Monitoring
Phil Booth for Future Music Magazine
article featured on Futuremusic.co.uk
Decent audio monitoring is essential for any studio, yet is often overlooked. In this first of a two-part series, engineer Phil Booth reflects on the only instrument in the studio to make a sound...
IF I WERE to ask for everyone's impression of what a particular instrument,
effect or even microphone sounds like, most of you would have a fair mental
picture (or sound) in their mind. Website forums and email groups are swamped
with users' opinions on the sonic merit of all sorts of gear, from acoustic
instruments and hard or soft synths, through to the alleged pristine clarity
of 192kHz/24-bit analogue to digital converters. But without the benefit of
reliable sonic binoculars, how can anyone be entirely sure they have the right
view of what the music world sounds like?
Enter the role of the loudspeaker, for without it, all these opinions would not be possible.
The studio monitor
The loudspeakers used for studio applications are often called reference monitors.
Their precise and uncoloured reproduction across the audible frequency spectrum
is what sets studio monitors apart from the sorts of speakers that accompany
the average hi-fi system.
It's not that hi-fi is a bad thing. In fact, high-end audiophile products are often equally at home in a pro studio. However, it's the majority of systems out there, indiscriminately shoved in the 'high fidelity' basket, that you must be wary of. Behind their fancy façade of flashing lights and exotic plastic are weak amplifiers with poor distortion specs, nasty drivers with limited frequency range and non-linear power output, plus a maze of ports that alter the sound like the spine-chilling resonances of a church organ.
With sub-standard monitoring, you'll end up altering your music to make your speakers sound good, only to find the mix sounds wrong when played on other systems. This is the main reason why certain speakers (and amplifiers) fail to meet the expectations demanded by audio engineers for any sort of critical work. We need monitors we can depend on to produce mixes that will translate as expected to the intended audience's listening environment: their home, car or even Walkman.
If you're considering a new amplifier and speaker system for your studio, get monitors designed with production work in mind, and leave the steroid-enhanced, neon lightshows for the lounge, or better still, on the showroom floor.
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Sound in motion
You've probably already seen your woofers (the largest of the speaker cones)
flapping in time with heavy bass and this back and forth motion is the essence
of how a speaker works. The pounding on the air molecules immediately in front
of the speaker generates a disturbance that ripples through the atmosphere at
a whopping 343m per second, or 1,235km/h (at 20 degrees Celsius).
To visualise this concept on a smaller, two-dimensional scale, toss a pebble into a still pond and see how the waves emanate outwards in all directions. The water itself doesn't travel anywhere as such. Rather, it moves up and down on the spot and it's the motion of the up/down wave motion from one section of water to the next that appears to travel onwards. A stadium of sports supporters doing a Mexican wave illustrates this brilliantly. Each spectator stays in the same seat, but as each person stands up and raises their arms before sitting down to allow the person next to them to do the same, a wave of activity can be seen to move around the stadium.
As sound waves, the series of high and low fluctuations of air pressure eventually arrive at our ears to push and pull our eardrums at the same rate as the source of the sound. Whether the sound actually occurs when the listener isn't there is probably a topic for Future Philosophy magazine, but take away the air (as in the vacuum of outer space) and there is no sound. Without a medium to transfer the sound wave energy, all that's left is an eerie silence.
Incidentally, the tighter the molecules are packed together, the faster the sound waves travel. Underwater, the speed of sound jumps to 1,482m/sec, while steel conducts sound at the breathtaking speed of around 5,960m/sec, or 21,456km/h! There's something to think about.
In the analogue audio world, the motion of high and low pressure of soundwaves
is represented by electrical current that alternates back and forth between
a positive and negative state, thus the term 'alternating current'. At the amplifier
stage, these tiny voltage levels are boosted before travelling down the speaker
wires to the connectors at the back of the speaker.
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The engine room
Since Earnest W Siemens first described the dynamic or moving coil transducer
in 1974, many inventors and engineers set about refining his concept into the
loudspeaker you see and hear today.
Despite the advancements that have been made to the initial design over the years, the basic principle of the 'magneto-electric apparatus' has remained unchanged right up to the present day.
The part(s) of the speaker that do the hard work are called the drivers, and are (usually) identified by a circular cone or dome (known as the diaphragm) mounted on to the speaker's front panel. At the very heart of the driver, connected to the back of the diaphragm, is the voice coil. This consists of a length of fine wire that's wound tightly around a hollow tube known as the former.
Anyone who remembers back to their school science lessons may recall wrapping a coil of wire across the face of a compass, connecting a battery and observing the electromagnetic effect as the direction of the needle shifts from its natural north position to sit perpendicular to the coil. A similar electromagnetic field is produced by the voicecoil once connected via speaker wires running back to an amplifier.
In a speaker driver, the voice coil is suspended inside a powerful, doughnut-shaped permanent magnet. For maximum sensitivity, the voicecoil is placed in close proximity (but not touching) the inner surface of the magnet. To prevent damage through contact, the voicecoil is carefully centred using a round piece of corrugated material known as the spider. Being semi-flexible, the spider also acts as a spring to return the voicecoil to its stationary position when no signal is being applied. The magnet, spider and voicecoil are mounted together on a circular metal frame called the basket.
