Unit 25: Sound Recording Hannah Jenkinson

Signal Paths and levels, audio-file formats

In order to understand signal paths, we must understand the
differences between analogue and digital sound.
 Analogue sound waves replicate the original sound
wave (what we’re recording).
 Digital sound waves only replicate the sampled sections
of the original sound wave.

The potential accuracy of an analogue recording depends on

the sensitivity of the equipment and medium used to record
and playback the recording.

Digital audio fidelity heavily depends on the rate at which the

recording equipment sampled the original sound wave over a
specified increment of time. Even with the newest technologies
and techniques, digital audio still cannot create exact
replications of an original sound wave.

Analogue and Digital Converters

Analogue to Digital Converter (ADC) and Digital to Analog
Converter (DAC) are very important components in electronic
equipment. Since most real-world signals are analogue, these
two converting interfaces are necessary to allow digital
electronic equipments to process the analogue signals.

ADC converts the analogue signal collected by audio input

equipment, such as a microphone, into a digital signal that can
be processed by computer. The computer may add sound effect
such as echo and adjust the tempo and pitch of the music. DAC
converts the processed digital signal back into the analogue
signal that is used by audio output equipment such as a
speaker.

Sound: Signal Paths

Analogue sound came to be from the
earliest technologies of sound recording.
The process involves using a microphone
to turn air pressure or sound into electrical
analogue signals and converting them directly onto analogue
tapes (large reels or cassettes) through magnetization or on
vinyl records through its spiral ‘grooves’.
Digital sound recording also requires
the step of turning sound into an electric
analogue signal, except it extends the
process to convert that analogue signal
into digital. This digital form of audio is
easily copied onto compact disks, hard
drives or uploaded online.

An Analogue to Digital
Converter (ADC) is a
system that converts an
analogue signal, such as a
sound picked up from a
microphone, into a digital
signal.
A Digital to Analogue
Converter (DAC) performs
the reverse function; it
converts a digital signal into an analogue signal.

A decibel is a unit used to measure the

intensity of a sound or the power level of
an electrical signal by comparing it with a
given level on a logarithmic scale. It is
measured in orders of magnitude, so
each mark on the decibel scale is a
previous mark multiplied by a value. On
the decibel scale, the quietest audible
sound is 0 dB. A sound 10 times more
powerful is 10 dB.

A normal conversation is about 60

dB, a lawn mower 90 dB and a loud
concert 120 dB. In general, sounds
above 85 dB are harmful, depending
on how long or often you are
exposed to them and whether or not you are wearing ear
protection. When recording sound, you need to be sure to
record at the optimum levels to capture the sound.

Optimum Recording Levels

To achieve optimum recording levels, you should set your
recording levels as close to the permitted maximum
level (PML) as possible, without reaching or exceeding that
limit. Reaching or exceeding the PML will
result in distortion of the signal.
Permitted Maximum Level (PML):
Highest volume a programme or a piece of
hardware can accommodate without
resulting in distortion or “clipping”. Digital
audio can come very close to the
threshold for PML without distorting; however, clipping will
occur once the peak has been reached.
Clipping is usually indicated with a red light.
If utilizing coloured level meters –
 Green is acceptable
 Yellow is still acceptable but approaching PML
 Red indicates clipping.
After recording, it is possible, with audio editing software, to
boost the levels if originally recorded too low. A weak signal-to-
noise ratio will raise the noise floor and boost the noise during
the recording.

