Q: Why do I hear echoes in a PA system?

 I recently saw a concert in a circular stadium. I could hear the sound come from different directions at different times. Why was that? Is it unavoidable?

By David Mellor, Course Director of Audio Masterclass

I remember myself going to an open-air concert where the same thing happened, and I know exactly why.

There were I think six or eight loudspeakers positioned at the edge of the audience area in a full circle, pointing inwards.

This might seem to concentrate the sound into the circle and thus be very efficient. Remember that sound is easily lost outdoors so you don't want to waste any power on the birds in the trees.

However this arrangement only works for people at the very center of the circle.

Anywhere else, you will hear the loudspeaker closest to you loudest. Then you will hear all of the other loudspeakers, less loud according to their distance, and delayed also according to their distance.

Sound travels around 30 cm (1 foot) per millisecond, so you only need a distance of 15 meters (50 feet) or so before the delay becomes noticeable. Any more than that and the echo becomes distinctly irritating.

The problem on this occasion was so bad that several people went up to the mix position to complain. In my experience, this is extremely unusual. However once the show had started there was nothing that the sound crew could have done.

The solution to this problem is that for a show 'in the round' the loudspeakers should be in the center, pointing out. In this way, no-one will hear sound from more than two loudspeakers (if they are exactly on the mid line between two cabinets) and since both of those loudspeakers are at the same distance, there will be no delay. Most people will hear sound from one loudspeaker only.

For more information, look up 'center cluster', which is a loudspeaker array that uses exactly this concept.

In a large stadium it may not be practical to have a cluster exactly at the center. In this case the loudspeakers should be in a circle entirely inside of the audience, pointing out. As much as possible each audience member should hear the output of only one speaker.


David Mellor
Course Director, Audio Masterclass

 Publication date: Thursday September 09, 2010
Author: David Mellor, Course Director of Audio Masterclass

Creating Alternate Arrangements with the Sound Forge Pro Regions List and Play List

What difference does an instrument or vocal make if you can't hear it?

 "Push it down so you can only just hear it, then push it down 3 decibels more". In what circumstances could this be good advice?

By David Mellor, Course Director of Audio Masterclass

I was reading the current issue of my favorite recording mag when this particular phrase sprung into vision... "Take the bass down so that you can just hear it, and then once you've done that take it down by a further 3 dB". You'll have to read the mag to see the original context, but it seems to me like an interesting idea for comment.

"Set the level so that you can just hear it" is a phrase I find myself saying quite a lot as a piece of advice. Not about the bass however, but about reverb. I find that this point is a good benchmark for reverb. Of course the requirements of the track might demand that the reverb is clearly audible. But would there be any value in having the reverb at a lower level, i.e. at a level where you can't hear it?

What does 'just hear' mean?

There are two ways of interpreting 'just hear it'. One is that when you listen to the mix in its entirety, you can just hear the instrument or effect in question. The other is that you can't actually pick out the sound in the mix, but when you switch it out you can hear the difference.

Sounds that you might want to 'just hear' include reverb, pad instruments and - yes - bass. With these instruments and possibly others it is well worth experimenting with the 'just hear' point, in both senses that I have described. You might end up choosing 'clearly audible' as the better option, but trying things out can help you learn things about the music you are mixing that you wouldn't pick up on by mere rational thought.

It's louder than you think

Another interesting experiment is to set an instrument at the lower 'just hear' point, where you really can only just hear it. Now solo it. It's louder than you thought, isn't it? How could you ever miss it in the mix? The human ear works in strange ways.

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

Quick Tip VIdeo: Audio Restoration in Sound Forge Audio Studio

Is there such a thing as maximum loudness? What if I want to go higher than that?

 As Roger Waters of Pink Floyd once said, "It doesn't have to be loud. We just like it loud." Well, some people would like it even louder. Is this possible?

By David Mellor, Course Director of Audio Masterclass

If you have a powerful enough amplifier and big enough speakers, there's no limit to how loud you can go. Well, I suppose when the air starts to ionize, that might be a problem. But other than that you can go as loud as you like, or as health and safety regulations permit.

But what about inside your digital audio workstation? How high can the level go?

Well the common answer to that question is 0 dBFS. 'dB' obviously stands for decibels; 'FS' stands for 'full scale'.

0 dBFS means zero decibels below full scale; in other words FULL SCALE - it's as high as the signal level can go.

But that's not entirely true...

Suppose you have recorded a signal and you optimized the gain of your preamp so perfectly that it peaks at exactly 0 dBFS. Surely that's as loud as it can go?

OK, let's try a test. Set the channel fader to 0 dB (which means that it doesn't change the signal level) and the master fader also to 0 dB.

Now try pushing the fader up on the channel. Push it up 10 dB. You will notice now that the meters by the master fader are now flashing bright red. That ought to tell you that something is wrong.

Indeed it is a warning. If you now bounce what you have done to disk, the output file will sound terribly distorted on playback. That is because you have tried to go beyond 0 dBFS. And you have failed because you can't go beyond 0 dBFS in the output file. The result is horrible-sounding clipping where the peaks of the waveform have been chopped off.

But go back to your master fader and lower it by 10 dB, or maybe a tiny smidge more just to be on the safe side. The red lights have gone out, haven't they?

As you listen to your signal, it sounds clean. Bounce it to disk and play back the output file. It still sounds clean, doesn't it.

So in the channel you have pushed the signal level up beyond 0 dBFS, which was supposed to be full scale. But since you lowered it again with the master fader you got away with it.

So what's happening?

What is happening is that 0 dBFS refers to full scale in the output file, not in the DAW. In fact, your DAW is probably capable of handling signals up to +30 dBFS, as long as you don't try to output them that high but bring them down again with the master fader.

That extra 30 dB is there as a safety margin, or 'headroom' as we like to call it. The point of having headroom is that you don't intend to use it, but it will get you out of trouble if you need it to.
30 dB is in fact loads of headroom. So there isn't any need for accidental clipping in the output file. Just make sure that the red lights in the meter by the master fader remain entirely off for the full duration of your track.

Over to you... let's hear your red light experiences (!)

Publication date: Friday January 22, 2010
Author: David Mellor, Course Director of Audio Masterclass

Korg Kronos Music Workstation -- The Players Speak Out

"How can I become a professional mix engineer?"

 An RP reader who is already a pro recording engineer wants to become a (highly paid) mix engineer. How can he achieve this?

By David Mellor, Course Director of Audio Masterclass

A question from an RP reader...

I have been working as a recording engineer for a few years. How can I take that big step to be a professional mixing engineer?

One thing that people working at a high level in pro audio have in common is that they don't tend to have that much in common!

In other words, everyone's experience is different, and what was the turning point in someone's career could be totally irrelevant to someone else.

In general however, we can start by saying that anyone who is already working as a recording engineer, and earning their living at it, is already doing pretty well. Congratulations on that.

But being a recording engineer isn't necessarily the most well-paid job in the world. Being a super-star mix engineer however can really bring in the dollars.

You could ask the question why is it that mix engineers are so well paid?

One reason is that record labels never really know why a record is a hit, particularly when it's a new artist.

If a new artist has a hit, was it the artist, the song, the producer, the hairdo, the dance moves... or the mix engineer?

