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Equipping Your Home Recording Studio

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Equipping Your Home Recording Studio Page 

Cover photo: AKG C414B XLII microphone

Equipping Your Home Recording Studio Page 

Equipping Your Home Recording Studio
© Audio Masterclass MMVIII

Equipping Your Home Recording Studio Page 


It is the dream of many musicians to have a recording studio in their own

home. Not so long ago this was far too expensive even to be a remote
possibility. But now, equipment and software is available at extraordinarily
reasonable prices, and for an outlay much less than a decent secondhand
car, you can have a home recording set up that is capable of fully
professional results.

The home recording studio equipment market is now quite mature. The
manufacturers know very well what they are doing and they make products
that will achieve good results for you, almost without exception. Indeed, it
is difficult to find a product these days that is actually bad. Even so, there
are ways of spending your money wisely and there are ways of spending
your money foolishly, and it’s not just a question of wasting money, it’s a
question of using your energy sensibly as well.

Ideally what you should do is take a short time getting your studio together
and getting it ready for work, and then you should just create music in it
- beautiful music.

Unfortunately, what happens with a lot of people is that they start to set up
their studio and somehow they never quite finish. There always seems to
be something that needs changing, something that needs improving - some
equipment that they haven’t got yet and they can’t make brilliant recordings
without it. Somehow they go on and on like this possibly for months,
sometimes literally for years. And in that time they never get round to
making any serious recordings. That is sad indeed.

Equipping Your Home Recording Studio Page 

The challenge is to work out exactly what equipment you need for the
type of work you want to do. Assess the equipment that is available on the
market and decide which are the ideal items for you. And the best way of
doing this is by having an understanding of what the equipment offers;
what the pros and cons are likely to be. If you read a recording magazine
and look at the glossy adverts for equipment, many of the manufacturers
will try to sell you their products on the basis of their ‘feelgood factor’.
Does the product look glamorous? Does it look professional? Will you feel
that you are a better person if you buy it and put it in your studio? They
tend not to sell the products so much on what they can actually do.

So that is the purpose of this book. To help you understand the differences
between the various types of products that are on offer and to help you
make an informed decision based on your own personal needs. We’re not
going to say ‘buy this particular piece of equipment’ because that’s not the
right way to do it.

That piece of equipment might be the best on the market, but there might
be a better one available tomorrow. That piece of equipment might be better
for one person but another piece of equipment might be better for someone

So when you read this book you will rarely see the names of manufacturers
or products. We have tried to avoid that as much as possible so that you
can absorb the knowledge in each chapter of the book, understand the
equipment that is on offer in each category, then you can see for yourself
what the pros and cons in those various pieces of equipment are. There

Equipping Your Home Recording Studio Page 

are cases however where it is virtually impossible to avoid naming certain
manufacturers and where that is important we haven’t hesitated to do so. So
let’s get started...

At the end of each chapter of Equipping Your Home Recording Studio you
should pause for thought and ask yourself how much knowledge you have
absorbed from that chapter.

Remember that buying a book doesn’t automatically put the knowledge

that book contains directly into your head. You have to take the time and
trouble, and spend the energy to absorb it fully. When you have read
and absorbed this book fully you will find that you are able to cut a clear
path through the jungle of confusion of all the vast variety of recording
equipment that is available today. You will set up your studio knowing that
the equipment you have bought is right for you and right for your budget.
You will get your studio up and running in a short period of time, and from
that point you will start to record. You won’t worry about upgrading your
studio because you’ll know that you have the right equipment already. You
will record day after day after day.

Each time you record you will gain more experience and within a short
time you will become the master of your own home recording studio. And
you will have the satisfaction of producing recordings that are professional
in quality, that please you, and have the potential for success in the market
place for music.

You will look at other people who are still struggling setting up their
studios, constantly tweaking and upgrading them and never getting round

Equipping Your Home Recording Studio Page 

to recording, and you will feel sorry for them. You will be the one that is
producing the goods. Read on…

Equipping Your Home Recording Studio Page 

2 Important information on how to use this e-book
3 Viewing this e-book
3 Copyright information
6 Introduction
12 Equipping Your Home Recording Studio
14 How to save money on your studio
18 Digital audio workstation software
20 How to choose digital audio workstation software
22 Digital audio workstation software features
22 Audio interfaces and Pro Tools
25 Plug-in and software instrument compatibility
29 Digital audio workstation websites
30 Choosing a computer
34 Computers and noise
35 Audio Interfaces
44 Audio interface websites
45 Plug-ins
46 EQ
49 Compression
51 Delay
53 Reverb
55 Pitch changing and correction
58 Mastering

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65 Software synthesizers
73 Software sampler websites
74 Really expensive sample libraries
76 Copy protection
77 Microphones
97 Directional characteristics of microphones
101 Microphone stands
102 Microphone Preamplifiers
110 Control Surfaces
115 External Hardware
120 Standalone workstations
130 Monitoring
149 Shopping list
152 Conclusion

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Equipping Your Home Recording Studio
Thousands of people have home recording studios and you are about to join
their number. How did they choose their equipment? Some went into music
stores and took the guidance of a sales person. Sales people in music stores
often have home studios too, but simply having a home studio doesn’t
mean that you are properly qualified to offer advice. Stores have to make
profits and what the sales person offers you may be the item that generates
most profit – or commission – and might not be the best for you. Indeed
the store might not even carry the best products among their stock. Top
manufacturers can sometimes be choosy about who stocks their products,
so you might be completely unaware of the products that really would
perform best for you.

Many home studio owners buy their equipment mail-order, influenced by

the advertising they see in home recording magazines. Clearly we want the
equipment manufacturers to enjoy a healthy level of business and continue
to bring out new products for us to enjoy. But the person who takes too
much notice of the advertising in recording magazines will spend a lot
more money than is necessary to build an effective home studio.

Reviews in recording magazines can be a great source of advice on what

to buy and what not to buy. But remember that a magazine’s profit comes
from the advertising. The price you pay just about covers printing and
distribution. So dare any magazine annoy their advertisers by printing
bad reviews? Actually some magazines won’t review bad products and
few reviewers would risk damaging their reputation by recommending

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bad products. Magazine reviews, therefore, can often be a good source of
information, but remember who is paying the bills.

But there is an even better way of choosing home recording studio

equipment. And that is to understand what the equipment does, how it
works, what it might do well and what it might do badly. If you understand,
then you are in control. And when you receive recommendations or read
reviews, you will be well able to assess their worth and act accordingly.
And that is the whole point of this book – to allow you to make informed
purchase decisions yourself. Having read this book, you won’t need anyone
else to tell you what to buy. You will know yourself what is best – for your
style of music, for the way you like to work, and of course for your budget.

Equipping Your Home Recording Studio Page 13

How to save money on your studio

You will avoid a lot of unnecessary expense on your studio if you realize
two basic facts:

• You need equipment of basic professional quality, not exotic


• When you have equipment of basic professional quality, the standard

of your recordings is totally down to you.

This book is written from vast experience of more than three decades of
recording. During that time it has been seen again and again that many,
many home recording enthusiasts are living under the delusion that what
they need to bring professional quality to their recordings is that elusive
‘next’ piece of equipment or software. And once acquired, perhaps at
considerable expense, there will be another new ‘next’ item that is essential
to get professional results.

It is amazing how so many people can be so wrong, but thankfully they

keep the manufacturers in business and keep prices down for the rest of
us. All you need to achieve professional recordings is equipment of basic
professional quality. Experienced and skilled engineers don’t need anything
more. They might like exotic and expensive audio toys, but they don’t need
them. And neither do you.

Another way people waste money is by continually upgrading their

recording system. The problem with this is that you are always trying to
hit a moving target. There always will be another upgrade available and

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every time you change one piece of equipment for another, you have to
learn it, familiarize yourself with it and then become expert in using it. By
the time you have done that, yet another upgrade will be the cover story in
the recording magazines. Even well known artists and bands can fall prey
to ‘upgrade-itis’. They struggle for years, putting their music together on
whatever equipment they can get hold of. Finally they land a deal and a
label licenses their recordings. The band then spends its advance money
on a state-of-the-art recording facility. But unfortunately the magic is lost
and they never have another hit. They needed to show some respect to the
equipment that actually did bring them success!

So, the moral to this is that you should buy wisely and aim to keep every
item you buy for a long time. Some equipment will last for decades with
minimal care and attention. A prime example of this the microphone. If
you buy a microphone today you should EXPECT that it is still working
perfectly in twenty years. So you really ought to buy a microphone with
a really good sound quality that you like. If you buy a microphone just to
get you by until you can afford a better one, well in twenty years’ time it
will still be reminding you of your unwise decision. And of course you will
be able to buy a better microphone now because of savings you will make
elsewhere through reading this book.

Some equipment and software just has to be upgraded, however. One prime
example is your computer and the software it runs. You always have to
upgrade that sooner or later, don’t you?

Let’s look at the computer/software problem from a slightly different

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angle. Let’s suppose you run a large broadcasting operation with dozens
of studios. Among them is a small audio editing suite with a computer and
software fit for the task. A number of engineers use that studio and you
know that you are getting value for your investment when that studio is
running for as many hours a day as possible, churning out programming.
Are you with the story so far?

Now let’s suppose that the software company updates their product from
version 7.4 to version 7.5. So a maintenance technician, who hasn’t
been working for you for very long, installs the update. But suddenly
all the menus are different; the shortcut commands have changed. It’s
incompatible with your other software on the same computer and it won’t
talk to the network. So, despite the fact that the update might have fixed
some bugs and offered one or two improvements, your studio has gone
from being a well-oiled machine to an unproductive source of nothing but

No, the best way to work is to decide which software and computer
you need to work most effectively, then install it and iron out any initial
problems. And once it is working, DON’T CHANGE ANYTHING. Once
the studio is up and running, the old saying of ‘If it ain’t broke, don’t
fix it’ surely applies. Wait until there have been significant advances in
technology and then renew, or consider renewing, EVERYTHING.

Clearly this makes sense in a professional working environment, and it

makes sense in the home recording studio too. You should buy the software
and computer you need and get it working - then just use it. Day-in, day-

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out, churning out the music. If you don’t tinker with it, you won’t have any
problems. And the sheer usability you will get from a reliable system that
you can trust will far outweigh any small benefit from minor upgrades.
And when you have worn it out, or there really is something better on the
market, change the whole lot. A three year cycle wouldn’t be too long.

In summary so far, this book is going to show you how to equip your studio
so that it is effective, doesn’t cost more than it needs to, and lasts you a
long time. Store assistants and magazine advertisements want to bleed you
dry of your money and then expect you to carry on spending. ‘Equipping
Your Home Recording Studio’ wants you to buy right then concentrate all
of your energies on recording!

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Digital audio workstation software

Your recording system will be at the heart of your home recording

studio, so that is where we shall start. Notice that we have started with
the software, not the computer. This is always the right order in which to
make your decisions. Your digital audio workstation software will be a key
component of your studio. The computer is merely a platform on which to
run that software.


There are two types of data stream that you may want to record in your
studio. One is audio, of course, and the other is MIDI. MIDI needs a little

Sound sources will be covered in detail later, but let’s skip forward for a
moment and consider your musical keyboard. You can play the instrument
directly and, assuming it is connected to an amplifier and loudspeaker, it
will make sweet music which you could record as a audio signal. Now,
if you look around the back of the instrument, you will see that it has a
socket called ‘MIDI OUT’. From this socket comes a stream of data that
represents the keys pressed on the musical keyboard, not the audio. In
fact you could run a cable from the MIDI OUT socket to the MIDI IN of
another keyboard and the same notes would sound from its audio output.
And you can record this data, or use it to play a ‘software instrument’ that
runs inside the computer.

The reason you might want to record MIDI is that because it records

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only key press and related data. You can change the type of sound that
is produced. So you can record a MIDI track connected to a software
instrument creating the sound of a grand piano. Then later you can change
the sound to an electric piano without having to re-record. You can also
easily edit the timing or pitch of the individual notes that you played.

MIDI is useful for synthesized and sampled sounds, nothing else. So real
guitars, drums, strings, woodwind and brass instruments must be recorded
in audio. Indeed many home recording studio owners have little or no use
for MIDI. How important or otherwise MIDI is to you may impact on your
choice of digital audio workstation software.

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How to choose digital audio workstation software

There are a number of digital audio workstation softwares on the market.

One, however, deserves special attention because it is far and away the
most popular in the professional industry. That is Digidesign Pro Tools. The
reason for the success of Pro Tools with professionals is that it has always
been simple to use. Professionals hate unnecessary complexity. Some
professional equipment is indeed complex, but that is because it has to be to
do its job. Otherwise simplicity wins every time because the user has more
brain-power to spare to concentrate on the music, which of course is the
whole point of the exercise. Other software developers battled feature-for-
feature in the home studio market and didn’t notice – or perhaps care – that
Pro Tools was racing ahead with the professionals.

Now one thing that is vitally important to remember is that although Pro
Tools is virtually the industry standard, that doesn’t necessarily mean it is
the best, or the best for your application. It is useful to make an analogy
with graphics software here. If someone writing a book called ‘Equipping
Your Graphics Studio’ did not prioritize Adobe Photoshop, they would be
making a big mistake. Photoshop is massively dominant in that industry.
Anyone who wanted to enter the industry without a knowledge of
Photoshop would find themselves left out in the cold, although they could
run a one-person business quite happily with other graphics softwares.

And so it is with Pro Tools. If you make a recording with Pro Tools (called
a ‘session’) you will be able to take it to just about any professional studio
and play it on their Pro Tools rig. If you make a recording with any other

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software, you will come across studios with compatible software far less

The message has to be very clear – if you want to be as compatible as

possible with the professional recording industry, then you must use Pro
Tools. If this is not important to you then you are free to choose any other
software that meets your needs.

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Digital audio workstation software features

We can take it for granted that digital audio workstation software sold into
the pro audio market can record, play, edit and mix effectively. In terms of
their functions, there is very little difference in what the major softwares
can achieve. The same applies to sound quality. All of the major digital
audio workstation softwares work close to the limits of human hearing. Any
deficiencies are so absolutely tiny that they are not worth bothering about.
Other matters are more important.

Audio interfaces and Pro Tools

All digital audio workstation softwares require an audio interface to

connect to the outside world. Yes, computers mostly do have audio inputs
and outputs but they are rarely of good enough quality for effective
recording. We will look at audio interfaces in more detail later. However,
there is an important point that must be made here. If you choose
Digidesign Pro Tools software, then you have to use a Digidesign audio
interface (or a compatible interface made by M-Audio. Digidesign and
M-Audio are both owned by the same parent company). Any other digital
audio workstation software sold into the pro audio market can, with very
few exceptions, work with any audio interface.

The question of why this is so is bound to arise. The answer is that

Digidesign is dominant in the digital audio workstation software market
and they can force users of their software to buy their audio interfaces. You
may or may not find this morally repugnant. However, the fact remains
that Digidesign’s software and hardware both work very well indeed and

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demand consideration. There is also something to be said for the additional
security provided by having the software and hardware come from the same
company – you can be sure that the combination will work as advertised.

Pro Tools LE (the nominally ‘lite’ but still incredibly feature packed version
of Pro Tools) together with any of the interfaces in the Digidesign Mbox
range is not expensive compared with other software/hardware options. It
is however limited in terms of the number of inputs and outputs it provides.
If you need more inputs or outputs than the Mbox (or M-Audio) range
provides, then you will have to turn to the massively more expensive Pro
Tools HD system, or buy a different digital audio workstation software that
can work with almost any interface.

