Basics of Acoustic Engineering

For

(​Indie Recording Depot​)

By

Alpha Victor​ (Andrew)

Producer @ ​Flute Cafe Music

 Introduction

Greetings, my name is Andrew and I will be your guide as we cruise through the basics of Acoustic
Engineering. My approach in explaining the topics will be in an informal question and answer format.
My knowledge in Acoustics stems from my background in Heavy Electrical Engineering. Which
means, aside from designing your quiet air conditioners, I can also design rocket engines, jet
engines and power plants. Acoustics is a significant part of jet engine design, the noise of a jet
engine is at hazard levels and if sound is not properly absorbed and diffused, it can self-resonate in
jet cavities leading to catastrophic results similar to the infamous “Galloping Gertie” from Tacoma,
Washington.

In electrical aeronautical engineering, we deal with sound dynamics of jet cavities and pressurized
cabins. In civil engineering, acoustic engineering is most commonly used in design of concert halls,
cinema halls, bridges, music studios and home theaters. For the sake of simplicity, I will approach
the subject from a musical perspective instead of an engineering perspective.

To many, acoustics feel like some kind of magic that makes a studio sound good. It is that invisible
technology that works for you night and day. Whether you are an engineer, audio enthusiast or a
musician, understanding of basic principles of acoustics can make a difference at many levels.
Please be advised, this is not a primer of what the best acoustic treatment is, instead it covers the
basics of acoustics without going into cumbersome details of a myriad acoustic engineering
products.
 Chapter 1: Basics of Acoustic Treatment

The field of music resonates with a common question:

What is the most ideal environment to listen and record sounds and music?

The truth, if you want purest sound, climb on top of the tallest tower in the largest field out there
where there are miles of nothing. That is the best place to listen and record music, provided you can
fight off the dragon. Yes, large open fields with miles of landscape, no obstacles, no parallel
structures, no reflections, no distortions, no standing waves, no high frequency absorption; just plain
pleasant and lively sound with a consistent frequency response.

So why do we not record or mix outside?

You can; however, rain, wind and flying birds above you might have other ideas. ​If you are lucky
enough to live in a quiet deserted area with miles of nothing, you just might get away with it.

Why is acoustic treatment needed?

If you cannot listen to music or record music out in the open, you are stuck listening to it inside halls,
theaters, opera houses, your favorite home theater or a basement. Acoustic treatment is mandatory
for professional auditorium buildings, without it, the reverb of big spaces and resonant characteristics
of rooms will make it difficult to decipher a single word of speech or a single musical note.

In hospital buildings, acoustic treatment matters a lot due to health risks caused by sound pressures
that could easily reach over 30 Decibels and cause anxiety and heart flutters. Our bodies are made
of water, and water resonates with the sounds around it. You can test it by playing music next to a
cup full of water. You will start seeing ripples in the water. Poor acoustic treatment in hospitals can
cause nervous breakdowns due to elevated anxiety levels caused by resonance, not to mention a
hazard to patients with sensitive ear conditions.
In homes, I have seen people invest thousands in sound systems for their home theaters but not
many bother to invest in proper acoustic treatment. A $5000 sound system will sound like a $100
sound system in an untreated room with poor acoustics. At that point, investing thousands in a
sound system is a sheer waste. Most homes are not architecturally designed for listening to loud
music. I ​have​ seen expensive sound systems in small rooms and large rooms alike. In small rooms,
the room ​resonances​ from reflections off parallel walls kick in after awhile and the whole sound just
becomes one large pool of mush. High power sound systems In larger rooms are tainted with reverb
and flutter echoes, phasing issues, boundary interferences and imaging issues. The end result is a
sheer atrocity on the ears.

So what is next best thing?

To find what is the best form of indoor studio environment, we must first understand what happens to
sounds inside rooms. Sound bounces off walls, ceiling and floors and create what we call
“reflections”. Some reflections are good, some are bad. The result of bad ones are called “Standing
waves”. Standing waves happen when reflected waves collide and converge to create a wave that is
relatively standing still . Standing waves from bass and low-end frequencies are especially notorious
from an acoustic point of view . Other problems are natural low pass filters like carpet, thick curtains
and clutter. Combination of all this is what creates that boomy, inaccurate, nasty closet-like sound
that we all dread as audio engineers.

To fix these, we need to absorb the extra sound energy created and diffuse the rest to minimize the
effect of standing waves and other sonic distortions. That is where acoustic treatment comes in.

The goal is to create and represent accurate sound.

Low-end absorption is a hard beast to tame, the lower the frequencies, the denser, more finely
porous, thicker, and more expensive material you need. That is where most of the money is spent.
As your room is treated properly, it gets closer and closer to accurate and lively sound.
How do we test when our room is accurate?

Most professionals train their ears to detect accurate sound, however, even the sharpest of ears can
only detect with about 80% of accuracy. To go further, the best way to detect is to place real time
microphones in areas of the room. Play broadband static with equal energy per octave (pink noise)
and analyze the sound curve at the listening end. If your curve matches, voila, you have accurate
sound, if not, find out which areas have unnatural boosts or dips. Once you find the problem
frequency ranges, you can treat (absorb or disperse) them.

Size of the studio

It is often contemplated, how big is too big and how small is too small of a room to listen and record
sound? Short answer, there is indeed a good compromised middle ground, where room size and
room ratios meet to create a decent overall balanced acoustic environment. The long answer is that
there is a lot of math involved!

Anyway, let’s start with a little high school fun!

speed of sound = (frequency of sound) x (wavelength of sound)

In order to be an effective studio, we must be able to handle low frequencies; let us take an example
of sound at a frequency of 30 hz. By handle, I mean give “room to breathe” or to flow unrestricted
where the ​dimensions of the room are larger than the wavelength of most frequencies ​in the human
audible range. When room is large enough, it weakens the resident resonant frequencies of the
room (more on this later).

Speed of sound is 1,130 ft per second, frequency is 30 hz, let’s do the math, yes cold hard math!

Wavelength = 1130/30 about ​37 ft

In an ideal world, a bit more than 37 ft of room (from the sound source), is important to let the sound
flow unrestricted inside a room for low frequencies around 30 hz. Which means an effective studio
could, in an ideal case, have about ​37 ft long unobstructed path from the direction of the sound
source​. Remember, this is not some magic number that is going to turn your studio into the Boston
Symphony Hall; several other factors - not just room size are considered as well. This is to give you
an idea that room size is important.
A larger room will have axial resonances (room modes, more on that later) that are lower in the
spectrum and less problematic, they will also have lower intensity resonances that build up over time
among other modes compared to smaller rooms. Larger rooms may, however, have some issues
with reverberation.

Not everyone has the money to get a larger studio, most studios are 20 ft on average, which means
sound waves of low frequencies will not have enough “room” to travel, they will double back,
resonate at higher intensities and create unpleasant sound. Which is why smaller studios, need a lot
more low- end absorption​ than larger studios.

