Welcome to No Limit Sound Productions

Company Founded
2005
Overview

Our services include Sound Engineering, Audio Post-Production, System Upgrades and Equipment Consulting.
Mission
Our mission is to provide excellent quality and service to our customers. We do customized service.

Monday, December 31, 2012

Happy New Year!!

Enjoy our new release of
"Solace" music album by Jordan
Listen to a sample or download here
View "Solace" sample trailer here
Purchase a CD here
Coming Soon to Pandora Internet Radio!!
"Like" us on Facebook.
  From all of us here at No Limit Sound Productions, we would like thank you for being with us and would like to you wish you all a

Happy New Year!!

Happy Holidays from all of us to all of you!

Coming Soon! "In Motion" music album by Jordan
Check out Jordan "In Motion" on Facebook
or
Enjoy "Solace" by Jordan
Listen to a sample or download here
View "Solace" sample trailer here
Purchase a CD here
  From all of us here at No Limit Sound Productions, we would like thank you for being with us and would like to you wish you all a

Happiest New Year!!


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

Friday, December 28, 2012

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

Monday, December 24, 2012

KORG monotribe - Jingle Bells / BakaOscillator

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

Friday, December 21, 2012

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

Schumann: Cello Concerto / Moser · Mehta · Berliner Philharmoniker

Q: I'm converting audio from 16-bit to 24-bit. Should I worry?

 A Record-Producer.com reader is importing 16-bit audio into a 24-bit session. He is concerned whether there will be any quality loss.

By David Mellor, Course Director of Audio Masterclass

Here's a protip for life in general - worry about the things that make a difference, don't worry about things that don't. Another - You'll die if you worry, and die if you don't. So just get on with life and stop worrying.

We get a lot of inquiries here at RP Towers from people with worries. The usual worry is that they don't have the right equipment. Sometimes they are justified - plugging a microphone into an interface with only line inputs is never going to lead to success. But no-one knows these things automatically so we are happy to help.

But quite often we find that people are worrying about things that don't really make any difference. We take it that anyone who follows RecordProducer.com and Audio Masterclass is primarily interested in making music and/or recordings of a professional standard. And by 'professional standard' we mean a quality that will satisfy a typical industry client, or sell into the market.

So let's have a look at the 16-bit to 24-bit conversion issue. Is it an issue at all really?

The first thing to consider is the significance of each of the bits. Those sixteen bits are used to quantify the voltage level of an analog signal in digital terms, in 65,536 steps (2 to the power 16).

So if at any instant the digital signal is at 1001010001101011 then that is a digital description of the instantaneous level of an analog voltage. In digital terms, the left-most 1 or 0 is the most significant bit. The right-most 1 or 0 is the least significant bit. Going from left to right, the signal is described with greater and greater accuracy, or you could say in finer and finer detail.

Sixteen bits are sufficient to describe a signal with a dynamic range of 96 decibels from its loudest to its quietest parts. (In real life it's a little less, but we'll stick with the theory.)
96 decibels is a HUGE dynamic range. Play some music really loud through your monitors. Now reduce the level by 96 dB. What can you hear? Nothing. Or as near to nothing as makes hardly any difference.

When a 16-bit signal is imported into a 24-bit session, all that happens is that eight more bits are added to the right of the previously least significant bit. In this case, they should all be zeroes as the 16-bit signal has no information here.

But what about dither?

Ah, there just has to be a complication....

One problem with digital audio is quantization distortion at very low levels. It sounds nasty and it is something to be concerned about. The way it is dealt with is to add a little dither noise right at the end of the recording chain. This eliminates the distortion and although adding noise sounds like a bad thing, it isn't - it's totally good. You will almost certainly have a plug-in for it.

So what difference would it make whether or not your 16-bit files were dithered?

Well if they are undithered what will happen is that quantization distortion will remain in the lower levels of the 16-bit signal.

If your 16-bit files are dithered, then all should be fine. Unless of course your DAW's import function adds dither without you knowing about it, in which case you have dithered twice so your signal is now noisier than it ought to have been.

The question is of course, do you know whether your 16-bit files are dithered? If they sound free from distortion at very low levels, they probably are. And if you can't hear any quantization distortion, well it doesn't really make much difference.

Dither is something that people can often get really worried about. Ideally you wouldn't add any yourself at all and leave it to the mastering engineer.

But why not explore dither for yourself - play around with some very low level audio -70 or -80 and below. Switch dither in and out; try different types of dither. When you hear what it does for yourself, you will be in a good position to make decisions.

In summary, if you need to import 16-bit files into a 24-bit session, just go ahead and do it. Listen closely - if they sound like they need dither then add it (at the 16-bit level). If they don't, then just leave them be. Why worry?

 
Publication date: Thursday March 24, 2011
Author: David Mellor, Course Director of Audio Masterclass

Thursday, December 20, 2012

NOW AVAILABLE FOR THE HOLIDAYS!! No Limit Sound Productions CD release

NOW AVAILABLE!!
on CD
  No Limit Sound Productions is pleased to announce the release a of new instrumental music album 
by Jordan.
Instrumental New Age music aids in stress management.  "Solace" album by Jordan takes you to an added dimension.  Solace is defined as a source of relief and cheer.  In the stress of your daily life you can relax with this New Age Instrumental Music.

Korg All Access: Greg Phillinganes demos the Korg Krome Music Workstation

Q: How should I use an equalizer with a limiter?

 Q: "Could you tell me please whether the equalizer should go first, then the limiter. Or should the limiter go first?"

By David Mellor, Course Director of Audio Masterclass

Whether an equalizer should be placed before or after a limiter depends on the purpose you are using the limiter for.