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Digidesign Mbox Pro audio interface, front and rear

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Plug-in and software instrument compatibility

Another important consideration when choosing digital audio workstation

software is plug-in compatibility. Plug-ins provide processes such as
equalization and compression, and effects such as delay and reverberation.
All digital audio workstation software comes with basic plug-ins or
equivalent functionality included. But it won’t be long before you want to
explore the amazing universe of plug-ins that is available. And, of course,
you may want to use software instruments.

Plug-ins and software instruments come from two types of source.

One source is the developers of the digital audio workstation softwares
themselves. The other is third-party software developers who do not
themselves sell digital audio workstation software. Third-party developers
want to maximize their sales so, in general, they make their products
compatible with all digital audio workstation software.

Digital audio workstation software developers, however, want to maximize

the attractiveness of their major product line, which is the digital audio
workstation software itself. So they provide plug-ins and software
instruments that can only be used with their own digital audio workstation
software. Fortunately the range of third-party plug-ins that is now available
almost makes this irrelevant, but it is still worth bearing in mind when
making your selection.

There are a number of plug-ins and software instrument formats. There

are a number of digital audio workstation softwares sold into the pro audio
market. There are two popular computer operating systems. It’s time for a

Equipping Your Home Recording Studio Page 25

run down of all the possibilities.

The digital audio workstation softwares you are most likely to come across
are these: Apple Logic Pro, Cakewalk Sonar, Digidesign Pro Tools LE,
Mark of the Unicorn (MOTU) Digital Performer and Steinberg Cubase. As
was said earlier, all of these softwares are capable of recording, playback,
editing and mixing to a fully professional standard. You will make your
purchase decision on industry relevance, audio interface compatibility,
plug-ins and software instrument compatibility, and of course ease of use.

Steinberg’s Cubase software uses its own VST (Virtual Studio Technology)
plug-in format and VSTi software instruments. Cubase is available for
Windows PCs and also the Macintosh. VST plug-ins and VSTi instruments
are specific to either the PC or the Macintosh. Software developers,
however, commonly produce both versions, although you should not
assume this is so.

Cakewalk Sonar uses the DXi plug-in format on the PC, which is based on
a technology developed by Microsoft. It is also possible to use VST plug-
ins and VSTi instruments using a ‘wrapper’ or ‘shell’ that translates from
one format to the other. Although wrapper technology can be very effective,
it has to be noted that it imposes an additional layer of complexity. When
software is updated or upgraded, for example, it isn’t uncommon for plug-
ins not to function until they are updated too. Using a wrapper increases the
possibility of this happening.

Digital Performer only runs on Macintosh computers and originally favored

the MAS (MOTU Audio System) plug-in format. Digital Performer can

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also use VST(i) and Apple’s AU (Audio Units) formats.

Digidesign’s Pro Tools LE uses their own RTAS format - ‘RTAS’ standing
for ‘Real Time Audio Suite’. Pro Tools LE runs on either the PC or the
Macintosh and RTAS plug-ins are generally available in both formats.
Pro Tools LE can also use VST(i) plug-ins inside a wrapper, as mentioned
earlier. Once again, however, this increases the possibility of plug-ins not
working properly, particularly after a software update.

Logic Pro belongs to Apple and therefore only runs on Macintosh

computers. It uses Apple’s own AU (Audio Units) plug-in format.

The range of plug-ins and software instrument technologies can be baffling,

particularly when wrappers are also taken into account. Fortunately a lot of
heat is taken out of the decision-making process by the fact that third-party
developers commonly make their products available in all of the formats. It
is the digital audio workstation software developers themselves who restrict
access to their own plug-ins and instruments, in the hope that you will find
their products more attractive.

Final considerations on choosing of digital audio workstation software...

As has been said earlier, all of the major digital audio workstation softwares
can fulfill the essential functions. They do differ, however, in the details
of their advanced or specialized features. When you are new to recording,
then it is the basic features that are important. You won’t know which
advanced or specialized features you need until you actually do develop a
need for them. And it is impossible to predict ahead of time which will be

Equipping Your Home Recording Studio Page 27

most important to you. The good news, however, is that all of the major
digital audio workstation softwares are fully capable of producing work to
professional standards. So you can choose any of those mentioned and be
confident that you will not be held back in any way.

There are some further points that might influence your decision. Price
is one of course, which your can check with your dealer. Community is
another – Sonar and Digital Performer have strong user communities in
North America. Cubase is probably stronger in Europe.

But there is one factor that is surprisingly important. Look at the screen
shots of the various softwares, which are commonly available on
developers’ websites. Now just think that you will be looking at the same
screen every day you are in the studio. Once you have made your choice,
it isn’t at all unlikely that you will stick with the same digital audio
workstation software over a period of ten years or more, through its various
updates and upgrades. This indicates that it is extremely important that
you choose a software that looks ‘right’ to you. Preferences in this respect
can be very individual. Your decision should be based on the clarity and
elegance of the screen layout as you perceive it. Remember... every day you
spend in the studio for the next ten years or more…

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Digital audio workstation websites (Logic Pro)

www.cakewalk/com (Sonar) (Pro Tools LE) (Digital Performer) (Cubase)

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Choosing a computer

Once you have chosen your digital audio workstation software, the next
step is to choose a computer. Unfortunately many people choose the other
way round, in which case this might rule out the software that would
have been best for them, simply because their computer will not run it. Of
course you might be considering using a computer that you already have, in
which case the choice is already made. This is not going to be a good idea

You should DEDICATE a computer to music and use a different computer

for all of your everyday tasks.

Think of it like this – the more software you install on your computer, the
more likelihood that there will be incompatibilities leading to unpredictable
behavior, instability and perhaps even crashes. Now imagine a computer,
fresh out of the box with only the operating system installed on it. Now
install your digital audio workstation software - and don’t install anything
else. Now you have a clean system with as little that can go wrong as
possible. There is nothing that will disrupt your work more than computer
problems so having a clean computer to work with will be a significant

Now the big question … PC or Macintosh?

People seem to get very worked up over this issue, but the computer is just
a box. It is your digital audio workstation software that you should care
most about.

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So if you have chosen Cakewalk Sonar, you will need a PC because Sonar
doesn’t run on the Macintosh. If you have chosen Digital Performer or
Logic Pro then you need a Macintosh, because neither will run on a PC. If
you have chosen Cubase or Pro Tools LE then you can run them on a PC or
Macintosh, so you have to choose.

There is one significant advantage in buying a Macintosh that should not

be underestimated. Only Apple makes Macintosh computers, so there are
very few different models. Software developers can thoroughly test their
products on every kind of Macintosh. There are literally thousands of
makers of PCs so it is absolutely impossible to test them all. Some software
developers have tried having ‘recommended PCs’ but they have found
that this isn’t good business because models change so quickly. So this
raises the question of if you want to choose a PC, how do you make sure to
choose the right one? There are two ways of doing this…

Firstly, although there are thousands of brands of PCs, many of them have a
lot of similarities. The most important component of a PC is the processor.
There are very few manufacturers of processors and fairly few models
available. It is very likely that a software developer might list compatible
processors, chipsets and even motherboards. It is an excellent idea to
look closely at the software developer’s website for such compatibility
information. If you buy a computer that is compatible with your software,
according to this information, then it is almost certain that it will work

The other way to buy a PC with confidence is to buy one from a

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manufacturer that specializes in making computers for the pro audio
market. In this case, the computer manufacturer will probably state which
softwares you can use. A variation on this is to go to a dealer and ask to buy
a complete system for recording music. Now, if any problem should arise, it
is the dealer’s responsibility to sort it out.

Whether you choose PC or Macintosh, there are certain common factors

that you need to consider. The most important of these is processor speed.

With a faster processor you will be able to record more tracks (although
some softwares are limited to a certain maximum) and run more plug-
ins and software instruments. It has to be said that even a slow computer
these days can run an awful lot of tracks and plug-ins. But it’s human
nature to always want more. Processor speed is one feature of a computer
that is not upgradeable, not by any convenient method anyway. So the
recommendation is to buy the fastest computer you can afford.

It is also worth investing in plenty of RAM memory. Computers often, in

normal use, require more RAM than is physically present. In this case the
processor will dump some data to the hard disk and then fetch it back later.
The hard disk is a much slower storage device than RAM and the computer
will ‘pause for thought’, sometimes for several tens of seconds. This is
irritating and more RAM is the cure.

You should also invest in a second hard disk drive. The reason for this is
that the operating system and digital audio workstation software resides on
your primary disk and will keep it quite busy shuffling digits around. With
a second disk, your audio can have space all to itself. You will be able to

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record and play back more tracks and perform more complex editing. When
considering the size of your audio disk you should consider how you are
going to back up your data. Disk drives ALWAYS fail sooner or later so
back up will be important. Unfortunately, the capacity of back up media has
fallen far behind the capacity of hard disk drives. So having a huge hard
disk that is difficult to back up is not necessarily a good idea. Bearing in
mind that it is a rare song that requires more than five gigabytes to record,
even a fairly small hard disk will store quite a few songs comfortably and
will be much more practical to back up.

Regarding data security, it is also possible to have an array of disks where

data is duplicated. If one disk fails then it can be replaced and the data
will automatically be reconstituted. Unfortunately, this does not safeguard
against fire, water damage or theft. You should also check that your digital
audio workstation software supports disk arrays, should you be considering
this as an option.

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Computers and noise

Computers are inclined to be noisy, and clearly noise and recording studios
do not go together well. The professional solution is to place the computer
in a separate ‘machine room’ which cures the problem completely. The
keyboard, monitor and mouse can be connected via a ‘KVM extender’,
‘KVM’ standing for ‘Keyboard Video Mouse’. These may seem like
specialist items which are not commonly advertised, but putting the
computer well away from your working area is strongly recommended.

There are also ‘silent’ computers. There are three sources of noise in a
computer – the processor fan, the case fan and the hard disk. The processor
and case fans can be replaced with more efficient models that do not churn
up the air quite so much. Silencing enclosures are available to cut down on
hard disk noise. Water cooling is also sometimes used.

The problem with ‘silent’ computers is the uncertainty of just how silent
they are going to be. Putting the computer in a different room is always a
100% solution. A silent computer may be an improvement on an ordinary
model, but not quite a total solution.

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Audio Interfaces

Your selection of audio interface will depend on the type of music you
intend to record, and your recording methods. The most important issue
in selecting an audio interface will be the number of inputs and outputs,
although there are other considerations that we will come to later. Digital
audio workstation softwares feature ‘tracks’. Each track is used to record
a single instrument or voice. (You can combine instruments and voices if
you wish but you will probably be running out of tracks if you are resorting
to this.) A typical digital audio workstation software may feature 32 tracks.
Some have as many tracks as you want, limited only by the performance
of the computer. It is highly unlikely though that you will have 32 inputs
and 32 outputs on your audio interface. Professionals might, but it is very
expensive and home recording studio owners will normally manage well
with fewer.

The smallest number of inputs it is practical to have is just one. However,

two is the normal lower limit. With two inputs you can route any mono or
stereo sound source to any track or pair of tracks within the digital audio
workstation software. So you can record one instrument, then overdub
another, then another, and so on until your song is complete.

You will also need a minimum of two outputs to connect to your

monitoring system (stereo amplifier and loudspeakers) so you can hear your
work as it progresses.

The drawback of this arrangement is that you can only record from a
maximum of two microphones or line-level sources simultaneously. If you

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want to record a full band, playing simultaneously, with each instrument
going to a separate track, then you will need an audio interface with more

8-channel interfaces are commonly available, and generally it is possible to

use multiple interfaces to achieve as many inputs as you need.

Multiple outputs are generally not as useful as multiple inputs. You would
need multiple outputs if you wanted to mix your recording through a
conventional mixing console. Multiple outputs are also useful if you want
to take a signal out from your digital audio workstation software, process it
externally in some way, then route it back in again.

There are three types of input that an audio interface may possess.
Microphone inputs accept the low-level signal from a microphone and
amplify it considerably. Line inputs accept a higher-level signal from a
mains or battery powered piece of equipment such as a musical keyboard.
The degree of amplification is less, or it may be none at all. Instrument
inputs accept the weak signal from an electric guitar or similar instrument.
Instrument inputs are sometimes known as ‘DI’ for ‘Direct Injection’.

There are so many possible combinations that it is difficult to generalize.

But you could come across an interface with eight inputs where each input
is switchable among all three possibilities of mic/line/instrument. Or you
might find an interface where four are line only. Or you might find an
interface where all eight inputs are switchable between mic and line, but
only two have instrument functionality. Your choice should be guided by
the range of instruments, and their numbers, that you intend to record.

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It is worth pointing out that some audio interfaces have NO microphone
inputs. Connecting a microphone to a line input should not cause any
damage, but the signal will be weak and noisy, or there may be no signal at

Where there are microphone inputs there should also be phantom power.
Some microphones need electrical power to operate. This may be provided
on an input-by-input basis or it might be globally switchable. Microphones
that do not need phantom power are mostly unconcerned by its presence,
so global switching. The exceptions are microphones that do not have a
‘balanced’ output. These are rarely sold into the pro audio market and are
not recommended. Some vintage ribbon microphones (and some modern
ones) may however be damaged, so proceed with caution.

Many audio interfaces have digital inputs as well as analog. You might
for example have an old digital audio tape (DAT) recorder with a tape you
want to copy. You could connect it through the S/PDIF digital input and
make a perfect copy rather than suffer slight degradation by going through
the analog inputs.

Some interfaces have ADAT optical inputs. ‘ADAT’ stands for ‘Alesis
Digital Audio Tape’, which was a popular recording format some years ago.
The ADAT itself does not survive, but its 8-channel optical data connection
continues to be useful. You might for instance have an audio interface with
four analog inputs plus an ADAT input. If you also equip yourself with an
8-channel preamplifier with an ADAT output, you can boost your input
count to 12. Very useful.

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Some audio interfaces have an input for digital synchronization. In
general, when you have one or two pieces of digital equipment in your
studio, synchronization isn’t something you need to consider. But when
you get three of more pieces of digital equipment, then they should all be
synchronized to a master clock generator. If your audio interface does not
have a sync input, then you won’t be able to do this. Having said that, it’s
rather advanced stuff to require a master clock generator, and its application
to home recording studios is comparatively rare.

The sound quality of digital audio is primarily determined by two factors.

One is the sampling rate and the other is the bit depth.

To convert an analog signal to digital, its voltage is measured thousands

of times every second. For technical reasons that we won’t go into here,
44,100 times per second is adequate. We would call this a sampling rate
of 44.1 kHz (44.1 kilohertz). 48 kHz is also often used. 44.1 kHz is the
sampling rate used in the compact disc format and most people find it
perfectly satisfactory. It has to be said, however, that 44.1 kHz is only just
good enough. There is very little ‘safety margin’ between 44.1 kHz and a
lower sampling rate where people really would notice degradation in the
signal. So to make sure that the sampling rate is fully high enough, we can
use 96 kHz, which is a massive difference in terms of the raw numbers.

But if you could hear a problem at 44.1 kHz, what would it be? And why
would 96 kHz solve it? If you could hear problems at 44.1 kHz, and most
people can’t, you would notice that the upper end of the frequency range
is limited to around 20 kHz (that is 20,000 vibrations per second). Most

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people can’t even hear up to 16 kHz so this is a small problem. You might
also notice ‘ripples’ in the frequency response at very high frequencies, and
phase anomalies where the timing at very high frequencies is very slightly
imprecise. Also there may be additional slight anomalies in level at very
high frequencies where conversion is not so precise. All of these possible
defects are VERY difficult to hear when equipment is working correctly,
but they do exist and they are measurable. Moving the sampling rate up to
96 kHz eliminates these measurable defects at a stroke. Some people still
feel however that 96 kHz is not enough, and indeed some audio interfaces
that sample at 192 kHz are available.