For a 20 ft studio, low-end sound waves will be restricted by about 17 ft (that doesn’t mean add 17 ft
long or deep panels; no). To make up for the missing distance, acoustic engineers have designed
many things from attack walls to air gap acoustic panels to porous polymers each of which handle it
in their own way. Which one is the best? Only your budget can decide as you can never have too
much low-end absorption.

More on spaciousness

The need for proper room size in good acoustics and inverse square law:

Intensity of sound = Power / Area of sound dispersion

which means intensity of sound drops the further away from the source you are.

Let's say there are 50,000 watt speakers, dispersion of sound depends on the area so when your
radius is R, the area of sound at that point is the area of the sphere at that point. Suppose that it is a
perfectly dispersed spherical sound, the area of the sphere at radius R would be:

4 Π R2 (four pi R squared)

Intensity 1 = Power / 4 Π R2

Now let's say you have more room to breathe and want to double the radius (2R):

Intensity 2 = Power / 4 Π (2R)2 = ¼ (Power / 4 Π R2 )

=> Intensity 2 = 1/4 (intensity 1)

This means when you move twice as far, the intensity drops significantly (a fourth) - ​this is the
inverse square law​. This also affects reflections. The higher the intensity at the impact point the
stronger the reflection. Which is even more reason why having enough room is important in acoustic
environments.

Not saying smaller rooms cannot be treated, they just have to treated differently. As we develop
further, importance of ​room ratios​ and geometry takes priority over sheer brute force sizing of
space. There is a sweet spot in the room size and ratios that can serve a good enough purpose to
be a good middle ground or starting point.

Let’s start by understanding some basic phenomenons that affect sound inside rooms.

Resonance, Resonant Frequencies and Introduction to Room Modes

Before we dive into extremely vast topic of room modes, let's take a moment to understand
resonance. Resonance happens when a force interacts with another medium that is either already
vibrating or can vibrate. The result is an increased intensity (greater amplitude).

When this driving force is periodic and matches the frequency of a medium that is vibrating, the
amplitudes will keep getting bigger and bigger and bigger indefinitely - unless there is some kind of
dampening going on.

The Swing conundrum:

Acoustic resonance like mechanical resonance is dependent on dynamics of simple harmonic

motion. Let's take a swing for example. When you push a child on a swing ​at the same rhythm as
that of the swing​, the kid keeps going higher and higher. That result is resonance and the ​frequency
of the swing is called the ​resonant frequency.

Now if you break the rhythm and start pushing the child at random times or from opposite end, the
swing will eventually not go as high and eventually lose the battle with friction and come to a stop.
In the case of a swing, the resonant frequency is also its fundamental frequency (lowest resonant
frequency). ​No matter how hard you push, you cannot make a swing, swing any faster​ without
changing its chain length. ​A swing will always have the same frequency (its called fundamental
frequency)​. A swing is a simple case, as it has only 1 resonant frequency. Vibrating mediums can
exhibit many resonant frequencies (overtones).

A note here, resonant frequencies are not the same as harmonics. Things that vibrate can produce
both harmonics and can have resonant frequencies of their own. Same principle applies to acoustics
of instruments and yes rooms. Even though structural elements of a room don’t vibrate as much, the
air trapped in the room can, resulting in resonant frequencies of their own (room modes). This
phenomenon is caused by standing waves that are created by reflections of sound waves from room
walls when a sound source (such as a speaker) is introduced.

Wine glass shattering conundrum

You must have all heard about wine glass shattering from high pitch sounds. This is due to
resonance. When external frequency matches the natural resonant frequency of an object,
resonance occurs and amplitudes get higher and higher causing distortions strong enough to break
the glass. The video link below will demonstrate this effect.

https://www.youtube.com/watch?v=17tqXgvCN0E

Every system has a resonant frequency​ and they can cause a lot of issues. This can be good and
bad. When dealing with jet engines, frequencies from jet engine systems can cause harm to
structures If proper sound treatment is not present.

Why do objects in nature have resonant frequencies?

A lot of research is happening in this area, but some of what we do know is that objects in nature are
subject to many forces, some explained and some unexplained that cause the atomic structure to
stretch and contract. When we drill down to the basics of resonance, it is a cycle of energy shifts
back and forth - potential to kinetic, kinetic to potential, electromagnetic to heat, photoelectric to
magnetic, etc. Any kind of energy shift that is cyclic in nature will cause the system or object to have
a resonant frequency. Objects in the universe are vibrating and that is what gives them their
resonant frequency.
 Chapter 2: Sound Reflections and Room Modes

Room modes are the ​core​ of the acoustics and acoustic treatment of confined spaces. Think of a
room like a music box, or a sound box of a guitar or a piano. It is hollow and filled with air. Perfect
environment for resonant behavior. It is true that each room is like a musical instrument with hidden
resonant frequencies that live within based on its geometry.

I intend to keep it fairly simple to explain this phenomenon without going into too much detail like
nodes and antinodes, I will use a basic mathematical approach to explain.

Geometry can be used to guess the resonant frequencies in a room at a basic level by ​calculating
the wavelength using the geometry​ of the reflections. This process can even be done mentally.

It is no secret that sound can bounce off walls, ceilings and structures; these are called reflections.
Although, they are not exactly reflections like light rays off a mirror, when we are dealing with sounds
in the human hearing range inside a room where wavelengths are somewhat around or within the
boundaries of the room - ​we can treat their behavior like light rays

If we treat them like rays, we can then ​use geometry​ to analyze the reflections in basic cases.
When a sound wave is reflected back from a wall or combination of walls - at a certain wavelength of
sound they will combine (superimpose) exactly on top of each other to deliver a standing wave. ​It is
called a standing wave because when two waves traveling in opposite directions meet, they come to
a halt, as in appearing to stand still but continue to transfer energy back and forth.

These standing waves are static frequencies and their multiples that resonate inside a room are
called ​“Room Modes”. ​Room modes are internal resonance of a room that exist when you turn on
your loudspeakers. These are, static frequency hums inside that get stronger when a particular
frequency or multiple of that frequency is played.

In other words, your room will start “singing” like a tuning fork at certain frequencies causing
additional interference with the sound being played in it.
Example:

Let’s say sound gets reflected back and forth between two parallel walls (axial mode).

If the room is 10 ft long, the round trip of sound back and forth is 20ft.

Which means ​a sound with 20 ft wavelength will fit nice and snug within that length​, delivering
a standing (stationary) wave at that frequency.

Now what sound has a wavelength of 20 ft? Let's find out!

1130/20 = ​57 hz​ (reminder: 1130 is the speed of sound)

This means when you excite the room with sound around 57hz, the room will start resonating
(humming) around that frequency and multiples of that frequency (harmonics). Room modes can
exist from 20 hz upto 300 hz in complex cases. After that they still exist but they are less
problematic.
There are 3 types of room modes

For simplicity we will take into account only the first harmonic (half wavelength, as in each leg of the
jump is equal to half of the wavelength)

Axial Mode:

The one we discussed in the above example is called “axial mode” (between two parallel walls), it is
the simplest of the cases and very simple to calculate. See illustration below:

See animated version of this illustration here!

