Lesson 1: Types of Microphones based on design

Reading time: 15 min

Learning outcome: You will learn the design and construction as well as types of
microphones used in recording audio.

Introduction

A microphone (frequently called a mic) is typically the principal gadget in an audio signal
chain. Basically, a mic is a transducer that transforms one type of energy (sound waves) into
another related type of energy (electrical signs). The nature of its pickup will regularly rely
upon outer factors, (for example, arrangement, separation, instrument and the acoustic
condition), just as on structure factors, (for example, the mic's working kind, attributes and
quality). These interrelated components will in general work together to influence the general
sound quality.

So as to manage the wide scope of melodic, acoustic, and situational circumstances that may
come to your direction (also your very own taste), an enormous number of mic types, styles
and plans can be pulled out of our "sonic tool kit." Because the specific attributes of a mic
may be most appropriate to a particular scope of utilizations, architects and makers utilize
their masterful gifts to get the most ideal sound from an acoustic source via cautiously picking
the mic or mics that fit the particular pickup application close by.

Despite the fact that it's quickly turning out to be something of an under-appreciated skill,
the determination and utilization of good quality mics will never be totally surrendered as
long as there are acoustic instruments and human voices to be recorded. Without making
light of the job of instruments like the synth and the drum machine - the two of which have
had a significant influence in empowering artists to move away from customary miking
strategies as methods for catching the sound of 'genuine' instruments - fundamental amplifier
systems are as yet a significant piece of the lives of most studio experts. Indeed, even
individuals who don't utilize an acoustic instrument frequently end up needing a mic.

Examining, for instance; probably the most innovative samplists around do practically the
entirety of their work with a convenient DAT machine and great quality mic. Also, obviously,
in case you're going to utilize any sort of vocals other than examined ones, you're probably
going to wind up needing a sensible mic and some information on where to put it (...so to
talk).

Luckily, there's nothing especially complex about mic innovation and, despite the fact that
there is a huge scope of models, for most viable purposes these separate into three or four
fundamental sorts, with maybe four diverse pickup reactions - and this last trademark is the
one we'll take a gander from the start.

Now remember there is this thumb rule to follow -

Good Musician + Good Instrument + Good Performance + Good Acoustics + Good Mic + Good
Placement = Good Sound!

The Design
Based on the design, we shall differentiate the microphones into three main categories
namely -

1. DYNAMIC MICROPHONES
2. CONDENSER MICROPHONES
3. RIBBON MICROPHONES
Dynamic Microphones

On a fundamental level, the dynamic mic works by utilizing electromagnetic induction to

produce a yield signal. The straight forward hypothesis of electromagnetic induction
expresses that at whatever point an electrically conductive metal cuts over the transition lines
of a magnetic field, a flow of electrons will be created inside that metal.

Dynamic mic structures, for the most part, comprise a hardened Mylar diaphragm of generally
0.35-mil thickness. Joined to the diaphragm is a finely wrapped centre of wire (called a voice
coil) that is definitely suspended inside an elevated level inductive field.

At whatever point an acoustic wave hits the diaphragm's face, the connected voice coil is
uprooted in relation to the abundancy and recurrence of the wave, making the loop cut over
the lines of inductive motion that is provided by a perpetual magnet. In doing as such, an
identical electrical signal is prompted into the coil and over the drives, thus delivering a simple
sound yield signal.

( P.C - agilemusicproject.com )

Condenser Microphone

Condenser mics work on an electrostatic rule instead of the electromagnetic principle utilized
by a dynamic or ribbon mic. The case of a fundamental condenser mic comprises two plates:
one dainty moving diaphragm and one fixed back-plate. These two plates structure a
capacitor (or condenser, as it is still called in the UK and in numerous parts of the world). A
capacitor is an electrical gadget that is fit for storing an electrical charge. The measure of
charge that a capacitor can store is dictated by its capacitance value and the voltage that is
applied to it, as per the equation:
 Q = CV

where Q is the charge (in coulombs), C is the capacitance (in faraday), and V is the voltage (in
volts).

At its most essential level, a condenser mic works when a directed DC power supply is applied
between its plates to make a capacitive charge. At the point when sound follows up on the
diaphragm, the fluctuating separation between the plates will similarly make an adjustment
in the mic’ capacitance. According to the above condition, if Q (the charge) is steady and C
(the capacitance) changes, at that point V (voltage over the diaphragm), will change in a
corresponding and in inverse manner.

( P.C - agilemusicproject.com )

Phantom Power

Most present-day proficient condenser (and some lace) mics don't require inward batteries,
outer battery packs or individual AC power supplies so as to work.

Rather, they are intended to be controlled legitimately from the support using a phantom
power supply. Phantom power works by providing a positive DC supply voltage of +48 V
through both sound conductors (pins 2 and 3) of a fair mic line to the condenser container
and preamp. This voltage is similarly dispersed through indistinguishable value resistors, so
no differential exists between the two leads. The −48-V side of the circuit is provided to the
case and preamp through the link's establishing wire (pin 1). Since the sound is just influenced
by potential contrasts between pins 2 and 3 (and not the ground signal on pin 1), the
painstakingly coordinated +48-V potential at these leads is in this manner not electrically
"noticeable" to the information phase of a fair mic preamp. Rather, just the descent,
substituting sound sign that is as a rule at the same time conveyed along the two sound leads
will be distinguished.

Ribbon Microphone

Like the dynamic microphone, the ribbon mic also works on the principle of electromagnetic
induction. Older ribbon design types, however, use a diaphragm of extremely thin aluminium
ribbon (2 microns). Often, this diaphragm is corrugated along its width and is suspended
within a strong field of magnetic flux. Sound-pressure variations between the front and the
back of the diaphragm cause it to move and cut across these flux lines, inducing a current into
the ribbon that’s proportional to the amplitude and frequency of the acoustic waveform.
Because the ribbon generates a small output signal (when compared to the larger output
that’s generated by the multiple wires turns of a moving coil), its output signal is too low to
drive a microphone input stage directly; thus, a step-up transformer must be used to boost
the output signal and impedance to an acceptable range.

Another relatively recent advance in ribbon technology has been the development of the
printed ribbon mic. In principle, the printed ribbon operates in precisely the same manner as
the conventional ribbon pickup; however, the rugged diaphragm is made from a polyester
film that has a spiral aluminium ribbon printed onto it. Ring magnets are then placed at the
diaphragm’s front and back, thereby creating a wash of magnetic flux that makes the
electromagnetic induction process possible.

( P.C - agilemusicproject.com )

Other alterations to traditional ribbon technology make use of phantom power to supply
power to an active, internal amplifier, so as to boost the mic’s output to that of a dynamic or
condenser mic, without the need for a passive transformer (an explanation of phantom power
can be found in the next section on condenser mics).
Questions

1. Explain Dynamic microphone.

2. Explain Condenser microphone.

3. Explain Ribbon microphone.

Next Lesson

In the next lesson, you will learn about various microphones based on their directional
properties.

Lesson 2: Types Of Microphones Based On Directional Properties

Reading time: 10 min

Learning outcome: You will learn the types of microphones based on its directional
properties.

Microphones fall into a few primary categories, as per their directional qualities. Their field
designs are best pictured on a polar chart, a kind of diagram where the yield in various
directions is shown by lines connecting points around the focal point of the diaphragm where
the yield is constant.

Omnidirectional Microphones

These preferably, react similarly to sounds originating from all directions. Fundamentally,
they are gadgets for estimating the weight of the air and changing over it into an electrical
signal. The most straightforward structures of coil and condenser small scale telephone work
right now. The diaphragm is available to the air on one side as it were. This kind of activity is
likewise normally called 'dynamic'.

Bidirectional Microphones

These measures the distinction in (pressure inclination) between the different sides of a
diaphragm, thus at two progressive focuses along the way of a sound wave drawing nearer
from the front or back. In the event that the mic is put sideways to the way of the sound, the
weight is consistently the equivalent on the two sides of the stomach, so no electrical sign is
created. The amplifier is in this manner dead to sound from the side and live to the front and
back.

Moving from the front to the side of the mic, the reaction turns out to be continuously lesser
at that point rises once more, and a diagram of the yield resembles a figure-of-eight or figure-
eight (minor departure from a term now and then used to depict this sort of amplifier). The
live edge is by and large viewed as being about 100 on each face. For sound arriving at the
receiver from the back, the electrical yield is like that for sound at the front, yet is actually
inverse in phase. The least complex structures of strip mouthpiece utilize this rule. They react
to the distinction in pressure on the two essences of a piece of aluminium foil. Condenser
figure-of-eight mics have two diaphragms, one on either side of a charged backplate.

Unidirectional Microphones

Cardioids (as the name suggests) have a heart-shaped response. This is obtained if the
electrical yield of a pressure - outcome worked amplifier is added to that of a pressure slope
mic with a response of comparable quality along with its forward pivot. The yield is multiplied
at the front, tumbles to that of the omnidirectional amplifier all alone at 90C to either side
and afterward to zero at the back, where the two are antiphase and offset. Different methods
for joining the two principles are conceivable: early cardioid mics really contained a strip and
a moving loop inside a solitary case. There was a generous signal of sound on the front and
some round to the side, yet continuously less past that.

Supercardioid - In the event that the acoustic pressure and its variants methods of activity
are blended in changing extents, a scope of polar outlines is delivered, going from
omnidirectional through cardioid, passing through supercardioid to hypercardioid, until lastly
resulting in bidirectional. The directional characteristics inside this range might be depicted
in two different ways. One is by the level of unidirectional reaction, demonstrating the
proportion of sound acknowledged at the front and back of the amplifier (for example
forward and behind a plane at right points to its directional pivot). This is 1:1 for
omnidirectional and bidirectional reactions (the two boundaries) and most extreme, arriving
at a pinnacle proportion of 13:1, at the middle of the road. Another characteristic of a
mouthpiece's directivity is its victimization circuitous sound: practically speaking, this also is
communicated as a proportion, here portraying the extent of the all-out strong point over
which the amplifier is adequately touchy to sound. Again it is 1:1 for omnidirectional
receivers; it ascends to 3:1 (resonation is 4.8 dB down) for both cardioid and bidirectional get
and is higher (somewhat over 4:1 with 6 dB dismissal of reverberation) for a reaction
somewhere between cardioid and bidirectional.

The blend of the Acoustic Pressure and its variance standards in various extents is helpful for
cutting the measure of reverberation and for dismissing undesirable commotions. A few mics
can be changed to various polar outlines; others are intended for one specific example of a
reaction. While picking between various field designs, note that the broadness of getting
limits as the weight angle part gets more grounded. A hypercardioid (additionally called cabin
portion) design is generally clear in the range where the victimization encompassing sound is
at its most noteworthy; the term 'unidirectional' might be applied to the range between
cardioid and hypercardioid.
 ( P.C - hookeaudio.com )

Questions

1. Explain different types of polar patterns

2. Explain Omnidirectional, Bidirectional and Unidirectional microphones.

Next Lesson

In the next lesson, you will learn about the properties of microphones.

Lesson 3: Microphone Properties

Reading Time: 2 min

Learning Outcome: In this lesson, you will learn about the different properties of
microphones.

Frequency Response

Down to earth, top-notch proficient mics don't have a reaction that is splendidly level. In more
established receivers pinnacles and troughs of up to 34 dB were normal. This was not simply
terrible work-manship: at one time it was hard to plan a mic for a level reaction at all audible
frequencies, and even today some deviations might be useful. Also, an individual example
may differ by 2 dB or so from the normal for its sort. Luckily, hardly any individuals notice
varieties of this request: in a reverberant equalization, they are effectively differentiated by
varieties because of the acoustics.

A few mics have a reaction that falls away in the bass to make them inadmissible for specific
symphonic instruments (and absolutely for a full symphony). Considerably more typical is a
loss of top, from around 12 kHz on certain receivers. Others have a high-recurrence reaction
that works out in a good way past this, however, with these the edge at which the receiver is
utilized might be basic: for a few, to acquire the evenest reaction the hub of the diaphragm
should point legitimately at the sound source.

Proximity Effect

The pressure-gradient mode of operation, by its very nature, produces a distortion of the
frequency response called proximity effect or bass tip-up: when placed close to a source the
microphone exaggerates its bass. The distance at which this begins to set in depends on the
difference in path length as the signal reaches first the front and then the rear of the
diaphragm. Bass tip-up occurs when the sound source is close enough for the path difference
to be a significant proportion of that distance and is due to the fact that, as it radiates
outward, the sound dies away. For spherical waves, it diminishes in proportion to the square
of the distance. It is most noticeable on voices (including song).

Questions

1. List the properties of microphones.

Lesson 1: Driver Design

(Reading Time: 20 mins)

Learning Outcomes:

1. To understand the build and importance of a driver.

2. To understand the different aspects of driver design.

Introduction to Drivers:

Within the recording process, our ability to evaluate and adjust sound is based on what’s
heard through the monitor speakers at a project studio or control room environment. In fact,
within the audio and video industries, the word monitor refers to a device that acts as a
subjective professional standard or reference by which program material can be judged.

Despite steady advances in design, speakers are still one amongst the weakest links within
the audio chain. This weakness is mostly because of potential nonlinearities that may exist
during a speaker system’s frequency response. What's more, connections with a room's
recurrence reaction frequently bring about pinnacles and plunges that may influence a
speaker's sonic character in manners by which are hard to anticipate. Add to this the
components of individual ''tastes'' in the sound, size, and configuration types, and you'll
rapidly find that speakers are likewise one of the most devices in a creative studio.

The expression "amplifier" can ask singular transducers (known as "drivers") or to complete
frameworks consisting of a nook consolidating at least one driver. To sufficiently repeat a
decent scope of frequencies, most amplifier frameworks require more than one driver,
especially for high solid weight level or greatest exactness. Individual drivers are accustomed
to reproducing different frequency ranges. The drivers have tagged subwoofers (for very low
frequencies); woofers (low frequencies); mid-range speakers (middle frequencies); tweeters
(high frequencies); and sometimes super tweeters, optimized for the best audible
frequencies.

The terms for different speaker drivers differ, depending on the appliance. In two-way
amplifiers, there is no mid-go driver, that the assignment of duplicating the mid-extend
sounds falls upon the woofer and tweeter. Home sound systems utilize the assignment
"tweeter" for high frequencies, though proficient sound frameworks for shows may assign
high-recurrence drivers as "HF", or "highs", or "horns". At the point when numerous drivers
are used in a framework, a "channel arrange", called a hybrid, isolates the approaching sign
into various recurrence ranges and courses them to the reasonable driver. An amplifier with
n separate recurrence groups is portrayed as "n-way speakers": a two-way framework will
have woofer and tweeter speakers; a three-way framework is either a blend of the woofer,
mid-range, and tweeter, or subwoofer, woofer, and tweeter.