The basket is also where the outer edge of the diaphragm is secured into alignment, via a ring made from foam, plastic or rubber, known as the surround (or suspension). The surround seals the rear of the driver once the driver is mounted in its enclosure and is sufficiently flexible enough to allow the diaphragm to vibrate back and forth. If this were not possible, the speaker could not function. It also provides additional springiness to assist the spider in pulling the diaphragm back into its stationary position when there's no signal.
Lastly, the surround helps dampen vibrations travelling along the surface to the outer edge of diaphragm itself.
When the electrical signal from the amplifier is applied to the voice coil,
it forms an electromagnet. Remember back to those science lessons again and
the effect when two magnets meet. Opposites poles attract and pull together,
while the same poles repel with force. As the signal alternates between positive
or negative to mimic the audio waveform, so the voice coil switches its electromagnetic
polarity between north and south to either push away from the surrounding permanent
magnet, or pull further inside it. It's this pushing and pulling that ultimately
drives the diaphragm like a piston to excite the air around the speaker and
produce the sound we hear. So there you have it! A speaker driver is really
just an electric motor that relies on alternating current to switch directions
at the blink of an eye.
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Frequency and the human ear
The distance measured between the two peaks or crests of the waveform (or the
time interval between a completed push/pull action of the driver) determines
the frequency or pitch of the sound we hear. The height or amplitude of the
waveform (or the distance the diaphragm travels, known as the excursion) determines
its intensity or volume.
While the speed of sound is fixed at a given ambient air temperature (343m/sec @ 20°C), a deep, low sound will feature widely spaced crests, meaning a relatively limited set will pass by our eardrums each second.
This number of cycles per second is described in Hertz, after the 19th-century German physicist Heinrich Hertz. If the cycles occur in the region of 20 to 200Hz, we recognise the sound as bass. As the number of cycles per second increases (and hence the spacing between wave crests is shortened), we perceive the sound as higher in pitch. In the region of 200Hz to 2,000Hz (or 2kHz) lie the mid-range frequencies.
As the cycles occur more rapidly and the waveform crests become even more closely packed together in the region of 2kHz and above, the sound is said to be high frequency. The frequency range for young and healthy human hearing is about 20Hz to 20kHz (20,000Hz).
Unfortunately, our ability to perceive the upper frequencies tends to roll off to around 17-18kHz as we grow older, while damage through excessive noise exposure may cause a dip or notch in the sensitivity around the 4kHz level.
Different drivers
A single driver with frequency response spanning 20Hz to 20kHz, or the range
of healthy human hearing, would be the Holy Grail in loudspeaker design. However,
different frequencies require different drivers, with each one optimised specifically
to cater to a particular part of the frequency spectrum (see the diagram below
to see how this works in practice).
Bass requires a lot of energy to produce a decent level, meaning larger elements are required to push such massive volumes of air. The largest driver in any loudspeaker is called the woofer, and is usually somewhere between 5 and 15 inches in diameter, depending on the bass response and volume output required. Above 2.5kHz, the woofer's size and weight becomes too sluggish at responding to the rapid fluctuations contained in high frequencies and is therefore quite poor at reproducing treble.
To take care of the high frequencies is the tweeter. It's much smaller and more delicate in design, to accommodate the ultra fast vibrations required, typically from 2 to 5kHz through to beyond 20,000 times per second.
While woofer cones are generally made from compressed wood pulp, polypropylene (plastic) or even Kevlar, tweeters are regularly made from metal alloys or silk pressed into a dome.
When a tweeter is used in conjunction with a woofer to reproduce the entire audible spectrum, the pair is known as a two-way system.
Sometimes speaker designers will accommodate the upper limitation of the woofer's response and the lowest end of the tweeter by adding a third (or fourth) driver, known as a mid-range. The system is now described as three-way (or four-way, etc, depending on the number of frequency bands covered by separate drivers). To separate the music into the frequencies best handled by the woofers, mid-range and tweeters, the audio signal is passed through a crossover. (For more on the crossover design, see the Crossing over box on p126.)
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Ported versus sealed
For the effective transmission of sound from a driver, the front face of its
diaphragm should be isolated from the rear. Otherwise, as the cone or done pushes
forward, air will rush around its edges to fill the low pressure void created
behind it, and vice versa, and so hardly any of the sound energy will make its
way to your ears.
In 1954, Acoustic Research released the AR-1, a bookshelf loudspeaker that was the culmination of efforts between Henry Kloss, an inventor and Edgar Villchur, an engineer and the company co-founder. Their significant development heavily influenced modern loudspeaker design and involved a simple procedure of mounting a redesigned driver into an airtight box or enclosure of a specific size or volume. This acoustic suspension method immediately allowed for a reduction in overall cabinet size from previous methods. Until then, it wasn't uncommon to find a bass speaker system for a cinema theatre would be up to 14 feet tall!
The acoustic suspension or sealed system exhibits a very smooth frequency response and is often favoured over other methods by many studio pros and home audio experts alike. The air inside the box acts as shock absorber that keeps the natural inertia of the driver under control for a tighter bass response with less distortion and overhang. Although smaller than its predecessors, the sealed enclosure still has to be relatively large to offer enough air volume for the driver to effectively compress, and it's typically inefficient, as more amplifier power is required to match the volume output of other enclosure designs.
The most common enclosure type, patented by Albert L Thuras in 1932, is the ported or bass reflex design (again, see the cross-section diagrams on p126). Rather than sealing the back of the driver from the outside world, the bass reflex box incorporates a tube or port with strict diameter and length dimensions that tune the box, like you would any instrument, to around the lowest frequency the driver can produce (known as its resonant frequency).