Yes, the mix engineer is considered.

Unfortunately the recording engineer doesn't always seem to get this same consideration. He is probably hidden under the producer's shadow, rightly or wrongly.

So if a mix engineer has worked on a hit record, he or she is part of the winning formula.
You don't change a winning formula.

And of course when the mix engineer has a hit, his or her manager will want to up the rates for the next job.

So if you are a recording engineer already, how do you 'upgrade' to being a mix engineer?

As I said, everyone's situation is different. But one common factor is that to get work in any area of audio, you already have to have done it successfully.

So no-one will give you work as an assistant engineer until you have already worked as an assistant engineer.

No-one will give you work as an engineer until you have engineered a track that gets released.

And certainly no-one will let you sit in the producer's chair until you are a successful producer!

So it sounds impossible, except that everyone has to get started somewhere. But how?

The answer is that people often get started by chance.
One day a studio's assistant engineer doesn't turn up. So the guy who has previously been cleaning up gets to put mics on stands.

One day the engineer's car breaks down, so the assistant engineer has to take over for a while.

One day the producer has flu, so the engineer gets to offer a few of his ideas.
Get the picture?

With a mix engineer, the situation is different. You're not going to get a big-paying job by accident.
But you do have to create that first big-selling mix that shows you can do it.

And how do you do that?

Answer - by taking every opportunity you can to mix, and get people to listen to what you have done.

So for example, if you're working on a session and the producer decides that the recording phase is complete, ask if you can have a go at mixing it, in your own time (like the middle of the night).

If you think you have done an amazing job, and be harshly self critical, get the producer to listen to it.

If he likes it, he'll take it away with him.

And...

Perhaps no-one, not even a seasoned mix engineer can make a mix that's better than yours. So your mix is the one that gets released.

Hey - you're a mix engineer!

OK, so I'm telling a story here. But it isn't that far from reality.

So all you highly-paid mix engineers out there - how did you get the gig?
In fact, if you're a successful musician, writer, engineer or producer we would love to know what happened to get your career really off the ground.

Tell us below...

Publication date: Friday February 05, 2010
Author: David Mellor, Course Director of Audio Masterclass

Chords in Finale 2010

How much power do you need to fill a venue with sound?

 Small venues need small amplifiers. Large venues need racks and racks of amps. But how do you know how much power is enough?

By David Mellor, Course Director of Audio Masterclass

Working in live sound can encompass venues of all sizes from a small and intimate bar all the way up to the biggest sports stadium seating 100,000 or more.

So is there a way to calculate how much power you need to fill the venue with sound?

Firstly, we need to be a little more precise about this. "Fill the venue with sound" needs quantifying. One way to do this would be to say that the system should be capable of a level of 100 dB SPL over the whole seating area of the venue. Or a map of levels could be produced that allows some areas to be louder than others.

So now let's consider the variables in the system...

Let's say that you have 1000 watts of power available. This will be supplied to loudspeakers that have a certain efficiency rating. If a loudspeaker can convert 1000 watts of electrical power to 20 watts of sound power, it is doing pretty well at 2% efficiency. The rest of the energy is wasted as heat.

So now we have 20 watts of sound power to play with. All loudspeakers focus their output to a greater or lesser extent. The more focused the output, the higher the level in the direction of 'throw'.

Now for the difficult part - reflections from the room...

Any room (in acoustics, 'room' means an enclosed space of any size) holds and contains sound energy to an extent. A reverberant room will allow sound energy to bounce back and forth. A well-damped room will absorb sound energy. The reverberant room will be louder for the same sound power input because you get the opportunity to hear the same sound several times as it bounces back and forth.

Plainly, we are talking about some difficult calculations here. But there is an alternative... good old-fashioned 'rule of thumb'!

I don't think you will find a better rule of thumb than that provided by the experts at Crown Audio who collectively probably have at least as much and possibly more experience than anyone else in providing amplification for venues of varying sizes, and for different purposes.

So here it is...

• Nearfield monitoring: 25 W for 85 dB SPL average (with 15 dB peaks), 250 W for 95 dB SPL average (with 15 dB peaks)
• Home stereo: 150 W for 85 dB SPL average (with 15 dB peaks), 1,500 W for 95 dB SPL average (with 15 dB peaks)
• Folk music in a coffee shop with 50 seats: 25 to 250 W
• Folk music in a medium-size auditorium, club or house of worship with 150 to 250 seats: 95 to 250 W
• Folk music at a small outdoor festival (50 feet from speaker to audience): 250 W
• Pop or jazz music in a medium-size auditorium. club or house of worship with 150 to 250 seats: 250 to 750 W
• Pop or jazz music in a 2000-seat concert hall: 400 to 1,200 W
• Rock music in a medium-size auditorium, club or house of worship with 150 to 250 seats: At least 1,500 W
• Rock music at a small outdoor festival (50 feet from speaker to audience): At least 1,000 to 3,000 W
• Rock or heavy metal music in a stadium, arena or amphitheater (100 to 300 feet from speaker to audience): At least 4,000 to 15,000 W

Crown also provide a calculator, but this does not account for the directional properties of the loudspeakers, nor for reflections in the room. Still, it makes for a good starting point. (You could bear in mind that they want to sell you more amplifiers!)

Opinions from live sound practitioners would be welcome.

 

Publication date: Wednesday January 12, 2011
Author: David Mellor, Course Director of Audio Masterclass

Quick Tip Video: Playing A File In Sound Forge Audio Studio

Q: What are the best filters for mastering?

 An RP reader asks, "I want to know the best filters for mastering. I want to know the best EQ."

By David Mellor, Course Director of Audio Masterclass

Well I would say that in any artistic pursuit there is no such thing as 'best'. It's all down to personal taste, and what pleases your client or market.

So by this logic there is no such thing as the best filter for mastering or the best EQ.

There is one exception to this in general in music and recording... That is where you want to copy a certain sound. So if a producer asks you to get the same bass sound as he has heard on his favorite Lady Gaga song, you will be off to a good start if you can find out which instrument and setting were actually used.

Back to mastering...

Suppose you have perfected your mix to the maximum and feel that it's the right time to start the mastering process.

You will reach for your favorite mastering plug-in obviously. Well that's what most people do.

But what you could consider doing is applying the various processes involved in mastering 'by hand', rather than just slapping a plug-in on the buss.

One favorite mastering process is multi-band compression. It's powerful, and you can really screw things up with it. But it can really help you fine-tune your master, if you are careful and know what you are doing.

So you could insert a multi-band compressor.

Or you could do it 'by hand'.

What you can do is use auxiliary sends to send copies of your mix to four new stereo auxiliary tracks. (I would really like to call them channels, which is what they are, but I'm using the language of Pro Tools for this example.)

Arrange things so that only these four new tracks go to the master fader, and none of your original mix.

Now, in each track insert a filter. Not an EQ - a filter, or filters. You might find your filters as part of your EQ plug-in. This is fine, just don't use the EQ part.

Set the filters to split up the frequency band into four, so one track handles the bass, one track the lower mids, one track the upper mids and the final track the high frequencies.

So for instance in the LF track, you could set a high-cut filter at 200 Hz. For the low-mid track you would set a low-cut filter at the same frequency (200 Hz) and a high-cut filter at say 1000 Hz.

