I did a recording session recently using a mixture of dynamic and condenser mics, and realised my desk does not have switchable phantom power for each individual channel — they're either all on or all off. As it happens, I had a second mixer and some external channel strips which I ran the condensers through, but, for the future, is it safe to apply phantom power to dynamic mics?
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Technical Editor Hugh Robjohns replies: People get very hung up about phantom power. As long as your mic cables are all wired properly (balanced, with the correct pin connections) and well made, and you are using decent XLRs everywhere — and all your microphones are modern — there is no problem at all.
In BBC radio and TV studios, for example, phantom power is provided permanently on all wall box connections. It cannot be turned off. And engineers are plugging dynamic, condenser and even ribbon mics in and out all day without any problems whatsoever.
Clearly, it is vital that dynamic and ribbon mics are properly balanced internally and well maintained, but this should be a given with any modern mic. The female connectors on good-quality XLR cables should have the contact of the earth pin socket (pin 1) slightly forward of the other two so that the earth contact mates first, and are designed so that the other two pins mate simultaneously. There is therefore little chance of subjecting the mic to significantly unbalanced phantom voltages.
There will be a loud 'splat' over the monitors when connecting a condenser mic as the circuitry powers up, but it is good practice to always keep the channel fader down when plugging in mics anyway. I don't disagree that plugging mics in with phantom off is a safe way of working, but I have never really bothered about it, and have never destroyed a mic yet — not even a ribbon, and I've used a lot of those over the years.
The important issue about ribbon mics is that it is safe to plug in ribbon mics on circuits carrying phantom power, provided the ribbon mics in question are compatible with phantom power. Some vintage ribbon mics employ an output transformer which is centre-tapped, and that centre tap is earthed. This arrangement essentially short-circuits the phantom power supply and can cause damaging currents to flow through the transformer, potentially magnetising it or even burning it out (although that is extremely unlikely). So it is sheer lunacy to be using vintage ribbon mics with centre-tap grounded transformers in an environment where phantom power is also used. Sooner or later, a ribbon will get plugged into a phantom supply by accident and will be permanently damaged. If you want to use vintage ribbons with centre-tap transformers in the same room as phantom-powered condensers, get the ribbons modified before it's too late.
The bottom line is that all modern mics with balanced outputs terminated with XLRs, whether they be dynamics (moving-coils and ribbons) and electrostatics (condenser and electrets), are designed to accommodate phantom power, and can be plugged in quite happily with phantom power switched on, provided you are connecting XLRs, not jack plugs/sockets. Some vintage ribbon mics, and any mic wired for unbalanced (sometimes also referred to as high-impedance) operation will be damaged by phantom power unless suitably modified.
I've been told that, if possible, you should always cut rather than boost when EQ'ing. So, for example, if you need more bass, you should cut the high- and mid-frequencies and raise the overall level, rather than simply boosting the low end. Is this true, and, if so, why?
Technical Editor Hugh Robjohns replies: Firstly, let me say that the governing factor when applying EQ should be how it sounds, so if it sounds right — regardless of whether you have used boosts or cuts, or both — it is right! Secondly, if you weren't meant to boost signals the designers of console and stand-alone equalisers wouldn't have provided their EQ controls with a boost side! So rest assured that boosting is allowed, and you won't be dragged off to the Sound Processing jail to be shot at dawn if you get caught using boosts.
Having said all that, I generally only use EQ boosts when I need to apply a relatively small amount of gentle, wide-bandwidth tonal shaping. So if the bass end needs a little lifting, then I would probably boost the bass a little using a wide-bandwidth shelf equaliser. On the other hand, if I needed to do something more dramatic in the way of tonal shaping or corrective EQ'ing, I'd almost certainly try to do it with cuts. The usual technique is to wind in a fair bit of boost and then dial through the frequency range of a parametric equaliser to help find the problem frequencies. Once located, reverse the gain control to cut the offending frequency area.
There are several good reasons why cutting is often a better idea than boosting, particularly when applying large amounts of EQ, such as is necessary when trying to correct the sound of something. The first is the issue of headroom in the EQ circuitry. Boosting quickly eats into the system headroom, and you risk transient distortion when fast peaks run out of headroom.
Next, if you need to use high-Q (narrow-bandwidth) filters, the ear seems to be very sensitive to their effect when boosting, but surprisingly oblivious when cutting. The result is that the effect of the EQ is often far more subtle and less audible if cuts are used rather than boosts.