This immediately enhances the box's bass output, and by allowing the driver to breathe (similar to blowing into an empty milkbottle) it becomes far more efficient, at around +3dB louder than a sealed system at the same amplifier output. This can also mean a ported system requires half the amplifier output as a sealed system for the same speaker volume.
The trouble is that bass reflex systems can run into problems with excessive ringing or peaks in the bass around the port-tuning frequency, particularly if the enclosure and port dimensions have not been designed properly. Also, without the airtight suspension of the sealed design to dampen the excursion of the driver, the woofer can lose its linearity or accuracy when tracking the audio waveform at louder volumes, or at frequencies below the resonant tuning of the enclosure.
Plus, there's nothing more disturbing than copping a blast of air or audible chuffing from the speaker's front port on every bass note. Lastly, if the port is mounted on the rear of the box, the speaker's proximity to the immediate wall or corner behind it becomes a larger factor in the level of bass output.
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It's a personal choice
Regardless of whether you choose sealed or ported, active or passive, subwoofers
or three-way drivers, all monitors sound different. Additionally, one person
will perceive sound differently from the next, so it's really up to you to choose
the right monitors for you.
Spend some time getting accustomed to your own monitor's distinctive quirks, in the same way you would learn to use any tool in your studio. Listen to numerous recordings you're familiar with and note how bass, treble, vocals, synths and guitars sit in the mix. If you're trying a set of studio monitors for the first time, and you've previously only worked with a hi-fi or PC system, you'll be amazed to discover the detail you've been missing all this time. With a firm understanding of your monitor's traits, you'll be well on your way to achieving the perfect mix. FM
Pros and cons of nearfields
The majority of monitors used in the studio are nearfields. These specialist
speakers are characterised by a very even or flat frequency response across
their natural range, and a sweetspot that's designed for the operator to sit
in close, hence the term nearfield.
The nearfield's relatively small enclosure design and a woofer diameter of
around five inches help reduce the effect of the room's natural resonances or
modes, so let you closely monitor the music without being influenced by reflections
bouncing back from the walls, floor and ceiling, thus messing with the monitor's
direct patch to your ear. This promotes a smoother frequency curve below 300Hz
but, as a consequence, it can be of great detriment to a deep bass extension.
If you're producing particularly bass heavy material and want to hear exactly
what's going on down below 80Hz, you may wish to consider nearfields with a
larger woofer diameter of eight inches. Otherwise, 6.5 inches is fine for most
purposes, or supplement a smaller box with a subwoofer. Just watch out for those
room modes though!
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Crossing over
Before the music signal can be played through a multi-driver speaker system,
it must be first divided up using a network of high- and low-pass filters, known
as a crossover. This splits the full spectrum waveform into the frequency bands
best handled by each specific driver. Without the crossover's high-pass filter
to protect the tweeter, its delicate diaphragm could be ripped apart and the
voice coil fried if it came in contact with the powerful low frequencies normally
reserved for heavy bass. To avoid detracting from the tweeter's performance,
a low-pass filter limits all but the lower register of frequencies on its way
to the woofer.
The filters in a crossover aren't effective at blocking all frequencies above or below its allocated cutoff points so there'll always be an overlap of frequencies shared by both drivers that can cause an uneven frequency response. This can be minimised by employing steeper filter slopes. So rather than using a high-pass filter circuit that drops off at the rate of -6dB for every octave below the cutoff of around 2kHz (NB: half or double the frequency to span one octave), a better solution is a -24dB/octave design. This way, in a mix containing bass below 250Hz, the level of the music signal actually making its way to the tweeter will be more than -72dB quieter, thus minimising the risk of driver overload.
Passive, powered or active?
Over recent years, the trend has shifted from using a pair of speakers with
an amplifier, to self-powered units. The obvious difference is the powered speaker
connects directly to a music source, such as your mixer or soundcard, without
the need for a separate amp, thus eliminating a spaghetti of cables. Still,
the powered monitor is often nothing more than a regular amplifier bolted to
the back of a passive speaker box and consequently offers no real sonic benefits.
The real advantage lies in an active system. It's similar to the powered - an amplifier is incorporated into the speaker enclosure - but active monitors use multiple amplifiers to power each driver independently. This means delicate treble is not disturbed by the rigours of producing bass. The crossover in an active system is fully electronic and presents far less distortion and phase anomalies than are typically introduced through inefficient passive crossover networks. By knowing each driver's exact performance statistics, manufacturers can tailor a sophisticated electronic EQ for a more even frequency response, or to suit your room conditions or personal taste. Some systems (like Mackie HR824) even actively track the position of the driver and apply more amplifier power should it differ from the input waveform.
Active monitors often appear expensive, but once a suitable amplifier is matched up with a pair of passives of comparable standard, the total cost of an active system can be similar if not cheaper.
Of course, there's nothing to suggest that a £1,000 active system will outperform a £10,000 passive system, but with the theoretical advantages outlined above, active monitoring appears the way to go.
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WHICH MONITORS FOR WHICH BUDGET?
There are more quality monitors than can be listed here, but these are decent
active models with built-in amplifiers at three different price points. If you
choose a passive set-up, team it up with quality amplification, such as the
Hafler P1500 or P3000 to bring out the best of your drivers...