Tangential mode​:

In tangent mode, sound bounces around the 4 walls instead of 2. These are less intense than those
with two walls back and forth; normally around half as strong as axial modes of the same room.
(How? - a discussion for another time). In a room, if sound bounces off four walls, it would generate
a standing wave with a ​wavelength equal to the length of the diagonal of the room​ in its basic
case.

In a basic case, in a 9 by 12 room, sound will bank tangentially, ​forming a standing wave of
wavelength 15 ft ​(Note: 9,12 and 15 are Pythagorean triples). So the resonant frequency of
tangential mode would be around a wavelength of a wave that fits snug in 15 feet is 1130/15 = ​75 hz
(approximately). 75 hz and its multiples is one of the resonant frequencies for a Tangential room
mode of a 9x12 room.

Basic tangential modes can be quickly approximated by calculating the diagonal of the room.

See Illustration below ​(see animated illustration here!)

Oblique mode:

This time sound bounces around ​six walls​ (four walls and ceiling and floor).

This is the weakest of the three modes and less problematic as they are about half as strong as
tangential ones. However, if a lot of tangential and oblique resonances are around the same
frequency, that particular frequency can still boom out. Which is why it is still important to take
oblique modes into account as well.

This case of a reflection is a bit more complex to compute geometrically because pressure at more
oblique angles is a lot less than the other two cases and there are six surfaces involved. In one of
the basic cases sound will approximately form a standing wave at a fraction (around inverse of
square root of 2, roughly about two thirds) of the distance of the “​large diagonal​” (for example, the
distance between top left and bottom right corners of the room in its basic case). See figure below.

If your room is 10 feet by 12 feet by 8 feet tall, the large diagonal will be about ​17 ft​.

Sound will form one of the standing waves at around roughly two thirds of it as wavelength, so about
11 ft . The frequency at 11 ft is 1130/11 = ​102 hz​ (approximately).

102 hz and its multiples will become the resonant frequency of the room in Oblique mode.
Here is an illustration for this:

See animated illustration of this here!

Basic oblique modes can be approximated by the inverse root 2 method explained below or even
mentally approximated by two-thirds approximation method as inverse of square root of 2 is very
close to the two-thirds fraction. The actual resonant frequency will be somewhat close to the two
approximations.
Final result calculations:

Note:

Of course, we may think of other ways sound can reflect and create standing waves.

Geometry gets extremely complicated​. While, using geometry we can approximate for basic
scenarios but they get complicated as we progress further and a more pressure driven approach is
needed.

Using calculus (partial differential equations) we can use a pressure or molecular displacement
driven wave equation to calculate resonant frequencies in confined spaces easily. After years of
caffeine driven calculus you will arrive at an equation we can all easily use.
Thankfully Dr. Rayleigh did it for us​ so we never have to.

Rayleigh Room mode equation:

The above equation is a bit more accurate than the traditional geometry methods, while both
methods will yield results around the same frequencies.

Lx, Ly, Lz are room dimensions -

Lx = Length

Ly = Breadth

Lz=Height

l​, m and n are integers normally within (0 to 4) that help calculate different room modes

l​, m and n correspond to axes x, y and z respectively.

For basic cases:

axial use [​l​ = 1 , m=0, n=0 ] or [ ​l​=0, m=1, n=0 ] or [​l​ =0, m=0,n=1 ]

tangential use [​l​ = 1 , m=1, n=0 ] or [ ​l​=0, m=1, n=1 ] or [​l​ =1, m=0,n=1 ]

oblique use [​l​ = 1 , m=1, n=1 ]

v is speed of sound 1130 ft/sec , result will be in hz.

If someone is interested in how this formula is derived, I can shed more light on the calculus and
wave equations involved if you pay for coffee .
Wave collisions and Resonant Boundary Distortions

A room can have many structures where sound can collide and create distortions​. Like a rock band
in a garage with a sheet metal garage door, big glass windows, flimsy paneling, poor quality low
hanging metal roofs, etc. You can literally whisper in these rooms and hear the unpleasant sonic
character of the room. When a wave collides against a resonating structure at a premature cycle
(less than its wavelength) at higher intensities, the collision can generate in-harmonic distortion (odd
harmonics and in-harmonic distortions) that pool up together and eventually create an unpleasant
sonic character.

Here is a diagram that illustrates generation of harmonic distortion that could arise as a result of
collisions with resonant boundaries (odd ones in this case). Be advised that the actual physics
behind resonant boundary distortions is much more complicated than the underlying illustration and
involves many factors other than just wave collisions. They include, but are not limited to factors
such as: elasticity, tension, material, Q , heat, forced vibrations, power of sound, pre-existing
standing waves and boundary interferences, etc.

In most cases, Partial tone distortions also arise from incomplete high intensity collisions with
unpredictable resonant boundaries. partial tones are known as ​in-harmonic distortions​. ​A series of
in-harmonic distortions can create unwanted mud and noise​. Think of it as an unpleasant sound
amplifying soundboard.

Standing waves and ephemeral in-harmonics generated as a result of resonant boundary distortions,
alter the sonic character of the room. Resonant boundary collisions are not as predictable as
non-resonant hard surface reflections (leading to room modes) but they can be understood to a
certain degree with principles of in-harmonic or partial harmonic distortions. In reality, it is hard to
predict exactly what kind of distortion would be generated; that depends a lot on the aforementioned
factors involved.
 Chapter 3: Sound Pressure, Sound Power, Sound Intensity

In acoustics, terms are often changed around; however, they have completely different meanings.

Sound Pressure

When we are referring to ​sound pressure​, we actually mean pressure that is measured in pascals.
The pressure of the sound wave itself (usually in air) or the pressure of the wave as it hits your ear
drums.

In humans, there is a threshold of ​20 micropascals​. Therefore, you won't hear a “sound” at a
pressure lower than that, however, some other animals still can.

Sound power​, on the other hand is, yes you guessed it correctly, ​wattage​ as in the power drawn by
the speakers or the source in order to create that sound. Like 10 watt speakers, 1000 watt
speakers,etc.

Sound intensity​ is power dispersed over an area. Governed by ​inverse square law,​ Sound Intensity
drops by a quarter with every doubling of the distance away from the source.

Now that we got that out of the way,

How are these entities linked? Is there a common ground, some common terminology we can
use?

Short answer, yes, there is. It is called ​DB, Decibels

Long answer, ​yes and no​. In all honesty, we are still struggling and arguing to find common ground.

While DB provides a sense of unity to the different aspects of measurements, decibel itself is a
relative unit; as in, it requires a reference mode. It is a logarithmic entity that requires comparing one
quantity to another quantity.
In case of Sound Pressure, we call it ​Sound Pressure Level​ (DBSPL)

When referring to sound pressure level, we compare pressure of the sound to the minimum pressure
required to hear a sound; in case of humans 20 micropascals (Po).

