adding bus processing?

SOS Forum Post

Reviews Editor Mike Senior replies: I find that bus processing of various kinds does wonders in pulling together a mix, and I do usually mix through a selection of bus processors for that reason. But let's not get carried away — a great mix will benefit as much (or even more) from bus processing as a mediocre one, so having access to decent bus processors doesn't really let you off doing a decent mix, because these days most serious engineers have access to decent bus processing!

If you start with all your 'polishing' bus processes in place at the start of the mix, you're likely to work less hard at getting the basic mix right in the first place. I think this was one of the lessons to be learnt from the On-line Mastering Shootout listening tests we did here in the SOS office for our March 2006 issue. Trying to fix mix problems with bus processing is fantastically difficult — improving any element of the mix usually involves compromising some other part. So, if you don't do the mix properly to start with, you'll find it very difficult to sort out any problems later using mastering-style processing. It's important to do the very best you can with your mix before you switch in any bus 'polishing', otherwise your final results will suffer.
My general advice would therefore be to avoid bus processing for as long as you can with a mix, so that you get it sounding as good as possible without any extra help. However, there are a couple of exceptions I would make to this.

The first is that, speaking personally, I usually patch in a full-band bus compressor over the mix while I'm creating my opening balances. Whether you would find this suitable as well will depend on whether you use a pumping compression sound as I tend to. If you plan to, then I'd suggest mixing with the compressor switched in — your balance decisions have to be different if you are intending to hit a bus compressor in this way, so you need to be able to hear what you're doing.

The second exception concerns using EQ rather than compression on the mix bus. It's a little trick I learned from our interview with Spike Stent, one of my mixing heroes, in SOS January 1999 (www.soundonsound.com/sos/jan99/articles/spike366.htm). He patches in a really high-quality EQ over the whole mix, boosting the 'air' frequencies so that he doesn't need to do this using lots of individual lesser-quality channel EQs. I use my Drawmer DC2476 mastering processor for this, so that I can stay in the digital domain. Spike was using a Massenburg EQ in preference to his 'low-quality' G Series SSL channel EQ, so the quality difference for him is probably less than for the rest of us — the difference between a really nice EQ processor like the Drawmer and the built-in digital channel EQs in a digital multitracker or software sequencer is much bigger.

What I would say though, even with regard to both these exceptions, is that you should always make sure to record completely unprocessed versions of mixes along with the processed versions. That way, if anything goes wrong with the bus processing, you don't need to completely redo the mix — you can just reprocess the unprocessed versions.


Published May 2006

KROSS 2 Video Manual Part 2: Selecting & Playing Sounds

Q. Is it best to synchronise all my digital gear using a word clock generator?

By Hugh Robjohns

Please could Hugh Robjohns write a comprehensive article explaining the operational advantages and disadvantages of using a word-clock signal to synchronise studio equipment as compared to alternative methods? Further, if the audio has been reference-clocked as it was recorded, does the replay chain (perhaps including multiple downstream signal processors) still require a synchronisation reference or is the clocking information embedded in the recorded data sufficient to hold all the downstream equipment in the correct relationship? Finally, with respect to jitter, will using an external master clock to synchronise the equipment chain prevent it?

SOS Forum Post

Technical Editor Hugh Robjohns replies: A new series of articles concerning various practical aspects of working with digital audio is planned for the near future, but in the meantime I'll have a bash at tackling your list of clocking questions.

A lot of equipment accepts only a simple word clock reference input, rather than AES or composite video references, purely because it is far easier and cheaper to implement. However, there is no significant technical advantage in using only the word clock format. Some might argue that the ability to daisy-chain a word clock signal around a number of devices using BNC T-pieces makes word clock superior since it provides a cheap and convenient way of distributing a reference clock. The problem is that while this approach can work in controlled situations, there are inherent dangers involved if the equipment isn't (or can't be) configured correctly, or the setup is changed without correctly re-engineering the chain. A proper star-shaped distribution of clock signals from a dedicated hub or master generator, using word clock, AES or a combination of both, is a far better and more reliable approach.

 
The Drawmer D-Clock provides a total of 20 word clock outputs, but is it the answer?

