Get Started With Recording, Mixing & Mastering - Guide To Starting Your Home Studio

Keith Scully 2015 KeithScullyMixing.
com
1.
INTRODUCTION
Thank you for downloading your free copy of my new eBook. If you've been passed
this eBook from a friend then I'm glad you've got it too and I hope it helps you get well
on your way to making some great music.
If you're reading this then the chances are that you are a musician, songwriter or a
budding engineer or producer looking to learn about the process of creating music and
getting it ready for release to the outside world. Throughout this book I'll be giving you
some advice on setting up your recording or production space, choosing equipment
and connecting everything together, as well as recording various instruments and
vocals. I'll also run through the mixing process, the mastering process, and finally, I'll
touch a little bit on distribution and how to get your music out into the world quickly and
easily. As you progress through the book I'll also point out some either free or very
affordable options available to you when choosing some hardware or software for your
home or project studio. If you've already got a fully functional studio then thats fine and
you should use what you know, but if you're completely new to this and don't have any
equipment at all to start with then this should help with the decision making process
when you're purchasing your gear and stocking up on software.
I would like to point out that this eBook is not an instruction manual in any way. It is
simply here to provide you with a fundamental understanding of each part of the
process along with some principles you can follow to get your music out there quicker
and get it sounding better. Above all, there are no rules when it comes to producing
music, the most important thing to remember from beginning to end is to enjoy it!
v1.0
Keith Scully 2015 KeithScullyMixing.com

2.
CONTENTS
YOUR STUDIO....................................................................................................3
The Room........................................................................................................................3
Hardware.........................................................................................................................7
Computer.............................................................................................................7
Audio Interface.....................................................................................................7
Microphone..........................................................................................................8
Headphones / Loudspeakers...............................................................................9
Software........................................................................................................................10
Digital Audio Workstation (DAW).......................................................................10
Virtual Instruments.............................................................................................11
Plugins...............................................................................................................11
All The Choice Is Great, Isn't It?....................................................................................12
MAKING CONNECTIONS.................................................................................14
XLR................................................................................................................................14
Jack...............................................................................................................................14
S/PDIF...........................................................................................................................15
RCA...................................................................................................................15
TOSLINK............................................................................................................15
MIDI...............................................................................................................................15
Balanced vs Unbalanced Connections..........................................................................16
Line Level......................................................................................................................16
RECORDING.....................................................................................................17
Overview........................................................................................................................17
Microphone Polar Patterns............................................................................................17
Phantom Power.............................................................................................................18
Proximity Effect..............................................................................................................18
Recording Levels...........................................................................................................19
Phase............................................................................................................................19
The Recording Process.................................................................................................21
Scratch Track.................................................................................................................22
Recording Drums...........................................................................................................22
One Microphone................................................................................................22
Two Microphones...............................................................................................23
Recording Bass Guitar...................................................................................................24
Recording Acoustic Guitar.............................................................................................25
Recording Electric Guitar...............................................................................................26
Recording Vocals...........................................................................................................27
MIXING..............................................................................................................29
Overview........................................................................................................................29
Preparation....................................................................................................................29
Track Order........................................................................................................30
Colour Coding....................................................................................................30
Track Markers....................................................................................................30
Editing................................................................................................................30
Getting To The Mix........................................................................................................32
Setting The Balance......................................................................................................32
Using EQ.......................................................................................................................33

3.
Using Compression.......................................................................................................35
Using Reverb.................................................................................................................36
Using Delay...................................................................................................................37
Final Thoughts...............................................................................................................38
MASTERING.....................................................................................................40
DISTRIBUTION.................................................................................................43

4.
YOUR STUDIO
THE ROOM
Most of us do not have access to the facilities or the resources to work in a room that
was professionally designed and professionally treated acoustically. If you're reading
this book then there's a good chance you are working from your bedroom or spare
room at home with no real recording space, or if you're lucky enough you might have a
garage, basement or some other large room in which you intend to record various
instruments in, as well as carrying out mixing and mastering tasks.
If you're going to be working solely on headphones with no recording being done in the
room then you have a huge advantage here because the room itself will have little or
no impact on the music. However, if you intend to record anything in the room and/or
use loudspeakers for listening back then it is worth reading on.
Because there's such a large variety of rooms you could use for this purpose it would
be impossible for anyone to provide you with all the instructions on how to treat your
room. The shape of the room, the size of the room, the floor and ceilings surfaces and
the construction materials of the walls all come into consideration. Because these
aspects can vary so much all rooms will have different sound reflection properties and
therefore each room will have its own unique sound to it, which may be for the good or
the bad.
An untreated room will have an uneven frequency response and the room you work in
is very important if you are using loudspeakers for playback because the room will
'colour' what you hear from the speakers when you're mixing or mastering. What this
mean is that the reflections in the room interfere with the direct sound from the
loudspeakers. The result of this is that you can't accurately hear what's being played
through the speakers which makes it hard to judge how your mix will sound in other
rooms on other playback systems. It is also very important when recording in that
space too because your recordings will have the sound of your room printed on them,
which is likely not a good thing if you haven't made the efforts to even out the response
of the room itself.
One huge misunderstanding is that your room should be covered from ceiling to floor
with foam tiles to kill all reverberation, which is completely wrong. There's a pretty good
chance that nobody will be listening to your finished music in a room with no
reverberation, so it doesn't make sense for you to create it in one either. Your intention
here is to create a neutral sound balance, to level out the frequency response of the
room. Almost every small space will have problems with resonating bass frequencies
which results in your room sounding a little boomy, so you need to bring these under
control first. With solid bare parallel walls you're also likely to have some flutter echo
happening in the room if you hit a drum, clap your hands or play guitar in the room so
this is something else you might have to deal with.
In this section I'll give you some basic guidelines on how to set up your room just
enough to get reasonable results at little cost so you can get straight to the music. To
begin with I want to mention the placement of your monitors in your room. Avoid
placing the monitors in the corners of the room. You should also avoid placing them
exactly halfway along any of the rooms dimensions and exactly halfway from ceiling to
floor. If you're in a rectangular room you should position the monitors facing down the

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long axis of the room. Monitors should be stood upright and not lying on their side,
which compromises the stereo image, and they should make up two corners of an
equilateral triangle, the third corner of which should be you in the listening position.
When dealing with sound reflections in a room there are two main forms of treatment:
absorption and diffusion. Absorption uses various materials to absorb sound, whereas
diffusion breaks up the sound waves so they're less troublesome.
The low frequencies are the toughest to deal with and require a high level of
absorption. To give you an idea of how much you need, for an absorber to be effective
at treating a 100Hz audio signal it should be approximately 30cm thick, or
approximately 60cm thick for a 50Hz signal. These values would vary a little depending
on the absorption properties of the material being used. Low frequencies should be
targeted before mid or high-range frequencies. For this I recommended investing in
some bass traps. These are quite big and much thicker than regular foam panels and
are normally cut for placing in corners of the room. If you only have the budget to treat
two corners of the room then you should treat the two corners behind the monitors first,
the two corners at the rear of the room should be secondary.
Once you've got the low end under control it is time to move on to dealing with higher
frequencies. Mid-range and high frequencies can be treated quite easily using acoustic
foam panels from companies such as Auralex. You should first prioritise what are
known as the 'mirror points'. These are the first sound reflection points in the room.
When the sound leaves the monitor it bounces back from these points towards the
listening position. Focus here on the walls to the right and the left of the listening
position as well as the ceiling above. To find the exact mirror point quickly you can sit in
the listening position and have a friend hold up a mirror and move it about the walls
and ceiling until you can see your monitors in the mirror, hence the phrase 'mirror
point'. Treat these points first then focus on the rear wall of the room to deal with
reflections coming from behind you.
Some soft furnishings in the home can also be very useful to absorb mid and high
range frequencies. Duvets can be great for this purpose and can be as good if not
better than some of the acoustic foam panels available on the market. The tog value of
the duvet will determine how well they absorb sound, the higher the tog value the more
sound is absorbed. Heavy curtains hung with folds can also provide some absorption,
particularly in the upper-mid and high frequencies.
Hanging thin drapes straight will not be sufficient.
Having a carpet or rug on the floor will also help
deal high frequencies, and the thicker the carpet
or rug is the more sound it will absorb.
Diffusion is another good way of dealing with high

frequencies. Placing some shelves on the rear
wall behind the listening position and part filling
them with a mixture of books, CDs and other
similar items works quite well at this if you don't
have the budget for some pre-made diffusers or
for the additional foam panels.
The diagram on the right is a good reference for

placing acoustic treatment in a typical room. Once
the bass has been dealt with the first points in the

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room to focus on are the orange areas. The purple areas should be the next for
treatment if you can stretch your budget a little further.
Just some final tips to finish off this section; If you can hang the absorption materials an
inch or two away from the wall this will increase their effectiveness. You should also
avoid covering any more than 40% of your room, you are likely to kill the sound if you
do so.

7.
HARDWARE
Computer
The first piece of hardware you'll need, and

which you've surely got lying around
somewhere, is a computer. This will be the hub
of your studio and it doesn't matter if it is a PC
or a Mac, you can make music on it with great
results either way. Some people will argue the
advantages of one over the other but my advice
is that if you're familiar with PCs then use a PC,
and if you're familiar with Macs then use a Mac.
What's worth mentioning here is that the

more processing power you have in the
computer the more processing you can
carry out during mixing sessions. For this
I recommend you use a computer with a
good CPU and plenty of RAM.
Ideally you should also have a separate

hard drive for audio. Storage is quite
inexpensive these days so you can get a
500GB or 1TB drive without breaking the
bank and this will allow you to free up your internal drive for just system use and for
running applications.
Audio Interface
To get your sounds in and out of the computer you'll need an audio interface to connect
to your computer and to work with your software. This is what you'll connect your
microphones, guitars and keyboards to when recording. You'll also be connecting your
headphones and/or loudspeakers to this too so you can hear your music play back.
Most software these days will work with a wide range of audio interfaces so there's lots
to choose from. If you're just starting out then I would advise you to stick with a 2-
channel interface. This will allow you to record 2 inputs at the same time. You'll also
need at least one of those two channels to provide you with what's called phantom
power if you intend on using a condenser microphone.
These audio interfaces will connect to your computer via USB and will give you the
internal pre-amps you need to record a variety of instruments. Technology
advancements over recent decades mean that we can get high quality recordings with
minimal cost and with very little fuss when setting up your recording system. The audio
interfaces I discuss here also carry out the conversion of analog audio into digital audio
and back again. The analog audio signal coming from an electric guitar or microphone
travels through an Analog-to-Digital Converter (ADC) to convert the signal to digital
audio, computerised 1s and 0s. When we've processed the audio in the computer the

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digital audio passes through a Digital-to-Analog Converter (DAC) to convert it back to

an analog signal which is sent to the headphones and loudspeakers.
A good choice of audio interface for starting out

with would be the Focusrite Scarlett 2i2. This will
provide you with 2 channel inputs, phantom power,
loudspeaker outputs, a headphones output as well
as monitor and headphones volume controls. This
audio interface should only cost you around 129
if you're shopping in the Eurozone, 99 if you're in
the UK or $ 149 in the US. To find out more click here if you're in Europe or here if
you're in the US.
If you're looking for an interface with a little bit more

to offer then take a look at the Audient iD14. Audient
use the same high-quality pre-amps in their home
audio interfaces as they use in their large-format
mixing consoles and on top of what you get with the
Focusrite above there are some more nice features
including a harmonically rich DI input as well as
their innovative ScrollControl feature that allows you
to control onscreen features, such as plugin
parameters, with the control knob on the unit. This
interface should cost you around 269 in Eurozone, 199 in the UK or $ 299 in the
US. Again, if you want to find out more click here if you're in Europe or here if you're in
the US.
If you're looking for something with a few more inputs, which you may need if you
intend on recording drum kits or if you want to multi-mic a few different instruments at
once, then you could look at the Focusrite Scarlett 6i6 or 18i8. You can also add to the
Audient iD14 by connecting the Audient ASP800 to it via an optical cable to add a
further 8 high-quality inputs.
Microphone
On a very basic level microphones convert sound pressure into electrical energy, and
there are 3 main types: dynamic microphones, condenser microphone and ribbon
microphones. Without getting into the technical side of things, they just use different
kinds of technology to do what they do. Dynamic microphones and Condenser
microphones are most popular, though some people would use ribbon microphones a
lot in a studio. Dynamic microphones are robust and relatively inexpensive in
comparison to other types. For this reason they are good for use in live sound but they
also have their place in the studio as well. Condenser microphones are more often
used in a studio setting, just like ribbon microphones.
Choosing microphones will depend heavily on what you intend to record in your studio.
I would suggest you start with a dynamic microphone or a condenser microphone. The
range of microphones available is also huge so you could do some research on various
microphones to see what suits you best. For now I'm just point you in the direction of
one condenser mic and one dynamic mic.