Sometimes a limiter is used as an extreme form of compression, for musical purposes. In this case you can place the EQ either before or after the limiter, depending on what you want to hear. Try it both ways and choose whichever you prefer.

However limiters are more normally used when you don't want the signal to exceed a certain level. This would be the case in mastering, and you would use limiters in live sound and broadcasting too.

In addition to the above, you may also want the signal to come up to a predefined level, not merely not to exceed it. In this case you would set the limiting threshold so that the signal frequently triggers the limiting action.

Since equalization can increase the signal level, then it must come before the limiter. If it comes after the limiter, then the signal may go higher than the level you have set.

You may be using EQ to reduce certain bands of frequencies, in which case the signal level will be lowered. Even so, the equalizer should come before the limiter, otherwise there is little point in the limiter being there in the first place.

So, for musical purposes place the EQ where you like. For signal control purposes, the equalizer should go before the limiter.

Publication date: Tuesday August 31, 2010
Author: David Mellor, Course Director of Audio Masterclass

Getting to know Kaossilator 2 -- Part 4 of 4

Wednesday, December 19, 2012

Phantom power - a great breakthrough in microphone technology?

 Capacitor microphones used to come each with their own power supply. Then phantom power was invented so that any number of mics can be powered from the mixing console. So why are some manufacturers returning to the old ways?

By David Mellor, Course Director of Audio Masterclass

If you have a very long memory, you will remember microphones such as the AKG C12 and Neumann U67 - tube microphones that came with their own dedicated power supplies. You plugged the mic into the power supply, the power supply into the mains, and took a feed of the mic signal via the power supply to the mixing console.

Doing this with one or two mics isn't any big deal. But what if you wanted to have twenty mics on an orchestral session? You would soon get tired of all that extra plugging.

But somewhere in Norway, way back in 1966, plans were afoot to change all of this.

Norwegian State Television at the time was already using a 48 V DV powering system for much of its equipment. They wanted to take advantage of this ready supply of voltage to power their capacitor microphones, rather than use a separate power supply for each one.

The Neumann company responded to this with their model KM 84. This is a small FET (field-effect transistor) microphone and was the first to be capable of being phantom powered.

Neumann's phantom power system uses a 48 V DC source and sends this equally to the hot and cold conductors of the audio signal cable, and thence to the microphone. The word 'phantom' is appropriate because you don't see the power supply - it can be contained within the mixing console and requires no additional wiring beyond the signal cable.

The word phantom is also appropriate because unless a microphone is designed to receive such power, there is no voltage difference between the two signal conductors. A microphone that is not designed to use phantom power simply will not notice it is there (as long as it has the usual output transformer).

Phantom power is supplied to the conductors through two well-matched 6800 ohm resistors. In fact one power source can be used, and voltage supplied to many microphones, each through a pair of resistors. This provides protection against short circuits - even if one mic is shorted to ground, only 14 milliamps can flow, which is peanuts to any properly designed power supply.

Once phantom power was accepted by the microphone manufacturing industry, the limitation on the number of microphones employed caused by all those individual power supplies was removed.

The strange fact is that currently there is an increasing number of microphones that are returning to the old ways. In the case of tube microphones, this can be justified - tubes require higher voltages. Also, microphone that are capable of handling very high sound pressure levels can benefit from a higher power supply voltage.

However in the vast majority of cases, phantom power works just fine. It is in fact a brilliant invention - brilliant in its simplicity.
Publication date: Sunday April 11, 2010
Author: David Mellor, Course Director of Audio Masterclass

Getting to know Kaossilator 2 -- Part 3 of 4

Comment of the week: Just how hard is it to 'get into the groove'?

 A RecordProducer.com reader, like so many of us, finds it difficult to get into the groove. Is groove a lost art?

By John Speed

John Speed of Montreal added this comment to our recent discussion on groove, offering additional insight on just how hard it is to 'get into'. Perhaps sequencers and quantization have taken too much of our attention recently...

"I think this is the first time I have heard someone really try to talk about the issue of groove. Lots of musicians say the word but few know how to get there.

I have been working with two professional musicians, bass and percussion, for three years now. The bassist is about 90% technician, a hard worker, perfectionist. We work hours on counting beats and bars and trying to hit the click track or, more so, make it disappear.

I am an intuitive singer songwriter, not a great player but competent. I have learned a lot from this work and my playing has improved but the band doesn't really "work"!!

Over three years of practice and gigs I think we have hit the groove for maybe a total of 3 minutes of playing time. When I play alone I hit it often. I recognize it immediately when it happens but it comes seemingly by chance and we do not seem to know how to get there by design, even though we often discuss the fact that we need to find this magic if the music is really to become acceptable. We owe this to our audience.

I tell myself to just keep working, it takes work, lots of work, and this is the only way there. I would like to hear from others on this subject, I feel like I am missing something very important. Many thanks for your good work."

Further comments on groove will be very welcome.

 Publication date: Thursday March 17, 2011
Author: John Speed

Tuesday, December 18, 2012

Getting to know Kaossilator 2 -- Part 2 of 4

My first self-built loudspeaker (disaster!)

 Everyone should build a loudspeaker at least once in their life. But for this would-be loudspeaker builder, their first attempt was something of a disaster...

By David Mellor, Course Director of Audio Masterclass

There are some things that you just have to do at least once in your life - for example see a total eclipse, perform a parachute jump, and make your own set of loudspeakers. Actually, I might give that parachute jump a miss. But I've done the total eclipse. I remember the immense sense of collective disappointment of the thousands of would-be viewers gathered together at Parc l'Eclipse (really!) near Cherbourg when a cloud obscured the event from view at the last moment. I can also remember the massive sense of disappointment when I heard what my first self-built loudspeaker sounded like.