The drawback of these higher sampling rates is that they make your
computer work harder. Twice as hard in fact at 96 kHz, so you can expect
to get half the number of tracks and be able to use only half the number
of plug-ins that you could at 44.1 kHz, approximately. It is also rather
irritating to start a project at 96 kHz, then find as the tracks build up that
the computer is running out of steam. In conclusion to this point, 96 kHz is
nice, and it is the future, but 44.1 kHz can be a lot more practical and few
people can tell the difference.

Now, to bit depth. CD-quality digital audio measures the signal to an

accuracy of 65,536 different levels. 65,536 corresponds to 2 to the power of
16, hence we call this 16-bit accuracy, or a ‘bit depth’ of 16 bits.

An analog audio signal possesses an infinite number of voltage levels, so

describing it in terms of just 65,536 is bound to result in some inaccuracy.
And this inaccuracy makes itself heard in the form of noise and distortion.

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The fact is though that 16-bit corresponds to CD-quality and very few
people are aware of any noise or distortion at all. But this is only when
things are 100% just so. In the real world of recording you always have to
leave some ‘headroom’ between your highest signal level and the maximum
level 16 bits can cope with. So in practice you might only be using 14 bits,
or maybe even just 12. At a bit depth of 12 bits, noise and distortion are
definitely audible. Clearly therefore it is better to use more bits during the
recording process, then the ultimate output to CD-quality can be optimized.

Because of this it is very likely that your audio interface is capable of 24-bit
operation. This significantly lowers noise and distortion and is a worthwhile
improvement. It does place a 1.5x extra burden on your computer, but it
provides more benefit at less cost than a 96 kHz sampling rate. In summary,
24/96 operation is desirable, 24/44.1 is practical, 16/44.1 will give you
most tracks, and most people won’t notice the difference in terms of

It could be possible to be concerned about the quality of the preamplifiers

and the analog-to-digital converters in audio interfaces. The fact is however
that it is very rare to find a unit sold into the pro audio market that is not
fully adequate in both respects. There are reasons why you might want to
use a different preamplifier, which will be covered later. However, most of
the potential problems in preamplifiers and converters were solved by the
1990s. Today there are issues that are vastly more important, such as choice
of microphone, and gaining the experience and skill to use your equipment
to best effect.

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There is one feature of audio interfaces that you should look out for,
because it really is very useful. And that is zero-latency monitoring.
Let’s look at it this way … imagine you are singing into a microphone
and listening to your own voice on headphones. Between microphone
and headphones the signal has to be digitized, pass through the computer
and then be undigitized again. This process takes a little time, several
milliseconds and perhaps more. This delay is short but distracting. In
fact it is very off-putting to sing while hearing a delayed version of your
voice. Expensive recording systems have a delay that is so short that it
isn’t distracting in the least. Home recording systems have yet to achieve
this standard. Fortunately many audio interfaces have ‘zero-latency

With zero-latency monitoring, the input signal is directed to the output

of the new audio interface, within the interface itself and in analog form.
Analog audio travels instantaneously so there is no delay whatsoever that
is even remotely perceptible to the human ear. There is one slight drawback
to zero-latency monitoring in that if you have applied any process to the
signal in your software, then you don’t hear it via the zero-latency path.
This is only a slight drawback and, while we wait for technology to get
faster, zero-latency monitoring is definitely a good thing.

The way the audio interface connects to the computer is significant, but
becoming less significant as technology progresses. Even so, it is worth
knowing something about the various technologies available. Basically we
want to get data into and out of the computer quickly. That means the most

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tracks, the highest sampling rates, and the greatest bit depth. The fastest
route to the computer processor is via a PCI card. This is a card that fits
inside the computer and talks almost directly to the processor. With some
PCI cards, the analog audio connects directly to the card. This is generally
thought of as not being a particularly good idea. The inside of the computer
is filled with electrical interference that can degrade an analog audio signal.
The better way to do it is to have the analog-to-digital and digital-to-analog
converters in an external box and send digital audio signals, which are
robust, to the PCI card.

The other methods of connecting the audio interface to the computer are
through the USB or FireWire ports.

USB is a general-purpose computer interface and was not designed to carry

signals. The original version of USB was quite slow and could be pressed
into carrying just a couple of stereo signals. The current version of USB
is very much faster and a number of manufacturers use it to carry multiple
audio streams. Despite the fact that USB was not purpose-designed for
audio signals, it now seems to work impressively well.

FireWire was purpose-designed to carry signals. Not just audio but video
too. We sometimes see FireWire therefore as the connection of choice
despite the fact that USB is now very fast. It is generally a safe option to
leave the choice of connection to the manufacturer of the audio interface. If
the interface has the features you want, it doesn’t matter so much whether
it is USB or FireWire. Indeed, some audio interfaces offer both. It is worth
pointing out that not all computers have FireWire ports and you may have

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to buy a FireWire card if this is what you want.

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Audio interface websites

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Digital audio workstation softwares have the basic functionality of

recording, playback, editing and mixing. To provide processes such as
equalization and compression, and effects such as delay and reverb, then
plug-ins are required. (Some digital audio workstation softwares integrate
basic functions such as EQ.)

Digital audio workstation software developers provide a range of basic

plug-ins as standard. These plug-ins are nearly always fully technically
competent. They provide the processes and effects they claim to provide,
usually with a full set of controllable parameters. They have a wide
frequency response, low distortion and low noise. It is in fact perfectly
possible to make a totally professional recording using only the standard
plug-ins. These are however reasons why you might also want to invest
in third-party plug-ins. Some of these are quite expensive so the reasons
should be good. We will look at each type of plug-in process and effect in

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EQ, or equalization, is the process best-served by standard plug-ins.

Some standard EQ plug-ins provide a greater range of features than even
the mightiest of traditional studio mixing consoles. Even so, there is no
shortage of third-party EQ plug-ins available. Some of these emulate
real-life hardware equalizers, often ‘classic’ models from the past. One
surprisingly strong reason for preferring a hardware emulation to a standard
plug-in is that the on-screen display looks nicer and is easier – perhaps even
a pleasure to work with. Although the appearance, or ‘skin’, of a plug-in
has no relevance to its sound, it’s just nice to work with nice equipment.
Sound engineers since the dawn of the art form have felt this.

There are also most substantial reasons. One is that however well-specified
one equalizer is, another one might have different frequency curves and you
might just prefer the other one. Neither is better, but they sound different
and this may be sufficient reason for preferring one over the other. Another
reason is that certain types of electronic circuitry provide ‘warmth’.

Warmth is a subjective term for mild distortion. Distortion changes the

sound of a voice or instrument. Equipment in an earlier era of recording
distorted the sound whether you wanted it to or not, so distortion was seen
as a bad thing. Modern digital equipment and software has almost zero
distortion so it captures sounds very accurately.

However, when used in small amounts with careful consideration, a little

warmth can be a good thing. So plug-ins that emulate the warmth of analog
equipment, in addition to their stated function can be very useful. Old

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vacuum tube equipment provided a very nice quality of warmth, so a plug-
in that emulates a vacuum-tube equalizer can be expected to be warm too.

In summary, the standard EQ plug-in supplied with your digital audio

workstation software should provide all the pure equalization you could
ever need. But a plug-on that emulates a classic vacuum tube equalize can
also give you extra warmth when you need it.

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A recreation of the classic Pultec equalizer in plug-in form, by Universal

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Once again, standard compression plug-ins are technically very good. The
prime objective of compression is to reduce the difference between loud
and quiet parts of the signal. This is no trouble at all to the compression
plug-in that came free with your digital audio workstation software.

Controling the level of a signal automatically from moment to moment is

an easy thing to do in software. It is however much more difficult in the
analog domain. There are five main methods of achieving compression,
and each has its own characteristic ‘sound signature’. The five methods
are diode bridge, variable-mu, optical, FET and VCA. In traditional analog
audio, a studio engineer would experiment with the equipment to hand and
form an opinion on which compression best suited each type of application.
He or she probably wouldn’t give a second thought to what technology
was inside a particular unit. In general the diode bridge and variable-mu
compressors bring warmth to a signal. The others are cleaner, but still
impose their own character on the signal.

When you first start recording it is a good idea to become completely

familiar with your standard compression plug-in. Then start to add more
exotic compressors to your collection. The variable-mu compressor is
probably the next one to go for as it has the most warmth. Vacuum tube
compressors work on the variable-mu principle so you should look out for a
tube emulation.

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Plug-in emulation of the Fairchild 670 variable-mu vacuum tube

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Delay has been used since the time of Elvis Presley and it is still widely
used now. Delay involves creating either a single or a repeating echo after
the original signal. It is child’s play for a competent software designer to
create a delay plug-in that works perfectly.

But … that wasn’t the way it worked in Elvis’s day. In the early days of
recording, delay was created using a spare analog tape recorder. Musicians
heard this effect on records and wanted to use it in their live performances.
The problem was that studio tape recorders were extremely expensive,
and extremely large and heavy. They were totally impractical to take out
on the road. But enterprising manufacturers saw the demand and set about
producing miniaturized delay units that were practical to use on stage.

Some were based on loops of magnetic tape, others on magnetic disks. At

least one had a can of oil at the center of its function! There were many
such devices, and the one thing they all had in common was that they were
imperfect. The echoes ‘wobbled’. Surprisingly, this created an even richer
sound, which was far more musically useful than a technically perfect
digital delay. Fortunately there are now several plug-ins that emulate analog
delays of the past, often with variable ‘wobble’ and degradation. They are
well worth investigating.

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Vintage analog delay plug-in emulation

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Of all the plug-ins that are supplied as standard with the various digital
audio workstation softwares, reverb is likely to be the least good. The
phrase ‘least good’ is used rather than ‘worst’ because they are unlikely to
be actually bad. Many hit records have been made in the past with reverbs
that were really rather awful, so that’s not necessarily a limiting factor. But
technology has now moved on to the point where even free reverb plug-ins
are quite good. But now, there are better …

In the 1980s it became possible to ‘sample’ the notes of a musical

instrument and play them from a keyboard. It is now possible to sample
an acoustic space and contain it within a reverb plug-in. These are known
as ‘convolution’ reverbs and there are several available. When you buy a
convolution reverb you will be supplied with several ‘impulse responses’
which contain the acoustic signatures of real rooms and auditoria.

It is possible also to sample the impulse response of a classic reverb unit, so

plate reverbs and early digital reverbs (some of which are surprisingly nice)
can be available too. Offbeat impulse responses are also available, such
as the back of a van. If you are really dedicated you can create your own
impulse responses and load them into your plug-in.

It is important to note that you can achieve basic professional recordings

even with an ordinary reverb plug-in. But it is a joy to use a really good
reverb, and it can put an extra ‘gloss’ on your mix that raises the standard
just a little bit higher.

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Altiverb convolution reverb plug-in

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Pitch changing and correction

Pitch changing plug-ins are commonly offered as standard with digital

audio workstation software. So if a certain note is out of tune, you can
manually shift it to the pitch it ought to be. You can also use pitch changing
as an effect. If you copy a track, then apply a pitch change of around 10
cents or so (a cent is one-hundredth of a semitone) then the result will be a
distinctly different sound, which you may like.

Correcting out-of-tune notes one by one is a time consuming business.

And you may need two or more tries for each note before you achieve
reasonable accuracy. Fortunately, automatic pitch-correction software
is available that makes this task instant. You tell the software which key
a song is in, and then all the notes of, say, a lead vocal will be shifted
precisely to the permissible notes in that key.

Automatic pitch correction is not a cure for all woes however. Firstly it
requires that the vocalist sings in tune to an accuracy of at least better
than one semitone; with some notes of the scale one quarter-tone. Many
vocalists will find this challenging. Secondly it doesn’t always work
– certain sections of the vocal will confuse the software and it will react
unpredictably. Thirdly, it can result in a rather mechanical performance that
is lacking in true musicality.

When you first start out recording, you should probably take the view
that you intend to capture great musical performances. You should aim
to work with singers who can sing in tune. And if you sing yourself, then
you should make every effort to improve your skills until your tuning

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is faultless. But at some point you may realize that professional results
are always going to continue to elude you unless you invest in a pitch-
correction plug-in. So, you don’t buy a pitch-correction plug-in right at the
start. You work with the best musicians you can get hold of, practice your
own vocal skills if necessary and THEN - if you have to - buy the plug-in
and apply a small amount of pitch correction as a finishing touch.

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Antares Auto-Tune pitch correction software

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Mixing is performed by sweetening all of the individual tracks of your

recording, then blending them together with the fader and pan controls.
This results in a stereo output that could be your finished mix. Alternatively
you may want to improve your mix using additional processing in the
master channel. For instance, your tracks may be blended perfectly, but the
overall balance of frequencies is dull. You could re-EQ all of the individual
tracks, or you could insert an EQ plug-in into the master channel.

But you can go further and compress the mix as a whole, by placing a
compression plug-in in the master channel. This can be any compression
plug-in or a specialist ‘bus compression’ plug-in if you prefer. You should
realize that bus compression is an art in itself. It won’t automatically make
your mixes better – you have to learn how to use it to get best results.

The other part of the mastering process is the optimization of subjective

loudness. Imagine that you make a mix, and adjust all of your fader levels
precisely so that the peak signal level is just a hair’s breath below making
the red light come on. In other words your mix is as ‘hot’ as it possibly
could be. Now compare it to a commercially-released recording in a similar
musical style. The chances are that the commercially-released recording is
a lot ‘hotter’. Its peak level isn’t any higher than yours, but somehow it just
sounds a lot louder. The reason for this is that it has been processed so that
the average level is higher, while leaving the peak levels where they were
– right at the top already

Compression is one process that can make a mix subjectively louder. By

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applying compression correctly you can bring up the low-level signals
while leaving the peaks where they are, thus increasing the average level
and therefore the subjective loudness. ‘Limiting’ is the same thing as
compression, just taken a step further. So a limiter plug-in can be used
to achieve even more subjective loudness. But even better than ordinary
limiters are specialized mastering limiter plug-ins that analyze the signal
and make it as hot as can be, without too much sonic degradation. Of
course, it is always possible to take things too far. Your mix might then be
as hot as the core of a nuclear reactor. Unfortunately it sounds terrible.

The last word in master processing is the multiband compressor. This splits
up the full frequency range of the signal into five or so separate bands. It
then compresses each band independently. This ensures that each band is
full to the brim with level and, overall there is hardly a fraction of decibel
wasted at any frequency. The multiband compressor is a powerful tool and
it can increase subjective loudness massively. However it can also change
the frequency balance of the mix, so you do have to use it carefully. If you
want to match the subjective loudness of commercially-released music,
then master processing is essential. But remember that it doesn’t do its
work all be itself and you have to be firmly in control.

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SSL bus compressor plug-in

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Waves L3 multiband mastering limiter plug-in

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Software Instruments

In a previous era of home recording you would have needed an expensive

keyboard synthesizer to create your ‘bread and butter’ sounds, and a
separate sampler to create your own sounds and load in library loops and
samples. Now both of these devices are emulated in software inside your
computer. The massive advantage is that software instruments can be much
cheaper. Also, since there is no physical product, many small companies
have sprung up, leading to an enormous variety of sounds being available.

To play your software instruments, you need a physical musical keyboard.

But since this doesn’t need its own internal sound generator, it can be
reasonably inexpensive. Such keyboards are often known as ‘master’

Choosing a master keyboard is quite easy. First you will decide how many
keys you want. A full piano keyboard has 88 keys, and master keyboards
are available with as many. This encompasses a range of just over seven
octaves. Five octaves or 61 keys is probably a more convenient size and
is certainly more common. You can buy smaller master keyboards, but
since their range is less then you will find yourself pressing the ‘octave
up’ and ‘octave down’ buttons often. The smallest keyboards do not lend
themselves to two-handed playing, if you have that ability.