Tannoy Reveal Actives
(£404*)
A confident performer for the budget-conscious studio owner who needs reliable
feedback on their mixes. Twin 50W RMS amplifiers discretely power a 6.5-inch
woofer and one-inch soft dome tweeter. They're a little shy below 60Hz (-3dB),
although this may be supplemented with an extra octave of bass extension from
the new Reveal Sub 10 subwoofer. They're also available in a passive version.
TEAC: 01923 819630
www.teac.com
www.tannoy.com
Alesis M1 Actives
(£499*)
The original passive Alesis Monitor Ones proved highly popular with home and
project studios in the late 90s. The newer M1 Actives have been redeveloped
with new high-tensile, carbon-fibre woofer cones and a front-mounted, twin-ported
enclosure that's powered by 75W RMS and 25W integrated amplifiers.
Numark: 01252 341400
www.numark.com
www.alesis.com
KRK V8s
(1,299*)
These are instantly distinguished by their eight-inch Kevlar woofer that gives
a light yet rigid and fast transient response, set below a silk dome tweeter
on a vented cabinet. They have all-active crossover electronics and independent
120/60W RMS amplifiers to cover both drivers. The V8's natural character aids
the consistent transition of your mixes to other systems, but the non-hyped
sound may not woo potential users upon first audition.
Protape: 020 7616 550
www.protape.co.uk
www.krksys.com
Mackie HR824
(£1,383*)
150W + 100W RMS active amplification, acoustically time-aligned drivers and
a waveguide-assisted high frequency system promote a tight, detailed response
with precise imaging and a wide sweetspot. With their passive radiator, the
824s feature one of the deepest frequency responses in its class (38Hz@-3dB),
making it a favourite among engineers, although the enhanced bass performance
may sound weird to newbies.
Mackie: 01268 571212
www.mackie.com
Genelec 1031As
(£2,298*)
A highly popular model featuring 120W/120W (RMS) bi-amplification, a solid eight-inch
woofer and one-inch waveguide-assisted tweeter, in a vented enclosure to provide
a beautifully detailed sweetspot and favourable bass. Their amazing clarity
can give the unwary a false impression of how mixes will sound on sub-standard
stereo systems. Used in media post-production and project studios.
SCV London:
020 8418 0778
www.scvlondon.co.uk
www.genelec.com
Quested Q210s, VS2205 and beyond...Out of reach from all but the serious professional
or lottery winner, the Quested range are found in recording, mastering, film
and TV post-production studios worldwide, including Abbey Road Studios in London.
Media Tools:
020 7692 6611
www.mediatools.co.uk
www.quested.com
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Jargonbuster
Hertz: The number of complete cycles from a positive to a negative state a waveform
makes in a second. So, with a bass note of 100Hz, the woofer cone vibrates back
and forth 100 times every second. (Frequencies over 1,000Hz are represented
in kiloHertz (1000Hz = 1kHz).
Mid-range (driver): Optional driver that caters for frequencies that are otherwise
overlapped by woofer and tweeter. By using a mid-range driver, the woofer can
be left to do what it does best (bass), and the overall system can deliver a
louder output without fear of overloading the tweeter.
Subwoofer: Special driver incorporating bulkier magnet and voice coil assemblies
to deliver subterranean frequencies, between 20Hz and 80Hz.
Sweetspot: The area in line with the front of speaker (on-axis) where the frequency
response sounds most balanced. As you move to either side of the front of the
speaker (off-axis), the frequency response deteriorates (particularly noticeable
with treble as bass tends to be less directional). High frequencies also drop
off with distance so a midfield to main monitor will project its mid range and
treble across a room, while a small nearfield monitor is best listened to up
close.
Tweeter: Driver optimised to deliver the highest frequencies in a speaker system,
from around 2.5kHz to over 20kHz.
Woofer: Usually the largest driver in a speaker system, woofers are capable
of reproducing a frequency range somewhere between 30Hz and 2.5kHz.
Who uses What?
Astral Projection: Genelec 1032 & Tannoy AMS8
Avalanches:A cheap ghettoblaster!
Basement Jaxx:Yamaha NS10M, JBL Control 10
Bentley Rhythm Ace:Mackie HR824
Chicane: A&R Red Box, Quested F11P
Enigma: Dynaudio BM15A
Fatboy Slim: Yamaha NS10M
Future Sound Of London:UREI 838 & Genelec
Kinobe:Tannoy Reveals, Yamaha NS10M
Lamb: Mackie HR824s
Laurent Garnier: Mackie HR824s
Lo-Fidelity Allstars: Auratone, Dynaudio M1 & Alesis Monitor One
LTJ Bukem: Genelec
Luke Slater: Tannoy System 12 & Yamaha NS10M
The Orb: Genelec
William Orbit: Yamaha NS10M
Orbital: Blueroom Minipods & Yamaha NS10M
Plaid: Genelec
Pop Will Eat Itself:Alesis Monitor Ones
Rae & Christian: Dynaudio BM15s & Yamaha N15 & NS10M
Roni Size: Dynaudio M2
The Shamen: Dynaudio M2, Richard Allen 8 Series II & JBL Control 5
The Strongroom: Coastal Acoustics Boxer 5 Series; Genelec 1029, 1031, &
(Orbital/Chemical Bros/ 1032; Yamaha NS10M
Prodigy/Underworld)
Way Out West: Genelec 1032s, Yamaha NS10M
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Choosing a pair of audio reference monitors is a necessary part of setting up a studio, but knowing the best way to extract their optimum performance is the secret to the studio built for business...
