For The Love Of…by Distant Fires Burning

Pleasure and honor I mastered this album

a0797296149_10Distant Fires Burning | For The Love Of…
Audiobulb Records (CS/DL)

Review by Darren McClure

The fourth album from this ambient project by Gert De Meester finds the artist stretching his processing skills into interesting territories. His background as the bassist in bands assists in the direction of Distant Fires Burning, focused as it is on the sounds of the Fender Jazz Bass. The bass guitar isn’t a common instrument adapted to experimental drone music, the only band I can think of that has put it to the fore is the British group Rothko, who use the low tones to paint sonic vistas as minimal as their namesake.


De Meester’s use of the bass often veers into these widescreen ambient soundscapes, but also uses the plucked strings to generate rhythms. Computer processing is evident on this recording, with granular textures and delays affecting notes and strums.

This album is…

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Four Noteworthy Compilations for Fall

excellent reviews as always

Sept Comps

top/left: End Family Separation – Fuzzy Panda Recording Company
top/right: Anthology of Electroacoustic Lebanese Music – Unexplained Sounds
btm/left: Earthen – Cold Spring
btm/right: Sichten 1 – Raster Music

Though we do not review compilations but extremely rarely, there is no denying that some which are floating in the most random air space call out for a serious listen. We located four recent, and forthcoming, that are strong contenders for your attention. The four collections below represent a wide ranging international scope in experimental and electronic music – something for everyone. Some of these albums include exclusive material, and otherwise historically unreleased tracks.


We start with the extremely timely set with a big heart, and a conscience, from Washington DC’s Fuzzy Panda Recording Company and their effort, End Family Separation. This collection includes strong efforts of ambient and abstract works by Tag Cloud, BLK w/BEAR, Small Craft

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The 10 Best Studio Monitor Speakers – Essential Buyers Guide 2018 


After your computer and DAW software, studio monitors are arguably the next most important component in any music production environment (with the possible exception of the audio interface.

When levelling up your music production, the importance of a decent set of monitors cannot be underestimated, since their role in accurately transmitting the sound from the outputs of your interface across the room to your ears makes them as vital a link in the chain as the interface itself.

dynaudio acoustics lyd 5 best studio monitors speakers

A good set of monitors should be the audio equivalent of a mirror – what you hear should be a true representation of what’s actually there. The more accurate the response, the more likely it is that, if it sounds good to your ears, itwill sound good to everybody else as well. The question is, what are you looking for in the ideal project / home studio monitor if your main output is electronic or computer-based music?

Monitor selection is a very personal thing – what works for one person may not work at all for another. You’ll be looking for something with a compact footprint that can be positioned fairly close to a wall without compromising the bass response, a deep bass extension for those kicks and sub basses, coupled with a non-flattering, even response across the mids and crisp, detailed highs. You want a monitor to accurately reflect the changes you’re making to your mix, the ultimate aim being to produce an end result that sounds good when played back in any environment, from your car, to the club, to the Bluetooth speaker you’re playing your iPhone through at your mate’s barbecue.


Monitors that exhibit a so-called ‘smiley curve’ response, with exaggerated bass and top-end responses designed to flatter the sound, may sound good while you’re working on them, but the end result is likely to be a lacklustre mix lacking in lows and highs. After all, while you’re working on your project, it’s what you hear with your own ears that steers the decisions you make when it comes to cutting or boosting certain frequencies and levels. So if your monitors are colouring the sound and making it sound more punchy, bottom-heavy and sparkly than it really is, you’re going to end up disappointed when you play your mix elsewhere.

“A good set of monitors should be the audio equivalent of a mirror – what you hear should be a true representation of what’s actually there.”

The difference, then, between studio monitors and the consumer speakers that might come with your hi-fi or computer, is that studio monitors are designed to reproduce audio frequencies as accurately as possible across the audible frequency spectrum, ideally at any volume level. While a totally flat response across the frequency spectrum is the Holy Grail of monitor manufacturers, it’s an inescapable fact that all monitors will colour the sound in some way. It can take a while to get used to the particular characteristics and sound of one set of monitors over another. The longer you use them, the more you get to know the quirks in their response curves and tailor your production decisions accordingly to ensure the best end result.

Achieving such a flat response can come at a cost, however, as it tends to take quality materials, high-end electronic components and well-executed design and engineering to pull this off, which is why a decent pair of monitors can be a bit on the pricey side. However, there are still plenty of options available that are capable of delivering more than adequate performance for those on a tighter budget.


Speakers corner – the evolution of the modern monitor

Before the advent of tape recorders, most commercial recordings were cut live to a master disk, often in a single take and with no editing. Because of this, there was little or no real need for accurate monitoring, as playback scenarios were few and far between. With the advent of tape recording, multitracking, overdubbing and mixing however, there came a heightened requirement for being able to focus in on the detail of what was being recorded and played back after the event.

Up until the late 1960’s, studio monitoring systems tended to consist solely of large, wall-mounted units with huge cones for the low frequencies and concentrically-mounted midrange cones or horns. Each component was fed by a crossover unit that split the incoming signal into the required frequency bands, usually at around 1000-1200 Hertz, so that the low frequencies were sent to the bass cone and everything else to the midrange unit.

munrosonic egg 150 inside diagram

By the 1970’s, however, studio technology had progressed to the point where nearfield monitoring was becoming more widely used, driven in part by the BBC’s need to monitor in the vans they were using for outside broadcasts at the time. The BBC’s LS3/5A and JBL’s 4310 compact nearfield monitors brought about a revolution in studio monitoring. These units were compact enough to be placed on the top of the mixing desk and could be listened to at much closer distances than their wall-mounted cousins. They also allowed the engineer to focus more on the sound coming directly from the speakers, rather than that being reflected off the walls and ceiling. So began a trend that has lasted up to the present day, the technology having progressed over the intervening years to the point where most ‘nearfield’ studio monitors are compact and affordable enough to bring professional-level performance within reach of the budget-conscious home recordist / producer.

Key features to consider

As always when purchasing any piece of recording gear, there are several factors that can influence where your money goes. Here are just a few of them…

Active vs passive

Passive monitors by definition need to be connected to a compatible power amp in order to work – a notable example is the famous Yamaha NS10, the iconic, white-coned nearfield monitor that dominated the professional studio market in the 80’s and 90’s, whose perfect partner was the famous Quad 405 power amp. The one main advantage of this approach was the versatility it afforded to customise your system – changing the amp could have a major effect, either detrimental or beneficial, on the sound of your monitoring setup. It also meant that the speaker units themselves were lighter and therefore easy to set up and move around.

These days, thanks to the progress made in the miniaturisation of electrical circuits and components required in loudspeaker design in general, most nearfield monitors are of the active type; i.e. they don’t require external amplification because all the necessary amps and crossovers required to drive the conesare built into the speaker casing along with the cones themselves. This is more convenient not just because the units are self-contained, but also because you don’t need to shell out an additional arm and a leg for a compatible power amp. Most active monitors also offer some form of adjustable room response compensation features, achieved by the use of internal digital signal processing chips (DSP) that can be a bonus when setting up and adjusting to your unique monitoring environment. You will, of course, require a power lead (and a mains socket of course) for each speaker, but that’s not really an issue in most modern studio environments. So because the vast majority of monitors on the market today are active designs, unless you specifically want to go the passive route, this is largely a decision that has already been sorted for you by the manufacturers.

Reflex / Transmission line

Smaller speakers are always going to struggle more to accurately reproduce bass frequencies than larger units, simply because of the wavelengths of low-frequency soundwaves. One way in which designers have sought to get around this is by building bass ‘ports’ into their speaker casings, an approach known as reflex design. These are specially designed ‘tunnels’ within the casing that exit via a hole – a bit like an exhaust pipe – somewhere in the casing of the speaker unit. If the bass ports are located on the rear of the cabinet however, this can cause issues with positioning of the monitors close to a wall.