Driver Design:

The most widely recognized kind of driver utilizes a lightweight cone, associated with an
inflexible container, or casing, using a flexible suspension that compels a curl of fine wire to
move pivotally through a tube-shaped attractive hole. At the point when an electrical sign is
applied to the voice loop, an attractive motion is made by the electrical flow inside the voice
curl, making it an electromagnet. The loop and in this way the driver's attractive framework
collaborate, producing a mechanical power that causes the curl (and along these lines, the
connected cone) to move to and fro, subsequently repeating sound heavily influenced by the
applied electrical sign originating from the speaker. Coming up next is a portrayal of the
individual parts of this sort of amplifier.

Fig 1. Cross-Section of a Driver Fig 2. Outline of a Driver

The diaphragm is typically manufactured with a cone- or dome-shaped profile. a spread of

various materials could also be used, but the foremost common are paper, plastic, and metal.
the perfect material would be stiff, to stop uncontrolled cone motions; light, to attenuate
starting force requirements; and well-damped, to scale back vibrations from continuing after
the signal has stopped. By and by, every one of the three of those rules can't be met at the
same time utilizing existing materials; along these lines, driver configuration includes
exchange offs.

For instance, the paper is light and ordinarily all-around damped, however not hardened;
metal is frequently made solid and lightweight, yet it's not normally all-around damped;
plastic is regularly light, yet commonly, the stiffer it's made, the less very much damped it's.

As a result, many cones are made from some kind of material. This will be a matrix of fibres,
including Kevlar or fibreglass; a layered or bonded sandwich construction; or just a coating
applied to stiffen or damp a cone.

Fig 3. Chassis (frame) of a 15 inch Driver

The crate, or edge, must be intended for unbending nature to stay away from disfigurement,
which may change the attractive conditions inside the magnet hole and will even cause the
voice curl to rub against the dividers of the hole. Containers are normally thrown or stepped
metal, albeit shaped plastic bushels are getting normal, particularly for modest drivers. The
edge likewise assumes an extensive job in leading warmth away from the loop.

The suspension keeps the curl focused inside the hole and gives a reestablishing power that
causes the speaker cone to come back to an unbiased situation in the wake of moving. A run
of the mill suspension comprises of two sections: the "creepy-crawly", which associates the
stomach or voice curl to the edge and gives the main part of the reestablishing power, and
the "encompass", which helps focus the loop/cone get together and permits free cylinder-
like movement lined up with the attractive hole. The spider is typically made from a
corrugated fabric disk, generally with a coating of cloth intended to enhance mechanical
properties. The name originates from the type of early suspensions, which were two
concentric rings of Bakelite material, joined by six or eight bent "legs". Varieties of this
topology included adding a felt circle to supply an obstruction to particles which may
somehow cause the voice curl to rub. A German company, Rulik, still offers a spider made
from wood. The choice of suspension materials affects driver life, especially in the case of
foam surrounds, which are susceptible to ageing and environmental damage.

Present-day driver magnets are quite often perpetual and made of artistic, ferrite, Alnico, or,
all the more as of late, neodymium magnets. A pattern in structure—because of increments
in transportation costs and a craving for littler, lighter gadgets (as in many home theatre
multi-speaker establishments)— is the utilization of neodymium magnets rather than ferrite
types. Very few manufacturers use electrically powered field coils, as was common within the
earliest designs. The size and type of magnet and details of the magnetic circuit differ,
depending on design goals. For example, the state of the shaft piece influences the attractive
cooperation between the voice curl and the attractive field and is in some cases used to
change a driver's conduct. A "shorting ring", or Faraday circle, could likewise be incorporated
as a thin copper top fitted over the post tip or as a significant ring arranged inside the magnet-
shaft cavity. The advantages of this are decreased impedance at high frequencies, giving
expanded treble yield, diminished consonant twisting, and a markdown inside the inductance
regulation that ordinarily goes with enormous voice curl outings. On the contrary hand, the
copper top requires a more extensive voice-loophole, with expanded attractive hesitance;
this decreases accessible motion, requiring a somewhat bigger magnet for proportional
execution.

Driver design—including the actual way two or more drivers are combined in an enclosure to
form a speaker system—is both an art and science. Changing a structure to upgrade execution
is finished utilizing attractive, acoustic, mechanical, electrical, and material science
hypothesis; high exactness estimations; and along these lines the perceptions of experienced
audience members. Planners can utilize space to ensure the speaker is frequently estimated
freely of room impacts, or any of a few electronic methods which may, somewhat, supplant
such chambers. A few designers shun anechoic chambers for explicit normalized room
arrangements expected to reenact genuine listening conditions. A few of the problems
speaker and driver designers must confront are distortion, lobbing, phase effects, off-axis
response, and crossover complications.

Next Lesson:

In the next lesson, we will learn

1. Different types of drivers.
2. How drivers work and specifications.

Questions:

1. What is a driver?
2. What are the different kinds of drivers?
3. How is a two-way system different from a three-way system?
4. List the different parts of a driver.
5. What are the three materials used to manufacture a cone?
6. What is Kevlar?
7. What are the two parts of a suspension?
8. Name the company that produced a wooden spider. Where did they originate from?
9. What are modern driver magnets made off?
 10. Mention a few things that need to be considered before designing a driver.

Suggested viewing:

1. Working of a Driver - https://www.youtube.com/watch?v=048tBZMt3eY

2. Making of a Driver Voice Coil - https://www.youtube.com/watch?v=6qjcIXaE7gc
3. Driver: Manufacturing unit - https://www.youtube.com/watch?v=Z_AHN2Bo78A

Lesson 2: Types of Drivers

(Reading Time: 15 mins)

Learning Outcomes:

1. Understanding different driver types, their specifications and application.

Full-range drivers

A full-range driver is intended to have the most stretched out recurrence reaction
conceivable. These drivers are little, ordinarily, 3 to 8 inches (7.6 to 20 cm) in measurement
to allow sensible high-recurrence reaction, and painstakingly intended to give low-bending
yield at low frequencies, however with decreased most extreme yield level. Full-range (or all
the more precisely, wide-run) drivers are most usually heard to openly address frameworks
and in TVs, albeit a few models are appropriate for greeting Hi-fi tuning in. In Hi-fi speaker
frameworks, the utilization of wide-extend drive units can keep away from the unwanted
association between numerous drivers brought about by non-incidental driver area or hybrid
system issues. Enthusiasts of wide-run driver Hi-fi speaker frameworks guarantee a
cognizance of sound, said to be because of the single source and subsequent absence of
impedance, and likely to the absence of hybrid segments. Spoilers commonly refer to wide-
run drivers' restricted recurrence reaction and unassuming yield capacities, together with
their necessity for enormous, intricate, costly walled areas, for example, transmission lines,
or horns—to move toward ideal execution.

Fig 4. Full Range Driver (Eminence Alpha 5-8)

Full-range drivers often employ an extra cone called a Whizzer: a little, light cone attached to
the joint between the voice coil and thus the first cone. The Whizzer cone expands the high-
frequency response of the drive and widens its high-frequency directivity, which could ideally
be enormously limited as a result of the external breadth cone material neglecting to stay up
with the focal voice curl at higher frequencies. The foremost cone during a Whizzer design is
manufactured so it flexes more within the outer diameter than within the centre. The result
is that the foremost cone delivers low frequencies and thus the Whizzer cone contributes
most of the upper frequencies. Since the Whizzer cone is smaller than the foremost
diaphragm, output dispersion at high frequencies is improved relative to an equivalent single
larger diaphragm.

Limited-range drivers are typically utilized in computers, toys, and clock radios. These drivers
are less detailed and less costly than wide-extend drivers, which they may even be seriously
undermined to fit into little mounting areas. In these applications, the sound quality could
also be a coffee priority. The human ear is amazingly open-minded of poor sound quality, and
along these lines the mutilation characteristic in restricted range drivers may upgrade their
yield at high frequencies, expanding lucidity when tuning in of vocable material.

Subwoofer

A subwoofer could also be a woofer driver used only for the lowest neighbourhood of the
audio spectrum: typically below 120 Hz. Because the intended range of frequencies in these
is restricted, subwoofer system design is usually simpler in many respects than for
conventional loudspeakers, often consisting of 1 speaker enclosed during an appropriate box
or enclosure.

To precisely repeat extremely low bass notes without undesirable resonances (ordinarily
from bureau boards), subwoofer frameworks must be firmly built and appropriately propped;
great ones are commonly remarkably overwhelming. Numerous subwoofer frameworks
incorporate force speakers and electronic sub-channels, with extra controls applicable to low-
frequency generation. These variations are referenced as "dynamic subwoofers".
"Uninvolved" subwoofers require outside intensification.

Fig 5. 4-inch Subwoofer

Woofer

A woofer may be a driver that reproduces low frequencies. Some loudspeaker systems use a
woofer for rock bottom frequencies, making it possible to avoid employing a subwoofer. Also,
a few amplifiers utilize the woofer to deal with centre frequencies, disposing of the mid-run
driver. This will be cultivated with the decision of a tweeter that reacts low enough joined
with a woofer that reacts sufficiently high that the 2 drivers include intelligently inside the
centre frequencies.

Fig 6. Neodymium Full-Range Woofer Driver

Mid-range driver

A mid-range speaker is an amplifier driver that recreates middle frequencies. Mid-range

drivers can be made of paper or composite materials, or they can be pressure drivers. On the
off chance that the mid-range driver is cone-shaped, it tends to be mounted on the front
baffle of an amplifier nook, or it very well may be mounted at the throat of a horn for included
yield level and control of radiation design. If it is a pressure driver, it is constantly mated to a
horn.

Fig 7. Eminence Beta 8A Mid-Range Driver

Tweeter
A tweeter could also be a high-frequency driver that typically reproduces the absolute best
waveband of a loudspeaker. Numerous sorts of tweeter configuration exist, each with
contrasting capacities as to recurrence reaction, yield loyalty, power taking care of, most
extreme yield level, and so forth. Delicate vault tweeters are generally found in-home sound
systems, and horn-stacked pressure drivers are basic in proficient sound fortification. Strip
tweeters have picked up prevalence as of late, as their yield power has been expanded to
levels helpful for proficient sound fortification, and their example control is advantageously
moulded for show sound.

Fig 8. Bullet 4-inch Full-Range Tweeter

Ribbon Tweeter

Ribbon tweeters came to existence during the same time as the ribbon microphones and
shared the same technology. The tweeter is built using similar elements as the traditional
driver, like the style of the monitor, etc. It's usually made up of an ultra-thin metal
diaphragm (made from the deposition of aluminium vapour). It is not too thick and results in
being flexible. This diaphragm is suspended in an electromagnetic field. They are usually
1/10th the weight of a standard dome tweeter and because they are driven evenly over the
whole surface they move in adjacent to the magnet and this is what causes the charge in the
field.

Fig 8.1 Ribbon Tweeter Dome

Coaxial Speakers
They are usually 2 or 3 way loudspeakers, where the tweeter and a mid-range driver is
mounted in front of a woofer. This design is famous due to the use of lesser area, without
compromising the sound wave path of the woofer offering symmetrical response both
horizontally and vertically. This results in a wider sweet spot and more consistency
throughout the room. Time-alignment correction is made in such designs so the sound from
the tweeter doesn’t arrive before the sound from the woofer.

Fig 9.1 Coaxial Speaker

Horn loudspeaker

A horn loudspeaker is a finished amplifier or amplifier component which utilizes a horn to

build the general proficiency of the driving component, commonly a stomach driven by an
electromagnet. The horn itself is an aloof part and doesn't enhance the sound from the
driving component in that capacity, but instead improves the coupling proficiency between
the speaker driver and the air. The horn can be thought of as an "acoustic transformer" that
gives impedance coordinating between the moderately thick stomach material and the
quality of low thickness. The outcome is the more noteworthy acoustic yield from a given
driver. The restricted piece of the horn close to the speaker driver is known as the "throat"
and the huge part most remote away from the speaker driver is known as the "mouth".
 Fig 9.2 Horn loudspeaker with a sealed box driver mounting

Horns have been utilized to broaden the low-frequency cutoff of a speaker driver—when
mated to a horn, a speaker driver can recreate lower tones all the more firmly. The flare rate
and the mouth size decide the low-frequency limit. The throat size is to a greater degree a
structure decision. Horns have been known to expand the frequency scope of a driver past
five octaves.

Horns have been utilized to alter the directional attributes of the delivered sound waves.
Even the inclusion point is the essential determinant of horn width, and vertical inclusion edge
decides horn tallness. Here and there hub execution will vary contingent upon the state of
the horn. Bargains in execution, for example, mutilations of the wavefront must be adjusted
against the plan objective.

Fig 9.3 Horn with transformer

Acoustic horns convert huge weight varieties with a little relocation into a low-pressure
variety with an enormous uprooting and the other way around. It does this through the
continuous, regularly exponential increment of the cross-sectional zone of the horn. The little
cross-sectional region of the throat confines the entry of air along these lines introducing a
high impedance to the driver. This permits the driver to build up a high weight for a given
dislodging. Thusly the sound waves at the throat are of high weight and low relocation. The
decreased state of the horn permits the sound waves to continuously decompress and
increment in removal until they arrive at the mouth where they are of a low weight however
enormous relocation.

Next Lesson:

In the next lesson, we will learn

1. Details of cabinet design.
2. Different types of enclosures.
3. Understanding Infinite Baffle.

Questions:

1. What is a reasonable diameter of a driver?

2. What is Whizzer?
3. Where are limited ranged drivers found?
4. What is a subwoofer? Explain in detail.
5. What is a woofer? Which are the frequencies they handle?
6. What is a tweeter? State its uses.
7. Explain horn loudspeaker in detail.

Suggested viewing:

1. Tweeters, Woofers & Sub Woofers -

https://www.youtube.com/watch?v=jhvXWpRmhvM

Lesson 3: Cabinet Design

(Reading Time: 15 mins)

Learning Outcomes:
1. Understanding different cabinet designs and operating types.
2. To understand differences in cabinet enclosures.
3. Understanding Infinite Baffle.