It was back in the hazy days of the 1970s, when the rule book of loudspeaker design was still some way from completion. It was then the fashion for guitarists to have a 'stack' comprising an amplifier and two 4 x 12 loudspeaker cabinets. I would have liked to have a Marshall stack, just like Jimi, but I couldn't afford one. So I bought eight 12-inch drive units, the cheapest I could find (far from the quality of the Celestion illustrated!), and several sheets of chipboard. I bought the chipboard because it was the cheapest material I could get, but I later learned that it is actually quite a good material for loudspeaker cabinets.

So I sketched out a design for a cabinet that was rather larger than the Marshall equivalent. I don't know why I did that... yes I do ' I just wanted my stack to be bigger than anyone else's! Then over a couple of evenings I put that first cabinet together, covered it with a cheap Rexine imitation, fitted the drive units and handles. Finially I wired up the drive units to the jack socket and screwed on the back. I have to say that it looked great. Of course I couldn't wait to hear it, so I plugged in my amp and guitar and performed my best Pete Townsend power chord.

Er... there was something wrong. It didn't sound good at all. All the drive units were working but the sound was just wrong. Over the next few days I came to the conclusion that I didn't know as much about loudspeakers as I needed to, so I decided to cut my losses and sell the cabinet. So I advertised it at a price that just about covered the cost of the materials. I soon got an enquiry from a local working mens' club, as they were called in those days. Since I'm an honest trader (with 100% positive feedback on eBay, as of writing!) I needed to give them a demonstration so that they could properly consider what they were buying - a speaker that worked but wasn't all that good. So I took the cabinet round to the club and played some music through it. The committee were satisfied and said they would have the pair. "The pair?", I don't know how that happened, but since I had the parts for the other speaker, I realized that if I wanted to get my money back, I would have to make another one.

So we shook hands, although they did take the opportunity of exploiting my youth and innocence and knocked down the price to about two-thirds of cost. I set to work building the other speaker. Once finished, I had to test it of course. And... it sounded great! It was exactly what I wanted a 4 x 12 cabinet to sound like, and I reckon it would have given the Marshall a run for its money. So I looked at the first speaker again to see what had gone wrong. I spotted the error straight away, when hours of looking hadn't helped just a few days ago. I had wired the four drive units in the conventional series-parallel way, but two of the drivers were wired in reverse phase. So effectively at any instant when two of the drivers were pushing at the air, the other two were pulling. I corrected my error, and I now at last I had my brilliant twin 4 x 12 stack. Except I had agreed to the sale and later that day the club sent a van round. I never did make any more 4 x 12 cabinets, but at least I knew that I could if I wanted to.
Publication date: Friday April 10, 2009
Author: David Mellor, Course Director of Audio Masterclass

Getting to know Kaossilator 2 -- Part 1 of 4

Monday, December 17, 2012

The Roland V-Piano Grand - will it put Steinway out of business?

 Roland's V-Piano mimics acoustic instruments with stunning realism. Will the new V-Piano Grand oust the conventional grand piano?

By David Mellor, Course Director of Audio Masterclass

In contrast to my London Olympics tickets that cost me upwards of £150 each and were the cheapest I could get my hands on, my ticket to Roland's V-Piano Grand UK premiere in the Britten Theatre of the Royal College of Music was free. Now that's an offer I couldn't possibly refuse!

I imagine Roland must have my address from the registration card I sent in when I bought my own V-Piano, in its original stage version.

But why would I buy a digital piano? I hate digital pianos. All of them, without exception. I can play a conventional piano for personal pleasure for hours. I even enjoy practising scales (some people enjoy working out at the gym, so why not?)

But a digital piano - well there just isn't any enjoyment to be had. OK, some of them do make a noise of reasonable quality. But they don't feel good to play.

But the one thing that digital pianos do have going for them is practicality. They are more compact, don't need tuning, and you can use them with a MIDI sequencer.

Digital pianos are also easier to record. Of course it is perfectly possible to get a great recording of a good-quality acoustic piano, well maintained and tuned, in a good studio or concert hall. But try doing the same thing with your upright at home. Suddenly recording just became a lot harder.

So although I love playing my Yamaha acoustic pianos (plural, but not at the same time), I wanted a digital piano for recording. I tried them all and bought the best - the Roland V-Piano.

Stage piano to grand piano

Roland describe the V-Piano as a 'stage piano'. It's the kind of thing you would play with a band, like you would once have played a Fender Rhodes.

Turning it into a grand piano however raises a big question...

Why?

The standard V-Piano has all of the advantages of digital pianos I listed earlier, which to summarize boil down to practicality. But a digital grand piano lacks the advantage of compactness and portability. And although a digital grand piano may be easy to record through its line out sockets, a conventional grand piano is easy to record too, in a decent acoustic space.

So, before the event, I wondered to myself what the point of the V-Piano Grand could be. The only answer I could come up with was that I expected it to be superior to a conventional grand in some way.

If you're not familiar with the V-Piano, now is the time to learn that it works by modeling, not samples. And it can model grand pianos, upright pianos, antique pianos. And pianos that don't exist - like pianos with three strings for every note, pianos with silver strings, pianos with a glass soundboard.

So there is some potential here for the V-Piano to be better than a conventional grand. But is that potential fulfilled?

The V-Piano Grand in concert

On seeing the V-Piano Grand in real life for the first time, I found it smaller than I expected. On a concert stage, you expect a piano to be of a certain size. Compared with a Steinway Model D, this was pint-size. OK, quart-size.

Pianist Daniel Tong took the stage and grasped the keys in a masterly fashion. And the sound that came out...