Master keyboards are available which are ‘weighted’ or ‘piano action’.

If you are a pianist then you will find the familiar feel comfortable. Few
master keyboards feel exactly like a piano because the keys do not extend
so far into the instrument and the point where they lever is closer to the

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fingers. That is a small issue however. If you are not a pianist then you will
probably prefer a keyboard that is not weighted. They are faster and more
agile. Your fingers will feel less fatigue too.

Master keyboards connect to the computer by USB or by MIDI. Getting the

system to work via MIDI is really easy, if your audio interface has a MIDI
IN socket. If it doesn’t then follow the instructions precisely to connect via
USB. It can be a little more tricky but you will be able to connect without
the additional expense of a MIDI interface.

When you investigate master keyboards, you will find some that have
several or even many controls. If you know exactly why you need a certain
control then this will be important to you. If not you can ignore them. The
essential controls are the musical keys. There should be a pitch bend and a
modulation control to the left of the keyboard, and there should be ‘octave
up’ and ‘octave down’ buttons, other than on an 88-key keyboard where
you don’t really need them. There should be a socket on the back for a
sustain pedal. Other than that, anything else you might want to adjust can
be done on the screen of your computer.

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MIDI master keyboard by M-Audio

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Software synthesizers

A software synthesizer is an instrument that generates sound signals using

mathematical algorithms, inside the computer. Most software synthesizers
mimic the techniques used by physical synthesizers of the past. Some
software synthesizers emulate the sounds of non-synthetic instruments
– sometimes amazingly well.

The classic synthesizer is the ‘subtractive’ synthesizer. It is sometimes

incorrectly referred to as an ‘analog’ synthesizer. Subtractive synthesis
can be performed digitally, and not all analog synthesizers are subtractive
synthesizers. The classic physical subtractive is the Minimoog. Like
other subtractive synthesizers it works by generating a sound waveform
that is rich in harmonics, then certain bands of harmonics are filtered
away, leaving the sound that you want. The Minimoog is available in the
form of a software instrument emulation, as are other classic subtractive
synthesizers of the 1970s and 1980s.

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A modern version of the Minimoog subtractive synthesizer

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Another form of synthesis is FM synthesis, popularized by the Yamaha
DX7 of the mid 1980s. The Yamaha DX7 was fully digital, even all that
time ago, so very convincing software FM synthesis is possible. It isn’t
even an emulation. It is FM synthesis. FM synthesis is useful for tinkly
electric piano sounds, basses and all sorts of weird and wonderful electronic

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Yamaha DX7 original FM synthesizer
(Photo used with permission from

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There are other synthesis techniques, many of which are available in the
form of software instruments. These include additive synthesis, wavetable
synthesis, physical modeling and more.

One exceptional application of software synthesis is in electric piano

emulation. The electric piano is exemplified by the Fender Rhodes and
Wurtlitzer EP200, both of which had many years of popularity. Both have a
hammer action, similar to an acoustic piano, but the hammers hit metal bars
called ‘tines’ instead of strings.

You could buy a secondhand Fender Rhodes from eBay. That might be a
MkI, MkII or later version. Its pickups will be adjusted in a certain way,
it may have the ‘Dyno’ modifications. It will be just one Fender Rhodes
piano. But the software emulation can go beyond that and provide many
of the sound textures that you would come across in different examples of
the real thing, and other models of electric piano also. And very convincing
they can be too.

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Fender Rhodes original electric piano

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The other software instrument is the sampler. A sampler can record any
sound, and then you can play back that sound at different pitches from
your musical keyboard. So you could sample the notes of a grand piano,
then play your samples from the musical keyboard in your home recording
studio. Getting a good grand piano sound in a home recording studio used
to be almost impossible. Now it is straightforward, and quite inexpensive.

In theory, therefore, you could sample your own sounds and build up your
own sample library. In practice however there is such an enormous range of
sample libraries available that you can have every sound under the sun right
there at your fingertips. If you do wish to sample your own sounds, which
is a great source of originality, then you must choose a sampler software
that allows you to do this. Some do not intrinsically have this functionality,
but you can import recordings you have made and edited in your digital
audio workstation software.

To choose a software sampler, first you should decide which is the one most
important sample library you want to be able to play. Then look at your
second, third and forth choices. Sample libraries are generally available in
only a limited range of formats so you need a software sampler that can
play the majority of libraries you want to work with. Although software
samplers each have their own range of facilities, strengths and weaknesses,
they are all – at least the well known ones – capable of doing the job
well. But the differences in sample libraries are huge. This really is the
right place to start. So your first port of call should be the sample library
companies or their distributors. Audition their products and see which you

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like best. Then look at which formats they support, then select a software
sampler that is compatible with your digital audio workstation software.

An acceptable short cut is to audition the sample libraries that a provided

free with the sampler software instruments themselves. If you like the
sounds, then you should buy the sampler.

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Software sampler websites (Emulator) (Kontakt) (HALion) (GigaStudio) (MachFive)

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There are also sample replay-software instruments. These are software
instruments that contain their own samples. You can’t load libraries into
them. They have a range of sounds for you to play from your musical
keyboard, and that is all that they do. This might sound limiting, but
in fact their simplicity makes them a joy to use. Where fully-featured
software samplers are complex and intricate, these simple sample players
are something that you will just go ahead and use. And you can’t have too
many of them!

Really expensive sample libraries

If you want to compose music for film or TV, then you will need an
orchestral sample library. These are available in all price ranges, but it is
instructive to note that one, the Vienna Instruments series, has a top-of-the-
range product that costs, at the time of writing, more than $10,000.

The Vienna Instruments series has been around for a long time, so this
proves that top composers are ready to pay this amount of money for the
best quality orchestral sample library.

Well, anyone can dream...

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Native Instruments Kontakt software sampler

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Copy protection

When you buy a plug-in or software instrument, you are usually only
allowed to use it on one computer at a time. One of the most common
methods of enforcing this is the iLok key, which you have to buy yourself
to protect software developers from unauthorized use.

Fortunately, the iLok works quite well and does not cause significant
difficulties in operation, unlike earlier copy protection systems.

All of your software authorizations reside on your iLok. Since the software
will not run without an iLok key with the necessary authorization present,
you can install the software on multiple computers. It will only run on the
one to which the iLok key is connected.

You should be aware that when your iLok key is loaded up with
authorizations it is very valuable. You should take great care of it and insure
it for its full value.

iLok key

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If you record purely instrumental music then you may not need a
microphone. Of course if you want to record vocals, a microphone will be
essential. Also if you want to record acoustic instruments, a microphone
will be required. Some acoustic instruments can be fitted with a pick-up
but this changes the sound and it won’t produce as natural a sound as a
microphone can.

Even if you use only electronic and digital instruments that could be
connected directly to your audio interface, you can benefit from having a
microphone. Connecting these instruments directly can result in a rather dry
and uninteresting sound. By connecting these instruments to an amplifier
- a guitar amplifier and speaker often works quite well - you can achieve a
sound which is more natural, more alive and more in keeping with sounds
of vocals and acoustic instruments.

In any case, if you don’t have experience of recording using a microphone

then you could, over a period of time, become expert in all aspects of sound
engineering, music production and recording, but this one blank area where
you haven’t experience of using a microphone prevents you from being a
complete sound engineer or producer capable of taking on any studio task.

How many microphones are necessary in the studio? In theory if all you
need the microphone for are vocals then you only need one microphone.
Even if you intend to record background vocals then you can layer them
one at a time so more than one microphone isn’t absolutely essential.

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But there are reasons why having more than one microphone is better.
Firstly, if you have two identical microphones then you can record in
stereo. It would be unusual to record a lead vocal in stereo, in fact very
unusual, but if you record an acoustic guitar in stereo it can sound much
better than the same instrument recorded in mono.

This applies to just about every acoustic instrument. When you mic up
an instrument in stereo with two identical microphones, you can afford
to capture some of the natural ambience of the room, which makes the
recording more realistic. If you have two microphones but they are not
identical then, surprisingly, you can still make stereo recordings. The left
and the right sides of the stereo image may sound a little different but if you
try this for yourself you will be surprised how well it can work and, with
a little equalization, you can make it so that few people would realize that
dissimilar microphones were used.

Another reason for having more than one microphone is so that you can
record more than one musician at the same time. You may, for instance,
have a singer and an acoustic guitarist. It is likely that you will get a
much better performance if you record them both at the same time, rather
than recording the guitar track first and then overdubbing the vocal. You
could record the vocal and guitar with just one microphone if you placed
it in exactly the right position to capture the correct balance of the two
instruments. However, it is difficult and time consuming to find the right
spot so it is much better to use one microphone for the singer and one
microphone for the guitar, or if you have three microphones you can use

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one microphone for the singer and two microphones for the guitar in stereo.
But probably the main reason for having more than one microphone is that
different microphones have different sound qualities.

There is, even now, no such thing as the perfect microphone that will
capture a sound with perfect accuracy. All microphones are flawed to some
extent, and in different directions. So one microphone might be good at
capturing the fine detail of a sound in crystal clarity while another might be
good at capturing the body and the fullness of the sound with warmth and

In a sense, microphones to the sound engineer are like all the different tools
an artist might draw or paint with. An artist might use charcoal, pencil,
water color, acrylic, oil paint, all the different brushes and all the different
tools available to add texture to their work and to contrast one area of a
painting with another. It is similar with microphones.

A certain microphone will suit a certain instrument. Another instrument

might be best served by a different microphone, and an experienced sound
engineer can build up a complete sound image using each microphone to its
best advantage and blending the result together into a satisfying whole.

When you have just one microphone you won’t be aware of how different
different microphones can be because you have nothing to compare
your one microphone with. When you have another microphone that is a
different model or make to the first one, you will hear straightaway that
they have different sound qualifies. On some instruments they might sound
very similar; other instruments will point out the differences. Soon you will

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develop a preference as you will use one microphone for certain purposes
and the other microphone for other purposes. And then you will start to
wonder what it will be like to have a third microphone, so you have another
sonic texture to play with. Once you have arrived at this point, there’s no
limit to the number of microphones you could aspire to have. You never
have to ask a sound engineer what they want for their birthday. Another
microphone will do nicely, thank you.

There are two types of microphones in common professional use. One

is the dynamic microphone, the other is the capacitor microphone,
which is sometimes known as the condenser microphone. ‘Capacitor’
and ‘condenser’ mean exactly the same in this context. The dynamic
microphone works like a miniature electricity generator. There is a
diaphragm which is a small thin membrane which vibrates in response to
sound traveling through the air. Attached to the diaphragm is a coil of wire.
When the diaphragm vibrates, that coil of wire also vibrates, and it vibrates
with the field of a magnet. When a coil of wire vibrates or moves at all
within the field of a magnet, an electrical current is generated within the
coil, which is the signal that forms the output from the microphone. It is a
very simple technique but very effective.

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Shure SM57 dynamic microphone

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The capacitor microphone is more complex. We don’t need to go too deeply
into how it works here but the main difference is that there is no coil of
wire. The the diaphragm is not weighed down by the burden of the coil
therefore it can be very much more responsive to the sound vibrations that
strike it. So whereas the diaphragm of the dynamic microphone is sluggish
due to the weight of the coil, the diaphragm of the capacitor microphone
is light and quick to move. This means that the capacitor microphone can
capture a much more detailed sound. Where you will hear the difference
most of all is in metallic percussion instruments.

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AKG C414B XLS capacitor microphone

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A dynamic microphone used on metallic percussion instruments such as
cymbals will sound dull. In fact it might sound so dull that it is completely
unrealistic and therefore unusable. The capacitor microphone on the
other hand will capture cymbals with perfect clarity. This is not to say,
however, that the capacitor microphone is always better than the dynamic
microphone. Sometimes the dynamic microphone just sounds nice on a
particular instrument.

Dynamic microphones tend to sound good on drums, that is on the

individual drums of the drum set. A capacitor microphone placed close
to a drum in normal modern drum recording technique will capture the
sound of that drum from close up very accurately. If you think about it
though, you would never put your ear that close to a drum so although the
capacitor microphone is capturing an accurate sound, it is capturing an
accurate sound that you would never actually hear in real life. A dynamic
microphone placed in the same position will capture a sound that, to the ear,
is much more like we expect the sound of a drum to be.

The same applies to electric guitar loudspeakers. The capacitor microphone

will capture an accurate sound, but the dynamic microphone may well
capture a sound that is subjectively more pleasing. Dynamic microphones
can be quite inexpensive and we are talking around $100 for a model that
is entirely professional in quality. In fact there are two models which are so
classic that they are found in all applications of sound engineering. They
are both made by the Shure company. One is the Shure SM57, the other is
the Shure SM58. These are very similar designs and the sound quality is

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very, very similar. The SM58 has the advantage of an integral pop shield
which makes it more useful for live vocals on stage.

There are plenty of other dynamic microphones in common use. One is the
Beyerdynamic M201 which, for a dynamic microphone, has a very good
sound quality. There are the Sennheiser MD421 and MD441, both of which
have been in the catalogue for years. Like the SM58, SM57 and M201, the
MD421 and MD441 both exhibit a good, strong sound quality.

One other model that is worthy of mention here is the ElectroVoice

RE20 which is commonly used in radio. The reason this model is popular
in radio is that subjectively it sounds good on speech. Often the choice
of microphone is not so much a question of accuracy, it’s whether it
just sounds good for your particular application, and for speech the
ElectroVoice RE20 has that quality.

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Shure SM58, Beyerdynamic M201, Sennheiser MD421 and MD441,
Electrovoice RE20 dynamic microphones.

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Capacitor microphones tend to be more expensive because they are more
complex. In recent years budget capacitor microphones have come onto
the market, many of them made in China. For some reason many of these
budget capacitor microphones strive to emulate high class and expensive
capacitor microphones. But they often have irritating defects in their sound
quality and they are definitely not the same thing.

There are other moderately priced capacitor microphones from established

microphone manufacturers that don’t pretend to be high class microphones,
but they do a good solid job at a reasonable price. One example of this,
although not the only example, is the AKG C3000B. There’s nothing fancy
about this microphone, it doesn’t have an exotic sound, it won’t flatter any
instrument or voice you use it on but it will turn in a pleasing performance
without defects every time you use it.

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AKG C3000B capacitor microphone

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The diaphragm of a capacitor microphone produces a signal that is
extremely weak and it can only travel a very short distance through a cable.
Therefore, every capacitor microphone has an internal amplifier to boost up
that signal so it can travel down the cable successfully.

Of course that amplifier needs to be powered; in fact the diaphragm needs

to be powered too because of the way the capacitor microphone works. So
the microphone input of the microphone preamplifier, or the audio interface
or the mixing console, will have what is known as phantom power.

Phantom power travels along the ordinary audio cable to the microphone.
All you have to do is switch it on when necessary. Some capacitor
microphones are of a special type known as electret microphones that can
be powered from an internal battery. In general their sound quality is not
quite as good as a normal capacitor microphone.

So, if you don’t have much money to spend on microphones, a reasonable

compromise would be to buy a dynamic microphone of professional
quality and a moderately priced capacitor microphone from an established
microphone manufacturer, and you would get the benefits of the differences
in sound quality between these two mics.

Ultimately, though, you should aspire to a capacitor microphone of real

quality, particularly for lead vocals. Having a really great vocal microphone
in your studio is important. If you can’t afford it right at the start, there is
no reason why you can’t make recordings of a good professional quality,
but having that really good vocal mic makes your recordings just that little
bit better.

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So what makes a good vocal mic? Firstly, let’s look at what makes an
accurate microphone. If a manufacturer wants to make a microphone as
accurate as possible and faithful to the original sound source then that
microphone will be a capacitor microphone. It will have a fairly small
diaphragm about 10 or 12 mm across.