You can choose a filter slope, but all the slopes should be the same. Start with 18 dB/octave.

If you can do that, the rest of the filter settings should be obvious.

Now when you mix the four tracks together then it should sound pretty similar to your original mix. I say similar rather than identical because it depends on precisely how the filters are designed whether everything will mix together 100% correctly. But since the mastering process changes the audio significantly anyway, we won't worry about that for now.

Next, insert a compressor into each track after the filter(s).

Guess what? You now have a multi-band compressor to play with, and you made it by hand.

So go on and play with it. Experiment with the crossover frequencies and slopes, particularly experiment with compression ratios, thresholds and everything else compressors can do. Use the faders to balance the levels of the four bands of frequencies.

I can guarantee that if you spend a few hours experimenting with this, everything you do with purpose-designed multi-band compressors will be very much better in the future, to the benefit of your mastering skills.

Back to the original question. Really it isn't a matter of having the best equipment or software. You need decent equipment and software, but after that it's all down to how you use it.

 

Publication date: Wednesday December 29, 2010
Author: David Mellor, Course Director of Audio Masterclass

New for Sound Forge™ Audio Studio 10: 24-bit/32-bit float, 192kHz audio support

Can curtains provide good soundproofing?

Posted on August 19, 2012

One of the rules of soundproofing (sound insulation or sound isolation if you prefer) is that sound is best blocked by reflection, using a barrier of high mass. Bricks, high-density blocks, concrete and plasterboard (gypsum board) are all good. So is thick glass. Particle boards are useful too in sufficient thickness.

If you were aiming for near-total soundproofing, then reflection is the way to go. But it is costly, disruptive to install, and small defects in construction can cause a significant loss of performance. There is a strong case to be made for calling in professionals with experience in recording studio construction. That’s going to cost.

But you can take the view that whatever the existing level of soundproofing in your studio, whether it is a bedroom, garage or shed, it can be improved to a useful extent at a relatively low cost and with little disruption.

Small improvements can be very worthwhile, especially if they don’t cost too much.

So, courtesy of Twitter, I came across a company that makes soundproofing curtains – Quiet Curtains.

Now you might think that curtains couldn’t possibly provide good soundproofing. After all, they are relatively lightweight, and some sound energy is bound to avoid them simply by going around the edges.

But that isn’t the point. They don’t have to provide perfect soundproofing. They just have to make things better at a reasonable cost. The technical documentation on Quiet Curtains’ website is impressive. And the price compares well to quality curtains custom made.
I’m not normally convinced by mere claims of performance, I like to see and hear the evidence for myself. And I do have some experience of curtains and soundproofing…

A year or so ago I bought some blackout curtains for the same reason anyone would buy them – to keep my bedroom dark until a reasonable hour during the summer months. I was surprised that a label on the packaging claimed that they also provided soundproofing. Nonsense, I thought, they’re nowhere near thick enough or heavy enough.

I live within earshot of a road that is busy from early in the morning, and I can hear it in my bedroom. Not too loud, but it’s there. But when I installed the blackout curtains, I was amazed at the difference in the subjective level of the noise from the road. If I had made a measurement, I doubt if it would have amounted to even a couple of decibels across the whole frequency band. But the difference at high-mid and high frequencies was clearly audible.

As I said earlier, perfection in soundproofing is expensive, and is often unattainable. Good soundproofing is also hard to achieve. But a difference in the level of soundproofing that makes a small but useful improvement can be surprisingly easy to achieve and – significantly – not cost too much.

David Mellor

P.S. The same applies to acoustic treatment, but that’s another story.

10 Tips For Faster Editing in Sound Forge Pro 10

Unlock your brain's hidden power!

 Creativity not flowing? You know it's in there, but can you get it out and into your recording?

By David Mellor, Course Director of Audio Masterclass

You probably know the feeling. You've got the first few lines of a new song, but somehow the rest just won't come. And the harder you try, the more your brain seizes up and blocks the free flow of creativity.

It isn't that your brain has suddenly lost its creativity. You have what writers call 'writer's block' (as though no-one else ever suffers from it). So what can you do?

Here are three brain unblocking techniques, all tried, tested and proven to work...

Lateral thinking

First popularized by Edward do Bono, lateral thinking seeks to work around a problem rather than hitting it head on.

Let's say you have written the first half of the lyrics of a verse, but you're stuck on the second half. OK, reach for a broadsheet newspaper. Or a fat thesaurus. Or anything that contains a lot of words in fairly random order (i.e. not a dictionary, although that's better than nothing).

Close your eyes, open up your random word source and put your finger on it somewhere. Take a look and see what the word underneath your finger is.

Now what you do is think of all the associations that word has with the topic of your song. Suppose the word was 'frog'. I don't know too many songs about frogs but I suppose Paul McCartney would tell you differently.

Frogs jump, they are often green, some are poisonous, they have long sticky tongues, they eat flies. Think of everything you can that is connected with frogs and look for links with your song. You'll be amazed how quickly you can come up with a creative idea that would otherwise not have occurred to you.

That's just one lateral thinking idea. Although de Bono's book is getting on in years now, it is still well worth a read.

Metronome practice

This one is completely different, but it demonstrates how the brain has power that is locked away and not available for conscious use.

Classical pianists often practise to the beat of a metronome. It is often wrongly thought that this is to help them play in time. Drummers would benefit from click-track practice, but playing the piano really should not be subject to such restrictions.

The true benefit of metronome practice is this...

Suppose you have a section of music that is difficult to play without hitting wrong notes. Slow practice is beneficial, but it doesn't necessarily lead to being able to play the passage at full speed.

So set a metronome to a tempo that is slow enough to play comfortably. Play the section a few times, then increase the tempo a couple of BPM. Play again a few times. Rinse and repeat, over and over.

Here comes the interesting bit...

At some point in the sequence of increasing tempi, you will find that your conscious brain can't handle all the data throughput. So some of the processing will be handed over to your subconscious brain. As you continue increasing the tempo, you will find yourself thinking less and less about where your fingers are going. They will seem to find the notes themselves.

Interestingly, this works for improvised playing too. Improvise to a slow beat, then increase the tempo bit by bit. Soon you will find music flowing from your fingers without conscious thought. It can almost seem a little spooky, as though someone else had taken control of your playing.

Riffing

The word 'riffing' can mean to play riff-based music. But it has another meaning related to improvisation.

Try this... Put a backing track together, or borrow a sample from someone else's music (just in private so no copyright issues). Four, eight, sixteen or thirty-bars work well. Make it loop round and round.

Now set up a mic and put some EQ, compression and reverb on it so that it sounds nice. Get a nice blend between loop and microphone on headphones (it seems to work better on headphones I find).

As the loop plays, start singing. Just start singing. Any old syllables will do and any old tune. Vary what you do. Go crazy.

Then start forming words. Force yourself to sing one word after another, in tempo with the music. Sing any word rather than let the mic fall silent. When the words are flowing, force them to start making sense.

Once again, your brain will be compelled to draw on its subconscious reserves. After a while of doing this, you will have the creative backbone of a new song, which you can further develop.

Over to you...

