I've recently moved into a new studio and have stumbled across a problem that's never concerned me before. Previously, my plug points have always been within arm's reach, but now, all my plug points are at floor level and not particularly accessible. What is handy, though, is the main power box with trip switches, which I've been using to turn all my gear on and off without touching their independent power supplies. Is this a really bad idea, or does it amount to the same thing as switching individual plug points on and off?
SOS Technical Editor Hugh Robjohns replies: Many studios commonly use a big master switch on the wall to kill and power up everything at once, although there are a few things to think about. First, if you're using the trips in the fuse box, I'm not sure they'd be adequate for regular use — it might be worth checking this with an electrician. Most studios seem to use a big cooker-style switch for the purpose.
Killing the power amp before everything else will help to protect your speakers, and turning it on after everything else will minimise the power surge when everything is trying to charge its power reservoirs. So long as the computer programs and the operating system have been shut down properly first, I wouldn't worry about using their front-panel power switches.
Depending on where your studio is, I'd talk to an electrician to make sure that the cabling and supply can withstand the kind of power surge you will get if you turn eveything on at once. This can pull in a considerably greater amperage for a few seconds than would be normal when everything's powered and running normally. I know of people who have burned down the 'studio in the shed at the bottom of the garden' this way!
Companies like EMO (www.emo.co.uk) can supply you with power distrubution boards that are designed to power up equipment in a specific sequence and time interval. They're expensive, but might be worth considering if you have equipment that draws a lot of power-on current.
I've noticed many contemporary recordings feature a prominent vocal track that's apparently dry, or are they just using a reverb setting that simply contributes to the general ambience of the mix, rather than sounding like an effect?
Assistant Editor Mike Senior replies: This kind of 'bone dry, but sitting nicely in the mix' sound that you can hear on a lot of pop productions is difficult to achieve, but here are some of my favourite tricks. The first is using reverbs with decay times well under a second (or dedicated ambience programs) so you don't get any audible 'reverb tail', even when you turn up the effects return. However, there are a number of other techniques I've found useful.
Try using mono, rather than stereo, reverb. Much of what makes reverb audible separately from the lead vocal (which is usually mono and in the centre of the stereo image) is its stereo-ness. Another easy thing to try is simply a mono delay, with little if any feedback, matched to the tempo of the track. Quite high levels of in-time delay will disappear into a track, especially if it's made quite dull by low-pass filtering or other EQ. (This also applies to reverbs — a duller reverb will hide behind the vocal better.) Bear in mind, though, that a dull reverb will dull the sound overall a little, so you can afford to increase the crispy high frequencies of the main vocal a little using EQ or psychoacoustic enhancement, particularly if you want that extra breathiness that seems to be so characteristic of pop vocals.
Furthermore, you can make the delays more diffuse by passing them through a chorus (mono again, though). Some people feel that subtle levels of double-tracking (real or automatic) can also make a vocal sit a little better in the mix, but this is a moot point, and you'll need to see whether this does the trick for you.
However, whatever effects you use, the most important thing to remember is to 'ride' your effects levels. There are a number of situations where this can be useful. Firstly, if you have a very sparse arrangement in your verse (meaning that a reverb or delay tail would be very obvious), fade the effects down just before any gaps in the phrase. It only needs to be done a little to be effective. Similarly, try using automation to duck the effect send during sibilant consonants — these in particular tend to show up reverb. And, of course, you can often bring up the reverb level if the arrangement gets fuller for the bridge or chorus.
For similar reasons, you can try altering the reverb time between the different arrangement sections — generally in pop music this will mean making it shorter for the verses and longer for the choruses. Of course, this relies on your reverb processor allowing you to change its decay time in real time without generating a whole load of zipper noise. If your reverb processor suffers from zipper noise, you'll have to use two different processors for the two lengths of reverb. Of course, this approach lets you differentiate the two reverbs in other ways as well, so it might be preferable anyway.
I'd like to measure how low the background noise is on my soundcard. How does Martin Walker produce the figures for his SOS soundcard reviews?
PC Music specialist Martin Walker replies: I normally use Wavelab 4.0 to provide rough-and-ready RMS background noise measurements as a comparison between review soundcards, although you can perform exactly the same procedure using Sound Forge or Cool Edit Pro. I record about 10 seconds of background noise with nominal gain, normally by leaving any input level control at its default or 0dB position, and if the soundcard has switched mic/guitar/line sensitivity, I make sure I'm using the line input.
I then use Wavelab's Global Analysis function to measure average RMS power. You'll need to pull the threshold setting right down to -144dB to make sure it includes the low-level noise, as at its default setting of -50dB it will simply assume that the entire recording is of silence and ignore it. In Sound Forge and Cool Edit Pro you can simply use the Statistics options, and again look at the average RMS power reading.
I normally get readings around -93dB for 16-bit recordings, and most budget soundcards achieve between -98dB and -100dB at 24-bit/44.1kHz, while more expensive cards such as Echo's Mona and M Audio's Delta 1010 manage -107dB to -110dB. The lowest I've measured to date is the Lynx Two, which turned in an amazing -116.5dB RMS. At 24-bit/96kHz all figures are normally a couple of dB worse due to the doubled bandwidth.
This only tests the A-D converters, of course, but since this ultimately determines the noise floor of all your recordings it's far more important than the D-A background noise. You can often disconnect the cable from the soundcard input during the test to make sure you only measure the card's noise, since in my experience it's rare that the noise will vary significantly depending on whether the input is open or short circuit.