A variation is the transmission line design, which gets around this problem by incorporating a sort of long, folded tunnel structure, lined with absorbent acoustic material, into the innards of the cabinet to handle the bass response, often terminating in the front of the speaker. This can significantly reduce low frequency distortion and deliver greater bass extension and loudness than a ported or sealed design of a similar size, meaning that you don’t have to crank the volume to elicit the correct bass response – the response is balanced whatever the volume level.

Frequency response

A good set of monitors with a so-called flat frequency response should theoretically give you the ability to reveal detail and make mix decisions such as subtle changes to EQ and compression more easily. The spec sheets will usually quote a range of frequencies that the speakers are able to reproduce, which will usually exceed those frequencies audible by the human ear, normally around 20Hz – 20kHz. Many monitors also offer options for tuning the frequency response of the monitors to match the acoustics of your studio space – this is done by the addition of digital signal processing (DSP) circuitry, which in turn requires internal analogue-digital-analogue (ADA) converters, the quality of which can have a big effect on the overall sound.

Woofers and tweetersyamaha hs7 front

One thing that all the monitors on our list have in common is that they all have multiple drivers. In other words, each speaker unit has at least two cones, one large one (or ‘woofer’) to handle the bass and lower-mid frequencies, and a small one (or ‘tweeter’) to take care of the upper mids and highs. In an active monitor, the incoming audio signal is split into two frequency bands – low and high – by a component called a ‘crossover’. The point at which the split occurs is known as the crossover frequency, and this varies greatly from monitor to monitor, anywhere from between 800Hz-2kHz. The low-frequency signal is then sent to the woofer and the high-frequency signal to the tweeter, each via its own power amp. The fact that each driver has its own amplifier is often what contributes to the surprising weight of an active monitor!

Ear Fatigue

Mix sessions can last for hours at a a time, so one very important thing to consider when choosing studio monitors is so-called ‘ear fatigue’, which essentially translates into how long you’re able to listen to them at a decent working volume before your ears start to get tired. A revealing sound is all well and good, but if this results in a sound that’s too harsh and unforgiving in the midrange, say, your monitors will be challenging to listen to for any great length of time. Because everyone’s ears are different though, one man’s sonic silk purse may be another’s sow’s ear, so the fatigue factor is something very subjective that’s going to be very hard to gauge without a thorough test of the speakers you have in mind. So if at all possible, try to get hold of a pair to ‘test drive’ for a while, so that any long-term compatibility issues between the speakers and your ears can be assessed. You’ll find it pays dividends in the long run, as you’re more likely to end up with a monitoring system that’s right for you, in that it’s effective and comfortable to work with for long periods.

Sub or no sub?

While perfectly capable as functioning as a stereo pair with good performance across the entire frequency spectrum, many monitors currently on the market are available as so-called 2.1 systems, consisting of two nearfield, desk-mounted stereo speakers to handle the upper and mid-frequency ranges and a separate, single sub-woofer, usually placed somewhere below the desk, to handle low frequencies. Because the sub is usually a large unit containing a large diameter cone, it’s often able to extend the low-frequency response of the system down to around 20Hz or so – the kind of bass you feel in your chest rather than hear. While this may sound great and reveal a lot more low-end detail than a solitary pair of desk-mounted nearfields, in small rooms the extended low-frequencies can create noticeable peaks and troughs in the bass response. For that reason, we’ve predominantly focussed our list on stereo systems, but highlighted whether or not a compatible sub-woofer is available from the same manufacturer.

10 Best Studio Monitors – Buyers Guide List 2017

So now onto the list of what we consider to be ten of the best studio monitors on the market today. Like all of the lists we feature on GTPS, we don’t believe it’s particularly useful to say that one device we’ve included is categorically better than another, and neither is this a categorical list of the ten best monitors available. Monitors are a very personal choice, and in this roundup of some of the market leaders we’re simply highlighting some of the best options currently available for various budgets and studio setups – everyone will, of course, always have their own personal favourites.

focal shape 65 best studio monitors speakers

1. Focal Shape 65

Focal’s Shape range of active monitors replaces their acclaimed CMS series, and is available in three sizes – the 40, 50 and 65, in order from smallest to largest. The largest of the range, the 65 is designed with a sideways-facing, 6.5-inch passive radiator cone – in essence an unpowered speaker cone that reacts to the sound pressure inside the cabinet. This dismisses the need for a bass port, the idea being that this makes it easier to position the monitors with their backs next to a wall, a commonly-required configuration compromise in a lot of small studios. There’s also low tweeter directivity, meaning that it’s not so essential to have the speakers placed at one particular optimal angle for best results. Sound-wise, with a variable crossover set at 160Hz and adjustable low and high frequency settings to optimise the response to their surroundings, the Shape 65 offers great performance for the price, and as a bonus the cabinets include built-in threaded holes so the speakers can be wall or ceiling-mounted using an optional accessory mounting kit.

Focal Shape 65 – Specs:

Design: 2-way active, with side-mounted, 6.5″ double passive radiator
Woofer: 6.5-inch Flax sandwich cone
Tweeter: 1-inch ‘M’ profile Aluminum-Magnesium dome
Frequency response: 40Hz – 35kHz
Wattage: MF/LF 80W, class AB, HF 25W, class AB
Dimensions: 14 x 8.6 x 11.2″ (355 x 218 x 285mm)
Weight: 18.7lb (8.5kg)
Compatible Sub? No



yamaha hs7 best studio monitors speakers

2. Yamaha HS7

Yamaha all but cornered the nearfield market back in the 80’s and 90’s with its legendary NS10M monitors. In a matt black wooden casing with a distinctive white bass cone, they came to be the monitors of choice for a huge percentage of studios worldwide, thanks to their uncompromisingly accurate response. The NS10M was discontinued by Yamaha in 2001, but its DNA lives on in the HS series, of which the HS7 is the mid-sized variant. Sticking with that white cone, the HS7 builds on its heritage, adding an active reflex design with a rear-facing bass port, and includes a pair of switches to boost or attenuate high and low frequencies by a dB or two, so as to tailor the response to the room you’re using them in. At a very reasonable price point designed to catch the attention of home studio owners (hence the ‘HS’ moniker) who like the idea of some white-coned Yammies on their desk, the HS7’s deliver a lot of bang for the buck.

Yamaha HS7 – Specs:

Design: 2-way active, bass reflex (rear ported)
Woofer: 6.5-inch cone
Tweeter: 1-inch dome
Frequency response: 43Hz – 30kHz
Wattage: LF 60W, HF 35W
Dimensions: 8.3 x 13.1 x 11.2″ (210 x 332 x 284 mm)
Weight: 18lb (8.2kg)
Compatible Sub? Yes – Yamaha HS8S


dynaudio acoustics lyd 5 best studio monitors speakers

3. Dynaudio Acoustics LYD 5

Dynaudio Acoustics are responsible for some legendary high-end speaker designs, including the acclaimed BM15A. For those looking to acquire a taste of Dynaudio quality in a more affordable package, there’s the LYD series, made up of the LYD 5 and its bigger brother, the LYD 8. Both available either in a resplendent white or standard black finish, the LYD 5 (the 5 denotes its 5-inch woofer cone size) packs quite a punch, belying its diminutive size with a beefy sound achieved in part by a rear-facing bass port. Any issues this may cause by having the speakers positioned against a wall can be offset by the ‘Position’ mode, while the Sound Balance control is an attempt to compensate for overly bright or dead rooms. There’s also a Bass Extension control to tailor the low end to your working environment. All in all then, a smart-looking, versatile monitor from a pedigree brand at a decent price – what’s not to like?