In the case of live sound, using EQ boosts tends to increase the risk of acoustic feedback. It's true that if you cut and then bring the level back up you are equally at risk, but in practice people tend to subconsciously know that raising the channel gain risks causing feedback and they take care. On the other hand, most people don't seem to associate turning an EQ gain knob up as adding gain — only as changing the tonality — with the result that as they try to make the vocalist sound a little sharper, the PA squeals embarrassingly!
I have just bought Emagic's EVB3, EVD6, and EVP88 software instruments. I also have the EXS24 MkII software sampler. I've fallen absolutely in love with all of these items and the opportunities they afford me in my studio. However, I want to use them with my band in a live setting, in both rehearsals and also on stage live. My guitar player's girlfriend has a four-year-old Gateway Pentium III laptop with 64MB RAM and Windows 98SE installed, and she has agreed to let me do whatever I need to so that we can use it for music. She doesn't have anything on there that needs saving, and we just want to wipe the thing clean and end up with Windows XP for an OS, 320MB of RAM and Emagic Logic Audio Platinum with the software I have mentioned.
However, she knows nothing about computers and neither do I. I have never deleted an OS before, I have never wiped a hard disk, and I have certainly never installed RAM on a laptop. I have the Windows XP disc that came with another PC, but I have no idea if this OS can be installed on another machine. What should I do?
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PC specialist Martin Walker replies: You can certainly start again from scratch with a PC laptop, but this isn't an easy task if you haven't done anything like it before. If you want to install more RAM, then you'll first need to check that a spare slot exists for further expansion — many laptops are shipped with all their memory slots already filled, so you might have to discard at least some of the existing RAM to install larger-capacity chips. It may be fairly obvious how to access the innards of your laptop to check, but a trip to the Gateway web site (www.uk.gateway.com) should yield an explaination of how to do this in a safe way.
If possible, installing the extra RAM is best done before anything else, so you know it's been recognised correctly, and can rule it out as a possible cause if you get problems later on. You'll need to get RAM that's compatible with what's already in the laptop, and once again the Gateway web site is the best place to look first, although the chances are that generic RAM from other suppliers may also be available at lower prices to suit your model.
If you chose to install Windows XP, you would need to buy another license for it, since it needs to be activated remotely by Microsoft for each individual machine if you want to use it for more than the 30-day grace period. In your situation, sticking with Windows 98SE would be a better bet, since on a laptop that's at least four years old, XP drivers may not exist for its various motherboard devices, and you already have a perfect right to re-install the existing operating system. Wiping the current contents would certainly be a good idea after all this time — you can do this as part of the Windows install procedure by using the Delete Partition option, but make absolutely sure you've first backed up any data belonging to your guitar player's girlfriend, or be absolutely sure that there's nothing she needs to keep, or you won't be very popular!
All PCs should have the latest motherboard drivers installed straight after Windows as a matter of course, since this will ensure that you don't have any 'exclamation marks' in Device Manager that indicate unknown motherboard hardware devices. If your guitar player's girlfriend still has the original driver CD-ROM that came with her laptop then this should include suitable drivers for Windows 98SE, but you should still try to track down the most recent versions from the Gateway web site if possible. Personally, I don't like having Microsoft Word installed on any music PC, since it has a habit of running background tasks behind your back that can interfere with the smooth running of music applications, but I described how to track these down and disable them in SOS December 2001's PC Musician feature (www.soundonsound.com/ sos/dec01/articles/pcmusician1201.asp).
However, even once you've re-installed Windows 98SE on a freshly formatted partition, installed the various chipset drivers, and added more RAM (if available), you're still faced with the prospect of running Logic Audio Platinum plus a selection of software synths and samplers on a Pentium III laptop. Most people now consider a desktop Pentium II 233MHz the absolute entry-level machine for running music applications, making your Pentium III laptop a good contender to run a selection of audio tracks, some plug-ins, and perhaps a synth or two, but you'll be really struggling to use all the software you mention simultaneously. You'll also need some sort of external sound device, since relying on the motherboard sound chip with its high latency and low audio quality is a bit risky, and some laptops have a habit of occasionally interrupting sound chip playback to do other things that you may not be able to trun off. You'll also need to disable any power-saving technologies to get maximum CPU performance.