Well it sounded like a piano. But I have to say that although the digital modeling of the V-Piano is wonderful, it sounded like a piano played through loudspeakers. I don't want to over-emphasize this point because it was only a little 'speakery', and for many purposes this would pass unnoticed. I am sure that in a blind test, many listeners would not be able to tell. But in terms of sound quality, there is no way that the V-Piano Grand is better than a conventional piano. Steinway's business model is safe, for the moment. However...

Have a go

At the end of the concert I hung around a little until most of the audience had departed. I and a few others then found an opportunity to hop up on stage and have a go on the V-Piano Grand for ourselves.

First up was a pianist of excellent ability, which gave me the opportunity to walk around the piano, as I would if I were selecting a mic position for a conventional piano. I expected the sound to be localized from the loudspeakers (you can see the grilles), but no, the sound was very full and appeared to come from the whole of the instrument.

I sidled up to the piano stool and took the next spot. I was surprised - the V-Piano Grand is very pleasant to play. This for me is the sticking point for all other digital pianos. They are not nice to play. But the V-Piano Grand plays very much like a conventional piano; the sound is alive and responsive. In terms of playing for pleasure, the V-Piano Grand could make an alternative to a conventional piano. It can't match a large, high-quality, big-name grand perhaps, but it's a contender against less highly-specified models.

Hotel lounge piano?

Having had my go on the ivories, I asked an onlooker what his interest in the V-Piano Grand was. It turned out he was a hotelier and he was interested in having one for his lounge. I'm not so sure that is what Roland had in mind, but a sale is a sale, so if they can convince him on the grounds of practicality, I'm sure he will be pleased.

Finally I had a chat with one of the Roland guys, who made everything clear for me... The purpose of the V-Piano Grand is to be a flagship product. Like car makers who produce a really high-end model in limited quantities. They don't expect to make much of a profit from their ultra-sporty or ultra-luxurious models, but the publicity they can get from them is invaluable, and the new technologies they develop can trickle down to their standard range of products.

Digital pianos are big business for Roland and they have an incredible variety of models - far more than you would expect unless you took a look at the catalog.

So in the future we can expect to see V-Piano technology trickling down to their more affordable models. I'm all for this - the digital modeling in the V-Piano is fantastic. The V-Piano Grand might not knock Steinway off its perch, but it does sound good and brings excellent playability to the digital piano market.

I don't suppose I'm going to buy a V-Piano Grand, but I'm very happy with my V-Piano (non-grand). In fact I think I'll go and practise some scales on it right now...
Publication date: Sunday July 17, 2011
Author: David Mellor, Course Director of Audio Masterclass

Quick Tip Video: Quick Copy And Paste In ACID Pro

Q: Do you have to use a good converter for a microphone to compete with the industry?

 A Record-Producer.com visitor asks whether a good analog-to-digital converter is necessary. Or will any old converter do?

By David Mellor, Course Director of Audio Masterclass

Question from an RP visitor...

"Do you have to use a good converter for a microphone to compete with the industry?"

Firstly, a little Level 1 explanation for newcomers to recording... A microphone needs a preamplifier to bring the level up from a few tens of thousandths of volts to around one volt. Then the signal goes through an analog-to-digital converter so it can be input into a digital audio workstation.

So firstly you have to use a preamplifier that is comparable with those used in the pro industry. That's another matter entirely, so I'll assume that this is already taken care of.

So, do you need a good analog-to-digital converter, which I'll call a converter for short, or will any old converter do?

Well, without doubt it would be nice to have the best converter in the world, that was comparable with the very best that industry pros use. There is no doubt that it is always the right thing to do to aspire to the ultimate standard available.

But what if you can't afford the best? Will your recordings be ruined?

One way of looking at this is to go back into the history of digital audio, back to the early 1980s. The converters they had then were primitive compared to what we use now.

Sometimes they were not even 'monotonic', meaning that an increase in voltage didn't always result in an increase in the digital numbers that came out.

Even so, many great recordings were made with such converters. Recordings that stand the test of time now.

I would venture to bet that even the worst converter that is sold into the pro audio market these days is by far superior to the best that was available then.

So, according to this logic, you don't particularly need to worry about the converter. Yes, buy the best you can afford and couple it with a good preamp. Then forget about it and work on your music, your studio acoustics, your microphone technique and your mixing skills. They will make infinitely more difference than anything else.

Publication date: Monday March 29, 2010
Author: David Mellor, Course Director of Audio Masterclass

Saturday, December 15, 2012

As classic an example of compression pumping as you will ever hear...

One of the potential problems of compression is pumping. And in this example it's about as bad as it gets.

By David Mellor, Course Director of Audio Masterclass
Let's dive right in with the example...




 There are many good features about this video, and it is extremely useful for its intended purpose.
But the audio has a problem. Two problems in fact.

The first is the noise. Clearly the audio is being recorded direct into the camcorder through its built-in mic. If it is a tape-based camcorder, then this could be the reason for the noise.

But also, there is a huge amount of pumping, due to the automatic gain control (AGC) of the camcorder.
When the player hits a note or chord, the AGC kicks in and quickly lowers the gain. But as the notes decay, the AGC relaxes and allows the gain to go up again. The result is that, between notes, the volume swells.

The solution of course is to record the audio separately onto a dedicated audio recording device, then sync it again in a video editing app. Or, if possible, use an external microphone and set the camcorder's gain manually, switching off the AGC.

Of course, either way is more fiddly than straightforward point-and-shoot. Although the results may be better, it might result in fewer videos from this YouTube contributor. Some might say that fewer but better videos would be the way to go. On the other hand, this pianist has set himself a mighty task of recording a lot of music that many people will undoubtedly find useful.

Publication date: Monday October 01, 2012
Author: David Mellor, Course Director of Audio Masterclass

Beethoven: Choral Fantasy / Abbado · Berliner Philharmoniker

Friday, December 14, 2012

Q: What is the right mic for hihats?