Dynaudio Acoustics LYD 5 – Specs:

Design: 2-way active
Woofer: 5-inch MSP cone
Tweeter: 28mm soft dome
Frequency response: 50Hz – 22kHz
Wattage: LF 50W, HF 50W
Dimensions: (170 x 260 x 211mm)
Weight: 12.5lb (5.7kg)
Compatible Sub? No


MunroSonic EGG 150 best studio monitors speakers

4. MunroSonic Egg 150 Monitoring System

Up there as contenders for most unusual speaker design, the MunroSonic Egg range of monitoring systems lives up to its name, in that it consists of a pair of uniquely egg-shaped passive speakers driven by a dedicated external analogue power amp that comes as part of the package, along with two high-quality, 3m cables to connect it all up. There is, naturally, a good reason for this departure from the norm – the company’s literature states that ‘the traditional wooden box has been replaced by a scientifically proven, curved enclosure that virtually eliminates diffraction and resonant effects that distort and smear the original sound’.

Combined with the free-standing amp is a control unit that features source-select inputs, active analogue crossovers, LF and HF trim pot equalisation for room and location set-up compensation and a re-defined mid-band control to emulate the mid-range response of both Hi-Fi and NS10 type speakers. The egg-shaped driver units sit on purpose built ‘nests’ that allow flexible positioning in all directions, aided by blue guide LED’s above each tweeter to help you locate the ‘sweet spot’ when setting up. The nests also allow sufficient room for the downward-facing bass port to expel the necessary air.

MunroSonic Egg 150 Monitoring System – Specs:

Design: 2-way passve, with downward-facing bass port and external analogue power amp
Woofer: 165mm Polypropylene Cone
Tweeter: 25mm dome
Frequency response: 45Hz – 20kHz
Wattage: MF/LF 50W HF 50W
Compatible Sub? No


Eve Audio SC204 best studio monitors speakers

5. Eve Audio SC204

The smallest monitor in Eve Audio’s product line, the SC204 combines a 4-inch honeyomb-structured driver with an AMT (Air Motion Transformer) ribbon tweeter of their own proprietary design that’s used throughout the whole Eve Audio range. RCA and XLR analogue inputs feed directly into Burr-Brown digital converters, and the signal is processed digitally before reaching the PWM Class-D digital amps that feed the drivers. Sophisticated stuff.
A unique addition is the front-mounted volume control – the sort of thing you’d normally expect to find on budget desktop computer monitors, except this knob controls a host of other features, including standby mode, filter selections, speaker symmetries, phase tweakings and even LED display intensity.
There’s a large rear rectangular port with smooth edges to extend the low-end response and minimise bass distortion, while the ribbon tweeter provides a crisp and detailed high end that’s bright without being harsh. All in all, these diminutive and affordable speakers punch way above their weight.

Eve Audio SC204 – Specs:

Design: 2-way active
Woofer: 4-inch cone
Tweeter: AMT RS1
Frequency response: 64Hz – 21kHz
Wattage: MF/LF 50W, HF 50W
Dimensions: 5.7 x 9.0 x 7.7″ (145 x 230 x 195mm)
Weight: 8.4lb (3.8kg)
Compatible Sub? Yes – Eve Audio TS107

PMC twotwo 6 pair best studio monitors speakers

6. PMC twotwo 6

British manufacturer PMC aren’t just known for the excellent quality of their speakers, but also for the transmission line principle that lies at the core of their design philosophy. Their version is named ATL (Advanced Transmission Line) and, in the twotwo range, places the bass cone near one end of a long tunnel within the casing, lined with material that absorbs the high and low-mid frequencies radiating from the rear of the cone. The lowest frequencies emerge from the end of the tunnel via a port at the front that effectively acts as a second bass driver.
PMC twotwo 6 rearThe twotwo range consists of three models – numbered 5, 6 and 8 after the size of the bass driver in each – and what sets them apart from their predecessors is the addition of DSP-based EQ, crossover and driver response adjustment, making them the only PMC nearfield monitors to feature EQ compensation apart from the range-topping AML series.
Throw in an LCD screen on the back to let you see what’s happening while you adjust the EQ settings, digital and analogue inputs and the ability to link monitors together with a Cat 5 ethernet cable that supplies duplicate EQ settings and audio to a second monitor and you have a class-leading package that can’t be ignored.

PMC twotwo 6 – Specs:

Design: 2-way active
Woofer: 6.5-inch cone
Tweeter: 27mm soft dome
Frequency response: 40Hz – 25kHz
Wattage: MF/LF 150W, class D, HF 50W, class D
Dimensions: 16” x 7.6 x 14.3” (406 x 194 x 364)
Weight: 18.5lb (8.4kg)
Compatible Sub? Yes – twotwo sub1 & twotwo sub2

adam a7x best studio monitors speakers

7. Adam A7x

Another brand that favours the ribbon tweeter design, Adam Audio’s version is known as the X-ART (short for eXtended Accelerating Ribbon Technology), so-called because of a claimed flat frequency response that extends up to an amazing 50kHz. Why would you want to go that far beyond the range of human hearing you might ask? Good question, but the idea behind the design is to achieve detailed, uncompressed highs and upper mids without being tiring over long listening periods, and it has to be said that the A7x reproduces transients with incredible clarity, creating an extremely precise sound. Universally accepted as one of the best monitors on the market, the A7x improves on the original A7 model in almost every aspect, from that updated tweeter and new amps to a redesigned bass and midrange driver. There’s an additional bass port on the front, and round the back you have controls for tweeter level and high and low shelf filters.

Adam A7x – Specs:

Design: 2-way active
Woofer: 7-inch Carbon/Rohacell/Glass Fibre cone
Tweeter: X-ART (Equivalent diameter 2”)
Frequency response: 42Hz – 50kHz
Wattage: MF/LF 100W PWM, HF 50W class AB
Dimensions: 13.5 x 8 x 11″ (337 x 201 x 280mm)
Weight: 20.3 lb (9,2 kg)
Compatible Sub? Adam Audio Sub 10 Mk 2


ATC SCM25A best studio monitors speakers


The SCM25A is ATC’s first ever compact three-way active studio monitor, but their previously available large / midfield designs have garnered accolades from every corner of the the music-making world, building a reputation for precise and detailed reproduction across the whole frequency spectrum at any volume level. The SCM25A continues the family tradition, and at a whopping 30kg per cabinet, you know you’re dealing with a quality product as soon as you (and three burly mates) lift it from the box!
Design-wise, it’s one of the few on the market to offer a choice between two different approaches to loudspeaker cabinet design. There’s a large port vent on the side panel adjacent to the woofer that can be plugged with a dense foam bung, transforming the speaker from a reflex design into a sealed cabinet design if needed.
With a clean and natural bottom end, the mid-range clarity shines through thanks to that renowned ATC soft-dome mid-range driver, and the highs are honest and detailed, allowing for easy analysis and precise adjustment of mixes without being overly fatiguing or clinical. There’s no escaping the fact that the SCM25A is one of the more expensive monitors on the market for its size, but the accuracy and quality of its performance really does justify the extra expense.