Overall, even after all this work, while you might find the updated laptop useful for rehearsals, I think you might regret trying to use it live. As many musicians will tell you, PCs and live venues can be an unhappy mix, and you really need to know your way around any computer in this scenario if the worst happens and you need to give it emergency first aid. While I don't want to sound too pessimistic, from a standing start with no previous experience I personally think you've chosen a tricky task, but if you're determined to give it a try it may be worth trying to find a local 'PC guru' to give you a hand. Good luck, whatever you decide!
I was told by a sound engineer that, when mixing, it is not a good practice to have all the channel volume faders way up and to have the master fader down, and that this applies both to analogue and digital consoles. Is this true? I thought that if none of the channels is clipping, then there should be no problem with putting them up and keeping the master fader at a lower level.
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Technical Editor Hugh Robjohns replies: This is all a question of optimising the gain structure — or, in other words, setting the correct level at each stage of the signal path from start to finish in order to avoid overloads or deterioration in sound quality — something which is rather more critical when using an analogue console than when using a digital one.
In an analogue desk, the most likely point of overload when mixing is at the mix buss amps, so you have to set the gain structure of the front-end (voice channels, mic preamps and so on) and mixer channels correctly to optimise headroom and noise floor when these signals meet at the mix buss. Lowering the master output fader won't affect the mix buss in any way, since the master fader comes after this part of the circuitry.
So, optimise the channel input gains with the channel faders at their unity positions, and pull down the channel faders (or reduce the input gains) if the mix buss gets too hot. In a recording/mix environment the master fader should ideally be on its unity gain position at all times unless you are fading out. In live sound situations you may want to have the master fader rather lower than unity to enable better level control of the PA system, leaving room to gradually crank the level up through the set, for example, or after the support act!
In the case of digital consoles, the situation is rather different, because the signal processing is done in a different way. In systems which employ floating-point maths, you cannot overload the notional mix busses however hard you try. With fixed-point systems the headroom is considerably less [if you read the answer to the next question you'll understand why — Ed] and it is potentially possible to overload mix busses, which is one reason why fixed-point digital console makers like Yamaha provide a digital attenuator before the EQ section of each channel.
Reviews Editor Mike Senior adds: I'd definitely agree with all that Hugh has said, although there's another thing which needs to be taken into consideration too: fader resolution. Most faders — analogue and digital — have their maximum resolution around the unity gain mark. This means that, around unity gain, moving the fader by a small amount effects a small change in level, while moving the fader by the same distance at the very bottom of its travel, where resolution is much lower, equates to a much larger change in level. When you're mixing (particularly with automation) very small changes in level can become very important, so it's useful to have the channel faders near the unity gain position for that reason.
The only way to do this and also keep the master fader near unity, however, is by using the channel input gains (or the digital channel attenuators on a fixed-point digital console) to get your mix into the right ballpark initially, only tweaking the faders once a rough balance has been set. On some digital consoles you can assign the attenuator levels to the faders temporarily to make this initial process simpler. If no channel input trimmer is available, you can sometimes adjust the output levels of the individual tracks from the multitrack recorder (or whatever your source for the mix might be) instead.
Unfortunately, many manufacturers don't provide the pre-EQ digital attenuators that Hugh mentions in their fixed-point systems, especially at the lower end of the market — all-in-one multitrackers in particular often don't include them — in which case you have no choice but to keep the channel faders quite low if you're going to avoid overloads, at the expense of mixing resolution. In the case of Roland's VS880, for example, there's no more headroom on the mix buss than on the individual channels, so you can easily encounter clipping at the mix buss. Furthermore, as on an analogue console, turning the master fader down doesn't avoid the clipping, it just reduces the level of the clipped signal!
Could you clarify the difference between floating- and fixed-point 32-bit operation in the digital domain. I know that floating-point systems allow for data to be handled at word lengths above 24-bit, which are then dithered back down. Does it also result in a greater dynamic range?
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Technical Editor Hugh Robjohns replies: Accurate digital audio capture and reproduction requires, at the very most, 24-bit resolution. The reasoning behind this is that a 24-bit signal has a theoretical dynamic range of 144dB, which is greater than the dynamic range of the human ear, so, in theory, a 24-bit system can record sounds slightly quieter than those we can hear and reproduce sounds louder than we can stand. There is therefore no need for A-D/D-A converters to work at resolutions higher than 24-bit.