After covering monitor history and their internal workings last month, this
time round I'll start with what to do with your brand new monitors once they're
out of the box. And the first step is to find them a suitable place in your
studio.
Sit down in your regular mixing spot. Ideally, the imaginary point directly
between each speaker's tweeter and woofer should be level with your eyes, or
more importantly, your ears. The table used for your mixing desk and/or PC might
appear the most logical and convenient placement, but sonically speaking, it's
less than ideal. The graphs you often see boasting a speaker's ultra flat frequency
response are measured from a point that's directly in front of the speaker,
known as on-axis.
As one leans to either side, or up and down, in other words off-axis from the
monitor's face, the frequency response will change with a notable drop in the
treble due to the highly directional nature of the tweeter. To remain on-axis,
special speaker stands may be needed.
You can make your own, DIY-style, using sheets of dense MDF (medium density
fibreboard), cut into lengths that, once mounted around a square base and top
plate, form a tall, four-sided column. For stability, throw several sandbags
down the central hollow. This will also help reduce speaker vibrations from
travelling down the columns and being lost in the floor. You may be familiar
with wall-mounted brackets that can be used to elevate rear speakers (as part
of a multi-channel home theatre system), but in context of accurate studio monitoring,
the close speaker proximity to the wall behind them can lead to an over emphasis
of bass output. More on that later.
The distance the two speakers are spaced apart is important too. Too close together
and each sound's precise placement within the stereo field will be hard to define.
Too far apart and you'll experience a strong sense of emptiness from the centre,
and extremely exaggerated stereo panning. Decide on a happy medium by placing
each speaker into the two corners of an imaginary equilateral triangle while
your head makes up the third point (see the diagram above).
Lastly, turn the toes of each monitor inwards so the front panel is once again
on-axis with your ears. High mounted speakers should be angled downwards. Tip:
If you can't see the sides, top, or bottom panels of each monitor cabinet, you'll
know they're in the correct position.
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On a final positioning note, be wary of using non-magnetically shielded speakers
near CRT (cathode ray tube) computer monitors or you might severely distort
the colours. The solution is to use
magnetically shielded speakers. These feature drivers with an additional magnet piggybacked on to the back of the regular magnet in the speaker to help cancel out stray magnetic fields from around the enclosure and with negligible impact on the speaker's performance.
To sub or not to sub?
Fully-fledged studios use large and frightfully expensive full-range monitors in conjunction with their nearfield monitors for making A/B comparisons before finalising a mix. The massive cabinet dimensions and large drivers contained in these full-range speakers provide a bass response that can't be matched by the small nearfield monitors we often find ourselves limited to using.
With the dawn of home theatre sweeping the average consumer into a sound technology
buying frenzy, more and more people are experiencing music enhanced with subwoofers.
This places a greater importance on proper bass management in your mixes.
One of the most common nearfields in use today is the now discontinued Yamaha
NS10M. Typical of its size, the NS10M features a frequency response extending
down to around 60Hz. Note a competent subwoofer system might bottom out at around
15Hz. This difference of 45Hz may not appear much, and that's true when referring
to the upper end of the frequency spectrum, but bear in mind, the frequency
value doubles each time one moves an octave along the keyboard.
Therefore, the difference between 15Hz and 60Hz actually equates to two whole
octaves of bass notes that can't be heard from the NS10Ms! That being the case,
although not crucial, if you're mixing modern music focused around heavy bass,
you might choose to consider supplementing your nearfields with a quality subwoofer
to help hear precisely what's going on 'down there'.
Be very wary of the cheap subwoofers thrown in with low-end home theatre and
PC multimedia applications. While they might sound great shaking the floor and
rattling the windows during cheesy Hollywood action flicks, their musicality
is generally poor, and they tend to boom and drone in a monotone manner from
their poorly tuned cabinets. Plus their weak amplifier and drivers fail to deliver
any definition or detail. For studio-quality subs, look towards the manufacturer
of the monitors you currently use. You'll find sub offerings by Genelec, KRK,
Dynaudio Acoustics and, more recently, Mackie.
If you do go the sub route (and by no means are they necessary for quality mixing)
always be wary of how your mixes might translate back to a regular, small hi-fi
system. Too much sub bass and not enough mid-bass presence in the studio may
result in mixes that sound as if they're lacking in bass on the average boombox,
and totally overwhelm a thumping club system. This is where your regular nearfields
switched to full range mode will come in handy to provide that smaller perspective.
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The road to hi-fidelity
One of the first recommendations offered to somebody wishing to set up a small home studio is to forget their hi-fi speakers and go out and invest in a pair of quality studio monitors. The term 'hi-fi speaker' is perhaps a misnomer, particularly when referring to the all-in-one package systems. While perfect for partying around the house, these units tend to feature a distinct lack of high frequency detail, a clouded mid range, plus an unnaturally hyped and boomy bottom end, all of which is in complete contradiction to their high fidelity banner.
Decent studio monitors (and decent hi-fi speakers too) exhibit far less colouration
that's prone to masking problem areas in your mix. The wonderful dynamic range
allows you to hear the precise degree of compression you might be adding, and
the notably flatter frequency response lets you make EQ decisions in good faith,
rather than compensating for the inadequacies of your speakers. But watch out
as you climb the quality stakes because the marked improvement in sound clarity
can make everything sound good. Possibly too good.