As to your second question, a digital recording has, by definition, to be clocked from a reference at the source. That reference is most often the internal crystal clock of an A-D converter. At each subsequent transfer of the digital audio from one machine to the next (assuming the use of AES, S/PDIF, SDIF3, MADI or ADAT interfaces) the clocking information is fully embedded and passed along with the audio. Assuming the equipment is configured to extract the embedded reference clock from its input signal, then it is not strictly necessary to provide separate reference clock signals from a master generator system. However, in larger setups there are significant practical and technical advantages in using a central master clock to provide stable references to the entire system.

Jitter is the enemy of all clocking systems because it introduces a degree of uncertainty in the timing of samples, which translates as a rise in the noise floor with various noise modulation effects, and often causes a blurring or instability in the stereo image at the A-D and D-A conversion stages. However, it should be noted that these jitter effects only become an issue at the points where audio is converted between the analogue and digital worlds — digital transfers between equipment are completely unaffected by even quite severe levels of jitter.

A good-quality master clock should have less intrinsic jitter than most individual devices, but that isn't a guarantee that you'll have a jitter-free system. There are three main causes of jitter: poor clock design, poor clock-recovery circuits (the part of the A-D/D-A converter which extracts the clock data from an incoming digital audio or reference signal), and the effects of the interconnecting cables. Of these, cable effects and poor clock-recovery circuits cause the most problems. The capacitance inherent in cables limits the slew rate of the data — how fast the signal can transition from one binary state to the other. At the output of a piece of digital equipment the data might switch from one state to the other in a beautifully crisp square wave, but by the time it reaches the input of another device the cable capacitance will have rounded it out into something looking more like a triangle wave. The clocking reference timing is generally taken from the points where the data transitions cross the nominal centre line of the waveform, and if these 'vertical transitions' have become sloping lines because of the cable capacitance, the precise point of transition becomes rather vague — that's jitter!

The greater the capacitance of the cable, the worse this problem becomes, so short, high-quality, low-capacitance cables will preserve clocking information far better than overly long, cheap, high-capacitance ones. Obviously, fibre optic cables don't suffer from electrical capacitance, but they have an optical equivalent, which is dispersion. If the optical quality of the plastic or glass is not optimised, the pulses of light can be degraded in such a way that the transitions between light and dark become (quite literally) blurred, and that causes exactly the same kind of jitter problems.

Fortunately, a good clock-recovery circuit can reject the effects of cable jitter, and some companies put a lot of effort into designing good jitter-rejecting clock recovery circuits. The problem is that most techniques which reject jitter to a high degree are very slow to respond and synchronise in the first place, so a practical compromise has to be reached, trading jitter rejection for responsiveness.

So, given that cables induce clock jitter, and that some jitter often seeps through the clock-recovery circuitry, it won't come as a surprise to learn that it is often better to use the A-D converter's own internal crystal clock as the reference, both for the conversion itself and the rest of the digital system, rather than use an external reference. This assumes that the converter has a good-quality low-jitter clock, of course. If it doesn't, you might get better results clocking from a better-quality external clock, although you are then at the mercy of the jitter-rejection capability of the device's clock-recovery circuit.

Sometimes there is no choice but to externally clock an A-D converter, as is the case when you need to synchronise several separate A-D converters for a multi-channel recording, for example. Using good-quality converters, linked with short clock cables to a common master reference clock would be the best and most practical solution in this case. The only alternative would be to run each converter on its internal clock, and then use sample-rate converters to resynchronise their outputs to a common reference — an expensive option, and one which might introduce a whole different set of unwanted artifacts!

Published May 2006

KROSS 2 Video Manual Part 1: Navigating The Interface

Q. Do I need two S/PDIF inputs?

By Mike Senior

 
The Roland M1000 digital line mixer will allow you to combine several S/PDIF signals.

If you have two bits of gear that you'd like to run into your computer interface via S/PDIF, do you need two S/PDIF inputs? I'm just curious as most soundcards/interfaces only have one S/PDIF input and output.

