Thursday, February 23, 2017
Wednesday, February 22, 2017
By Hugh Robjohns
I recently purchased a Mindprint DTC and I've noticed that it produces quite a lot of hiss. It is not so obvious to the ear at first but after individually compressing instruments and playing them together, it becomes quite nasty. Also, if you are monitoring the signal from the DTC, you notice quite a large amount of noise on the analyser. I purchased the 24-bit S/PDIF module to find the same happens there. Whilst I was sold this as a mastering tool I am wondering if this is a design flaw. I have read some good reviews (including yours from SOS June 2002), but on a few web sites, similar problems are mentioned. I have done all the regular troubleshooting, such as changing cables and checking the power distribution of my system, and it all looks fine. I am wondering why my DTC is so noisy and if it is supposed to be.
Technical Editor Hugh Robjohns replies: This is tricky to answer without actually hearing the problem you are complaining about and knowing how you are using the product. However, the DTC remains a favourite processor of mine, and I've not had any serious noise problems when I've used it, so I would suspect either an operational problem or a faulty unit.
As it uses valves, it is possible that you have a faulty one, which could lead to excessive noise. Changing the valves is not difficult and replacements aren't expensive — but it might be worth getting the product properly checked-over by a qualified technician, in case anything more serious is wrong. Unlikely, but it's always best to get it checked.
Perhaps the more likely problem is an operational one. Setting an appropriate gain structure is important to optimise the signal levels through the unit. The other thing that intrigues me is that you say: "It is not so obvious to the ear at first but after individually compressing instruments and playing them together, it becomes quite nasty". Noise will always add and build in level, so processing individual instruments with the DTC will always produce a noisier result than simply processing the final mix. But I wonder if, in fact, you are over-compressing the instruments.
Ideally, you could send a short extract of some affected material, but it may be easier and quicker to get the unit checked-out or compared with another unit to make sure all is working as it should.
I've always found Mindprint to be helpful in resolving issues like this, so giving them a call might be a worthwhile step to take as well. Mindprint +49 6851 9050.
Tuesday, February 21, 2017
Monday, February 20, 2017
Q. When I mix in Cubase, my mixed-down songs do not sound as good in playback as they did before I saved them. Why?
By Hugh Robjohns
When I mix in Cubase, my mixed-down songs do not sound as good in playback as they did before I saved them. I listen before and after saving; in the same room with the same monitors; with the same quality conversion, and without any digital clipping on any busses. I usually mix 32 channels down to eight stems, and then into a stereo output buss.
Does analogue summing provide a more accurate representation of the multi-channel mix, an enhancement to the sound of the mix, or both? Can I get to the first option by simply capturing the pre-save monitor output with an accurate stereo A-D converter? And can most of the benefit of analogue summing be derived from creating stems in the DAW before saving, then routing just the stems out for analogue summing? Alternatively, can those benefits be gained by making all adjustments to channel levels in the software, and then running the software-adjusted signals through a summing device that makes no further gain adjustment? If so, are the gain controls on most analogue summing devices necessary?
Technical Editor Hugh Robjohns replies: Wow, that's a lot of questions to get through! Bear in mind that I'm not a Cubase expert, so I can only talk to you from the analogue summing perspective.
The first query, about an audible difference between a 'live' session and a replay from the saved file, is an interesting one. The audio components of a session in progress will typically be working within a 32-bit floating-point domain, whereas the output D-A will only accept a 16- or 24-bit fixed-point signal. So a conversion process is required which will involve some form of dithering. It is well known that some dithering algorithms have a more audible effect on some material than others — particularly where some kind of noise-shaping is involved. So it is possible that the differences you hear are the result of the conversion from the floating-point format to a truncated and dithered fixed-point one.
However, it is probably more likely that the differences you hear are due to the way you are monitoring the 24-bit mixed file. If you are using a different program to replay these files, that alone may explain the change — perhaps there is some kind of internal processing going on that has an audible effect. If you are monitoring through Cubase, it might be because the file is being transcoded back into a 32-bit floating point format within Cubase, and then converted yet again to 24-bit for the monitoring output. That kind of multiple conversion process is likely to result in a sonic degradation, no matter how subtle.