  An RP reader wants to capture his hihat in high fidelity. So what is the right mic for the job?

By David Mellor, Course Director of Audio Masterclass

If the question is, "What is the right mic for hihats?", then another question could be, "Right for what?" Everything is subjective in the listening experience, and ultimately it's all about opinions. Or rather, it's about what sells.

So let's start with the textbook answer...

The hihat falls into the class of instrument known as metallic percussion. Bash two pieces of metal together and you are going to get a lot of high frequencies coming out, which you will want to capture accurately.

High frequency sound causes the diaphragm of the microphone to vibrate back and forth rapidly, so you need a mic with a diaphragm that is capable of vibrating very easily at a quick rate.

That therefore rules out the dynamic mic. Dynamic mics have a coil of wire attached to the diaphragm, which makes the diaphragm heavy giving it a certain amount of inertia. Although dynamic mics often have a useful 'presence peak', they are not renowned for crystal clarity at high frequency.

So it's going to be a capacitor mic then. Large or small diaphragm? Tube or transistor?

Let's start with tube or transistor first. The reason for the continued existence of tube microphones is the 'thickened' sound they produce. Who wants a thick hihat? Strike that one.

Now, large diaphragm or small? Well with the large diaphragm we are once again in inertia territory, and it is often thought that the resonance of a large diaphragm, even if well controlled, can smear high frequencies.

So, going through the possibilities, we come to the textbook answer that we should use a small diaphragm capacitor microphone on the hihat, which will have a transistor internal amplifier because that's all that's on the market these days (unless someone knows different?)

But that's the textbook answer. What's the real-world answer?

Well I had the experience a while back of running out of microphones. There were not enough small-diaphragm capacitor microphones to go round and something had to give. What is the instrument, I thought, that least matters if it has the wrong mic?

Aha - the hihat!

So that is where I made my compromise. I can't remember where the small diaphragm capacitor mic went, but on the hihat was a Shure SM58!

Now for the hihat, this is about as un-textbook as you can get. But you know what? It didn't matter. In the mix, no-one would have cared what mic the hihat had, although to be fair it was a rock music arrangement. With other styles of music it might have been a different issue.

Has anyone else had success using the 'wrong' microphone? Or did the 'right' mic ever not perform as expected?

Publication date: Thursday March 31, 2011
Author: David Mellor, Course Director of Audio Masterclass

Lyric Spacing tips in Finale

Sampling rates for sample libraries

 I always set up my projects at 24 bits/44.1 kHz, but recently I noticed how sample libraries are offering 32 bits/96 kHz. Do you think we have to use them?

By David Mellor, Course Director of Audio Masterclass

The answer to this question depends on whether you use sample libraries or create them.

If you are a user of sample libraries then 24/96 is good enough for any purpose. 24/44.1 will sound perfectly OK to 99.9% of potential listeners and the other 0.1% probably wouldn't know unless you told them.

There is however an area where 24/96 can be audibly better than 24/44.1, and that is where samples are used transposed down from their normal pitch. The higher sampling rate will help preserve higher frequencies.

It is worth saying however that sometimes the artefacts of sampling are exactly what is wanted. If, for example, you wanted to achieve a retro 80s sound.

In theory a 24-bit sample should offer a dynamic range of 144 decibels, which is wider than the ear can cope with. A 32-bit sample could in theory extend this to 192 dB. Bigger numbers are always better but, once again, few would actually hear the benefit.

If you produce sample libraries, then it is best to go for the most excellent recording quality possible. That way you will keep up with, or ahead of, the competition. You will win in the numbers game and safeguard yourself as much as possible against further advances in technology.

Publication date: Wednesday June 16, 2010
Author: David Mellor, Course Director of Audio Masterclass

Thursday, December 13, 2012

The simplest version of IO

Why your preamp should have an impedance selector. You're missing out if it doesn't...

 Some microphone and instrument preamplifiers have a variable impedance selector. What kind of difference will it make to your sound? What are you missing if you don't have it?

By David Mellor, Course Director of Audio Masterclass

Why your preamp should have an impedance selector. You're missing out if it doesn't...

Some microphone and instrument preamplifiers have a variable impedance selector. What kind of difference will it make to your sound? What are you missing if you don't have it?

Here is an excellent example of a preamplifier with a variable impedance selector, the Little Labs Multi Z PIP. Before I go further, let me tell you that it costs around $600 - I wouldn't want you building up a desire for one and then finding out you can't afford it!

Yes, it's a glorified DI box, but what glory... there is a level control, which is a bonus compared to most DI boxes, but also there is this all-important variable impedance selector, known here as 'Input Circuit Select'.

What does it do?

To put it simply, if the switch is set to 'hi Z', then the input impedance is high and the unit draws hardly any electric current from the sound source, which I'll assume is a standard electric guitar.

If the switch is set to 'lo Z', then the unit will attempt to draw a large current from the pickups of the guitar.

Now where the difference arises is in the ability of the pickups to supply current. A guitar pickup isn't very good at providing current, so where the hi Z position isn't asking an awful lot, and the pickup is quite comfortably able, the lo Z position demands rather more current than the pickup can successfully supply.

So what happens? Well imagine if the pickup was shorted out by a piece of wire. In this case the Z would be so low as to be zero. In this situation, the pickup will provide no voltage, hence there is nothing to amplify. So a low Z, but not zero, will lower the output voltage.

That doesn't sound good, does it? But what happens in reality is that the pickup is more capable of supplying current at certain frequencies, according to its design, than others. So the loss in voltage is frequency selective.