These are just three ways out of many to access the brain's hidden reserves. The human brain is an incredible resource that is often left largely untapped.

So how do you tap into your hidden brainpower?

Publication date: Monday July 11, 2011
Author: David Mellor, Course Director of Audio Masterclass

Lyrics in Finale 2011

How good is your drummer? Put him to the test in five different ways.

 A good drummer is worth his or her weight in platinum. But how can you know for sure just how good they are?

By David Mellor, Course Director of Audio Masterclass

A good drummer has the quality of 'snap'. It's impossible to put into words what snap is. But if a drummer has it, you know straight away. If you're wondering whether a particular drummer has snap or not, he doesn't.

But any way of analyzing the quality of a drummer's timing and precision has to be a good thing. 'What gets measured gets improved' is an old saying.

The Beatnik RA1200P Rhythmic Analyzer will do just that. It has a touch-sensitive pad on which the drummer can show off his skills. The analyzer's display will show each stroke visually with an accuracy down to 512th notes. Yes really.

The unit can review stroke accuracy and analyze transitions between changing subdivisions and display separate historical accuracy summaries for each.

Of course the unit is adaptable for different skill levels. No point in starting off on the John Bonham setting.

You can find out more about the Beatnik RA1200P Rhythmic Analyzer at www.tuners.com.

By the way, we have no connection with this company other than we found the product interesting.

Publication date: Sunday April 11, 2010
Author: David Mellor, Course Director of Audio Masterclass

Finale music notation software: Staff Layout in Finale 2011

Is it cheating to use Auto-Tune?

 The Auto-Tune debate is unlikely to go away. Those who can, sing. Those who can't use Auto-Tune.

By Jim Covington

RP reader Jim Covington contributes to the great Auto-Tune debate...

"I have been involved with the music for over 50 years and in the music industry and education fields for nearly all of those years. Back before there were processors available to the masses, there were artists concerned with merely expressing their art.

Artistry does not mean 'without flaws'. It is merely an interpretation of something that conveys a view point, flawed though it may be. It pleases some but not all. That being said, the advent of computer processors brings on the age of 'non tolerance.

Processors are incapable of impartial judgment. The programs to run the processors are based on a series of 'on' and 'off' signals. Hence a 'yes' or 'no', a 'black' or 'white', a 'right' or 'wrong', and finally 'perfection' or 'imperfection' society which has spilled into every aspect that the society is built upon.

Have you noticed how intolerant the world's people have become when it comes to any 'gray' or flawed areas? An example might be as follows. In school, a student is considered a disgrace, a failure if they do not get 'A's' or at least 'B's'. 'C' being the standard measurement of 'average' is not good enough, flawed. It represents the 'gray' area. An area between perfection [success] and imperfection [failure].

In the music industry, to be specific, with the advancement of technology, engineers have begun to see that there can be a level achieved that was once thought beyond their grasp.

An example might be the 'lack of noise' which was once thought beyond the grasp of recording studios and which today is easily achieved applying today's technology. Now with the ever advancing software, the public has a chance to enter that magical, unique 'club', of the creational entity [the record company].

The software enables enhanced personal creation by the masses, though they be personally flawed, their personaL artistic uniqueness exposed for the first time in history where the whole world is able to hear and see.

So it comes down to the question and issue of using the latest technology to express a person's creation. Yes, just as Leonardo da Vinci was entitled, and in many ways encouraged, to use the technology [unique chemical pigments] of that time that not only enhanced the brilliance of nature's colors, but expressed his interpretation of nature, though it did not truly express nor exposed nature's flaws.

In addition, because of their chemical makeup the paintings tended to last much longer and adhere longer to materials than other painters pigments. Was he cheating? The world did not think so. So it would seem, with Auto Tune..... the singer, producer, or engineer is simply expressing their interpretation of the song [vocal tonal perfection ] which will please as many of the audience that they are attempting to reach as possible.

It is not cheating to use the technology of the day and age, UNLESS there is an attempt to defraud, to state openly and defiantly that the 'art' presented is the 'true' example of 'fact' and not merely the expression of the creators. It would not hurt the 'true' artist to be willing to state in some printed form, the technology used in creating the artful expression.

Should the artist is unwilling to do so, then they would need to be exposed, as was the case of the singing duo, 'Milli Vanilli'?"

Thanks Jim.

Publication date: Sunday April 11, 2010
Author: Jim Covington

Finale's ScoreManager Controls Staff Appearance and Playback

When should you use Auto-Tune, and when should you avoid it like the plague?

 Auto-Tune is a powerful vocal tuning software. It can turn an out-of-tune vocal track into a pitch-perfect pro performance. But are there times when you would be better off not using it?

By David Mellor, Course Director of Audio Masterclass

When you should not use Auto-Tune

You should definitely not use Auto-Tune when you are working with a great singer backed mostly by acoustic instruments, or electric instruments such as the electric guitar or (genuine) Fender Rhodes piano.

The reason for this is that if you have a great singer, what could you possibly want to do to meddle with their performance? Tuning is subjective and there is no such thing as 'accurate'. Accuracy in tuning means perfect alignment to the even-tempered scale that is a compromise already. Take away the subtle shadings of pitch that a great singer will use, and you have taken much of the artistry out of the performance.

If you are working with a singer who is dreadfully out of tune, Auto-Tune can't offer much help. In order to recognize which notes the singer intends to sing, they must be performed with an accuracy of better than half of the step between adjacent notes. For some notes in the scale, this means an accuracy of better than a quarter of a tone. Manual retuning will be necessary.

When you should use Auto-Tune

When you have a singer who is a little less than perfect, backed by electronic or digital instruments that are always exactly in tune, Auto-Tune will probably help. The comparison between the perfect tuning of the instruments against the slight approximations of the singer is often uncomfortable to listen to. Auto-Tune can make the singer as accurate as the instruments.

When you should carefully consider whether to use Auto-Tune

When you need to get the work done quickly. If you need to get the work out of the door in a hurry, and often there are occasions when this is necessary, Auto-Tune can help speed things up, rather than spend time on retakes.

In summary, Auto-Tune and other pitch correction software can be very useful. But be careful that they don't take the soul out of the performance.

Publication date: Saturday April 04, 2009
Author: David Mellor, Course Director of Audio Masterclass

Quick Tip Video: Show Me How in Sound Forge Audio Studio

What frequencies do different instruments produce?

 Is it possible to relate frequencies to instruments? For instance, will the voice will be more present with a boost at 3 kHz? What about other instruments?

By David Mellor, Course Director of Audio Masterclass

Yes you can find charts that show the frequency ranges of various instruments.

There is a problem however...

The chart will show the frequency range of the double bass as 40 Hz to 200 Hz; the piccolo from 630 Hz to 5 kHz, approximately.

However this does not tell the whole story.

These are the fundamental pitches of the notes that can be played on these instruments. The chart takes no account of the harmonics of the instruments.

All musical sounds possess harmonics. If you play a single note on any string or wind instrument, a fundamental frequency will be produced, which is the note you hear. But also there will be frequencies that are whole-number multiples of the fundamental frequency. These are the harmonics.

And the harmonics of any acoustic instrument range right up to the limit of human hearing and beyond. Even for a double bass.