Understanding Decibels
The levels of decibels have the role of measuring the sound
pressure level (SPL). Sound decibels above 80/90 dB can
cause temporary hearing loss depending on the length of
time and how often the sound exceeds this limit. Sound
decibels at 140 dB and above can cause permanent
hearing damage. Sounds at 3 dB are not perceptible to the
human ear. The more power put in, the higher the decibel level.
Optimizing output
 Enclosure
 Surface area
  Sound dampening
Understanding Clipping
Clipping occurs when the sound exceeds the PML and the
sound wave goes beyond the top line causing a sharp bend in
the sound wave.
How to avoid clipping
 Choose the right amplifier – a large amplifier is generally a
better choice as playing at a high volume is not stressing
it like it would with a smaller amplifier.
 Set your gain properly
 Avoid bass boost
To get a high-quality recording and avoid clipping you must set
the gain properly. This means testing the loudest point of the
recording while taking into consideration the distance of the
microphone. You set the gain/volume at least three decibels
below the clipping point. Peak levels are usually between -12db
and -6db in interviews as it gives some leverage for natural
changes in dynamics. The same applies for recording music –
average levels should come at about -18db, however this is just
a guideline. Levels should remain constant throughout a
recording to avoid feeling the changes in dynamics. If recording
with an amplifier, bass boosting should be avoided as it can
cause clipping.
Digital sound recordings are usually recorded at 16-bit (the
quality of a CD) or 24-bit. Both are fine but 24-bit is easily more
achievable.
dbFS is a way or measuring peak levels of recording. The
maximum dbFS for most digital audio recordings is 0db so if a
recording exceeds that, the dbFS would only reach 0db. When a
recording exceeds 0dbFS, distorted clipping occurs.
Due to all the considerations put in place, it is vital to have a
level metre to constantly display the average and peak levels
when recording sound. If you are recording onto a DAW (Digital
Audio Workstation), your DAW is likely to include a level meter
as one of its features.

Audio File Formats

The Broadcast Wave Format (BWF) is a file format for audio
data. It can be used for the seamless exchange of audio
material between different broadcast environments and
between equipment based on different computer platforms. As
well as the audio data, a BWF file contains the minimum
information – or metadata – which is considered necessary for
all broadcast applications. The Broadcast Wave Format is based
on the Microsoft WAVE audio file format, to which the EBU has
added a “Broadcast Audio Extension” chunk. Audio file formats
are digital files stored on a computer system. With audio files,
there are two types of compression that are used to reduce file
sizes: lossless and loss compression.

Lossless and Lossy

Lossless compression means that
as the audio file size is compressed,
the audio quality remains the
same – it does not get worse. Also,
the file can be restored back to its
original state. FLAC and ALAC are
open source lossless compression
formats. Lossless compression can
reduce file sizes by up to 50% without losing quality.
Lossy compression permanently removes data so that the
original state cannot be recovered. It is used for transferring
audio data. For example, a WAV file compressed to an MP3
would be lossy compression. The bit rate could be set at 64
kbps, which would reduce the size and quality of the file.
However, it would not be possible to recreate a 1,411 kbps
quality file from a 64 kbps MP3. With lossy compression, the
original bit depth is reduced to remove data and reduce the
file size. The bit depth becomes variable. MP3 and AAC are
lossy compressed audio file formats widely supported on
different platforms. MP3 and ACC are both patented codecs.
Ogg Vorbis is an open source alternative for lossy
compression.

UNCOMPRESSED
Uncompressed audio is audio without any compression
applied to it.
Waveform Audio File Format (WAV) – A WAV file is
a raw audio format created by Microsoft and IBM. The
format uses containers to store audio data, track numbers,
sample rate, and bit rate. WAV files are an example of
uncompressed lossless audio and can take up quite a bit of
space, coming in around 10 MB per minute with a
maximum file size of 4 GB.
Audio Interchange File Format (AIFF) – Stands
for “Audio Interchange File Format”. AIFF is a file format
designed to store audio data. The format was developed
by Apple Inc. and is most commonly used on Apple
Macintosh computer systems. The audio data in most
AIFF files is uncompressed pulse-code modulation (PCM).
Pulse-code Modulation (PCM) – A method used to
digitally represent sampled analogue signals. It is the
standard form of digital audio in computers, compact
discs, digital telephony and other digital audio
applications.
Compact Disc Digital Audio (CDDA) – Also
known as Audio CD, is the standard format for audio
compact discs. The standard is defined in the Red
Book, one of a series of Rainbow Books that contain
the technical specifications for all CD formats.