The hi Z setting, captures the full range of frequencies of the guitar, the mid Z and lo Z positions will capture different frequency responses, in general with less high frequency energy.

Yes, the impedance selector is a kind of EQ control. But it's an EQ that is very dependent on the characteristics of the pickup, and every pickup will react differently.

So being able to select different impedances makes the instrument/preamp combination a kind of a symbiosis, where one reacts to the other to create a unique sonic character - you can't get this with EQ alone.

So with a guitar such as an old-style Stratocaster with a three-position selector switch, the three impedance positions on the Little Labs Multi Z PIP allow a total of nine sound combinations.

So three times the fun. Let's have fun!
Publication date: Monday June 01, 2009
Author: David Mellor, Course Director of Audio Masterclass

Those sticking-out things on the sides of your head - what are they for?

 The pinna of the ear helps to collect sound, obviously. But why does it have those complicated folds? And why do we only have two ears and not three?

By David Mellor, Course Director of Audio Masterclass

For holding your sunglasses on would be one answer. Helping locate, or localize, the direction of a sound source is another. Much of the design of the human body is down to the requirements of living in the wild on the plains of Africa tens of thousands of years ago.

In those days there were priorities other than getting a deal with a record label. First and foremost was personal survival, closely followed by the survival of the species. So on any encounter with something new and unfamiliar there were three possible courses of action... eat it, escape from it, or mate with it!

The ear is adapted by evolution to help us survive, by pinpointing sources of danger, and by helping us communicate. So the pinna, which is the proper term for the ear 'lug', helps collect and localize sounds. The shape of the pinna, like a miniature satellite dish, collects sound over a relatively large area and funnels it into the auditory canal to the ear drum. At the same time, the body of air it partially encloses resonates at frequencies around 3.5 kHz, thus acoustically 'amplifying' these frequencies.

This helps us hear speech more effectively - not the fundamental tones but the sounds that mark the differences between words.

The main localization function is provided by the fact that we have two ears. But the pinna also helps. By being forward-facing it helps us differentiate between front and back, which the mere fact of having two ears does not. Also, the complex folds cause reflections that subtly boost and cut different frequencies according to the height of a sound source.

So although we do not have any specific detector for the height of a sound source, we can in fact localize quite well in the vertical dimension. So you can tell instantly whether that strange sound you hear is a rat nibbling at your boot laces, or a bird about to dive bomb you with its payload.

Which reminds me of a joke. Did you know that Captain James T. Kirk of the Starship Enterprise did in fact have three ears? His left ear, his right ear, and his final front ear! (Final frontier -- get it??)
Publication date: Sunday February 27, 2011
Author: David Mellor, Course Director of Audio Masterclass

Beethoven: Missa solemnis / Blomstedt · Chor des Bayerischen Rundfunks · Berliner Philharmoniker

Wednesday, December 12, 2012

Q: What is a buss?

I have heard that mixing consoles have busses, but I can't see anywhere I can control or operate a buss. Could you tell me please what a buss is?

By David Mellor, Course Director of Audio Masterclass

In audio we use the words 'bus' and 'buss'. Either spelling is acceptable, however for this answer we will use 'bus' to mean the road vehicle; 'buss' to mean the component of a mixing console that is under discussion.

Think of a road bus at the outer end of its route in the suburbs of a city. Initially it is empty, but as it wends its way to the city's heart, it picks up passengers at every stop. Eventually it gets to its destination and everyone gets off.

A mixing console buss is similar (in an analog console). It is a wire or metal rod that starts at Channel 1 at the left of the console and ends up at a group output or master output on the right. Let's say that we are considering the buss for Group Output 1.

As it traverses the console, it picks up signals from any channels that are routed to that buss, just like the road bus picks up passengers.

When the buss arrives at Group 1, it delivers all of its signals.

Of course, you can only take an analogy so far. The buss in a mixing console doesn't move, and it doesn't return anything back to the channels (there is no reverse direction to its route).

An analog mixing console has one buss per group, plus one buss for each of the master outputs, plus one buss for each auxiliary send that it has.

Digital mixing consoles mimic the busses of analog consoles. Digital audio workstations also employ the same concept.

In conclusion, wherever audio signals are mixed together, in either the analog or digital domains, you will find busses.

Publication date: Tuesday June 15, 2010
Author: David Mellor, Course Director of Audio Masterclass

How To Delete Only Certain Items in Finale

How to make a Microphone Pop Filter for under $10.00

The video is introduced by explaining what a pop filter is and why a person who records podcasts should build one. At 1:30 the instructor introduces the materials needed to make the filter. At 2:00 he begins instructing viewers on how to make the microphone filter beginning with the treatment of the nylon...

By YouTube Genie, video courtesy YouTube
The video is introduced by explaining what a pop filter is and why a person who records podcasts should build one. At 1:30 the instructor introduces the materials needed to make the filter. At 2:00 he begins instructing viewers on how to make the microphone filter beginning with the treatment of the nylon.


At 4:00, the instructor shows viewers how to assemble the nylon and the embroidery hoop. He cautions people who are working alone that they must be clever in stretching the nylon tightly over the hoop without an extra pair of hands for help. At 8:05, once the filter is in place, the instructor begins to talk about the quality of recording vocals.

If a person speaks nearer to the microphone, his or her voice sounds deeper. When speaking further from the microphone more of the high notes are caught by that transmitted resulting in a higher pitched sound. Ultimately, the filter is able to prevent the hisses and pops that occur with a naked microphone. Often times, though the speaker is unaware of a problem when he or she listens back to what has been recorded, every p, t and s sound stands out with a distinct popping or hissing.