The art of mixing is all about creating an aural landscape by taking sounds
and carefully placing them in the foreground, background and anywhere in between.
The more clarity your monitors provide, the more detail you will be able to
pick out in the background.
Let's assume a lovely high-end pair of monitors allow you to make out the normally
faint sound of a pin drop. What superb clarity! As the mastering engineer for
the project, I query the client's inclination towards haberdashery only to learn
the sound of the pin hitting the floor is supposed to be a major part of the
song. Not a problem it would seem, as the studio monitors reveal the sound without
a hesitation and so the album goes off for CD duplication.
Now take the same mix and play it on a mediocre consumer hi-fi system. What
a disaster! Without the same pristine clarity of the studio system, the pindrop
effect seems to have been lost in the mix. Could our excruciatingly expensive,
state-of-the-art monitors really have been lying to us all this time?
It's this false sense of security that can arise when mixing on high-end monitors
that has many engineers resorting back to their trusty middle-of-the-road Yamaha
NS10Ms (or similar) as a reality check on how their mixes are likely to translate
to the masses. This is especially crucial in nailing the all-important vocal
balance to ensure it hasn't slipped into the background.
It's like cruising down the musical freeway on a beautiful spring afternoon
in your fancy exotic convertible with a magnificent mountain range stretching
along one side and the glistening ocean on the other. The birds and bees are
seen going about their business among the native wild flowers, and there's not
a single cloud in the sky to rain down on your parade. Life in your studio with
your flashy monitors simply can't get better!
Now back in the real world of consumer hi-fi where the beautiful serenity has
been replaced by a washed-out sky and heavy fog. Suddenly your focus is drawn
to just keeping all the musical signposts visible. With high-quality monitors,
one can easily make out the vocals, for example, even if they're sitting far
back in the mix. However, a ghettoblaster just won't have the same level of
detail and therefore it's crucial to have the vocal mix balanced just right,
or your mix will steer itself off the road!
When panning your sounds across the stereo field, keep in mind also that some
of your intended audience's sound systems may not exhibit imaging as sharp as
your own monitors. It may be necessary to exaggerate any panning to ensure the
effect is noticed. Pay particular attention to any sounds sitting around the
11 o'clock and 1 o'clock pan positions (with 12 o'clock being the centre or
mono position of the pan knob). Any slight stereo positioning can easily be
lost among the true mono sounds in a mix.
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Room to move
The shapes and materials used in the construction of acoustic instruments heavily govern their tonal character, while the surrounding walls, floors and furnishings simply become an extension of the instrument itself by inducing deep resonances (so you get peaks and ringing in the lower frequency response) called modes (see diagram below). Each room sounds different, so establishing proper bass levels in your mix can be difficult.
At the same time, delayed early reflections of mid to high frequencies interact
with the direct sound to inflict comb filtering (sharp notches in the frequency
response) that resembles a subtle, stationary flanger or phaser effect. If there's
too much reverberation build-up, it's likely to mask the true make-up of the
mix.
As a result of the room modes and reflections, our near perfect monitors adopt
an entirely different flavour once removed from the manufacturer's echoless
testing facilities. Regrettably, the very room devoted to your studio operations,
despite all its good intentions, can seriously tarnish the reputation of even
the most elite monitors
The motion of sound is best understood when compared to the action of waves
in water. As soundwaves, they consist of series of high crests and low troughs
of air pressure that ripple through the atmosphere. When two independent waves
come together, there are interesting and important consequences.
Two crests (or two troughs) will sum to become bigger and therefore louder. These waves are said to be in-phase. Should a crest and a trough interact, they are said to be out-of-phase, and subtract from each other, and if the waves happen to be perfectly equal in frequency and volume (amplitude), yet completely out-of-phase from each other, they'll cancel each other out altogether to leave no sound (see the When waveforms meet box above).
Low frequency sound waves below 300Hz slosh around the room like water in the
bath. When a frequency is emitted with a wavelength (distance between two waveform
crests) that's precisely double the distance between two opposing walls, upon
hitting the wall, the rebounding wave switches phase and meets up with the second
half of the oncoming waveform perfectly in-phase to reinforce it.
This is the known as the fundamental resonant mode of the room. Additional modes
spaced further up the frequency scale, known as harmonics, occur at frequencies
that are multiples of the fundamental.
What were once proudly advertised as totally flat monitors are now looking considerably
bumpy. Of course, a room is three dimensional so the calculations should include
frequency values for all opposing walls, floor, ceiling and diagonals. This
can be performed fairly simply using the mode calculator at www.mcsquared.com/metricmodes.htm.
Irregular shaped rooms work best as they present plenty of modes to even out
the overall response. On the other hand, a cube-shaped room (3m x 3m x 3m),
or a room where the long walls are exact multiples of the short walls (5m x
2.5m x 2.5m) suffer the worst. While containing fewer modes, their peaks are
spaced distinctively apart, plus each mode will be heavily reinforced by the
soundwaves bouncing around similar distances.
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Standing still
Have you ever walked around the room to discover the bass sounding particularly heavy in one spot, while elsewhere there's a considerable lack of bass? This occurs with the way modal frequencies are bounced between parallel walls. Because their wavelengths are directly related top the room's dimensions, the point where there's maximum air excitation (due to the convergence of in-phase waveforms) occurs in the same places in the room, while elsewhere, conflicting waveforms will be cancel themselves out.