SOS Forum Post

Reviews Editor Mike Senior replies: The problem with inputting multiple bits of S/PDIF gear into a single unit is that the digital clocks of all the different pieces of gear need to be synchronised, otherwise you'll get nasty digital clicks. A soundcard with two S/PDIF inputs would either have to have word-clock inputs and outputs to synchonise it with the two input sources, or it would have to incorporate built-in sample-rate conversion. Good sample-rate conversion isn't cheap or easy to implement, so most budget soundcards (and hardware recorders too) won't be able to include it. That's why you will generally find only one-in/one-out S/PDIF interfacing where there's no word-clock I/O provided.

You can run more than one unit into a computer through a single S/PDIF input, but the signals need to be mixed together first, bearing in mind that there's the possibility of overload from the combined signal. There are a number of different devices available that can do this for you. Two that I can immediately think of are the Roland M1000 digital line mixer (SOS June 2003) and the Mutec Smart Merge clock router (January 2004). Although they have slightly different takes on the problem, both are useful problem-solvers, offering all the necessary sample-rate conversion, and should cost you around £500.

If you just want to switch between sources without having to un-plug and re-plug, the Smart Merge also includes this function, but is perhaps a little overspecified for just this task. You might want to look for a simple digital source switcher instead.


Published May 2006

KORG Pa1000: more power - new arranger keyboard from KORG

Q. Where can I find a software GM sound module?

By Sam Inglis

Bandstand is Native Instruments' new software GM synth.

It should be simple, but nowhere can I find a General MIDI (XG/GS) software sound module for VST. Can you help? A freeware one would be nice...

Dave Wallace

Features Editor Sam Inglis replies: I don't know of any freeware examples, but Edirol make a software sound module called Virtual Sound Canvas that provides a complete GM2 and Roland GS sound set and is inexpensive — Virtual Sound Canvas Multi-pack, which includes VST, Direct X and stand-alone versions, costs just £49 in the UK.

A more upmarket, but pricier alternative (although it lacks GS support) is Native Instruments' new Bandstand, which was reviewed in last month's SOS (www.soundonsound.com/ sos/may06/articles/nibandstand.htm). It comes with a 2.5GB sound library (which can't be said of your average GM module) and costs £150.

Published June 2006

KORG Pa1000: more power - new arranger keyboard from KORG

Q. What do the controls on my Preamp do?

By Hugh Robjohns

Photo: Mike Cameron

I've been looking up information on some external preamps, and I found that most of them have two controls — Gain and Level. Most console preamps have just a Gain control, so what is the Level control for?

SOS Forum post

Technical Editor Hugh Robjohns replies: Depending on how the preamp in question is designed, as a rule, the Gain control sets either the input level to the (fixed gain) preamp or the actual gain of the preamp, the latter being the more common arrangement. The Level control sets the amplitude or level of the unit's output. So Gain and Level are essentially the equivalent of the channel Gain control and the fader on a mixing console channel respectively.

Preamp Gain controls are often indented and change gain in steps of, say, 5dB, while the Level control is normally continuous. Having an output level control on an external preamp allows you to send a suitable level to a recorder. It also means you can tweak the overall gain structure of your signal chain, so that you could, for example, drive the preamp quite hard into saturation while ensuring the output level won't overload the recorder or DAW input.

Published June 2006

KRONOS LS artists first impressions

Q. Are all convolution reverb plug-ins created equal?

By Martin Walker

I'm looking to get a decent reverb for my Pro Tools rig, and there seems to be lots of choice, and quite a variety of prices — is it just the shipped library of impulse responses that makes the difference, or is there a difference in the software? So far I have looked at Waves' IR1, Audio Ease Altiverb, Wizoo's W2 and Trillium Lane Labs' TL Space.

SOS Forum post

SOS contributor Martin Walker replies: Many musicians must have at some time wondered whether there's any audible difference if you load exactly the same impulse response into several different convolution reverb engines. Well, different reverb plug-ins can sound slightly different, although you might only hear this difference on high-quality monitors in an acoustically treated room.

 
The price of convolution reverb plug-ins, such as Audio Ease's Altiverb and Wizoo's W2, has a lot to do with the library of IRs they come with.

Perhaps surprisingly, with some convolution reverb engines you may also hear an audible improvement if you convert the impulse response itself from 24-bit to 32-bit floating point format — all you need to do is load the 24-bit IR into your audio editor, re-save it in 32-bit float format, and then load it into your normal convolution reverb plug-in. This doesn't add any extra resolution to the file, but can ensure that any rounding errors or gain adjustments during the convolution process itself are minimised.