The sound of a speaker framework will change when heard in various acoustic conditions,
Speakers of various plans and working sort will generally sound altogether different from each
other, in any event, when heard in a similar room. Walled in area size, a given number of parts
and driver size, hybrid frequencies, and structure reasoning contribute incredibly to these
distinctions in sound quality.
 Fig 10. Detailed speaker cabinet design

Professional speaker enclosures, in general, are one of two design types: air suspension and
bass reflex. An air-suspension speaker enclosure (Figure 11) is an airtight system that seals
the air in its interior from the outside environment. This system type (which is usually utilized
in ‘‘bookshelf’’ designs) generally provides a robust, ‘‘tight’’ bass response, while often being
rolled off at the acute low end. The bass-reflex or vented-box design (Figure 12), makes use
of a tuned bass porthole that's designed into the front or rear of the speaker enclosure. This
enables the atmosphere inside the enclosure to combine freely with the surface air in such a
way as to act as a tuned resonator (which serves to acoustically boost the speaker’s output at
the acute lower octaves).
 Fig 11. Air suspension design (Adam T7V) Fig 12. Bass Reflex Design (Adam A8X)

Types of Enclosures

The creation of a decent high-fidelity loudspeaker necessitates that the speakers be encased
because of various essential properties of loudspeakers. Simply placing a solitary powerful
loudspeaker in a shut box will improve its sound quality drastically. Present-day loudspeakers
walled in areas ordinarily include numerous loudspeakers with a hybrid system to give an all
the more uniform recurrence reaction over the sound recurrence extends. Different systems,
for example, those utilized in bass reflex walled in areas might be utilized to broaden the
valuable bass scope of the loudspeakers.
 Fig 13. Different Cabinet Designs

The idea of the enclosure can influence the effectiveness and the directionality of a
loudspeaker. The utilization of horn loudspeakers can give higher productivity and greater
directionality, however, boundaries can lessen the fidelity of the sound. Line arrays can give
some directionality.
 Fig 14. Infinite Baffle

The expression "infinite baffle" is regularly experienced in conversations of loudspeaker

installations. It envisions a loudspeaker mounted in an endless plane with boundless volume
behind it, yet in useful use may allude to a loudspeaker mounted in the outside of a level
divider with a significant volume of air behind it. Due to the flexible properties of the
loudspeaker suspension, it will even now display its regular free-cone reverberation yet will
be liberated from the diffraction impacts saw with a little box speaker, and liberated from the
impacts of the pressure of the air behind the loudspeaker cone.

The motivation behind the loudspeaker fenced in area is to forestall waves produced from
the front and back of a drive unit meeting and consequently offsetting one another. The
conspicuous way to keep the front and back waves from regularly meeting is to broaden the
baffle. Several designs have been made with an open-back or di-pole format; however, the
low-frequency performance on these models is very poor, and would only be suitable for
lead guitar amplification in a PA situation.

Folding the baffle into a full box, thereby completely containing the rear emitted wave is a
much more popular design. This is the method of enclosure employed by the majority of
speakers within a low budget range because they are relatively easy to manufacture,
especially in bulk, are suitable for speakers of virtually any size; and give an agreeable sound
quality. As the baffle has no edges it is commonly referred to as the ‘infinite baffle’. Drive
units selected for use with these enclosures need to be carefully selected, with a weaker or
more compliant cone suspension than others, as the trapped air within the box acts itself as
suspension under the pressure of cone movement. For this reason, this type of arrangement
is also called ‘acoustic suspension’.

However, because of this trapped air behind the cone, the movement of the cone is modified,
resulting in untrue movement of the driver cone. This causes unwanted colouration to the
sound waves emitted from the front of the drive unit- i.e. the one that the listener hears. The
damping effect of the trapped air on the cone decreases compliance and results in the natural
resonant frequency of the enclosure (a point at which the frequency of sound is reinforced
by the synchronous vibration of the cone and enclosure) being reached.
This effect is reduced with larger volumes of the enclosure as a given cone excursion exercises
less compression on a larger air volume than on a smaller one. So larger boxes have lower
resonant frequencies with the infinite baffle enclosure. In the same way, a smaller driver
cone also compresses a given air volume less than a large cone, hence the cone size also
affects the enclosure resonant frequency. The cone mass also affects resonance as smaller
masses have a higher resonant frequency, to begin with. The object should be to have as low
a resonant frequency as possible, preferably outside of human hearing limits (below
20Hz). Hence box size (therefore air volume); cone size and cone mass need to be carefully
selected in this type of enclosure.

Next Lesson:

In the next lesson, we will learn

1. Speaker polarity, monitoring configurations.
2. Near & Far-Field Monitoring and its technicalities.
3. Damping & Distortion.

Questions:

1. Explain the importance of cabinet design.

2. Differentiate Air suspension and Bass Reflex cabinet design.
3. State the different types of cabinet designs.
4. What is an infinite baffle and where are they mounted?
5. Explain the damping effect.

Suggested Viewing:

1. What port size to choose - https://www.youtube.com/watch?v=nxKkM4bXPFI

2. Importance of speaker cabinet - https://www.youtube.com/watch?v=UTqpdUhXDAo
3. Ported and sealed cabinet - https://www.youtube.com/watch?v=dtQdO2rJWH8

Lesson 4: Speaker Polarity, Monitoring and Calibration

(Reading Time: 60 mins)

Learning Outcomes:

1. To understand speaker polarity and its connections.

2. Understanding concepts involved in monitoring such as volume, different
configurations, etc.
3. Understanding different speaker and headphone monitoring types with their
specifications.
4. Damping and distortion specifications.

A typical oversight that can influence the sound of a multi-speaker framework is to wire them
out-of-phase concerning one another. Speaker extremity is said to be electrically in-stage at
whatever point one sign that is similarly applied to the two speakers will make their cones
move in a similar way (either decidedly or adversely). At the point when the speakers are
wired out-of-stage, one speaker cone will move one way while different moves the other
way.

Speaker polarities can be effortlessly tried by applying a mono sign to both or the entirety of
the speakers at a similar level. On the off chance that the sign's picture seems to begin from
a point straightforwardly between the speakers, they have been appropriately wired (in-
phase). On the off chance that the picture is difficult to find and seems to begin past the
external limits of a sound system speaker pair or moves as the audience moves their head,
it's a decent wager that the speakers have been inappropriately wired (out-of-phase).

An out-of-phase speaker condition can be effortlessly amended by checking the speaker wire
polarities. The ‘‘hot’’ lead (þ or red post) leading from each amp channel should be secured
to the same lead on its respective speaker (Figure 15). In like manner, the negative lead (black
post) ought to be associated with its separate lead for every speaker in the framework.
Speaker cable is by and large colour coded, with white or red being positive (+) and black
being negative (-). If no colour-coding is present, heavy-duty power cable or other cabling
types that are suitable for speakers can be used. These cables will frequently have an indented
edge (or set of edges) or have a printed white band that is commonly associated with the
negative lead post. Speaker wire measures ought to consistently be as hardcore as is
conceivable or viable. The #18 wire is viewed as the base for lengths of under 250–500
metres, while #14 is viewed as the base length that ought to be utilized for 500 and 1000
metre runs. (Note: The littler the check number, the thicker the wire; in this way #14 is thicker
than #18.)

Two explanations behind expanding the thickness of the conduit as cable length increments
may be:

Every single cable has opposition, which will increment with length. More slender cables, for
the most part, have more noteworthy opposition esteems, implying that more force will be
disseminated in the cable and will, subsequently, be inaccessible to drive the speaker.
 Fig 15. Banana Plugs connected to an amplifier Fig 16. Banana Plug breakdown

The higher the cable opposition, the lower the speaker's compelling damping factor. Damping
factor is identified with how well the intensifier can control the movement of the speaker
cone. A brought down damping will frequently bring about lost snugness, definition, and
clearness in the low end. Once more, thicker conductors will have less opposition and along
these lines help to limit damping issues.

Monitoring

When mixing, it's significant that the engineer is situated as intently as possible to the centre
of the sound field (considering the producer, artists, and other people who are additionally
putting the effort to be in the ''sweet spot'') and that all the speaker volumes are balanced
similarly. For instance, if the engineer is closer to one speaker than another, that speaker will
sound stronger and might be enticed either to pan the instruments toward the far speaker or
boost that whole side of the blend to equalize the volumes. The resulting mix would sound
properly centred when played in that room, but in another environment, the mix might be
off-centre. As a brisk check against this, the engineer ought to consistently ensure that a
comparing visual distinction accompanies a discernible volume contrast between speakers on
the primary yield VU or show meters. Another protection against topsy turvy levels is to
screen pink clamour (or a test tone signal) from every speaker in the sound field to watch
that they're similarly uproarious (either by doing a snappy discernible check or by putting a
mic in the inside listening position). In the last case, the yield level from every speaker can be
perused and coordinated, utilizing an SPL meter or VU meter on an extra reassuring input.

Monitor Volume
Before continuing, I’d like to revisit another important factor — volume. During the record
and mix downstage, it’s important to keep in mind that the Fletcher-Munson curves will
always have a direct effect on the frequency balance of a mix. Since our ears hear recorded
sound distinctively at different observing levels when checking at uproarious levels, our ears
will effectively see the extraordinarily high and low frequencies in the blend (sounds great
doesn't it?). But, when the blend is played back at lower levels, (for example, over the radio,
TV, or PC), our ears will be substantially less delicate to these frequencies and the bass and
extraordinary highs will presumably be lacking (leaving the blend sounding far off and
dormant). Dissimilar to during the 1970s, when unbearably high SPLs would, in general, be
the standard in many studios, ongoing decades have seen the decrease of screen levels to an
increasingly moderate 75–90 dB SPL. (A decent general guideline is that if you need to yell to
convey in a room, you're checking excessively boisterous.) These moderate levels offer a
decent trade-off for mixing, as they all more precisely speak to listening levels that are
probably going to be experienced in the normal home (i.e., the Fletcher-Munson bends will
be all the more firmly coordinated). Ear weariness and potential ear harm because of delayed
presentation to high SPLs by industry experts can likewise be kept away from at these levels.

Monitor Configuration

Getting the best generally sound out of a mix, another observing worry that has picked up
significance is the need to tailor the mix to the expected room/speaker setup (i.e., mono-
stereo, mono-surround, and stereo-surround). It's imperative to recollect that a huge level
of your potential clients may initially hear your blend over a PC or AM/FM radio in mono.
Hence, if it sounds great in stereo yet poor in mono, it probably won't sell also because it
neglected to consider these media. The same may go for a surround-sound blend of a music
video or highlight discharge film in which legitimate consideration wasn't paid to phase
cancellation issues in mono as well as stereo (or the other way around).

The lesson of this story is this: To forestall potential issues, a blend ought to be painstakingly
checked in the entirety of its output arrangements to guarantee that it sounds great and that
no out-of-phase parts are incorporated that would offset instruments and possibly debase
the balance. The most generally acknowledged speaker setups are mono, stereo, and
surround sound.

Mono

Even in this age, much of the buying public will first experience a mix in monaural (mono)
sound (Figure 17). In other words, they'll hear your song over the radio, on TV, in a lift, on the
PC, and so forth. For these reasons, record companies, producers, and everyone else involved
in the process will often place a great deal of importance on mono compatibility and the
overall sound of a mono mix. Truth be told, it's normal for a different mono mix to be made
to guarantee that it'll sound as great as possible for its proposed medium.
 Fig 17. MONO Configuration

Stereo

Since the time of the practical advancement of the 45°/45° record cutting procedure,
stereophonic (stereo) sound (Figure 15) has managed the turntable. Over the years, stereo
has also grown to govern FM radio, the CD player, and TV. Consequently, the making of a
quality stereo mix is critical with L/R balance, in overall frequency balance, dynamics, depth,
and effect. The mixing environment should be acoustically and physically symmetrical (within
reason) to encodings ensure that the L/R balance and overall imagery is accurate within the
stereo sound field. Apart from this, it's constantly a good idea to check for mono
compatibility. Phase cancellations can make instruments or frequencies in the range just
vanish at whatever point a mix is added to mono. The best devices for decreasing phase
mistakes are acceptable mic procedure, a phase meter or X/Y scope show, and obviously, your
ears.
 Fig 18. STEREO Configuration

Fig 19. Various scopes Lissajous phase shapes

It ought to be noticed that numerous bigger consoles and DAW programs or potentially
modules will regularly incorporate a phase scope show. This readout can be used to show
phase by sending the left channel to the vertical (Y) trace and right to the horizontal (X) trace
inputs (Figure 19).

The subsequent showcase brings about a ''Lissajous'' follow that produces such shapes as:

A slope that rises from left to right at a 45° angle shows an L/R signal, showing that the
combined signal is in-phase.
A slope that rises from right to left at a 45° angle shows an L/R signal, showing that the
combined signal is out-of-phase.

Surround Sound

With the advent of 5.1 surround playback in home and audio ‘‘theatres,’’ surround sound
(Figure 20) has grown into a major professional and consumer entertainment market. The 5.1
name refers to the five, full-range channels (left, centre, right, surround left, and surround
right), plus a sixth sub-bass channel (containing a narrow frequency response of 5–125 Hz).

DVD recordings and sound plates regularly utilize 5.1 encodings as Dolby Digital (a plan that
encodes the discrete 5.1 data into a solitary bitstream, known as AC-3). Most players can
interpret or course this sequential bitstream to an outside decoder in different advanced
surround positions.

Fig 20. Surround sound monitoring configuration

The discussion over how to mix for surround has raised the temperature of numerous panel
gatherings and Web Chat conversations. The individuals who plunge into these conversations
will in general fall into three camps:

The individuals who advocate that music ought to be engaged in the front L/C/R mix while
setting the encompassing and embellishments into the back surround field

The individuals who like to abandon such customary ideas and take a gander at the sound
field as an entire 360-condition, where instruments and mix components can fall anyplace in
the surround sound field

The individuals who will in general surrender the choice over to the craftsman, maker,
engineer, and record company.

When managing discrete 5.1, similarity issues aren't normally a significant issue, as it's
comprehended that media will be played back on a discrete playback framework. In such a
circumstance, it's basic for a different stereo/mono-perfect mix to be developed from the
soundtrack or creation mix. Notwithstanding, should the mix be encoded in Dolby ProLogic (a
plan that encodes the surround data into the L/R stereo track utilizing complex phase
connections), care ought to be taken to guarantee surround, stereo and mono playback
similarity.

Monitor Speaker Types

To get the most ideal trade-off in sound balance, a few monitor speaker alternatives are
frequently accessible as a reference during a session and/or mix. Frequently, a console or
monitor control framework will let you select between speaker/monitor types, with each set
usually having its related amplifier for power and level coordinating adaptability. These types
incorporate far-field, near field and headphone monitoring.