9.
The Rode NT1-A is a large-diaphragm condenser

microphone mainly used for vocals but also works well with a
variety of instruments. It has gained a reputation for being a
microphone that punches well above its weight in terms of the
quality of performance you get for the price you pay. This
microphone is also sold as a full kit that includes a great
shockmount and a quality pop shield (we'll get to these in a
bit more detail later). You can expect to pay around 189 in
the Eurozone, 139 in the UK or $ 229 in the US for this
whole kit. To find out more click here for Europe or here for
the US.
The Shure SM57 really is a classic microphone. It is a workhorse

of a mic that I can safely say you can't go wrong with. This
dynamic microphone is great on snare drums and on guitar
cabinets. It is also useful on vocals and I've even heard of people
getting good results using it on a kick drum, so you can see the
range of tasks this mic can be put to good use on. This mic is a
must-have whether you're in a small home studio or if you're in a
studio at the top end of the recording industry. The cost of the
SM57 is around 109 in the Eurozone, 80 in the UK and $ 99 in
the US. For more information click here if you're in Europe or here if you're in the US.
You'll also need a microphone stand or two depending on how many microphones you
intend to use at any time so pick up some standard boom-arm microphone stands from
your local retailer.
Headphones / Loudspeakers
When looking for headphones or studio monitors (loudspeakers) it really is down to

your own personal taste. I'm not going to refer to you to any particular set of
headphones or monitors, I'll just point you out a few sets of each that you can start with
on your quest to find the perfect headphones or monitors for you. All of them are
popular choices and all come with favourable reviews from a variety of sources.
Headphones:
Beyerdynamic DT-880
AKG K-702
Sennheiser HD-650
AKG K-271 (closed back headphones)
Monitors:
KRK RP6 RoKit
Adam F7
Yamaha HS7
Focal Alpha 65
Just a few points to bear in mind; most of the headphones I've mentioned above are of
an open back or semi-open back design which are good choices for the reference
monitoring and critical listening required for mixing and mastering duties. If you are

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recording with headphones you may wish to use a closed back design to avoid spill
from the headphones hitting the microphone you're recording with. For this reason I've
added a closed back option to the list above. When choosing monitors take a look at
the frequency response of the speaker before purchasing, particularly bass
frequencies. This will be somewhat dependent on the woofer size, but one speaker
may be able to produce 40Hz when another may only reach 55Hz as its lowest
frequency. This might be a major deciding factor if bass is important in the music you're
producing.
My advice when it comes to headphones or monitors is made up of two key points;

Consider the environment you're working in - For example, if you're working in
a room that has poor acoustic properties, such as one with highly reflective
surfaces and little sound absorption, then it may be counter-effective to use
monitors as the sound produced by the speaker will be badly affected by the
reflections from all around the room. One might be better off with headphones
rather than monitors in a situation like this. Also, if you're predominantly
working at night time then headphones might be the better option if
loudspeakers are likely to cause disturbance.
Get to know your speakers and headphones Whatever you choose, spend
the time to get to know your speakers or headphones really well. Take out a
bunch of your favourite CDs and spend time listening to them. Listen to as
much music as you can on them. The longer you spend listening to music on
your headphones or monitors the more you'll become familiar with how they
behave. Things to look out for are how they respond to transient sounds such
as drums, how bright or dark they are in terms of the energy they have in the
higher frequencies, how much bass and low end they produce, how wide the
stereo imaging is and how clear the front-to-back depth is in the music you play
through them.
SOFTWARE
Digital Audio Workstation (DAW)
The Digital Audio Workstation

(DAW) is the most important piece
of software in your studio. This is
where all the recording will be done
in the computer and it is where all
the editing, mixing and mastering is
done afterward.
Your options will depend a little on

whether you are using a PC or Mac
computer. Some of the most popular
DAWs on the market are Pro-Tools, Logic, Cubase, REAPER, Live, SONAR, Studio
One and Reason. All are compatible with both platforms except Logic which is Mac-
only and SONAR which is PC-only.
I work in REAPER myself and I would highly recommend it to anyone looking for a new
DAW. If you're cost-conscious then REAPER might be a good option for you too
because of its inexpensive license fee of just $ 60, despite it becoming one of the most

11.
popular DAWs out there over recent years. Pro-Tools is somewhat of an industry
standard though when it comes to professional studios, and some DAWs are also more
suited to working in specific genres of music.
Most of the DAW manufacturers offer a trial version or even a free version with a
limited feature-set so I would recommend you try out most, if not all, of the above
DAWs and see which one best suits your way of working. Once you've selected your
DAW then move on and get to know it as well as you can. Being familiar with most of
the common shortcuts for example will slice chunks out of the time it takes to complete
a production.
One common misconception among those new to all this is that music produced in one
DAW will sound much better than music produced in another DAW. I can assure you
now that this is not true. Whether it is Pro-Tools, REAPER, Logic or Cubase, neither of
these DAWs directly effect sound of the music. What you do in the recording, mixing
and mastering is what defines the end result. With that out of the way you should be
free to go about your evaluations before choosing your DAW.
Virtual Instruments
Virtual Instruments allow you to create a variety of sounds that otherwise you would
need dedicated hardware or a physical space to record yourself... and possibly a really
nice set of microphones too. If you're short on instruments or recording space then you
can look at a huge range of software instruments to fill those gaps. For example, if you
don't have the space and equipment to record drums then you could try Steven Slate
Drums, EZdrummer or Addictive Drums to lay drums down in your tracks. If you need a
grand piano or a range of popular amps for your guitar then there's something for these
too. There's also a massive range of synthesizers and orchestral instruments available
out there. Take a look around and make use of any trial versions before you purchase
to be sure that it is just what you are looking for.
Plugins
This is where it gets almost too easy to become lost in all of what's on offer. The audio
industry is flooded with plugins. To start with you've Equalisers, Compressors, Limiters,
Gates, Distortions, Reverbs, Delays, Enhancers, Transient Shapers, Pitch-Shifters,
Tuners, De-essers, Stereo Imagers, Meters and more. Within all that you've got
developers offering numerous versions of each, and they all claim to offer something a
little bit better than the others.
So where do you start? Well I recommend you start with what's already included in
your chosen DAW, the stock plugins. That might sound entirely unexciting but every
DAW comes with a bundle of very useable plugins and tools. In fact, it's entirely
possible to create a great sounding mix with stock plugins alone. If you can't create a
mix that sounds good with just stock plugin then it is not the plugins that are the
problem, you just need to keep developing your craft to improve as an engineer.
However, if you are short on one kind or another you can see what joy you get with the
third-party options. Before you go spending your hard earned money you've got to
make sure you're looking for the correct plugin format. The main formats out there are
VST, AU (Audio Unit), RTAS and AAX. RTAS and AAX are used in Pro-Tools whereas

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AU is used in Mac workstations. VST is used in various PC and Mac DAWs. Check
which plugin format your DAW can host, and just as I suggested with the DAWs, try
before you buy.
Some of the big names out there for plugins are Universal Audio, Waves, Slate Digital,
SoundToys, FabFilter, iZotope, Softube and Sonnox. Many of them will regularly run
sales throughout the year but otherwise they are likely to cost you anything from
around $ 100 to as much as $ 300 or more per plugin if you're buying them individually.
Universal Audio plugins also require their own dedicated hardware which further
increases your outlay. I use a mixture of stock plugins and various plugins from the
manufacturers listed above, but I also use a number of other plugins alongside those
that are either free or much more affordable. To help you save money I've listed some
of my recommended plugins from the 'under //$ 100' range:
EQ PSP NobleQ ($ 69), Cockos ReaEQ (FREE), IK Multimedia T-RackS Classic

Equalizer (FREE), Tokyo Dawn Labs Slick EQ (FREE), Melda Productions MEqualizer
(FREE), lkjb Luftikus (FREE)
Compressor Massey CT5 ($ 79), Klanghelm MJUC (24), Klanghelm DC8C (20),
Variety of Sound ThrillseekerLA (FREE), vladg/sound Molot (FREE), Antress Modern
compressors (particularly Seventh Sign, Lost Angel, Fire Chainer, & Leveling Amplifier)
(All FREE).
Limiter LoudMax (FREE), vladg/sound Limiter No6 (FREE)
Distortion / Saturation Softube Saturation Knob (FREE), Klanghelm SDRR ( 22),

Massey TapeHead ($ 69), Plug&Mix Analoger ($ 49), Voxengo Tube Amp (FREE)
Reverb Valhalla Room ($ 50), Valhalla VintageVerb ($ 50), Variety of Sound epicVerb
(FREE)
Delay Cockos ReaDelay (FREE), Variety of Sound NastyDLY (FREE)
Modulation / Chorus Acon Digital Multiply (FREE), Blue Cat Audio Chorus (FREE)
Some of the plugins listed you might find a bit too complex in the beginning with all the
parameters available, particularly the vladg/sound plugins, so maybe leave these until
some time later. Most of these plugins come in VST format and the others are available
in RTAS, AAX or AU. If you feel stuck because your DAW won't support the VST or AU
format then you try Blue Cat Audio PatchWork which is a plugin adapter of sorts that
allows you to use VSTs in Pro-Tools for example.
ALL THE CHOICE IS GREAT, ISN'T IT?
Well... Yes, and No. Advances in technology mean we can now have quality tools at
our disposal and an extremely well equipped studio in a small space such as a
bedroom, something that could only be dreamt about a few decades ago. The
abundance of equipment on the market has also driven down the cost of gear so
everyone can get involved, which is great for music. However, the problem we have is
that it gives us too much choice. We can get caught up using tonnes of different plugins
and not getting to know any one of them particularly well. We can also spend long
hours debating the best audio interface on internet forums when in fact all of them are

13.
perfectly useable for creating professional-sounding music. Then you have software
developers marketing their new plugin as that one missing piece you were looking for
to get that sound you were always after, which is clever marketing from the sellers but
completely takes the focus away from the ability of the user. None of this is particularly
helpful when it comes to helping you develop your craft.
My advice to you is to pick your DAW, audio interface, microphone and headphones (or
monitors) then move on. Accept what you now have and then focus on using those to
learn your craft and get better at recording, mixing and mastering music. When it
comes to plugins just pick one distortion, one delay, one reverb, one limiter, one de-
esser, etc. Your most important tools are your EQ and compressor so for these I would
allow two of each, one with a clean transparent sound and one with some character,
such as one modeled on a piece of analog hardware. Stick to the plugins you've
selected and use just these on every project you work on for the next six months. This
way you'll get very familiar with how they work and what sounds you can achieve with
them, and the spotlight is taken off the plugin and onto you.
Focus on you, not on the gear. Find out your weaknesses and learn to improve on
them, and always maintain that mindset you'll still be learning in ten or twenty years
time. Never think that the gear is stopping you from making pro-sounding music. I can't
stress this enough and I promise you that if you follow this approach then you'll
become a much more accomplished engineer.

