Isn't it time you tried a REALLY different microphone?

OXSA Outbound | February 19, 2013

You've tried all the usual microphones and are tired of their sound? Why not try something that is really over the edge...

If you're into microphones then you might have noticed that there is a certain 'sameyness' about the standard models.
You might choose to use a dynamic mic, a ribbon, a small- or large-diaphragm capacitor, or a tube mic, maybe even a vintage model.

Each type of mic has its own characteristic sound, but within types they sound quite similar. Yes there are differences between large-diaphragm capacitor mics, for instance, but they are not huge differences, like the differences between mic types.

So to get a sound that is really different, perhaps it would be an idea to choose a mic that stands out from the crowd.

And of course we have an example - the Coles 4104 commentator's lip mic. We saw this example in an eBay auction. Here are some more tasty photos...






By the way, we don't have any connection with the seller other than we asked his permission to use the photos.

The Coles 4104 is a noise-canceling microphone. It subtracts sound arriving from a distance while leaving sound immediately in front of the microphone untouched.

This makes it ideal as a sports commentator's mic, where there is likely to be a lot of background noise. The mic is held with the upper guard piece touching your top lip. This makes it a no-brainer for a non-technical person to use the mic.

You could try noise canceling for yourself with two directional microphones - place them back to back and flip the phase of the rear mic. Speak into the front mic from a close distance. Since background noise arrives at both mics more or less equally, flipping the phase of the rear mic makes it cancel out to a significant degree. But since the sound of your voice is much stronger in the front mic, it hardly cancels at all.

But the Coles 4104 has another trick - it is very good at handling the pops and breath noise that you get when a mic is used close to the mouth. It's a design that other manufacturers might consider taking a look at.

Oh, and there's one more feature - this mic is insensitive at the sides. This means that two commentators can sit next to each other and leakage will be minimal.

You have already heard this mic on many occasions on TV. Even beyond the realms of sport it is useful for outside broadcasting in general.

As well as its useful features for its intended purpose, this mic has a characteristic sound all of its own. You won't find another microphone that sounds like it.

The sound is amazingly clean considering how close to the mouth it is used. You couldn't say that it is an accurate sound, but it's something that could be used in many contexts as a contrast to the standard mic sound.

There's another use for it in live sound - you know that you occasionally hear a song that features a distorted vocal, either all the way through or in segments? (Can we blame John Lennon for starting that?).

Well if you use a distortion effect on stage you will find that the high gain involved increases the risk of feedback significantly.

But if you use the Coles 4104 for this purpose, then since it rejects the sound coming from the speakers, it is very robust against feedback.

In summary, this mic is excellent for its intended purpose. But it also has an interesting sound that might find a place in your studio, or perhaps even live.

Note: This auction is now closed. The winning bid was £217 UK pounds.

Apogee Duet audio interface features soft limiting to prevent clipping

OXSA Labs | February 18, 2013

Should a pro engineer prevent clipping through correct gain setting? Or can soft limiting be relied on to save the day?

"This superior analog design prevents the digital clipping that causes distortion by instantaneously rounding off transient peaks before they hit the analog-to-digital converter. Soft Limit allows several more decibels of apparent level to be recorded while subtly providing an analog-like warmth to the sound." - Apogee

Apogee features 'Soft Limit' on several of their audio interface products, including the new Duet. The idea is that if you set the gain too high, Soft Limit will prevent digital clipping. Not only that, it adds warmth to the sound too. But is this a good idea for the budding engineer or producer?

What is clipping?

Clipping occurs in a recording when the preamplifier gain is set too high. The signal attempts to go over the maximum level that can be stored in a digital audio file (0 dBFS) and as a result the tips of the waveform are sheared off flat. The sound is distorted and harsh. Note 'harsh', not 'warm'.

How can clipping be prevented?

The usual way is to do a level test before recording. So the singer sings a few bars, the drummer goes round each drum, etc. - The level of each performer is tested and the gain of each channel set so that clipping does not occur.
But...
It is commonplace for players and singers to perform louder in an actual take than they do during the level test. The engineer will anticipate this and set the gain lower than the level test might have indicated. The engineer may choose to set the gain 6 decibels lower than the setting that was just below clipping. This is known as leaving 6 dB of 'headroom'. The more unpredictable the signal level, the more headroom should be allowed.
Headroom is an almost foolproof method of preventing clipping, in the hands of an experienced engineer.

So why Soft Limit?

Although the wise engineer will allow plenty of headroom, there is always the possibility that he or she might have underestimated. If so, the recording will be clipped. If the recording is being made under studio conditions, then the answer is to lower the gain and go for another take. But if it's a live recording, then the clipping has done its damage and is permanently fixed in the recording.
But if a limiter is used between the preamp and audio interface, or an Apogee interface with Soft Limit used, then an instance of unexpectedly high level doesn't have to mean disaster. The recording won't be entirely clean - it will be limited or soft clipped and will sound as such. But that's a lot better than it being clipped.

Should you rely on Soft Limit?

In professional terms, no. An engineer who can't set the gain correctly should find another profession that doesn't require such a high degree of care, attention and precision. Keeping Soft Limit as a last fallback position is OK, but relying on it is completely the wrong thing to do.

What about the 'warmth' Apogee claims?

If Soft Limit provides a kind of sound texture that you like, then there is no reason why you shouldn't use it. Bear in mind however that any processing that is used at this stage is locked into the recording and cannot be undone. You might consider that it is more flexible to make clean recordings that you can process any way you like during the mix.

More decibels?

"Soft Limit allows several more decibels of apparent level to be recorded, as Apogee says."
Let's say that you push the level 3 dB over the clipping point. Soft Limit will round off the top 3 dB of your recording, leaving the peaks just below 0 dBFS, so there is no actual extra level. The lower levels that would not have clipped will however be 3 dB higher, at the expense of the overall accuracy of the recording.

Takeaway...

Soft Limit is a useful tool as a last backstop of protection, but a professional engineer would hardly ever need to use it. If you like the sound of Soft Limit, then it's fine to use it, as long as you realize that it can't be undone later in the mixing process.

How to pan an acoustic piano

OXSA Outbound | February 14, 2013

Should the bass notes be on the left and the high notes on the right, or the other way round?

Keyboard orientation

If you sit at a piano keyboard, then the bass notes are on your left and the high notes are on your right, so it would seem that this would be the correct way to record the instrument. Pan the bass notes left and the high notes right.