Moving on to analogue summing, I would say it generally sounds 'different' rather than 'better.' Analogue summing is inherently less technically accurate than digital summing (when performed correctly), but often those technical imperfections are musically and sonically attractive — in much the same way that the high levels of harmonic distortion created by some valve preamps are pleasing to the ear. So the answer to your question is: yes, in some circumstances analogue summing enhances the sound of a mix.
One thing that analogue summing provides for free is mix headroom. A huge number of people 'mixing in the box' do so with very high peak levels on each source channel; typically well up above -6dBFS. This results in the mix-buss accumulator having to accommodate signals with sample values above 0dBFS and the master output fader has to be pulled down to compensate. Although this absence of mix headroom shouldn't be a problem in theory, it appears that some systems (and plug-ins) do have a problem with it, and a great many users have found that leaving more headroom on the source channels avoids the problem completely, producing far sweeter-sounding mixes as a result. The old analogue concept of having 20dB or so of headroom was arrived at for a very good reason and has stood the test of time. The concept translates perfectly to the digital world, with the same practical and sonic advantages.
You asked if you could capture the pre-save monitor output with an accurate stereo ADC. Well, yes, you could, but it won't improve anything on a technical level. The output from Cubase is transcoded internally from 32-bit floating point to 24-bit fixed-point samples. These are then passed to the digital output and in turn to the D-A converter that feeds your monitoring. Routing that analogue signal back through yet another A-D converter to get it back into the digital domain gains nothing at a technical level and is likely to degrade rather than enhance the signal. Of course, it is possible that you may like the sonic character imposed by that degradation, in the same way that people like valve preamps.
Next you asked if most of the benefit of analogue summing can be derived from first creating stems in the DAW before saving. The answer is a qualified yes. The advantages of mix headroom and the sonic colouration will be the same whether you mix individual channels or stems in an outboard analogue summing box. However, if the mixing of stems created within Cubase produced samples greater than 0dBFS, the quality loss resulting from a lack of mix headroom in the digital domain would remain.
Finally, you asked if most of the benefit of analogue summing could be derived by making all adjustments to channel levels in the software, and then running the software-adjusted signals through a summing device that makes no further gain adjustment? The answer here would be yes again, because by adjusting the level of source channels before mixing, the advantages of mix-headroom are preserved, and the mix may well sound better as a result.
The gain controls on those analogue summing devices that include them are there to allow the user to optimise the gain structure for whatever source equipment is being used. Some D-A converters operate with peak levels of +26dBu, while some only reach +12dBu, so an input gain trim to optimise the gain structure is both necessary and important if you can't determine the source input levels.
Saturday, February 18, 2017
Friday, February 17, 2017
By Hugh Robjohns
Years ago I touched a guitar that was plugged into an amp against a radiator, and it blew the amp and melted the guitar string. Afterwards someone told me that this was caused by a ground loop, but I've never actually understood what that means.
SOS Forum Post
Technical Editor Hugh Robjohns replies: That wasn't a ground loop — that was a faulty amp with a missing safety earth. There was no loop, because there was no earth at all until touching the radiator provided the missing link. Had you sat on the radiator and then picked up the guitar, you might well have been playing Emaj7 on a harp on a fluffy white cloud by now!
This is a distressingly common and life-threatening situation often caused by guitarists (or their so called 'technicians') in a futile effort to stop audible hums.
A ground loop is different — it occurs when there is more than one ground path between two items of equipment. Usually, one path is the screen of an audio cable connecting the two pieces of equipment and the other path is via their chassis safety earths in the mains plugs. Inside the equipment, the audio screen earth is often linked directly to the chassis earth, hence the possibility of a loop. If the two bits of equipment are plugged into the same mains socket, their chassis safety earths are effectively tied together at the same potential, and so there is unlikely to be any circulating ground current, despite the apparent ground loop. However, if one item is plugged into a different mains socket, its chassis safety earth might be grounded some way away from the other equipment's earth, and there can be a small difference in potential voltage between them. Silly as it sounds, earth is not the same everywhere. The potential voltage difference between their two chassis earths can cause a small current to flow, and since the earth provides a reference for the audio electronics, that flowing current causes the earth reference voltage point to vary slightly. This can be heard, usually as a low-level hum or buzz.