The internal amplifier of this accurate microphone would need to boost

up the signal without changing or distorting it in any way, and to do this
the principal amplifying device would be a field effect transistor (FET),
which will give an extremely transparent performance. With this accurate
microphone we could record the lead vocal of a song and the recording
would sound almost perfect. So why would there be a problem with that?
The answer is that we want the vocal to sound more than perfect. We
want the sound not just to be as it is, we want it to be better than it is, and
curiously that involves using a microphone that is less than perfect.

This microphone will have a large diaphragm, maybe 25-30 mm across.

A large diaphragm is less accurate than a small diaphragm. The reasons
for this have to do with the additional mass and also cancellation effects
for any sound which doesn’t arrive exactly head on at the diaphragm. But
subjectively, for vocals in particular, the large diaphragm simply sounds
nicer. There’s no way to measure this, it just does sound nicer and we
accept that.

Also, for a vocal microphone, ideally the internal amplifier should use
a vacuum tube as its principal amplifying device, rather than an FET.
The vacuum tube has a tendency to produce a mild distortion, which

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does change the signal so it is not accurate any more, but subjectively
the ear interprets that as ‘warmth’. So the large-diaphragm vacuum tube
microphone will add warmth, presence and fullness to the vocal sound,
which is preferable to it being merely accurate.

There are large diaphragm microphones that have FET amplifiers. One
good example is the Neumann U87. This is a classic microphone which,
for many years, was considered to be the best microphone to chose for
vocals. It also has many instrumental applications. Its rival was the AKG
C414 which was rated almost as highly for vocals and could be used very
effectively on almost any sound source.

The AKG C414 also worked very well in a stereo pair and, though this
microphone has been around literally for decades, you will still see it
widely used for drum overheads. Both the Neumann U87 and the AKG
C414 are still available new, the C414 in updated versions. They are quite
expensive, fortunately not as expensive as they used to be.

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Neumann U87 large-diaphragm FET capacitor microphone

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A good example of a large-diaphragm vacuum tube microphone is the
Neumann M147. In general, vacuum tube microphones can’t be operated
from phantom power so they require their own special power supplies,
which are supplied with the microphone. So you connect the microphone to
the power supply, connect the power supply to the mains and also connect
an output from the power supply to your preamplifier, audio interface
or mixing console. The Neumann M147 is a lovely warm-sounding
microphone - but at a price, and you can pay even more for yet better
microphones if you want to.

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Neumann M147 large-diaphragm vacuum tube microphone

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There is another type of microphone called the ribbon microphone. This
works in a way which is very similar to the dynamic microphone. The
difference is that it doesn’t have a coil. The diaphragm is made from a
corrugated material which is made conductive and is called the ribbon. This
is suspended in the field of a magnet so effectively the ribbon is its own
coil. It is a coil of only one turn but a coil nonetheless.

Because there is no coil as such, the diaphragm (the ribbon) can be light
and responsive. The dawback, historically speaking, is that it is difficult to
suspend the ribbon well and ribbon microphones have tended to be fragile.
There are exceptions. The Beyerdynamic M130 and M160 are ribbon
microphones that are quite robust.

Ribbon microphones were perhaps not developed as much as they could

have been because of the rise to dominance of the capacitor microphone.
However, the ribbon microphone has its own characteristic sound which in
some contexts can be very desirable. The ribbon microphone’s sound has
a subtle fullness. It captures detail but without over-emphasising it. It is a
sound that needs to be experienced. It would not be reasonable to suggest
that your first microphone should be a ribbon microphone nor even your
second or third, but somewhere down the line there will be a place in your
microphone collection for a ribbon microphone, perhaps more than one.

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Beyerdynamic M160 ribbon microphone

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Directional characteristics of microphones

Different microphones have different directional characteristics. Some

microphones are almost equally sensitive all the way round. They are
called omnidirectional microphones. Some microphones are sensitive at
the front and at the sides but they are not sensitive at the back. They are
called cardioid microphones. Some microphones are sensitive at the front
and equally sensitive at the back but not at all sensitive at the sides. They
are called figure-of-eight microphones. In between the cardioid and the
figure-of-eight pattern there is the hypercardioid microphone which is very
directional towards the front and has a slight sensitivity towards the rear.

When you become deeply involved in sound engineering, and particularly

when you record bands and orchestras, then you will use these directional
patterns, or polar patterns as they are often called, to highlight certain
instruments with certain microphones while rejecting other instruments
from those same microphones. It is a significant skill that takes some time
to acquire.

When you are working in a home studio, the polar pattern is not so
important. Most microphones are in fact cardioid pattern because this is the
most generally useful. The figure-of-eight microphone is virtually useless
in a home studio context because home studios tend to be on the small side.
And because the the rear of the microphone is equally as sensitive as the
front you will often tend to pick up too much ambience from the room. The
hypercardioid pattern would not be a problem in this respect.

There is also a difference in sound quality between microphones that are

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directional. ‘Directional’ means microphones with cardioid, hypercardioid
and figure-of-eight polar patterns.

A directional microphone will tend to boost low frequencies when it

is used close to the sound source. This is called the proximity effect.
Omnidirectional microphones do not have the proximity effect. You can
expect when you use a cardiod microphone close to the sound source, such
as a vocalist, that the bass frequencies will be enhanced. Generally this is
not a problem, firstly because you will appreciate the additional warmth
that those low frequencies bring. Secondly you could equalize them away
if you wanted to. One thing you might like to try, however, is to record
the lead vocal with a cardioid microphone, which will give enhanced
low frequency warmth, and background vocals with an omnidirectional
microphone. This way the lead vocal will stand out warm and full, and the
background vocals can stay more in the background as they should be.

Finally regarding choice of microphone, we must consider vintage

microphones. Vintage, of course, is another word meaning ‘old’. Why
would you want an old microphone when you can buy a new one? The
answer is that we are accustomed to hearing many great recordings that
were made in the past on exactly these old microphones, so we have
become used to the sound of old microphones and we seek to emulate it
today. The old microphone, or the vintage microphone, isn’t necessarily
better, and it is certainly not better because it has ‘matured’ over the course
of time. That would not be true. But we have almost been programmed to
like these sounds and, although you could buy a microphone today that

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claims to be able to replicate the sound of a vintage microphone, some
people will not be happy unless they have the real thing.

There are two ultimate classic vintage microphones by which all other
microphones are judged. One is the AKG C12, the other is the Neumann
U47. Both of these microphones are large-diaphragm vacuum tube
microphones. That’s the only way they knew how to make microphones in
those days.

Engineers who have access to these microphones love their sound. It is

astonishing that a microphone can be 40 years old or more and still give a
brilliant sound that people aspire to, but it is so. If you wanted to have an
AKG C12 or Neumann U47 in your collection then firstly you would have
to wait for one to become available, and sound engineers and studios don’t
easily part with them. Then you would have to buy it at whatever price was
being asked. Typically, for one microphone, that price might be $10,000.

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Neumann U47 vintage large-diaphragm vacuum tube microphone

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Microphone stands

You will obviously need a microphone stand. This should be a boom stand,
which has an upper section that can be angled with respect to the vertical

There are unfortunately a number of cheap microphone stands that are

sold into the pro audio market that are insufficiently sturdy. They tend to
be floppy in use and soon fail. Unless you need a whole forest of stands to
record an orchestra, it is a wise investment to buy the best quality stands,
which are not expensive in comparison with the microphones you will use
them with.

Microphones are normally supplied with a stand mounting clip. Normally

this clip is perfectly adequate and you do not need an elastic suspension.
Elastic suspensions are useful where there are vibrations in the floor of your
studio, but otherwise they are of little benefit, they tend to be expensive and
they are often a nuisance to use.

Microphone clip and elastic suspension

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Microphone Preamplifiers

Microphone preamplifiers have been a popular topic recently. Indeed some

people seem to think that the selection of microphone preamplifier is a life
or death issue. They will say that if you don’t have the right microphone
pre-amplifier then you can’t possibly make good recordings, and of course
the right microphone preamplifier is the one that they use (and chose from a
magazine ad!).

This is an issue that demands serious consideration. Firstly because

microphone preamplifiers can be very expensive. Secondly, your audio
interface probably has microphone preamplifiers built in already, so why
would you want to pay a lot of money for a microphone preamplifier when
it is something you already have for free?

The microphone preamplifier is an important part of the signal chain. The

reason for this is that the output of a microphone is very low and rather
delicate. Typically the output could be somewhere between 1 millivolt and
10 millivolts. A millivolt is one thousandth of a volt. Of course it can go
higher. A microphone used to pick up a bass drum will have a very much
higher output level. Even so, low output levels are the norm.

The microphone preamplifier has to boost up these low levels up to around

about 1 volt. We call a level of around about 1 volt ‘line level’, and the
rest of the equipment that you use prefers to operate at that level. There
have been excellent microphone preamplifier designs around for decades.
Think about it. When you listen to a recording by Frank Sinatra or Nat
King Cole, do you find yourself complaining about the quality of the

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microphone preamplifier? No, absolutely not. This tells us that perfectly
adequate microphone preamplifiers were available as far back as the 1950s.
Professional recording equipment from the 1950s, 1960s, 1970s, 1980s and
onwards includes microphone preamplifiers that are fully adequate for the

However, until more recently, microphone preamplifiers had to use a

transformer to boost up the signal level before it met any of the active

A good transformer is an expensive component, therefore at a semi-

professional level where microphone preamplifiers omitted the transformer,
the sound quality was not particularly good. Preamplifiers without a
transformer tended to be excessively noisy.

This is perhaps where the debate about microphone preamplifiers started,

when there was a genuine problem to be solved. However, microphone
preamplifier designers and electronic component manufacturers were able
to come up with techniques that allowed microphone preamplifiers without
transformers to sound every bit as good as those that did have transformers,
and even better. And once the need for this expensive component was
removed, microphone preamplifiers of professional quality - and that is
fully professional quality - started to be incorporated into equipment that
you will find in a home recording studio.

Quality of a microphone preamplifier is defined in three major ways. One is

the frequency response.

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The range of frequencies we normally consider in professional audio is
from 20 Hz at the low frequency end of the scale to 20,000 Hz or 20 kHz
at the high frequency end. We can perceive frequencies lower than 20 Hz.
Indeed it is difficult to assess the point where you stop hearing them with
your ears and start feeling them in your entire body. However, frequencies
below 20 Hz tend not to be musically useful, and in fact can often be more
trouble than they are worth. Having said that, it is no trouble at all to design
audio circuitry to handle frequencies as low as you wish. It is not rocket
science, it is a simple matter of applying established design techniques.

At the high frequency end of the scale, a frequency of 20 kHz is beyond

the range of most people’s hearing. High frequency sensitivity decreases
with age. A child might be able to hear up to 20 kHz but by the age of 20,
that response has probably dropped to maybe 17 or 18 kHz; by the age of
40 down to 14 or 15 kHz and so on. This change is hardly noticeable. The
fact is, however, that most audio equipment can accommodate a frequency
range of up to 20 kHz with ease. That is fully adequate for the vast majority
of people. And so with microphone preamplifiers, to design and build a
microphone preamplifier with a full frequency range from 20 Hz to 20 kHz
is absolutely no problem. The designer would have to do something very
wrong not to achieve that.

Distortion used to be a significant problem in audio up to around the 1960s,

and in fact distortion remained a problem in analog tape recorders until
they largely fell out of use. Distortion changes the shape of the waveform
of the signal and therefore the way it sounds. It also adds frequencies that

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were not originally present. A reasonable benchmark for an acceptable
level of distortion is 0.1%. This means that the level of the frequencies that
are added by distortion is 1,000 times lower than the level of the original
signal. This is the same as a difference of 60 dB.

To get a feeling for this, it is worth trying an experiment if you have

digital audio workstation software already. Play some music through
your system with the fader set at 0 dB. 0 dB means no cut and no boost.
Set a comfortable listening level on your amplifier or on your headphone
volume control. Now lower the fader to -60 dB. This would be the level of
distortion mixed in with the signal if the distortion percentage was 0.1%.
This is very low and few people would be able to notice it, but in fact it
was possible to achieve 0.1% distortion decades ago. These days, electronic
circuit designers can achieve much better distortion figures, so the amount
of distortion generated in a microphone preamplifier that is competently
designed is so low that it is almost not there. Once again, this is not rocket
science, this can be achieved with standard design techniques.

Noise is a problem with microphones and microphone preamplifiers. The

reason for this is that microphones have a very low output level. Electrical
signals are carried through cables by electrons in the copper wire, but the
electrons have a random vibration of their own. That vibration gets mixed
in with the signal and we hear it as noise. It is there whether we like it or
not and there is nothing the circuit designer can do to remove it, so the
electronic circuit designer’s job is not to make the noise any worse.

Because of this intrinsic noise caused by the electrons themselves, there is

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a limit to the noise performance of a microphone preamplifier. This is a law
of nature that cannot be broken.

In microphone preamplifier specifications you will see a figure for

‘equivalent input noise’ or EIN. To find the equivalent input noise you
would measure the noise level at the output of the preamplifier and
subtract the amount of gain that has been applied to the signal. The lower
theoretical limit for equivalent input noise is in the region of -129 dBu.
What that means is that the noise is 129 dB below the standard reference
level that we use, which is 0.775 volts. A good microphone preamplifier
can approach that lower theoretical limit within 2 or 3 dB. And we are not
talking about an expensive preamplifier here, we are talking about one that
can be built with standard components at very low cost. So, if you compare
a preamplifier that approaches the lower theoretical limit within 3 dB with
another one that approaches the lower theoretical limit within 2 dB you will
find it very hard to tell the difference in terms of noise level.

What this is telling us is that in this day and age it is straightforward to

design and build a microphone preamplifier that is very accurate, and it can
be done at low cost. So it is likely that the microphone preamplifier that is
in your audio interface already is as accurate as you could ever possibly
need. Paying more money for a preamplifier is not going to bring you an
audible improvement in accuracy. In fact, of all the things that you can do
to improve your recordings, changing your microphone preamplifier is one
of the things that has the least effect.

There is an exception, however. What we have been looking at so far

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is accuracy. Accuracy is objective, it can be measured. Microphone
preamplifiers can be measured to see how accurate they are.

But what if you said, “Let’s not bother about accuracy, let’s just have a
preamp that sounds good”?

It is a fact that a degraded frequency response never sounds good. It is

a fact that a poor noise performance never sounds good. But a little bit
of distortion in a signal can often be perceived as an enhancement to the
original. This slight distortion is often referred to as ‘warmth’. Warmth has
a subjective quality that can’t be measured.

Different circuit components exhibit different kinds of warmth and so do

different circuit designs. The classic example is the vacuum tube. Vacuum
tubes were the only amplifying devices available until the invention of
the transistor. The transistor is very good for making accurate electronic
circuits. In fact they have to be accurate because if a transistor circuit is
poorly designed and causes distortion, that distortion sounds terrible to
the ear. There is nothing nice about transistor distortion. Vacuum tube
distortion, however, is a totally different thing. It is perfectly possible to
design an amplifier using vacuum tubes that is very accurate. But you may
not want that because a little bit of vacuum tube warmth can sound very
nice indeed.

So here we are largely out of the realm of objective measurements. Yes, we

still want a wide frequency response. Yes, we still want low noise. Yes, we
still want reasonably low distortion, but that small amount of distortion will
give us a warmth that is a totally subjective effect that cannot be judged by

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any measurement technique.

So, we can separate out microphone preamplifiers into two types. One is
the type that strives for accuracy, and it will use transistors, field effect
transistors or integrated circuits as its amplifying devices. Because this is
very well known science you can expect the level of accuracy to be very
good, even at a low price point.