DBSPL = 20 Log (P/Po)

Po​ = Lowest pressure for ears 20 micropascals

P = actual pressure

Looking at the logarithmic equation it is easy to see that when ​pressure doubles, sound pressure
level increases by +6 DBSPL​ as 20 log 2 equals 6

In case of ​Sound Power we call it DBm

DBm = 10 Log (W/Wo)

W = actual wattage

Wo = reference wattage (normally at 1 milliwatt, 600 ohm)

We can see that when power doubles, power level (DBm) increases by ​+3 DBm

Ham radio operators can most likely relate to it. DBm is used more in engineering for impedance
matching than acoustics but it is still relevant. However, DBm gives birth to another quantity called
DBu or DBV

Analog system operators will relate to this. Power is a square function of voltage, hence we can
derive a relationship, providing resistance is constant. In our example, let’s say 600 ohm.

Since power is a square function of Voltage, it is safe to say that:

DBv = 2 x DBm in logarithmic terms

DBv = 20 Log (V/Vo)

V= actual voltage

Vo = 1 Volt (or 0.7volt in some cases called as DBu)

Easy to see that ​when voltage doubles, DBv increases by 6

Analog musical systems rely on DBv or DBu more than DBSPL which is mostly used in engineering.

All that aside, we have also heard the terms RMS, DBFS, DBA and Phon.

What are RMS, DBFS , DBA, Phon?

As digital era descended on us, we needed a new unit to which the digital analog converters could
relate.

DBFS is Decibels Full Scale. There is ​absolutely no mathematical relation between​ DBFS and
DBu. But we do have guidelines; in Europe ​about 18 DBu is seen as the same as 0 DBFS

In the digital realm, 0 DBFS is the maximum level and ​everything else is negative​. (If you notice,
other DB units are non negative and their minimum value is 0 with relation to the references).

RMS​ is another method to measure sound levels, in this case it is root mean square of the voltage
function in Analog systems over a period of time. Mathematically it is integral calculus, but you get
the point.

In electrical engineering terms, a 6V RMS AC supply will shine with the ​same brightness​ in a bulb
as a steady 6V DC supply.

DBA​ is pretty much the same as DBSPL but measured with an ‘A filter’. To those unfamiliar with the
A filter, it highlights frequencies between 2000 and 4000 over others, that is around the ​resonance
modes of our ear canal.​ DBA is used in design of concert halls and is quite relevant in the field of
musical instruments and acoustics.

Phon

Phon relates to psychoacoustics and also in civil engineering. It relates to the DBSPL of a 1 khz
tone. In civil engineering, it is used in design of speed deciphering systems or speech isolating
systems. Hope that demystifies some of the issues around terminologies.
Now that we understand some terminology and also know how to calculate room modes, where do
we go from here? First step is to plot them all out in a table and look at the data in order from low to
high frequencies.

What to do with the Room Mode data

When we are looking at an ordered list of room modes, for simplicity sake, let's deal with axial
modes of a 10 x 10 x 10 room (ice cube).

We already established that wavelength of a standing wave in a 10ft room is around 56 hz. Let’s say
we have already calculated the other overtones:

1. 56 hz

2. 56 hz

3. 56 hz

4. 112 hz

5. 112 hz

6. 112 hz

7. 168 hz

8. 168 hz

9. 168 hz

10. 224 hz

11. 224 hz

12. 224 hz

13. …

It is easy to see that this is a terrible scenario! We have multiple modes at the same frequency and
the gap between frequencies is much larger than 20 hz on both sides ( 56 - 112 - 168 ). This means
that these frequencies will stand out quite a bit as there is little to no masking going on.
To a human ear, tones stand out more if there is no tone around 20 hz below and 20 hz above in
frequency; a phenomenon called masking. When tones are within 20 hz apart on both sides, it will
be hard to isolate the tones​. Now let's take a look at axial modes of a 10 x 12 x 8 room (Lunchbox).
Let's assume we have precalculated all of this for us and ordered it sequentially.

Frequencies in Hz

1. 47

2. 56

3. 70

4. 93

5. 112

6. 140

7. 141

8. 168

9. 189

10. 224

11. 234

12. 281

13. …

We can see that the frequency response of the lunch box is a bit more distributed than the icecube
one. It still has some problems though. We can see that frequencies around 140s are more than 20
hz away from their neighbors, 112 and 168. To add to it, frequency 168 is more than 20 away from
141 and 189 and so forth. So even though we don't have all three overlapping modes, we still have
an isolation (not enough masking) issue.

There are room sizes where such issues don't exist or are at a minimum like the bonello sizes or the
bolt footprint (more on this later); they still dont always rectify all problems.
Truth is, even though theory shows some golden (sweet spot) areas, we are not near a defined size
rule due to the many variables involved.

Room Sizes, Room Ratios and Room Geometry

(Is bigger, better?)

When we talk about a room, let’s first ask, what is the room for? For simplicity, let's take a balanced
approach; a decent, all purpose approach (listening, mixing, tracking, performing). We know that
size of the room affects room modes and different sizes are going to have different room modes.

If it is a fresh build, it is a good idea to look at room sizes first and see what fits the budget; material
and architectural restrictions.

To answer the age old question, is bigger always better?

The short answer is Yes it is, however it comes with strings attached.

In two rooms that are proportional (same ratios) a bigger room will always be better than the smaller
room. Even though bigger rooms have reverberation issues, when compared to strong room modes,
room modes win in their villainy.

In that sense, bigger is better but is bigger by a few inches better? Not necessarily, It is too small of
a difference to tell them apart.

The room size difference has to be large enough (capacity differences of around 3 cubic feet and
above) to be able to clearly tell the differences between the two rooms. (Reminder: ​To a human ear,
tones stand out more if there is no tone around 20 hz below and 20 hz above in frequency; a
phenomenon called masking).

Consider a room much larger than 10 x 12 x 8. Let's say the room is 35 x 33 x 29 ft. The room
modes will be weaker in the larger room, though they will still exist. Room modes will always exist in
a room any size, any shape; even in a spaghetti shaped room.

In a much larger room, room modes will be weaker. Why? Due to the fundamental axial mode for the
35 ft room mentioned above, which will be under 20 hz. That is below the hearing spectrum of the
human ear and their overtones, but even though they will be very much in the range of 20-250hz,
they will be weaker.
Overtones in general are weaker than the fundamental, hence making them less problematic.
Another case for larger rooms is the inverse square law - sound intensity dissipates with distance.
Every double the distance, intensity drops to a fourth.

Room Ratios

Now that we have established that going as big as possible within the budget is a feasible thing to
do, next thing to consider is what dimensions should they ideally be? This has been contemplated
for about a hundred years and yet we don't have a solid solution. We have some theories that get
very close (Bolt, Bonello, Trevor) however, they still fail in certain scenarios and building conditions.

Many different types of approaches have been considered when deciding on a room ratio for an
ideal all purpose audio room. The starting point for all are room modes.