Ideally, the solution is to make sure that everything is earthed at one central point, so that everything shares the same common earth reference point. The easiest way to do that is to plug everything into a star arrangement of plug-boards fed from a single socket (assuming suitable power capacity). If that can't be done, the safest solution is to break the loop by isolating the audio cable screens at one end. The cable is still screened, but there is no longer any possibility of a loop, so the hum currents can't flow around it. Inserting transformers in the signal path can also break the loop, and this solution is common in outside broadcast and live sound rigs. DI boxes feature transformers for this purpose too.
Problems arise when uninformed people decide to break the loop by removing the safety earth in the mains plug instead. This does break the loop, obviously, so any related hum will disappear. But it also means that the equipment is no longer earthed, and thus any fault that occurs in the equipment is now life-threatening! Sometimes you don't even need a fault to cause dangerous problems, though. Most equipment has filtering on the mains input to stop mains-borne noise getting in (or out). If you disconnect the mains earth in the plug, the nature of that filtering is such that the (previously earthed) chassis — and everything connected to it — 'floats' up to half mains voltage (making it about 115V in the UK). This means everything that should be safely earthed — all the exposed metalwork, including guitar strings — now carries a life-threatening voltage.
Going back to your guitar incident, the strings on the guitar are supposed to be earthed through the guitar lead to the socket on the amp. That, in turn, is usually connected to the amp's chassis earth, and thus through the mains plug to the mains safety earth. Metal radiators are connected to the mains safety earth point too, as is all house plumbing.
So if you have a guitar amp with the safety earth disconnected in the plug, the chassis is likely to rise to 115V, and everything that should be at 0V (earth) is now cooking with gas! Rest the guitar strings on anything that really is earthed (the radiator in your example, but a mic stand holding a mic that is earthed through its cable to a mixer is another very common alternative), and you have mains power now flowing directly through the equipment to find a real ground. This is almost certainly going to severely damage or destroy the amp, the guitar and — if you get yourself between the two — you as well! The classic way for budding pop stars to die is because of an amp with the mains safety earth disconnected. They rest one hand on their guitar strings and reach up with the other to hold a properly earthed mic. That leaves them with 115V effectively straight across their heart. Sweaty hands make very good conductors and it only takes a few milliamps of current flow to stop the heart. Musicians are electrocuted in this way every year, and while I'm all in favour of Darwinian evolution, sometimes the good guys get caught out too.
The lesson is that you should never remove the earth from a mains plug. If you have hum problems, break the screens on the audio cables or use isolating transformers in DI boxes.
Published August 2006
Wednesday, February 15, 2017
Tuesday, February 14, 2017
By Sam Inglis
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...
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.
Monday, February 13, 2017
Saturday, February 11, 2017
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.
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.
Friday, February 10, 2017
Thursday, February 9, 2017
I have a five-piece drum kit with the three standard cymbals. I am trying to get the best sound possible on a very small budget. I am willing to learn new techniques (and I have to anyway) but I do want to keep prices low. I'm planning to order a Rode NT1A which I hope to use as a single mono overhead, but I was also thinking about a pair of AKG C1000s instead. I am considering a Samson Q Kick mic for the bass drum and a Shure SM57 for the snare. My budget may not stretch to the SM57 for now — can I manage without it? I'm aiming for an acoustic rock sound.
SOS Forum post
Reviews Editor Mike Senior replies: The mics you mention are all solid choices overall, but I'd suggest you go for stereo overheads as opposed to mono. In fact, I'd rather go for stereo overheads to start with, rather than a mono overhead and kick drum mic.
You can manage very well with just a pair of overheads, although you're more at the mercy of your drummer's skills than you are with a full multi-mic setup. After that, I'd probably suggest that a kick-drum mic would be the next best investment, followed by a snare-drum mic. Your choices for both of these should be fine, and you should be able to find lots of other uses for both mics in the studio other than just drum miking.