Audience orientation

But what if you are a member of the audience at a piano recital or concert? The piano will be sideways-on, with the player at the left. Since the bass notes have longer strings, their sound comes from the full length of the instrument. The high notes have shorter strings and thus are heard mainly towards the keyboard end. Therefore the high notes come from the left, and the bass notes extend all the way to the right, the reverse of what the player hears.

What the player actually hears

Since the notes of the piano keyboard extend all the way from left to right in even semitone increments, it may seem that the sound from the strings should be the same way in the stereo sound stage from left to right. So the lowest note A is all the way to the left; the highest note C is all the way to the right; Middle C is in the middle and every other note is positioned in proportion.

This however doesn't take account of the fact that in nearly all pianos, the strings cross over to save length (in a grand) or height (in an upright). It is a very rare piano where the strings are all parallel.

So when you sit down to play the piano, the low notes are 'left-ish' and the high notes are 'right-ish', but there isn't any precise sense of positioning. There is however an even spread from left to right, and this is something that should be captured in a recording.

To pan stereo mics from the player's perspective all that is necessary is to pan the mic that is pointing to, or closer to, the bass end of the piano to the left. Pan the other mic to the right. If the piano is to be mixed in with other instruments then you probably don't want it to sound as wide as the distance between the speakers, so pan the channels inwards to get the width as you want it.

What the audience actually hears

In the concert hall, only people in the front couple of rows hear any directional information from the piano at all. Any further back and the direct sound from the instrument is virtually a point source. All of the stereo information comes from the ambience of the auditorium.

So you might consider this if you want your recording of piano to sound like a Beethoven sonata. The close mics should be panned almost center and the ambient mics panned hard-left and hard-right. If you're not using ambient mics, then you will need to add a very natural-sounding reverb.

Takeaway...

Sometimes the nature of the equipment we use leads us to create a sound that doesn't correspond all that closely to what would be heard in reality. It is always useful to think about what would be heard naturally without the mics, then aim to mimic that sound in the finished recording.

Universal Audio Twin Finity combined tube/transistor mic preamp

OXSA Labs | January 13, 2013

Is this dual tube/transistor preamp the ultimate in preamp flexibility? Or does it completely miss what could have been its finest trick?

Although some will say that it is best to capture a very accurate sound on recording, and process it later any way you like, there's a certain magic in getting exactly the 'right' sound at source. In general, transistor preamps provide all the accuracy anyone could want (providing the mic is accurate too of course), and vacuum tube preamps give a kind of warmth that is difficult to achieve any other way.

So why not have the best of both worlds? The Universal Audio 710 Twin-Finity Tone-Blending Preamplifier & DI Box (to give it its full title) seems to do just that. It combines a transistor preamp with a completely separate tube preamp, and blends the two together, controllably, into a single output. So you can have the transistor sound (which is actually no 'sound' at all), the tube sound, or anything in between.

One problem however is that to get exactly the right sound, you have to know what you want beforehand. This takes experience. It would be quite easy to undercook or overcook the warmth. Then you're stuck with what you have done for the rest of the recording process all the way to mix and master. (You could add extra warmth by other means if you wanted to, but that would seem to negate the whole point of the unit.)

And this is where the Twin-Finity gets it totally wrong. Rather than blending the outputs of the two preamps into a single output on the rear of the unit, there should have been two outputs - one for the transistor preamp, one for the tube. They could have been recorded onto two tracks and the blend could then be adjusted at any later point in the recording process.

Still, it's an interesting device and it is always useful to have more tone colours to play with. And Universal Audio should be applauded for their innovative thinking.

Can you now use nearfields to completely replace your main monitors?

OXSA Labs | January 13, 2013

Any pro studio would have both main monitors and nearfields. But now you can have both in one package.

In the olden days of recording, a pro studio would have a pair of really good loudspeakers so that they could hear every detail of what they were recording.

It's a law of the universe that really good speakers also have to be large, otherwise a) they can't reproduce the bass, and b) what bass they can reproduce isn't as clean as it should be.

But there would also be a tiny speaker equivalent to what you would find in a 1960s or 1970s transistor radio.

That was used to hear the sound the way a typical listener would hear it.

But by the 1980s, people had hi-fi systems at home. Well they were called hi-fi. The fi wasn't generally as hi as all that, and the speakers were usually quite small. About the size of nearfields, coincidentally.
Actually it's not a coincidence, because nearfield monitors became popular to emulate this new 'medium-fi' experience in the home.

At the same time, engineers discovered that by having the monitors really close, the acoustics of the control room became less of an influence.

So in time, monitoring on nearfields became pretty much the standard.

But there is a problem...

If you need to have large loudspeakers to reproduce a recording really accurately, wouldn't you be making a mistake by monitoring and mixing on small nearfields?

Well it depends on your philosophy of sound really. Should the sound on the recording be perfect in every way, or should it be adapted to suit the likely listening conditions?

You'll have to make your own judgment on that because there isn't one correct answer.

But if the sound quality of nearfields is a concern, why not bring the main monitors closer?
Well you can do that, but then you won't be able to check what domestic reproduction would sound like because you have now occupied the space where the nearfields could have been.

But now you can have the best of both worlds in one package. Well, two packages for stereo.

Welcome the Focal SM9, which is like a nearfield and a main monitor in one box.
This trick is achieved by providing 8-inch and 6.5-inch bass and midrange drive units, plus a tweeter, on the front panel, and an enormous 11-inch passive radiator on the side.

A passive radiator is like an unpowered loudspeaker drive unit that works a bit like the port in a bass reflex enclosure, but better and without the 'chuff'.

And what's more, you can switch the SM9 into 'main' or 'nearfield' mode, via a 'Focus' switch.
In nearfield mode, the larger drive unit is disconnected, and the crossover frequency between the two small drive units shifted to give the 'small speaker' sound, appropriate to fine-tuning mixes for domestic consumption.

Whether or not this idea catches on we will have to wait and see. But in my mind it is a fascinating development. I wouldn't be surprised to see other manufacturers jumping onto this promising bandwagon.
P.S. Why do monitors always have to be black?

The X-ART tweeter of ADAM Audio loudspeakers (It ISN'T a ribbon!)

OXSA Labs | February 15, 2013

Q: When is a ribbon tweeter not a ribbon tweeter? A: When it's an eXtended Accelerating Ribbon Technology tweeter in an ADAM Audio loudspeaker.

ADAM Audio has set a problem for itself by describing its tweeter as a kind of ribbon tweeter. It seems, judging from Internet comments, that many people think it is a ribbon tweeter. But it isn't, quite.