14.
MAKING CONNECTIONS
To get all of your audio equipment communicating with each other you need to start
connecting them together, and although a lot of the gear connecting directly to the
laptop or desktop computer can be done with the ever-reliable USB connection you
may need to use an assortment of other plugs and sockets to connect various
instruments and sound devices to your audio interface. In this section we'll take a look
at the most common connections.
XLR
The XLR is the industry standard when it comes

to balanced audio connectors. You're likely to
find XLR connections on the majority of `
microphones, particularly professional-grade
microphones. They are also used to connect
loudspeakers.
There are a variety of forms of the XLR connector but

the one we're most concerned with when working with
audio is the 3-pin version. One pin is used for 'ground'
(or 'earth') (Pin 1) and the other two pins, Pin 2 and
Pin 3, are used for 'hot' and 'cold' signals respectively.
On the audio interface you're using in your home or project studio you are likely to find
at least 1 or 2 XLR inputs which you will generally use for microphones. If you are
using a condenser microphone it will require phantom power, which is 48 volts DC, and
the balanced XLR connector will allow you to transmit that power to the microphone
once you hit the phantom power button on your interface or mixer.
JACK
OK the Jack is not the official name for this connector

but it is one of the most common terms we use for it.
The 'jack' is a phone connector and is correctly
referred to as a TRS connector. TRS stands for Tip, Ring,
Sleeve, which are the 3 conductors in the connector.
You might notice in the image on the left the two black
rings around the connector. This will tell you that it is a
TRS connector. There is also the TS version which has
just one black ring, the one closest to the tip. They look just like the TRS connector
except for that one difference. TS stands for Tip, Sleeve, and this connector has just
the two conductors. The Sleeve is the 'ground' and the Tip & Ring carry the audio.
TRS connectors are used for Stereo or

Balanced Mono connections, whereas
TS connectors are for Unbalance Mono
connections.
There are a number of different sizes

15.
when it comes to 'jack' connectors but the most common one we come across in audio
production is the 6.3mm (1/4-inch) connector. You'll be familiar with this if you play
electric guitar because this is the connector we use to plug the guitar into an amplifier,
pedal FX or DI box. This connector is also used for headphones, synthesizers and
electronic keyboards, as well as loudspeaker outputs on a home audio interface.
It is also worth briefly mentioning the 3.5mm (1/8-inch) connector. This is the one used
for headphones or earphones on an iPod or any other portable audio device.
S/PDIF
S/PDIF stands for Sony/Philips Digital Interface and is generally pronounced 'ess-pee-
diff' or 'spih-diff'. Where the previous two connections we looked at were analog
connections, meaning they transmit audio in low-power electrical current, the S/PDIF
connection is a digital connection meaning it transmits audio in digital 1s and 0s.
The S/PDIF format is best used over short distances and generally uses two types of
connector, the RCA and the TOSLINK.
RCA
The RCA connector is used for coaxial cable

connections and the the connector is often called a
phono connector. The White cable carries the Left side
of the audio signal (or the Mono signal) and the Red
cable carries the Right side of the audio signal. On each
connector the pin in the center carries the audio signal
and the collar around it is the 'ground'. In some Audio-
Visual (AV) connections you might see a Yellow cable too, this carries the Video.
TOSLINK
The TOSLINK connector is used for fiber optic

connections. It is commonly known as an 'optical cable'.
These cables can fail or become permanently
damaged if the cable is bent tightly. However, they do
not suffer electrical problems such as ground loops or
electromagnetic interference.
MIDI
MIDI stands for Musical Instrument Digital Interface. The

MIDI cable does not carry any audio signal, it carries
event messages like pitch, velocity, notation, and more.
These messages are sent from a MIDI keyboard controller
or a MIDI instrument to a sound device or sequencer that
triggers the audio we hear.

16.
Over recent years the MIDI connection has been increasingly replaced by MIDI over
USB, so many MIDI keyboards you are likely to see in the home studio now are
connected directly to the computer by USB.
BALANCED VS UNBALANCED CONNECTIONS
You might have noticed I made use of the words 'balanced' or 'unbalanced' when
discussing the various connectors. A balanced connection has one ground and two
audio conductors, such as the XLR or TRS connections, whereas an unbalanced
connection has one ground and just one audio conductor, such as the TS or RCA
connections. Balanced connections protect the audio from external noise and electrical
interference and allow the use of longer cables.
Most modern audio systems allow balanced and unbalanced connections to be mixed
successfully, but you can have problems with certain outputs and inputs if you are
connecting an unbalanced plug to a balanced socket, or vice versa. For this reason I
recommend that you check the manual that came with the unit to be clear on what
connection you should make.
LINE LEVEL
The term 'line level' is often spoken of in discussions on audio connections. Line Level
refers to the strength of an audio signal being transmitted between audio devices and
is the highest level signal before amplification. For Consumer audio devices such as a
CD player line level is around -10dBv, but on Professional devices such as mixing
desks or signal processors line level is around +4 dBu. You might find a selector switch
on some equipment allowing you to select -10 or +4, which allows you to calibrate it to
the other gear you're using with it.
Microphones output what is known as 'Mic Level', which is a weak audio signal, and
instruments such as electric guitars or synthesizers output what's known as 'Instrument
Level' which is also a weak signal. Both Mic Level and Instrument Level require a pre-
amp to come up to Line Level. There's also Speaker Level which is the strength of the
signal after being amplified. This is higher than Line Level and is used to drive
headphones and loudspeakers. Be careful not to send a line level signal to a pre-amp
expecting a Mic or Instrument level signal, and never plug a speaker level signal into
any device expecting anything less than a speaker level signal.

17.
RECORDING
OVERVIEW
This is where you start laying down your greatest musical achievement yet. At this
stage your goal is to capture a good song really well.
In the home or project studio you're unlikely to be recording large ensembles with multi-
mic setups, particularly if you're working with just the one microphone to start with. But
that's a good thing right now. You've first got to master the craft of using a single
microphone effectively, then you can move on and add a second or a third.
Always remember that a bad recording engineer will produce a poor-sounding record
even when using top-of-the-range gear, but a good recording engineer can produce an
excellent-sounding record with just modest gear, and I include home studio equipment
when I say 'modest gear'. So much of the equipment available today is capable of great
results so bad recordings are generally a result of poor engineering, not poor recording
equipment. Again, focus on you and not the gear.
When recording, there's not just one way of recording any type of instrument or vocal.
There's no wrong way either. The only rule is to make sure it sounds good.
Experimenting with microphone positioning is essential to finding out the best way to
record each sound in the room you're in, so don't just set it and forgot it. Spend the
time getting things perfect before you commit the recording. You should be prepared for
tweaking your mic placement until you get it just right. The more you do this the better
you'll get at recording.
In this section I'm going to explain a little about the recording process and then provide
you with guidelines on how to record some of the most commonly used instruments.
Before that I'm just going to run through a few things you should know.
MICROPHONE POLAR PATTERNS
Every microphone will have whats known as a polar pattern, which is the microphones
directional sensitivity to sound pressure. This might sound complex if you've not heard
of it before but believe me it is not. There's a number of different patterns but the main
three used in the studio are Cardioid, Omnidirectional and Figure-of-8. The plot images
here will help me explain...
CARDIOID OMNIDIRECTIONAL FIGURE-OF-8

18.
What these plots are really trying to do is give a 3D representation of a microphones

sensitivity in a 2D graph, which might be why they come across as being a little
confusing. When looking at these plots try not to get too caught up in the signal
attenuation figures (-25dB to -5dB) and don't try calculate exact angles or anything
either just yet. Think of the microphone as being in the middle, just as in the first image
on the left, and think of 0 as being the sound source the microphone is aimed at.
What a cardioid microphone will do is pick up everything in front of it as well as quite a

bit of sound from the sides. The cardioid microphone will also reject sound from behind
it, which is useful when you want to avoid another sound and your mic placement will
be key to this. Cardioid is the most common polar pattern in microphones, however
they suffer from 'proximity effect' (which I'll get into shortly) and 'plosives' which are
short, strong blasts of air on words beginning with Ps and Bs.
An omnidirectional microphone is equally sensitive to sound pressure all around it. This
gives a more natural sound and they are not affected by proximity effect and suffer less
than cardioid with plosives too. 'Omnis' tend to pick up a lot more of the room, so they
are at their best when used in a great sounding room. These are commonly used in
stereo mic configurations.
The figure-of-8 (aka bi-directional) microphone is equally sensitive to sound to the front
and from behind it, but rejects sound from the sides. These also provide a 'natural'
sound but like cardioids they suffer from proximity effect. All ribbon microphones are
figure-of-8 by their nature, and like omnis they are often used in stereo mic
configurations.
There are other patterns such as Supercardioid, Hypercardioid, Subcardioid, etc, but
they're more or less variations on those detailed above and you are unlikely to be using
these in the home studio.
PHANTOM POWER
Phantom power is DC electrical power that is required by certain microphones in order

to operate and is distributed through the audio cables from a mixing console,
preamplifier or audio interface. The voltage is typically 48v DC and the microphone will
use as much power as it needs. To power a microphone in need of phantom power
simply engage the button for +48v on the audio interface or preamp you are using.
Phantom power generally doesnt affect audio quality and is safe to use but you
shouldn't supply phantom power to microphones that do not need it, especially ribbon
microphones.
PROXIMITY EFFECT
This is the build-up of bass or low frequency energy in a sound that happens with a
directional microphone (one with a cardioid polar pattern) is placed too close to the
sound source. The easiest way to see this is when recording a vocal. Ask the vocalist
to sing or speak at a reasonable distance from the mic, and while doing so steer them
closer to the microphone. You'll immediately hear the bass tip-up. This can work nicely
to fatten the vocal but too much and the vocal will sound boomy which is not ideal at

19.
all. It doesn't just happen with vocals though, try it on an acoustic guitar too.
This may not be a huge problem when you're recording a track with just a few vocals or
instruments but over 15, 20, 30 tracks or more it could get problematic.
RECORDING LEVELS
Once you've got your USB audio interface hooked up it should be pretty simple to get it
working with your chosen DAW. Check the manual for your DAW on how to set which
audio device you're using and to set the audio ins and outs (also known as I/O). When
you add a new track to your DAW session and arm it for recording then you should see
the incoming signal from your microphone flash up on the track meter on screen,
peaking and dropping with the sounds hitting the mic.
Before you record anything set the gain on your audio interface so that you're getting a
good level. By good level I mean not too quiet but also not peaking right up at the top of
the meter. When working with digital systems such as your DAW your track meters are
likely set to measure in dBFS (dB Full Scale), where 0dB is at the very top of the meter
and trying to go beyond that will immediately result in clipping. I'd recommend
recording with your signal peaking around -10dB for largely transient sounds such as
drums and for all other sounds I'd recommend recording at an average level of around
-20dB.
When recording to an analog medium such as tape the medium itself can create quite
a bit of noise, such as tape hiss. Recording engineers in this scenario have to get a
stronger signal onto the tape in order to increase the signal to noise ratio. If the
recording was too quiet it would be washed with noise when played back. Recording
directly onto digital storage in your computer does not have the same problem, so
there's no reason to be overloading the inputs and recording at really high levels.
You've got to leave some headroom on your recordings. This will make managing gain
levels during the mixing session a hell of a lot easier too.
Don't worry about how loud your recording is at this stage in comparison to
commercially released tracks. Commercial tracks have been mastered so they've been
prepared to sound that loud. In the recording and mixing stages it is much more
important to maintain a reasonable degree of headroom, and leave raising the overall
level of the song to the mastering stage.
PHASE
Dealing with phase is more of a concern when recording with more than one
microphone on a sound source, so when you're recording with just the one microphone
don't worry about this right now. However, as soon as you add another microphone to
the mix you'll need to consider this.
Phase is basically the peaks and trophs of a soundwave, which is fine on its own, but
problems might occur when you've got two microphones on the one source and when
you run the soundwaves from each microphone side-by-side and they don't line up
right.