Fairfield monitors regularly include enormous, multi-driver amplifiers that are fit for
generally conveying exact sound at moderate to high volume levels. On account of their
enormous size and fundamental plan, the enclosures are by, and large soffit mounted
(incorporated with the control room to diminish reflections around and behind the walled-in
area and to build by and large speaker proficiency).

These large-driver systems (Figures 21) are frequently utilized during the recording stage in
light of their capacity to securely handle high levels (which can prove to be useful should a
receiver drop or a vocalist choose to be charming and shout into a mic). They're additionally
extraordinary for tuning in to a mix at noisy levels to hear the effect that it'll have on the floor
or in a loud system. Truth be told, specific types of music depend on bass levels that must be
provided by such a system at moderate-to-high SPLs. (Note: It's important to know about the
threat of long haul presentations to such sound levels.)

Fig 21. Fairfield Monitor (Adam S3H)

Near Field Monitoring

Although far-field monitors are commonly the best reference at high listening levels, hardly
any frameworks are outfitted with speakers that can convey ''clean'' sound at such high SPLs.
For this explanation, generally expert and venture studios use near field monitors that more
practically speak to the sort of listening condition than John and Jill Q. The term near field
alludes to the position of little to medium-sized shelf speakers on each side of a work area
working condition or on (or marginally behind) the metering extension of a creation console.

These speakers are commonly set at nearer working separations, permitting us to hear more
of the immediate sound and less of the room's general acoustics.
 Fig 22. Near Field Monitor (Avantone CLA-10)

As of late, nearfields have gotten an acknowledged standard for monitoring in practically all
regions that identify with sound creation for the accompanying reasons.

Quality near field monitors precisely speak to the sound that would be imitated by the normal
home speaker framework. The arrangement of these speakers at a position nearer to the
listening position lessens undesirable room reflections and resonances. On account of an
untuned room, this assists with making a more exact monitoring condition. These moderate-
sized speaker frameworks cost altogether not as much as their bigger studio reference
partners (also the decreased amplifier cost because less wattage is required).

Headphones

Headphones (Figure 23 and 24) are also an important monitoring device, as they remove you
from the room's acoustic condition. Headphones also offer incredible spatial placement in
that they let the engineer or producer place a sound source at basic situations inside the
sound system field without reflections or other natural impedance from the room. Since
they're portable, you can take your favourite headphones with you to rapidly and effectively
look at a mix in a new domain.

It ought to be noticed that while headphones take out the acoustics of a room from the
monitoring situation, they don't generally give a genuine portrayal of how sounds will carry
on through amplifiers (particularly for imaging). Monitoring through headphones will also
frequently accentuate low-level sounds like reverb and different impacts more than
amplifiers in a room. Therefore, tuning in to a mix over both monitor types will frequently be
valuable.
 Fig 23. Beyerdynamic DT 990pro Fig 24. Audio Technica ATH-M40x

Speaker Specification

Speaker specifications can be hard to genuinely decipher because there are numerous factors
in the way speakers can be measured. The most ideal approach to assess a speaker is to hear
it out. However, as sound specialists, we generally demand thoroughly analyzing the specs.

Frequency response specification

Understanding of frequency response details regularly is difficult because too little

information is provided. The genuine issue with frequency response specifications is the
number of factors included. The separation at which response is estimated, the size of the
room, the axis of estimation, the nature of the test signal, the force level utilized for the test.
These factors, and more, influence the deliberate frequency response. Similarly, as with other
speaker evaluations, there is no standard and to be significant, the maker should give the
frequency response and the greatest number of dB that the response fluctuates, comparative
with a reference frequency. The reference frequency is normally thought to be 1 kHz unless
otherwise specified.

Fig 25. Freq Response (Dt 990pro) Fig 26. Freq Response (ATH-M40x)

For an exact detail, an appraised speaker may have a frequency response of "20Hz to
20,000Hz, +-4dB". This determination is sensible, informative, and prone to be practical.
Determinations of "20Hz to 20,000Hz" infer close to 1dB variety between the level at 1,000Hz
(the basic "0dB'' reference recurrence) and the degree of any recurrence from 20Hz to
20,000Hz. All things considered, a speaker that performs with just 4dB of variety from 20Hz
to 20,000Hz is viewed as excellent. A decent speaker determination will incorporate a chart
of recurrence reaction. A chart will show the most extreme deviations from a perfectly "level"
response, and where the deviations happen.

Amplifier and Speaker power rating

Amplifier power is evaluated in "RMS". PMPO, which stands for peak momentary peak
output, not to be confused with the actual amplifier power. PMPO is cited for gear like the
smaller than expected framework and ghetto-blasters to make them look more remarkable;
"10,000 Watts PMPO!" simply demonstrates how noisy a framework can sound for a concise
second. RMS (Root Mean Square) is the normal nonstop force yield an amplifier is fit for
creating; power yield an amplifier can deliver reliably over broadened periods.

A decent amplifier (strong state) should give at any rate 20 Watts RMS. In a perfect world,
you ought to follow the suggested amplifier power indicated by the speaker maker.
Notwithstanding, an amplifier with a more powerful evaluation than the suggested
specification won't harm the speakers. The speaker framework sheet may contain an
announcement like this: Recommended amplifier power....10 - 70 Watts RMS and you are
facing a challenge by utilizing a bigger amplifier than 70 Watts RMS.

The speaker particularly suggests a base force however they are not firm, they do in general
be characteristic of the force required for sensibly good bass yield. As a rule, the "power"
rating of a speaker framework is its capacity to absorb sound power and make an
interpretation of it into sound and warmth, while the "power" rating of an amplifier is its
capacity to create sound power and convey it to a heap. These force appraisals truly portray
two distinct qualities, so the utilizing of the force evaluations produced for amplifiers to depict
speaker power limit is most likely invalid and can be deluding.

Speakers' capacity evaluations are additionally estimated as "peak" or "constant" power and
you ought to know about the two terms. "Peak" power implies that is the most extreme they
can handle before the peak level. Most amplifiers' capacity rating is Watts per channel (which
implies per speaker). Amplifier evaluations regularly incorporate "top" power also, which is
useful.

Advantage of Large Amplifiers

The limiting factor in coordinating an amplifier to a speaker where the intensity can cause
the voice curl to move excessively far, which may deform or tear the cone, arch, especially
with a woofer. When an amplifier’s power is higher than the speaker's "capacity limit" this
will not be an issue as long as the amplifier powers the speaker. While you may figure a bigger
amplifier would "wear out" a speaker much quicker, this isn't found in every case.

In certain occasions, a bigger amplifier may be more secure for your speakers. At the point
when you attempt to acquire high listening levels from a little amplifier, the amplifier may
arrive at its capacity cutoff points and start to "cut". At the point when this occurs, the highest
points of the amplifier's electrical waveforms (its capacity yield) are cut and extreme
distortion happens. Rather than moving consistently, the voice loop of a speaker which is
driven by an amplifier that is "into cutting" will move into one position and remain there,
move back and remain there, and so forth.

Under these conditions, the speaker draws more electrical force, yet changes over less
capacity to acoustical force than during normal (unclipped) activity. A bigger amplifier could
convey higher listening levels before it cuts, and even though more pinnacle force would be
taken care of to the voice loop, the vitality would be scattered as sound (air movement) rather
than heat. Likewise, the speakers can’t observe a small amount of that high force except if
you play at most extreme level constantly. It just takes a couple of watts to get an average
level from the speakers; the rest is there to handle the dynamics of music.

Efficiency

The productivity of a speaker system is a proportion of how well the speaker can change over
electrical vitality into sound (acoustic vitality). Speaker effectiveness differs generally. The
most noteworthy proficiency buyer speaker systems are about 20% effective; 4% can be
viewed as high productivity. Proficiency has no immediate relationship to the nature of sound
proliferation, and many lower productivity speakers have superb frequency reaction, low
distortion, and so on. Some high-productivity speakers, while they might be stronger than
lower proficiency speakers (given a similar force), may not sound as great. Proficiency is hard
to quantify, hard to assess, and only here and there determined by numerical worth so a
related worth is indicated rather than affectability.

Sensitivity

Sensitivity is estimated regarding dB/watt/meter or dB/2.83 volts/meter which implies that

specific tumult (1 dB) will be accomplished with a standard sign level (2.83 volts or 1 watt @
8 ohms) at a standard good ways from the speaker (1 meter). Affectability is identified with
proficiency, however, there is an important contrast; effectiveness thinks about the aggregate
sum of sound force (counting sound returning from the top, base, and sides of the speaker)
to the electrical info power, though affectability gauges just the sound weight level along with
the front-focus hub of the speaker.

Since it depicts which speaker will be stronger at the focal listening position, the affectability
rating is more helpful and affectability is estimated with a predefined input signal. While the
affectability rating is helpful, you ought to understand that, similar to control evaluations,
affectability estimation strategies can and do change. On the off chance that two speakers'
sensitivities are estimated comparatively, the speaker of higher affectability will be stronger.
Then again, the lower-affectability speaker having a powerful appraising might have the
option to create more sound than a higher affectability speaker with a low force rating.
Affectability evaluations give just a sign of the relative uproar that can be delivered and have
little to do with sound generation quality.

Dispersion and Polar response

The polar chart is a commotion bend drawn on a 360-degree round matrix, where the
distance from the centre of the circle shows relative level. Sub-frequencies are estimated and
outlined as the speaker is pivoted, giving a progression of "bends" that speak to the relative
frequency reaction at different precise positions. In straightforward terms, polar charts
resemble perspectives on a speaker, with nonexistent lines that show how well various
frequencies are "anticipated" in two opposite planes. Since scattering and polar reaction are
firmly related, they are now and again utilized reciprocally. Another name for scattering and
the polar reaction is "off-pivot reaction".

A speaker determination may peruse, "scattering 120-degrees"; this proposes on the off
chance that you move along a circular segment anyplace inside 60 degrees on either side of
the speaker's middle pivot, the frequency substance will remain equivalent to it along with
the middle hub. A superior detail would give the most extreme number of dB that the reaction
differs (not the commotion change, however the adjustment in frequency reaction). For
instance, a speaker may be indicated as having "120-degree scattering, +-6dB somewhere in
the range of 40Hz and 16kHz". A surprisingly better strategy to determine scatter is with
"polar reaction" graphs.

Fig 27. Horizontal and Vertical Polar Response

Polar graphs uncover what sound the audience hears while moving around the speaker. Issues
that are handily observed on polar charts are "radiating" (a narrowing of scattering at higher
frequencies), and "campaigning" (a progression of serious changes in level as the audience
moves over a circular segment before the speaker). Horizontal scattering maybe is more
critical than vertical scattering, however, both ought to be wide.

Keep in mind, if the speaker is worked horizontally, as a shelf speaker on its side, the vertical
scattering turns into the horizontal scattering.

Along the edges, most speakers will "tumble off" fairly in generally speaking level, especially
at higher frequencies. Speakers with more extensive scattering are probably going to sound
better, not exclusively to audience members on the sides yet to those before the speakers.
Additionally, wide scattering helps by keeping sound more "even" when sitting or standing.
Because of development and material science, the horizontal scattering and vertical
scattering barely ever coordinate, yet this has little effect on the speaker's performance. Most
little speaker systems have wide scattering at low frequencies, however tight radiating
scattering at high frequencies. A perfect speaker system will have even scattering at all
frequencies.

Impedance Specifications

Each speaker model has an electrical detail called the "ostensible impedance". For the most
part, it is 8 Ohms, which implies that, excluding the speaker wire, the amplifier is given 8 Ohms
of opposition (impedance) in passing its electrical sign through the speaker. The word
ostensible implies that 8 Ohms is the normal impedance. Impedance has nothing to do with
quality. 4, 6, 8 or 16-ohm speakers can be utilized with practically any amplifier accessible for
home use.

On the off chance that few speakers are wired in equal, amplifiers may distort, blow a breaker,
or overheat and even wear out. Two 4-ohm speakers or four 8-ohm speakers, wired in
parallel, comprise a 2-ohm load, which is about equivalent to putting ten feet or wire over
the yield of your amp… a near short circuit!. On the off chance that you intend to utilize a few
sets of speakers, or if your speakers are appraised at 4-ohms or less, be certain the amplifier
can handle the heap. Once in a while arrangement or arrangement/equal wiring to the
speakers will give the required least impedance. With any multi-speaker load, some
corruption of performance might be inescapable because of collaboration so for best
outcomes, all speakers ought to be indistinguishable.

Ostensible impedance rises to the real impedance, over a restricted portion of the speaker's
frequency go due to impedance changes with frequency. To give a superior sign of the real
impedance, a diagram of impedance versus frequency now and then is recorded with the
speaker particulars. The facts confirm that the lower the impedance, the more force a speaker
will draw in general, however the way that impedance changes with frequency don’t imply
that frequency reaction will be lopsided. This is because the effectiveness of the drivers
likewise fluctuates with the frequency and because the mix of fenced-in area reverberation
with the hybrid network will, in general, make up for the adjustments in power draw that may
be recommended by average impedance bends.

Damping factor

The impedance of the speaker will likewise influence what is known as the "damping factor".
This is characterized as the proportion of the impedance of the speaker to the yield
impedance of the amplifier. In this manner, if the speaker impedance is 8 Ohms, and the
amplifier yield impedance is 0.05 Ohms, at that point the damping factor is 8 isolated by 0.05
= 160. High damping factors generally imply that the bass reaction will be very much
characterized ("tight"), while a low damping factor will bring about a free sounding bass.

Tight or free bass involves preference; one isn't better than the other. Cylinder amplifiers
frequently have low damping factors, for instance, 10, contrasted with a strong state, which
may add to their normally free bass reaction. Cylinder amplifiers are regularly depicted as far
as "warmth" or "detachment", and it tends to be a charming impact. Regardless, the detail
sheet for the amplifier will once in a while list the damping factor, so on the off chance that
you pick a low impedance speaker, a higher damping factor on the amplifier might be
fundamental if you like the "tight" bass sound. Dropping the speaker impedance from 8 to 4
Ohms would decrease the damping factor by half for some random amplifier.

Distortion specifications

Distortion in speaker systems might be brought about by a wide range of factors that are
extremely mind-boggling and are difficult to completely depict. While symphonious distortion
generally indicated as T.H.D. (Total Harmonic Distortion) is a typical determination,
intermodulation distortion (generally indicated as I.M.) is only here and there indicated.
Numerous different factors, for example, non-linearity at low listening levels, may be
considered "distortion" however not many such things are ever recorded on detail sheets.
The way that rigorous distortion details are discarded is no deficiency of the producer as there
is real trouble concluding how to gauge numerous inconspicuous distortions in a manner that
can be seriously deciphered by the purchaser, that is adequate to different makers, and that
is repeatable in other testing circumstances.