What is normally regarded as a ribbon tweeter has a flat metallic diaphragm suspended in a magnetic field. Its advantage over the more usual dome tweeter is that the ribbon works as both radiating diaphragm and coil, with every part of the ribbon being driven directly. It can have a frequency response up to 100 kHz that only your pet bat can hear.

Adam Audio's tweeter, although it has the word 'ribbon' in its name, is more properly called an 'air motion transformer', as invented several decades ago by Dr. Oskar Heil.

The essential difference between this and a conventional drive unit is that the diaphragm of a conventional driver moves as fast as the air in front is required to move to create the desired acoustic wave. The air motion transformer can however move air four times faster than the speed of the diaphragm.

Adam Audio has incorporated this technology into its X-ART tweeters. The diaphragm is folded into a concertina shape that is squeezed by the incoming audio signal. Compression and expansion of the folds of the concertina force the air in and out much faster than the motion of the diaphragm itself.

The advantages of this are, according to ADAM Audio, that the tweeter has "unprecedented clarity and pristine transient reproduction" and also "avoids the typical breakup/distortion and subsequent dynamic limiting at higher frequencies of stiffer voice coil designs". A further bonus is that the X-ART's equivalent of a coil is in direct contact with the air and thus can be cooled more efficiently.

Also, by folding the diaphragm, it can cover a much larger area while remaining compact in terms of its aperture to the air. A smaller diaphragm aperture has a larger angle of dispersion, which is a desirable feature for a tweeter in many applications.

Whether this technology can be shown to be subjectively better than a conventional tweeter will be down to its ultimate acceptance in the pro-audio marketplace. Or not. It is however a fascinating development and, as always, we applaud technical progress.

Do plug-ins sound like the analog equipment they emulate?

OXSA Outbound | February 06, 2013

Everyone would like an 1176 or LA-2A compressor in their studio. But the originals cost so much that plug-in emulations are usually the only viable alternative. But do they sound the same?

A noted outboard manufacturer with a powerful software development team once ran a comparison test between hardware compressors and their software emulations. Even with technology that is now several years old, the results were impressive. It was VERY difficult to tell which recording was hardware and which was emulation. In a properly controlled test it is likely that most people, even experienced engineers, wouldn't have been able to tell the difference reliably.

So, according to this evidence, plug-ins can sound exactly like the hardware they emulate. But that doesn't tell the whole story...

Firstly, just because a software developer is able to make an interface that looks beautifully like a battle-hardened compressor from the 1960s doesn't mean that their emulation is as good. Some software emulations are, as mentioned above, almost exactly like the real thing. Some, unfortunately, are nothing like it.

Anyone buying a plug-in emulation of a compressor would need access to the real hardware to make a realistic comparison. And it's worth bearing in mind that real-life examples of the same compressor don't always sound the same. Circuit designs change over time; components differ; age affects different units in different ways.

Secondly, but equal in importance, is that is isn't just the steady-state settings that govern the subjective 'sound' of a compressor, or any other audio processor. It's what happens when you adjust the controls. How a unit reacts to user input is just as much part of the sound. Think of a compressor or an equalizer as a musical instrument and you'll get the picture.

But perhaps it doesn't matter how good an emulation is. Most software developers of repute offer trial versions of their products. If you download a plug-in and find that you can do great things with it, it doesn't matter how close an emulation it is to any analog hardware. And if you don't like it, then if it really is a 100% accurate emulation, you wouldn't have liked the original hardware anyway.

There's a certain joy in having nice equipment, just for its own sake. And a plug-in that looks good is surely more pleasant to use than one that doesn't. But ultimately if the sound of a plug-in pleases you, and more importantly the results you achieve please your client or market, then that's all that's needed.

How to compress a bass guitar that varies in level

OXSA Outbound | February 10, 2013

It's a rare bass guitar recording that doesn't vary in level to some extent. In most cases, this adds nothing to the musicality of the recording and just makes the track harder to mix.
If you have a bass guitar recording that varies in level, it is likely that it will be too loud at some points in the song, too quiet in others. There is no one position of the fader that seems correct all the way through.

Possible solutions

There are several ways to handle this problem (other than ignoring it and convincing yourself it is a product of the musicianship of the player)...

• Fader automation
• Clip-based gain
• Compression

Fader automation would be possible, but when the problem could exist on a note-by-note basis, i.e. quiet-loud-quiet etc, it could get very fiddly. The same applies to clip-based gain, which might involve splitting the track into dozens of separate segments, many as short as just one note. Compression might seem like an obvious solution, but it is important to consider the alternatives.

Identify what you want to achieve

Compression is often used because it just sounds nice, or to change the dynamic structure of individual notes (which is very applicable to bass in the appropriate context). But here we just want to get the level to be consistent, so that the track is easier to mix, and has a more solid foundation in the low-frequency region.
In this case therefore compression is only used to correct the varying levels of notes. Any other changes should be minimized.

Fix the big problems first

A quick look through the waveform display, zoomed to make individual notes visible, will show up any major problems - notes that are massively too loud or too quiet. These are best fixed using clip-based gain, or whatever similar technique your DAW provides. By doing this, you can preserve the sonic texture of these notes. Otherwise, a) the very loud notes would be extremely compressed, and b) you would end up compressing all but the quietest notes, so the bulk of the bass guitar track would be compressed when it really isn't necessary to do so.

The gain reduction meter is your friend

Since what we want to achieve is to smooth out the variations in level, the compressor should only kick in on louder notes. Most of the time it should be inactive. The gain reduction meter will show you this. Your aim should be that quite a lot of the time there is no compression at all. If the gain reduction meter shows compression all the way through, then you are compressing too much for the purpose of level correction.
If your compressor has a threshold control, then you will adjust the threshold and ratio to get a good control over the notes that are too loud in level. The gain make-up control can then be used to bring up the overall level, so that the quieter notes are raised up.
If your compressor doesn't have a threshold control, then you will balance the input control and ratio to achieve the same thing, always with an eye on the gain reduction meter.

The result

Since the object of the exercise was to control the level of the bass, the end result should not sound obviously compressed. It should just sound more consistent in level. This will almost certainly be easier to mix than it was before, and the finished mix should sound more solid and more confident.
Takeaway - When compressing to control dynamic range, the gain reduction meter should go all the way down to zero in the quiet sections of your original recording. Otherwise you're compressing too much and changing the texture of the instrument.

Why your studio door should not have a latch

OXSA Outbound | February 10, 2013

How do you keep your studio door closed? If it uses a latch then you may be slowing down your sessions. 