20.
For example, in the first image below on the left you'll see two soundwaves. They both
run perfectly alongside each other, so the peaks and the trophs in the waveforms are
completely aligned. These signals are 'in phase' and reinforce each other so this will
sound perfectly fine.
In the image below in the center you'll see that the bottom signal is delayed and runs
slightly behind the top signal. These signals are not quite in phase. Phase generally
affects the low-end more than the high-end, so it affects bass frequencies most. The
resulting sound might come across as 'thin' and lacking in bass energy. Sometimes
signals out of phase to certain degrees will just sound 'weird' in comparison with what it
would sound like with an improved phase relationship between the two.
The final image below on the right shows signals that are completely 'out of phase'.
These signals will cancel each other out and leave a very weak sounding result, if
anything at all. In fact, if these signals were exactly the same without any slight
variations at all the result would just play back as complete silence.
IN PHASE PARTIALLY OUT OF PHASE OUT OF PHASE
On recording hardware and software you might see a button with what
looks like a circle with a diagonal line running straight through it from
bottom-left to top-right. It will look something like the one in the image to
the right here. This is the polarity inversion button. When pressed, this
button will flip the polarity of the soundwave on that channel. What this means in a very
simple way is that peaks will become trophs and trophs will become peaks. This is a
useful tool when checking, or trying to improve, phase relationships between channels.
When using two microphones on a source the distance between each mic and the
instrument itself will be a major factor in deciding the phase relationship between the
signals from the two microphones. With that in mind, if you can set both mics at the
same distance from the source then you should be on the right track. For example, if
you have a drum kit and you have two microphones set up as stereo overheads (we'll
get into this more a little later on) then measuring the distance from the right overhead
to the snare drum, and then ensuring the distance between the left overhead and the
snare drum is exactly the same, should help to make sure the phase relationship
between the two channels is strong every time the snare drum is hit. Sound travels at
approximately 1 foot in just 1 millisecond, so it can be difficult to get it right all the time.
However, when recording it is good practice to get tracks recorded with phase in good
shape so keep this in mind.

21.
THE RECORDING PROCESS
Recording, which is also commonly known as 'tracking', should take place once all pre-
production has been completed. Pre-production is the writing and arrangement of the
music. The music and lyrics should all be written and the arrangement of the song
should be fully mapped out. Any electronic drum or synth parts should be programmed
at this stage too. Performing musicians should also be well rehearsed and all
instruments should be properly tuned before recording begins.
Before you commit to recording any sounds make sure they sound good. If you've ever
heard the phrase 'fix it in the mix', forget about it. If you consider it acceptable to take
average recordings and try make them great during the mixing stage then you're doing
it wrong.
Listen actively to instruments and vocals to figure out what it is that makes it great and
find the perfect place to position the microphone to capture that. This may take a few
rounds of trial and error, but that's OK. Listen to how things sound through your
monitors or headphones. If necessary, record a sample of say 30-60 seconds and then
play it back and listen. If it is possible to improve upon then go back and re-adjust the
microphone or the instrument, or maybe even the position of the vocalist then go
through it again listen to it actively through whatever monitoring system you use and
if necessary record a sample and play it back. This process should be followed for all
instruments and vocals being recorded. If it sounds as good as you want it to sound
then you're ready to record some real takes.
If you've got just one microphone then you don't have to worry about choosing the right
one, but if you have two or more then it is worth knowing that all microphones have
their own sound. They may sound full, thin, bright, dark, etc., all have their own 'in-built
EQ'. You might often hear people ask what the best microphone to record a certain
instrument or vocal is. There's no single answer to any of these questions because
there are so many subjective opinions. If you've got a really great room, a high-quality
interface with great converters and you're ready to invest some serious money into a
top class microphone then I recommend that you do your research on various
microphones and demo whatever ones you can get access to. Otherwise the answer is
this: The right microphone to use is the one that, out of all the microphones you
have available to you, fits best with the sound you are trying to record. This is
how to approach microphone selection and you may soon find that it is particularly
important when recording vocals. If you've got two microphones try both out side by
side and see which one suits best and move on.
In the home studio, without the locker packed with microphones and the space to fit an
orchestra, you'll be recording instruments and vocals one at a time. This way of
recording is known as overdubbing.
When recording each instrument it is important that the performer is comfortable. The
technical side of recording is crucial but most of the what comes across to the listener
of a finished track is a great performance, so it is vital that the musician is in the right
place to perform at their best.
For each instrument you may want to record a number of takes, then before beginning
the mix you can edit to keep the best performances of each instrument by choosing the
best take or editing together the best parts from various takes. Your DAW software
should have a system in place for recording takes so refer to the manual. When

22.
recording you can run through full takes (recording the song from beginning to end in
one go), or recording section by section (ie. Intro, chorus, verse, etc.) and taking a
number of takes from one section before moving onto another. The way you choose to
record takes depends entirely on how comfortable the musician is performing the music
and whether he/she is more confident recording full takes or recording section by
section.
SCRATCH TRACK
This is a very basic track that you'll record first then overdub everything else to. By
'very basic' I mean a recording of just vocal and an acoustic guitar for example, most
likely to a click track in your DAW. During the rest of recording process you'll record
drums, bass, electric guitar, keyboards, etc, alongside this scratch track. It is a guide
for the rest of the recording.
As the scratch track is often seen as 'just a guide' it can be easy to think that it is not
that important, but you'll hear every now and again about some successful commercial
track where the engineer will say that the guitar or vocal was actually the first take done
in the scratch track. Strangely enough this take actually turned out to be the best take
so they decided to keep that one in the final mix. What I'm trying to say is don't treat the
scratch track as unimportant. Record it well. If you're just engineering the song and not
performing yourself you can use this perception to your advantage. With the performers
thinking it is 'just a scratch track' it can relax the musician or vocalist which can lead to
great moments early on in the process. It is also the period where musicians aren't
burnt out after 10 hours recording so handle the scratch track as well as you'll handle
everything after it.
After the scratch track is down, the order in which you record everything else should be
determined by the importance each instrument has in the song. If the drums drive the
track from start to finish then maybe the drums should be recorded next. If the drums
are not a huge part of the song then move onto the bass or piano or whatever is next in
terms of its importance in the song.
Throughout the rest of this chapter I'm going to give you some guidelines on recording
the most common instruments. I'm also going to assume you've got just one
microphone, but I'll briefly mention some other ideas just in case you have another
microphone or a DI box you can use too. When experimenting during the recording
stages don't be afraid to trust your ears and trust your own taste.
RECORDING DRUMS
One Microphone
Recording drums with one microphone is

not the most common way of recording
drums, but it is possible to get a very
useable sound if you approach it the
right way.
With one microphone on the drum kit

you should take a little time to find the

23.
optimal placement for the microphone. Drums are generally a huge part of most songs
so you shouldn't rush this. It might mean some extra experimentation when you have
just the one mic but it'll be worth it. Trying to record a whole drum kit with just one
microphone, getting a nice balance of all parts of the kit, can be a huge learning
experience for any engineer.
To start off try placing the microphone out in front, facing the center of the kit maybe 2
or 3 feet away away, at a height just above the top of the kick drum. You can vary the
height a little to get more or less of different sections of the kit, for example if the kick
drum is too prominent try raising the mic a little and vice versa. If you want you can
push the microphone closer too or even place it in the middle of the kit, just above the
kick drum and pointing towards the snare. Experiment with the positioning, small
adjustments to the position of the microphone can make huge differences in sound.
Because drums are loud acoustic instruments your room will play a major part in the
sound of your drum recording so if the sound of your room is affecting your drum sound
in a negative way then reduce the distance between the microphone and the drum kit
to get more of the drums and less of the room. If you're lucky enough to have a great
sounding room then the opposite applies, don't be afraid to pull the microphone back
out a little into the room to get a greater sound.
Two Microphones
If you've got an audio interface similar to those I've written about earlier in this book
then you'll have two inputs for recording two microphones at once. By using two
microphones to record drums it helps you pick up a bit more of the sound of the kit and
also gives you a bit more stereo width to work with when mixing.
There are a few different techniques you can use for recording a drum kit with two
microphones but the one I'm going to explain here is called the Recorderman
technique. For this technique you should use two cardioid microphones and both
microphones should be of equal distance from both the snare drum and the kick drum
to get the most from this setup.
The first microphone should be placed straight above the snare drum pointing directly
down at the center of the snare. It should be approximately 32 inches from the snare
which is about the length of two drumsticks held in-line. The second microphone should
be placed over the right shoulder of the drummer, also pointing at the snare drum and
the same distance away.

24.
Both mics need to be equidistant from the snare and the kick drums so you can check
this by getting a long piece of string or a cable and following these steps: Take one end
of the string and use a piece of masking tape to stick the string to the center of the
snare drum, then take the other end of the string and with a piece of masking tape
again stick this to the kick drum at exactly the point where the beater strikes the drum.
Lift the string towards the microphone above the snare until both legs of the string are
taut, then pinch the string where it touches the microphone. While continuing to pinch
the string there, pull the string across to where the second microphone is positioned.
The point at which you are pinching the string should touch that microphone too while
both legs of the string still taut. If not then adjust position of the microphone so that it
does, making sure it is still pointing at the center of the snare.
By using the string or cable in this way you can make sure that both drums are of equal
distance to each mic. This will ensure that the kick and the snare remain center of a
good stereo image of the kit, and it will also help keep the recording of the snare drum
and kick drum in phase. You'll remember earlier in this chapter of the book when I
explained a little about phase in audio signals. When you introduce a second mic into
the recording each sound will hit the microphones at slightly different times resulting in
phase differences. Ensuring the distances are equal will help keep the phase
relationships here in good order.
Another benefit of using the Recorderman technique in the home or project studio is
that due to the positioning of the microphones it records a lot of the kit and little of the
room, which is good because in most of these studios the rooms are not ideal
environments for recording drums and so you won't have the wonderful room acoustics
of a large purpose-built commercial facility. The positioning of the mics also means that
the cymbals will be understated a little, which is good because in many drum
recordings the overhead microphones pick up too much of the cymbals and this results
in the recordings from the overhead mics sounding too harsh, which in turn makes
them difficult to work with when mixing.
The Recorderman method is a great technique to get you started in stereo drum
recording.
If you find that recording drums with one or two microphones doesn't really cut it for
your production then your other options are to use a bigger multi-mic setup if possible,
use samples, or try one of the virtual instruments mentioned earlier in the book, such
as EZdrummer, Addictive Drums or Steven Slate Drums. With these you can program
every drum hit on each part of the kit or you can use drum parts from their libraries. It
may not have the same feel as recording a drummer who has rehearsed and
performed particularly for that song but these software packages provide high-quality
recordings of some top class drum kits so they can work really well.
RECORDING BASS GUITAR
Bass guitar recording is relatively straight-forward. There's also a few ways to do it in

the home studio. The first way is the easiest, and probably most common, which is to
plug the bass into an instrument input jack on the audio interface and record straight
into the DAW. This will give you the direct 'dry' bass signal. Some will love the direct
sound, however some musicians won't want this as their 'bass sound' but there's more
to this and I'll explain in a moment.