If you're going to measure your own soundcard, beware of the odd design with noise gating — it's not unknown for cheaper cards to provide false figures because they completely mute the input circuitry until a low-level signal is detected. Others may give false readings because they generate a short spike as they enter record mode (early versions of the Echo Mia did this), or occasionally because there's a DC offset, as in the case of Creative Labs' Audigy.
However, although comparing background noise figures is certainly one way to look at soundcard audio quality, you shouldn't rely on it to tell the whole story. Other factors such as frequency response, distortion levels and clock jitter all enter the equation, which is why using your ears is always the ultimate test. Some of the soundcards I've reviewed have had higher than average background noise levels, but still had subjectively good sound quality. In fact, some soundcard manufacturers have told me that during the design process they deliberately chose the version that sounded best, whether or not it had the best measured sound quality.
I'm looking for some advice on connecting line inputs to a Mackie eight-buss desk. I have a 16-way studio box going into all of the XLR/mic inputs of the desk, connecting the studio to the control room. Is it OK to plug a keyboard into the wall box using a jack-to-XLR lead, or would I be better off getting a second wall box that goes to the line inputs of the desk? I could always remove four of the XLRs at the desk and swap them for jacks as I never use all 16 mic inputs at once, but I'd prefer not to do this if I could get away with it.
Technical Editor Hugh Robjohns replies: Although most mic inputs will be able to cope with the outputs from a keyboard (if you turn the input gain down), it's not a particularly good idea. There's a danger that phantom power will fry the output of the keyboard, or at the very least cause loud bangs when you plug things in.
It would be far better either to dedicate a bunch of your mic tie-lines as line-level links, or to install a second wall box for line levels. If you choose the former option, the easiest and best way would be to purchase four Neutrik TRS sockets, which should be drop-in replacements for the XLR sockets on the wall box. Wire them in the same way, preserving the balanced line, and at the desk end replace the XLR plug with a TRS quarter-inch plug, again, wired in the proper balanced manner.
I have some questions concerning the physical frequencies of notes. 1) I'd like to be able to calculate every frequency of every separate note. 2) After some research, I discovered that there are two standards: one placing the reference at 440Hz and the natural, where the reference isn't exactly 440Hz. Which of these is most commonly used? Is there a difference between every instrument, or are they all tuned to the 440Hz of a piano (that is, if the piano is tuned to exactly 440Hz)? 3) Is the difference inaudible if the two standards are mixed together?
Rudy (via email)
Assistant Editor Mike Senior replies: 1) It might help to have a look at the graph printed in the Using Equalisation article in SOS August 2001. This gives rough instrument ranges against a keyboard, with the frequencies of the white notes given above. If you want to work out the frequencies of any of the black notes, simply multiply the frequency of the white note below it by 1.0594631.
2) I'm afraid that the subject of tuning is much more complicated than that. Even assuming that you want the semitones within each octave evenly spaced (a tuning called equal temperament, only one of many tunings available even in the rarified world of western classical music), then you'll still find that different orchestras in different coutries have different preferences. This is why good electronic tuning devices will give you the option to set the tuning reference to suit yourself. And remember, as well, that most orchestras will tune to the piano in a concert, even if it is tuned slightly off, as it's quicker to retune the orchestra than to retune the piano, so the tuning of a given orchestra can change even during a single concert.
3) I don't know about you, but I hear it as unpleasant if an instrument is out of tune by even 10 cents (a tenth of a semitone) in a track. Assuming a 440Hz standard, if you played your concert A 10 cents sharp or flat, you'd be playing a note at about 442.5Hz or 437.5Hz respectively. This would suggest to me that mixing instruments tuned to A=440Hz with instruments tuned at A=435Hz would make the overall performance sound unpleasantly out of tune. However, assuming for a moment that it's a full orchestral performance, it would depend what instruments were tuned wrong. Even if his or her instrument were tuned slightly out, any violinist worth their salt would almost certainly hear the anomalies and alter their playing to compensate, and would also avoid playing open strings (in fact, open strings are discouraged as a matter of course in some professional orchestras, I believe). Most good wind instrument players would probably do the same. Keyboard instruments, harps, and tuned percussion, however, don't allow any such tuning compensation during performance in most cases, so those would probably stick out a mile if tuned to the wrong reference.
I have a Rode NT2 microphone and I put a couple of silica gel sachets in its box. Someone told me that this will only work for a couple of months, though, and someone else said that you have to put the sachets in the oven — is that true? Are there any other things I can do to stop moisture getting onto the diaphragm of the mic?
SOS Forum post
Technical Editor Hugh Robjohns replies: If you plan to store your microphone for a while, putting some silica gel sachets in the box is a reasonably good idea to ensure the capsule remains nice and dry. You can get suitable sachets from most camera shops if you need them.
The idea of the gel is obviously to absorb moisture, so it will keep the air dry if you put a sachet in an enclosed space such as a mic box. Over time the gel will become saturated and will be unable to absorb any more water, though how long this takes depends on the environment it's in.
Some gels have a colour indicator that shows when the gel is saturated. All you then have to do is put the sachet in a warm, dry airing cupboard for a few days — the longer the better really — to encourage the trapped moisture to evaporate, leaving the gel ready to resume service with your microphone. Personally, I wouldn't put sachets in the oven in case the sachet envelope catches fire!