Indeed, even the estimation of intermodulation distortion, a basic undertaking for power
amplifiers in particular, is hard for speakers. Intermodulation regularly results from the
"Doppler impact". A speaker recreating a high and low frequency simultaneously will create
undesirable "sidebands'' (new frequencies that were absent in the information signal, and
that are non-melodic to the ear). Intermodulation in an amplifier might be determined at
numerous frequencies; 60Hz and 6 kHz, 70 kHz, and so forth, all of which have legitimacy.

In any case, if a two-way speaker is estimated for intermodulation at 70Hz and 7kHz, the
odds are that these frequencies are being recreated by two distinct drivers (70Hz by the
woofer and 7kHz by the tweeter), so intermodulation isn't likely. If a similar speaker were
estimated at 40Hz and 300Hz, the two of which frequencies are taken care of to the woofer,
extreme intermodulation could happen, with perceptibly upsetting outcomes. Along these
lines, regardless of whether an intermodulation rating is given on a speaker, it might be
important. Symphonious distortion might be the consequence of overdriving the speaker (an
excessive amount of intensity), of "undulating" speaker cones, or both. The symphonious
distortion detail is valuable since it very well may be definitively looked at on different
particular sheets. Stage distortion frequently is discussed and only from time to time
characterized.

Next Lesson:

In the next lesson, we will learn

1. Working and responsibilities of a crossover and its uses.
2. Overview of Two and Three-Way Crossovers.

Questions:

1. What is a banana plug and describe its uses?

2. What are the different types of monitoring configurations?
3. Explain MONO, Stereo & Surround sound configurations in detail.
4. What are the different monitor speaker types? Explain briefly.
5. State difference between Near and Far-Field Monitoring.
6. Why do we use headphones for monitoring? State the different types.
7. What is the frequency response? Explain a frequency response graph.
8. What are the advantages of using large amplifiers?
9. Explain dispersion and polar response.
10. Explain Damping factor and Distortion specifications.

Suggested Viewing:
1. Mono, Stereo and Surround explained - https://www.youtube.com/watch?v=-XP4MkaZylU
2. Difference between Mono and Stereo - https://www.youtube.com/watch?v=RVgc7v-4n8k
3. Positioning surround sound - https://www.youtube.com/watch?v=3sjocJvK6rI
4. Understanding speaker impedance - https://www.youtube.com/watch?v=gvaojICThzg
 Lesson 5: Crossover Networks
(Reading Time: 15 mins)

Learning Outcomes:

1. Understanding working and responsibilities of a cross over.

2. Overview of Two and Three-Way Crossovers.

Since singular speaker components (drivers) are more productive in some frequency ranges
than in others, distinctive driver sizes and types are frequently utilized in the mix to give the
ideal frequency reaction and level yield. For instance, huge measurement drivers, (for
example, 8 and 18 inches) produce low-frequency information more productively than at high
frequencies; medium-sized speakers, (for example, 4 and 5 inches) work best in the midrange
frequencies, and little speakers, (for example, 1 to 2-inch stomach sizes) duplicate highs more
successfully.

These speakers are frequently associated with latent hybrid networks, which forestall flags
outside a specific frequency run from being applied to its assigned out the speaker. Inactive
networks utilize frequency-specific inductors and capacitors to part the frequency into a few
frequency bands. This arrangement gives smooth progress from speaker to speaker by
steering input flags over the hybrid frequency to the mid-and/or high-frequency driver while
directing signs underneath the hybrid frequency to the bass driver or drivers.

Fig 28. Speaker crossover network

 Fig 29. Passive two-way crossover system: (a) crossover/amp layout and (b) frequency
response curves showing crossover frequencies of 1500 Hz

If a speaker system has just a single crossover frequency, it is known as a two-way system,
because the signals are separated into two bands. In like manner, if the signal has two hybrid
frequencies, it's known as a three-way system. The Westlake Audio BBSM-10 monitor
speaker, for instance, is a ported three-way system that incorporates two 1000 woofers for
the bass, a 6.500 driver for the mid-frequencies, and a 1.500 stomach vault tweeter for the
highs with the hybrid frequencies being tuned at 600 Hz and 4 kHz, individually.

Certain plans incorporate hybrid level controls that decide how much energy is to be sent to
the centre and high-frequency drivers. This lets you make up for different room situations
and/or insufficiencies (for instance, an absorptive room may require more high-frequency
energy than a live room).

Electronic hybrid networks, called dynamic hybrids, vary from traditional latent hybrid
systems in that the line-level sound sign is partitioned into different frequency bands. Each
levelled signal is then taken care of to its capacity amp, which thus is utilized to drive the
particular bass, mid, and/or high-driver components. Such a system is, for the most part,
alluded to as being bi-enhanced or tri-intensified, contingent upon the number of hybrids and
force amps that are required per channel.

Such a system has several advantages:

• The crossover signals are low in level, implying that inductors (which can present
discernible ringing and between balance distortion) can be dispensed with from the plan.

• Every frequency go has its capacity amp, so the full intensity of every amplifier in the
separate speaker proficiency range will be accessible (implying that the drawing of
unnecessary current in one area won't influence the sound in another frequency run).
For instance, how about we expect that we're taking care of a 100-W power amp through
an inactive crossover network to a low-end high-frequency speaker system. If the low
frequencies are pulling 100 watts of intensity from the amp and a high-frequency signal
goes along that requires an extra 25 W of intensity, the amplifier won't have the option
to supply it. Both the lows and the highs will get distorted. These monitoring necessities,
be that as it may, could be met without bringing about distortion by utilizing a functioning
crossover network to take care of a 100-W Amp for the low-end speaker and a different
25-W amp for the highs.

• The crossover focuses for either an uninvolved or a functioning crossover network are
commonly 3 dB down from the level portion of the reaction bend. Frequency extending
outside the channel's passband will be constricted according to an incline (normally 6,
12, 18, or 24 dB for each octave with a 12-dB/octave slant being the most widely
recognized). The system's crossover focus will be dictated by the driver plan and type;
nonetheless, the more regularly chosen frequencies are 500, 800, 1200, 5000, and 7000
Hz.

Crossover Elements
The capacitor has a lower impedance at high frequencies. It acts to block low frequencies and
let high frequencies through.

Fig 30. i.High-pass filter

The inductor has a lower impedance for low frequencies. It acts to block high frequencies and
let low frequencies through.

Fig 30. ii.Low-pass filter

A capacitor and inductor in arrangement act to block both extremely high and exceptionally
low frequencies.

Fig 30. iii.Band-pass filter

Two-Way Crossover

Blends of capacitors, inductors, and resistors can guide high frequencies to the tweeter and
low frequencies to the woofer. This adds up to channel activity. A two-way crossover network
separates the frequency run between two speakers.
 Fig 31. Two-Way Crossover

Three-Way Crossover

A blend of capacitors, inductors, and resistors can guide high frequencies to the tweeter and
low frequencies to the woofer. This adds up to channel activity. A three-way crossover
network partitions the frequency between three speakers.

A capacitor has a lower impedance for high frequencies. In arrangement with the high-
frequency speaker (tweeter), it acts to block low frequencies and let high frequencies
through.

Fig 32. Three-Way Crossover

The inductor has a lower impedance for low frequencies. In arrangement with the low-
frequency speaker (woofer), it acts to block high frequencies and let low frequencies through.

Questions:

1. What is a crossover?
2. Explain High, Mid and Low-Frequency Signals.
 3. Explain active and passive crossovers.
4. What are the crossover elements?
5. Explain High-Pass, Low-Pass and Band-Pass Filter.
6. Explain Two-Way and Three-Way crossovers.

Suggested Viewing:

1. What is a crossover and how does it affect speaker configuration -

https://www.youtube.com/watch?v=iAIgK6Vf4TM

2. Two-way and Three-way Crossover -

https://www.youtube.com/watch?v=jVDmvQG2xsQ

Lesson 1: Introduction to Signal Processors

(Reading Time: 15 Minutes)

Learning Outcomes:

1. Understanding Signal Processors

2. Analog and Digital signal and processing

3. Basic comparison between Analog and Digital

What are Signal Processors?

Have you ever come across the terms ‘compressor’, ‘EQ’, ‘Gate’, ‘Threshold’, ‘limiter’,
‘expander’, etc.?

These can be analogue hardware or digital software which is a DSP (Digital Signal Processor).
But what exactly are these? These are devices or processors which modify an audio channel
in the analogue or digital domain. These are used for sculpting sound and to give it character
and distinguish it. This is why it is generally seen that most audio channels in a project have
an array or chain of signal processors to it and go through continuous change before it reaches
perfection according to its purpose.

Back in the day when the advent of digital technology did not take birth, signal processors
were analogue hardware. They would be put on a signal chain with other rack mount effects
across the studio. Typically, they would not be cheap, as a result of which studios with a
convincing budget not only had a lot to offer but also dominated the scene.

Today, with the boost of technology, Digital Signal Processors are being introduced in the
audio market almost every day in the form of digital emulations of old analogue hardware.
This makes it much easier in terms of space and budget required for a studio or an individual
to dive deep into the world of sound.

To make it easy for a better understanding let us break down the word “Signal Processing”.
The signal is information and processing is operation. So, any operation done on specific
information is Signal Processing. The signal goes through analysation and modification.
So, it can be said that analogue signal processing is a continuous wave that changes over time.
It can be denoted by a Sine Wave. Below is an image representing an analogue signal on the
X/Y axis.

Fig 1. Analog Signal (X-Axis- Time, Y-Axis- Voltage)

Like said before, when some operation happens on that information is when it can be termed
as ‘Processing’. Hence, when processing happens on that analogue signal it is then called as
analogue signal processing. This processing analyses and modifies the signal. The easiest way
to understand processing happening on an analogue signal can be understood simply by
taking a microphone and passing it through an amplifier to the speakers.

The signal coming out of the microphone has low amplitude. When it goes through the
amplifier its amplitude is high then goes to the speaker for everybody to hear. The amplifier
is the analogue signal processor here because it is modifying the voice coming out of the
microphone from low amplitude to high amplitude and further the output via the speaker.

Analog Processors have come back to the industry with a boost in the form of software
emulations and are being used by everyone in every setting. It shows the importance of using
old tools and techniques while combining them with the present technology to make the best
use of both worlds.

These devices generally alter a source’s relative volume levels (e.g., equalization and
dynamics), also many analogue devices can be used to alter time-based effects. Spring and
plate reverb units add their distinctive sonic characteristics and can be found in new markets.
Signal processing has become an increasingly important part of modern sound and music
production. Signal Processor changes or modifies an audio signal both analogically or digitally.

In today’s world, digital audio has set signal processing on fire by offering an almost unlimited
range of effects that are available to the musician, producer, and engineer.

A big advantage of working in the digital domain is that software can be used to achieve a
wide range of effects like reverb, echo, delay, equalization, dynamics, pitch shifting. The
work of processing a signal digitally is achieved by mixing logic circuits in a building-block
fashion. The logic blocks follow binary computation rules that operate according to a program
algorithm.

The numeric value of sampled audio can be altered when these are combined. After a
program has been configured (from internal ROM, RAM, or system software), complete
control over a program’s setup parameters can be altered.
As the process is fully digital, the settings can be saved and duplicated at any time upon recall.
Amazingly the overall quality and function steadily increase also becomes available and more
cost-effective. Bringing a huge amount of production power to both a beginner and a
professional.

To understand DSP (Digital Signal Processing) more easily, we can say that the wave here is
discrete, unlike the analogue signal where the wave was continuous. This discrete waveform
carries information in binary form.

The information is carried through a combination of 1 and 0 (Binary Number). Processing is

done on digital signals. The wave here is denoted by Square wave unlike Sine wave for
analogue.

Fig 2. Digital Signal (X-Axis- Time, Y-Axis- Binary Number)

In today’s digital world, analogue is converted to digital and then back to digital.

The most popular standards are VST (PC/Mac), DirectX (PC), Audio Suite (Mac), Audio Units
(Mac), MAS (MOTU for PC/Mac), and TDM (Digi design for PC/ Mac).

The computer’s CPU carries out the DSP functions. With the speed and power of modern-day
CPUs, this has become less of a problem; however, when operating on a complex session
there is a possibility for the computer to still run out of DSP steam.
 Fig 3.Analogue to Digital, Digital to Analog

Running out of DSP streams can be dealt with in several ways. A lot of digital audio
workstations come with freeze or lock functionality that paves the way for processing to
release the CPU for other real-time operations and avoid unnecessary load.

A DSP accelerator card that can be plugged into your computer acts as a dedicated plug-in
processor to share the processing workload with the CPU. When using Steinberg’s VST plugin,
it’s possible to share the CPU workload over several networked computers using their VST
Link (V-Stack) protocol.

Fig 4. Analog signal chain

Questions:
1. What are signal processors?
2. Name a few basic common sound processing signal processors.
3. What is the full form of DSP?
4. What do signal processors do?
 5. Do a brief comparison of old analogue signal processors and present-day usage of DSP.
6. Signal is __________ and processing is _________.
7. The signal goes through __________ and ____________.
8. What is an Analog signal? Explain its processing with an example.
9. Which wave denotes DSP?
10. Explain the whole concept of DSP.

Suggested Resources
1. Audio Processing
2. Digital and Analog Signals (Austin Lutz)
3. Analog vs. Digital As Fast As Possible

Next Lesson:

In the next lesson, we will learn,

1. Types of Signal Processors
2. Computer processing areas

Lesson 2: Types of Signal Processors

(Reading time: 15 Minutes)

Learning Outcomes:
1. Computer Signal Processing areas
2. Types of Signal Processors

Let us take into consideration that in the digital domain the computer is also a digital signal
processor. It analyses the signal it receives and performs vivid signal processing operations.
Hence, a quick go through of some of the computer signal processing areas can be beneficial
for an in-depth understanding. These are as follows:
1. Speech processing
2. Sonar
3. Radar
4. Spectral density estimation
5. Statistical signal processing
6. Digital Image processing
7. Data Compression
8. Video Coding
9. Audio Coding
10. Signal processing for telecommunications
11. Biomedical Engineering
12. Control systems

So, it can be said how vividly Digital Signal Processing is being used in multiple ways, in
multiple platforms.
Any device that processes audio has an inbuilt DSP. The Apple AirPods Pro uses a DSP that
measures the ear canal and then adjusts the performance of the earbuds to optimize the
audio quality. Sony’s 360 Reality Audio maps the ear and adjusts its signal to achieve the best-
desired results.