However, as far as the overheads go, the AKG C1000s wouldn't be my first choice, to be honest. They're designed primarily for stage use — unlike phantom-powered condenser mics, they use a back-electret condenser design which means they can run off battery power — and I've found they can sound a bit nasal. Looking at the options within your budget, a better pair of mics to go for would probably be the Samson CO2s or Red5 Audio RV4s, either of which will set you back only £99 per pair in the UK. A slightly more expensive option would be a set of SE Electronics SE1As, which are £199 per pair. Any of these would be a good starting point, and, again, you should find lots of uses for them as your studio grows. The only potential fly in the ointment is that none of these mics can be battery-powered, so you'll need at least two input channels with phantom power to get them to work. You don't say what you're recording to, but two channels of phantom power isn't asking very much!
In terms of techniques, Hugh Robjohns feature on recording drums from SOS February 2003 is an excellent starting point — www.soundonsound.com/ sos/feb03/articles/drummiking.asp. In it he discusses not only mic selection and positioning, but also room acoustics, mixing and effects, and the vitally important and often overlooked matter of making sure the kit is properly set up and tuned in the first place! Searching the SOS web site for 'drum recording' should turn up lots more helpful information. In particular, we frequently tackle drum-related issues in the course of our Studio SOS visits — in March 2003 (www.soundonsound.com/ sos/mar03/articles/studiosos0303.asp), March 2005 (www.soundonsound.com/ sos/mar05/articles/studiosos.htm) and as recently as April 2006 (www.soundonsound.com/ sos/apr06/articles/studiosos_0406.htm). It's not really surprising that the topic comes up so often — recording a full kit in a home studio environment is a bit of a challenge, but with a bit of know-how and some experimentation, you should be able to get good results.
Tuesday, February 7, 2017
Monday, February 6, 2017
By Dave Lockwood
Electrical noise — hiss, buzz and hum — is something that plagues every electric guitarist to some degree, but noise comes in a variety of forms and it is important to establish exactly which kind(s) you are experiencing in order to devise an appropriate solution. Most noise in an electric guitar rig emanates from one or more of five different sources: amplifier self-generated hum and/or hiss; hum or buzz picked up by the guitar itself; self-generated noise from any pedals/processors in the circuit; gain structure-related noise, such as cascaded distortion stages; and ground-loop-related hum. If you think you are suffering from noise that isn't generated in one of these ways, I'd like to hear about it!
The most efficient way to track down noise in a guitar system is to think of the amplifier or studio monitor system as the end of your signal chain and work systematically back from there. If you don't do this, you have no idea whether the noise you are hearing from the amplifier is being generated within the amp itself, or being picked up by the guitar and fed to the amp. You could end up taking steps to solve a problem you don't have, as well as completely failing to solve the one you actually do have. If your amp or monitoring system hums or buzzes excessively with no input connected to it, then you've got an equipment malfunction. That is beyond the scope of this article, so for these purposes I'll assume that that part of your rig is clean. From here on, I'm also going to treat amps and recording processors (Line 6 Pods and the like) as the same, because it is the 'upstream' noise of the guitar itself and related systems that we are interested in.
When tracking down noise it always pays to initially reduce your system to the minimum number of components, so begin by connecting your guitar directly to a single amp or recording processor via a screened cable, set the volume of the amp or monitor system to a normal operating level, turn the guitar's volume control all the way down and just listen. If there is any more noise than there was before the guitar was connected then the cable is at fault. With nothing connected, the amp's input jack will be automatically short-circuited to ground; with the guitar connected, but turned down, the input is again shorted, but at the other end of the cable, so the cable is the only variable.