25.
Another option is to use a microphone to record the bass amp. If you've got your own
signature bass tone with your amp then that's great, and you should record that. Amps
are great for the character and tone they give in comparison to the dry direct sound.
When recording a bass amp try placing the microphone about 6-12 inches from the
speaker. If you've got a cabinet with multiple speakers then place it in front of the best
sounding speaker. Pointing the mic directly at the center of the speaker cone will give
you a very in-your-face sound whereas pointing it a little to the right or left of the center
of the cone will produce a warmer sound. Try experimenting too with the distance by
moving the mic a little closer to the speaker or further away from the speaker. Moving it
further away is likely to reduce the low end energy of the amp a little whereas moving it
closer will increase the low end energy picked up by the mic, but be mindful of
proximity effect.
The only problem with recording bass in the home studio is that if you are recording
just the amp then it is likely you'll miss out on some of the really nice low end energy
the bass has to offer, unless you've got a large amp that you can play really loudly. I
don't recommend doing this either if you're likely to disturb a lot of people. Also, if you
have a microphone similar to the ones I've mentioned earlier, then they are not
necessarily designed for bass sounds which means you'll be left short on bass energy
here too, even when the mic is right up close to the cabinet.
A popular way of overcoming this is to record the direct sound and record the amp
sound. The direct sound will carry the really low end, and some of the clarity in the
higher frequencies, while the bass amp will bring all the character and tone that comes
with it. These should be recorded on separate channels in your DAW but be wary of
potential phase issues since as you are recording two tracks of the same sound
source.
To record both sources at the same time you can use a DI Box with a 'Link' or 'Thru'
socket available. Plug your guitar into the Input jack on the DI box and connect the
Output of the DI box to the input of your audio interface. In terms of sound, this is more
or less the same as connecting your guitar directly to your audio interface. However,
with the Link or Thru socket you can take a feed of this direct signal and connect it to
the input on your guitar amp. With a microphone on the guitar amp you can record the
amp sound when the guitar is played as well as the direct signal through the DI box.
If you're not hung up on using your own amp then this opens up a world of opportunity
to experiment with bass sounds by using a virtual instrument. Line 6 POD Farm, IK
Multimedia AmpliTube, Native Instruments Guitar Rig, Waves GTR, Avid Eleven are all
great options that will give you realistic bass amp tones along with the additional effects
and processing options they provide.
RECORDING ACOUSTIC GUITAR
When recording acoustic guitar you need to find the right balance between various
elements of the sound of the acoustic guitar being played. There's the sound coming
from the sound hole, the sound of the pick on the strings, the sound of fingers sliding
along the strings on the neck of the guitar and there's the overall sound of the guitar in
the room it is being played in.
If you've got an electro-acoustic, which is an acoustic guitar with an internal pickup and

26.
preamp that you can plug a standard guitar cable into, then I would advise against
plugging this guitar into your interface to record. Recording direct will produce a thin
sound that will sound a lot less natural in a recording. This system is really just used in
live performances on stage.
To start off recording the acoustic guitar I would suggest you place the microphone
between 8-12 inches from the guitar. The height of the microphone should be level with
the neck of the guitar and the microphone should be placed in front of the neck
pointing slightly inwards toward the 12th fret on the guitar. This should provide you with
a nice clean sound with some of the brightness of the strings. Use this as your starting
point and experiment from there.
If you want to get more of the body of the guitar you could place the microphone at the
same height but in front of where the fretboard meets the sound hole and point
outwards toward the 12th fret on the guitar. This will allow the mic to pick up more form
the sound hole.
Whatever mic placement you chose you should be aware that the guitar, the
microphone, how the guitar is played and what is being played on the guitar will all
affect how you place the microphone. The song and the sound you have in mind is also
highly important.
Other alternatives for mic placement would be to place the microphone at the same
distance from the guitar a few inches lower than the guitar pointing just below the
sound hole, or a few inches above the guitar pointing downwards at just above the
sound hole. You can adjust the distance between the microphone and the guitar by
moving the mic a little closer or a little further away, but just be careful of pushing the
mic too close to the guitar at the sound hole which will produce a very boomy sound
which is as a result of the proximity effect I've mentioned before.
There are stereo techniques for recording acoustic guitar too but I'd recommend
becoming proficient with just the one microphone for now just to start developing your
skills recording an acoustic guitar. If you're thinking about getting some stereo width
from your acoustic guitar in your production then you can double-track the acoustic
guitar, which is to record a second performance just like the first one but on a different
channel in your DAW and you can use panning in the DAW during the mixing stage
(more on this later in the Mixing chapter) to spread these out over the stereo spectrum.
More often than not a single mono acoustic guitar track thats recorded well will be just
what you need, even when acoustic guitar is a major part of the production.
RECORDING ELECTRIC GUITAR
Some of the methods for recording electric guitar are very similar to what I explained
earlier about recording bass guitar.
The most common way of recording electric guitar is by setting up a microphone at the
guitar cabinet and recording the sound of the speaker. If you've got a cabinet with
multiple speaker cones then find the best sounding speaker and place the microphone
right up close to it. Experiment with the position of the microphone in terms of which
part of the cone it is pointing at. Pointing directly at the center of the cone may sound
harsh, whereas pointing it further from the center and closer to the edge of the cone will
sound a lot warmer. If you feel it is necessary you can try recording a few bars of guitar

27.
in each of a few different positions on the cone then play them all back before deciding
what position you like best.
That generally covers a lot of what recording electric guitar involves. If you've got more
equipment available such as various amplifiers, cabinets and guitars then you can try
out different combinations to find the guitar tone you're looking for.
Recording electric guitar through virtual guitar amplifiers and effects pedals has
become extremely popular in recent years. Line 6 POD Farm, IK Multimedia
AmpliTube, Native Instruments Guitar Rig, Waves GTR, and Avid Eleven all provide a
great selection of realistic amps and effects to help shape the tone you want, so they're
definitely worth trying out. They may be your only option if you're in an environment
where the loudness of an amplifier could be a problem.
One technique worth serious consideration when recording electric guitar is to record
the dry sound as well as the sound from the amplifier, just as I recommended when
recording bass guitar. The reason for doing so differs a little however. When recording
both of these signals from a bass guitar the idea is that both tracks would be blended
together at the mixing stage, whereas with electric guitar it is unlikely you'll use the dry
sound for blending with the amp sound. With electric guitar the dry sound can be very
useful as a backup. If the tone of the recorded guitar amp doesn't fit in well with the rest
of the track once all instruments are recorded then the dry electric guitar signal can be
run through one of the virtual amplifiers to create another amp sound without having to
go back and re-record all the guitar parts. Being a virtual guitar amp this sound can be
continually adjusted until the final mixdown is done. The dry sound can also be re-
amped through another physical guitar amp later on if necessary. All this flexibility can
be invaluable in some sessions and mixing engineers may request the dry tracks if
there are problems fitting guitar recordings into the mix of a song.
RECORDING VOCALS
Vocal recording is most often left until last. One of the reasons for this is that the music,
instrumentation and arrangement will affect the vocal performance. Once all the music
tracks are in place the vocal is ready to be recorded.
As far as the average listener is concerned the vocal is the most important part of any
song, in most musical genres anyway. For this reason it is even more important that the
singer is well-rehearsed, is comfortable and that the right microphone is used if you
have a few different mics to choose from.
The position of the microphone in the room is hugely important and the microphone
should be set up on a mic stand away from any solid walls. This will reduce any
potential problems created by reflections from nearby surfaces. If you have acoustic
treatment then this should be placed at the sides of where the microphone is positioned
,and also behind the singer. If you don't have any acoustic treatment then items from
the home such as duvets and sleeping bags can be hung in those areas to control
reflections. You can also put down a rug on the floor below the mic stand and singer if
you're working in a room with a hard floor. I'm not suggesting that you try and make the
room completely dead but I would highly recommend you follow these steps in order to
maximise the ratio of the direct vocal sound to room ambience. When recording vocals
you should be looking to keep room ambience in the recording to a minimum,
particularly in home and project studios which often have less than ideal acoustic

28.
properties.
Always use a Pop Shield (aka Pop Filter or Pop Screen)

between the singer and the microphone. If you don't use a
pop shield then the sudden blasts of air created by plosive
'b' and 'p' sounds will overload the microphone resulting in
'pops' in the recorded sound. If your microphone doesn't
come with a pop shield then I recommend you buy one,
they're inexpensive and well worth the investment.
The height the microphone is positioned is worth some

consideration too. If you place the microphone a few
inches above the singer's mouth then it will make the
singer lift their head a little and therefore open up their
airwaves to give a more fuller vocal performance.
The singer should be positioned around 6-8 inches from the microphone, however for
more intimate performances the singer may move a little closer. I'd recommend that
you do not position the singer any more than 12 inches from the microphone because
the effect of room ambience on the vocal recording will increase the further the singer
is from the microphone.
Another advantage of using a pop shield for vocal recording is that you can position the
pop shield to limit how close the singer can get to the microphone, for example if the
pop shield is set 3 inches from the mic then the singer cannot get any closer to the mic
than 3 inches. This offers you some protection from higher levels of proximity effect
which would cause the vocal to sound boomy when using a cardioid microphone.
Proximity effect varies as the singer's distance from the mic changes, and you don't
want this ruining what would otherwise be an excellent vocal take.
Experiment with the height of the microphone placement and the distance of the singer
from the microphone to find out what best suits the vocalist and the song. A good
vocalist who is experienced in recording should have good microphone technique
meaning they are good at controlling their own distance to the microphone throughout
each section of the song.
Once you've followed the advice throughout this section and have managed to
successfully avoid excessive room ambience and plosives then you're well on your way
to getting a great vocal recording.

29.
MIXING
OVERVIEW
So once you've got all the recording done it is time to mix all those individual tracks into
a single stereo audio track that sounds great. This is very much a creative process but
one that requires a lot of technical skill and in my opinion is by far the most difficult part
of the music production process to master.
In this section I'll explain the mixing process, the most important tools used in mixing
and provide you with some common techniques to get you started with these too. Do
remember however that every song is different so some tools will be more important to
some songs than others, and how you use each tool will be different depending on the
song. What I'll do is provide you with some basic principles to begin with.
Before I go any further I'm

going to run through a few
things in relation to your
DAW. In your DAW you're
likely to have two main
windows, a Mix window and
an Edit window. The Edit
window is the one that
displays all the tracks listed
from top to bottom down the
left hand side of the screen
with all their audio waveforms
extending across towards the
right hand side of the screen.
The Mix window is the one
that displays all the tracks in
the session laid out from left
to right with each track
containing a volume fader,
pan pot, mute and solo
buttons, and various slots
for inserts and sends. This is
the window that looks similar to the layout of a mixing console.
In case you have little or no experience using DAWs I'm going to briefly describe some
of the tools available on each channel. The main one as I'm sure you know well already
is the volume fader which controls the volume of the audio playing back through that
particular channel. Then there's the pan pot (or sometimes a slider), and turning or
moving this left or right will allow you to place the audio from that channel in any
position across the stereo field. The Mute button simply mutes that channel and the
Solo button mutes every other channel so you can only hear the audio playing through
the solo'd channel.
PREPARATION
Before you start to mix it is essential that you set yourself up for creating the best mix

30.
you can produce by preparing your mix session in the right way. Preparing your
session will save you from wasting time trying to locate various sounds and various
sections of the song so that you can focus more on important decisions.
I understand that this kind of thing is not hugely exciting, but it is important to get this
work out of the way so you can focus entirely on the creative aspects of trying to shape
the mix.
Track Order
Lay out your tracks across the mix window. Keep all your drums tracks together, all
your guitar tracks together, all your background vocal tracks together, etc. Then choose
the order for them that you think will best suit your way of working. For example,
starting from the left I would have all my drum tracks, then bass, then all guitar tracks, if
I've got keyboard or strings then they'll be next, then vocals, and so on. In each group I
also have a typical order to the tracks, for example with drums I'll start with kick tracks
then snare tracks, tom tracks, overhead tracks and finally room tracks.
During this process you should also correct all labels on the individual tracks where it is
required so that your track labels read like 'Overhead L', 'Overhead R, 'Bass', 'Guitar 1',
'Lead Vocal', 'Backing Vocal 2', etc.
Colour Coding
Your DAW should give you the option of colouring the channels in your session.
Choose a colour for each group of instruments and apply them in the session. For
example your drums might be green, your guitars blue and your vocals red. The choice
of colour is not important but it helps you differentiate the various groups so you can
locate instruments easier when scrolling through your session.
I recommend using the same track order and colour-coding on every session because
it will help you move quickly around the session to find what you need. When you're
sessions grow bigger and bigger this becomes even more important so that you're not
wasting valuable time trying to find certain sounds.
Track Markers
Most DAWs will give you the option of setting track markers which are markers that are
placed at any given point in the timeline at the top of the edit window. These markers
should be set to indicate the start of various parts of the song on the timeline, such as
'Verse1', 'Chorus1', 'Bridge1', 'Verse2', 'Chorus2', etc. Shortcuts will then allow you to
jump to individual markers at any time so you can immediately switch between different
sections of the song throughout the mixing stage.
Editing
Your DAW will have a range of tools available to you for editing audio waveforms.
These tools will allow you to do things like split audio regions, move regions, cut/paste
regions, fade in or out regions, cross-fade regions, etc. You are likely to have other