DSP boxes are also used to correct performance for bookshelf speakers, headphones, and
calibration of virtual surround sound systems. The operation of Analog processors happens
directly on the electrical signal, while digital processors operate mathematically on its digital
representation. The processing list goes on. Though these are some of the most common
signal processors to manipulate sound.

1. EQUALISATION (EQ) - It adjusts and balances the frequency within an electrical signal
(Analog). Though with the advent of DSP, EQ is also used digitally. If the sound isn't
good without EQ, then one will never end up with anything but second best. The only
time we should ever use EQ to 'save' a sound is when we have been given a project to
work on that was recorded by a lazy engineer. However, EQ is massively used in the
form of random experimentation by DJs and live electronic artists to cut and boost
frequency to manipulate and play around with an audio signal.

2. Compression- A compressor manipulates the dynamic range between the quietest

and loudest parts of an audio signal. The louder signals are attenuated and the quieter
signals are boosted. However, there are no hard and fast rules to achieve the desired
sound.

3. Audio Filter- This has an audio frequency range of 0 Hz to 20 kHz. It can amplify, cut
or pass audio signals. It can be said that a filter is a frequency-dependent amplifier
circuit.

4. Limiter- It limits signal levels from increasing beyond a specified level. It is also a
compressor that limits the level of an audio signal to a certain threshold.

5. Chorus- The chorus effect can make a single instrument sound like several instruments
are being played. It adds some thickness to the sound and is often described as 'lush'
or 'rich'

6. Phaser- It is an effect processor that creates a series of peaks and troughs all over the
frequency spectrum. It results in sweeping sounds.

7. Flanger- This produces whooshing sound by doubling the input signal and playing both
back together. As a result, there is a slight difference in phase and delay.

8. Delay- This signal processor is also a cool effect unit like some of the previously listed
above. It records an input signal into storage and plays it back after some time. A
digital delay is accomplished by storing sample audio directly into RAM. After a defined
length of time usually measured in milliseconds, the sample audio can be read from
memory for further processing or direct output.

9. Reverb- This is also called reverberation. Reverb has to do with the sense of space. In
the simplest sense, reverb simulates the reflections of sound off walls by creating
 dense echoes or delays of the original signal. Since walls absorb sound over time, the
delays or reflections in a reverb decay over time.

10. Echo- This happens when the sound waves reflect. It is used in processing audio in the
desired way. It has a delay that is directly proportional to the distance of the reflecting
surface.

11. Expander- Expansion is the process by which the dynamic range of a signal is
proportionately increased. Depending on the system’s design, an expander can
operate either by decreasing the gain of a signal (as its level falls below the threshold)
or by increasing the gain (as the level rises above it).

12. Noise Gate- This allows a signal above a selected threshold to pass through to the
output at unity gain without dynamic processing. Once the input signal falls below this
threshold level, the gate acts as an expander and effectively mutes the signal by fully
attenuating it. Crucial points as to why Noise Gate is used are the reduction of leakage
between instruments and eliminating noise from an instrument or vocal track.

Fig 5. Rack MountSignal processors

Fig 6 Digital signal Processor- Software Compressor

Questions:
1. Mention at least 5 computer processing areas.
 2. Mention a few companies that use DSP in their products.
3. Give examples of signal processors.
4. Explain in brief what is Delay?
5. What is Reverb?
6. Explain Noise Gate?
7. Explain the concept of Phaser and Flanger.
8. What is EQ?
9. Explain Compression and Limiting.
10. What other things are DSP boxes used for apart from its common areas?

Suggested Resources:
1. Signal Processing 101
2. Katie Gately: Sound Processing

References:

Triggs, Robert. “What is a DSP?.” soundguys.com. https://www.soundguys.com/what-is-

dsp-28013 (accessed July 17, 2020).

Lesson 1: Introduction

Learning Time: 5 min

Learning Outcomes: Students are able to route and mix audio signals with the usage of
onboard signal processors and outboard gears.

A Console is considered as the heart of a sound recording and mixing studio. A console that
sits at the heart of any facility that works with audio, such as recording and mixing studios,
live sound venues and film sound mixing and broadcast studios. As the entire sound
engineering process mostly happens in between two transducers, the presence of a console
is inevitable.

The advantage of a console is its capacity to route the audio signals to various signal paths
inside the console and different outboard gears, whichever is connected with the console.
With the help of a console, you will be able to make different successful audio signal chains.

The basic functions of an audio console can be broadly placed in two categories:-

1. Mixing Audio:-

Audio consoles mix audio signals together. Audio consoles are also called mixing consoles.
Consoles have the capacity to mix multiple audio signals that come through different channel
strips. Mixing of audio is, technically, a more elaborate process, which includes changing the
timbre of incoming sounds so they fit into a mix.

2. Routing Audio:-

The standard operating level of a console is +4dBu. The tracking or mixing section of a console
is well structured with many features that allow it to route and shape the incoming audio
signals. These features will be available at every channel and known as Channel Strips.
Next Lesson: In the next lesson you will learn the usage of channel strips during the signal
routing and mixing

Questions:

1. Name 2 basic functions of audio consoles.

2. State difference between Mixing and Routing.

Lesson 2: Channel Strip

Learning Time: 35 min

Learning Outcomes: Students will be able to understand the usage channel strips during the
signal routing and mixing

The channel strip is the path through the audio signal flows. The audio signal flow begins at
the input stage and ends at the channel fader. The incoming audio signal can be routed to any
other part of the console or even out of the console where the other audio devices that
connected with the insert ports, auxiliary sends and returns, sub-groups and other
connectivity ports.

When a microphone is plugged in, it is plugged into a channel strip. When a recorder or DAW
return is brought back to the console for mixing purposes, each return is brought into a
separate channel strip. A mixing console might look scary at first sight, with its hundreds of
rotary controls and switches, but it seems less so if you know that the rotary controls and
switches you find on one channel strip are identical to those on other channel strips. If you
are grabbing the awareness about a channel strip and it’s audio signal routing ways, You will
be able to handle the whole console easily.

A channel strip contains the following features typically:-

1. Connectivity Ports
2. Input Gain Section
3. Equalizer Section
4. Auxiliaries
5. Monitor Bus
6. The Pan Knob
7. Solo and Mute Switches
8. Subgroup switches and Main Stereo switch
9. Channel Fader
10. PFL and AFL
The position of Equalizer, Auxiliary, PFL and Sub-Group sections can be changed as per the
design concept of a premium/large scale console.

1. Connectivity Ports

The connectivity port of a channel strip mainly contains:-

a. Mic Input – XLR3 (female)

This input slot is for connecting mic level signals. XLR connectors are available in male and
female versions. XLR male connectors are specifically used for outputs and XLR female
connectors are used for inputs. An XLR3 has 3 conductors (pins) in the connector that are Hot,
Cold and Ground. It allows it to carry a +48 volt Phantom power when required. Phantom
power is supplied to the microphone on pins 2 and 3 of the XLR cable, while the ground is
supplied by pin 1.

A dual-core cable (two signal wires and a separate ground wire) will be used to establish the
connection with an XLR3. A dual-core cable (two signal wires and a separate ground wire) will
be used to establish the connection with an XLR connector.

Picture Courtesy: Wikipedia

 Picture Courtesy: slideshare.net

b. Line Input - TRS (1/4” Jack)

This input slot is for connecting line-level signals. TRS connectors are available in male and
female versions. Both male and female connectors being used for both inputs and outputs. A
TRS has connectivity has 3 conductors in the connector i.e. Tip, Ring and Sleeve. A dual-core
cable (two signal wires and a separate ground wire) will be used to establish the connection
with a TRS connector.

Picture courtesy: www.sweetwater.com

c. Insert - TRS (1/4” Jack)

Insert is a direct out as well as direct in. An insert point is used to insert outboard equipment
into the signal chain of the channel strip. This port allows inserting outboard gear like a
compressor or any other dynamic processor.

If the insert slot is a single TRS female connector, you can use a ‘Y’ cable to insert an audio
device.
 Picture Courtesy: Amazon.com

‘Y’ cable has a TRS connector corresponding to 2 TS connectors. Means 1 balanced end and
2 unbalanced ends.

Insert process: Insert the TRS connector to the insert slot of the console and now the TIP of
the TRS connector will be ready to send the audio signal and the RING of the TRS will be ready
to receive the audio signal back to the console.

Now you can connect the TS corresponding to TIP of TRS to the input port of the outboard
gear and connect the TS corresponding to RING of TRS to the output port of the outboard
gear. Inserting of a device into a channel will be extending the capacity of that specific channel
only.

d. Direct Out

Mixing Consoles give the user the option to take the signal out of the board just after the
preamplification. The signal is split at this point, with one copy of the signal sent out of the
board from the direct out. The other copy of the signal going through the rest of the channel
strip. The direct out can be the form of a TRS female connection or a DB25 connection for 8
channels as a group.

In premium consoles, the direct out also can carry the on-board processed signal.

The direct out can be directly routed to a standalone multi-track recorder or an audio
interface with more input capacity to establish a multi-track recording. Direct out doesn’t
have a knob on the channel strip section.

The direct out is the cleanest, least noisy output of a mixing board, and it’s often used in parts
of the console that are faulty or noisy, in order to achieve the cleanest output available.

2. Input Gain Section

Once an input source is connected with the input port, the next step is to make the incoming
audio signal relevant as per the standard operating level.

To make the incoming signal well, there are some supportive features attached on the Input
Gain Section.

Preamplifier/Preamp.

The preamp is an electronic amplifier that changes a weak electrical signal to a line-level
signal strong enough for further processing. Microphone signals are usually very low from the
standard operating level, so a lot more gain is required. Preamps are usually capable of driving
30dB to 60dB gain to make the weaker mic level signal strong. To support the gain
management, there are certain features available along with a preamp.

Before we introduce a signal into the channel strip, we first have to make sure what kind of
signal are we plugging into the input of the channel strip, is it a weak or soft signal of a
microphone? is it a strong signal from a keyboard? is it a very loud signal from a synthesizer?.

When a signal is introduced into a channel strip, it has to meet certain requirements that are
given below.

Phantom Power

As you learned a condenser microphone does not generate the power to convert acoustic
energy to electric energy itself. In this case, you have to drive the power of +48volts to the
condenser microphone to operate it. That power is called the Phantom power. You can find
a Phantom Power switch in the preamp section of every console to provide the required
power to the condenser microphones. In addition to powering the circuitry of a microphone,
traditional condenser microphones also use Phantom Power for polarizing the microphone’s
transducer element.

a. TRIM/PAD

The PAD stands for “Passive Attenuation Device” Passive refers to an electronic circuit that
requires no power to operate. Attenuation means making the level of the signal smaller. The
pad is commonly found in a microphone preamp. A pad also can be found in a capacitor
microphone as switchable. It allows reducing the amount of incoming signal as mentioned on
the board. This is very useful to attenuate incoming instrumental level
signals.

b. High-Pass Filter

A high-pass filter is an electronic filter that passes signals with a frequency higher than a
certain cutoff frequency and attenuates signals with frequencies lower than the cutoff
frequency. The amount of attenuation for each frequency can be varied as per the filter
design. A high-pass filter also is known as a low-cut. Means, a high-pass filter allows high
frequencies to get through while filtering or cutting low frequencies. A high-pass filter can be
used to prevent undesired air-blows, pops etc.
c. Phase Invert Switch

With this switch, you can shift the phase 180 degrees forward or backwards in time and also
flip the polarity 180 degrees relative. While toggling the phase switch, you will be able to
listen to the sonic difference on the incoming signals.

d. Mic/Line Switch

This switch is a selection switch that allows selecting the source of an input signal whether
mic input or line input. This option is not a common one for all the consoles, mostly available
on premium consoles.

e. Flip Switch

This switch is switched to toggle with the source and tape(DAW) input signals. And this switch
will be available only on In-Line Consoles.

f. Level Meter

Most of the premium consoles have a meter per channel to indicate the strength of the
incoming signal. Most of the consoles have an overload indicator also per channel.

The Preamp Section – An Overview

The preamp section as a whole is used for level setting, and fixing the incoming signal by using
Preamp, PAD, Phase switch, Hi-Pass filter and Phantom Power switch. Once the signal is set,
the preamp section is rarely touched again. In fact, once the signal is set, there is no need to
touch the preamp section at all, unless the source of the signal is changing. The preamp is
very important because any loss of quality at this stage cannot be regained.
 Picture Courtesy: digitalsoundandmusic.com

3. Dynamics and Equalization Section

Further down the channel strip, after the signal is sent at the preamp section, we will be able
to start modifying the incoming signal the way we want. This where the second part of the
channel strip comes in.

Dynamics

Not all the consoles have a dedicated dynamic section per channel strip, only with premium
consoles like Harrison Series 12 or SSL Duality. Usually, dynamics modules are expensive and
having on each channel strip will significantly increase the cost of the board as a whole.

The dynamic section consists of a dynamic compressor and expander. These modules change
the amplitude of the audio signal dependent on the signal’s incoming amplitude.

A compressor attenuates the parts of the signal that cross a certain level, and an expander
attenuates the parts of the signal that do not cross a certain level.
 Picture Courtesy: SSL Duality SE 48

Equalization

The equalizer section consists of an EQ, and on higher-end consoles, two filters for either side
of the spectrum. After the audio passes the dynamic section, it is sent to the EQ, which boosts
or attenuates certain frequencies and leaves others unaffected.

The parametric equalizer is the most flexible type in equalizers. It’s midrange bands usually
have three adjustments: gain, centre frequency, and quality factor Q(bandwidth).

A parametric equalizer allows the engineer to add a peak or notch at an arbitrary location in
the audio spectrum. Adding a peak can be useful to help an instrument be heard in a complex
mix or to boost a particular frequency range. Notches can be used to attenuate unwanted
frequency ranges.
 Picture Courtesy: Harrison Consoles

4. Auxiliary Section

The auxiliary section mainly has two areas of the workflow. Those are known as sends and
returns.

Sends

Auxiliary sends allow the signal to be sent independently to some other location such as a
signal processor (mostly time-based), headphone amplifier or to a stage monitor (MOH).

An output from the console which is used to send (route) the signal to additional external
processing or monitoring.

Individual channel strips on a console will have several auxiliary send channels, each of which
usually goes to the master auxiliary send knob or fader before being sent out of the console.