COMPRESSED LOSSLESS FILES

Compressed lossless audio files are audio files that
have been compressed with no loss of quality.
Apple.m4a – Apple Lossless data is often stored
within an MP4 container with the filename extension,
m4a. This extension is also used by Apple for lossy
AAC audio data in an MP4 container.
Adaptive Transform Acoustic Coding (ATRAC)
– This type of advanced lossless file is a “scalable”
lossless audio codec that records a lossy ATRAC3 or
ATRAC3plus stream, and supplements it with a stream
of correction information stored within the file itself
that allows the original signal to be reproduced if
wanted.
Moving Picture Experts Group 4 (MPEG4) – A
file with the M4A file extension is an MPEG4 Audio file.
They’re most often found in Apple’s iTunes Store as the format
of song downloads.
Windows Media Lossless – WMA is a series of
audio codecs and their corresponding audio coding
formats developed by Microsoft. WMA Lossless
compresses audio data without loss of audio fidelity
(the regular WMA format is lossy).

COMPRESSED LOSSY FILES

Compressed lossy audio files are audio files that result
in lost data and quality from the original version.
MP3 – MP3 is a very popular way of storing audio data.
It samples audio data by using a lossy technique that
cuts out parts of the sound that the human ear can’t
hear. Once it has lost this inaudible data, it then applies
a further compression technique, called Huffman
encoding, to make the file even smaller.
Vorbis – Vorbis is the lossy compressed audio codec
which is typically transported in Ogg files. “Ogg
Vorbis” refers to both parts together: an Ogg format
file containing audio compressed using Vorbis.
Windows Media Audio (WMA) – WMA is a series
of audio codecs and their corresponding audio coding
formats developed by Microsoft. It is a proprietary
technology that forms part of the Windows Media
framework. WMA consists of four distinct codecs: the
original WMA codec, WMA Pro, WMA Lossless and WMA
Voice.
ATRAC – Essentially the same as lossless ATRAC files
but are more compressed so some of the quality is lost
but the size of the file will be relatively smaller.

Mono & Stereo Recording and Replay

In monaural sound one single channel is used.
It is a single channel or track of sound created
by one speaker. It can be reproduced through
several speakers, but all speakers are still
reproducing the same copy of the signal.
Monaural sound was replaced by stereo or
stereophonic sound in the 1950s, so you are
unlikely to come across any monaural equipment for your
home.
In stereophonic sound, commonly called
stereo, more channels are used (typically two).
It is the reproduction of sound using two or
more independent audio channels, through a
symmetrical configuration of loudspeakers, in
such a way as to create a pleasant and natural
impression of sound heard from various
directions. You can use two different channels
and make one feed one speaker and the second
channel feed a second speaker (which is the
most common stereo setup). This is used to create
directionality, perspective, space.

How stereophonic sound may be

displayed/laid out in a theatre.

MICROPHONES

Omnidirectional microphones – These

types of microphones capture sound from all
angles. Because of their non-directional design
and zero rejection, these mics capture nuances
better, resulting in a more natural sound. These
mics are best used in studios and other venues
with great acoustics and can also be used for live
recording of multiple instruments, as long as the
noise level is low. The downside is that they lack
background noise rejection and are prone to
monitor feedback which makes them the unsuitable mic choice
for loud and noisy venues.
Bidirectional microphones – Bidirectional
microphones, also called figure-of-eight
microphones), can capture sound of both the
front and back, while rejecting the two sides. This
front and back sensitivity makes them ideal for
stereo recording and for capturing two or more
instruments. They are essentially similar to
omnidirectional mics, but without the sound
rejection on two sides. The bidirectional mic is
commonly used on ribbon mics or on some large
diaphragm condenser microphones.
Unidirectional microphones – Unidirectional
microphones are mics that only pick up sound
with high gain from a specific side or direction of
the microphone. If a voice is spoken into a
unidirectional microphone, they must speak into
correct side, normally called the voice side, of the
microphone in order to get good gain on the
recording. This is in contrast to omnidirectional microphones,
which pick up sound equally from all directions of the
microphone.
Cardioid microphones – Cardioid mics
capture everything in front and block everything
else. This front-focused pattern will let you point
the mic to a sound source and isolate it from
unwanted ambient sound, making it the ideal
microphone choice for live performance and
other situations where noise reduction and
feedback suppression are needed. Mic position is
very important as these types of mics add subtle
sound coloration when the source is off axis.
Hyper cardioid microphones – These mics
have the same front directionality but have a
narrower area of sensitivity compared to
cardioids. This results in improved isolation and
higher resistance to feedback. Because of their
enhanced ability to reject noise, you can use
these for loud sound sources, noisy stage
environments or even for untreated recording
rooms. However, back rejection is a bit
compromised, so you will need to position
unwanted sounds like stage monitors and drum kits on the
dead spot sides.