ATC SCM25A – Specs:

Design: 3-way active, with side-mounted bass port
Woofer: hand-built 7˝/164mm short-coil carbon-paper cone
Mid: hand-built 3˝/75mm soft-dome
Tweeter: 25mm neodymium soft-dome
Frequency response: 47Hz – 22kHz
Wattage: LF 150W, MF 60W, HF 25W
Dimensions: 10.4 x 16.9 x 114.5” (264 x 430x 369mm)
Weight: 66lb (30kg)
Compatible Sub? No

Barefoot Sound MicroMain 45 best studio monitors speakers

9. Barefoot Sound MicroMain 45

For those lucky souls where budget is no object, Barefoot Sound’s range of high-end monitors have to be high on the list of considerations. Barefoot are kind of the monitor equivalent of the Rolls Royce, but nestling down near the more affordable end of their product spectrum (although still not exactly cheap at £6k a pair) the MicroMain 45’s are a stripped-down version of the flagship MiniMain12’s. These three-way active monitors are probably equally suited to midfield and nearfield use owing to their size – and the 8-inch bass driver, twin 2.5-inch midrange cones and 1-inch ring radiator tweeter certainly get the job done.
Barefoot’s approach to monitor speaker design has always been to deliver outstanding sound quality, wide dynamic range and bass extension at every stage of the audio production process, and the MM45 sums up that philosophy in a nutshell. Now though, Barefoot speakers offer a range of preset EQ options in the shape of MEME voice emulation technology, meaning that, with just the flick of a switch, you can get your monitors to emulate the sound of sweetened ‘Hi-Fi’ speakers or, with the ‘Old School’ setting, the classic Yamaha NS10M. Amazing.

Barefoot Sound MicroMain 45 – Specs:

Design: 3-way active
Woofer: 8-inch aluminium cone
Mid: 2 x 2.5-inch aluminium cones
Tweeter: 1-inch ring radiator
Frequency response: 40Hz – 45kHz
Wattage: LF 250W, MF 180W, HF 180W
Dimensions: 11 x 15.5 x 11” (279 x 394 x 279mm)
Weight: 37.5lb (17kg)
Compatible Sub? Barefoot MicroSub
Available from: KMR Audio +44 (0) 20 8445 2446

Genelec M040 best studio monitors speakers

10. Genelec M040

Finnish company Genelec have a fine studio monitor pedigree, stretching back to the early 80’s when their large, wall-mounted designs were the speaker of choice for a huge number of professional recording studios worldwide. Still going strong, it’s now possible to get the Genelec name in your own studio for a fraction of the price of their hefty ancestors with the brilliant, small and mighty M040 (and even smaller M030).
Genelec have thrown an impressive number of acronyms into the design of the M040. There’s the Directivity Control Waveguide (DCW™), which ensures flatness of the overall frequency response, the Laminar Integrated Port (LIP™) moulded into the cabinet to aid faithful bass reproduction, Intelligent Signal Sensing (ISS) that automatically enters standby mode if no signal is detected for 30 minutes, and the Natural Composite Enclosure (NCE™), which refers to the environmentally-friendly makeup of the cabinet itself. Elsewhere, you’ll find simple-to-use room response compensation controls, in the form of dip switches, mounted on the rear of the cabinet, to tailor the response to any acoustic environment.

Genelec M040 – Specs:

Design: 2-way active, ported
Woofer: 6.5-inch cone
Tweeter: 1-inch metal dome
Frequency response: 44Hz – 21kHz
Wattage: MF/LF 80W, class D, HF 50W, class D
Dimensions: 13.3 x 9.3 x 9″ (337 x 235 x 229mm)
Weight: 15.4lb (7.4kg)
Compatible Sub? Genelec 7040A


How to create wider sounding mixes



  • Stereo width is an illusion that we either capture in a recording or create within a mix
  • To make wide sounding mixes, capture width when you record or create width inside your mix

What is stereo width?

Stereo width is an illusion of the left-to-right dimensions of the sound field (i.e. sound stage or panorama) in a recording, perceived by a listener.

Imagine yourself standing on a sidewalk listening to busy downtown traffic. You are hearing cars constantly rushing back and forth, left to right and right to left. This “field of sound” you are hearing is quite wide. Because we hear binaurally (another future topic), you’re hearing a three-dimensional world of sounds from left to right, front to back, and even up and down. But let’s simplify for a moment and focus on the fact that cars are coming from one direction or the other and we largely are hearing that back and forth movement across the sound field because we have two ears and we can sense where each of the sounds are coming from.


Imagine that you capture a few minutes of what you are hearing on the sidewalk with a stereo audio recorder, go back to a quiet place to listen to what you recorded. Setting aside mic quality, technique and other factors for the moment, you will hear a rough representation of what you experienced on the sidewalk because you will have captured the scene in stereo. Your recording will reveal how the traffic moves from left to right and right to left across the sound field captured by the recorder. Show that recording to someone else, who wasn’t there when you recorded it, and they will be able to imagine to a degree what it was like standing on the sidewalk because the recording has enough information in it to recreate a panorama where they can hear the traffic moving back and forth, similar to how you experienced it.


Now imagine if you use a mono (i.e. monophonic or monaural) voice recorder to capture what you heard on the sidewalk, instead. You will hear a much different representation, because your recording is missing a massive amount of information. A mono voice recorder only has one microphone (i.e. one ear) so it cannot capture stereo information. Your recording will reveal a mush of engine noise, maybe the odd vehicle horn, but everything is crammed together and it sounds like varying levels of noise. Show that recording to someone else, who wasn’t there when you recorded it, and they will find it challenging to imagine being on that sidewalk. The resulting panorama from the mono recording simply does not contain enough information to do that.


Stereo width therefore depends upon stereo information being captured and presented back to the listener. When you think about that traffic scene, what makes the scene stereo? It is all about differences between what you hear from one side of the scene to the other. Stereo width is that simple. You create stereo width by creating or enhancing the difference between the left and the right sides of the sound field that you are presenting back to your listeners.


One more example before we move on. Picture yourself sitting in front of a stage with two musicians standing side by side, a few feet apart. Forgetting the room acoustics and other factors for a moment, you focus on the two musicians, and from your vantage point, they are relatively close together. You therefore perceive them both coming largely from the center of the sound field. Now imagine that each musician walks to the opposite end of the stage. Now you perceive them as being distinctly separate where you hear one largely off to the left, and the other way off to the right. The musicians have just “widened the panorama” presented to you simply by separating themselves further, relative to you. Now imagine replacing yourself with a stereo array of microphones. See how microphone placement relative to the subject(s) can make a big difference in this situation? You can drastically alter what you present to a listener – which impacts their perception of width –  simply by changing the placement of the subject(s) relative to the microphones.


This is how stereo width within a mix works. When you create a mix, you are building a sound field with layers of sound. Stereo width is an illusion of the left-to-right dimensions of the sound field. The more you can create differences between what you present on the left versus what you present on the right, the wider your mix will seem to a listener.


How to Create wide Stereo mixes

Bottom line:

  • You can only enhance stereo width if it exists to some degree in your mix
  • Make your mix sound wider using differences in left vs right gain, time, pitch, tone or polarity

Mono sound is single-channel sound, and stereo (stereophonic) uses two channels. If you reproduce monophonic sound over two loudspeakers, you will hear the sound coming from a narrow area between the two speakers (in the center). This is called a phantom image because it can be quite palpable to a listener. It can sound almost as real and distinct as a sound that comes from just the left or the right speaker on its own. The phantom image sounds focused (narrow) because each loudspeaker is presenting the same information, and there is nothing that our brain can use to establish a difference between sounds coming from the left versus the right loudspeaker.

Does your mix sound too narrow? Width is a key component of modern mixes because most music is mixed in stereo. Many listeners listen in stereo (e.g. with earbuds). We perceive a mix as being too “narrow” when we sense that the music is coming from the center when it seems that it should not. That is to say, a center-centric mix can be perfectly fine if it sounds natural, but you know you have a problem when you or your listeners sense that things are too narrow, where the mix sounds too confined and the panorama is not expansive or convincing enough to draw them in.


You can make your mixes sound wider by maximizing the difference between what you present in the left versus the right channel. The “difference” can be in time (arrangement of notes or elements, and includes phase as well), gain, pitch (arrangement of notes and/or tuning), tone, or polarity.