Assuming all is well with the cable, now turn up the guitar's volume to maximum, hold the strings in a normal playing fashion and listen again. If you hear no more noise than before, congratulations; you must have a fantastically well-screened guitar and the perfect guitar-recording environment. The rest of us will be hearing at least a bit of buzzing and maybe a bit of 50/60Hz hum as well. Move the guitar around over an area of a few feet either way to see if the hum goes away. The level of hum is usually directly related to the guitar's proximity to any large mains transformers in the room. If you are using conventional (non-hum-cancelling) single-coil pickups and you are within the radiated field of a mains transformer, you will get hum. Exactly how much depends on the gain in your system and your proximity to the source. If you can't work out the origin of the hum field, try switching off everything except your amp (or monitor system, if you are DI'd) and then switch things back on one at a time to see when the hum reappears. When it does, see if you can re-site the offending item further away. The only solution is physical separation, as the amount of additional screening required to keep induced hum out of the pickups would actually prevent the guitar working at all. Of course, if you are using humbucking pickups, you are in the clear on this one, but the chances are you'll still have some 'buzz'.
Buzz has a lot more high-frequency content than hum. If you are unsure which you have, try turning your guitar's tone control all the way down; if the noise mostly goes away, you are dealing with buzz rather than hum. Buzz will also often be greatly reduced when you touch the strings or any other metal part of the guitar, sometimes accompanied by an audible click, whereas hum will remain unchanged. The common explanation for why noise goes away when you touch the strings or metalwork is that you are adding to the overall amount of screening. I'm not so sure about that, because certain types of noise actually get louder when you hold a guitar close to your body without touching the strings. This suggests to me that the player's body is, effectively, conducting the interference into close proximity with the guitar. The noise goes away when you touch the strings because that interference is safely conducted away to ground.
Unlike hum, which is generally induced directly into the pickup coils themselves, buzz gets in everywhere, so any bit of unshielded wiring can be the source. Even guitars with humbucking pickups will often still buzz. This is, understandably, very frustrating if you've just shelled out for a set 'noiseless' pickups for your Strat and find out the instrument is just as noisy as before; it no longer hums, but the amount of buzz is unchanged because the noise is getting in via the control cavity and the unscreened wiring rather than the pickups. The only answer is to screen every part of the internal electronics with copper foil or conductive paint, which is then connected to the earth side of the circuit. Do not attempt to screen the pickups themselves, or even the pickup covers, however, as this will alter the sound.
Screening will make a major improvement, but if you are using single-coils with a high-gain setup or lots of compression, you will still have some noise pickup. Buzz is often sensitive to the angle at which you hold the guitar, however, so you can always try to find the 'null point' at which the noise is least intrusive and simply do your best to keep the guitar at that angle whilst recording. It sounds crude, I know, but pro Strat and Tele players have worked that way in the studio for years because, until recently, there were no hum-cancelling single-coil pickups that sounded enough like the real thing to make the trade-off worthwhile.
The most common sources of buzz are TVs and CRT computer monitors, computers themselves and lighting dimmer switches. Just occasionally you'll also find a poorly designed external power supply for some piece of equipment in your rig that puts buzzy noise back onto the mains and thereby affects everything in the room. So, switch the TV off, use a flat-screen (non-CRT) monitor if possible, site your computer over four feet away, dump any noisy PSUs and use only conventional incandescent lighting.
If you've done all of those things and you've still got a nasty, edgy-sounding buzz, then the chances are that there is a lighting dimmer involved somewhere. The trouble is, it doesn't have to be your dimmer — lighting dimmers can affect you from an adjacent room, or a room above or below you. And it doesn't even have to be the dimmer itself — the cable running between the dimmed lamp and the dimmer control can also emit interference and this is often routed across the middle of the room within the ceiling void. Dimmers make most noise when they are actually dimming, so the noise will improve slightly when you turn the dimmer all the way up, but only switching it off altogether will make the interference go away. Dimmers that work on an entirely different principle and do not create electrical interference are just starting to appear on the market — watch this space for news.
Over the next few issues we'll tackle sources of noise beyond the guitar itself: pedals, cascading gain stages and earth loops. Dave Lockwood
If you hear electrical noise (including crackling) when you move the tremolo, that's probably because the tremolo springs or some other non-moving part of the tremolo is grounded, but the strings are connected to it only via the tremolo pivot points, and if these don't have a very low electrical resistance at all times, you'll get noise that will vary as the resistance varies. Spraying a contact enhancer such as DeOxit on the pivot points can help, but in some tremolo designs it may be best to use a thin, very flexible wire to ground the moving part of the tremolo providing you can find or create a suitable attachment point, such as a tag washer fixed under one of the existing screws.