Then we have the microphone preamplifiers with ‘character’ and these

will be the vacuum tube preamplifiers. They will sound different to the
accurate preamplifiers and you may prefer that sound. Also, one vacuum
tube preamplifier will sound different to another. Which one you prefer
is your own personal choice. No measurement can tell you which is best
and nobody else can tell you which to prefer. You have to make your own

In summary, the microphone preamplifiers in your audio interface, or

in your mixing console if you have one, are in all likelihood perfectly
adequate to make recordings of professional quality. Any time you spend
worrying about this point would be better spent worrying about something
else. But when you are confident with recording, you are in control of your
equipment and have a good appreciation of the sounds that you are working
with, then you will aspire to have a character vacuum tube preamplifier
- perhaps more than one.

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Universal Audio Solo/610 vacuum tube microphone preamplifier

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Control Surfaces

One interesting question is whether you should have a conventional mixing

console in your own home recording studio, either an analog console or a
digital console. The answer to that is probably not. Not when you are just
starting out. Somewhere further down the line you might decide that having
a mixing console will be advantageous, but that will be at a point when you
know exactly what you want to do with it.

There is always the possibility however of adding a control surface to your

system. The control surface works with your digital audio workstation
software ,so rather than having to adjust levels by dragging a fader with a
mouse, you can do it with a real fader. This mimics one of the key features
of the conventional mixing console.

With a conventional mixing console, all the controls are laid out right
in front of you and when you want to adjust something you just reach
out and adjust it - it becomes instinctive after a while. With digital audio
workstation software however, you always have to look closely at the
screen, direct the mouse pointer at exactly the right square millimeter and
then concentrate on what you are doing to make the correct adjustment.

On a conventional mixing console it’s perfectly possible to adjust two

controls at the same time. You might want to do this when you’re balancing
two instruments together, a little more of one, a little less of the other. You
can adjust both instruments at the same time in real time on a physical
console, whereas in software you would have to adjust one then adjust the
other, and then fine tune one and then fine tune the other, and go backwards

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and forwards like this in a completely non-instinctive way.

So the convectional mixing console can be instinctive and intuitive. Digital

audio workstation software is never completely that, even for experienced
users. You always have to think before you act. The control surface restores
some of the intuitive aspects of operation.

Control surfaces come in a range of sizes. A small one may have eight
faders, a large one might have twenty-four, and professional ones can be
even bigger. When you have twenty-four faders then it is feasible to have a
fader for every track that you record. This makes mixing a very natural and
comfortable operation. Just balance the faders until it sounds right. If your
control surface only has eight faders this doesn’t mean that you can only
control eight tracks because you can swap the faders to different groups of
eight tracks. There are usually convenient buttons to do this. Because those
faders will be motorized, then whatever the positions of the faders on the
screen of your computer, the positions of the faders on the control surface
will move to match.

Using a small control surface in this way is not quite as easy as using
a control surface with more faders but it is a workable and reasonably
intuitive solution. One feature to look out for, and it is a useful advantage
if you can get it, is the ability to lock one or more faders. So on a small
control surface you might lock one fader to the lead vocal, which is
something that obviously is very important, and then when you swap
groups of tracks on the other faders, the lead vocal will always stay where it
is. Not all control surfaces have that feature however.

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Euphonix MC Mix 8-fader control surface

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Some control surfaces are made by the developers of the digital audio
workstation softwares themselves. In this case you can expect them to be
tightly integrated, to have all the controls that you need and not to have
any superfluous controls that don’t do anything. There are also third-
party control surfaces which can be used with a range of digital audio
workstation softwares. You must check which softwares are compatible.

With third party control surfaces you might not be able to control all
the features of your software you would like to. Likewise, there may be
controls on the control surface which do not match to any control on the
screen of your computer. Also there is the question of how to control plug-
ins. One plug-in can have ten, fifteen or more controllable parameters. You
will use different plug-ins on different tracks and even quite a simple mix
could involve twenty plug-ins or more.

Some control surfaces manage this reasonably well. The controllable

parameters of plug-ins are placed beneath your fingers in a logical way and
you can easily switch from one plug-in to another to make adjustments.
With other control surfaces however, control over plug-ins might not be
quite so easy, so logical or so comprehensive. You could find yourself
returning to the mouse to adjust plug-in parameters via the computer

One important question you have to ask yourself about control surfaces is
are they worth the money? When you buy your digital audio workstation
software it comes with all its functionality built-in. You can do anything
that the software is capable of using the keyboard and mouse of your

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computer. Having a control surface will not allow you to do anything
extra. The only one thing that you can say for it is that you can adjust two
parameters at once but even then it might not be the combination of the two
parameters that you really wanted to adjust.

So in essence, the control surface is all about how you feel when you’re
interacting with your music. Do you like to have your hands on physical
controls or are you happy watching the screen very closely and using the
mouse and the keyboard? Some people can record and mix perfectly well
using the keyboard and mouse. They tend to have the ability to imagine the
sound that they want and then set about achieving it. With a control surface,
you are that little bit more free just to ‘feel your way’ around the track and
experiment, and not plan so much in advance. The only way you will really
know whether a control surface is for you is to try one out for yourself.

One odd point about control surfaces is that the digital audio workstation
software you buy might cost a few hundred dollars. The larger control
surfaces can cost a few thousand dollars. You could be spending that money
on more microphones, so think carefully.

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External Hardware

A few years ago, if you wanted to have a recording studio at home, your
equipment would be comprised of independent items of hardware. The
centerpiece of your studio would be the mixing console, traditionally a
large object taking up a lot of space in your control room - but impressive
for studio visitors. Indeed many commercial studios market themselves
on the basis of which mixing console they have. Along with the mixing
console, you would have external compressors. Only the biggest and most
expensive mixing consoles have compressors built-in, and even then you
would find reasons to use outboard compressors often.

Outboard equalizers are not so necessary because a mixing console has

an equalizer on every input channel. Even so, on a moderately low-cost
console these equalizers would be limited in function and it would be worth
supplementing them with an outboard equalizer, with better functionality
that could be used when necessary. For reverberation, there would be an
outboard digital reverberation unit and you could have one or more effects
units. Often these effects units would have reverb built in but quite often a
dedicated reverb unit could be superior in quality.

So every studio from large to small only a few years ago had an effects
rack, and in the effects rack would be processors such as equalizers,
compressors, noise gates and perhaps other dynamics control units, and
effects unit including reverb, delay, possibly pitch changing and others. But
now, clearly, all of these processes and effects can be obtained as plug-ins
which will work with your digital audio workstation software, and the state

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of the art has developed to an extent where these plug-ins are of very high
quality and they have excellent functionality.

What has happened therefore is that the base level of outboard hardware
equipment has almost disappeared because there is now no need for it. You
can get the same and better in the form of plug-ins. So if you have a full
suite of plug-ins working with your digital audio workstation software then
you can rest assured that there is no real need for external hardware. You
can make a recording and mix of fully professional quality just with your

It is human nature however always to seek improvements and when you

have a wonderful software studio with great plug-ins and you are indeed
producing work of professional quality that you are satisfied with, then you
will still want to go one better, and it is a definite possibility that you could
achieve a wider range of sonic possibilities by using external hardware
together with your software based system. There is a hesitation here in
saying that your work could actually be better because your software is
already capable of professional results. But it is the case that the range
of possibilities is wider so you could encompass a wider range of sound

Because plug-ins are so well developed now and they work to a state
of technical perfection, what we would be looking for in any outboard
hardware would not be any superior technical perfection because that
would not realistically be possible. What we are looking for however is
character and texture. We want something that is different to technical

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perfection but is pleasing in some way. And for that, as in the case with
microphones and microphone pre-amplifiers, we have a tendency to look
back to the equipment of the past, particularly equipment from the 1950s,
1960s and perhaps even 1970s.

You could ask the question – why do we have to go into the past? Why
can’t we find the answers today? Well the fact is we could find the answers
today if we wanted to, but what we are trying to achieve here is something
a little bit different. The sounds of the past are rooted in our consciousness
and in our memories. We’ve all heard recordings by Elvis Presley. We’ve
all heard recordings by The Beatles. We’ve all heard recordings by a whole
range of artists that were made on equipment that was designed in the
1950s, 1960s and 1970s and we like those sounds. It may be perhaps that
if you found the performance of a particular artist pleasing, then a large
part of the reason for that would be the intrinsic musicality of that artist.
But also lumped in with that is the sound of the equipment that they were
recorded with, and because the state of the art was not so well developed in
that era, the sound quality is subject to certain degradations.

We always like to have a wide frequency response, we always like to have

low noise, but distortion here is the key. Distortion comes in a whole range
of subjective qualities from extremely warm and pleasant and enhancing to
music, to harsh and destructive and completely undesirable. And within that
whole spectrum of possibilities there are certain sounds that are without
doubt pleasing to the ear.

So, looking at the equalizer for instance, the ways that they made equalizers

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back in the old days were not necessarily the same as they make them now.
For instance, in the old days an equalizer might have been designed using
coils of wires known as inductors. Inductors let low frequencies through
while blocking high frequencies, so they are a useful means of control.
Inductors have their own particular sound because unlike other circuit
components such as resistors and capacitors, inductors can very easily
create distortion. If an inductor, for instance, has a metal core to make it
more efficient, then that core becomes magnetized. If it is magnetized too
strongly then it can create distortion, and the character of that distortion is
different to the character of distortion that you would get from, for instance,
a vacuum tube.

So present day manufacturers have realized that there is a lot of inspiration

to be had from the past, and a range of equipment is available which uses
these old techniques but brought up to date. The idea is that the elements
that were good about the old equipment are kept and anything that was bad,
such as noise for instance, is corrected and, although it would be possible to
create a plug-in emulation of a vintage EQ for instance, there is always the
feeling that it doesn’t quite sound like the real thing. There is a lot of truth
in this. If you put a plug-in emulation next to the actual equipment that it
is meant to emulate, you will quickly realize that although the plug-in gets
close and has the obvious characteristics well-captured, the subtleties tend
to get lost. It isn’t just a case that there are differences in sound quality,
it seems to be the case subjectively that some of the richness and texture
of the experience are lost. Certainly the feeling of operating real physical
equipment has something to it too, which cannot satisfactorily be emulated

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using a mouse and a keyboard interface.

We’ve talked about EQ but the key piece of outboard equipment for the
modern recording studio is very likely to be the compressor. The reason
for this is that the compressor is so good at adding life, warmth, punch,
vigor and sparkle to your music so you could have a plug-in that provides
technically accurate compression and that will give you some of these
benefits. Or you can have a plug-in that emulates a variable-mu vacuum
tube compressor and that will give you more of these benefits. Or you could
have a real variable-mu vacuum tube compressor in your effects rack and
that will give you the ultimate in what compression can be, and of course
there are more than one of these kinds of units on the market and they all
sound different to each other. So potentially there are few limits on the
range of possibilities that you can achieve through outboard hardware.

We have mentioned the variable-mu compressor. There is also of course

the diode bridge compressor, the optical compressor, the FET compressor
and the VCA compressor and, while you can find emulations of all of
these types as plug-ins, you can also go out and buy the real hardware
equivalents. Although this is costly, your work will benefit from the

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Standalone workstations

You don’t need a computer to make recordings. There are quite a number
of recording workstations available that require no computer connection at

Before we look at workstations themselves let’s think about computers and

what problems they might cause for your recordings.

Traditionally it has been perfectly possible to make recordings without

computers at all. Indeed, in the 1960s and 70s computers were not used for
audio at all outside of research laboratories. The fact that we can listen to
perfectly good recordings made in that era tells us that the computer is not
an essential recording tool.

So why do we use it? The answer to that is that professional recording

equipment has always been very expensive. The mixing console in a first
class recording studio, for instance, might have cost as much as $400,000.
Smaller mixing consoles more suited to home recording studio use have
been made, of course. Still, they were expensive compared to the amount
of money that most people wanted to pay.

Multitrack recorders have also been expensive. A multitrack analog tape

recorder as used in a first class professional studio once would have cost
in the region of $50,000 to $60,000. And then enterprising manufacturers
brought out smaller versions of the multi track recorder which you could
buy for something around $10,000. Clearly we are still talking about a lot
of money. And to get any recording equipment that did fit comfortably into

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a home studio budget, well the facilities were extremely limited and the
sound quality was poor.

So, in the era prior to the computer you had a choice. Pay not very
much money and have an inadequate studio capable only of rough demo
recordings. Pay quite a lot of money and have a semi-professional facility
which was capable of reasonably good recordings, although maybe not
quite in the first class. Or you could hire a professional recording facility;
you certainly wouldn’t be able to afford to buy that kind of equipment.

The computer crept gradually into our lives in recording. Firstly it was
used for MIDI sequencing, where the notes that you play on a MIDI
keyboard are recorded into the computer, but only the key press data, not
the audio. But then that can be used to replay your music using MIDI
sound modules and samplers.

Initially that was all the computer could do, but gradually it acquired audio
capabilities as well. Firstly it became capable of editing stereo audio, and
people would pay a huge amount of money for the necessary software and
interfaces to be able to do this. But then as technology improved it became
possible to record more and more tracks and eventually have multitrack
recording. The cost of the hardware came down and eventually we came
to a point where the computer was fully capable of performing a complete
multitrack recording and mixing it. And what’s more, it could do that to
an equivalent standard of the equipment in a fully professional recording
studio from only a few years earlier.

So to summarize this, the reason we use the computer is because it saves us

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Why are computers inexpensive compared to professional recording

equipment? The answer to that is that professional recording equipment
only has a very small market. Even in the heyday of commercial recording
studios, top class facilities were only numbered in the hundreds in the entire
world. So a manufacturer of mixing consoles, for instance, only had a
small market to sell their products into. Therefore their products had to sell
for a sufficient price to sustain their business. And that applied to all the
other types of equipment as well.

With the computer, however, the computer is a general-purpose machine

which you can use for a whole range of tasks. So computers sell into
offices, in quantities which are thousands of times greater than they sell
into recording studios. So we have economy of scale here; computer
manufacturers have such a huge market that they can make their products
really quite cheaply, so although it might seem expensive to you to buy a
computer for, say, $2,000, in reality the capability that you’re getting inside
that computer is incredible.

There is a drawback to this. The computer is a general-purpose machine.

This means it is not specifically dedicated to audio, and because audio is
a small market it sometimes hasn’t received the attention that we ideally
would like. So the computer is optimized for general office tasks because
there are vastly more computers used for that kind of function than audio.

The computer is excellent for accessing the internet. The computer is also
very good at graphics, even though that is a fairly minority application at

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a professional level. The reason why a computer is good at graphics is
because the file structure of an image isn’t that much different to the file
structure of a word processor document. They both contain data and the
computer can handle data without caring what kind of data it is.

Except for audio. Audio is a different kind of data because it is time-

dependent. There is no time dependency in a word processor document.
You want it to open quickly, scroll quickly; you want the computer to be
responsive to your key presses, but the computer is perfectly adequately
fast to respond as quickly as you could possibly wish. It’s the same for
graphics files. A graphics file can be much, much bigger than a word
processing document so it might take a little bit longer to open. It might
take a little bit longer to perform whatever actions you want to perform on
that image. So you might find yourself waiting and wishing it could be a
little faster. But it will happen in the end. The file will open, the process
that you want it to carry out will finish and you will have the result that you
hoped for.

Audio is different because it’s time-dependent. If you record one single

audio signal from a microphone then it starts at time zero, finishes at time
five minutes, for instance, and time flows at a the normal rate between the
beginning and the end point of that segment of audio. So it means that the
computer has to keep up, and that is not the way computers are designed.
A computer, for instance, has to pay attention to your keyboard input.
The computer has to display graphics and information on the monitor.
The computer has to control data coming to and from the hard disk. The

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computer has to respond to demands of the operating system. And at the
same time it’s got to fit in your audio signal in real time with no delay. If
there is a hold-up at any point, your recording is ruined. And that’s just for
one audio signal. To try and do that for two tracks, four tracks, eight tracks,
16 tracks, or today 32 tracks, it gives the computer a lot of work and it isn’t
optimized for signals.