Let’s look at gain first. Panning is used to adjust the “gain difference” between left and right. Pan hard left, and you have full gain in the left, and no gain in the right. This is the easiest way to maximize left-to-right difference. Take two different sounds in your mix, pan one hard left, the other hard right, and you have maximum separation, maximum difference between left and right. The more you “pull them in” by panning closer to center, the more you reduce difference information, bringing them closer and closer to perceived center.


You can use timing to create left-to-right difference by delaying one channel relative to the other. You can take a mono string section track in the left channel, add a 25ms delayed version to the right channel, and with some gain adjustments you have created a very wide, very stereo sounding string track. If you experiment with this, a 15ms delay is a good starting point. As you pull it towards 0ms, you will hear the delay getting harder to discern and other things start to happen as the sounds fuse together, which can be good or bad. I recommend listening in stereo for the effect you want but then listen in mono to make sure you haven’t created a mono-compatibility issue in the process. If you hear the tone change drastically where it is clear in stereo but muffled in mono, that is an issue. Also listen to the attack portions of notes. Sharper attacks (guitars, vocals etc.) will only tolerate shorter delays because if the delay is too long you’ll get a flam effect which can be distracting to a listener. Sounds with softer or slower attacks will often work well with over 20ms of delay.


You can create tonal differences between left and right by equalizing each channel differently. A common technique with mono piano tracks is to EQ the left to have more bass and lower mids and the right to have more upper mids and treble so that there is a sense of movement from left to right as the player moves up and down the scale.


You can use tuning differences between left and right to create width. This is partly how chorus effects work. Take a mono track in the left, and then add a detuned version in the right channel to create width.


A mix engineer uses all of these methods different ways to either create or enhance stereo width. The ideas are almost endless. You can keep it simple or get really creative depending upon the situation. For example, putting reverb on one side of the field but not the other is often used on guitar tracks. Another option is putting vibrato on one channel and not the other, or mono chorus on one channel or the other. Hammond’s popular “Leslie” effect, captured in stereo, works effectively because it uses the Doppler effect which essentially creates timing and tuning changes between what’s captured in the left and right microphone.




This is absolutely key. To perceive width, we need a reference point. Making every sound in your mix super-wide will not necessarily lead to an engaging or musical mix. Certain anchor points in a mix that are kept central, such as a lead vocal, a bass guitar, a kick drum etc. are essential to creating a wide mix because we will judge the width of the panorama relative to those central elements! Never underestimate the power that mono, central elements have in enhancing the perceived width of your mix by how they create contrast.



Remember: Width must exist within your mix before you can enhance it in a meaningful way.

Please note that I consider these next tips to be shortcuts. If your goal is to truly understand stereo width to create musically wide-sounding mixes, these shortcuts will mainly help you by allowing you to easily experiment. They are a time saver that can be useful in some mix situations after you have learned the concepts above.

Stereo wideners, ambience retrieval, or other forms of spatial enhancers have become very popular. They employ different combinations of the techniques above to broaden and deepen the sound stage. They are a quick fix. I caution against using them across a whole mix because that can greatly distort the sound field you have spent so much effort to build if you are not careful. I have lumped ambience retrieval in with stereo widening but they are often different processes, so research any processor first to understand what it will do to the sound. Use these products sparingly, and consider only using them on certain elements within your mix that would benefit from widening or ambience retrieval effects.


“FreeHaas” and “FreeOutsider” are free plugins offered by VescoFX. They are well worth experimenting with. I recommend using them sparingly, on one or two elements in your mix at the most. FreeHaas adds a Haas Delay (see Haas effect) which you can adjust to your liking. FreeOutsider is a much more obvious beyond-the-speakers effect. You can combine them with tone and gain adjustments to maximize the difference between left and right, thereby maximizing the width of certain elements of your mix.


Mid/Side (a.k.a. sum and difference) processors are widely misunderstood and are often thought of for stereo widening, but I caution that merely adjusting the ratio of Mid to Side will not work that well unless the source has a lot of difference information in it already. This is why many spatial enhancers include a Mid/Side adjustment control to allow you to adjust how much of the widening effect you hear, after the spatial processor has done the initial work of creating more difference information in the first place. If your mix has a lot of difference information, but does not sound wide enough to you, Mid/Side adjustments probably cannot help. If your source is completely mono, Mid/Side will do absolutely nothing and if it is close to mono, then boosting the side doesn’t do much either. I expand upon this much further in Part 1 and Part 2 of my Mid/Side articles.

Widening your mix beyond the loudspeakers

While much of the effort in a mix is in creating a wide field within the loudspeakers, you can create the illusion of going past them as well, to further extremes. There is risk to this. If you overdo it, it will lower the quality of your mix.


Take any sound and add it to two channels in your workstation. Pan one channel hard left, pan the other hard right. Press play and you’ll hear the sound coming from the exact center. Flip the polarity of one of the channels and listen to how the sound changes. It will sound strange – very wide depending upon your listening environment. The reason is because yet again, you have created another difference between left and right – this time it is a difference in polarity! This is a completely unnatural sound and for most people it is fatiguing after a while. But it is also completely incompatible in mono. Collapse your mix bus to mono, and everything disappears because the left and right channels cancel each other out. While absolute polarity differences like this are unnatural, they can also be useful.  Experiment with them, particularly on occasional ambient sounds or even reverb effects, and with the other techniques that I mentioned above to help push the sounds a little further outside the speakers.


By combining various techniques, we can recreate more complex combinations of binaural cues that our brain uses to determine where a sound is located, allowing you to present a three-dimensional sound stage to your listeners. This is worthy of a separate article but here is a simple example. When we hear a sound coming from our left, our left ear hears it slightly louder, slightly sooner, and slightly brighter than our right ear does. Our brain uses that combination of differences in loudness, timing and tone to perceive where that sound is located within a three-dimensional sound field. Our brain analyzes that, along with the information about ambient environment (early reflections and the left-vs-right loudness, timing and tonal differences of sounds in the environment) to map out how large a space is and where the sound is located within it. The more location information we can present to our brain, the more we can create the perception of location be it within or beyond the loudspeakers.

It is not easy to recreate these types of cues over loudspeakers because of bleed (crosstalk). If you sit between any two loudspeakers, you will hear both loudspeakers with both ears to varying degrees whereas with headphones, each ear only hears one speaker. It is therefore much easier to recreate these complex cues when you are mixing for playback over headphones because you won’t have to overcome crosstalk. Creating these cues for headphones can be as easy as using a binaural microphone array to capture the source, and then working to preserve those cues throughout the production process. You can also add the cues after the fact with an HRTF (Head Related Transfer Function) processor.

When mixing for playback over loudspeakers, the only way to effectively recreate these cues is to find ways to minimize the perceived crosstalk. There are crosstalk cancellation technologies available such as QSoundAmbiophonics and BACCH that accomplish this to various degrees. I will plan expand more on this in a future article. These types of filters are well suited to situations where your listeners are sitting in a fixed position, in front of two loudspeakers which makes them well suited to gaming applications or small portable devices. BACCH might be the most promising from a pure performance perspective but unfortunately at the time of this writing it is priced outside of the range of most users and productions.

Outside of crosstalk, there are other challenges to reproducing a three-dimensional sound field over two loudspeakers. There is no way to know exactly where your listener will be placed, but we can be certain that most will not stay in the same position. We also cannot know what performance level their listening room and sound system is capable of, whether it is mono, stereo or surround. We can’t even guess at how it will be configured (EQ or tone, speaker placement etc.). Therefore, if you plan to build complex binaural cues into your mixes, carefully consider all of the possible end points (ear buds, loudspeakers, television, movie theater etc.), and how your music will be consumed (while gaming, while driving, while travelling, while housecleaning). In many cases, building these complex positional cues into your mix is only worth the effort if you can do so without reducing the sound quality for other listeners in the process. This is why checking for mono compatibility and loudspeaker vs headphone compatibility is so important.