Expensive recording tools can prevent this problem, but if the podcast is only a hobby, the recorder might not want to spend too much money on a solution to the popping. To make a microphone pop filter for under ten dollars, all that is needed is a four-inch, wooden, embroidery hoop, a one foot length of slip tubing, a ten inch by two inch moldable gauge, a ratcheting zip tie a single black nylon, clip pliers, scissors and a screw driver.

This cheap solution will resolve every pop and hiss making the final recorded podcast sound professionally made. The assembly of the filter takes under ten minutes. To begin, the builder should take the nylon and cutting the toe away, then the leg. The remaining nylon tube should be cut lengthwise and laid flat. The length of the nylon will be roughly twice the breadth. Folding it once over will make a square and provide a double barrier.

The builder should then take the nylon, folded in half and stretch it over the small round of the embroidery hoop. Once it is stretched tight, the larger round of the hoop should be slipped over the nylon on the small round and tightened with the screw fastener. After the hoop is assembled with the nylon stretched tightly over it, the excess fabric should be cut away. Taking the slip tubing and fastening it to the base of the microphone stand, the builder provides an anchor point for the hoop.

The tubing is best positioned roughly two inches from the microphone at a obtuse angle to the mesh. Once the tubing is in place, the hoop can be fastened to the area just below the microphone using the zip tie. The zip tie can securely join with the tubing by lining up the screw fastener and end of the tube. The excess on the zip tie should be cut away. For a professional appearance the moldable gauge can be slipped over the tubing to make a uniform black pop filter, once assembled a podcast can be recorded free of hissing and popping sounds.

Publication date: Friday December 31, 2010
Author: YouTube Genie, video courtesy YouTube

Tuesday, December 11, 2012

How To Hear Garritan Percussion Sounds in Finale

Why are the mics we use on guitar cabs so obviously the wrong shape?

 Whoever designs microphones clearly thinks that they all should be microphone-shaped. But isn't it about time we had something more appropriate to the way we use them?

By David Mellor, Course Director of Audio Masterclass

Think of a microphone.

OK, I'll tell you the image that's in your mind. It's either a thin near-cylindrical end-address microphone, or a fat near-cylindrical side-address mic.

You're thinking of a Shure SM57 or a Neumann U87!

Well, something like that.

The mics we use today are clearly descended from those designed in the 1940s and 1950s. And the classic shapes of the 'pencil mic' and 'bottle mic' are without doubt the most popular. Alternatives are very thin on the studio floor, or live stage.

I was watching some old concert footage on TV the other day and I noticed the mics on the guitar cabs.

They were clearly the Shure SM57 model, and they were simply hanging down in front of the cabs by their cables.

Of course, this can be seen as the lazy man's way to mic a cab. Or perhaps there weren't enough stands available.

But clearly when used in this way, an end-address mic is pointing in the wrong direction.

The problem wouldn't arise if we had microphones that were specifically designed for guitar cab miking.

If, for instance, a mic could be made in a rectangular shape with a side-address diaphragm, it could dangle from the cable with the diaphragm pointing at the speaker cone.

Or perhaps a new type of stand could be devised to fit onto the cabinet, rather than taking up floor space. Surely that would be better than using a conventional stand.

Perhaps the mic and 'cab-stand' could be one integrated structure.

Sometimes I wonder whether we are too set in our ways and we need a few 'crazy' ideas to refresh the process of making music.

Having mics that differ from the conventional forms could be a start.
Publication date: Friday February 12, 2010
Author: David Mellor, Course Director of Audio Masterclass

Handel: Concerto grosso in G / Haïm · Berliner Philharmoniker

Monday, December 10, 2012

How to keep your studio business open by recognizing the client life cycle

 If you have a studio business, doubtless you want it to continue for decades to come. But if you don't understand the client life cycle then you'll soon be in trouble.

By David Mellor, Course Director of Audio Masterclass

If you have a recording studio business, then congratulations to you! It's a hard trick to pull off successfully when everyone has a pro-quality DAW at home.

But there are some types of recording that you really do need a professional environment for, with soundproofing and acoustic treatment too.

Having a studio is not the same as keeping a studio. Many businesses, of all kinds, do not fully appreciate the client life cycle.

When your business is new, you will spend a lot of time attracting clients. Some of those clients will be one-offs, others will come to appreciate the quality of what you do and come back again and again.

Gradually you will acquire more and more regular clients and they will provide most of your income.

You might come to view one-off clients as a bit of a nuisance. They don't understand how you work, they ask for unusual things, you're not sure whether they will pay on time etc.

Eventually you might find yourself working only with your group of regular clients.

Now let's look at things from the client's point of view...

Every client is a new client at the beginning of their relationship with your business. This is the 'birth' end of the client life cycle.

Eventually the client will become mature and will use your services again and again. It's win-win because you get regular income and the client gets a service of known and repeatable quality.

But eventually the client will 'age'. They might outgrow the service you can provide and be forced to go elsewhere. They might find a cheaper solution. They might go broke or otherwise go out of business. They might actually retire. This is the 'death' part of the client life cycle.

So if your business relies on a group of regular clients in the mature phase of their life cycle, you cannot expect this to go on forever. One by one they will drop off and die.

And now you are faced with the unfamiliar situation of having to attract new clients - something you perhaps have not done for years.

The solution, for any business, is never to rely on regular custom. You have to attract new clients all the time. There should be a regular cycle of birth-maturity-death among clients and you should expect and welcome it.

Many studio businesses have failed over the last ten years or so. In many cases this could be because they haven't understood the client life cycle properly.

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

IO In action Sequence Segment

What is a 'natural sound' in audio?

 Do your recordings sound natural? Or do they sound 'microphony', electronic or digitally processed? How can you tell?