31.
tools like tab-to-transient which will help you jump right to the front of transient peaks in
the audio waveform which is useful for editing drums for example, and you will probably
have pitch-shifting or pitch-correction tools too. If you're not yet familiar with your DAW
I would suggest reading the manual which will explain all the editing tools available to
you.
Use your editing tools to cut out silences in the audio regions. This will help tidy your
session and make it easier to see the arrangement of the song in the edit window.
Listen closely to all the audio tracks and cut out any audio glitches, background noises,
pops or clicks that shouldn't be there. You should also listen out for any sound that cuts
in or out too suddenly and add a fade in or a fade out. What you should be trying to do
here is to deal with any sounds that distract you from the song itself. If some sounds
are out of time a little you can slide them left or right to line them up a little better with
the other instruments. However, be careful when moving tracks containing 'bleed' (the
sound of other instruments in the background) from other sources.
This is also a good time to check the phase relationships for any instruments recorded
with multiple microphones, or with a direct signal and an amp feed taken at the same
time. In the Recording chapter I discussed phase, and it might do no harm to glance
over that section again. When preparing to mix there are a few ways to improve phase
relations but the most common way is to flip the polarity of the audio waveform on one
of those tracks. You might remember the polarity inversion button from earlier. Some
DAWs come with this button on the track panel beside the volume fader but if your
DAW doesn't have one here then you might find this button on a simple stock Gain,
Trim or EQ plugin and you can insert one of these plugins onto the relevant track to
make use of the polarity inversion switch. An example of how you would check the
phase of an instrument recorded with two microphones would be to listen to both
channels together, then flip the polarity on one channel and compare that to how it was
beforehand. If they sound better together when polarity on the second channel is
inverted then you can leave it like this, but if the instrument doesn't sound better then
you can return to the original sound by hitting the polarity inversion button again.
Generally, if the polarity inversion improves the sound of an instrument you will find that
the instrument will have more 'body' with a better low-end frequency response and it
will also sound more defined. Out-of-phase recordings will often sound thin and
sometimes 'weird'. Getting the phase right on instruments recorded through multiple
sources is critical and I can't stress enough how important it is to spend the time getting
this right. Improving the phase relationship can dramatically improve the sound of an
instrument in the production. No matter how much you try to EQ or process an out-of-
phase instrument it will never sound good, so get the phase relationship right to begin
with.
When editing it is important that you do not go crazy editing everything just for the
sake of editing. Every edit you do should be done with intent and purpose. Don't go
moving every drum hit onto the grid, adjusting the timing of every bass note, tuning a
vocal until every word or note sung is perfectly in tune. You will kill the performance.
Editing should be seen as getting the song ready to be mixed, not as a process you
take advantage of to 'fix' everything about a recording thats not perfect. If the
performances have been rehearsed well enough they should be recorded well anyway.
Editing should not be seen as a process to make up for the lack of a good performance
or a good recording. Spend hours editing your song and you'll be sick of it before you
even start mixing, which I can assure you is not good at all.

32.
GETTING TO THE MIX
Before you start mixing you need to have a plan. You should have a pretty good idea of
how you want your song to sound once it is finished being mixed, then you work out
how to get it there. If you don't have the end result in mind then you're just throwing
faders around and twisting knobs on various devices aimlessly.
There aren't many rules at all in mixing but if there was just one then that would be
'Balance'. Mixing is all about balancing the sounds you're working with. That includes
setting volume levels across all the faders of your session, but also includes getting the
various instruments and vocals to fit with each other in various other ways. This will
become clearer a little bit later on.
If you were painting a picture you first have to consider the medium on which you're
painting, ie. the size of the paper or canvas. You can also do this with mixing, though
not in the same physical sense, there's a finite area you have to work with when
crafting a mix too. You've got three dimensions to work within. Firstly there's the stereo
field which is all the way from the left to the center to the right, it is the space between
your left and right speakers. Then there's top to bottom which is determined by the
frequency content in the audio, the bottom being low bass frequencies and the top
being high treble frequencies. Finally there's the front to back depth which is developed
through your use of volume as well as spacial effects like reverb and delay. With all
three dimensions considered, it is like having a 3D box to work in and a huge part of
mixing is how you position all the musical elements inside this box.
Throughout the rest of this chapter I'm going to run through some of the processes of
the mixing stage and explain how to use EQ, Compression, Reverb and Delay along
with volume and panning to help shape your mix. There are other tools and effects but
these are the most important and it you can utilise volume, panning, EQ, compression,
reverb and delay in the right way you can produce a really good mix.
SETTING THE BALANCE
It is time to get to work. Turn up some faders a little and start building a rough balance
with just volume and panning. Just a moment ago I used the idea of mixing within a 3D
box, well volume and panning lets you control the front-to-back and left-to-right
dimensions. When positioning sounds within this space the panning controls where
between the speakers you place each instrument, and the volume controls how close
or how far or how close the sound is. At a low volume it will sound far away but at a
high volume it will be right up front.
Start with the most important instrument, turning up the fader and setting a reasonable
level. Make sure you leave plenty of headroom because not only do you want to avoid
clipping on that channel but once you continue to add instruments to the mix the overall
volume on the master fader is going to increase and you don't want that to be clipping
either. Once you've set the volume you can then pan it to where you want it in the
stereo field. This first instrument is likely to be the drums for a lot of people but some
engineers will start with the lead vocal then move onto the drums after that. When the
drums are done move onto the bass and the lead vocals, setting levels for each. The
lead vocal, bass and drums are generally the most important instruments in most
popular genres but if thats not the case in your song then start with whatever

33.
instrument is the most important and then progress through every other instrument in
order of importance, bringing each one into the mix, setting a level and positioning it
with the pan control.
When panning there are no strict rules but generally the most important instruments, ie.
the kick drum, snare drum, bass and lead vocal are kept in the center. Where you put
everything after that is completely up to you, just be aware that if you were packing
stuff into a physical box there are only so many things you can fit in one place before
you run out of room. When mixing you should remember this too because if you have
too many instruments panned dead center then it is going to get too crowded there and
therefore more difficult to hear individual instruments clearly. So leave space there for
the important stuff and push everything else to the sides to some degree. If you've
recorded drums with one of the methods I described in the Recording chapter then you
will not have individual snare and kick drum tracks, you'll either have a single mic
covering the whole kit, that I would recommend you keep in the center, or you have two
'overhead' mics from the Recorderman setup that you can pan out to the left and right
to give you some stereo width. However, if you've recorded with this two-mic setup I
recommend you don't pan fully to the left and fully to the right because with the lack of
additional microphones capturing the kick and snare you can end up a bit light in the
middle of the stereo spectrum and the kick and snare will become weak in the mix. Pan
them to equal distances left and right, say 30% left and 30% right, to keep the kick and
snare strong in the middle and still get some stereo width on the drum kit.
Work your way through the rest of the instruments until all of them have been
positioned in the mix.
Make sure you don't have everything loud and up front because then there's no depth
in the mix, and if everything is loud then nothing stands out. This might be difficult to
get to grips with in the beginning because we're so used to hearing loud commercial
productions every day so you might instinctively push everything up front. Remember
that the commercial mixes have been mastered and prepared to sound that loud before
being released. Your mix hasn't been mastered yet so just focus on building a nice
balance of all instruments in the performance that sounds good and that has depth to it.
USING EQ
The equaliser (EQ) is the tool used for

adjusting the frequency content of a
sound and is inserted into the channel
of the track you want to work on. In a
very basic way an EQ lets you select a
frequency somewhere along the audio
spectrum and either cut or boost the
volume of that frequency.
Before using EQ it is essential to understand the frequency spectrum. This spectrum is

20Hz to 20kHz and covers the range of human hearing. There are a number of
different bands within this range: Low Frequencies (20Hz-250Hz), Low-Midrange
Frequencies (250Hz-2kHz), High-Midrange Frequencies (2kHz-8kHz) and High
Frequencies (8kHz-20kHz). There's no standardised breakdown but it is generally
accepted that the bands are in or around these frequency ranges. Regions of the
spectrum generally have a number of subjective terms people use to describe the

34.
sound. Get 100Hz right and you'll get a nice fat bass but set too much and it gets
'boomy'. Get 400Hz right and it sounds 'warm' but too much makes it sound 'thick'.
Having 1kHz too high will make it sound 'nasal', or having 4kHz too high makes it
sound 'harsh'. Getting familiar with using these types of subjective terms can help
decide what it is you want to change about the sound.
Most DAWs will ship with a bundled parametric EQ whether it is EQ III in Pro-Tools,
ReaEQ in Reaper, or Channel EQ in Cubase or Logic. These EQs will let you select
which frequency you want to work with, will let you boost or reduce the gain on that
frequency and will also allow you to control the bandwidth of your gain changes using
the Q control. These EQs will also have a number of bands with each band allowing
you to affect a different frequency. Each band will contain a filter which will determine
how that frequency is affected. The most commonly used filter is the peaking filter (also
known as a bell filter) which increases or decreases the volume of a chosen frequency.
This frequency will be the center frequency of a bell curve in the audio with the width of
this curve being determined by the bandwidth control. You will also have the option of
using a high-pass filter (also known as a low-cut filter) which removes everything below
the cut-off frequency you've chosen, or a low-pass filter (or high-cut filter) which
removes everything above the chosen cut-off frequency. The last of the most common
filter types are the low shelf filter, which increase or decreases all frequencies below
the chosen frequency, or a high shelf filter which increases or decreases all
frequencies above the chosen frequency.
The approach I recommend you follow when using EQ is to cut away anything you
don't want or anything that doesnt sound good and leave everything thats great about
it. The stock EQs mentioned above are perfectly good for this. It is too easy to start
pushing up frequencies to make subjective tonal adjustments, and there is scope for
this, but it should be secondary to removing all the clutter. Think of it in a similar way to
sculpting a statue from a marble block, you have to cut away all the unnecessary stuff
to finish up with something great. Because you are taking away energy in the sound
the overall volume level of that sound will decrease after it is been EQ'd, so if you've
cut away quite a bit then adjust the output volume control on the plugin to match it to
what the volume of the sound was before you made any changes to it. This is easily
done by pressing the bypass button on the plugin and listening to the 'before' and 'after'
versions of the sound and then adjusting the plugin's output volume if necessary until
you perceive the loudness of each version to be the same.
Balance is a key word here again. For example, adding a little in the 200-500Hz region
can add some warmth to an electric guitar but adding too much will make it sound
muddy whereas on the other hand taking away too much will make it sound thin and
weak. What you also need to balance is how instruments interact with each other so
they can all work in the mix without overpowering one another in various frequency
regions. For example, if you have too much 1kHz energy in one instrument then it will
be more difficult to hear 1kHz frequencies in other instruments, so it is about carving
out space in various instruments so others can be heard too. Finding the right balance
will be very much dependent on the instrumentation and arrangement of your song.
One tool on a standard EQ that I recommend you make use of on almost every track is
the high-pass filter. Most recorded instruments will carry some energy in the low
frequency range that you might not even know is there, and that you don't need. Over
the whole mix if all tracks are carrying small amounts of this energy this adds up a lot of
unnecessary low frequency information that makes the whole song sound flabby. Try
using a high-pass filter on every track apart from kick drum and bass guitar to cut away

35.
everything below approximately 100z and you'll immediately hear the bass end of your
song get much clearer.
How you EQ your tracks depends on how they are recorded, how many tracks are in
your mix and how you want the final mix to sound. Learning how to use EQ takes a lot
of time and practice so stick at it and experiment. My final piece of advice when using
EQ is to avoid drastic cuts and boosts when you're starting out, so stick to within
approximately 5dB for adjustments. The bandwidth of the cut or boost, which is
determined by the Q setting on the EQ, will also sound more natural with a wide
bandwidth than a really narrow bandwidth. This is particularly important when boosting
frequencies.
As you EQ various tracks you should go back and review your main balance of the
song. Some adjustments might be needed to the instruments you've been working on
after you've made changes to the audio.
USING COMPRESSION
Compressors control the dynamic range of

an audio signal. What this means is that they
tighten up the range between loudest
moments in the sound and the quietest
moments in the sound. In a way it makes the
audio waveform more dense. Though a
compressor can serve many purposes in
mixing, one of its primary uses is to even out a
performance. For example, some words in a
vocal might be too loud whereas other words are too quiet. If you compress the vocal a
little it will help even out the singer's performance. So basically the compressor is a
volume control. It is like you moving the volume fader up and down while audio plays
back, only it is a lot quicker to react to the audio and you get to control how it reacts.
Once again the stock compressor in your DAW is a good one to get started with. On
the compressor you'll see a number of parameters. The most important ones are
Threshold, Ratio, Attack, Release & Output Gain. The Threshold control lets you set
where you want the compressor to begin compressing the audio, so when you lower
the threshold the loudest parts of the performance that pass the threshold will be
compressed based on how you set the other parameters. The Ratio parameter lets you
control how much it compresses. A setting of 1:1 means no compression will take
place, whereas a 4:1 ratio means that for every 4dB that passed the threshold only 1dB
will reach the output of the compressor. This will result in the audio coming out a little
bit quieter because the compressor turns down the volume of those peaks. It is for this
reason that you use the Output Gain control to turn it back up again so you can
increase the volume of the audio to match what it was before compression, but have a
more even performance. The amount that the compressor reduces the volume by is
shown on the Gain Reduction meter. The Attack and Release controls let you control
how quick or how slow the compressor reacts to the audio once it passes the
threshold, and then how quickly or slowly the compressor releases and allows the
audio to return to its original volume after it has been compressed. Experiment with the
attack and release parameters and set them to taste, just be careful of creating any un-
natural artifacts if you intend on retaining any natural feel in the performance.