We want to see them more evenly distributed as in no overlapping room modes and spread apart
within the 20 hz sweet range. Earlier statistical approaches were based on averages and standard
deviations. They gave us a starting point but a bit more musical approach was needed. Bonello
decided to plot them on a scale of third octaves. Which is what gives us the bonello scale. Bolt’s
approach and as of many others, was statistical. Bonello’s approach does give us sweet spots in the
Bolt range as well but they deviate after a while. So the bottom line is that there is a zone where
Bonello and Bolt meet that is considered a sweet spot.

That size is normally around ​9 ft by 17 ft by 23 ft.​ A ratio of around 1: 1.8: 2.6.

Bonello arrived at this by plotting many room modes per third octaves and comparing the response
to his main criterions that the curve should be monotonic (always increasing and always decreasing
when traced backwards) He also plotted it against the other two conditions we discussed above (20
hz apart on both sides and not overlapping). The result was a curve that satisfied both his criterion
and criterion of others. Even though the apparent ideal size of 9, 17, 23 is still considered by many,
there is still a pretty sizable flaw in his analysis. He treated all room modes, axial, tangential and
oblique the same. Tangential and oblique modes are weaker, namely 3DB and 6Db weaker than the
axial ones and hence, should have lower priorities in real time. He also did not take into account
building materials and structural integrity.
A lot of modern approaches start at a baseline of the bonello graph but branch off from there. Many
studies are still ongoing to find a solution that works with building components, materials and
prioritization of different modes. The bottom line, however, is still the same. Avoid boxes (cubic
rooms) and also try to avoid having two of the same dimensions in the room. Try to go as big as you
can with the budget restrictions.

Room shapes

To splay or not to splay - do angled walls matter?

The answer again is, Yes and No.

Angled walls matter to a certain degree as they give more surfaces for sound to bounce off and
reduce the intensity but they will still create room modes. Only this time room modes will be much
harder to predict without industrial strength calculus involved! While rectangular shoe box room
modes can be calculated mentally, calculation of irregular rooms will involve a computing strength
required to solve a bitcoin puzzle! Since each room is different, having irregular walls only introduces
elements of surprises.

For tracking rooms, they still might be a good idea to have - as in, if you already have them, don’t
bother breaking them down (put down the sledgehammer), just work around it and take advantage of
it. If you don’t have them, don’t bother adding them as the angles should come from room treatments
and not hard walls.
 Chapter 4: Speaker Boundary Interference (SBIR)

Speaker placement and listening positions are also among some challenges involved in room
acoustics. How far you place the speakers from the walls can affect what frequencies get cancelled
out (nulled), dipped (lowered) or boosted.

When you mix a delayed audio signal with the original audio signal -

you witness a phenomenon called comb filtering.

It depends on how far apart the signals are timewise (phase shifted) that will determine what
frequencies get boosted and what get dipped. The resulting effect ​looks like a comb​ on the graph ( a
bunch of notches )

A secondary effect of comb filtering is ​SBIR​ (Speaker Boundary Interference).

Low frequencies have a notorious reputation to bounce off the back wall (wall behind the speaker)
because low frequencies have larger wavelengths closer to our room dimensions. Higher
frequencies do bounce off as well but air pressure in the back of the speaker for higher frequencies
(1000 hz and up) is significantly lower than the lower frequencies. That is why boundary interference
has little effect on higher frequencies.

When a signal is delayed by 180 degrees, it will cancel out the original wave when mixed with it.
See Illustration below or ​see animated illustration here!

NOTE: There is a ​common misconception​ regarding this phase shift, some believe that when
sound reflects off a wall it undergoes a phase change of 180 degrees (pi) and that is what causes
SBIR; this is inaccurate.

A ​Sound signal does “NOT” change phase when it reflects of a wall​. Wave physics illustrate
that sound waves (longitudinal pressure waves) will not undergo a phase change upon reflection like
transverse waves. A compression will reflect as a compression of air and a rarefaction will reflect as
a rarefaction (void). Those two factors are what leads to buildup of Pressure Antinodes on the walls
in a room.

So why when a wave is reflected off walls cause a phase shift?

Keyword is “shift” and not “change”. While they have the same meaning in a sense, a shift is not a
change. ​A shift is a “delay” and “change” is instant.​ and a change will affect the sound wave
instantly.
Example: How does a flute work?

When waves inside a pipe instrument, like a ​flute ​hit the open end, they undergo an instant phase
“change” . The wave instantly changes phase by 180 because the compressed air blooms out of the
open end of the flute and the void that is created travels back up the flute like a bubble. A
compression changes to rarefaction upon reflection of the open end and the frequency of this
oscillation of air back and forth is what makes a flute work.

In the case of a rigid wall, there is no room for air to “bloom out”. A compression will return as a
compression, hence there is no phase change.

However, when reflections cause a delay, they mingle with the original signal occasionally causing
constructive and destructive interference. They can cause boosts and dips in certain frequency
ranges. The extreme case (a null) is illustrated in the above animation.

So how to fix SBIR issues?

Solution is rather simple, however hard to implement at times. The ideal solution is to mount the
speaker inside the walls (flush), making rear wall reflections a non issue. The entire wall then acts
like a driver for the speaker. One giant baffle. However, not all speakers are built for wall mounting
and making the speakers properly flush with walls is an expensive process.

The next best thing is to go as far back as you can to the back wall and calculate the distance to ​find
out what frequency lives behind your speakers​. For example if you are 1.5 ft from the wall, the
frequency that lives behind your speakers is four times that (because the frequency that could cause
a null is a quarter wavelength, as shown in the illustration above).

So eight feet leads to a frequency of around 750 hz and 750 hz is the villain frequency that lives
behind your speakers in that case. A thin absorption panel behind can take care of that issue quite
easily as the frequency is higher. The other way is to have your speakers quite far away from the
walls (but this is only possible if you have enough space).
 Chapter 5: Reverberation and Clear Speech Standards

We all may know that the effect of sound lingering after the source has stopped generating sound, is
called reverb. Reverb happens when sound reflects off other surfaces and reaches our ears at
different times. We have already discussed the resonant effect of reflections (room modes); reverb is
yet another side effect of reflections.

Human ear is able to intelligently interpret ​Phonemes​ (basic unit of intelligent speech) within 25 to 60
milliseconds. Meaning sounds must be at least 25 milliseconds apart for the ear to differentiate the
sounds separately. Anything before that is seen as uni-source and unidirectional (coming from a
single source from a single direction).

Before the 25 millisecond barrier, the precedence effect or ​Haas Effect​ is observed, that is, if two
similar sounds arrive within 25 milliseconds of each other, the ear will not be able to tell the sounds
apart and the direction of the sound will appear to come from the sound that arrived first, even if the
sound that arrived later was relatively much louder than the previous sound. In most cases, ​this
effect is considered a bad thing for speech intelligibility​, however this phenomenon can be exploited
to create a richer depth in sound mixes where a little intelligibility can be sacrificed for the sake of
musical and acoustic nuance (but that is a topic for another discussion).