Essential ambient production tips

Article from

Inspired by the likes of Philip Glass and Brian Eno, ambient music is as much about creating mood as it is creating melody.

Fortunately, computer users can now call upon an arsenal of ambient-friendly production tools – MusicRadar is here to explain what they are and how to use them.

mixing image

1. If all the soft sounds and smooth vibes get a little too much, try some juxtaposition. Ambient heroes The Orb are fond of this technique, and whether it’s a squealing guitar, devastating synth hit or ridiculous vocal sample, they’re not afraid to toss something a little unusual into the mix.

2. Getting off-the-wall sounds doesn’t have to involve spending hundreds on sample downloads and libraries – there are plenty of interesting sounds happening all around us all the time. If you’ve got a mic and a laptop – or any portable recorder – take a field trip and record some of nature’s bounty. Running water’s always good for a laugh, but remember: your equipment should stay dry, even if you don’t…

3. Second-hand record shops are great places to find sounds. You may even find that your local charity shop has an untapped collection of oddities just waiting to be snapped up by the enterprising samplist. From records featuring nothing but steam engine noises to children’s story albums, there’s an abundance of weirdness out there for the taking.

4. Samples are a constant source of inspiration, but it’s easy to discount one because it doesn’t fit the feel of your track when you first try it. If you’re short on fresh ideas, try running short bursts of a sample through a delay effect. Using this method, it’s possible to come up with some great abstract noises that sound nothing like the original source material.

5. If your tracks are jam-packed full of synthetic-sounding virtual instrument patches and everything’s starting to sound too ‘computery’, consider bringing in some natural sounds or using a few real instrument parts. Even if they’re from ROMplers, it should help take some of the unnatural edge off.

6. Recordings of natural sounds such as rainfall, waves, wind and fire are great for filling out a mix because they’re basically noise, and as such, they have a wide range of frequencies. They shouldn’t be too loud or they’ll overpower the mix, but use them with care and they can be extremely useful.

7. Noise is a useful synthesis tool – if your synth features a noise oscillator, you can use it with a fast-attack amplitude envelope to create your own percussion sounds. This sounds artificial, but in a lo-fi way, and works especially well when teamed with a high-quality reverb.

8. If you’re using long, sustained sounds, such as pads, your mix can lack movement if these elements are too static. By subtly altering tuning, pulse width or filter cutoff over time, you can create more organic sounds that will enhance the mix rather than make it sound lifeless.

“Recordings of natural sounds such as rainfall, waves, wind and fire are great for filling out a mix because they’re basically noise”

9. If you’ve got a sample that you want to play for longer than its duration, you have two basic options: you could timestretch it, which will most likely introduce unwanted audio artifacts, or loop it. Crossfade looping is the best way to get seamless loops, but if this isn’t possible, you can recreate the effect yourself by fading between two audio tracks in your mixer.

10. To make a pad sound particularly evocative, try modulating the filter cutoff with a shallow LFO as well as a big, sweeping envelope. This will give the sound a great deal of movement and works superbly when combined with a delay effect.

11. When working with vocals, you can have a lot of fun with pitchshifting. When pitching vocals around, it helps to use a plug-in with a formant control – this helps vocals retain their characteristics or, conversely, can be used to alter them radically. Check out Smoky Joe, a lo-fi formant processor.

12. With modern audio sequencers, it’s easier than ever to cut up vocals and other rhythmic sounds in order to fit them in with the groove of your track. When cutting sounds up in your sequencer, remember to zoom in to make sure you’re cutting the file at a point where the amplitude is zero – otherwise known as a ‘zero crossing’.

13. When deploying your newly-sliced rhythmic samples, it’s not always best to have your sequencer’s snap control active. You might find that pulling samples forwards along the track a little makes them fit in better with the rest of the groove, and having the snap control turned off also makes programming human-sounding rhythms easier.

14. Silky bass guitar tones are a common sound in ambient dub, but if you don’t have a real bass guitar to hand, you’ll have some trouble getting the same smooth sound. Bass ROMplers such as Spectrasonics Trilogy and Bornemark’s Broomstick Bass are your best bets for recreating this kind of thing.

15. Whether you’re composing in stereo or surround, it’s important to use the available panoramic space properly if you want to create a sense of size. If your track has drums, you’ll probably want to pan these around the centre, but with synths and effects you can afford to use the space more creatively, so try panning them around.

16. Most DAWs have simple pan controls that only enable you to pick one position in the stereo panorama. If you’re looking for slightly more control, a stereo imaging plug-in such as mda Image or BetaBugs Moneo can be used to control the position and filter setting of each channel or tweak them as a mid/side pair, respectively.

17. To add a natural stereo panorama to mono samples, you could do a lot worse than give Voxengo Stereo Touch a try. This effect uses a delay algorithm to create a convincing stereo effect that’s guaranteed to revitalise any dodgy old mono sounds you might have lying around.


18. Reverb is one of the most important tools you have for creating a sense of space, so if you’re making ambient music, it pays to take the time to get it as sweet as you can. A good start is to use a high quality reverb – Ambience isn’t just free, it’s one of the best reverb plug-ins out there.

19. It can be tempting to just stick reverb on a few tracks and leave it at that, but that wouldn’t be using this powerful effect to its full potential. Using high damping values, large room sizes and long reverb times will create a big sound that, when combined with judicious EQ, can create a ‘far away’ kind of effect.

20. When using reverbs, if you want to create a softer, more ethereal effect, use less of the dry signal in the output. You can do this by turning the wet/dry ratio up, or, if you’re using a send effect, by setting it to pre-fader and turning the source channel’s main volume level down.

21. If you’d rather have a brighter, closer effect, then make the reverb’s damping less severe, reduce the room size and turn down the delay time. This works especially well in conjunction with stereo enhancer effects such as the Voxengo Stereo Touch plug-in.

22. Many interesting effects can be created by rendering out reverb and delay tails minus the original dry sound, then applying creative processing to the tail. Filters work particularly well for this kind of thing and, once processed, the new sound can be played back alongside the original version, or replace it altogether.

23. Finally, when programming synth patches, don’t discount the creative potential of your instrument’s reverb section. With a long, lush reverb, even the smallest synth squelches or blips can be turned into pleasingly tonal atmospheric effects. Of course, if your synth effects truly suck, you can always use a separate reverb or delay plug-in instead to create the same effects.


24. Delay is a pretty common effect in atmospheric music like ambient, but for ambient dub, a full-on feedback delay, such as Ohm Force’s excellent OhmBoyz effect, is just the thing.


25. Dynamic use of feedback delay is useful for creating long, evolving rhythmic effects. By automating the feedback control on a delay plug-in, you can build to a crescendo or create weird rhythmic effects.

26. Getting that distinctive morphing dub delay effect can be done by adding either a filter or distortion component to the feedback loop – easily done in OhmBoyz, as it has both. If you’re using a delay effect in Reaktor or another modular environment, you can add these elements yourself, though it’s advisable to put a level limiter after them to ensure the feedback doesn’t get out of control.

27. Delay effects work well before a reverb, though too much of either will swamp the mix. However, it’s possible to tame these effects with automation – set the reverb’s wet level to 0%, automating it so that it comes up as the end of the delay tail is playing. This way, you’ll be able to use both the delay and the reverb, without having too much of either going on at once. As an advanced alternative, you could use sidechain compression to duck the start of the reverb (using the source signal as the key input), and setting the release time appropriately, thus achieving the same effect automatically.

Silencers by Sonologyst

Excellent review of Sonologyst’s album “Silencers. The Conspiracy Theory Dossiers”.