By David Mellor, Course Director of Audio Masterclass

The other day, I found myself advising someone that their recording of speech was good but it didn't sound natural. I further advised that the sound quality they had achieved was commonly heard on the radio, but it would be tiring to listen to for a long period, if the recording was part of an audio book for example.

It's worth reflecting for a moment on what we would consider naturalness, in a recording, to be. Fortunately we have examples of natural sound around us all the time, so there is plenty of material for comparison.

Perhaps the most useful natural sound is the human voice. Our ears are very closely attuned to the sound of the voice and we hear it all the time, and - most importantly - pay close attention to it. Of course I do mean the human voice as produced from a human larynx, throat and mouth, traveling directly through the air to your ears, not via a loudspeaker.

Let's consider therefore how we can compare the natural human voice with the sound of the voice reproduced via a loudspeaker.

Firstly, the person we choose to provide our hypothetical example of human speech should have a reasonably normal quality of voice. Professor Stephen Hawking writes excellent books on cosmology, but he isn't going to make a good example. Neither would a 40-a-day smoker. But we don't have to be too choosy. Apart from a few wayward examples, almost anyone would do.

Now we have to consider context. Should we consider the example of a friend spotting you from the other side of a busy road and shouting you a greeting? Well we could, but it doesn't have a lot of commonality with anything we would be likely to do in audio.

What about a lover whispering sweet nothings into your ear? That might be a desirable scenario, but the sound of the voice at extremely close range is difficult to mimic accurately. It's an interesting challenge, but we need something simpler.

So what about someone talking to you in a normal voice from a distance of two meters? That's just over six US feet.

This is a good test because it is a commonplace situation with which we are all very familiar. Also, it is practical to simulate with audio equipment. Bear in mind that most loudspeakers have at least two drive units and a certain amount of distance is required to allow the sound to integrate. A distance of one meter wouldn't be enough as small changes in listening position produce significant changes in perceived sound quality (and that is something to consider when using near-field monitors).

So imagine this... There is a visually opaque but acoustically transparent curtain in front of you, behind which there is a person, ready to speak from a prepared script (or you could do it in the dark). And also there is a loudspeaker, mounted with its central axis at the same height as your volunteer's mouth and as close as possible to one side. Through this loudspeaker will be played a recording that this same person made earlier. An assistant has previously checked that everything is working and that the levels are very similar.

So now you hear a voice. Is it human or is it the loudspeaker? Now you hear another voice. Or is it another voice? Is it perhaps the same sound source? Or has the source changed but there is so little difference that you can't tell?

You could carry out this experiment for real. Or you could consider it to be a test of naturalness in audio, and have this thought in your mind next time you need a recording to sound natural. Listen to your recording and ask yourself whether you would be fooled.

Although the human voice is the supreme test of naturalness in audio, it is also worth considering whether your recordings of acoustic instruments, including drums, sound natural. And if they don't sound natural, should you be trying to get closer to a natural sound, or are you trying to improve on nature?

Of course, naturalness isn't always the requirement. But it is a very useful benchmark of audio quality. Listen to your recordings closely and ask yourself which aspects don't sound natural. And whatever doesn't sound natural, ask yourself whether it is a defect, or an improvement.

Publication date: Monday May 23, 2011
Author: David Mellor, Course Director of Audio Masterclass

Saturday, December 8, 2012

IO Instant Orchestra Multi Making

Q: How can I make my drums and my bass guitar sound heavy?

 My drums and bass guitar sound loud and heavy when I play them, but they don't sound heavy enough when I make a recording. How can I make them heavier?

By David Mellor, Course Director of Audio Masterclass

Ah... you're suffering from lightweight drums and bass syndrome! This is very common in recording.

Let's look at the bass guitar because this is more likely to be suffering from this problem.

The principle cause of LBGS (lightweight bass guitar syndrome) is the way the ear perceives sound. When you plug your bass guitar into an amplifier and speaker and turn up the volume, your ear doesn't only perceive loudness, it interprets that loudness as heaviness too.

The microphone doesn't. It picks up the sound exactly as it is - merely loud.

When you set the gain correctly on the preamp, the resulting recording will be hardly any heavier than if you had set the volume control on the amp to 1.

Indeed, it might have no apparent heaviness at all.

So what's the cure?

The solution to this problem is to recognize that loud sound stresses the ear and causes distortion in the hearing process. So in other words, you're not hearing the sound as it actually exists, you are hearing your ears' interpretation of the sound.

Having realized that, you can begin to restore lost heaviness by adding distortion into the process.

This is best done at source. So if you have a powerful, clean bass amp, you need to exchange it for one that creates more distortion. So you need tubes rather than transistors, and low-power rather than high-power, so that when you turn the amp up, it distorts more.

Your speakers too might need attention. Speaker technology has 'improved' since the 1960s and it is possible to design and build drive units that are ultra clean at high sound levels.

But this doesn't produce a heavy sound in a recording. It is better to use drive units of 'old school' design where the cone bends more and produces a more distorted, but heavier, sound.

You might choose not to use an amplifier and speaker at all, and record through an amp simulator, or use an amp modeling plug-in.

Amp modeling is an amazing technique that can mimic the sound of real amps and speakers. The problem can be that although the modeling seems good, the result is lightweight.

Here you can improve the sound by putting the modeled signal through an amp and speaker, which could be your monitor system with appropriate settings of the controls. Mic this from a distance of a meter or more so that you pick up some of the ambience of the room.

You can use this alone, or mix it in with the original modeled signal. If you mix it, consider time aligning the two signals so that you don't get cancelation of some frequencies.

This technique will require a lot of experimentation to get the sound exactly right. But what you will end up with can potentially blend clarity and heaviness in exactly the right proportion.

Publication date: Tuesday June 29, 2010
Author: David Mellor, Course Director of Audio Masterclass