How many doors does your studio have? Possibly two, one between the control room and the recording area, and another directly out of the recording area for reasons of fire safety. (Bear in mind that if you carry out professional work in your studio, then in many jurisdictions a whole new book of fire safety regulations applies.)

Most doors in normal everyday life use a latch mechanism to keep them closed. So it wouldn't be unusual to see a similar mechanism in a studio door.

However, having a latch causes a certain amount of inconvenience. Whenever anyone uses the door, the latch makes a noise. In fact it makes two noises, one when the door is opened and another when it is closed. Yes you can open and close the door very carefully, but does every musician and singer who visits your studio have to be instructed in how to use a door? And will they remember?

This problem was solved way back in 1962 by the BBC (British Broadcasting Corporation - which probably operates more studios than any other single organization in the entire world). The solution was to having a spring closing mechanism, supplemented by a magnetic strip to provide an airtight acoustic seal. No latch is necessary.

Anyone working in a BBC studio can enter and leave the control room (which they traditionally call a
'cubicle') or recording area completely silently, without the risk of latching noises being broadcast to the nation.

This may be a comparatively small point in the whole universe of recording studio design and operation. But if people can come and go in complete silence, it will surely allow your sessions to run much more smoothly.

 By the way... The door in the photo - it's at the BBC!

Why do we monitor on moving coil loudspeakers?

Moving coil drive units are simple, efficient and generally work quite well. The drawback however is that the 'motor' - the magnet and the coil - is in the center of the diaphragm, and the diaphragm must be stiff so that vibrations travel quickly outwards from the center to the edges. This must happen without the diaphragm bending, otherwise the center could be moving outwards while the edges are still moving inwards as a result of the previous half-cycle of the waveform...

By David Mellor, Course Director of Audio Masterclass

Strictly speaking, the loudspeaker is the entire thing consisting of drive units, crossover and enclosure (cabinet). So here I will refer to the moving coil drive unit, which is a component of virtually ever loudspeaker you will ever hear.

The moving coil drive unit consists of a magnet, a coil of wire which when the signal is applied to it becomes an electromagnet, and a cone-shaped diaphragm that pushes against the air creating sound waves.

Moving coil drive units are simple, efficient and generally work quite well. The drawback however is that the 'motor' - the magnet and the coil - is in the center of the diaphragm, and the diaphragm must be stiff so that vibrations travel quickly outwards from the center to the edges. This must happen without the diaphragm bending, otherwise the center could be moving outwards while the edges are still moving inwards as a result of the previous half-cycle of the waveform. This effect is known as 'break up' because the diaphragm is not moving as a whole but various sections are moving independently of each other.

In the low frequency drive unit - the woofer - this causes audible distortion, which is greater at higher frequencies. It is a fact that distortion of this kind always exists in a moving coil drive unit. There has to be a compromise between the diaphragm being stiff and being light. Obviously if the diaphragm were made from 10 mm steel it would be perfectly rigid, but it would also be heavy and the drive unit would be inefficient.

Another kind of drive unit - the electrostatic - does not suffer from break up at all, hence distortion can be almost entirely absent. The flat diaphragm of an electrostatic drive unit is driven over its entire surface area, not just from the center, hence there is no need for it to be stiff - there is no way it can bend.

The trouble is that electrostatic drive units are expensive and inefficient, therefore electrostatic loudspeakers are sold only in very small quantities.

So even though an electrostatic loudspeaker, used as a studio monitor, would tell you the absolute truth about what's on your recording, it isn't representative of the real world. So we monitor on moving coil loudspeakers simply because they tell us what it's going to sound like for the listener.

Publication date: Tuesday November 30, 1999
Author: David Mellor, Course Director of Audio Masterclass

Why you should put the soloists behind the choir for recording

For the amateur recordist, or even the professional, this is a very common recording scenario. Choral singing is immensely popular and the opportunities for recording are plentiful.

In a concert, which for a classical choir will not be amplified unless it takes place outdoors, the solo singers are at the front of the stage, and the choir further back.
This is entirely logical because the audience wants to hear the soloists very clearly, and of course see them too.

But in a recording session - session as opposed to live recording - then you can position the singers wherever you want, within the limitation that they have to feel comfortable so they can give a good performance.

So you might indeed consider placing the soloists behind the choir. Why?
Consider this... Suppose you set individual mics for the soloists, then a stereo pair of mics to cover the choir. This will work fine, except that the solo mics are also pointing at the choir, so they will pick up the choir at a significantly high level.

This limits your options when mixing because when you raise the level of a soloist, then the portion of the choir behind them goes up in level too. And also consider that if the mics are positioned optimally for the soloists, then the sound they pick up from the choir probably won't be all that good.
But if the soloists are behind the choir, then the soloists' mics are pointing away from the choir, and the choir are singing in the opposite direction.

This will give you absolute flexibility in setting the levels of the soloists' mics.
You might say that the choir mics will now pick up the soloists. Well from experience, I can say that this won't be such a problem.

Remember that in a recording session, you as an engineer have a great say in where the musicians are positioned. Take every advantage of that.

By David Mellor, Course Director of Audio Masterclass

April 11, 2010

Speech is one of the most difficult sound sources to record well. Why is this so? What should be done to achieve a perfect recording?

How to record speech to radio and audio book standards

Speech is one of the most difficult sound sources to record well. Why is this so? What should be done to achieve a perfect recording?

Familiarity

The main reason why recording speech is difficult to do well is that we are so familiar with how it should sound. How many people do we talk to during the course of an average day? With every encounter we become increasingly familiar with the sound of the human voice.

Playback through loudspeakers

 Recorded speech cannot sound exactly like a real person. One reason is that moving coil loudspeakers (which are the kind in most common use) always sound a little bit 'loudspeakery'. Even a highly accurate electrostatic loudspeaker has different directional characteristics to the human voice.

Microphone distance

Another reason is that the distance at which microphones are commonly used for speech is much closer than we would normally listen to another person from. For speech, a microphone is almost always positioned much closer than one meter away from the mouth. One meter is about the minimum listening distance in a real-life social interaction, other than between intimate partners.

Background noise and ambience

The combination of the human ear and brain has a remarkable ability to filter out and ignore background noise and ambience. Microphones however do not have such abilities. Any background noise or ambience is permanently printed onto the recording, and heard very clearly on playback. Worse still, speech has silent gaps and pauses during which background noise and ambience can be heard. Music on the other hand tends to be more continuous and can obscure a certain amount of noise. Music also commonly benefits from ambience, where speech is almost always better recorded quite dry.