In sound engineering however, you will find that certain bands of frequencies can be used to enhance different instruments. Experiment with the frequency control of your EQ and you will find that certain settings can make the trumpet more 'trumpety', the violins more 'violiney' etc.

And indeed the human voice can often be brought out with an EQ boost around 3 kHz, even though this is way beyond the range of fundamental pitches that anyone could sing.

So, take frequency charts with a degree of caution. Experimentation, listening and good judgment will always bring you better results.

 

Publication date: Friday July 23, 2010
Author: David Mellor, Course Director of Audio Masterclass

George Enescu - Romanian Rhapsody No. 1 Op. 11

The nature of analog magnetic tape

Magnetic tape comprises a base film, upon which is coated a layer of iron oxide. Oxide of iron is sometimes, in other contexts, known as 'rust'. Er.. we used to record on rust?

By David Mellor, Course Director of Audio Masterclass

Magnetic tape comprises a base film, upon which is coated a layer of iron oxide. Oxide of iron is sometimes, in other contexts, known as 'rust'. The oxide is bonded to the base film by a 'binder', which also lubricates the tape as it passes through the recorder. Other magnetic materials have been tried, but none suits analog audio recording better than iron, or more properly 'ferric' oxide.

There used to be several manufacturers of magnetic tape - Ampex (later called Quantegy) and BASF (later Emtec) were among the front-runners. But gradually most dropped out as the market shrank. Emtec tape is currently still available (and not just on eBay!).

Tape has been manufactured in a variety of widths. (It has also manufactured in a range of thicknesses - so-called 'long play' tape can fit a longer duration of recording on the same spool, at the expense of certain compromises.).

The widths in common use today are two-inch and half-inch.

Oddly enough, metrication doesn't seem to have reached analog tape and we tend to avoid talking about 50 mm and 12.5 mm. Other widths are still available, but they are only used in conjunction with 'legacy' equipment which is being used until it wears out and is scrapped, and for replay or remix of archive material.

Quarter-inch tape was in the past very widely used as the standard stereo medium, but there is now little point in using it as it has no advantages over other options that are available.

Two-inch tape is used on twenty-four track recorders. A twenty-four track recorder can record - obviously - twenty-four separate tracks across the width of the tape, thus keeping instruments separate until final mixdown to stereo.

Half-inch tape is used on stereo recorders for the final master.

The speed at which the tape travels is significant. Higher speeds are better for capturing high frequencies as the recorded wavelength is physically longer on the tape.

However, there are also irregularities (sometimes known as 'head bumps, or as 'woodles') in the bass end. The most common tape speed in professional use used to be 15 inches per second (38 cm/s), but these days it is more common to use 30 ips (76 cm/s), and not care about the massive cost in tape consumption!

At 30 ips, a standard reel of tape costing up to $150 lasts about sixteen minutes.

Hmm. Compare that with the cost of hard disk recording space these days...

Publication date: Tuesday April 10, 2007
Author: David Mellor, Course Director of Audio Masterclass

Quick Tip Video: Change Pitch Of Event In ACID Music Studio

What happens when a microphone preamplifier clips?

 Although any competently designed microphone preamp should function perfectly when used within its capabilities, sound engineers have a habit of not always being sufficiently careful. Here's a scenario where things can go wrong...

By David Mellor, Course Director of Audio Masterclass

Although any competently designed microphone preamp should function perfectly when used within its capabilities, sound engineers have a habit of not always being sufficiently careful. Here's a scenario where things can go wrong...

Suppose a microphone is being used to record natural-sounding speech. With the speaker at a reasonable distance from the microphone, the output could be around -60 dB most of the time, with respect to the kind of signal that the mixer or recorder needs, which is around one volt. So the engineer applies 60 dB of gain.

However, although speech has a mostly consistent average level, it is also very 'peaky'. There are frequent transients that vastly exceed the average level, often by as much as 30 decibels. So when these come along, as they will, they are also amplified by 60 dB. If this kind of level were amplified correctly it would reach over thirty volts! But in practice, there will simply not be enough voltage available in the preamp's power supply range. So the signal is clipped.

This clipping takes place over a very short time period. However, at the moment it occurs, the preamplifier's ability to correct distortion, through the technique known as negative feedback, disappears. So there is a momentary and intense burst of distortion. Also, the preamplifier is now destabilized and will take a certain time to recover. All of this is audible, and since different designs will react in different ways, they will certainly sound different - some better than others.

Ideally, the preamp would have metering that would detect this and make clipping known to the engineer. However, most mic preamps have only rudimentary metering, some have none at all, and others have metering that measures the level after the damage has been done! There is clearly some improvement necessary here.

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

Quick Tip Video: Extract Audio From A CD In ACID Music Studio

What is the no-go area around a crossed pair of figure-of-eight microphones?

 A crossed pair of figure-of-eight microphones is sensitive to the front, sensitive to the rear, but elsewhere there is a 'no go' area? Why is this?

By David Mellor, Course Director of Audio Masterclass

It is common to record in stereo using a coincident crossed pair of figure-of-eight microphones.

'Coincident' means that the diaphragms of the mics are as close together as possible. 'Crossed' means that one mic points left, the other points right, separated by 90 degrees.

'Pair' means there are two of them :-)

A coincident crossed pair always works, if you can find the right position for the mics, and captures a good clear recording.

When figure-of-eight pattern mics are used for the crossed pair there is an interesting feature - since the mics are equally sensitive to the front and to the rear, the rear is effectively another 'front'.

This is commonly exploited in drama recording where a group of actors can divide themselves between front and rear, and in the resulting recording they will all seem to be located together.

However there are two 'no-go' areas around a crossed pair of figure-of-eights.

Sound that approaches the mics from the sides has the unfortunate effect of striking the front of the diaphragm of one mic, and the rear of the diaphragm of the other. So the two diaphragms are pushed and pulled in opposite directions.

In other words, the signals are out of phase, or inverted, with respect to each other.

When this is played through loudspeakers or headphones, the net result will be one eardrum being pushed inwards while the other is sucked outwards, and vice versa.

There is no sound in nature that does this, so the human hearing system has difficulty to interpret what it is hearing. It sounds distinctly odd.

So the moral of this story is never to place sound sources in either of the two quadrants to the sides of a crossed pair of figure-of-eight mics.

This is fine for direct sound sources where you have control. But you have little control over reflections reaching the sides of the pair.

Out of phase reverb is a characteristic of crossed figure-of-eights and perhaps accounts for the fact that although in theory this configuration should produce perfect stereo sound, in practice it doesn't quite live up to that ideal.

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

Quick Tip Video: Beatmapper In ACID Pro

Is SCSI dead (yet)?

 With FireWire and USB-2 now commonplace and reliable, is SCSI still with us? Does it have a future in recording?

By David Mellor, Course Director of Audio Masterclass

One of the misconceptions about music and sound technology is that if you haven't heard much about something for a while, then it must be dead. This is closely linked to the attitude that you can only create good music if you have the latest equipment (not true!).

SCSI, in terms of general computer connectivity, was the predecessor to FireWire and USB. For most purposes, SCSI was a pain in the neck compared to these more modern and slick standards.