On my PC I can certainly hear an increase in focus and transparency during reverb tails after performing this tweak on the excellent Pure Space IR libraries from Numerical Sound (www.numericalsound.com) when they are replayed through both Voxengo's Pristine Space and Waves' IR1 plug-ins, although your mileage may vary with other plug-ins, as it depends on their internal resolution. Try it with your own plug-in — apart from a 50-percent increase in IR size your processor overhead will be identical, so you've got little to lose.
Nevertheless, these are tiny differences. The price of a convolution reverb plug-in has more to do with its versatility — how many clever controls there are to manipulate the impulse responses to make them as useful as an algorithmic reverb — and, more importantly, its bundled library of IRs.

Although creating IRs is certainly not rocket science, and can be done by anyone with a mic, balloon and pin, doing it to professional standards in world-class acoustic spaces is an expensive and time-consuming business, so those convolution reverbs like Altiverb, Waves IR1, and so on that come with a huge library tend to be several hundred pounds more expensive than those that don't. Third-party IR libraries tend to be expensive for the same reason.

Some people may point out that you can download loads of free IRs from various sites on the Net, but although I applaud all the effort that's obviously gone into many of them, and they are certainly a good way to add to your collection, they mostly tend to be 16-bit files, which will restrict the dynamic range of your reverb to 96dB even if your audio tracks have been recorded with 24-bit resolution. Many of the free IRs I've tried have also been truncated, while a few have exhibited hums, or excessive background noise.

Ultimately, as always, you tend to get what you pay for. All four plug-ins you mention have very good reputations, offer plenty of scope for user adjustments, and are bundled with comprehensive IR libraries. However, they all differ slightly in their feature set and scope, so if you can, the very best way to choose is to demo them at a suitable dealer so you can try out different interfaces and get a better feel for the content of the libraries. After all, while some plug-ins are subject to fads and fashions, you'll still be using a good reverb for many years to come.

Published June 2006

Dean Forrest and Adjua from “Back Beat Sound System” create with KORG

Q. What are the characteristics of vintage mics?

By Hugh Robjohns

I've been browsing a vintage microphone site and it got me thinking: what kind of characteristics are actually offered by vintage mics? Can the same sound be achieved with modern mics and EQ? Isn't most of the 'vintage sound' due to tape and valves rather than mics?
 
The sought-after sound of the classic vintage mics is partly down to the fact that microphones used in professional studios many years ago would have been of particularly high quality to start with — and quality tends to age well.

Via SOS web site

SOS Technical Editor Hugh Robjohns replies: A good vintage capacitor mic sounds much the same as a good modern equivalent, and the same goes for ribbons and moving coils. Having said that, there has been a tendency over the last decade or two to make modern mics sound brighter, partly because the technology has improved to allow that, and partly because of aural fashion.
Also, professional mics that are now considered vintage were usually pretty expensive in their day — studios and broadcasters bought very high‑quality products — and that high‑end quality generally persists despite the age of the microphones.

Most of the vintage mics you'll find on those kinds of sites, though, are either valve capacitor mics or ribbons, and they both have inherent characteristics of their own that a lot of people revere. Ribbons have a delightfully smooth and natural top end, while high‑quality valve capacitor mics often have mid‑range clarity and low‑end warmth. These qualities can still be found in some modern equivalents if you choose carefully.
Some of the vintage character is certainly attributable to recording on tape, replaying from vinyl, and the use of valves and transformers. But some is also down to the construction of the microphone capsules and the materials used, not all of which are still available in commercial products today.


Published January 2011

Havian30 Video Manual: Global Media Updates

Q. How can I learn to create drum parts?

By Mike Senior

I'm just starting out in learning to record audio but am beginning to expand on what I want to do. Though I'm now fairly competent at using my DAW of choice (Reaper), I'm finding it really difficult to create drum parts. What would be the most straightforward way for a complete beginner to get into and learn about this?
Sara Willis, via email

SOS contributor Mike Senior replies: In a word: loops. There are two basic things you have to contend with when putting together great drum parts. Firstly, you have to obtain good performances: whether you're wanting the sound of live drums or electronic drum‑machine timbres, the nuances of the performance or programming of the part play a vital role in creating a commercial sound in almost any style. Secondly, you need to be able to control the sonics well enough to build up a decent mix once all the other parts of your arrangement are in place. The reason I recommend loops as a starting point is that it simplifies the process of dealing with these issues. All you have to do is find a suitable loop and then learn how to adjust its performance or sonics where the unique circumstances of your music require it.