SONOLOGYST Silencers - Lo res album cover for web

Naples native Sonologyst (Raffaele Pezzella) has recorded for labels like PeopleSound, Eighth Tower, Petroglyph Music, Sirona Records and Sillage Intemporell – but this is his first on Cold Spring. Martin Bowes mastered Silencers: The Conspiracy Theory Dossiers which is available on limited edition CD (digipak/booklet) and in digital format via Bandcamp. Ten ambient sci-fi tracks snake and wander remotely as heard on the title track with evasive drone and pitchy contortions. Both Singularity and Monotape set the dramatic scene here, like a refined installation or film soundtrack of warped sonic waves, a geiger counter and lots of mystery. This has a similar vibe as themes experimented on the mid 90’s ambient project SETI (Savvas Ysatis and Taylor Deupree), though here it’s more documentary-type exploration and less fantastical. On Nocturnal Anomalies there’s a disturbance, an alien being of sorts, just whaling over a hybrid hiss. This is illustrated clearly…

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Mastering to Cassette Tape

For tape mastering service please contact me via email: (Raffaele)

Also visit the Unexplained Sounds Group label for experimental music and streaming radio programs:

Beyond the Nostalgia

The anatomy of the Compact Cassette Tape

I grew up listening to vinyl and cassettes; I’m not that old, but growing up all we had was vinyl and cassettes to listen to. The first time I recorded anything, it was on cassette. I remember my dad letting me use his Hi-fi to tape some of our vinyl records and things from the radio when I was a child and when I grew up and wanted to record some music with friends, we all pitched in and bought a 4-track cassette portastudio. After bouncing tape tracks on that, and using one of the tracks to print timecode so it could trigger playback on my sampler, the mixdown went to another (stereo) cassette deck. That’s really when I became interested in getting the best sonic fidelity from cassette tape.

So why go back to using cassettes? It’s 2014 and today is the second year “Cassette Store Day” is observed around the world; why is this format all of a sudden making a (albeit small) comeback? I think people are becoming more interested in having a tangible product. Tapes are compact enough, are portable and require less maintenance than vinyl as a playback medium. Those who are getting into recording with tapes are realizing that there are many variables involved, and it’s possible to use the limitations of the format to get a unique sound.

Those new to cassettes are also finding out that it’s an inexpensive format to record with and reproduce. You can get tapes made for a lot less than Vinyl and CDs. For those who don’t currently own a cassette deck or walkman, you can find a cheap one in the used market. A good turntable in comparison, will usually set you back quite a bit more than a portable cassette player and good luck making it portable. I’ve seen some of the really cheap Sony cassette players on eBay sell for as little as $5 (although I seriously recommend those without one do a little bit of research and get something better than these low-quality players; you can score a very decent portable tape player for not much more, trust me).

The limitations of the format can give your project a “throwback” feel, and just like vinyl, you can’t replicate the sound of cassette tape effectively in the digital realm (but I wouldn’t be surprised if a plug-in developer comes up with some sort of emulation if cassettes keep getting popular). If you want your project to sound like it’s on vinyl or cassette, you have to put it on those formats; it’s similar to why some people still take pictures with film to this day. Technically, digital pictures are cleaner and sharper than film, but aesthetically, some people like the look and feel of photographic film. Like film, there are many variables that affect its quality; cassettes don’t all sound the same (different frequency response between types and the various formulas of tape that were produced and not all of them have an excessively “hissy” sound to them for example).

Mastering for Cassette (the right way)

If you’re going to create a master cassette to possibly be the source for cassette duplication, you should do it the right way. I wish I could tell you it was as easy as heading down to the nearest Goodwill, spending $30 on a used deck and recording your tape as hot as possible. You will get saturated recordings on cassette, sure, but it’s probably not going to sound great (yes, cassettes can sound good!)

My approach to mastering a cassette is to aim for a similar level of quality that was achieved in the peak days of the media. It was typical for Mastering Engineers then to audition a cassette after mastering to it, and make tonal and dynamic range adjustments as necessary to make the cassette recording sound as good as possible before it hit the bin loop duplicator for mass production.

It Tapes TAPES!

Up until the early 1990’s, cassette bin loop duplicators were analog devices and they used a cassette master tape as the source, often this source master tape was “emphasized” a bit for the format. Digital bin loop duplicators started to become popular in the early to mid 90’s and these used a digital source, usually from DAT or hard drive using first generation ADCs/DACs. In the peak days of commercial cassette production, a degree of effort went into creating the source master cassette or digital source, since it was known that high-speed cassette duplication would degrade the quality of the tape copies to a degree and with the usage of noise reduction systems like Dolby B, emphasis was made between 4k – 10k to make up for the loss of frequencies in that range when encoding the source tape with the Dolby B NR system. Since it’s hard to predict how the NR profile will affect each recording, it was typical to make adjustments after a few test recordings.

These days, cassette duplication services will accept digital files (.wav, .aiff, .mp3, etc.) They should have an engineer on hand to make sure the cassettes that are being made sound as good as possible, but chances are they’ll just transfer your files “flat” using your source files. Ideally, they should make frequency adjustments to the source as needed if the tapes that are being made don’t sound optimal. If we’re talking about the sound of throwback Hip Hop tape releases, consider that the dynamic range of those older albums was bigger than the releases of today; people weren’t smashing levels as much as we do these days, so that’s going to have an impact on the way your tape will sound.

When mastering to cassette, I use the full resolution 24 bit masters to feed the recording deck an unbalanced line out from my mastering console, and drive the input to my cassette deck to allow the cassette format’s saturation characteristics to give the material that “crunch” that you might be familiar with, especially with older cassette releases from the 90’s, for example. The saturation that’s achieved on tape will help give it a sound of cassettes from back in the day and it will sound slightly different than your digital release.

On the processing side, it’s always useful to make test recordings and see what they sound like afterwards, and tweak your processing chain to get the best sound for this format. I usually like the way recordings sound with a little bit of compression focusing on clamping down percussive peaks slightly, and the UAD Fairchild is one that I like often. The UAD Neve 33609 sounds good as well, but it also depends on the material. Brightening the mids and highs is also something I’ll do, and for that my go-to is usually the UAD Pultec Pro. This is just a starting point for me, so if this doesn’t sound right I will try different bus compressors and EQs, then make a few recordings on tape and settle on whatever sounds best.

Overloading a cassette deck’s circuitry isn’t the same with all available cassette recorders out there. Higher end Hi-Fi and professional decks equipped with Dolby HX Pro are able to record hotter levels (about 6dB) on tape without added distortion, this also means we can saturate more tastefully. I have a restored Tascam 122 mkII recorder, which was a typical workhorse mixdown deck in many Mastering studios back in the days when record labels were interested in putting out the best possible sounding cassette releases. Many of the tapes I have to this day have a Dolby HX Pro logo on them, to suggest that the cassette master was mastered on a deck equipped with it and many were also encoded with Dolby B (although as tapes age, I find they sound better with Dolby B disengaged, even though they might have been encoded with it).

Dolby HX Pro was considered to be a major update to the compact cassette format when it started to be used in the early 80’s. Playback decks don’t have to have HX Pro built in to be able to play tapes that were recorded using this technology; it’s a process that happens during recording. Essentially, cassette decks equipped with HX Pro are able to produce louder cassette recordings with less noise than those that aren’t. Some high end consumer recorders like the Nakamichi Dragon, considered by many to be the best consumer cassette recorder ever made, didn’t use HX Pro because the quality of the recording head was so good that it could achieve similar recording levels with minimal noise and distortion. However, unlike professional-grade decks like those made by Tascam, bias selection isn’t automatic on the Dragon and it must be set manually for each tape type; cassette decks that automatically adjust bias for each type of tape do so by identifying a series of indentations on each cassette tape that is loaded. Scarcity of parts for servicing and cost of repairs (if you can find someone reliable that can do so) these days also make the Dragon not ideal for professional use.