But there is one area where the latest versions of SCSI - and yes, it is still evolving even now - excels, and that is in lightning-fast hard disks.

SCSI has two advantages over the more common FireWire and USB methods of connecting hard disks (and even over the latest internal SATA interface).

The first is that it is a parallel interface. This means that data is transferred over many wires simultaneously rather than just one pair of conductors. The consequence of this is that several bits can be sent down the line simultaneously, whereas in a serial interface such as FireWire, each bit has to wait its turn in line.

Now serial interfaces do have their good points. One is that it is easier to use longer cables. But ultimately, over short distances, a fast parallel interface will always be faster than a fast serial interface.

The other advantage that SCSI still has is that SCSI hard disk drives are themselves imbued with intelligence.

So the computer simply shoves a load of data off to the SCSI drive, and the drive itself manages all the 'housekeeping' work. It's one less load for the computer's processor.

If you would like examples of where SCSI is still used, then firstly look no further than the internet. The highest speed servers are equipped with SCSI disks, even though they are more expensive, gigabyte for gigabyte. They can simply handle heavy data loads better.

The other example is from Digidesign. If you have a Pro Tools HD Accel system (the best), then if you also equip yourself with a Digidesign SCSI-128 kit, then you will be able to achieve a massive track count of 192 tracks! Yes, that is one hundred and ninety-two tracks.

What you'll do with all those tracks is a good question, but thanks to SCSI you'll be pretty sure you won't run out!

Publication date: Thursday February 10, 2005
Author: David Mellor, Course Director of Audio Masterclass

Quick Tip Video: Using the Chopper In ACID Music Studio

What important feature do analog tape simulators lack?

 It's a strange world where you can buy a secondhand analog tape recorder for less than the price of some tape emulation plug-ins and devices. But which will give you the most authentic sound?

By David Mellor, Course Director of Audio Masterclass

The early days of digital were so exciting. Suddenly you could record without discernable noise and distortion. Prior to that, an important part of the recording process was managing the vast amounts of noise and distortion produced.

And weren't those ultra-clean digital recordings soooo good?

Err. no. They were not good. It was interesting for a time to be able to record cleanly. But after a while the novelty wore off, and the 'dirty' sound of analog became a longed-for attraction.

So we started buying old vacuum tube gear. That helped. Then plug-ins became available that would 'grungalize' our squeaky-clean digital signals.

And then came analog tape emulators.

But what is it that analog tape emulators actually do?

Analog tape recorders produce a certain kind of distortion. It increases with higher signal levels. It is also symmetrical on positive and negative peaks in the signal, leading to the generation of strong odd-order harmonics.

A single-ended tube amplifier on the other hand is asymmetrical and produces both even-order and odd-order harmonics.

So a tape emulator sounds different to a tube emulator (or real tubes) and adds another useful tool to the audio toolbox.

Of course, analog tape recorders were also noisy. Tape emulators generally give an option whether you have the noise or not. Believe it or not, it is sometimes useful to have it.

But there's one feature of analog tape that emulators lack. And that is wow and flutter.

One tape emulator actually says this is a good thing, to lack this intrinsic component of the analog tape sound!

Let's be clear... wow is a bad thing. Wow is a noticeable cyclic up/down change in pitch. It is truly horrible.

But flutter is another thing totally. Flutter is a fast and mostly irregular change in pitch. It's far too fast to be perceived as a pitch change. Instead it adds something wonderful to the signal...

Sidebands!

You get sidebands with tubes. Put a clean 1 kHz sine wave into a tube amplifier and turn up the gain. You will get additional frequencies at whole-number multiples of the input frequency.

You will also get intermodulation products with more complex relationships.

But the thing is that they don't move. They are frozen and locked to the input signal.

In analogue tape, not only do you get sidebands due to the distortion, you get massively more, and more complex, sidebands due to the flutter. And they change continually.

This is what gives analog tape its incredible rich, warm and involving sound.
But the emulators miss that.

But the emulators often miss that. So far...

Publication date: Monday August 10, 2009
Author: David Mellor, Course Director of Audio Masterclass

Quick Tip Video: Add Audio FX In ACID Music Studio

Why classical music is superior to popular music, and always will be

 Popular music is here today, gone tomorrow. Each new recording gets more and more stale as time goes by, where classical music can always remain fresh and new.

By David Mellor, Course Director of Audio Masterclass

Yes classical music is superior to popular music. Not in every way perhaps, but it has one advantage that popular music threw away years ago.

Let's start with Elvis Presley. Some say that there was no music before Elvis. Clearly there was, but clearly too he was important.

But the 'was' in that last sentence is very significant. Elvis, contrary to the belief of some, is dead. All we have of him is his recordings. And a few Elvis impersonators.

The recordings are great. But the problem is that every time you play one of Elvis's records, it's always the same. However great a record may be, it gets increasingly tired and stale as time goes by.

Even if Elvis were alive today, he wouldn't be the same Elvis. Not the youthful innovator and shaker of pelvic bones. Nor the reinvented Elvis of the 1968 NBC TV special. Nor the 'fat and cuddly' Elvis who could still enthuse an audience, although some of his performances didn't exactly do the songs justice.

If Elvis were alive today he would still be singing his old hits. But it wouldn't be the same. The original records are the definitive performances, and they will never be equaled or surpassed by anyone.

So that is the problem - recording!

Somehow, popular music has focused on the record as being the definitive version of a song. Whoever is the first to get a hit record with a song defines how that song should sound for eternity. Anything else can only be an imitation.

Now let's compare that with classical music. In the heyday of classical music, there was no recording. So the only way to preserve music was to write it down.

And the written score has come to be regarded as the definitive version of any piece of classical music.

We can't go back in a time machine and hear a performance directed by Mozart. All we can do is perform the music as best we can from the score.

But that is surprisingly advantageous. Because there is no definitive recording of any piece of classical music (even modern works regard the score as the original, even if the composer has conducted a recording), it is open to anyone to give their own individual interpretation of that music.

And performance styles can change over the years. No matter how much historical evidence we can gather, we will never know for sure how music was performed prior to the era of recording.

So, because there is no definitive version of Mozart's 40th symphony, for example, we can go on performing it and re-interpreting it forever.

But popular music... well we'll always have to listen to those old recordings, because the recording is the definitive version and it can never be bettered.

Still, let's not dwell on negatives. I propose two solutions to this problem...

One, that we all go out and buy violins.

Or, maybe we shouldn't be so focused on recording. Maybe it's the song that is most important, and a recording should merely be seen as a version of that song. Anyone can come up with a new version, and no version is considered to be the definitive recorded performance.

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

Haydn: Symphony No. 98 / Koopman · Berliner Philharmoniker

Why your music should be LESS tuneful

We all love a good tune. But there are some areas of music where tunes are positively not required and will only be to your disadvantage.

By David Mellor, Course Director of Audio Masterclass

What is it about tunes anyway? If you listen to music from earlier centuries you will realize the the concept of a tune was something that had to be discovered.

Even J.S. Bach, genius though he was, went through his entire career with hardly a tune to his name. At least what we would call a tune today.

Now Wolfgang Amadeus Mozart was certainly capable of writing tunes. When he felt like it. Mostly he didn't, Neither did Beethoven. Mostly mind, I'm not saying there are not exceptions.