 
Just type 'sample' into the 'quick search' box at the top right‑hand side of the SOS home page to access an enormous archive of sample‑library reviews.

Finding a good library really shouldn't be hard. I've been reviewing loop collections for the magazine for ages now and I know that there are loads of really good ones available, catering for just about every musical genre imaginable. My first suggestion would be to go back through the magazine's sample‑library reviews: typing 'sample' into the 'quick search' field at the top right‑hand side of the SOS web site should pull them up out of the magazine's online archives for you. Anything with a four‑ or five‑star review is definitely worth investigating, but don't part with any cash before you've had a careful listen to the manufacturer's audio demos, and you should be as picky as possible in looking for exactly the right sonics for your needs. Don't just listen on your laptop's speaker or earbuds — drag the demo files over to your studio system, and if example loops are provided, try those out within a test project. This is what I regularly do as part of the review process, and it can be very revealing. Lining the demos up against some of your favourite commercial records may also help you narrow down the choices.

As far as the library format is concerned, I suggest you look for something based on REX2 loops, because these beat‑sliced files typically offer better tempo‑matching and rearrangement opportunities than the time‑stretching formats (such as Acidised WAV or Apple Loops). I don't think there's much sense in getting involved with any of the virtual instrument‑based libraries at this stage: while they can increase your flexibility in terms of sonics and programmability, they can also add a great deal of complexity to the production process, and I imagine you've got enough on your plate already with learning about all of this stuff! Often, loop‑library developers structure their libraries into 'suites', with several similar loops grouped together, and this can make it easier to build some musical variation into your song structure. There are also libraries that include supplementary 'one‑shot' samples of some of the drums used, and these can also be very handy for customising the basic loops, as well as for programming fills, drops and endings manually.

 
If you drag a REX2 file into Reaper's main arrange window, it'll automatically match itself to the project's tempo and present you with a series of beat slices. These slices make it easy to rearrange the performance, and also provide you with a lot of extra sonic options at mixdown.

Faced with a shortlist of good‑sounding REX2 libraries, the last consideration is whether the performances really sound musical. This is the most elusive character of a loop library and it's an area where the SOS review can provide some guidance. My usual barometer in this respect while reviewing is whether the loops make me want to stop auditioning and immediately rush off to make some music, so thinking in those terms may help clarify your thinking. It's also a good sign if the drum hits in the loop seem somehow to lead into each other, rather than just sounding like isolated events, because this can really make a difference to how a track drives along.

Once you've laid hands on some decent loops, you can just drag files directly onto a track in your Reaper project and they should, by default, match themselves to your song's tempo. Because each drum hit will have its own loop slice, it's quite easy to shuffle them around to fit existing parts. Just be aware that sounds with long sustain tails may carry over several adjacent slices. Map out a rough drum part by copying your chosen loops, making sure that Snap is 'on' so that the loops always lock to bar‑lines, but then be sure to also put in some work introducing fills and variations, so that the listener doesn't get bored. There are lots of ways of varying the loop patterns: edit or rearrange the slices; substitute a different loop from the same 'suite'; or layer additional one‑shots over the top. A lot of people think that using loops inevitably makes repetitive‑sounding music, but with most REX2 libraries there's no excuse whatsoever for letting this happen. (If you want to listen to an example of a drum part built with REX2 loops, check out my Mix Rescue remix from SOS October 2008 at /sos/oct08/articles/mixrescue_1008.htm, where I completely replaced the band's original drum parts in this way.)

The REX2 slices can also assist when it comes to adjusting sonics at the mix, because it's easy to slide, say, all the kick‑drum slices onto a separate track for processing. This is such a useful technique that I often end up doing it manually with loops at mixdown, even when they're not REX2 files! The Mix Rescue I did in SOS November 2010 (/sos/nov10/articles/mixrescue‑1110.htm) is a good example of this, and with that one you can even download the full Reaper remix project from the SOS web site if you want to look at how I implemented this in more detail.