However, when it comes to processing sound within a digital system, there needs to be some headroom to accommodate the fact that adding two 24-bit numbers together can produce a result which can only be described using 25 bits, and adding 30 or 40 such numbers together can produce something even bigger. At the other end of the scale, the mathematical calculations involved in complex signal processing like EQ generates very small 'remainders', and these have to be looked after properly, otherwise the EQ process effectively becomes noisy and distorted. The natural solution is to allocate more bits for the internal maths — hence 32-bit systems.
Fixed-point systems use the 32 bits in the conventional way to provide an internal dynamic range of about 192dB. Systems that use fixed-point 32-bit processing (like the 0-series Yamaha desks) usually arrange for the original 24-bit audio signal to sit close to the top of that 32-bit processing number to provide a lower noise floor and slightly greater headroom for the signal processing. (Incidentally, a 192dB SPL is roughly equivalent to two atmospheres' pressure on the compression of the wave and a complete vacuum on the rarefaction.)
Floating-point systems also use 32-bit numbers, but organise them differently. Essentially, they keep the audio signal in 24-bit resolution, but use the remaining bits to denote a scaling factor. In other words the 24-bit resolution can be cranked up or down within a colossal internal dynamic range so that, in effect, you can never run out of headroom or fall into the noise floor — there is something like 1500dB of dynamic range within the processing, if the maths is done properly.
Most high-end consoles and workstations employ floating-point maths because (if properly implemented) you can get better performance and quality in the computations. Most budget/low-end consoles and DAWs use fixed-point processing because it's easier and faster, and can be implemented in hardware more easily.
I have a melody which I am absolutely sure I made up when I was a kid and I want to use it in a composition I am working on. However, the things kids make up can be 'derivative' at the best of times, and I need to confirm that I hadn't heard it before. I am fairly sure that I did write it myself because I have never heard it since in the following 20-odd years, but it would be nice to be sure.
Do you know of any melody databases that could help me find out if I have lifted the tune from somewhere? I found one on the web (www.melodyhound.com) where you enter the Parson's Code of your tune and it checks a database, but I have no way of knowing how exhaustive their database is. Can you offer any advice?
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Reviews Editor Mike Senior replies: There are a number of melody databases accessible on-line, using a variety of different methods to catalogue and recognise melodies, though none of them is by any means exhaustive. You're probably no worse off humming your melody to friends and family of different ages to see if it rings any bells. However, if you are planning to release a track containing this melody, you may need to take the issue a bit more seriously.
Disregarding the significant moral issues here, the legal situation in the case of similar melodies, as I understand it, is very much a grey area, and far less straightforward than in cases of mechanical copyright infringement (using a sample without permission). If the context and lyrics have completely changed, then the prosecution would be obliged to show not only a serious similarity in both note pattern and rhythm, but would also probably have to show that you had sufficient access to the material from which it was taken. An interesting case study is the 1984 copyright infringement case against the Bee Gees, where unknown songwriter Ronald Selle accused the brothers Gibb of lifting the melody from his song 'Let It End' for their hit 'How Deep Is Your Love'. You can find information on the case by searching the web for 'Selle v. Gibb'. Alternatively, Columbia Law School's Music Plagiarism Project web site, which can be freely accessed at www.ccnmtl.columbia.edu/projects/law/library/entrance.html, features a wealth of information on music copyright infringement cases in the US over the last 150 years, in the form of documents, scores, audio and video files.
In the Bee Gees case, though the melodies in the two songs were agreed to be all but identical (even Maurice Gibb mistook part of Selle's song for the Bee Gees own while on the stand) and Selle had written and copyrighted 'Let It End' several years before the Bee Gees wrote 'How Deep Is Your Love', the claim of copyright infringement was rejected by the judge. Why? Because Selle couldn't prove that the Bee Gees could have heard his song before they wrote theirs, and couldn't rule out a common source which inspired both songs. 'Let It End' was rejected by a number of record companies and never released, and Selle was forced to admit that there were similarities between both songs and a number of others, including a previous Bee Gees hit. It would seem that in this kind of case, the onus is on the party making the accusation of plagiarism to show not only that the similarities in melody are such that they can only be explained by copying, but also that the alleged copyists could definitely have had access to the original melody and that they couldn't have both copied it from somewhere else.