The heyday of the tune however started with Franz Schubert. He wrote a lot of songs, and somehow songs seem to demand real proper tunes more than purely instrumental music.

Skip forward a hundred years or so and we have the popular songs and musicals of the twentieth century. Chock full of tunes. Real tunes. Real tunes that ordinary people sing in the bath and whistle in the street.

And we have instrumental tunes too. TV programs commonly have title themes are are real, whistleable tunes.

Somewhere along this timeline it has become a common idea that music must have a tune. But this simply is not so. Go back to Beethoven and you will see that his work consists largely of musical 'figures' which are elaborated upon. Yes, you can whistle the tune from his ninth symphony, but when do you hear anyone whistle the 'fate' motto from the fifth.

And, getting to the point, in fact most music used on TV does not have and does not require a tune. In fact it requires not to have a tune.

If you really listen and concentrate on the music used in TV programs, you will find that the vast majority is tuneless or possesses only a rudimentary tune. Certainly nothing you would want to sing in the bath.

Exceptions are title themes, and 'motto' themes that commonly occur in dramas. But mostly TV music is really rather tuneless.

So go ahead and make some music without tunes. You'll find that tuneless music is very much easier to sell into the TV market.

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

Quick Tip Video: Record Audio In ACID Pro

How many sound waves can you fit into your studio?

 Do you think your studio is big enough to fit all of your sound in? You might be surprised to learn just how BIG sound can be...

By David Mellor, Course Director of Audio Masterclass

Obviously this is a trick question of some sort. But what is the trick? It's in the answer - although you might be able to fit thousands of sound waves into your studio space simultaneously, there are situations where it might not even be possible to fit one!

The explanation is in the range of wavelengths of audible sounds. Any sound consists of a series of high-pressure and low-pressure regions traveling through the air. The wavelength is the distance between two adjacent highs or lows, or any two corresponding points on the wave.

At the extreme, the shortest audible wavelength, corresponding to a frequency of 20,000 Hz (20 kHz), is around 17 mm. At the low frequency end we could consider 100 Hz, the corresponding wavelength being around 3.4 meters (that is just over 11 feet, but come on - this is science in the 21st century!).

So can you fit something 3.4 long in a straight line into your studio? Perhaps, just - even if diagonally!

OK, let's think about 50 Hz, and this isn't even deep bass - not even close to the left hand end of the piano. Now we are looking at a wave some 6.8 meters long. This is getting serious.

If we go all the way to the generally accepted limit of audibility, which is 20 Hz, then the wavelength is a massive 17 meters! Few people have rooms that can accommodate wavelengths so long.

However, all is not lost, because sound waves are quite capable of bouncing around the room, in effect 'folding' themselves over. But the subjective effect isn't all that impressive. Bass in a small room sounds 'constricted', which it is. On the other hand, bass in an auditorium is allowed to 'breathe' and sounds much more pleasant and natural.

Small rooms are very difficult to treat acoustically, and the major problems are in the bass end.

The advice - move into bigger premises and let your bass breathe!

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

Kronos Music Workstation - EP-1 Electric Piano - In The Studio With Korg

Digital copies - identical clones, or only nearly perfect?

 Why a copy of a digital recording is meant to be a perfect replica of the original, and why in practice it isn't.

By David Mellor, Course Director of Audio Masterclass

In the old days of analog tape, one of the most significant problem areas was in copying tapes. An original analog recording can sound very good. But a copy doesn't sound quite so good, and a copy of that copy is starting to sound distinctly 'iffy'.

Each generation of analog copies increases the noise level and the distortion percentage. The frequency response suffers too as inaccuracies in the original are further increased by frequency response inaccuracies in the copying process.

When digital recording came of age in the early 1980s, it seemed like a solution to all copying problems. A digital recording consists simply of a list of numbers, stored on tape. It may be a long list, but it is just that.

Numbers can be copied identically, unlike electric or magnetic signals where there is always a degree of inaccuracy. In theory therefore, a digital copy is identical to the original and sounds just as good. It is sometimes called a 'clone'.

However, digital tape recordings are subject to errors, even in the original. Where data has not been recorded correctly, it must be compensated for. 'Error correction' makes use of additional data in the recording intended for just this purpose - if some data is bad, then the error correction data can be used to reconstruct the original. This is OK, the sound of the recording is not affected at all.

Error concealment is another matter. This occurs when the damage is too much for the error correction system to completely reconstruct the original data. What happens now is that the system 'guesses' what the data should have been. This doesn't necessarily sound too bad, but it's not quite the same as the original. This is called 'error concealment'.

Any copy made of this recording will treat the error-concealed data as though it were absolutely correct, and any further copy made from the copy will compound the problem.

In practice therefore, it will be found that a digital copy is not as good as the master, and just like in the old analog days, there will be one 'true' master, which is the original, and copies will always be imperfect to a degree, compared to that.

Hard disk systems are better in this respect, simply because it is in the nature of computers to copy data all the time. Consider the computer systems used by a bank. If there were any errors in the data, then concealment would be tantamount to fraud!

Generally, when all is working well, a disk copy will be 100% identical to the original. This is thanks to the more robust error correction system used, and the 'verification' process where data is checked to be correct immediately after recording.

Even so, glitches do happen, and the original recording should be kept safe just in case.

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

Quick TIp Video: In Line MIDI Editing In ACID Music Studio

What are the small faders for on a mixing console?

 A description of the technology and function of the small faders found in inline monitoring mixing consoles.

By David Mellor, Course Director of Audio Masterclass

Since we are considering an inline console, each channel module has two signal paths:

• The input signal, which is the signal from the mic that is being recorded to multitrack
• The monitor signal, which is the output of a single track of the multitrack recorder, the track number of which corresponds to the channel number (usually).

On some consoles, the small fader is normally set to control the level of the monitor signal. So the large faders are used to set recording levels to multitrack, and a temporary monitor mix is set up on the small faders.

On other consoles, this - as a normal condition - is reversed. All inline consoles allow the input and monitor signal paths to be 'flipped', i.e. reversed.

At this point, it is worth saying that the other facilities of the channel can be allocated to either the input or monitor signal paths, or shared.

So for example, you could place the EQ in the input signal path if you wanted to EQ the signal before it went down to tape.

On the other hand if you wanted to record the signal flat, you could put the EQ in the monitor path and use it to temporarily sweeten the monitor mix.

This applies to the dynamics section and auxiliary sends too.

Publication date: Thursday January 01, 2004
Author: David Mellor, Course Director of Audio Masterclass

Quick Tip Video: Creating A Submix In ACID Pro

How Do CDs Work?

How do CDs work? How should you look after a CD? A knowledge primer for better CD mastering...

By David Mellor, Course Director of Audio Masterclass

Browse CD players and recorders...
Article courtesy CDman Disc Manufacturing

Like gramophone records, the information on optical discs are recorded on a spiral track. However, with a CD the laser starts reading the disc from the inside ring (table of contents) and ends up on the outside. When play back starts, a laser beam shines on the ridges and lands on the data membrane layer. If you look at the image on the right you can see the data layer moving in grey.