All the master auxiliary knobs or faders are available on the master section of a console.

The audio signal through an auxiliary send may be pre-fader, i.e. independent of the input
fader on the console and maybe postfader i.e. dependent on the level set by the input fader.

The way the audio signals are sent through the auxiliary bus from different channel strips can
create a separate mix than the main output of the console. The balance of instruments or
sounds probably won’t be the same as the main mix.

For example, a musician may need a particular balance in their headphones or stage monitor.

That’s what auxiliary send knobs are for. You can use them to tap off a bit of sound from any
or all of the channels on the console at any level you require without messing up the main
mix.
The audio signal sent through auxiliary via pre-fader mode will be useful when the aux sends
and the master output needs independent audio streams.

Audio signals sent through auxiliary via post-fader mode to an aux send master will be in
proportion to the level of the input fader. This is the most common way to send a signal to an
external processor such as ‘reverb’ or ‘delay’ or any other time-based effect processors.

Postfader sends allow the engineer to maintain a consistent relationship between Dry and
Wet signals and it is preferable for an ideal mixing.

Post or pre-fader switches will be available at each auxiliary send options of a channel strip.

Auxiliary Returns

We have discussed the ways of sending audio signals through auxiliary sends to master sends
that correspond to the auxiliary output port of the consoles. After processing the signal it is
usually sent back to the console through the auxiliary return port that is available at the
connectivity area(back panel) of the master section.

In this way, the engineer will be able to control the dry(original) signal and the wet only signal
separately as required.

The wet only signal that comes through an auxiliary return port also can be routed to different
sub-group outputs if required.
 Photo Courtesy: Rupert Neve

5. Monitor Bus

All inputs, either recorded audio from a recorder, the audio interface returns coming on to
channel strips, or audio being currently recorded, coming on to their own channel strips, can
be sent to the monitor bus. The signal on the monitor bus is then sent to the monitors for
listening.

This signal path is very important for an in-line console for monitoring purposes, as tracking
and mixing processes happen through the same channel strip.

6. The Pan Knob

The knob that allows the amount of signal going to the left and right channels called a pan
Knob. This function is very important at various points of signal routing. At the time of
tracking, a pan knob allows to send signals through individual subgroup outputs, and at the
time of mixing and monitoring a Pan-knob allows the audio signals to send whether to Left or
Right channels. This pan knob is only suitable for stereo streaming of audio signals.

When the signal from a channel strip reaches the pan knob, it splits into two, a left channel
and a right channel, and the signal passes equally to the two if the pan knob is kept in a centre
pan mode. The separate Left and Right signals then pass on separately to the channel fader
and then to the Master bus, where the left signal goes to the Master Mix Left bus and the
right signal goes to the Master Mix Right bus. These L/R signals will be available on Master
Stereo output port and Control Room Stereo output port.

When the pan knob is turned completely to left, called panning the signal “hard left” and
turned completely to right called panning “hard right”.

The pan knob is a panoramic potentiometer and is often called a pan pot.

7. Solo and Mute Switches

A solo switch is a switch that activates the selected channel audible through a master bus by
silencing all other channels. The feature is very helpful to analyse a particular sound during
sound mixing.

The mute switch is a switch that activates silence on the output of a particular channel.

8. Subgroup Bus Switches and Main Stereo Switch

Subgroup buses also called group buses or buses. As buses are audio signal paths, subgroup
bus switches allow you to route the signals as per the requirement. These are assignable
buttons.

These buttons are often located near the channel fader and the pan knob, allowing the signal
routing easily. The user will be able to route multiple incoming signals to a single bus or
multiple incoming signals through individual subgroups.

For example, all the incoming signals are routed through the master bus to create a stereo
mix or all the incoming signals are routed through group buses to record separate audio
tracks.

All these assignable buttons are corresponding to relative master faders, located at the
master section of a console. When the user lifts those faders, the audio signal will be available
at the output port of the relative buses as per the amount lifted.

The master output port or control room output port can be connected to the monitor
speakers. And the subgroup out ports can be connected to the individual input ports of
recorders or audio interfaces for multi-track audio recording.

It is possible to record the blend of an entire strings section in a stereo track by choosing the
same subgroup path for all the incoming string instruments to create a strings sub-mix. Or it
is possible to record violins, violas, cellos and contrabass through separate subgroups to
create individual sub-mixes.
The stereo bus switch is also located near the channel fader and pan knob. This allows a user
to route the incoming signal that comes through the channel strip features to the main stereo
output.
Picture Courtesy: Digital Sound and Music
9. Channel Fader

The Channel Fader will be the last point for the incoming audio signal on a channel strip before
it enters the master section as per the signal routing. The channel fader simply boosts or
reduces the incoming audio signal by a certain amount at the final stage and is used to
perform quick level adjustments for tracking and mixing.

Premium consoles have 100 to 120mm faders, whereas entry-level consoles have shorter
faders, about 60mm.

Picture courtesy: promarchive.ru

The unity position of the fader labelled 0, which means that it is boosting/attenuating the
incoming signal by 0dB. When a fader is at unity, it is passing the signal through it, unaffected.

Numbers marked above the unity position are +ve, indicating that they boost the incoming
signal by a specific amount in decibels. Most of the consoles allow their faders to perform 6dB
boosts, and some allow 10dB boosts.

Numbers marked below the unity position are -ve, indicating that they cut/attenuate the
incoming signal by a specific amount in decibels. The lowest number here is infinity, and this
is effectively the same as muting the audio on the channel strip.

Adjusting the level of the signal on the channel strip while the audio is playing or being
recorded is called “gain riding”.

The length of the fader pull-down that performs a 5dB change is the same as the length of
the fader pull-down that performs a 10 dB change between -10 dB and -20 dB.

The difference in attenuation increases as we go down the fader. This shows us that the fader
has a high resolution around the unity point, and that small movement of the fader at this
point results in small amplitude changes.

As we go away from the unity point, the same small movements on the fader can result in
larger changes. The fader should be kept as close to the unity point as much as possible.
 Picture Courtesy: Presonus

It’s very rarely that a fader is taken above the unity point, and it’s something that isn’t
practised and generally avoided. This is because pulling the fader up increases the noise floor
along with an increase in amplitude.

Next Lesson: Next lesson is all about AFL and PFL, very important features that are especially
helpful for live sound operations.

Question:

1. List the features available on the channel strip.

2. What is the use of busses?
3. What is the use of Groups?

Lesson 3: PFL and AFL

Learning Time: 10 min

Learning Outcomes: Students/Users will be able to understand the audio signal streaming
method of Pre-fader and Post-fader listening.

PFL stands for Pre Fader Listen and AFL stands for After Fader Listen. Both are different kinds
of solo buttons.

Instead of soloing the signal of a particular channel by muting all other channels, it simply
routes the audio to some other selected output, such as a headphone for the engineer to use.
This is especially very useful in live sound scenarios, where the engineer might not want to
solo some channel playback on a PA system, but listen to it soloed on a headphone without
affecting the main mix.

In this case, a PFL/AFL output is chosen, usually, a separate output like a headphone output
and any track with the PFL/AFL switched on copies it’s signal to the selected output while
leaving the way in which the channel strip signals are reaching the mix bus unaffected.
PFL allows you to monitor the channel in question's signal level at a point immediately prior
to the channel fader, and will, therefore, include any EQ or dynamics that might have been
applied on that channel. Thus when setting up a channel's input gain using PFL, it's important
to bypass any EQ and dynamics processing, otherwise, you won't know what the actual
headroom is at the front end. On mono channels, PFL is mono. On Stereo channels, PFL should
be stereo, but some cheap desks derive a mono PFL signal for both mono and stereo channels.
AFL is similar to PFL in function, but takes its signal from a point immediately after the channel
fader, showing the level of the channel's contribution to the mix. AFL is also mono on mono
channels.
Solo, more correctly known as Solo-in-Place (SIP), is an after-fade listen taken from after the
pan control as well as the channel fader. It is, therefore, a stereo signal even on mono
channels. The idea is to allow the monitoring of a channel signal when panned to its
appropriate position in the stereo image. SIP is usually achieved by monitoring the main mix
bus and muting all the channels other than the one you pressed the SIP button on.
However, this means that you can't use SIP while mixing because it destroys the mix on the
mix bus, muting aux channels as well as main channels. (PFL and AFL only affect the signal
routed to the monitor outputs.) That's why SIP is often described as 'destructive solo
monitoring'. Usually, you'll want to solo a channel and hear it with any associated effects
returns, so selected channels can usually be made 'safe' from the SIP function so that they
continue to contribute to the mix when all the other channels are muted. A lot of desks have
a single 'solo' button somewhere near the fader which can be configured to provide any or all
of these functions.

Learning reference:

1. https://dbbaudio.com/2017/soloing-afl-vs-pfl/

Next Lesson:

Next lesson is the master section of a console which carries the secondary input and output
functions.

Lesson 4: Master Section

Learning Time: 10 min

Learning Outcomes: Students/Users will be able to understand the audio signal paths which
are linked between the channel strip and master section.

The master section of an audio console can be considered as the other end in comparison
with the input section of the console. The preamp section readies the signal to be used on the
console, and other audio equipment. The master section gives us the final output or mixed-
signal and lets the engineer decide what to do with the signal. There are many routing options
available here as well.

All the master controls are available here:

a. Auxiliary Send Master Controls

As we discussed earlier at the auxiliary section, sending audio signals from the channel strips
via auxiliary buses that reach to the auxiliary master section as customised cue mixes, audio
signals to the time-based effect processors etc. The auxiliary master controls of different
auxiliary buses are located in the master section. The user will be able to control the amount
of auxiliary master sends corresponding to the output port of the related auxiliary buses that
are located mostly in the rear panel of the console.

The user will be able to establish connections between the master auxiliary send and time-
based effect processors, headphone amplifiers, in-ear systems and floor monitors.

If the channel strip has 6 auxiliary buses on board, the equivalent numbers of auxiliary master
controls will be available at the master section of a console.
 Picture Courtesy: TOFT

b. Auxiliary Returns

Auxiliary returns bring the wet signals from the time-based effect processors most of the time.
This is an input section you can find there in the Master Section.
 Picture Courtesy:dummies.com

Here the user can decide whether the wet signals to be routed to the subgroups or to the
main mix buses because the incoming audio signal is a wet only signal.

And the user will be able to set the gain of the incoming wet signal by using the auxiliary
returns rotary control.

Picture Courtesy:slideplayer.com
 c. Subgroup Master Faders

At the time of tracking or mixing, the engineer routes the audio signals via subgroups for
multitrack recording or for pre-mixing purposes. The subgroup master faders allow those
signals to reach the subgroup output ports as per the fader position by decibels. Means,
whatever the audio signal that routed through the subgroup will be controlled by this
subgroup master faders.

The user will be able to listen to a specific subgroup signal by pressing an AFL button that is
associated with each subgroup fader. This subgroup master faders are located in the master
section near the master mix fader.

The user can establish the connections between the subgroup output ports and input ports
of a multitrack recorder or an audio interface. Insert options are available for all subgroup
outs in premium consoles.

d. Master Fader

A master Fader is a fader that controls the overall level of a summed audio output during
mixdown is called a master fader.

The blend of multiple channel outputs is coming to the master fader when it's routed to the
master fader. There will be an insert port corresponding to the master output port, that allows
you to make modifications by inserting a compressor or a graphic equalizer to apply the
effects to the entire mix if required.

Learning reference:

1. Setting your home recording studio

https://www.youtube.com/watch?v=qmoyRRZUQPQ&feature=emb_logo

Next Lesson:

Next lesson mentions what are the different types of Consoles.

Questions:

1. What is the purpose of Aux sends master control?

2. Why are returns important?

Lesson 5: Types Of Consoles.

Learning Time: 20 min

Learning Outcomes: Students/Users will be able to understand different types of consoles

which are equipped with different features and methods of the workflow.

As per the features and specifications, the consoles can be classified mainly into:-
1. Split Consoles
2. In-Line Consoles
3. Broadcast Consoles
 4. Live Consoles

1. Split Consoles

Audio consoles used in sound recording studios take inputs from various sources, route those
input signals to a recording device such as a magnetic tape multi-track recorder, digital multi-
track recorder or multi-track audio interface.

This is for the recording process and known as Tracking.

The audio from the recorder or from a digital audio workstation is then brought on to the
console again, to adjust the levels and tweak the recorded audio, summed as a “mix”. This
process is called Mixing.

Consoles that allow tracking and mixing on separate sections of the console are called Split
Consoles.

A split console usually contains a group of channel strips for tracking and mixing separately.
In between the tracking and mixing section, the master section will be placed.

Picture Courtesy: www.ballstatemusic.com

Advantage: At the time of tracking, the user will be able to assess the tape signals equalised
to ensure the end tonal result.

Disadvantage: The user will not be able to utilise the entire channel strips for the mixing
process.

2. In-Line Consoles

A console that allows tracking and mixing through the same channel strip/section is called In-
Line console. In-Line Consoles start from the range of premium consoles. The basic concept
of an in-line console is that input and monitoring functions are combined on the same channel
strip rather than having separate sections.
 Picture Courtesy: Audient Consoles

The benefits of an in-line console are that, during recording, the Tape(or DAW) returns are ‘in
line’ with the input section of the channel strip. During Mixing, the channels are all available
as inputs, allowing all tracks to be mixed from all available inputs.

Yes, that is one of the main advantages, that the user will be able to utilize the entire channel
strips during the mixing process.

An In-Line Console allows establishing a signal chain with a patch bay which helps to insert
outboard gears during tracking and mixing perfectly.

Picture Courtesy: Audient Consoles

The console that shown above is a small format in-line console.

It has 8 group buses and the entire in-line features that allow it to utilize Preamps, 4band EQs
etc.
3. Broadcast Consoles

Consoles are designed for specific purposes, and these changes have a lot to do with routing
options. Broadcast consoles are used to mix audio for radio, television and internet
broadcasts.

Audio owes a lot to the broadcast tradition since it was one of the first applications of audio
processing. The first recording consoles were broadcast consoles, which started undergoing
modifications that eventually turned the broadcast console into an audio console.

Broadcast consoles differ themselves between their different applications, so a console used
for TV will be different from the one used for radio. There are, however, ways in which
broadcast consoles are similar to each other in a way that’s different from recording consoles.

a. Broadcast consoles don’t have to have recorder returns on/for every channel, because
these aren’t required. Certain channels of audio are broadcast live, and others are played
back from a recorder or audio interface, but unlike a recording console, the final output is not
going to the master bus from the tape returns or audio interface. Direct outs on all channels
are also optional since no audio is going to the recorder or DAW for us to be able to choose
the simplest, least noisy path for it.

b. Broadcast consoles have the ability to perform what’s called a ‘mix-minus’, also called
‘clean feed’ or ‘select audio return’. A mix-minus setup ensures that a mix outputted to a
certain device does not contain the output of that device. It is simply the console MIX, MINUS
the input of the console.