Shotgun microphones – Shotgun mics, also

called Line and Gradient, feature a tube-like
design that make their polar pattern even more
directional than hyper cardioids. The capsule is
placed at the end of an interference tube, which
eliminates sound from the sides via phase
cancellation, resulting in a tighter polar pattern
up front with longer pickup range. They are most
commonly used for film and theatre but also
make great overhead mics for capturing thing
like singing groups, choral and drum cymbals.
Coil microphones – The moving coil
microphone, commonly called the dynamic
microphone, is one of the most widely used
forms of free-standing microphones. It is widely
used for vocals for musical performances as
well as for many other applications. The
dynamic microphone is also simple in its design and as a result
good microphones offer good value for money.
Ribbon microphones – While these mics are
no longer popular, ribbon mics were once very
successful particularly in the radio industry. The
light metal ribbon used in these mics allows it
to pick up the velocity of the air and not just air
displacement. This allows for improved
sensitive to higher frequencies, capturing
higher notes without the harshness while
retaining a warm, vintage voicing. Interest for
Ribbon mics have since returned, especially
since modern production ribbon mics are now
sturdier and more reliable than their older
counterparts, making them viable for live multi-instrument
recording on venues where noise level is manageable.
Condenser microphones – Condenser mics
have a thin conductive diaphragm that sits
close to a metal backplate. This configuration
works like a capacitor wherein sound pressure vibrates the
diaphragm which in turn changes the capacitance to produce
audio signal. Since they use capacitance instead of actual
moving coils, fidelity and sound quality is improved, making
these mics ideal precision recording in the studio.

MICROPHONE ACCESSOR
IES
Microphone accessories can be combined with
microphones to emphasise the desired effect
of the audio obtained from the microphone.
One of the simplest recording gadgets is the
pop filter. Weather a foam filter, a
suspended stretched fabric, or a metal screen,
it is positioned between the vocalist and the microphone to
block plosives – those percussive P and B sounds that result in
annoying low frequency bumps. While the first pop filters were
little more than nylon pantyhose stretched over a coat-hanger
wire, even the least expensive commercially available pop
screens available today do a significantly
better job at blocking plosives without
attenuating the high frequencies on your
vocals.
Portable vocal booths are shields made
from acoustic absorption material that sit
behind the microphone and prevent sound
moving past the mic from reflecting off of the
surfaces of the room and reaching your
microphone later than the direct sound, which
causes delaying. Portable vocal booths
provide many of the benefits of a tradition room treatment,
without the commitment of sticking foam up on all walls of your
room.
It may seem as though a microphone stand does not
have a serious impact on the sound of your
microphone, but it is a critical element for getting the
best possible sound from your microphone. It is vital to
have a mic stand that’s both extremely stable and
easy to adjust. Unlike liv mic stands which are
engineered to be lightweight and easy to move, studio
mic stands tend to be heavier and have longer reach.
As well as providing extra stability, added weight
makes the mic less likely to be affected by floor-transmitted
vibrations.
Cloud Microphones Cloudlifter Mic
Activators are phantom-powered in-line
preamps that boost signals near the source,
providing you with a stronger, cleaner and
fuller sound before your preamp. Regardless
of what type of microphone you boost with a
Cloudlifter Mic Activator, the extra 25dB at
the source minimizes the problems
associated with mic-level signals, allowing you to run longer
cables with less noise and signal degradation. Cloudlifters are
even more impressive when you use them on low-output
dynamic microphones.