During playback, the number of revolutions of the disc decreases from 500 to 200 rpm (revolutions per minute) to maintain a constant scanning speed. The disc data is converted into electrical pulses (the bit stream) by reflections of the laser beam from a photoelectric cell.

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Cutaway view of the laser pickup. Depending on whether the laser beam hits a ridge or a land, the laser beam is reflected and received by the photoelectric cell. The disc data is converted into electrical pulses (the bit stream) by reflections of the laser beam off of a photoelectric cell.

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When the laser beam strikes "land", the beam is reflected onto a photoelectric cell. When it strikes a "ridge", the photocell will receive only a weak reflection. Thus the photoelectrical cell receives series of light pulses corresponding to the ridges and lands in the disc. These light pulses are the foundation of binary 'digital' data. A simple substitution for the weak signal "0" and the in-focus signal "1" results in a pure digital playback without alteration, every time, without failure or degradation.

In music playback, a D/A-Converter (digital to analogue converter; DAC) converts the series of pulses (binary coding) from a decimal place to a waveform which can be then processed for amplification. The longer the decimal place the better the sound. Current standard CD audio is 44,100 pulses per second and 16 bit (decimal places) in digital word length. Thus a 24 bit system sounds all that much better, in fact DVD audio is set to allow 24 bit AND pulse at 97,000 times per second! Go DVD go!!!

The Compact Disc player mechanism. The laser pickup reads the disc from below.

Thanks to this optical scanning system, there is no friction between the laser beam and the disc. As a result, the discs do not wear, no matter how often they are played. However, they must be treated carefully, as scratches, grease stains and dust might intercept or diffract the light, causing whole series of pulses to be skipped or distorted. This problem can be solved, as during the recording the Cross Interleaved Reed Solomon Code (CIRC) is added, which is an error correction system that automatically inserts any lost or damaged information by making a number of mathematical calculations. Without this error correction system optical disc players would not have existed, as even the slightest vibration of the floor would cause sound and image distortions. 

Scanning the disc (part II)

When the laser beam hits land, all of its light is reflected and the cell gives off current. When the laser beam shines on a ridge, half of the light hits the upper surface and the other half hits the lower down service. The difference in height between the two places is exactly a quarter of a wavelength of the laser beam light, so the original beam is totally eliminated by the interference between the beam reflected from the surface of the disc and the beam reflected from the ridge. The photocell does not produce current.

It should be noted that the ends of the ridges seen by the laser are "ones" and all lands and ridges are "zeros"; thus turning on and off the reflection is one, steady state is a string of zeroes. As it is not possible to have two ones next to each other, Eight to Fourteen Modulation (EFM) is used to convert 8-bit data bytes to 14 bit units that always have a minimum of 2 and a maximum of 10 zeros between ones. This makes the pits/ridges and lands separating them 3 to 11 bits long, no less, no more. This conversion is done in hardware using a ROM lookup table. To connect these 14 bit units 3 merge bits are used to make sure that there are no "ones" too close to each other. In audio, the third merge bit is used to make sure that the cumulative lengths of the lands and ridges stay equal in the long run, otherwise a low frequency component is created that the processing amplifiers can not handle. Thus 8 data bits are actually 17 channel bits on the disc, but called 16 bit for naming conventions.

There are 20,000 tracks like this one on one compact disc.

The scanning must be very accurate because the track of ridges is 30 times narrower than a single human hair. You can see the "ridge" in the llustration above -it is the DARK ROUND CIRCLE. When the laser light is over top of it, the light 'splits' in two, causing a weak signal. There are 20,000 tracks on one audio compact disc. The lens which focuses the laser beam on the disc has a depth of field of about 1 micro;m (micrometer = one-millionth of a meter).

It is quite normal for the (compact) disc to move back and forth 1mm during playback. A flexible regulator keeps the lens at a distance of +/- 2 micro;m from the rotating disc. For the same reason, a perfect tracking system is required. The complex task of following the track is controlled by an electronic servo system. The servo system ensures the track is followed accurately by measuring the signal output. If the output decreases, the system recognizes this as being "off track" and returns the tracking system to its optimum state.

Many CD players use three-beam scanning for correct tracking. The three beams come from one laser. A polarized prism projects three spots of light on the track. It shines the middle one exactly on the track, and the two other "control" beams generate a signal to correct the laser beam immediately, should it deflect from the middle track.

The disc

The CD is a plastic disc 1.2mm thick and 12cm in diameter, with a silver-colored surface that reflects laser light. The maximum playing time for music recorded on compact disc is 74 minutes. The CD has several layers. First, to protect the 8 trillion microscopically small pits against dirt and damage, the CD has a plastic protective layer. On the top of this layer the label is printed. Then there is the reflecting aluminum coating which contains the ridges. Finally, the disc has a transparent carrier through which the actual reading of the disc takes place. This plastic forms a part of the optical system. Mechanically, the CD is less vulnerable than the analogue record, but that does not mean that it must not be treated with care.

The CD has several layers. Notice how the ridges contain binary information.

The protective layer on the label side is very thin: only 0.002mm. Careless treatment or granular dust can cause small scratches or hair cracks, enabling the air to penetrate the evaporated aluminum coating. This coating then starts oxidizing immediately at that spot. If the CD is played extensively, it may be advisable to protect the label side with a special protective foil, which is commonly available in shops.

A CD must never be bent, so care should be taken when removing it from the jewel case. Even slight bending causes stress fractures. The aluminum then becomes deformed, causing some ridges to be blocked. As a consequence, error correction always has to be applied in that area, affecting the final sound.

The reflecting side of the CD is the side that is read. People tend to set the CD down with the reflecting side up. But the more vulnerable side is not the reflecting side but the label side. On the label side, the reflecting layer with its ridges has been evaporated. The sensitive layer on the reflecting side has been protected better than the one on the label side. It is therefore better to store CDs with the reflecting side down. It is best to store the CD back in the jewel case, where it is safely held by its inside edge.

Never write on the label side, even with a felt-tipped pen. The ink may penetrate the thin protective coating and affect the aluminum layer.

If a smear, however small, remains on the CD, much information is lost.

CDs are easily scratched, and should never be cleaned with just any cloth. CDs should be cleaned radially: not along the grooves, but at right angles to the direction of the grooves. If a smear, however small, should remain on the CD, running along the direction of the grooves, much information would be lost. It is advisable to use special CD cleaner that operates with a rotating brush at right angles to the direction of the grooves.

Many people think that the digital CD is produced completely digitally, but this is not always the case. Many CDs have an analogue master tape as their source tapes still kept in the library of the record company, used in the past to make records. The quality of a CD made from analogue tape can be surprisingly high. A CD recorded, processed and dubbed digitally does not always sound better than a CD produced with one or two analogue processing stages.

To indicate what stages have been treated in what ways, a useful three-letter code is used on recordings. The letters represent: the recording, the editing/mixing process, and dubbing, respectively. They are printed on the CD and/or on the insert label in a rectangular box. There are three possibilities: DDD (completely digital CD); ADD (analogue recording, digital processing and dubbing); and AAD (analogue recording and processing, digital dubbing). Many CDs carry the ADD or AAD indication. This does not mean that they are inferior to the DDD CDs!

[Much information courtesy Philips - co-inventors of the CD]

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