Taming the Hissing Beast

Dolby NR (Noise Reduction) is an often misunderstood subject by many new to the format. Most consumer decks and portable players come with Dolby NR B. Many high-end consumer and professional decks often came with both B and C. For the sake of simplification, B reduces hiss during recording a bit less hiss than C, which extends the noise reduction frequency down to about 100 Hz. Both were part of an encoding (recording) and decoding (playback). If you record your tapes using B, the playback deck should also be set to B (and the same goes when using C). Commercial cassette tapes used the B profile, while C was aimed towards home recording gear. Fostex used the C system in many of its multitrack cassette and reel-to-reel recorders, so it was useful to have a stereo mixdown deck that was able to encode and decode both noise reduction systems.

Dolby B was developed in the late 60’s to help minimize tape noise. Dolby C was developed in 1980, and HX Pro came soon afterwards. By then, tape formulas had advanced quite a bit. As I mentioned earlier, not all cassettes sound the same and this is because there are different types, which are made with different materials that act as a magnetic element.

Before we go on to the different types of tapes, something that should be mentioned is bias. Bias is an inaudible, high frequency signal that is applied during recording. This signal is mixed in with the audio signal that is being recorded and moves it to the linear portion of the tape, so that the audio signal is recorded faithfully. The bias signal changes amplitude depending on the type of tape being used (lower bias for Type I, higher bias for Type II and even higher for Type IV tapes). Cassette decks either set bias automatically by reading indentations of the cassette shells themselves, or they allowed users to set the bias curve themselves, on these types of decks, there are controls usually labeled “normal” (for type I) “chrome” (for type II) and “high/metal” (for type IV).

Type I: This was the first type of cassette tape that was manufactured. The magnetic element in this type of cassette is gamma ferric oxide (commonly known as “ferric tape”). These kinds of tapes are usually labeled “normal bias” and tend to be noisier (more hiss) than Type II cassettes, but a lot of cassette tape enthusiasts prefer the sound of a well-made Type I tape for recording, like the Sony EF series, because it tends to warm up low frequencies in a way that Type II tapes don’t, and are able to record at slightly higher levels without saturation. High frequencies aren’t as bright as they can be on Type II tapes, which may be a desired effect depending on the type of music being recorded. When using one of the better Type I cassettes, it might be useful to use Dolby B (or C, which may produce slightly warmer recordings, but keep in mind what I said earlier about both NR profiles and their availability on consumer decks).

Type II: Developed not too long after Type I cassettes, this formula uses chromium dioxide and is commonly referred to as “chrome” tape. Type II tape is able to reproduce brighter high frequencies with less hiss, but it also reduces the response of low frequencies slightly. When using a high quality Type II tape, you may find that you’ll end up with better recordings when you don’t encode your recordings with a Dolby NR profile, and perhaps bump up the low end and the mids a little bit on your source recordings before hitting the tape.

Type III: This formula, known as “ferro-chrome” combined both “ferric” and “chrome” formulas on the same tape in hopes to get the best of both worlds: the better bass response of Type I and the better high frequency/reduced noise of Type II. The Type III had a short life span, from about the mid 1970’s to 1980. One of the main problems with it was bias; should you set your deck to normal (Type I) or chrome (Type II) bias? Those decks which set bias automatically would default to normal, and after a couple of years of consumers testing out this type of cassette (and manufacturers of cassette decks watching closely), they discovered some flaws, like the chrome layer of the tapes coming off with heavy use. They also discovered that when it came down to sound quality, the Type III didn’t offer an obvious improvement over the Type II cassette for those users that had decks that were able to adjust bias manually; many users felt that Type II, with its higher bias setting performed better. Manufacturers were reluctant to incorporate a middle ground bias setting for Type III in their tape decks because of the flaws being reported by consumers. They might have, if consumers would have bought into this particular type of cassette, but it was never popular and it struggled making worthy sales throughout its short life.

Type IV: Towards the end of the 1970’s, a completely different formulation of tape hit the market. This one used metal particles instead of oxides and consumers immediately saw a benefit from it. Known as “metal” tape, the Type IV was able to record even louder signals with less distortion in the upper frequencies than the Type II and the low frequencies also sounded better. This increase in quality did not come without some negative effects. Head wear was increased as the metal particles are more abrasive than oxides, and it was a bit more difficult to erase previously recorded material from it. The cost of these tapes was often more than double the cost of an average Type II cassette but it was worth it for a lot of users who heard the improvement in quality over the previous types of cassettes. It wasn’t long before manufacturers started including a metal bias selection in their decks, which happens to be an even higher bias signal than that which is used for the Type II cassette. It was definitely the best of all the types when it comes down to sonic fidelity.

After reading all of this, don’t you feel like giving your DAW a nice big hug? Isn’t it nice these days to just throw a good chunk of cash into a box with excellent Analog-to-Digital converters? Writing this article took me back to a time where you had to put in a lot of time and effort into getting decent recordings on cassette tape. I also remember lots of frustrating times with the format, like tapes stretching, dropouts and tapes being chewed up in the transport. I also remember what cassettes sound like when you play them loud through a nice system; the ones that were done right sounded excellent. With that, I can say that I see why this format is becoming increasingly appealing to artists, especially those that want a lo-fi feel from their recordings and are looking for that familiar vintage sound of the format.

Sometimes limitations inspire creativity, and the compact cassette tape format definitely had a lot of them.

Pan for ultimate width

Panning is the most crucial step for getting a wide stereo image.

Panning lets you place individual instruments, or even certain frequencies of instruments, in a particular spot within your stereo image—and go as wide as you wanna.

Always make your panning decisions based on your entire mix. There’s a few different approaches to panning, but no matter how you use them, they’re key to getting a wider mix.

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Here are some quick tips and rules for getting your panning pristine and achieving width in the mix:

Low frequencies are the heart of a groove and drive your rhythm, so keep them straight down the middle.

Keep your low end in the middle

Don’t pan your lower frequencies. Low frequencies are the heart of a groove and drive your rhythm, so keep them straight down the middle.

Keep your L and R balanced

Our brains naturally want to center stereo images, so keep the L and R channels balanced to avoid confusion in the phantom center.

Always pan with your ears, not your eyes

The only thing that really matters is how it sounds. When panning, close your eyes and listen until you hear that perfect sweet spot.

Even if the volumes of your L and R channels are balanced, if one side has more sound competing for the presence zone this can cause the stereo image to sound off balance.

Keep your lead vocals in the center

Keep your lead vocals to the center as well unless you have good reason to do otherwise. You want that lead vocal front and center to really let it shine.

Reverbs for multi-dimensional sound

Reverb is a classic mixing tool for adding width, but also that third dimension to your mix: depth.

By adding depth to your stereo image, you’re also expanding the stereo image as a whole. Reverb will give you more room for every sound to breathe and settle into the mix.

There are many different ways to use reverb and add space to your mix, but any reverb technique will add some degree of depth and spaciousness to your mix. And there are many types of reverb. Each is capable of adding a distinct vibe and depth to your mix.

Choosing the perfect type of reverb to give that extra space without drastically changing your audio’s character will take some practice. But when it comes to width, Hall reverb is a good place to start.


Don’t stop there though… all types of reverb can do wonders for adding three-dimensionality depending on your mix and production style. It can be useful to experiment with different reverbs for different tracks in the mix, or alternate dry tracks with reverb treated tracks. With small amount of effect, that can add unpredictable and variable spaciousness during the final mix.

Hot Tip: Using reverb with a short decay time will add a subtler reverb effect. It’s great for when you want to add width and depth without changing the overall character of a sound.

Some examples where reverbs are used with a creative and functional approach:


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