Picture Courtesy: Audio Technology Magazine

For Example, imagine a caller who has called into a radio station using his/her
telephone. The telephone is used to send audio to the broadcast station from the
telephone’s microphone/mouthpiece, and to receive audio from the broadcast
station, which is heard through the telephone’s earpiece. We will call this the
bconsole-out or tele-in. Now, getting the audio from the caller’s microphone (tele-
out) into the broadcast console and then broadcasting it is a fairly simple process.
 We can also route the studio audio to the caller by routing the RJ’s voice on the console
to a bus that feeds the output on the broadcast console(bconsole-out). This is as
simple as routing the broadcast console’s output to console-out, but this will also
contain the caller’s own voice, which will reach him/her with a certain amount of
delay, which can be distracting.

Mix-minus allows the engineer to phase invert a signal on it’s way to a particular
output bus, so that its phase cancels the signal that isn’t to be heard, which, in case, is
the caller’s own voice. This is a common need in broadcast, and it’s very often that a
signal needs to be mix-minus, like in news that reaches the broadcast studio from a
remote location.

c. Broadcast consoles don’t have as many auxiliary sends and returns as recording
consoles do. This is because there aren’t many mixes needed.

d. The EQ section is far less detailed, with fewer parameters, and often only contains
switchable low and high cut filters.

e. A tone generator is usually available on the console for setting levels. This is a
luxury on recording consoles and only the most expensive ones have this additional
feature. But a tone generator is a must on broadcast consoles, even cheap ones.

f. The level meter is often far less detailed and is mostly found on the master bus only.
This has to do with the limited dynamic range.

g. Broadcast consoles also have a ‘detent mode’, so if you have your channel
fader all the way down, pulling it up slightly will send the channel signal to a separate
auxiliary bus that can be monitored separately, so the engineer can hear what she/he
is about to bring up into the broadcast mix.

In the present, technology has changed a lot towards easy to work or user-friendly ways.

IP based broadcast consoles have introduced an OTT (over the top) workflow being
commonly used.

Live Consoles :

Live sound consoles are used in sound reinforcement for public address systems.

As the name suggests, a live sound console is structured to support live events.
Instruments, vocals or speeches present on the stage called stage outputs, which are
plugged into the console as input sources and those signals routed through different
parameters of equalizers, dynamic processors added with time-based effects finally
routed to a mix output. The mix output goes to the main speakers.

a. Live sound consoles have matrix outputs. A live sound console isn't only expected to
give us FOH mix, but also monitor mixes to the artists on stage. Performing artists need to
hear themselves clearly, and since the FOH speakers are placed facing away from the artists
on the stage, they can not provide the musicians with a reliably clear sound, and the
musicians' performance often depends on how well they can hear themselves play. On-stage
monitors are provided for this purpose. Moreover, each musician requires his/her own mix,
so the drummer might want only the drums on his/her monitors, but the guitarist might want
everything except for the drums (which are loud and can be heard quite clearly on stage
without any additional amplification) and second guitar (hearing another guitar in their
monitor mix is something most guitarists find very distracting).

The live sound console is designed to output many different mixes, called matrix
outputs. One matrix out (and sometimes several) go out to the PA system, with other
matrix outputs going to different monitor mixes. Often, just one mix is sent to a
separate monitor console on stage, where a second engineer uses another, smaller,
live sound console to create monitor mixes for the musicians on stage.

b. Live sound consoles must be built to last. Live venues are dusty, hot, and it isn't beyond
the realm of possibility for it to rain. Moreover, live sound venues are often powered
by generators, which makes the power supply unreliable, and prone to fluctuations.
Live sound consoles must be built to handle these conditions, should be dustproof,
able to perform normally, hot or cold, and should be able to handle minor power
fluctuations.

c. Live sound consoles are also handled roughly. They are often shockproof and come
with handles built-in to the sides.

Picture Courtesy: Avid

d. Live sound consoles have AFL/PFL functionality and sometimes have detent mode
built into them to avoid the engineer bringing in audio that doesn't have to be played
back through the PA.

e. Live sound consoles may or may not have a detailed EQ and dynamics section, but
they almost always have a very basic noise gate on all channel strips, because not all venues
have perfect electricals, and often have grounding issues.

References:

1. The Audio Mixer: Key Features https://producelikeapro.com/blog/audio-mixer-key-

features-functions/
 2. Lawo designs and manufactures advanced networking, audio, video and control
technologies for broadcast production, on-air, live and theater applications.

Questions:

1. Why are the live consoles not having in-line features?

Lesson 1: Cables and its Types
(Reading Time: 10 minutes)

Learning Outcome: Different types of Cables and Connectors and their design.

Overview: An introduction to different types of cables and connectors used in audio signal
flow and understand the balanced and unbalanced audio.

Introduction

A cable includes at least two wires braided side to side plaited together to shape an insulated
assembly called a single unit cable. A cable is used to convey an electric current, and sound
cables are utilized to carry sound signals, which are alternating current (AC).

One of the types of cables utilized in sound is called ‘Sheathed Multi-core Cables’, which have
a progression of inward conductors, encompassed by a protecting layer of the rounded
leading shield, encompassed by a coat called an insulator.

Some audio cables require one centre wire (internal conveyor) encompassed by a coincided
shield. This is known as a ‘Single Core Cable’. Some audio cables are ‘Double Centre Links’.

Cables and Types of Cables

Sound cables are generally made of two protected conductors (wires) encompassed by a wire-
work shield which reduces the hum or noise in cables. Outside the shield is a delicate plastic
or elastic protecting coat.

(P.C - A beginners guide to audio cables and signal processing)

Balanced and Unbalanced Audio

Balanced audio uses three conductors to carry signals. Two of the conductors convey
negative and positive signals (sound is an AC signal), and the third is utilized for establishing
the ground to them (earthing). With an unbalanced signal, there are just two conductors.

Generally, +4 dB balanced signals are suitable for longer runs. The high gain and isolated
ground make it good for a cleaner, noise-free signal.

If cable runs are shorter, there should be no problem with RF interference or noise on
unbalanced, -10 dB signals. If you have RF problems in your studio, you probably have some
kind of ground fault, and it's a good idea to have it checked out by an electrician.

Typical studios have both +4 and -10 dB gear, so the important thing here is to make sure gain
stages match throughout the signal chain. The biggest cause of noise and distortion in audio
is mismatched gain.

Cable Connectors And Chassis Connectors

Some connectors are part of cables; they are called ‘Cable Connectors’. Other connectors are
built into equipment chassis; they are called ‘Chassis Connectors’. Cable connectors mate with
(plug into) chassis connectors.

Several types of connectors are used in audio. We'll describe them below.

The 1/4-inch phone plug is used to connect unbalanced line-level or instrument-level signals.
This plug is part of a cable used with guitar amps, mixers, electric keyboards, electric guitars,
and some power amplifiers. A guitar cable has a phone plug on each end.

The tip of the plug is soldered to the cable's centre conductor; sleeve or long cylinder is
soldered to the cable shield.

The RCA or phono plug is also used to connect unbalanced line-level signals. It's commonly
seen in stereo equipment. The centre pin is soldered to the cable's centre conductor; the cup
is soldered to the cable shield.
A 3-pin professional audio connector (XLR) is used with cables for balanced mics and
balanced equipment. The female connector (with holes) plugs into equipment’s output. The
male connector (with pins) plugs into equipment’s input.

XLR connector has 3 pins (male) or 3 holes (female). The pins or holes are numbered 1, 2, 3.
In any XLR connector, pin 1 is soldered to the cable shield; pin 2 is soldered to the "hot" lead
(usually red), and pin 3 is soldered to the remaining lead.

How do you know if a 1/4" phone jack is balanced or unbalanced, mono or stereo?

You need to look at the specifications in the equipment manual. For example, if the jack is
labelled "AUX SEND", look up the specification for the AUX SEND connector in the manual. It
will tell you if it's balanced or unbalanced.

For headphones, the tip is soldered to the left-channel lead (wire); the ring (just behind the
tip) is soldered to the right-channel lead, and the sleeve is soldered to the common lead. For
balanced line-level cables, the sleeve is soldered to the shield; the tip is soldered to the "hot"
lead, and the ring is soldered to the remaining lead.

Some mixers have INSERT jacks that are stereo phone jacks; each jack accepts a stereo phone
plug. The tip sends a signal to an audio device input; the ring returns signal from the device
output, and the sleeve is ground. These signals are unbalanced.

If your recorder or mixer has unbalanced mic inputs, but your mic and mic cable is balanced,
put/fix up an adapter cable with a female XLR on one end and a 1/4" phone plug on the other
end. It's called a female XLR to 1/4" adapter cable.

Speaker Cable:

This cable connects a power amplifier to loudspeakers. A typical assembly is a banana plug to
banana plug with a 2-wire lamp cord between them. Or, each end of the cable might have a
Speakon connector, phone plug or bare wires.
Speaker cables are normally made of unshielded lamp cord (zip cord). To avoid wasting
power, speaker cables should be as short as possible and should be heavy gauge (between 12
and 16 gauge). Number 12 gauge is thicker than 14; 14 is thicker than 16.

Unbalanced Patch Cord (mono patch cord)

This is a phone-to-phone cable with a 1-conductor shielded cable, which is the same as a
guitar cable. It is used to connect unbalanced equipment. For example, patch cords can
connect a mixer to external devices such as an effects unit, recorder, equalizer, power
amplifier, etc.

Balanced Patch Cord (stereo patch cord)

Stereo phone to stereo phone (Fig. 20). Also called TRS to TRS. There's a 2-conductor shielded
cable between the plugs. It can be used to connect balanced equipment.
It is also used as an ‘INSERT’ cable for effects. The tip sends a signal (unbalanced), the ring
returns signal (unbalanced), and the sleeve is the common ground for send and return.

Some Useful Common Cable Nomenclature

The instruction manuals for your equipment tells how to connect your components. Usage of
shorter cables reduces hum but should be long enough to be able to make changes.

These connections are okay:

• A Phono plug to a phono jack.
• A Stereo phone plug to A stereo phono jack.
• RCA plug to RCA jack.
• XLR male to XLR female / XLR female to XLR male.
• Mic level to mic level / Line level to line level.
• Mic to the mixer mic input. Use an XLR mic cable.
• Mic to mic preamp input. Use an XLR mic cable.
• Mixer aux send to effects input. Use a balanced or unbalanced phone-to-phone.
• Effects output to mixer aux return or FX return. Use a balanced or unbalanced phone-
to-phone.
• Mixer main out to power amplifier line in. Use a balanced or unbalanced phono-to-
phono or XLR mic cable.
• Mixer main out to a powered speaker line in. Use a balanced or unbalanced phono-
to-phono or XLR mic cable.
• Mixer monitor sent to monitor power amplifier line in. Use a balanced or unbalanced
phone-to-phone cable.
• Power amplifier out to the speaker. Use a speaker cable.
• Speaker to the speaker (to play two speakers from one amplifier output). Use a
speaker cable.

Question:
1. Differentiate between balanced and unbalanced audio.
2. Describe a structure of a cable using a diagram.

Citation:

1. Bartlett, Bruce. “Beginner's Guide to Audio Connections.” cdn.shopify.com.

https://cdn.shopify.com/s/files/1/0247/3799/files/beginners_guide_to_audio_conn
ections.pdf (accessed July 17, 2020).

Next Lesson:

In the next lesson, you will learn different types of digital cables and connectors used in the
audio signal flow.

Lesson 2: Digital Cables

(Reading Time: 5 minutes)

Learning Outcome:
 Understanding of different types of digital cables and connectors and their design.

Overview: An introduction to different types of digital cables and connectors used in the
audio signal flow.

Digital Cables

We shall discuss some common cables and connectors that we find in the world of digital
audio.

Optical Cable (Toslink)

A fibre-optic connection used to impart advanced audio signals from a source segment to a
sound processor, for example, an A/V recipient. Impenetrable to basic electrical, magnetic
and RF obstruction, it offers incredible speed and transmission capacity, however, links are
generally more expensive than coaxial links which are utilized for a similar reason. As far as
execution, there is nearly nothing (if any) perceptible distinction in sound between the two.

(P.C - www.precsound.com)

Coaxial Cable (SPDIF)

An advanced interface design developed by Sony and Philips (henceforth, SPDIF) is used to
impart computerized sound signs from a source part to a sound processor, for example, an
A/V recipient. Coaxial alludes to the link structure and not the connection technology, which
sadly results in some disarray with other video and systems administration links that utilizes
a coaxial structure. For digital sound applications, coaxial cables are more practical than
optical and similarly as skilled, however, can be influenced by electrical obstruction.

(P.C - www.precsound.com)

HDMI

A digital connection that can carry both digital video and audio signals over a single cable.
HDMI can support up to eight channels of audio at multiple sampling rates, and cables can
extend for as long as 15 meters. HDMI is used primarily as a video connection, but the
specification allows for audio, video and control data, and future products will incorporate
these functions.

(P.C - www.precsound.com)

IEEE (Firewire)

IEEE-1394 is capable of high-speed throughput of up to 800mbps and is fast enough for digital
audio transfer and compressed video. In an audio context, 1394 sends digital audio signals
from a source component to an audio processor, such as SACD and DVD-A signals.

(P.C - www.precsound.com)

Ethernet

Ethernet is the most common connection in home networking. Cables are terminated in a
plastic modular plug called an RJ-45. This is a slightly larger version of the common telephone
plug, the RJ-11. Ethernet plugs are even used in wireless networks, to connect wireless NICs
to devices that don't have NIC capabilities built-in. Ethernet cables are ranked by "category",
with higher CAT numbers offering greater throughput.

(P.C - www.precsound.com)

USB
A digital connection designed as a "plug-and-play" standard for PC peripherals and network
devices. The original implementation (USB 1.1) offered only 12mpbs of throughput, but the
current USB 2.0 offers 480mpbs. Newer USB 2.0 devices are backwards-compatible with USB
1.1 devices, but only at the latter's slower throughput.

(P.C - www.precsound.com)

Questions

1. What is a coaxial cable?

2. Explain SPDIF, Firewire and Ethernet Cables used in Digital Audio.