Published January 2011

Havian30 Video Manual: Song Play

Q. Which sub-$230 audio interface should I buy?

By Matt Houghton

I've just bought an iMac and need a simple audio interface that I can connect via a Firewire cable. I want something simple and good quality to give me some decent guitar and vocal recordings, but only have around $230 to spend. Do you have any recommendations, and should I be looking on the second‑hand market?

Chris Lyons

SOS Reviews Editor Matt Houghton replies: Your question raises a few issues that need unpacking before making specific recommendations. Firstly, you don't need to think in terms of a Mac or PC interface, as most 'serious' budget interfaces will run on Mac or PC (Apogee and Metric Halo are the only companies I'm aware of that make their interfaces specifically for Mac but not for PC, and they come in above your budget). You've specified Firewire, but the fact that you're using an iMac means that you could consider either Firewire or USB interfaces. For the purposes you describe, either would be fine, unless you have other particularly bandwidth‑hungry devices running via the USB ports. The only other limiting factor, in terms of what will work with your computer, is going to be the version of Mac OS X you're using, and whether a given audio interface has drivers that support it.

 
Both the M‑Audio Fast Track USB and the Novation Nio come with bundled software, and at reasonable prices. The Fast Track USB comes with a version of Pro Tools M-Powered and would be a good choice for those on a very low budget for simple recording projects, while the Nio's I/O complement enables more flexible monitoring than some other budget interfaces.

You also ask whether you should consider purchasing a second‑hand interface, presumably with the intention of getting more bang for your buck. Personally, I'd happily go second‑hand, but the usual caveats apply: make sure it's a legitimate seller, check things out before you buy and so on. More importantly, some older interfaces won't be supported by OS 10.6 (Snow Leopard), and that's worth bearing in mind even if you're running 10.5.x, as you may need to update at some point in the future. In other words, you can find a bargain, but you need to find out what your money's paying for!

If you're recording guitar, it makes a difference whether you plan to record acoustic or electric guitar with a mic, or electric guitar or bass via DI. The former requires two mic inputs to record guitar and vocals simultaneously, whereas the latter requires only one mic input and a separate instrument input.

With this in mind, let's consider some specific interfaces. Some come with a bundled DAW of some sort, but as you'll already have GarageBand with your Mac, you may or may not think that important:

• Line 6 UX1: I've seen this online for under $150, and not only does it offer an ​XLR mic input, it also features a dedicated high‑impedance (Hi‑Z) guitar input and Line 6's rather good Pod Farm amp and effects modelling software. Their UX2 is similar, although it offers more inputs, so you could record in stereo if you wished to. The 'street' price of the latter is at the top of your budget if you buy it new, but obviously will come within budget if you go second‑hand.
• M‑Audio Fast Track USB: Although Pro Tools 9 now works with any audio interface, most M‑Audio interfaces still come bundled with a 'lite' version of Pro Tools 8, which makes them a good way to get into Pro Tools on a budget if you're keen to do so. At the time of going to press, you could get this interface bundled with Pro Tools M‑Powered Essential v8) for around $100 (street price).
• PreSonus Audiobox USB: This interface offers two mic/line/instrument inputs, and can be found for under $150. It comes bundled with PreSonus' Studio One DAW software, which works on both Mac and PC.
• ESI U46XL: The U46XL generally sells for under $200 and includes two stereo line inputs, as well as the mic and high‑impedance inputs you require. You also get a bundled copy of Steinberg's Cubase LE DAW.
• Novation Nio 2/4 USB: This includes effects software, much of which is aimed at guitarists. It's available for around $199. As well as the two mic/instrument inputs, there are two stereo outputs, presented on both RCA phono and headphone jacks, which makes this interface rather more flexible than the others in the list when it comes to monitoring.

That covers the basics, and all the above offer better audio quality than is built into your iMac. But before you jump in head‑first, I'd recommend that you read the in‑depth article on the subject that appeared in SOS September 2008 (/sos/sep08/articles/audiointerfaces.htm).