Shock mounts are the object that hold the

microphone in place to avoid it being damaged or
heavily effected by sudden movements. It also
shields the microphone from all the small
vibrations that may be picked up from a room
such as noise and ambient sounds. Essentially, it
provides ambience from being overwhelming in
the audio recording, making it feel more
naturalistic. Shock mounts are well adapted for
interior locations and are commonly used in radio/podcast
recording.
A microphone clip is another object that holds
the microphone in place. They are clipped onto
the boom arms of mic stands. However, they are
also used without a stand when used in situations
such as a commentary, where they act as part of
a headset. They are a useful accessory for
directing the microphone in a specific way so that
it effectively picks up high-quality vocals.
A windscreen is an accessory usually made from
foam and designed for outdoor conditions. Its
purpose is to encapsulate a microphone so that if
it is hit by wind, distortions and crackling from the
wind will be cancelled out. They are used for
things like recording ambient sounds and dialogue
in exterior settings.
EQUIPMENT
Sound Recording Studio Equipment
When recording sound in a recording
studio, there is a list of necessary, basic
and obvious equipment needed to allow
the recording to run smoothly. This
includes – a computer (to store the
recordings), headphones (to be able to
hear your recorded sound),
microphones (to obtain a high quality
sound), DAW, audio interface, studio monitors (speakers,
to play the recorded audio at a high quality), cables (to make
all the connections between each piece of equipment) and
microphones stands and other accessories desired.
There are many more examples of studio equipment that are
less commonly used/found but still exist to enhance the sound
recording in a studio. Examples of these are:

Power Conditioner – A power conditioner

consolidates power for the entire rack down to a
single cable instead of having a half-dozen power
cables sticking out the back of your rack from
each unit. This provides for a cleaner, more
organised workspace.
Rack Mount – A rack mount allows multiple
sound sources, which is a major difference from
an amateur to professional sound recorder, where
a typical ‘desktop’ recording can only provide a small number
of simultaneous tracks.
Preamp – A preamplifier (preamp) is an
electronic amplifier that converts a weak
electrical signal into an output signal strong
enough to be noise tolerant and strong enough for
further processing. Without this, the final signal
would be noisy and distorted.
Snake Cables – Snake cables are simply a way of
organising XLR cables so that your studio
becomes less cluttered. Working in an organised
studio environment is vital in order to get your job done
efficiently without running into any blockage.
Uninterruptible Power Supply (UPS) – A UPS essentially
functions as a back-up battery, giving you several minutes of
power to shut your computer down safely in the event of a
blackout. This allows you to avoid losing any work and to be
able to save work before shutting everything down.
Reflection Filters – Reflection filters offer a
workable and more affordable alternative to
“real” acoustic treatment. Intended mainly for
vocal recording, this device allows you to
capture sound reflections before they even
enter the room, avoiding the hassle of treating
an entire studio.
Diffusers – Room diffusion is an important
element in acoustic treatment plan because it
creates a nice, natural ambience without
removing too much of the “liveliness” from the
room. Diffusers do this by scattering whatever
sound energy exists in the room allowing all
frequencies to disperse randomly, rather than build up
unnaturally in certain spots.
Bass Traps – Bass traps offer broadband
absorption across the entire frequency
spectrum and are particularly good at
absorbing lower frequencies which cause the
majority of problems in any studio, especially
smaller rooms.
Acoustic Panels – Acoustic panels are great
at absorbing frequencies in the low-mid to high
range. They are more importantly particularly
good at taming standing waves which have a
tendency to cause major acoustic problems in
rooms with parallel walls, where sound
reflections bounce back-and-forth in the same spot.

Location Based Equipment

Broadcast Recorder
TASCAM’s HS-P82 is an example of a
location-based sound recorder that offers 8
tracks and microphones for high quality
recording yet built for the rigors of location
recording with reliable solid-state performance. Standard XLR
microphone inputs include phantom power (running off the
power provided from the input) and analogue limiting, with MIC
& LINE trims controlled from recessed front-panel controls.
Handheld Recorders
There are a variety of semi-professional
handheld recorders you can get that can be
plugged into cameras. However, you would
be more likely to use a broadcast recorder for
a professional shoot then this type of
handheld device. A handheld recorder is a
small device designed for on-the-go recording of audio of any
kind which includes music, sound effects, interviews,
conversations etc.
Unlike traditional voice recorder, portable or handheld
recorders offer some better audio quality, more advanced types
of microphone builds, additional features, extra memory space
and more. A good example of a handheld recorder is the Zoom
H1 made by Zoom Electronics. It offers the essential features
of a portable recorder including a stereo X/Y mic configuration
(which is great for quality), broadcast WAV files, MP3’s and a
built-in reference speaker. It is compatible with microSD cards
and is battery powered.
Another good example of a portable/handheld audio recorder is
the Yamaha Pocketrak PR7. It provides the same audio
quality of 24-bit/96kHz recording. This handheld recorder is
often used particularly for live music recording but will work
with just about any use.