Popping and blasting

'P' and 'B' sounds cause the mouth to emit a short pulse of air. When a microphone is quite close to the mouth, this pulse can strike the diaphragm, causing an audible and unpleasant 'pop' in the signal. Similarly, other speech sounds can create blasts of air that distress the diaphragm, causing unpleasant sounds that would never be heard in a normal person-to-person conversation.

The simplest way to guard against popping and blasting is to move the microphone from the direct line of fire of the breath, but still point it at the mouth. So, for instance, if the person speaking holds their head level, and the microphone is raised to the height of the forehead and pointed down at the mouth, popping and blasting will hardly ever occur.

This sometimes looks a little odd, but this is because the conventional microphone position directly in front of the mouth, so commonly seen, is in fact wrong.

Another method, although it is not 100% effective is to use a pop shield consisting of an open mesh material stretched over a wire or plastic frame. A third method is to use a windshield, which is an open-cell foam material shaped such that it can be placed over the grille of the microphone (including any openings behind the diaphragm). Once again this is also somewhat less than 100% effective.

Another solution, although more costly, is to use a more highly directional microphone, such as the Sennheiser 416, which works on the interference tube principle. Because it has a very narrow pickup pattern, it can be used at a greater distance from the mouth where any air blasts will have dissipated before they can strike the diaphragm. The drawback here is that the sound quality of interference tube microphones is generally not as good as other designs.

Sound insulation and acoustic treatment

To get a really good, professional sound on speech, it is necessary to have a recording area with good sound insulation, and acoustic treatment. Sound insulation will limit the level of background noise coming from outside. Acoustic treatment will reduce the amount of ambience within the room. Clearly it is vital not to have any sources of noise within the room.

It is quite common to use small booths for speech. However, it is difficult to acoustically treat such a small space without it sounding audibly 'boxy'. A larger, room-sized space, with acoustic treatment, can often sound better although clearly it will cost more to set up properly.

Choice of microphone

The most accurate type of microphone is the small-diaphragm capacitor microphone, particularly models designed to have an omnidirectional pickup pattern. This could be recommended to anyone who is keen to achieve the most natural sound possible.

However it is more normal to try and achieve the 'expected' sound, rather than the most natural sound. Large-diaphragm capacitor microphones are often used. They are not as natural-sounding, but they have the sound people expect to hear, so they are popular for speech.

Vacuum-tube capacitor microphones (sometimes called 'condenser microphones' - condenser is the old word for capacitor) can work really well for singing as they create a sound that is perceived to be 'warm'. However, they can sound a bit buzzy on speech, and the sound can be tiring to listen to over a long period. The same applies to vacuum-tube microphone preamplifiers. Having said that, the vacuum tube sound might be useful for radio, where a sense of exitement is often required.

Summary

Although it would be nice to be able to provide a summary in one sentence, in reality all of the points explained above require attention to achieve the best possible sound on speech. There is however one final point, which is that being easily satisfied is not the route to success. Only when you have achieved the ability to make recordings of speech that are fully equal to the best audio book standard can you say that you have truly mastered this difficult skill.

Publication date: Monday December 24, 2012
Author: David Mellor, Course Director of Audio Masterclass

Everything you need to know to set up a home recording studio

What do you need to know to set up a home recording studio? How can you make your studio capable of professional results? How can you avoid expensive mistakes?

By David Mellor, Course Director of Audio Masterclass

Everything you need to know to set up a home recording studio...

'Everything', in this sense, is a bit like 'infinity' - it's difficult to gauge its extent. But Record-Producer.com is here to help. Undoubtedly you would like your home recording studio to be capable of professional results. But the very fact that the studio is in your home adds further difficulties.

Here are just some of the issues facing would-be home recording studio owners...

• Soundproofing. If your studio disturbs others, it is not going to be a practical place to work in. Also, if sound from outside disturb your monitoring, or get onto your recording, then you are not going to achieve successful results.
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• Acoustics. The smaller a room is, the harder it is to achieve good acoustics. You need good acoustics for two reasons - firstly so that sounds you capture with a microphone sound good. Secondly so that your mix will sound great wherever it is played.
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• Traffic. This doesn't affect everyone, but if you have ever tried to record a rock band at home, you will know what impact the simple coming and going of people and equipment can have. Also, imagine if you had an expensive session booked with a pro singer. How would it be affected if other people were coming and going around the house? One of the key features of successful commercial studios is that they can handle traffic of vehicles, people and equipment. And once in the studio, they are a calm place in which to work.
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• Interruptions. Sessions in successful commercial recording studios never get interrupted for any reason - there's a story about Bono from U2 not wanting to be disturbed by a persistent caller. It turned out to be the Pope! Achieving a non-interrupted state in a home recording studio is much more difficult.
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• Equipment. Home studio owners can rarely afford the same equipment as commercial recording studios, although items such as microphones can sometimes be hired in, depending on where you live. Also, people who work around commercial studios gain insider knowledge - they see people coming and going with different equipment and get to learn what works, what doesn't and which equipment is good for which purpose. Manufacturers schmooze them too and let them borrow stuff to try out.
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• Practicality. Commercial studios spend up to 10% of their budget on installation and wiring. How many home recording studio owners spend as much? The result of a neat installation is a studio that is slick and versatile, saving vast amounts of time and energy.
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• Cost. The cost of providing all of the above is significant. Any home recording studio owner has to balance out how much income they can make from their studio, or how much they are willing to spend on their hobby.

You may have realized that to cover everything you need to know to set up a home recording studio is too large a concept to fit into a single feature.

But Record-Producer.com would like your input. What are your concerns regarding setting up a studio? What questions would you like to ask?

Record-Producer.com plans a major exploration of this topic, with much of the information being sourced from people who are successful in the industry.

Please send your one most burning question about setting up a home recording studio to david.mellor@record-producer.com

The ten Record-Producer.com visitors who send in the most searching and practical questions will receive access to the eventual fruits of our labors free of charge!Submissions for this offer have now closed.

 Publication date: Saturday November 05, 2005
Author: David Mellor, Course Director of Audio Masterclass 

What is a condenser microphone?

Description of the condenser microphone and explanation of its relationship to the capacitor microphone

 

Microphones can be separated into two basic types - those that work on magnetic principles and those that work on electrostatic principles. The former are known as dynamic microphones, the latter as capacitor - or condenser - microphones.