36.
Some compressors have a more simplified approach with one knob for Peak Reduction
(sometimes labeled Gain Reduction or Compression) and the more you turn this the
more compression you get. These compressors are likely to have a second knob to
control the output gain so you can readjust the loudness of the audio after compression
has taken place. The famous Teletronix LA-2A compressor is one such compressor.
Alternatively you might find another design that offers Input and Output controls as well
as the Ratio options and Attack and Release settings. This type of compressor has a
fixed threshold and the more you increase the input the more you drive audio into
compression and the more gain reduction takes place. The Output control again lets
you readjust the level of the audio. The hugely popular Universal Audio 1176LN
compressor is designed in this way.
If you are completely new to compression then for basic balancing purposes I would
suggest using the simpler designs for instruments such as vocals and guitars. When
working on transient-driven sounds such as drums I would suggest using a different
compressor with more control over attack and release times. Just be careful not to kill
the transient (the initial attack of the sound) and I would suggest setting the release
setting in time with the tempo of the song so that the gain reduction on the audio is
allowed to return to 0dB on the meter just before the next beat. I would also
recommend that you look for relatively conservative gain reduction on most instruments
of approximately 3 or 4 dB. Once you've developed a better understanding of
compression you can experiment more and more with how these tools affect the audio.
When compressing audio the average volume of the sound increases so its easy to
think that what you're doing sounds better, because our ears tend to make us think that
louder is better. This is why its hugely important to set any make up gain carefully so
that you are not fooled by any potential loudness boost.
Like anything else you do when recording or editing or mixing, use a compressor with
purpose and don't just insert a compressor on every track simply by default. If the
volume of an instrument feels unstable when you've set your fader level during the
initial balance then it may need a compressor to help even out the performance.
Just as I recommended with EQ, when you add compression to the various tracks in
the mix you should go back and review your balance again and make any necessary
fader adjustments.
USING REVERB
Reverb is made up of millions of repeated reflections of the original sound. Reverb can
help you develop the sense of depth in your mix as it generally helps push instruments
back further into the mix. There are a number of different types of reverbs such as
Room, Spring, Chamber, Hall, Plate, etc. Your choice of reverb depends on the type of
music you're working with as well as your own taste. Using reverb on a sound allows
you to put that sound into a certain space. This can be purely subjective and you can
use big spaces to make individual performances in the mix sound bigger, but
sometimes you also need to think about whether the size of the space is appropriate
for the music. For example, an orchestral piece wouldn't sound right in a small room,
just like a punk rock song would sound unusual in large concert hall.
When using reverb in a mix it is generally used in a Send-Return setup. What this
means is that instead of using reverb as an insert on a track, like you would with EQ or

37.
compression, you use a 'Send' on the track to send a copy of the audio from that track
to another track. The track you send it to is another track in your session on which you
have a reverb effect inserted. This is your return track. The reverb inserted on the
return track should be set to 100% wet. This is done using the Dry/Wet control, which
basically lets you blend the original signal and the effected signal. You don't need any
of the direct dry signal because this will come from the original audio track so once this
control is set to 100% set the fader on the return track to 0dB then you can go back to
the original track and adjust the level of the send. The send level determines how much
of the audio from that track is sent to the reverb track. The more you send to it the
louder the reverb will be. When using a Send on any channel the send itself can be set
to 'Pre-Fader' or 'Post-Fader'. For the purpose of adding reverb to a sound the Send
should be set to Post-Fader, which means that if you pull down the level of that
instrument then the level sent to the Reverb return will also be reduced. You generally
don't want a situation where you've pulled out an instrument from the balance but the
reverb you've added to it continues in the background.
One of the main controls on a reverb is Pre-Delay. This determines how long of a delay
there is between the direct sound and the start of the reverb. It is used to set a short
time lapse between the two which is useful to separate the reverb from the direct
sound. Sometimes when you put a reverb on a sound it can lose some clarity or
definition when there's no pre-delay. Setting a short pre-delay helps bring some of that
clarity back because the direct sound is allowed to play on its own for a very short time
before the reverb begins. Another important control is the Revert Time, sometimes
labeled RT60 or Decay, which controls the reverb decay time.
Hopefully your reverb plugin has plenty of presets to try out. With the likes of EQ and
compression I don't use presets because the plugin doesn't know anything about the
audio you play through it so presets are not that useful, but with reverb the presets can
be great and once you find a preset close enough to what you're looking for you can
tweak it if necessary before adding it to your track. Once you've got your reverb plugin
set up, go back to the original audio track and pull the send level all the way down then
play back the song and slowly bring the send level back up to introduce some of the
reverb until you've got it just where you want it in the song. Adding too much reverb can
wash all over a nice balanced mix so be careful.
USING DELAY
A delay is a repeat of the original sound. Just like reverb, delay can help you develop
the sense of depth in the mix by putting instruments in a defined space and pushing
them further back into the mix.
The most important parameter on a delay plugin is the Delay Time. This determines
how long it takes for the repeat to occur after the initial sound. You can freely select this
time period by choosing how many milliseconds (ms) you want to delay it by, or you
can set it by musical times such as quarter-notes, eight-notes, sixteenth-notes, etc.
which uses the tempo your project is set at to keep delay repeats in time. The
Feedback parameter on the plugin determines how many repeats you hear.
One common way of using a delay is to set up a mono slap delay. This is generally a
single repeat at a short delay time of somewhere between 60-120ms. Adjust the delay
time so that it works well with the track. It can help put vocals and instruments in a
space without resorting to reverberation. You can experiment with its placement in the

38.
stereo field too, leave it in the center behind the vocal or use it on a guitar thats panned
to the left but pan the delay itself to the right. Another common use to set up a stereo
delay to add lots of width to a sound. You can do this by using a stereo delay plugin or
by setting up two mono delay plugins but have the delays panned to opposite sides.
You can set up the timing again by musical time such as an eight-note on one side and
a sixteenth-note on the other, or set it loosely yourself by having 90ms on the left and
105ms on the right. Experiment with the delay times until it sounds good in the track
and also experiment with the feedback, though I would suggest increasing the
feedback slowly and with caution because some delays can get really loud as the
repeats regenerate over and over.
Like reverb, delay is most commonly used in a Send-Return setup with the send on the
audio channel set to post-fader. Once you've got the delay plugin set up the way you
want it you can play back the song and then slowly bring up the level of the send to add
in the delay.
Delays are often used quite discreetly in a track to give a sense of space and width,
however it is possible to get really creative with delay so don't be limited by that if you
feel a certain instrument or a certain section of the song calls for more extreme use of
delay as an effect.
FINAL THOUGHTS
When your mix is finished it should be well balanced. All instruments should be heard
clearly, particularly the most important ones in the production, and any reverb or delay
effects should be mixed in at appropriate levels so that they are not distracting the
listener from the song (unless that was your intention from the outset of course).
I mentioned that when creating your balance of the song that you should leave lots of
headroom when introducing all the instruments to the mix, and I can't stress how
important this is. You should aim to have your final mix peaking at no higher than -6dB
on your master fader, so when setting initial balance levels don't start off by setting the
first instrument peaking at around this level because as you bring in the rest of the
instruments the overall volume of the project is likely to increase. Start off by setting the
first instrument to peak at a much lower level, say around -12dB or -16dB, and work the
rest of the instruments around this. If this is too quiet to listen to then turn up the
monitoring volume on your audio interface. You want to avoid a situation where you've
got all instruments in the mix and you're pushing the master fader to its limits.
When mixing in your DAW everything is done on the computer and it becomes
strangely easy to get caught up in everything you see on the screen. This also applies
to editing and to mastering. You try to 'fix' waveforms you see in the edit window that
don't 'look' right, you try to set volume levels at a particular 'number' on the fader or try
to have audio signal peak at a certain number on the scale, you try to pan something a
very particular 'degree' on the stereo spectrum, you judge what a compressor is doing
to the sound by what the gain reduction meter is 'showing' you, or you wonder if the
curve you've just created in the graphical display of your parametric EQ 'looks' right.
Some of this stuff is really irrelevant in one sense. It can turn into mixing based on what
you see, rather than mixing based on what you hear. The point I'm trying to make is
that you should use the information on the screen to help you through the process but
make all final decision on the basis of what you hear. Every so often close your eyes or
turn off the screen and listen. Make critical decisions based on what you're hearing,

39.
without the distraction of the screen. At the end of the day we're not creating a visual
product, we're crafting something that sounds great, right?
Over time you're inevitably going to acquire more plugins to add to your arsenal of tools
and effects. While there is an element of excitement to buying and trying out new
plugins I think there also needs to be an element of control over these acquisitions in
order for you to get the most out of the plugins you have. Before you go adding a ton of
EQs or compressors take a look at what you already have. I mentioned before that the
ones that come with your DAW are often quite good and well worth sticking with. If you
do add some third-party plugins do some research and add one or maybe two but don't
go overboard. Clean processors will have no 'sound' to them whereas character
processors will have some form of analog-modeled saturation to them. Pick one clean
EQ and one character EQ, and the same with compressors, then use just the EQs and
compressors on every project for the next six months. Get the know them really well,
how they react to various types of audio, and discover what it is you really like (or
dislike) about them. Having 15 compressors ready to go in your DAW usually means
you waste time picking various ones and messing around with them before
progressing, whereas having two compressors you know extremely well will help you
focus on your own skills and help you get the mix finished quicker.
Finally, I would suggest limiting the time you have to mix any given song, particularly if
you are someone who is likely to continually tinker over a project in an attempt to
achieve the perfect mix. No mix is ever finished so you have to draw the line
somewhere. The longer you spend trying to mix the song, the more difficult it becomes
to remain objective when making mixing decisions. Try to mix quickly but without
rushing things and getting sloppy. Those first few hours are generally the most
productive and will get you most of the way there anyway. Set yourself a limited
number of hours to get the mix done and stick to it, then move onto the next stage.