CABLES
Analogue Cables
Balanced cables have 3 wires:
 Signal (+)
 Signal (-)
 Ground
They are designed to cancel out those interference and
electrical hums.
Unbalanced cables have only 2:
 Signal
 Ground
They also have a bigger chance of picking up radio interference
and noise.
The addition of a 3rd wire in a balanced cable is exactly what
makes noise-cancellation possible.
In terms of performance:
Balanced cables are relatively immune to noise from
interference such as radio frequencies, electronic equipment
etc. Which is why they’re the standard for pro audio.
When audio enters a balanced cable both the (+) and (-) wires
receive identical versions of the signal, the only difference is,
polarity of the (-) wire is inverted.
As the two signals travel along the cable, both wires gather
noise, the same way unbalanced cables do, however before re-
combining them at the opposite end of the cable, the polarity of
the negative wire is flipped back, to once again match the
positive signal.
With the noise pattern now on opposite polarities, they cancel
each other out, leaving the original signal noise-free .
On either end of a balanced analogue cable, you will find 1 of
3 connectors:

 XLR Male – which connects to various

hardware inputs
This is used to connect to things like an
audio interface or pedal board. It has 3 pins
which are designed to fit into the sockets of
an XLR female
 XLR Female – which connects to the
microphone, and various hardware
outputs
Generally used to receive output signals. It
has 3 holes which are designed to connect to the 3 pins of an
XLR male cable.
 TSR – which connects to both inputs and
outputs
The TSR cable has 3 key features: the tip,
the ring and the sleeve.

The 3 contact points of each of these connector wires are what

carry signals from the positive, negative and ground wires.

The XLR connection is another balanced connection. The most

commonly used format in audio is the 3 pin XLR. The XLR is
used across a broad range of musical and audio applications,
due to the fact that it is both balanced, and a very secure
connection.
As previously mentioned, female XLR plugs usually receive
output signals from devices (for example a microphone) and
male XLR plugs are generally used to plug into inputs (a mic
preamp on a mixing desk for example), with male XLRs
plugging into female sockets and vice versa. Common uses are
for mic cables, monitor speakers, audio interfaces, PA
applications and much more.
The XLR connector was originally invented by James Cannon,
founder of Cannon Electric, which is why you may hear them
referred to as “Cannon Plugs”. Cannon Electric branded this
new plug the Cannon X-Series. They later added a model with a
latch to lock the cables together called the Cannon XL. And
finally, a model with a hard to break rubber compound
surrounding the female socket called the Cannon XLR.
Therefore, XLR stands for Cannon X-Series connector with Latch
and Rubber.
Other types of cables include –