Internal view of the Neumann BCM 104

A capacitor is a device that stores energy by separating positive and negative electrical charges. 'Condenser' is a old word for 'capacitor', so a condenser is a capacitor, hence a condenser microphone is a capacitor microphone. In common usage however, it is not uncommon to call a large-diaphragm, vintage vacuum tube microphone (or its modern retro-designed replica) a condenser microphone. It is however rare for a small-diaphragm capacitor microphone to be referred to as a condenser microphone.

In technical terms...

The capacitor in a capacitor microphone (or condenser microphone) consists of the diaphragm, which picks up sound vibrations, and a fixed back plate separated by an air gap. A 48-volt 'phantom' power supply (or a power supply unit dedicated to the microphone) separates positive and negative charge between the diaphragm and the back plate, thus charging the capacitor.

Sound vibrations picked up by the diaphragm cause the distance between the diaphragm and the back plate to change in proportion to the movement of the air molecules in contact with the diaphragm. Since the value of the capacitance of the diaphragm/back plate system is related to the distance between the diaphragm and back plate, then the capacitance of the system will change in proportion to the incoming sound wave. Also, since the voltage between the diaphragm and back plate changes in proportion (actually inverse proportion) to the value of capacitance, the system will produce a voltage in response to the incoming sound wave.

In practical terms...

The essential difference between the capacitor microphone and the dynamic microphone is that the diaphragm of a dynamic microphone has to be attached to a coil of wire that moves within the field of a magnet (or it could alternatively, in principle, be attached to a magnet, the field of which passes through a coil) in order for the microphone to function. The diaphragm of the capacitor mic has no such burden and is therefore much more free to move in response to the incoming sound wave.
A capacitor microphone, or condenser microphone, therefore has a sound texture that in most cases has a higher degree of clarity than a dynamic microphone. Capacitor microphones will pick up the overtones of metallic percussion instruments very much more clearly than a dynamic microphone would. They are also noticeably demonstrate crystal-clarity on most other acoustic instruments. The differences between the capacitor microphone and dynamic microphone are less marked on membrane percussion instruments (drums) and electric instruments such as the electric guitar.
One small additional point is that because a dynamic microphone works on magnetic principles, it can pick up hum through magnetic coupling when positioned close to the transformer of an electric guitar combination amplifier-loudspeaker cabinet. Moving the microphone away from the transformer will usually reduce the hum to near-inaudibility. However, a capacitor microphone works on electrostatic principles and is not sensitive to this source of hum at all.

Phantom power

Capacitor microphones need electrical power to function, firstly to charge the diaphragm and back plate, secondly to power the essential internal amplifier. The signal from the diaphragm is extremely weak and has to be amplified within the microphone, before sending it down the cable to the preamplifier, audio interface or mixing console.

Some types of capacitor microphone, known as electret, or back electret, have a permanent electric charge, but they still require power for the amplifier. Many electret microphones can however operate from an internal battery.

Power is normally supplied directly from the microphone preamplifier, audio interface or mixing console. Power in this form is known as phantom power, because no power supply is visible.

However, vacuum-tube microphones, of the type that may archaically be referred to as condenser microphones, require higher voltages and thus normally have their own power supply.

Why distortion techniques MUST be part of your recording vocabulary

 Do you hate over-mastered, distorted recordings? That might just be why your work isn't selling.

By David Mellor, Course Director of Audio Masterclass

Please take a quick read of 'The Worst-Sounding Hit Record Ever' and listen to the audio. You might care to read the comments too, many of which are insightful.

The song is Firework by Katy Perry and clearly it is massively distorted. I had heard the song many times on TV, radio and my car stereo (courtesy of the car's other occupants) and I had always enjoyed it, although the strings - to my mind - are a little harsh.

But I didn't notice the distortion until I listened to the song from CD, on my favorite pair of extremely hi-fi headphones. Then I listened on my hi-fi speakers and again on my near-field monitors.

Distortion. Intense distortion.

At first I blamed the mastering process. Generally I find the effects of mastering for loudness merely unappealing. But this was appalling.

But I listened more closely. My feeling is that it is the kick drum sample that is distorted, and the whole mix is compressed to give a pumping effect around the kick.

I don't like it. I don't like it at all. I am sure that the song could sound so much better given the benefit of clean recording and mastering.

But you can't argue with...

Quadruple platinum.

A quadruple platinum award means Firework has sold over 4 million copies, and that's only in the United States. The sales figures in other major music markets are equally impressive.

It might be possible to speculate that, without the distortion, Firework might have achieved quintuple platinum, but with my hand on my heart I just don't feel that is so. There is something about the entirety of the artist, the song, the recording, the mastering (and the marketing) that put together has given Firework's creators their deserved success.

This brings my to my point...

My point is...

If this extreme distortion, that many listeners find unpleasant, is a component of such a successful record, then it has to become part of your recording technique.

Suppose for instance you make a recording, and the client says it isn't exciting enough. (Don't expect the client to use technical language, or be specific about what they want.) If you can't bring yourself to turn up the warmth on the kick to thermonuclear levels, then the client will find someone else who can.

And if it's not the kick, then it could be some other instrument, perhaps even the vocal. Mastering has for a long time been the char-grill of the recording process, so it might happen there.

So, I would say that anyone who doesn't like this kind of sound has three options...

• Lose clients who want the sound of heavy distortion
• Do it reluctantly, when pressed

Or...

• Embrace extreme distortion, make it your own, make it artistic and make it good!

So although the ability to make a clean, undistorted recording of any types and combinations of instruments and voices is absolutely essential, so is the ability to manipulate audio in any way that pleases the market. A knowledge of distortion techniques and their artistic application should be part of any engineer's skill package.

Publication date: Monday September 05, 2011
Author: David Mellor, Course Director of Audio Masterclass

Does limiting make your audio clip and distort?

 Push your levels too high and you will break the bounds of your system's capability. But will a limiter make things any better, or simply act as another source of distortion?

By David Mellor, Course Director of Audio Masterclass

Here's an experiment... Take a section of cleanly recorded audio that is well below the level at which the red lights come on.

Now raise the level, raise it some more, then raise it some more. Your system almost certainly does have the ability to raise the level by as much as you want, although you may have to rummage among your plug-ins to find it.

But by whatever means, raise the level until the red light starts to flicker. Listen carefully for changes in the sound. Continue raising the level until the red light is solidly on and there is absolutely no doubt that your sound is distorted.

What you are listening to is the sound of clipping. Any digital audio system has an upper level beyond which the signal just cannot go. We call this 0 dBFS - zero decibels with respect to full scale.