Published January 2011

Havian30 Video Manual: Sounds

Q. What's the best order for recording a band?

By Paul White

I'm going to be recording my band's new album and it's my first big project, so I'd like a bit of advice. It's a rhythm & blues band consisting of drums, rhythm guitar, lead guitar, acoustic guitar, bass guitar and lead and backing vocals. I have a limited number of mics, so I'll be recording sources separately. The question is: in what order do I do this? The band are pretty tight, but I don't think the drummer would like playing to a click track with no reference to the song itself. I seem to get mixed views on this: I know the normal way is drums first, but what would be best for this band?

Via SOS web site

SOS Editor In Chief Paul White replies: It always gives the best feel if the band play together, even if it's only drums plus a rough mix as a guide of the other instruments and a vocal, recorded down one mic, as these can be replaced later. Of course, you also need to minimise spill, so using POD‑style devices and headphones for guitars and bass can help in getting the guide parts down and avoids the need for a mic, though you may need a small mixer. Otherwise, put the drums in a different room from the guitar amps. Some people manage by recording just the drums and bass together, but trying to do the drums on their own is asking for problems, as the feel will never be quite right.

If the music is of a type where the drummer is happy to play to a click track, you could always record the separate parts to a click or guide drum loop, then have the drummer put the drums on last. But, again, you could lose all your feel that way.

SOS Reviews Editor Matt Houghton adds: With a genre like rhythm & blues, the feel of the playing is so critical to getting a good result. There are plenty of great recordings where most parts were played with everyone in the room to get a good vibe. As Paul suggests, you need the rhythm section, in particular, to be really hitting the groove, and with that in mind, I would certainly want to track at least the drums and bass together as a starting point. You could have both playing in the same room, using a solid gobo or two to provide separation between the kit and the bass amp, with the amp separated from the kit, but with the bassist standing in a position where he and the drummer have good visual communication. Alternatively, put the bassist in the room, but DI his bass, perhaps running it through an amp simulator for monitoring purposes. If you get an amazing take, it would be simple enough to use a good amp simulator such as IK's Ampeg SVX or Softube's Vintage Bass Room, or re‑amping, to make it fit in the mix, but you also have the opportunity of laying a new bass part over the drums.

While you could take the 'playing together' principle further, and put guitars, keys, vocalists and whatever else you wish in the same room, you'll often find that achieving acceptable separation becomes problematic, and maybe even a problem that outweighs the benefits of having everyone perform together. In my experience, you're better off having a vocalist and guitarist either in a different room than the drums, or in the control room, with the guitar DI'd and/or run through an amp simulator, so that the whole group is doing a live take into the monitor mix. You can always track those parts, in case you get a moment of magic — with the guitars, as with the bass, amp simulations or re‑amping are valid approaches here — but you'll still have the option to overdub those parts later, and many musicians will be glad of the chance to try a few different takes.

What else you record in what order will depend on the other musicians. Does the vocalist want to hear the other parts? Do other musicians take their cue off the vocals? If you make sure you record the guide parts as you go, those questions become less of an issue, and while I often advocate starting with vocals (or other primary elements) when mixing, I don't usually find that it makes such a difference when recording.

 
Recording the various instruments in a band in the right order is important, as it will affect the feel of the track. Photo: Richard Ecclestone

When it comes to tempo and click tracks, your approach will depend very much on the band in question. When you say 'tight', that might mean that the drummer can keep a rock‑solid tempo, or it might just mean that all the musicians can keep in good time with each other. In my experience, some bands that sound tight can actually accelerate and slow down considerably during a track, which may or may not be a good thing. The one thing that I would say, though, is that the thresholdsfor what seems acceptable when playing live is different than when listening to a record that gets played again and again. So having a click track or guide track for the drummer might be useful. Really, the best advice I can give is to discuss this with the drummer, go with what they feel comfortable with and simply be alert to any problems.

If you do choose to use a click track, my advice would be to feed it to the drummer alone, and then have everyone else lock in with him or her, which is what a good live band will usually do, after all. And do make sure that the headphones don't leak that sound into the overhead mics, which is something I hear a lot on material sent in to SOS!

Of course, it's perfectly possible to overdub drum parts, but I invariably find that when doing this you lose almost all of the magical glue that holds a track together. In this style, more than most, that will probably prove unacceptable. In fact, I can only recall one occasion where I've done it and obtained a satisfactory result.


Published December 2010