The Stereo Mini Jack Connector –

Balanced
This is a small 3.5mm male connector often
found on computer headphones, headset,
microphone, and speakers, which connects to
the computer’s sound card.
TRS vs TS Connectors (Jacks)
TS cables have one signal conductor (positive)
and ground. They are generally used for mono,
unbalanced signals, often associated with
guitars and amplifiers. TRS cables have two
signal conductors (positive and negative) and
ground. A TRS connector is typically used for
mono unbalanced signals as well as stereo
signals. They are identical in terms of their appearance.
The RCA Connector – Unbalanced
An RCA connector, sometimes called a phone
connector, is commonly used to carry audio
and video signals. They often come in colour
coded pairs as they are used to connect to a
left and right speaker. However, nowadays their function has
been replaced by stereo mini jack connectors.
Midi Cables
A musical instrument digital interface
cable is used to connect keyboards and
other electronic musical devices to
computers. While the MIDI cables are
labelled as ‘In’ and ‘Out’ plugs, they do
not work if they are connected to the
same labelled MIDI ports on an electronic
instrument.
Digital Cables
The two most common types of digital
audio cables are coaxial and optical digital
audio cables. Coaxial digital cables are very
similar to the legacy, RCA analogue type which most
consumers have experience using.
USB
USB A to B cables are used as midi cables for
electronic instruments. They are the
updated, newer style of midi cables as they
are generally easier to configurate as they
can plug straight into a computer. Firewire
This outdated cable was used for high
speed transfer of audio and for carrying
visual material.
Thunderbolt
The Thunderbolt cable is used in modern
day Mac laptops – MacBook Pro’s. They
are similar to USB-C cables but are more
high speed. All thunderbolt ports in
computers act as USB-C ports but not
vice-versa.
Recording in Interior and Exterior settings
When recording in different settings and environments it is
important that you have the necessary equipment to be able to
gain high quality audio.
When recording outdoors, the ambience of the natural sounds
often adds depth and interest to your final product. It can also
create a sense of character and make you seem
more authentic.
There is some equipment that applies to most
locations, whether that be interior or exterior.
You will need a computer with a DAW, a good
example is the MacBook Pro with Logic Pro X
software on it. XLR cables are usually required
in order to be able to connect your equipment. Good cables to
use that can be extended quite long are Hosa XLR-115 XLR3F
cables as they stretch 15 feet and are often used as stage
cables. A pre-amp may be necessary to remove any
background noise that may occur during recording. A good
example is the Avalon VT-737SP. You will also need a USB type
A to B cable with a USB type B to thunderbolt adapter in order
to connect the interface to the MacBook. Lastly, an audio
interface is necessary and the Focusrite Scarlett 8i6 is a good
example.

Recording interviews in a field

The first step is to create a minimalist setup with the
basic yet necessary equipment. You will probably
require a portable field mixer to layer all the sounds
into one to create the ambient feel. The best type of
microphone for outdoor use is a shotgun
microphone, for example the Sennheiser MKE 600, as
they reject sound from the rear. However, if your
interviewee is not staying in the same position for the
interview, a handheld dynamic microphone may
be better suited, such as the Shure SM58-LC.
A good microphone accessory to pair this with is a boom pole,
such as the RODE Boom Pole 3, as they can withstand noisy
outdoor areas.
To cancel out unnecessary background noise, it is
important to have good wind protection, for
example, in terms of recording, a windscreen is
useful in this environment as it reduces any chance
of wind effecting the microphone’s capsule. A good
choice is the Neumann WS2.
Although most background noise is wanted to be
eliminated, capturing ambient sounds using a
portable recorder, such as the Rolan R-05 Studio,
allows you to create the atmosphere of the setting in
subtle ways to add to the overall depth of the shot.

Recording interviews in a large room

Often, large and spacious rooms invite the presence
of echoes. When recording in an echoing room with
lots of sound reflections, it is important to eliminate
the muffling this can create in the audio. The best
choice of microphone in this setting is a cardioid
microphone as it will only capture the input vocal
audio without picking up any reverb. A good
example would be the Audio-Technica AT4053B.
A sound blanket is a good piece of equipment to
use as it adds depth and volume to the spacious
room, allowing the sounds to bounce off of these
materials rather than the structure of the building, thus
eliminating echoes.
A reflection filter seems necessary but will not be
needed when paired with a cardioid microphone.
However, as you will be near your interviewee, you
should instead use a pop filter. A good example is
the Musician’s Gear Double 6”.

Recording interviews in a busy street

Similar to recording audio in a field but more
background noise will be noticeable. A shotgun
microphone such as the Audio-Technica
ATR6550X Condenser is a good choice as it is a
smaller version but has all the same features and
no long cables, which is ideal for public, outdoor interviews. In
this setting, it is better for the interviewer to hold the
microphone when speaking to the interviewee so that the
microphone can pick-up high-quality audio from a direct source
rather than capturing surrounding sounds too.
A windscreen may be helpful in this situation as
it will help decrease the distortion and breathing
noises when placed near an interviewee. A good
example is the Rode Ws7 Deluxe Windshield.