Fortunately the sheer speed of modern computers mostly gets round that
problem, but it is a rare computer user who has never seen a window pop
up on the screen saying that the audio can’t play because the disk is too
slow, or something is too slow. That would never happen in equipment
designed purely for audio.

The other problem with the computer is it is so complex. Computers

have become more and more capable, but at the cost of more and more
complexity. So in the 1990s, for instance, it would be possible to own a
computer and examine the files on the hard disk and know exactly what
every file was there for. Data files, operating system files, program files,
the lot. We couldn’t possibly expect to do that now. Even with just the
operating system installed and basic application software there are so many
files on the hard disk that no one person could ever possibly understand
what they all do, but they all do something. Well, we believe they do.

This complexity leads to a certain amount of unpredictability and

unreliability, so you might find that your computer is running perfectly
fine one day and the next day, for no particular reason, it’s not giving you
the performance that you want. This might be, for instance, because the

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operating system has automatically updated itself and some element of that
update is clashing with your application software. That can happen. Also,
it has to be said, some computer users are their own worst enemies. As has
been said elsewhere, you should dedicate a computer to music. You should
install the operating system and your music applications, and keep it clean
of everything else. But many people pay little attention to what they install.
They’ll buy a magazine and it has a CD or a DVD with some software on
it and they will install that software on their computer, or even download it
from the Internet. This is, quite frankly, begging for problems to occur. If
you can keep your computer clean then it is much more likely to give you
reliable performance.

There are many people who find the complexity, the unpredictability and
the unreliability of computers off-putting and they would like to get away
from using a computer for music completely. And these people are ideal
candidates for standalone recording workstations that don’t require a
computer connection, and this is why they are fairly popular.

Let’s look at what a standalone recording workstation will do for you and
later we will consider the potential disadvantages. A standalone recording
workstation is a one-piece unit that combines mixing and multitrack
recording facilities. Sometimes they can also mix the output onto a CD
in an internal CD burner. Some of these units are quite small and may be
capable of perhaps eight tracks recorded onto the internal hard drive. The
hard drive itself would be capable of recording many more tracks but the
physical makeup of the equipment is such that it has been designed as an

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eight track unit and you will probably see eight faders on the top panel, one
for each channel, one for each track.

You will be able to record signals from a microphone or a line level source
onto the hard disk. You will be able to monitor while you overdub more
instruments and vocals. Ultimately, when you have filled all eight tracks,
you will be able to make a mix using EQ and perhaps even compression,
and perhaps reverb as well, provided as standard.

A small unit like this will be capable of very good sound quality; there’s no
reason why it shouldn’t. However, you can see from the limited number
of tracks that it is only suitable for simple recordings. If all you want to
record is voice and guitar then that’s fine; you don’t need anything more

But if you want to record a full band then clearly this kind of small unit is
not appropriate. You should also consider that just because a unit has eight
tracks, or however many tracks, you can expect it to have that many inputs.
This is usually not so; these devices usually have fewer inputs than tracks,
so you can’t record all eight tracks at once.

Moving up to the larger units, there are standalone recording workstations

with 24 tracks, for instance. And with 24 tracks it is indeed possible to
produce a richly textured recording. In fact if you can’t produce a good
recording with 24 tracks then it is saying something about you as an
engineer rather than the limitations of the equipment.

With more tracks you can also expect more sophistication, so the provision

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of EQ, compression, reverb, etc. you can expect to be of a higher standard.
A fully professional standard in fact, and as good as any of the standard
plug-ins you would receive with digital audio workstation software on your

Although there are several manufacturers of high end standalone recording

workstations, there is one that deserves a special mention, and that is
Yamaha. Other manufacturers may make equipment that is as good as
Yamaha’s equipment, but the one thing that Yamaha does is that it also
makes equipment that is fully professional. So if you go to a stadium rock
concert you wouldn’t be surprised to see a Yamaha mixing console. If
you go into a large broadcasting organisation you wouldn’t be surprised
to see Yamaha mixing consoles in there too. So the advantage of buying a
Yamaha standalone recording workstation is that you will learn the Yamaha
way. If you want to progress your career in sound engineering, into live
sound or broadcasting in particular, then knowing the Yamaha way will be
like having a foot on the first or second rung of the ladder, whereas if you
know some other manufacturer’s way there is no progression from that.
This may or may not be important to you, but it is certainly something
worth considering.

So the advantage of the standalone recording workstation is that you

decide to do some recording, you switch it on, you record, everything
goes smoothly with no problems, and you don’t have to use the analytical
parts of your brain in the same way as you do when you’re working a
computer. The drawback is that the quality of the end product is likely to

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be a little less than if you had used digital audio workstation software on
your computer. This is not because of frequency response, distortion or
noise. It is not because of any defects in the provision of EQ, compression
or reverberation. But in general the standalone recording workstation can’t
match software for detailed editing and manipulation.

So, for instance, if you wanted to replace one note of a take in your digital
audio workstation software, that would be no problem. If you wanted to
do the same thing on your standalone recording workstation you might find
it time consuming, difficult, or impossible. With the standalone recording
workstation you are far more reliant on the intrinsic quality of your
musicianship and performance to get things right as they go down onto the
disk. With the computer you have very much more opportunity to modify
things after they have been recorded.

Of course there is no reason why you can’t have the best of both worlds.
Some people do this. You could have a standalone recording workstation
for your initial recording. You would do this because you’re free from the
computer, it’s an instinctive process to record and you can concentrate on
your music more. But then, when you’ve finished the actual recording
process and you’re coming into editing, manipulating and mixing, you can
transfer the files to a computer. Some workstations make this easy; with
some you would have to use a more long-winded manual process.

In summary there are many reasons to say why using a computer with
digital audio workstation software is the way to go. But for some people
that is not the right solution. Standalone recording workstations will

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probably always be in the minority but it is likely that they will continue to
have a place in the home recording studio.

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You may have a domestic hi-fi system with good loudspeakers and you
listen to it for the pleasure of listening. If the speakers sound good to
you and you enjoy the music that you listen to then that’s OK. You have
everything that you need. But monitoring in the recording studio is very
different to listening for pleasure. Firstly, you will be listening to your own
music, or your own recordings of other people’s music, rather than listening
to recordings made by someone else. Secondly, you need the monitoring
system to tell you whether your recordings are good enough.

So for a home hi-fi system there is absolutely nothing wrong with buying
a pair of loudspeakers that flatter the music, that make it sound better and
allow you to enjoy it more. For recording studio monitoring you need a
pair of loudspeakers that tell you the truth, warts and all.

You see, if your recording studio monitor loudspeakers flatter your

recording then you will think that your recording is better than it really is,
and you will declare yourself satisfied at a point before the recording is
really finished. Then when you give that recording to someone else, send
it to a record label perhaps, they will hear it without the benefit of your
flattering loudspeakers. They will hear it as it really is and they may not
like what they hear.

So it is vitally important that your recording studio monitor loudspeakers

tell you the truth, the whole truth and nothing but the truth. And if the truth
hurts, well, you just have to develop your recording skills further so that
you can make your recordings better.

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Professional engineers always monitor on loudspeakers. There are two
reasons why a professional engineer might occasionally use headphones.
One is because the artists in the studio are using headphones and he or she
needs to be able to hear what they can hear on the headphones, so it’s just
a quick check. Secondly, sometimes you need to be able to listen in very,
very close detail to something if you suspect there might be some noise or
distortion on a recording for example. And if you listen on headphones
then you can hear more detail than you can on loudspeakers. So just for
that check it is worth having a pair of headphones around.

The problem with using headphones when you’re mixing is that what
sounds like a good mix on headphones doesn’t necessarily translate to a
good mix on loudspeakers. No-one really seems to know the reason for
this, but it is certainly true. You can learn to mix on headphones quite well
but when you take the headphones off and listen on loudspeakers you will
hear things that just need that final tweak to make them sound good. Oddly
enough mixes that sound good on loudspeakers also do tend to sound good
on headphones, so fortunately it does work that way round, which is just
as well considering the vast number of people who listen to music through
headphones on their MP3 players.

So headphones in the studio will be useful, both to you and any artists who
record in the studio. When you are recording vocals, for example, your
singer will wear headphones to listen to the backing track while singing
into the microphone. The reason that headphones are necessary is so that
the backing track doesn’t get recorded onto the vocal track, otherwise that

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would make the recording difficult or impossible to mix.

There are two types of headphones and both have their uses in the studio.
One type of headphone completely encloses the ear so that the sound
coming from the ear pieces is completely isolated from the outside world.
This has the advantage that very little sound comes in from the outside
world and also very little sound escapes into the outside world. This is
useful when you are recording vocals because you don’t want the sound
from the headphones to leak into the vocal microphone. So by using
headphones that completely enclose the ear this problem can be minimized.

The downside of this type of headphone is that the enclosed space created
forms a resonating chamber which will accentuate certain bands of
frequencies. So the sound quality of the completely enclosing type of
headphone is not the best. It certainly can be adequate, and for headphone
monitoring while recording through a microphone it works perfectly
well. But for the best sound quality it is necessary to use the open type of

Firstly, they don’t completely enclose the ear; they sit on top of the ear.
Secondly, the back of the headphone is more open to the air than the
completely enclosing type, where the back is firmly closed. Clearly this has
the disadvantage that there will be some leakage from the headphones and
if you use this type of headphone while you’re recording with a microphone
then, yes, there will be spill from the headphones into the microphone; that
is unavoidable.

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The advantage is, though, that the sound quality of the open type of
headphones is very much better. It’s a clearer sound, it has a better balance
of frequencies, and it is just more natural. So therefore if you have the
need to use headphones to make any kind of subjective judgement of audio
quality then the open headphone is the better choice for that purpose.

Audio interfaces and mixing consoles commonly have headphone sockets,

so you could easily find that you can just plug your headphones in without
having to pay any special attention to acquiring that facility. This is fine if
you’re working on your own, but if you’re working with someone else then
there will be occasions when both of you need to listen on headphones. So
if you only have one headphone socket and you need to drive two pairs of
headphones, what do you do?

The answer to that is to get a splitter, which comes in the form of a small
connector which you plug into the headphone socket and the connector
itself has two sockets to which you can plug in both pairs of headphones.
A two-way headphone splitter like this is effective. One thing you should
look out for, however, is that both pairs of headphones need to be identical.
If they are different then one pair will be louder than the other, sometimes
quite significantly louder.

Also it’s possible to have two pairs of headphones that look identical but
they have different impedances. This means that one will naturally draw
more current from the output than the other, in which case the one that
draws more current will be louder. So it is important to check that both
pairs of headphones are the same make and model and also that they are the

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same impedance, and then they will both be at the same level.

A two-way headphone splitter is practical. It is possible to buy or

make headphone splitters of three ways, four ways, five ways, six ways
perhaps, but there are two problems. Firstly, does the headphone output
have enough power to drive six pairs of headphones? It depends on the
headphone output and it depends on the headphones. It may be possible, it
may not be possible; you’re not going to find out until you try

There is another problem, however. With so many pairs of headphones

plugged into a headphone splitter there is a strong chance that one pair
won’t be plugged in properly. This is so easy to happen with jack plugs and
sockets. And if one pair isn’t plugged in properly, it may prevent all of the
headphones from working. And when there are so many headphones and
cables and people about it is just so easy for one cable to get pulled slightly.
It’s only a couple of millimeters that makes the difference; you really have
to look hard to see it.

So really, when you require more than two headphones to be in use at the
same time then you need a proper headphone amplifier. Fortunately these
are not too expensive. You can buy a four-way headphone amplifier, which
has separate sockets for each pair of headphones and also has separate
volume controls for each pair of headphones, so it cures all the problems
you could potentially face. There are other ways of having multiple
headphones in the studio, but the headphone amplifier really is the simplest.

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4-way headphone distribution amplifier

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Let’s move on to loudspeaker monitoring. In the old days of recording,
some would say the classic days, it was normal for a recording studio
to be fitted with a huge pair of monitor speakers. These would usually
be mounted into the wall, driven by a big enough amplifier. These
loudspeakers could produce an enormous quantity of sound.

There were two arguments for this; firstly, the artists, or the band, need to
react to the sound that they produce. Everyone likes loud music and if the
music is loud in the recording studio then it makes everyone feel better, and
they can perform better and end up with a better recording.

The second reason for having large loudspeakers was that engineers were
concerned that the speakers that they used should be more accurate than
the speakers that people use at home, which is true. And they wanted to
be able to hear the low frequencies of the recording in detail. It is a fact
that to produce low frequencies well at adequate volume you need large
loudspeakers. The laws of physics dictate that small loudspeakers cannot
reproduce low frequencies as well as large loudspeakers can.

So, if an engineer monitors on small loudspeakers, there could be

something really horrible going on in the low frequency end of the
recording that he wouldn’t hear. And then that recording might get
pressed onto a compact disc, it gets sold to a real hi-fi enthusiast who has
an excellent pair of large loudspeakers at home, and he can listen to the
recording and hear that problem that the engineer missed.

So for these reasons, recording studios used to have large loudspeakers.

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But there is a problem with large loudspeakers. Although it is desirable
to have loudspeakers that tell you the truth, the whole truth and nothing
but the truth, it is quite possible that the studio’s loudspeakers are so much
better than the average customer’s loudspeakers at home that the mix that
the engineer makes in the studio really doesn’t translate very well to a
lower quality playback system.

So for this reason it became common, particularly during the 1980s, rather
than use the main studio monitors engineers would mix using so-called
near-field monitors. A near-field monitor is a smaller loudspeaker that is
placed very much closer to the engineer’s ears, at about arms’ length in fact.

The expression ‘near-field’ means that the loudspeaker is so close to the

engineer that he or she hears more direct sound from the loudspeaker than
reflected sound from the walls of the control room. This means that it is
possible to ignore the acoustics of the control room and just concentrate on
the direct sound. It might seem odd, but the sound source doesn’t have to
be very far away from you before you hear more sound reflected from the
room than the direct sound from the source. So that is the reasoning behind
the near-field part of this arrangement.

Having small loudspeakers meant that the engineer was listening to an

effectively smaller sound, a sound which is more similar to what the
listener will hear at home on an average system. So the engineer is basing
his or her judgement on something that is much closer to the listening
experience. Engineers will say that you have to work harder to get a mix
sound good on near-field monitors and that extra work will translate to a

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mix that sounds better at home.

One thing that there never has really been full agreement on is how good
quality should these near-field monitor loudspeakers be? If they are
wonderful in quality then surely the engineer is getting a different sound
to people listening at home. So perhaps near-field monitors should just be
average loudspeakers, or perhaps they should actually be loudspeakers that
are sold into the domestic hi fi market.

This is, in fact, often done. The most famous near-field monitor
loudspeaker during the 1980s, when near-field monitoring first became
popular, was a hi-fi loudspeaker, and it didn’t sound particularly good.
Nonetheless it proved a very effective mixing tool and countless mixes
were made using this near-field ‘monitor’ loudspeaker.

One thing that can probably be said with a degree of truth is that if you
want your near-field monitors to be imperfect to reflect the imperfection of
inexpensive domestic hi-fi loudspeakers, then they need to be ‘averagely
imperfect’ so that they are reasonably representative of the majority of hi-fi
loudspeakers. So you can’t just pull any pair of hi-fi loudspeakers into the
studio and expect them to make good monitors. You would have to be sure
that you had selected a pair that had an average range of defects.