The areas of perfect cancellation are known as room nodes, and are characterised
by the areas completely lacking in bass, while the areas between the dead spots
or nodes will peak at double strength, and are called anti-nodes. Check out
how standing waves form using the 'both sides closed' model at http://home.a-city.de/walter.fendt/physengl/stlwaves.htm.
From the animation, you can see how each harmonic introduces more and more nodes
(and anti-nodes) at strategic places. As each node appears to remain stationary
in the room, the term 'standing wave' is born.
The potential tragedy when working within the space of a room node is innocently
cranking up the EQ with zero effect due to the perfect phase cancellation at
that particular frequency, only to discover your mix is bursting when played
in a different shaped room. Likewise, being blissfully unaware of your room's
natural resonances and anti-nodes, there's a good chance you'll be EQing these
frequencies out of your mixes, only to have your tracks sound disfigured when
played in another venue.
All is not lost
All this talk of modes, nodes and anti-nodes sounds more like something from a sci-fi novel than a relaxed music studio, and thankfully there are ways to minimise their effect. The ultimate would be a custom designed room with an angled ceiling and non-parallel walls, but for starters, be sure to not position your monitors up hard against the walls, and especially away from corners, as this tends to reinforce the room modes.
For persistent standing wave issues, specially tuned Helmholtz resonators, or
bass traps, are available and they can be installed along walls and corners
to soak up the problem frequencies before they get a chance to rebound and mess
with the direct waves. This will serve to provide a more accurate frequency
balance in the listening position.
Up and above 300Hz, the wavelengths are too far short for modes and standing
waves to be of any noticeable concern, and the focus turns to reflections and
tiny echoes that we recognise as reverberation. This may be treated using acoustic
tiles along walls opposite the speakers and anywhere there's a direct reflection
from the speakers to your ears.
There's no need to get carried away with treating the entire room. Acoustic
tiles are only effective above 1,000Hz, so eliminating these frequencies altogether
will leave you a room that's decidedly lacking in high frequency detail and
a notably muddy bass response due to the remaining modal influence. Balance
is the key.
Your best weapon against problematic room acoustics and individual monitor traits
is by simply knowing how your speakers sound in your personal studio space.
Listen to all your favourite CDs through your monitors.
That's the results you're striving for from your own mixes. Having an active
awareness of the acoustical room influences will undoubtedly help your mixes
translate to the wealth of different playback systems.
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Horizontal vs vertical?
Should your monitors sit on their sides, or upright? After years of sighting Yamaha NS10Ms lying horizontally across meter bridges of top recording studios pumping out Number 1 hits (and even more absolute flops), does it really matter which way your monitors sit?
A couple of points against horizontal placement:
• Having the tweeter positioned close to hard surfaces, such as a table, shelf, or your mixing desk, can generate delayed reflections of the mid to upper frequencies. Recombining the slightly out-of-phase reflections with the direct sound can cause sharp notches or dips along the frequency response that resembles a comb, and hence the term comb filtering.
• If you sway from side to side in front of a vertically aligned box (ie, tweeter over the woofer), the distance the sound travels from both drivers to your ears remains unchanged for a wide sweet spot. When the box is on its side, it's common for you to be positioned closer to one driver than the other. Where the two drivers overlap and share sounds around the crossover point, the separate soundwaves combine slightly out of phase from each other. With a 3kHz crossover point, the wavelength will be around 11cm. This means if one driver is heard from a point that1s 5.5cm further away than the other, the entire frequency will disappear due to wave phase cancellation. Incidentally, some manufacturers recess their tweeters into the cabinet for better acoustic alignment with the woofer.
• If you're mixing with horizontally positioned monitors, ensure they're swung around to face you rather than firing straight ahead to maintain consistent driver-to-ear measurements.
• Placing the speaker box horizontally so that the woofer and tweeter are side-by-side risks the frequencies shared by both drivers (around the crossover point) arriving at the listener's ear at different times. This can result in a dip or notch at that frequency due to cancellation of the non phase-aligned waveforms.
• Placing the box vertically so the tweeter sits directly above the woofer encourages the shared frequencies to reach the listener in perfect phase alignment,
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Check the specifications for a set of passive speakers and note the sensitivity rating, for example: 89dB@1wattRMS /m. This means the speaker will output 89dB at a distance of 1m with just a single watt RMS of power. Not bad considering the recommended safe listening level across eight hours is 85dB. To achieve the slightest perceivable increase in volume (+3dB), double the amplifier power as shown below...
1 watt RMS = 89dB/m
2 watts RMS = 92dB/m
4 watts RMS = 95dB/m
8 watts RMS = 98dB/m
16 watts RMS = 101dB/m
32 watts RMS = 104dB/m
64 watts RMS = 107dB/m
128 watts RMS = 110dB/m
256 watts RMS = 113dB/m
As you can see, a whole lot of amplification is required for not a lot of volume
gain. So if 1W will happily output around 90dB, why do we need more?
Music is full of ultra-fast transients that call upon additional power for the
briefest of moments. Bass requires substantial power to accurately pump such
massive volumes of air, and an amp working well within its limitations is likely
to outputting lower distortion, noise, and a cleaner output. When deciding on
an amp, look for the more realistic RMS power.
Between 80-100W RMS per channel (or more) is ideal for nearfield monitoring.