Coming back to the basics of reverberation and clear speech:

Contrary to phonemes, ​syllables​ can take up to 200 milliseconds for the ear and brain to
decipher​. In theory, a clear speech broadcast must be at least 25 milliseconds apart and must not
linger past 200 milliseconds. (Hence the Telecommunication Union and European Broadcast
standard for 200 ms RT60 reverb time standards for broadcasting control rooms).

The ​International Telecommunication Union (ITU)​ standards are often extended to music rooms and
mixing rooms. Most mixing control rooms target to have an RT60 Reverb time of under 200
milliseconds.
RT60 Reverb time​ is called the time for a sound signal to decay by 60 db in a large room after the
sound stops. While RT60 is mostly relevant in larger rooms, RT60 standards are still often extended
to smaller rooms. In smaller rooms early reflections within 10 to 15 db matter more than the overall
reverberant field of sound of 60 db. ​Which is why smaller rooms focus more on treatment of early
reflections and larger rooms focus more on critical distances and the overall reverberant field.

A bit more about RT60:

Wallace Clement Sabine, a Harvard based acoustic engineer of the Boston Symphony Hall
established a relationship between the reverb time RT60, the volume of the room and total surface
absorption of the room.

Today we deal with absorption in units of ​Sabin​.

Absorption of a surface in Sabins equals the area times the absorption coefficient of the surface.

Each building material has a sound absorption coefficient measured from 0 to 1 Sabins per sq ft in
most cases ignoring sound diffraction from edges (another topic of discussion).

For example a solid wooden door has an absorption coefficient of 0.14 at a frequency of 125 hz.

The formula empirically derived by Sabine is:

RT60 = 0.049 x Volume of Room in cubic feet / total absorption of the room in Sabins

Total absorption = sum of Area of each surface multiplied by absorption coefficients.

Octave Bands and Absorption:

In acoustics, we always deal with sounds in "Octave Bands"

31.5 hz, 63 hz, 125 hz, 250 hz, 500 hz, 1 khz, 2 khz, 4 khz, 8 khz, 16 khz

Each frequency has a different reverberation time, this is due to the fact that each frequency has
different wavelengths and different travel times as discussed previously. Ideally, according to
broadcast standards, the goal is to achieve similar reverberation times in each frequency band and
in a perfect world, around 200 ms or 0.2 seconds.
Example:

Let us analyze a brick and mortar room with a concrete floor with dimensions:

14 ft x 14 ft x 10 ft.

The room has a volume of 1960 cubic ft, hence the total absorption needed for room to have a
Target RT60 of 0.2 seconds is 412 Sabins.

Total Absorption needed to achieve the broadcast standard will equal to

( 0.049 * 1960 ) / 0.2 = 480 Sabin

Now let’s calculate how much ​actual absorption​ is in the that room:

The absorption coefficient of Brick and Mortar is around 0.1 around 500 hz

The absorption coefficient of concrete floor is around 0.05 around 500 hz

Total Absorption of Concrete floor = 14x14 * 0.05 = 9.8 Sabin

Total Absorption of The Brick and Mortar Ceiling = 14x14 * 0.1 = 19.6 Sabin

Total Absorption of 4 Brick and Mortar Walls = 4x (14x10) * 0.1 = 56 Sabin

Total Absorption of the room = 9.8 + 19.6 + 56 = ​85.4 Sabin

As you can see, ​we are about 400 Sabin Absorption short of achieving our goal of 0.2 second
Reverb time ​which means we need to find absorptive material to cover for about 400 Sabin of
Absorption. This is in addition to dealing with room modes. So ideally you want to find absorptive
material closer to your problem room modes while still covering about 400 Sabin worth of absorption
in the above mentioned room.
Reverb in small rooms

In smaller rooms we deal with early reflections, within 15 milliseconds to 25 milliseconds, to stop
comb filtering we discussed earlier. Please be advised that this is not the same as flutter echo as
commonly misunderstood. A flutter echo is separated usually by more than 60 milliseconds; it is a
strong delayed reflection between two parallel walls. Flutter echoes tend to go away as we treat
smaller rooms for early reflections but the treatment itself does not focus on flutter echoes; it focuses
on comb filtering. Broadcasting Union suggests that a signal should decay by 10 db (in some cases
15 db) within the first 15 milliseconds for certain frequency ranges.

The mirror trick:

To find early reflection zones, put a mirror on each side of the wall and ceiling and if you see your
speakers in the mirrors from your listening position, that will be your point of first reflection and the
first to be treated. Absorptive or scattering material is usually required at this location. With
absorption we target for sound level to drop by 10 db or more and for scattering, the goal is to make
the sound travel 17 ft or more before reaching our ears.

Why 17 ft? Its simple, divide 17 ft by speed of sound 1130 and you get 15 milliseconds (the
broadcast recommendation).

Those are some guidelines at a very high level. Of course, it depends on the purpose of the room. If
you want a room to sound “reverby”, you can control how much absorption you add so you can get
your target reverb times. This is only recommended for tracking instruments and not necessarily for
control room mixing. You dont want natural reverb times of the room messing around with the
reverbs you added with the plugins tarnishing your sound image. For control rooms, the reverb times
should be rather tamed to below 200 milliseconds as recommended by the broadcasting standards.
 Chapter 6: Sound Diffusion

The blaring question - Is Diffusion necessary?

There is some debate around diffusion; some say diffusion is a must and some say it is a luxury and
not needed in most cases. While there is some ambiguity surrounding the issue, due to the
confusion between diffusion and scattering, there is no doubt that diffusion is indeed a very effective
and proven way to liven up a room without losing accuracy. Diffusion was first effectively
demonstrated by a German physicist Manfred R. Schroeder.

Diffusion can be used alongside absorption to treat reflective rooms without removing too much
sound energy; the result is a more diffusive and lively room over a drier or dead room. Diffusion
allows high frequencies to remain in space longer compared to absorption. If the room is overtreated
with absorption, too many high frequencies could get absorbed, resulting in a dead, boomy and dry
room. Diffusion can also help minimize flutter echoes in smaller rooms. In theory, a room needs a
combination of both diffusion and absorption.

Diffusion Vs Scattering

Difference between Diffusion and Scattering is often under misunderstood. Diffusion and Scattering
are remotely related, as in they both involve use of reflective surfaces, but they are also conceptually
very different. Diffusion is the process by which sound intensity and sound pressure is eviscerated
evenly. In other words, diffusion is a controlled capture and re-release of sound energy. ​Diffusion
operates on principles of Phase shift ​and Boundary interference (we discussed earlier) causing
waves in different phases from adjacent point sources to mingle and create a normalized spread of
sound pressure over an area. This is also known as Huygen’s principle of diffusion.

The same principle is also used by concert array based PA systems, where speakers are stacked on
top of each other or to the side. The array of speakers is able to control the sound energy released;
they are able to diffuse sound energy evenly in the concert hall by creating managed phase shifts
between speakers.