40.
MASTERING
Mastering is the final stage of preparation before your music is ready for release. It is
about preparing it for the release medium, whether that is digital download, CD or vinyl.
If you've got a number of songs for an EP or an album then mastering also involves
preparing these songs to have a consistent sound from track to track. It requires a lot of
critical listening and one of the key priorities of the mastering engineer is to ensure that
the music is going to sound good on as many different playback systems as possible.
At this stage it is a good idea to get a more experienced engineer to master your
music. Even if it is only for the 'fresh' ears that haven't been buried in the writing,
recording and mixing of the music so far, this will bring a new perspective to things.
However, if you want to carry on and master your own music yourself then you can.
The process here normally involves adding some gain if the finished stereo mix is too
quiet, tweaking the EQ balance on the whole track, compressing if required to control
overall dynamics before limiting to bring the volume up to commercially acceptable
loudness levels. The signal flow may also involve some form of stereo wideness
processing or some form of clipping where necessary.
While you're mastering your own music it is important to be aware that you should only
be using EQ and compression to make minor tweaks. For EQ you should only have to
make changes of a dB, or even half a dB, where it is required to help balance the
whole track. With compression, if it is even required, you should only use very low
compression ratios while avoiding very short attack times to produce minimal gain
reduction in an unobtrusive way. It should be used just to glue the song together a little
more or add just a bit more control to the dynamics of the song. Your finished stereo
mix should be as good as it can be before the mastering stage. If you notice that some
parts of the frequency spectrum need some large cuts or boosts, or if the dynamics
need more work, then go back to the mix again and make any necessary changes then
render/export your new mix, import that into your mastering session and start over
once more.
Loudness is an important part of the mastering process. It is also important to make

sure that you don't push things too far when limiting your tracks. I've discussed EQs
and compressors in this book already but not limiter. A limiter is basically a compressor
with an extremely high ratio that clamps down heavily on the peaks of the audio that
pass the threshold. The limiter should be the last processor in your chain and I would
recommend that limiting decisions be made while playing back the loudest and busiest
part of your song, which is likely to be the final chorus in many cases. Several limiters
are set up with just two main controls. That could be Threshold and Output Ceiling,
where Threshold simultaneously raises the output volume while lowering the threshold

41.
and Output Ceiling controls the maximum output volume. Alternatively, you might have
another one that lets you set the Gain and the Output Ceiling, where the gain increases
the loudness of the audio and pushes it toward the Output Ceiling which limits how loud
the audio can go. When setting the output ceiling on a limiter never going all the way
up to 0dB, set it anywhere from -1dB upwards but no higher than -0.3dB. The limiter
should also have a gain reduction meter, just like a compressor. Keep an eye on this
meter and try not to remove any more than 6dB with the limiter when you're increasing
the loudness of your song. If you're taking over 8dB off with the limiter then you either
need to make some adjustments to your mix or you're simply just squashing the life out
of your song. Of course you want your music to be loud enough to compete with the
rest of the music out there but it is so important too that your music still retains plenty of
dynamics and life.
When a lot of people think of mastering they think of increasing the loudness and
tweaking different aspect of the sound, but mastering doesn't just involve the sonic
aspect of the audio processing. Like I said at the beginning of this chapter, it is about
preparing it for the release medium. Firstly there's the bit depth and sample rate.
Before you export your finished song from your DAW you need to decide on what
format you want to export it at. For example, if you want to release your music on CD
the audio must be rendered at 16-bit 44.1k. If you've recorded your audio at 24-bit and
you intend on releasing your music on CD then you will need to reduce the bit-depth to
16-bit when you're exporting your track. However, before you do so you must add
Dither to your finished track. Dither seems to confuse a lot of people but it is really quite
simple. Dither is a very low level noise that you won't hear but thats used to cut out any
unwanted distortion produced during the process of reducing the bit-depth of an audio
track. Some limiters give you the option of using dither and selecting the bit-rate that
you intend to export at. If your limiter doesn't offer this then there are plugins that will
do it for you and you can insert one of these after the limiter. Should you find yourself in
a position where you have no plugins that offer dithering options then you may find a
last resort in the 'Export' window in your DAW. If you're not reducing the bit-depth when
exporting your audio then you don't need to add dither. Dither is also not required when
reducing the sample rate, so if you recorded at 96k or 48k and you are exporting at
44.1k then you don't need to use dither.
I've mentioned the bit-depth and sample rate required for CD audio, but it is likely you'll
want to prepare your tracks for release online. If you're uploading your track directly to
a website you should check what file formats are supported. If you're using a digital
distributor to release your music then you should check what file formats they require.
Most will accept 16-bit 44.1k audio files but now more services offer the chance to
upload at 24-bit with sample rates as high as 96k.
When choosing a file type on export always go for WAV. It is the most popular loss-less
file format. Don't export in lossy formats such as MP3. Go for the highest quality audio
file that your online service or digital distributor accepts.
For releasing your music on CD you may need to export your project as a DDP image.
This is a file set that you can upload to the plant for CD duplication. Depending on the
DAW you are using you may or may not have this facility available to you. Many DAWs
don't have this option but Presonus Studio One Professional, for example, has a
project mastering page that you can export DDP images from. If your DAW doesn't
have this facility then you may need to invest in some additional software or get in
touch with someone who can do this for you.

42.
To finish this chapter I'm going to explain a few bits of information you're likely to need
before distributing your music, the ISRC code and the UPC/EAN. The ISRC is the
international code for identifying a recording. This is the code used to identify royalty
payments for the rights holder. These are generally available for free so look up the
provider in your local country or region. Your distributor may be able to provide this to
you if you don't already have any. The UPC/EAN is the barcode. Obtaining your own
barcode can be expensive so thankfully most distributors can provide these too for a
small charge or in some cases for free. This information should be embedded in your
DDP image or provided to your distributor if they are not arranging it for you.

43.
DISTRIBUTION
So your music is finished and now it is time to get it out to the rest of the world. In this
section I'm assuming that you don't have a record label at all and that you are releasing
your music yourself.
You need to choose whether to release your music on a physical medium or by digital
download, or a mixture of both.
If you plan to release music on a physical medium then CD is the most common option.
Distribution on CD is great because although CD sales have declined in recent years
there's still a lot of people out there who like to get a physical product. You can sell CDs
at all your shows, you can approach local independent retailers to see if they will stock
your music, or if you can reach an agreement with a distributor it will allow you access
to some of the bigger music stores.
Should you opt for digital release you have a number of channels to choose from.
Digital release is highly recommended given the potential exposure your music can get
online. Firstly you have a number of services you can use yourself for free, some of
which are;
SoundCloud (www.soundcloud.com)
ReverbNation (www.reverbnation.com)
Bandcamp (www.bandcamp.com)
All of these services allow you to set up your profile and stream your songs online.
Potential fans can stream your music for free on these websites. ReverbNation and
Bandcamp also allow you to sell your music directly, for which they take a percentage
share of the revenue. Bandcamp will even allow you to let your fans pay what they
want for your music.
If you want to make your music available on some of the larger platforms like iTunes,
Spotify, Google Play, Amazon, etc. then you will need an aggregator. These are digital
distribution companies who will get your music for sale worldwide on these sites. There
are lots of digital distributors out there. Some of the most popular are;
AWAL (www.awal.com)
CDBaby (www.cdbaby.com)
TuneCore (www.tunecore.com)
These distributors all allow you to sell and make money from iTunes, Spotify and
others. They have varying costs attached so do some research on each one before
deciding which company suits you and your music best. Some have a music store app
available for selling your music on Facebook, and CDBaby offer worldwide CD
distribution too.
YouTube is another great choice for getting your music out there. Although it is a video
platform you can still use it to stream your music, even if you haven't made a music
video. You can create a slideshow of band photos or album art for example. YouTube
has become a major search engine for discovering music in recent years.
Once you've got your distribution set up you need to promote it as much as possible.
This can be done in various ways in your local music scene and online through social
media. To help ensure your music gets to the right people you should tag your songs
correctly on sites like SoundCloud and Bandcamp, etc. including genre, styles, moods,
location, and more. You can also submit it to music discovery or music

44.
recommendation services such as Last.fm and AllMusic. If you're just putting out your
first releases then I'd recommend making a least some tracks available for free to help
spread your good work.

45.
THANK YOU FOR READING
So that's it. If you've made it this far then you're clearly interested in developing your
skills for recording, mixing and mastering your songs. In this book I've tried to cover the
main areas in music production. There is of course a lot more to all of this, so much so
that many books could be written on each step of the process alone, but what I've done
here is give you a ton of information and advice about each topic to help get your music
out there quicker and get it sounding better. For anyone who has written some songs
but never jumped into any of this stuff before I hope this book has inspired you to set
up your own home studio, produce your own music and share it with people all over the
world.
If you've found this book helpful in any way I'd love to hear about it. To send me some
feedback or ask me any questions you can reach me at keith@keithscullymixing.com.
I'll finish up by saying that learning to record, mix and master music takes time and
effort. Every producer, recording engineer, mixing engineer and mastering engineer out
there is learning new things all the time, even after many years. So keep working at it
and don't let any setbacks get in your way. If you're anything like me you'll find it
extremely satisfying to continually learn more & more about doing something you love.
Once again, thank you, and I wish you all the best in your musical endeavours.
Now go make some great music,
Keith Scully
KeithScullyMixing.com

Get Started With Recording, Mixing & Mastering - Guide To Starting Your Home Studio

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Keith Scully 2015 KeithScullyMixing.

Keith Scully 2015 KeithScullyMixing.com

Keith Scully 2015 KeithScullyMixing.com

Keith Scully 2015 KeithScullyMixing.com

Keith Scully 2015 KeithScullyMixing.com

Diffusion is another good way of dealing with high

The diagram on the right is a good reference for

Keith Scully 2015 KeithScullyMixing.com

Keith Scully 2015 KeithScullyMixing.com

The first piece of hardware you'll need, and

What's worth mentioning here is that the

Ideally you should also have a separate

Keith Scully 2015 KeithScullyMixing.com

digital audio passes through a Digital-to-Analog Converter (DAC) to convert it back to

A good choice of audio interface for starting out

If you're looking for an interface with a little bit more

Keith Scully 2015 KeithScullyMixing.com

The Rode NT1-A is a large-diaphragm condenser

The Shure SM57 really is a classic microphone. It is a workhorse

When looking for headphones or studio monitors (loudspeakers) it really is down to

Keith Scully 2015 KeithScullyMixing.com

My advice when it comes to headphones or monitors is made up of two key points;

Digital Audio Workstation (DAW)

The Digital Audio Workstation

Your options will depend a little on

Keith Scully 2015 KeithScullyMixing.com

Keith Scully 2015 KeithScullyMixing.com

EQ PSP NobleQ ($ 69), Cockos ReaEQ (FREE), IK Multimedia T-RackS Classic

Limiter LoudMax (FREE), vladg/sound Limiter No6 (FREE)

Distortion / Saturation Softube Saturation Knob (FREE), Klanghelm SDRR ( 22),

Delay Cockos ReaDelay (FREE), Variety of Sound NastyDLY (FREE)

ALL THE CHOICE IS GREAT, ISN'T IT?

Keith Scully 2015 KeithScullyMixing.com

Keith Scully 2015 KeithScullyMixing.com

The XLR is the industry standard when it comes

There are a variety of forms of the XLR connector but

OK the Jack is not the official name for this connector

TRS connectors are used for Stereo or

There are a number of different sizes

Keith Scully 2015 KeithScullyMixing.com

The RCA connector is used for coaxial cable

The TOSLINK connector is used for fiber optic

MIDI stands for Musical Instrument Digital Interface. The

Keith Scully 2015 KeithScullyMixing.com

BALANCED VS UNBALANCED CONNECTIONS

Keith Scully 2015 KeithScullyMixing.com

MICROPHONE POLAR PATTERNS

CARDIOID OMNIDIRECTIONAL FIGURE-OF-8

Keith Scully 2015 KeithScullyMixing.com

What these plots are really trying to do is give a 3D representation of a microphones

What a cardioid microphone will do is pick up everything in front of it as well as quite a

Phantom power is DC electrical power that is required by certain microphones in order

Keith Scully 2015 KeithScullyMixing.com

Keith Scully 2015 KeithScullyMixing.com

IN PHASE PARTIALLY OUT OF PHASE OUT OF PHASE

Keith Scully 2015 KeithScullyMixing.com

THE RECORDING PROCESS

Keith Scully 2015 KeithScullyMixing.com

Recording drums with one microphone is

With one microphone on the drum kit

Keith Scully 2015 KeithScullyMixing.com

Keith Scully 2015 KeithScullyMixing.com

RECORDING BASS GUITAR