There are several ways to design a loudspeaker. One way is to completely

enclose the drive units inside the cabinet. So all of the rear radiation from
the drive units, for the low frequency drive units in particular, is contained
within the cabinet. If it were allowed to escape then it would simply cancel
out some of the sound coming from the front of the drive unit.

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The closed box loudspeaker, sometimes known as the acoustic suspension
loudspeaker, performs very well; it is a good design. But one problem is
that the diaphragm of the loudspeaker now has to fight against the air that
is trapped inside the cabinet. Air is springy, so when the diaphragm of the
drive unit is pulling in, the air inside is compressed and tries to push it back
out again. When the diaphragm is pushing out into the open air, the air
inside the cabinet is rarefied and the air pressure outside tries to push the
diaphragm back in again. The net result of this is that the low frequency
response of the closed box loudspeaker is not as good as one might want it
to be.

But there is a solution, and that solution is to open up what is called a port
in the box. The port is an opening in the cabinet connected to the inside
by a tube. The tube could stick out of the box but for convenience the tube
sticks into the box. This creates what scientists call a Helmholtz resonator
and the cabinet can be tuned to resonate at a certain range of frequencies.
‘Resonate’ means that the air inside the cabinet will vibrate easily over that
range of frequencies.

So, if the designer tunes the cabinet to resonate at some range of low
frequencies at a point where the response of the drive unit has started to fall
off, then that in fact can lift the low frequency response of the loudspeaker
as a whole. So if you compare a closed box loudspeaker and this other type
of loudspeaker, which is called a bass reflex, of the same size, then the bass
reflex loudspeaker will have more bass.

Because of this, the majority of loudspeakers sold into the domestic market

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are bass reflex loudspeakers. There is a drawback that the bass can become
rather boomy, so whatever energy you put into the loudspeaker in the
low frequency area gets funnelled into the band of resonant frequencies.
In extreme cases it can seem that whatever you play through those
loudspeakers, the bass only has one note. So the bass reflex loudspeaker
has more bass than a closed box loudspeaker, but you can’t expect the
sound quality in the low frequency end to be quite as good.

So, to bring this into recording studio context, which should you choose?
Should you choose a closed box loudspeaker which will give you a more
accurate sound quality and tell you what is on your recording, or should
you choose a bass reflex loudspeaker which in fact is going to flatter the
base content of your recording?

This is one situation where we have to go against the doctrine of nothing

but the truth. If you monitor on closed box loudspeakers, you are hearing
something that is different to what the vast majority of customers who will
buy your recording will hear. So you don’t really know what it’s going to
sound like on their systems. It will probably have too much bass because
their speakers have more bass than yours. So for this reason it is usually
advisable to choose bass reflex speakers as your studio monitors so that
you’re more in line with potential listeners to your music.

Let’s turn to the amplifier, because the amplifier obviously is the other
component of the monitoring system. Once again the amplifier does not
directly affect your recording, but since it affects what you hear then your
reaction will, to a certain extent, be based on anything the amplifier does to

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the sound. So the amplifier needs to be accurate.

Power amplifiers are known science. Electronic circuit designers can easily
design a power amplifier with a fully adequate frequency response, ultra
low distortion and ultra low noise. The only questions that are left are how
much does it cost, how big is it, how reliable will it be, and does it have a
fan (the fan will create a little bit of noise)? And, of course, how powerful
should it be?

You probably know already that amplifiers are rated in terms of watts. The
watt is the unit of power. So how many watts do you need for an amplifier
in your home recording studio? The answer to that largely depends on the
loudspeakers. Some loudspeakers are more efficient than others at turning
amplifier watts into actual sound watts in the room. So if you have a pair
of loudspeakers that is inefficient you will need a more powerful amplifier.
As a guideline, however, in a small room it is likely that an amplifier that
is capable of 100 watts per channel will be fully loud enough for any
reasonable level that you could require on near-field monitors.

You also have to consider the power handling capability of the

loudspeakers. If you look at the back of a loudspeaker it will give a power
rating in watts. What this means is that you can feed that much power in
terms of an average music signal into that loudspeaker and the loudspeaker
will cope with it comfortably without blowing. If you put less than that
amount of power in the loudspeaker will be very happy, the sound level
will just be lower. If you put more than that amount of power in you risk
damaging the loudspeaker.

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But the odd thing is that you are more likely to damage a loudspeaker by
using an amplifier that is lower in power output than the rated capacity of
the speaker, than an amplifier that is higher in power output than the rated
capacity of the loudspeaker.

That sounds odd.

Suppose the loudspeaker is rated at 100 watts. Suppose you have a 50 watt
amplifier. You might think that there is no way on earth you could put too
much power into that loudspeaker and damage it. Conversely you might
think that if you connect a 200 watt amplifier to that loudspeaker you will
blow it up in a microsecond.

The way it actually works is this... A near-field monitor loudspeaker

will typically have two drive units; one for the low frequencies and one
for the high frequencies. There is actually very little energy in the high
frequencies of a normal music signal. Most of the energy is in the mid
and low frequencies. So if you put a music signal of 100 watts into that
loudspeaker it is likely that something like 80 or 90 watts will go to the
low frequency drive unit and maybe only ten or so watts goes to the high
frequency drive unit. Now, imagine a 50 watt amplifier. You might find
with a smaller amplifier that you’re struggling to get enough volume in the
room; it’s just not loud enough. So you increase the volume and increase it
and increase it and increase it, until you have what you feel is an adequate
level. Unfortunately you might have increased the volume so much that the
amplifier has gone into what we call ‘clipping’.

This means that you’re asking the amplifier to go louder than it is actually

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capable of. So it clips off the peaks of the wave forms, causing sharp
corners. You should be able to hear when the signal is clipped, but if you
ignore it trouble will follow. The sharp corners of the clipped peaks of the
waveform contain massive high frequency energy, much more than there
would be in a normal music signal. Now, all of this energy is going to the
high frequency drive unit, and it could easily be more than that drive unit
can handle, which is going to lead to a blown high frequency drive unit. So
for this reason it is not advisable to use an amplifier that is rated lower in
power output than the loudspeaker is rated in power handling.

Let’s consider a higher powered amplifier now. Let’s suppose a 200

watt amplifier is feeding a loudspeaker with 100 watts capacity. Clearly
this amplifier is capable of blowing that loudspeaker; there is no doubt
about that. But it is your fingers on the volume control that will prevent
destruction from happening. You will operate that amplifier well within
the loudspeaker’s capacity to handle power. And when you do this, that
amplifier is cruising; it’s not stressed in the slightest; there’s absolutely no
risk of clipping and it is delivering a perfectly clean signal. The only thing
you have to listen out for is signs that the loudspeaker is stressed. As soon
as you start hearing the loudspeaker producing distortion you need to back
off. The amplifier is still clean but the loudspeaker is telling you that you
can’t go on much longer before it will blow.

It isn’t so odd to operate an amplifier well below its rated power output
capability. Imagine you’re driving a car, you’ve bought a car with a one
liter engine. That car will go at 70 miles an hour (112 kilometers per hour),

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but imagine the kind of ride you will get, and how long that car is going
to last. It’s going to be worn out before you’ve driven it for more than a
couple of years. Now, imagine a car with a three liter engine. Imagine what
that feels like driving at 70 miles an hour. Smoooooth!

So the car with the one liter engine is at the limit of its capability, and it
shows. The ride is rough and the engine is strained. The car with the three
liter engine is capable of going much faster than 70 miles an hour, so 70
miles an hour is easy. The ride is smooth and the engine is comfortable.
It’s exactly the same with power amplifiers. And they don’t pollute the
atmosphere as much.

When you have a pair of near-field monitor loudspeakers that you need
to drive from a power amplifier, they’re known as passive loudspeakers.
Some loudspeakers, however, have amplifiers built inside them. These are
called active loudspeakers, which you can drive directly from the output
of your audio interface as long as it has a volume control. The benefit
of having the amplifier in the same package as the loudspeaker is that
everything is under the control of one manufacturer, so that manufacturer
can do everything in its power to take a line level signal and turn it into
sound in the room in the most accurate way possible.

With passive loudspeakers the designer of the loudspeaker doesn’t

know which amplifier you’re going to use. Even if it was from the same
manufacturer there are different ranges and there is no way of knowing
which combination you have. So the active loudspeaker is more
controllable from the design point of view and therefore you can expect it

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to be better than a passive loudspeaker.

However you wouldn’t choose a loudspeaker because it was an active

loudspeaker, you would choose it because you felt it was good for
monitoring in your recording studio. So if you find a loudspeaker good to
work with but it’s a passive loudspeaker then that’s fine. Buy an amplifier
and use it in the conventional way.

One problem that many home recording studio owners have is deciding
which loudspeaker to buy. How do you know what is a good loudspeaker
when you haven’t heard them? And even if you can get the opportunity
of hearing them, unless you have a great deal of experience in a recording
studio how do you know which one is going to turn out to be good for

The answer to that is that there are two good sources of information.
Firstly, the manufacturers that sell into the pro audio market. This is a
well established business and any manufacturer that wasn’t capable of
making products that would be accepted in the marketplace would have
gone bust a long time ago. So you will find that the manufacturers that you
commonly find in product catalogues actually do know what they’re doing,
and their products will be good. Of course it is always possible that one
manufacturer’s product might be slightly better than another’s but we are
talking about the difference between 89% and 91%. They’re already well
up there.

The other useful source of information is from established recording

engineers. It isn’t that you can just ring them up and ask them which

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monitors they use. But if you keep your eye on recording magazines then
commonly they will feature interviews with engineers, and the engineers
will say which equipment they prefer. This is golden information,
particularly if you can relate their choice of equipment to an actual
recording that they’ve made so that you can hear for yourself what results
that equipment is capable of. So, if a certain recording engineer who has
made a multi million selling record uses XYZ brand for monitoring then
that demonstrates that XYZ brand is capable of making multi million
selling recordings.

But what you will find, if you research extensively, is that different
engineers have their different preferences and you can find examples of all
the products in the pro audio catalogue being used to produce professional
products. What this is saying is that if you buy a pair of monitors of
professional quality then they will be capable of giving you good results.
Successful recordings are more about experience and skill than the actual
monitors that you use.

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200 watts/channel stereo power amplifier

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Nearfield monitor loudspeakers

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Shopping list
This is the equipment you will need, or need to consider, to set up your
home recording studio...

Microphone: At least one of professional quality which may be a dynamic

or a capacitor model. One dynamic and one capacitor microphone will
provide a contrast in sound qualities. Two identical microphones are
preferred for stereo. A large-diaphragm vacuum tube capacitor microphone
is preferred for vocals.

Microphone stand: Select a model that is sturdy, where the boom will not
sag in use.

Microphone cable: XLR to XLR cable, 5 to 10 meters in length, one for

each microphone.

Audio interface: Should have microphone inputs as well as line inputs.

Select a model with as many inputs as sound sources you intend to record
simultaneously. If you intend to use Digidesign Pro Tools LE digital audio
workstation software, you will require a Digidesign or M-Audio interface.

Digital audio workstation software: Apple Logic Pro, Cakewalk Sonar,

Digidesign Pro Tools LE, Mark of the Unicorn (MOTU) Digital Performer
or Steinberg Cubase.

Computer: Select a computer that is compatible with the digital audio

workstation software you wish to use. For reliability, you should dedicate
this computer to audio and not use it for anything else.

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Plug-ins: Start with the standard processing and effect plug-ins that come
with your digital audio workstation software. When you have mastered
these, consider whether you need additional plug-ins. A good convolution
reverb plug-in is always desirable.

Software instruments: Choose according to your own musical

requirements among synthesizer, sampler and sample replay instruments.

Sample libraries: Familiarize yourself with the library that comes with
your software sampler. When you feel that your library is not fulfilling your
needs, consider third-party libraries. Bear in mind that the best orchestra
libraries are extremely expensive.

iLok: Most plug-ins and software instruments are copy-protected. The iLok
key is a common method of copy protection.

Microphone preamplifier: The microphone preamplifiers in your audio

interface are probably sufficiently accurate for professional requirements.
Consider also purchasing a ‘character’ vacuum tube preamplifier.

External hardware: Explore the plug-ins that came with your digital
audio workstation software first, then consider third party plug-ins (which
are less expensive than hardware). After that, consider a vacuum-tube
compressor, then perhaps other external hardware units.

Control surface: Since the control surface doesn’t add any functionality
to your digital audio workstation software, it is suggested that you start
without one. Buy a control surface when you feel the need to control your
mix through physical faders.

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Standalone workstation: If you don’t like working with computers, the
standalone workstation might be better for you. Choose one with enough
inputs to record as many instruments simultaneously as you need. Also
consider the number of tracks.

Headphones: Choose a closed-back pair of headphones that completely

encloses the ears for monitoring while recording through a microphone.
Choose open headphones for better sound quality while bearing in mind
that they will leak some sound energy.

Headphone amplifier: A headphone amplifier is not necessary for one

or two pairs of headphones (for two, use a headphone splitter adaptor).
A headphone amplifier is advantageous where more than two pairs of
headphones are used simultaneously.

Monitor loudspeakers: Choose a pair of near-field monitor loudspeakers

of a make and model that is sold into the professional audio industry.

Power amplifier: If your monitor loudspeakers are active, then you do not
need a power amplifier. Otherwise, choose a power amplifier of at least 100
watts per channel. More then 300 watts would normally be excessive for a
small home recording studio. A power amplifier that has fan cooling will
make a slight noise.

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The purpose of this book is to help you choose recording studio equipment
that is going to give you the ability to produce recordings of professional
quality at home. We’ve done this, not by telling you which particular
items of equipment you should buy, but by telling you what the options
are, what the equipment does, how it works, what different sound qualities
are available. And now you should know, for instance, exactly why you
might choose a large diaphragm capacitor microphone or why you might
choose a small diaphragm capacitor microphone, or indeed even a dynamic
microphone. And this applies to all the other studio equipment and software

But here you have to make a choice. Many people who invest in home
recording studio equipment choose to make that an end in itself. So
although it would seem that the purpose of having a home recording studio
is to make music, for many people the purpose of having a home recording
studio becomes adding to it, improving it, upgrading it, changing it,
researching the latest equipment... In fact, anything but actually getting on
with the job of making music.

If this is what you like then that is fine. If the studio is the end in itself,
your ambition is to have the ultimate studio and you enjoy exploring the
technology, there is no reason why you shouldn’t do this - as long as you
understand what you’re doing, why you’re doing it, and the reason you are
spending your money in this way. But don’t let it happen by accident. It has
happened to many home recording studio owners who desperately wanted

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to have the ability to produce their own music, that they got sucked into
the technology and put all their effort into their studio and improving their
studio. So they never actually got around to making any music with it, or
perhaps they just played with it now and again and did little bits of things
but never really finished anything.

So if for you the emphasis is music, you must be clear. What you need to do
is to get your home studio up and running, get all the equipment you need,
get it set up so that is working properly. If at that point you find that there
are any inadequacies you should cure them as soon as you possibly can so
that you can reach a point where you can say, “My studio is finished!” and
at that point you should cut yourself off from sources of distraction that are
telling you to buy more and more equipment.

Ignore them and forget about them. Go into your studio, switch on the
equipment and make music. And tomorrow, go into your studio, switch on
your equipment and make music. And the next day, go into your studio,
switch on your equipment and make music. See, already we have three
days’ worth of music, whereas many other people have achieved nothing
more than three days’ more studio tweaking.

Hopefully you have enjoyed reading this book and hopefully it has made
equipping a home recording studio very much more clear for you. And
hopefully too, it will set you on the path to creating recordings of the
professional quality that you seek and deserve.

Thank you for reading.

Equipping Your Home Recording Studio Page 153