Scattering​, like diffusion, also results in a spread of sound energy, however it ​operates on principles
of reflection from hard surfaces​ as opposed to temporal phase shifting in diffusion. In scattering
there is no phase change as sound waves do not change phases after reflection from hard surfaces.
Scattering is a result of such reflections from flat angled surfaces and may also include diffractions
from edges. While both methods, diffusion and scattering spread energy around. The key difference
is that in scattering, sound just bounces off in different directions with similar intensity without any
phase changes. Whereas diffusion, there is an actual redistribution and re-release of sound pressure
as a direct result of phase cancellations.

Scattering relies on inverse square law for reflections to diminish over time. Scattering can be
unpredictable in many cases, and may also create unpredictable room modes if scattering surfaces
are large enough. Diffusion on the other hand has a controlled and predictable response.

So when to use Scattering and when to use Diffusion?

Scattering in other words can be seen as stronger (shinier) and diffusion can be seen as softer
(warmer). While both have their own purposes, some products can do both scattering and diffusion,
simultaneously (more on this later).

Scattering is used in situations where more shine and sparkle are needed on the high ends. For
example, tracking rooms can benefit greatly from controlled scattering. Diffusion is used where
environment needs to be softer, controlled and predictable. For example, mixing control rooms,
home theatres and other listening rooms.

Diffusion can be used in both tracking rooms and mixing rooms, however strong scattering solutions
are not recommended for mixing rooms.
Basic QRD Diffuser

In the scope of this article, we will discuss a type of diffuser called the Quadratic Residue Diffuser
(QRD Diffuser originally developed by Robert Schroeder) also commonly known as a quadratic
diffuser.

We have established that diffusion operates on phase changes, so how exactly does a QRD Diffuser
do that? Let's find out!

Example:

Let's take a foot long solid wooden strip about 2 inch thick and a foot long hollow wooden strip of,
let’s say, 1 inch deep, 2 inch thick and glue them next to each other as shown. What you will get is a
very basic N2 diffuser pattern.
How big does a diffuser have to be?

Due to size restrictions, diffusers usually target low-mids to mid-high frequency spectrums. To
diffuse low frequencies, the diffuser itself would have to be very large as low frequencies have very
long wavelengths.

If an obstacle is not around the size of the wavelength of sound at a particular frequency, sound
wave would act as if the obstacle wasn’t there, as in, it would reflect back from the wall ignoring the
three dimensional geometry of the diffuser. The diffuser has to be at least as large as the
wavelength of its lowest target frequency. To target really high frequencies, the geometrical
elements on the diffuser would have to be very small; borderline microscopic sizes and narrow
region acoustics would take effect. Narrow region acoustics is another big area of discussion. In
most cases a diffuser is a few feet wide and a few inches deep.

How does a diffuser work?

Diffuser relies on hollow cavities (wells) to create a phase shift. When sound of certain frequencies
enter these wells, they get reflected back with a delay (a temporal phase shift) and when they mix
with the signal that is in phase (as the one reflected from the top of the solid strip or from a different
depth) it creates a phase cancellation scenario. When these two adjacent wave fronts come out of a
point source ​(i.e. a slit-like opening)​, the phase cancellation creates a new pressure distribution (a
wider, combined disk-like 180 degree spread as speculated by the ​Huygen’s principle​).

Each point source on the previous wavefront (wavelets) becomes source of another wavefront in
that direction.The result is that the sound feels evenly dispersed in all directions in the nearby area.
How is phase cancellation a good thing in this case?

The phase cancellation scenario​ described above resembles the speaker boundary interference
condition we discussed earlier. Sound that gets reflected from the back of the speaker and mixes
with the main signal of the speaker and whatever frequency has the same wavelength as four times
the distance of the speaker from the wall ( quarter wavelength rule ) gets cancelled or phased out.
Except in this case, it is a good thing as it is ​happening to higher frequencies and not at the listening
position​. This time it is happening at the walls where diffusion or attenuation is needed. This is a
more controlled phase cancellation.

In a diffuser, these ​wells (cavities) are engineered at specifically quarter wavelength depths​,
targeting particular frequency ranges.
Quarter Wavelength Significance

You may have heard the term quarter wavelength thrown around during acoustic discussions. What
does it really mean and how is quarter wavelength significant here?

Quarter wavelength is important in wave physics as a wave has maximum amplitude at quarter
wavelength from its starting position and minimum amplitude (0) at half wavelength. Considering that
a wave started at 0 degrees (origin). Quarter wavelength is also the 90 degree (pi by 2) marker on
the timeline and Half wavelength is 180 degree (pi) marker.

So what is so important about reflections from the quarter wavelength marker?

When a wave reflects back from a quarter wavelength distance, it has traveled exactly 180 degrees
(90 degree round trip!). It is delayed and temporally out of phase by 180 degrees when compared to
the non reflected wave. The comparison is important because the wave itself does not change
phase upon reflection compared to itself, but when compared to another non delayed wave, it is 180
degrees out of phase ( exactly opposite waveforms). When these two waves mingle, they are out of
phase by 180 degrees causing a complete phase cancellation (a null). While for waves to be
completely out of phase is the extreme case scenario that usually defines the lower and upper limits
of operations of the diffuser, partial phase cancellation still occurs for other frequencies in the
operating range of the diffuser.
Design of a N7 Quadratic Residue Diffuser

Now that we understand the operating principle of a well diffuser, let us find out how we calculate
these depths. The depths of the wells of a QRD Diffuser is calculated from a symmetric quadratic
function proposed by Schroeder.

Where x is the position of the well from left to right starting from 0.

N is a prime number

% is the remainder operator (or mod)

The calculation itself is quite basic, divide x square by a prime number and the ​remainder​ (​hence
the name Quadratic Residue​) is the depth, however, the beauty of the function is in the choice of
the function itself. Choice of this function by Schroeder makes perfect sense, it gives a scalable and
symmetrical pattern on the X axis and it gives results that are very close to the wavelength of
frequencies of the human ear’s sensitivities.

Since we are striving for symmetry here, as in the symmetry of diffusion that leads to a symmetrical
dispersion of sound energy, this function is very apt and yet very simple.
Let us do some basic calculations for a N=7 QRD Diffuser, also called QRD N7

For position Position 0, the remainder is 0

Position 1, the remainder is 1 inch (translates to a 1 inch deep well in fig below)

Position 2, the remainder is 4 (translates to a 4 inch deep well in fig below… etc.)

Position 3, the remainder is 2 (3^2 divided by 7, leaves a remainder of 2)

Position 4, the remainder is 2 (4^2 divided by 7, leaves a remainder of 2 again)

Position 5, the remainder is 4

Position 6, the remainder is 1

Position 7, the remainder is 0

Position 8, the remainder is 1

Position 9, the remainder is 4

…. And so forth

As you can see the sequence is quite predictable and symmetric,

Let us make a QRD diffuser based on the above pattern (Fig below shows a top cross section of a
common QRD N7 diffuser)

(See Illustration below)

The deepest well​ (in our case 4 inches) dictates the lowest operating frequency that it can diffuse.
Applying the quarter wavelength rule, that would mean a wavelength of 16 inches (1.33 ft) total.
Dividing the speed of sound by 1.33 we get 845 hz. Which means that the diffuser will work on
frequencies above 845 hz

The shallowest well​ (in our case 1 inch) dictates the highest operating frequency that it can diffuse.
Applying the quarter wavelength rule again, we get 3390 hz.