When used in combination with either of the other pickups, the RW/RP pickup creates a parallel-connected humbucker. Noise, which is induced into the coils only, is cancelled out as the two coils are, effectively, out of phase, whilst the strings are sensed by the magnets, initially out of phase (due to the reverse polarity of one of the pickups) and then restored to in-phase by the reverse winding. The net result is a clean signal with no noise. It works a treat, but only in switch positions two and four, which, ironically, are the ones that you would rarely choose for high-gain work.
Effects PedalsDesigned by East Sound Research of Denmark, Carl Martin's Vintage range of pedals is designed to offer both tonal quality and mechanical longlevity, hence the chunky cast metalwork, chicken head knobs and mechanical bypass switches. All three models reviewed can be powered by means of a standard 9V battery or an external power supply, and the battery compartment is easily accessible without tools by means of a lift-out flap on the underside of the case. Rubber feet keep the pedals from creeping and they're heavy enough to stay put without being too heavy to carry around in your gig bag. The input and output jacks are on the edge of the case away from the player.
Finished in surf green, the Surf Trem (£54.99) is the simplest of the pedals and has just two controls other than the mechanical bypass switch and red indicator LED shared by the other models in the series. This is a straightforward tremolo effect, taken directly from the Carl Martin Tremovibe, with depth and speed controls that cover all the range you're ever likely to need. Depth adjusts the amount by which the level is modulated while Speed sets how fast the level wobbles. While this pedal doesn't do anything unusual, it does work extremely smoothly with no obvious noise or unwelcome thumping, and the bypass switch seems pretty quiet too.
The Crush Zone (£54.99) is an old school distortion box with controls for output level, tone and distortion. It has a raunchy, raspy character reminiscent of the old MXR Distortion Plus that straddles conventional overdrive and fuzz. A tone knob can be used to smooth out the tone but even at the minimum drive setting, you can't really clean up the sound by backing off the guitar's volume control, so this is definitely a 'step on it and blaze away' kind of pedal. Within its genre, Crush Zone delivers exactly what is expected of it. It's great for power chords or solos but is perhaps less well suited to blues or country rock.
There's little detailed technical info on the Red Repeat (£79.99). Apparently it was derived from the Carl Martin Delayla pedal and, from the sound of it, it is either an analogue (charge-coupled delay line) delay pedal or an extremely good emulation of one. My money is on true analogue, though none of the available literature comes out and says it in so many words. In addition to controls for Echo Level, Time and Repeat (feedback), it also has a tone control that can be used to darken the sound of the repeats. At long delay times (the longest of which is 600ms), there is a little background noise in evidence, though rolling off the top end using the tone control masks this pretty well. The repeats get darker and grittier each time around, which is actually a lot more musical than the pristine delays of a standard digital delay pedal, and at more practical delay times of 400ms or less with just a bit of top rolled off, the effect is exactly right.
Unlike tape echo units (or their solid-state equivalents), which may use multiple heads to produce multiple delay taps, this one generates a single delay that can be made to repeat to a greater or lesser extent using the Repeat control. If you crank this up too far you get the familiar swirling, out-of-control effect so beloved of dub producers, but normally you'd keep away from such extremes. The old analogue delay sound is definitely something special and the Red Repeat captures it perfectly.
While these pedals offer nothing radically new, they are sensibly priced, extremely solidy built and the sounds they produce are exactly right for their respective genres. It is still worth checking out the usual suspects such as Boss, especially where distortion is concerned, as that company offer so many different variants of fuzz, distortion and overdrive. However, I respect the 'built like a tank', no-nonsense approach taken here, and I can see no reason why these pedals shouldn't outlast their owners. Paul White
SUMMARY: Retro sounds and styling from a well-respected effects designer.
First Line +44 (0)1392 493429.
Saturday, February 4, 2017
Friday, February 3, 2017
By Martin Walker
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.
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