In fact, having up to double the amplifier power than the speakers are rated
is much healthier than underpowering your speakers to the point of clipping
where the amplifier1s maximum output is exceeded which resulting in flattening
off the tops and bottoms of the audio waveform.
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Safe monitoring levels
A quality studio monitoring system will rarely reveal signs of stress and therefore it can be particularly difficult to judge its true loudness. Our ears naturally succumb to loud noise by backing off their sensitivity in what is known as a 'temporary
threshold shift'. This causes us to no longer hear faint sounds.
It's a natural form of compression, if you will, brought on through the fatigue
of the muscles that operate within the inner ear. As the sensitivity diminishes,
we're inevitably tempted to crank up the volume pot to compensate, and only
exacerbate the problem.
Whenever you suffer any hearing numbness or ringing (tinnitus) after being subjected
to a sudden, loud noise, or noise built-up over time, it's a sign that damage
to the ear has occurred. Fortunately, in most cases, our ears are able to repair
themselves and should return to normal following several hours of rest, but
the threat of permanent damage is a lingering concern.
Music, in all its dynamically fluctuating glory, across an eight-hour exposure
period is best kept below an average of 85dBA. An SPL (sound pressure level)
meter (available from an electronics hobby shop) can help you establish safe
monitoring levels. If it saves your sense of hearing for the long term, it may
be the best musical investment you ever make.
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Troubleshooting
If you've got problems with your monitors, before you rush off to pay someone else to fix them, see if it's a problem you can possibly work round yourself...
Poor bass response, exaggerated stereo width, and a hollow mono image?
Check your (passive) speaker connections for correct polarity. If the wires
are reversed on one speaker, its audio phase is thrown180 degrees out from the
other speaker. This forces one speaker cone to push forward, while the other
speaker pulls back. This leads to soundwave cancellation immediately between
the two speakers because one driver is sucking the energy away from the other.
Amazingly enough, this is a common problem amongst audio-related shops that
should really know better.
No sound?
If a speaker stops working, first check its connections, then swap it with
the other speaker to ensure there's not a problem with the rest of your system.
If it's a single driver within the speaker that's cut out, a wire may have worked
loose internally, or the fine wire that makes up the voicecoil may have burnt
out and will require a full driver replacement.
Cracking sound at low volumes?
Swap with another speaker to confirm fault is speaker related. When a speaker
is overdriven, the voicecoil can heat up and distort until its molten insulation
begins to rub against the inner surface of the magnet. A slight, even press
against a suspect woofer driver diaphragm can confirm this if its movement feels
gritty rather than smooth. The only solution is to replace the driver and watch
the levels!
Cleaning
A gentle wipe over the cabinet with a clean cloth slightly dampened with
water should be all that's needed to maintain your monitors' showroom shine.
You can safely wipe over the woofer too if it's plastic and rubber surround.
Be wary of touching the delicate tweeter diaphragm. Steer away from even the
mildest detergents that can leave residues that may build-up and destroy the
surround or other components over time.
Recovering dented dustcaps and diaphragms
Young (and old) children seem to love pushing their fingers into undesirable
places. The woofer dustcap's sole purpose is to keep, you guessed it, dust from
entering the voicecoil and magnet assembly, while a tweeter's domed diaphragm
is purely built for sound. It's possible to suck dents out using a vacuum cleaner,
but be sure to set it to its lowest power setting with the hose vent opened
slightly to reduce the pressure. Why were you touching it anyway?!
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Jargonbuster
Active: Used to describe processing that involves powered electronics, rather than relying on existing power. Active speaker systems use electronic crossovers to divide the full spectrum signal into bass and treble (and sometimes midrange) frequencies prior to powering separate, dedicated drivers. In direct comparison, passive crossovers use frequency-dividing filters that rely on the electrical power of the music signal itself to function.
Hertz (Hz): The term named after Heinrich Rudolf Hertz (1857-1894) used to describe
the pitch or frequency of an oscillating wave by counting the number of cycles
(ie. the completed motion of a waveform crest followed by a trough) that occur
in a second. The healthy human ear can detect soundwaves ranging from around
20Hz to 20,000Hz. Frequencies above 1,000Hz may be abbreviated to kilohertz
(kHz), so 3,000Hz equals 3kHz.
Phase: The position of a waveform at any given point of time in relation to
another point, such as its stationary rest position or another waveform.
Imaging: The ability for a pair of speakers to reproduce the precise position
of a sound within a stereo sound field. Ideally, sounds should appear from pinpoint
locations from imaginary positions between the two speakers. Unfortunately,
due to inaccuracies in the stereo signal path, the recording medium, effects
processing, amplification, crossovers, and even the drivers themselves, the
perfect left and right waveforms can become disfigured to result in a stereo
image that is blurred and indistinct.
Tweeter: The smallest driver in a speaker system. It is capable of vibrating
the air at a very fast pace, from around 2kHz to over 20kHz, that's 20,000 times
per second!
Woofer: The biggest driver in a speaker system and is used to handle frequencies
from anywhere between 20Hz and 3kHz, depending on size and design. Dedicated
bass systems (below 120Hz) are called subwoofers. Incidentally, the term 'subsonic'
actually refers to a speed less than that of sound, so a more correct term to
describe sounds that fall below that of audibility would be infrasonic or the
infra-woofer, but I can't see anyone adopting my cause just yet!
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thanks to FUTURE MUSIC magazine!