The total width and height​ (not shown in the cross section above) dictates the low cut-off
frequency of the diffuser. Since our diffuser is 18” wide by 18” tall, it should be able to catch
frequencies that are shorter than the wavelength of 18 inches (1.5 ft). In this case that would be
1130 divided by 1.5, that is 753 hz. It is important that this number is less than or equal to the lowest
operating frequency of the diffuser because if the diffuser is not big enough, it might not be able to
trap the low operating frequency, which would defeat its purpose.

The width of the well,​ in our case two inches, dictates what the highest frequency is that can get
inside the wells and not be absorbed. Usually narrower openings are better but this number is a
tricky one because if it is too narrow ​then narrow region acoustics kick in​ and the signal will get
absorbed or attenuated (converted to heat via viscous drag). In that case it will act like an absorber
for high frequencies which is not good for a diffuser.

Narrow region acoustics is yet another large area of discussion, especially relevant in jet engine
design and other heavy electrical designs. The width of the opening is usually half the wavelength of
the highest operating frequency but can be engineered to be narrower with special care for reasons
explained above.
If the width of the well is too wide then frequencies will not exhibit Huygen’s phenomenon of
diffusion​, as wave sources have to be adjacent “point sources” (narrow openings or slits) for the
diffuser to work correctly. If too wide, frequencies would just reflect back without much effect.

What is the significance of Fins and Fin thickness of the Diffuser?

Fins are required to build the diffuser. They are the planks that are attached vertically to the
horizontal board to create the wells (cavities) of the diffuser. Without those planks there would be no
diffuser, there would only be a board. The only consideration is that the fin should not be very thick
(wide) or the top edge of the fin blade would act like a reflector. It needs to be a thin plank so that the
top edge would not reflect the target frequencies for the diffuser.
They are normally 0.6 - 0.7 inch wide (only a wave of 22k hz or higher would reflect off that edge).
Goal is to catch the wave in the wells; not scatter them from the top of the fins.

That is most of the fundamentals of diffusion. The diffusers are designed to be scalable, as in, they
can be connected, extended and used in either dimensions- horizontal or vertical. They can also be
connected with other larger, more complex diffusers as in N23 QRD Skyline diffuser, etc. Overall,
diffusion is a good solution to eliminate some acoustic problems in mixing rooms without absorbing
too much sound energy.
 Chapter 7: Diffusion Vs Scattering

Diffusion works on phase cancellations and Scattering works on reflections from hard angled
surfaces (there is no phase cancellations that happen on the walls in scattering). There is no
“theoretical diffusion” happening as a ​byproduct​ of scattering; however, some audio enthusiasts tend
use the non-technical definition of “diffusion” as ‘spreading around (dispersion)’, so in a
non-technical way, ​there is sound energy dispersion happening in both cases​.

If both methods spread sound around, why or how do we pick one over another?
Diffusion and Scatting both have similar yet very different applications. I will however, explain what
effect each of them have on a room and when or why one might need one over the other.

Scattering is usually stronger (higher in energy levels) than diffusion because reflections that happen
from hard scattering surfaces are in phase and carry similar energy levels as the incident waves.
After spreading them around, scattering relies just on the inverse square law to diminish the
reflections over time, so the high frequencies ring a bit longer compared to diffusion - which has
both, phasing and inverse square law working for it. In diffusion, phase cancellations happen right at
the walls. Sonically, diffusion creates a “reverse soft-knee” kind of effect where the db levels are
brought down gently like a gradient while sound still disperses around evenly (Huygens).

It can also be understood as Scattering is “shinier, sparklier” and diffusion is “softer, warmer”.

Another important factor is that ​Diffusion is predictable, it has a guaranteed operation on every
frequency in its frequency band​ and the result is also a perfect 180 degree even dispersion
regardless of angles of incidences. Diffusers are also very easy to build compared to predictable
Scattering solutions that can run up the cost quite a bit. Cheaper diy solutions like popcorn walls and
ceilings or random stuff stuck to the walls are a waste of time and aesthetics for a professional
mixing control room.
To get scattering to be predictable, requires a lot of engineering work as the surface curvatures and
sizes would need to be carefully designed by engineering methods or some surfaces might not
deflect at proper angles, they might have an unbalanced dependency on the angles of incidence,
they could also have size disparities that might create holes in frequency responses. The worst of
all, an improper scattering surface ​might not have an even, controlled dispersion​ which is never an
issue in diffusion.

So what is the advantage of scattering?

Aside from the unique sonic character of scattering​, Scattering solutions (even poorly designed,
cheap ones) could be quite beneficial in tracking rooms​ but a poorly designed or incorrectly done
scattering solution can cause problems in mixing rooms. Good scattering solutions are not cheap.
Diffusion is mostly used in mixing rooms for more predictability, but they can be interchanged
wherever necessary.

How effective are Reflective curved walls as a substitute for scattering or diffusing solutions?

My opinion on them is, If you already have them, great, most solutions can work with it; but if you
don’t have them, building them might not be the most efficient use of time and budget unless they
are engineered with proper curvatures based on your room size. You are better off not building it
without proper specifications as it could create gaps in your room modes, only in this case you won't
know where (or what frequencies) they are without an exhaustive test. The budget required to build
an structurally safe, aesthetically pleasing and effectively engineered walls is going to go beyond
what other effective solutions might be able to offer.

As mentioned before, Ideally all curved or splayed surfaces inside a professional mixing room should
be treated with proper materials. Exposed glazed concrete or other shiny angled or curved surfaces
can create unpredictable problems.
Are there any DIY (Do it yourself) solutions to Acoustic problems?

Common sense is a very powerful tool when it comes to acoustics for musical purposes. For
engineering purposes, there are no DIY solutions. In engineering, there is a need for proven
methods, for example we cannot use makeshift solutions inside a Jet Engine unless it is an
emergency. Design of intensive care units also need solutions that adhere to guidelines; music
industry however, is quite forgiving.

In the Music industry, there are quite a few less than ideal solutions that have been used effectively;
by effectively, I mean that the final result was aesthetically pleasing.

I have seen aspiring artists make vocal booths out of PVC pipes and moving blankets. I have also
seen dense foam mattresses and memory foam pillows used creatively to create absorption. I have
seen, partially cooked brick walls and heavy bags of clay and sand used creatively and I have seen
music recorded inside a car, generate millions of views and plays.

Are these solutions effective? - maybe, maybe not but a good ear will always go a long way.

My advice to DIYers is to think simpler and to not get carried away. Don’t waste your time building
random things, focus on the basics and always use your ear and hopefully have another person with
a good ear, judge before and after changes. In the music industry, there is no way to guarantee that
your song will be enjoyed by the masses based on the acoustic treatment involved.

In the end, if one is truly serious about their art, they will take care of it and acoustics is the soul of
music that deserves that attention.

***********