If you try to raise the signal beyond this level, the tips will be clipped, causing distortion.

By the way, this test should be done at a low monitoring level as it will produce a lot of high frequency energy that could potentially damage your tweeters.

To avoid clipping, you might consider using a limiter. Set correctly, this will prevent the signal level reaching 0 dBFS, so no clipping can occur.

Well, that is the theory. In practice it depends on making the correct settings.

Think of it like this... imagine that you could control the level by hand very quickly so that the signal never went above 0 dBFS. Every time a loud section comes along, you lower the level so that the signal stays within bounds. When it goes quiet again, you can raise the level back up again.

That's exactly what a limiter does. But you can make a limiter respond very much more quickly.

You might therefore be tempted to set a really short 'attack' time - that's the time it takes for the limiter to respond; it has nothing to do with making your signal sound more attacking subjectively.

But when the speed of response approaches 50 milliseconds or less, something interesting happens. The limiter is now fast enough to respond to single cycles of the waveform, at low frequency.

At this point, the limiter has the capability of changing the shape of single cycles of the waveform.

There is a word for that - distortion.

So you have a dilemma. Set the attack time too long and fast peaks may get through and clip. But set it too short and the limiter itself causes distortion. It isn't clipping, but it can sound as bad.

The moral is to be very careful when setting an attack time of less than around 100 milliseconds, to be on the safe side. Listen very carefully, watch for red lights, and perhaps limit a little lower than 0 dBFS, then normalize afterwards to bring the level up to peak.

Publication date: Tuesday May 26, 2009
Author: David Mellor, Course Director of Audio Masterclass

Flute, clarinet and other woodwind - how much key clicking is too much?

 Try close-miking a flute or any other woodwind instrument and you will pick up a whole load of key clicking. What can you do? How much is too much?

By David Mellor, Course Director of Audio Masterclass

We received a recording of solo flute at RP Towers recently. The sender was enquiring whether the amount of key click noise was too much.

Originally these instruments had no keys, and therefore no key click noise. That however was more than a hundred years before the invention of recording so there was no-one around to appreciate the benefit.

But wily inventors added first one key, then another, all in the interests of player convenience and extending the range of the instrument. In modern instruments, all or nearly all notes are produced using keys, rather than stopping the holes directly with the fingers.

Inevitably this causes a mechanical noise that is easily picked up by sensitive microphones. So what can you do?

The first thing you can do as a recording engineer is consider where the normal listening position for a woodwind instrument would be. Would it be ten meters away or more if the instrument were in an orchestra? Would it be two or three meters if you were sitting in the front row in a small recital? Would it be a couple of centimeters from the instrument if you wanted to get the same experience as a close mic?

For some reason that acoustic science finds it difficult to explain, key click noise seems to disappear with distance. You would almost never become aware of it in an orchestral concert.

But the close mic sound is very attractive. If you must have that sound, what can you do about the key clicks?

The first thing to do is have the instrument serviced. Players tend not to notice the gradual degradation in performance of an instrument between services. They hear the key clicks every time they practice and cease to notice them.

This might not be practical and you have to deal with the instrument as it is.

You could try asking the player not to click so much. This will probably produce no noticeable benefit and the performance will suffer because the player is now distracted.

One thing you can do successfully is move the microphone towards the source of musical sound and away from the mechanism causing the clicks.

Woodwind instruments produce sound from three places - the mouthpiece or reed, the opening at the end (the bell in the clarinet and oboe), and from any holes in the instrument that are not covered. This last source changes from note to note, and you will hear this clearly if you listen from close range.

So if you move the microphone closer to the mouthpiece or reed, or to the opening at the end, you will increase the amount of musical sound slightly and decrease the level of the clicks. You can also experiment moving the microphone around the instrument to the side that has fewer mechanical parts.

Two points...

1. Don't expect miracles. Expect a small improvement at best.
2. Moving the mic will change the sound quality, perhaps not for the better.

There is however another solution that trumps all the rest, and that is to accept these noises as part of the texture of the instrument.

All acoustic instruments have texture and this is something that gives an instrument its character.

This can come as a surprise if you have learnt recording on synthesized and sampled instruments. But texture can be a large part of the 'soul' of a recording and something to be welcomed, in reasonable quantities.

We would like to hear about your experiences with the 'texture' of acoustic instruments.

What non-musical sounds do you like? And what drives you crazy?

If you can e-mail us an example, we would love to hear it and perhaps publish it on the site - newsletter@recordproducer.com

Publication date: Friday February 26, 2010
Author: David Mellor, Course Director of Audio Masterclass

Would YOU pay $18 for a Waves plug-in

 If an offer seems too good to be true, then it's probably a scam. Would you fall for this one?

By David Mellor, Course Director of Audio Masterclass

If I had a dollar, or better still a UK pound, for every time I've won the Nigerian National Lottery, then I would have as much money as if I had won a real lottery. Weight loss, hair loss, loss of sexual function - there's a cheap cure for just about every medical condition there is. According to my e-mail inbox!

And then there are the phishing attacks. There's not a day goes by without some bank that I've never dealt with inviting me to 'update my account details'. Occasionally I get one from what seems to be my bank, but of course it isn't. It's just someone who wants to trick me into giving them my password.

But I was shocked the other day to receive a phishing attack email in the name of Waves, the renowned developers of high quality, but expensive, plug-ins. Apparently I could buy Waves plug-ins for as little as $18. Wow, if that were true it would be amazing!

What shocked me was that I thought that phishing attacks only involved banks and other institutions where account information could be valuable to a criminal. But Waves? How could anyone make much money from knowing people's Waves account details? Well I'm not a criminal mastermind so I wouldn't know. (We don't need ideas, thank you.)

(It did occur to me that it might be fake software, but I never seem to get e-mails for any kind of fake software, so I guess there isn't much of a profit in it.)

What worried me further was the thought that this kind of attack could spread through other audio manufacturers and software developers. Since this is an area I need to keep in touch with, it would make my life a nightmare.

Since I am hardened, through experience, to phishing attacks I know what to look for. One key piece of information is shown in the status bar below the message when I hover on a link - does the status bar show the correct website? If it doesn't, then to me that could suggest that the email is a phishing attack.

And lo and behold... the links in the Waves email go to trailer.web-view.net

So I typed in trailer.web-view.net into my browser, which got me, "403 - Forbidden: Access is denied". This was not looking good and I felt even more strongly that this is a scam, and clicking any link would invite trouble of all kinds.

Except...

It isn't a scam at all. It is a genuine offer from Waves. I didn't click on the link but instead typed www.waves.com into my browser's address bar, which is good anti-phishing practice.