Although I'm no legal professional, I have been told that this last point can give rise to a defence against accusations of copyright infringement, namely to cite a piece of music that's in the public domain, such as an old classical work or a piece of folk music, as the source of your contested melody. Quite a few composers of high-profile music have sought out this kind of copyright-free music in the past, as an insurance policy in case anyone should have a pop at them. Accusations of plagiarism are more often than not groundless (at least, I'd like to think so), and the more successful a composition is, the more likely it is that someone will believe that the money and fame should belong to them. Having a piece of copyright-free music to claim as a source could squash any suit before it gets off the ground. Again, there are databases which can be used to look for suitable material, but they're rare, notation-heavy, and all of them index the melodies differently. Finding instances of a given melody is a long-winded and painstaking job (I speak from first-hand experience on this one!), even if you have access to the books and you have a musicology degree.
If you're worried that your melody is a knock-off, go ahead and record it anyway. When you listen to the finished article, with your lyrics, your arrangement and your performance, you may well find that it is your song after all.
I have a compressor with inserts to access the side-chain. What is the side-chain and what would you use these inserts for?
Technical Editor Hugh Robjohns replies: All dynamics processors — compressors, gates and de-essers — use a gain-controlling device to alter the instantaneous level, and thus the dynamics of the input signal. The gain controller could be a solid-state VCA, a valve, or an opto-resistor, depending on the design of the unit, but whatever the type of device, it has to be controlled by a circuit which looks at the signal and decides how much to reduce its gain. This control circuit is known generically as the 'side-chain'.
The reason a lot of dynamics processors provide access to the side-chain signal is to allow additional external signal processing to modify the signal that the side-chain is working with, and the usual process is an equaliser. It is worth emphasising at this point that we are talking here about full-band compressors — whatever we do to the side-chain signal, the level of the entire input signal is affected when the compressor operates. In multi-band compressors, the input signal is split into several frequency bands and each is processed separately, which is a very different thing and should not be confused with the approach I'm describing here.
Consider the situation where a very bass-heavy mix needs to be compressed. The amount of compression is determined by the amount of energy in the side-chain, so in a bass-heavy mix, the amount of compression is going to be heavily influenced mainly by the bass signals because that is where most of the energy is. In extreme cases, the result might be heavy gain reduction on each kick-drum hit or bass-guitar note, which might not produce the required overall control at all.
One solution is to insert an equaliser in the side-chain to filter out or reduce the side-chain signal's low-frequency energy. The result is that the compressor side-chain now determines the amount of compression on the energy in the mid- and high-frequency ranges, which will probably produce a more natural and consistent sound.
So, because the side-chain tells the compressor what to do, reducing the level of a frequency band using an equaliser in the side-chain also reduces the compressor's sensitivity to energy in that frequency band.
Another technique using equalisation of the side-chain is to deliberately increase the compressor's sensitivity to certain frequency bands, and the classic example here is the de-esser. Sibilant singers, for example, tend to produce peaks of excessive energy in a fairly narrow frequency band somewhere between 2kHz and 8kHz. Ideally, such problems should be avoided through careful mic selection and positioning, but if you are faced with a recording already containing sibilance, you can often deal with it quite effectively using a de-esser, or a compressor with an equaliser inserted in its side-chain.
By using the side-chain equaliser to boost the frequency region containing the sibilant energy, the compressor is made more sensitive to that frequency region. As a result, even quite small increases of energy here will result in quite large amounts of compression, thus reducing the audible impact of the vocalist at the moment of sibilance. It can take a little juggling of equaliser boost, threshold, ratio and attack/release settings, but when properly optimised this system can prove extremely effective.
Side-chain equalisation can also be used with gates and expanders in the same way. Consider a snare-drum mic which has a lot of spill from both the hi-hat and the kick drum. If the energy level of hat and kick is sufficiently high, it might prove impossible to find a threshold setting which allows the gate to let the snare through but hold back the kick and hats. One solution would be to reach for the side-chain equaliser again — and adjust it to remove both the low kick-drum frequencies and the higher hi-hat frequencies, leaving just the mid-range snare drum frequencies. A pair of high- and low-pass filters are ideal for this job, but high and low shelf filters will usually work as well. With only the mid-range snare drum signal left in the side-chain, the gate is left in no doubt when to open and close, and setting the threshold is now remarkably easy!
These are just a few obvious examples of side-chain equalisation, but it is a powerful technique which can be used in a wide range of applications to solve problems or to make dynamics processing more effective and easier to control.
In practice, it is useful to be able to listen to the output from the side-chain equaliser to help fine-tune the EQ settings — doing it 'deaf' is not ideal. A lot of hardware and software compressors incorporate some form of side-chain equalisation as standard, and these units usually have a 'Listen' mode which routes the side-chain signal to the output to allow auditioning of the EQ settings.