You can indeed buy the Renaissance EQ for $38, and AudioTrack for a mere $18!

Clearly someone has made a big mistake here because the e-mail showed every sign of being a phishing attempt. But if you want to grab yourself a bargain, now is the time to type www.waves.com into your browser.

P.S. If anyone has an interesting e-mail scam to report, I'm sure we would all benefit from hearing of it. I'm afraid I did once fall for 'Someone has left negative feedback for you on eBay'. Fortunately I realized very quickly and changed my password. Could have been nasty though.

Publication date: Friday April 08, 2011
Author: David Mellor, Course Director of Audio Masterclass

Q: How can I record pro-sounding drums in a small room?

 Is it possible to record a high-quality sounding drum track in a small space? Or am I just wasting my time recording in a room with a low ceiling? I am using high quality microphones (worth a few thousand collectively), a good drum-set with very high end Zildjian cymbals and I know how to properly mix a kit. Nor am I ignorant of phase problems and how to deal with them. Nonetheless I have never been able to achieve a drum sound in a small room that sounded any better than just 'good'. Is it possible? What should I do?

By David Mellor, Course Director of Audio Masterclass

From time to time we receive a question that is really difficult to answer, and this is one of them.

It would be nice if we could give a magical solution that would allow drums to be recorded to a professional standard in a small room, but the best we can do is offer ideas that will help you progress in a professional direction, if not get all the way there.

The first thing to realize is that the sound of professionally-recorded drums is the sound of professionally-recorded drums in a large room, not a small room. At least five or six meters in the smaller horizontal direction.

And by 'large room' we also mean 'large room with a high ceiling', preferably four meters or more.

Such a room is way beyond what you would normally find in anyone's house or apartment.

It also helps for the large room to have good acoustics. And of course pro studios are professionally acoustically designed, so this is what you would expect.

'Good acoustics' means having a controlled reverberation. The reverb field should be diffused with many weak reflections rather than few strong ones. There should not be too much reverb, and its frequency response should be smooth. I say 'smooth' rather than 'flat' because it is normal to have a longer reverberation time at low frequencies rather than high, because low frequencies are less easily absorbed. This is what we find in rooms all the time, and the human ear tends to like what it expects.

Small rooms tend to show the effects of standing waves too, where certain frequencies are highly emphasized compared to others.

What we consider to be good acoustics for recording also implies that the direct sound and reverberation are separated somewhat in time. So imagine a single drum strike. In a large room the direct sound will travel several meters to the nearest surface, other than the floor, then reflect. In a small room, it will travel over a much shorter distance and the reverb will build up much more quickly.

So the problems with normal domestic rooms are these...

• A few strong reflections rather than many weak ones


• Uncontrolled frequency response of the reverberation


• Standing waves


• Inadequate separation in time between the direct and reverberant sound

Although it will be impossible to turn a small room acoustically into a large one, you might consider at least making some improvements.

So you could use acoustic treatment to fix any standing wave problems. Normally this is done with tuned panel absorbers or Helmholtz resonators.

You could control and diffuse the reverb at the same time with suitable acoustic treatment. Oddly enough, the installation of bookshelves (with books!) can work well since they are partially absorbent and have irregular surfaces that break up what reflections that remain.

Controlling the frequency response of the reverberation normally implies giving special attention to low frequencies, since these are the most difficult to absorb. If only porous absorption is used, then only high-mid and high frequencies will be absorbed leading to an unpleasant imbalance. Panel absorbers are used for the lows because they take up less space than porous absorbers and can be tuned to the required range of frequencies.

Having done all that, what remains is that your room is small, and the reflections occur within a short time of the direct sound. You can never prevent that, but what you can consider is to make the room less reverberant altogether, which means having more absorption. Since the problem is not in the drums or cymbals but in the reverberation, then it makes sense to eliminate what is causing the problem. Perhaps not completely, but enough to make a difference. Since the quality of plug-in reverbs is excellent these days, you can always add any ambience you consider to be missing.

If you consider all of the above carefully, then you will see that making improvements is going to take an awful lot of hard work.

There is an old English proverb... "If at first you don't succeed, try, try again."

A more useful version adds at the end, "And if you have tried and tried again, it's really time to give up and find something more practical!"

It may indeed be more practical to find somewhere else to record your drums, or hire a professional studio to record drum tracks.

Publication date: Saturday December 18, 2010
Author: David Mellor, Course Director of Audio Masterclass

Q: How can I edit a song so that it is shorter?

 A RecordProducer.com reader has a song that is too long. How can he edit it so that it is shorter?

By David Mellor, Course Director of Audio Masterclass

A RecordProducer.com has a little problem...

"I would like some advice on editing. I have to cut a song from 7:45 to 4:00 because it is too long. Where should I start the editing process?"

Too long for what?

My first question here would be, "Too long for what?" If this is a song for commercial release then it is certainly too long. If you are going to write a song longer than Bohemian Rhapsody then it would really have to be something rather special.

If this is the purpose of the editing, then you need to think about what is it about the song that will make people buy it? All you have to do then is cut out the rest.

For instance, it might be that some of the duration is taken up by chorus repeats at the end. You could have the chorus once at the end and fade quickly at the start of the first repeat.

Some of the duration might be taken up by an instrumental solo. It's a rare solo that is the 'hook' of the song and the reason for purchase. Cut it out, throw it on the studio floor and trample it to death.

Some songs are burdened by too much introduction. Perhaps you can start straight in with the verse, or at least cut the introduction down.

If after all of these cuts the song is still too long, then you're going to have to lose a verse or two. Choose the verse or verses that say the least; that add the least value to a potential purchase.

Cutting for TV

It may be however that the song is to be used as part of a soundtrack for some kind of TV usage. For a commercial, trailer, program intro or drama soundtrack. In this case you have much more freedom. You should identify exactly what it is that makes this song unique; the reason why the producer chose it. You can now cut out pretty much everything else, and perhaps even repeat the good bits.

In this situation it is often not a good idea to try and keep a bit of each section of the song. For example, there is little point in keeping four bars of the middle eight. Just dump the whole thing.

Recomposition

However you approach editing the song, please don't think that editing is a mechanical process, or something to just 'get out of the way'. Think of editing as a process of 'recomposition'. The end result should sound, to a new listener, like the song was originally written that way. And if your edit is really good, it might be even better than the original recording!

Publication date: Friday April 22, 2011
Author: David Mellor, Course Director of Audio Masterclass