La Martiniere College, Lucknow

Digital Academic Services

Class 12 Subject Mass Media and Communication

Name/Number of Chapter: Chapter 4 Recording Radio Programs

Module Number 11
Dates From: 22nd July, 2021 To: 28th July, 2021

Topic: Transmission Equipment (Part 2)

Objective: There are three main objectives:
1. To introduce students to Radio Production.
2. To explain how a radio station is set up.
3. To explain the various equipment used in radio production.

Notes:
Amplifier
Amplifiers may be classified in a number of different ways: according to bandwidth (narrow or wide); frequency
range (audio, intermediate, or radio frequency); or output parameter requirement (voltage or power).
Wide-band radio-frequency amplifiers are not needed for audio signals unless a frequency-modulated system is
used. Amplitude-modulated signals for sound broadcasting should have a radio-frequency bandwidth of ±10
kilohertz though on medium waves it is often limited to ±5 kilohertz (total bandwidth of 10 kilohertz). High-
quality frequency-modulated audio needs a bandwidth of about ±100 kilohertz.
The main problem with radio-frequency amplifiers in receivers is the possibility of cross modulation—that is, the
mixing of two information channels, which can occur if an undesired modulated signal enters the radio-frequency
input together with the desired signal.
Amplifier is the generic
term used to describe a
circuit which produces
and increased version
of its input signal.
However, not all
amplifier circuits are
the same as they are
classified according to
their circuit
configurations and
modes of operation.
In “Electronics”, small signal amplifiers are commonly used devices as they have the ability to amplify a
relatively small input signal, for example from a Sensor such as a photo-device, into a much larger output signal
to drive a relay, lamp or loudspeaker for example.
Amplifiers can be thought of as a simple box or block containing the amplifying device, such as a Bipolar
Transistor, Field Effect Transistor or Operational Amplifier, which has two input terminals and two output
terminals (ground being common) with the output signal being much greater than that of the input signal as it has
been “Amplified”.
An ideal signal amplifier will have three main properties: Input Resistance or (RIN), Output Resistance or
(ROUT) and of course amplification known commonly as Gain or (A). No matter how complicated an amplifier
circuit is, a general amplifier model can still be used to show the relationship of these three properties.

Sound Mixer
An audio mixer is a device with
the primary function to accept,
combine, process, and monitor
audio.
Mixers are primarily used in
four types of environments:
live (at a concert), in a
recording studio, for broadcast
audio, and for film/television.
An audio mixer can come in
either analog or digital form.
There are two builds of mixers
• In-Line

An in-line mixer means there are two paths per channel. For example, in a recording environment, an in-line
mixer can use the same channel to send and receive sound to and from a digital audio workstation (DAW).
• Split Monitor

A split monitor console has one path per channel. Each channel can be used to either send or receive sound to or
from a DAW. Both of these builds have two areas for us to explore: the channels and the master section.

An audio mixer may also be referred to as a “console,” “desk,” or “board.” All three of these terms are synonyms
with “mixer.” Additionally, we must define “signal.” Signal is the generalized term for any audio passing through
a mixer. This could be a vocal, drum, bass, synth, guitar or another instrument. All of it is referred to as signal.
Regardless of the application, build or form of the mixer, most are structured in a similar fashion. Let’s examine
the parts, features, and prices of audio mixers.

Parts of a Channel
• Input Section

The input area of the channel strip may

accept any or all of the following levels
of signal: mic, instrument, -10 line level
or +4 line level. Depending on which level of signal is desired to be accepted, specific knobs and/or switches
must be engaged and used. For example, if a microphone is wired into a channel, the channel must be configured
to accept mic level signal. Then, the mic pre knob must be raised in order to add gain to the signal.
Wiring a guitar, bass or another instrument may use a separate input jack (probably for an “unbalanced” ¼”
cable). In this case, there may be a switch or button labeled “DI” in the input section. If so, pressing this button
will tell the mixer to accept the instrument level signal (rather than mic level or line level).
The mixer will also be able to accept +4 line level, -10 line level or both. This normally takes the form of
deselecting all switches and buttons but may vary between consoles. Some mixers have a knob referred to as the
“line trim.” This serves the same purpose for line level signal as the mic pre does for mic level signal: it adds or
subtracts gain at line level.

• EQ

Most mixers have an equalization (EQ) area on the channel

strip. The amount of flexibility and precision of frequencies
that can be adjusted in the EQ area is typically related to the
overall sophistication of the board. For example, most budget
and moderate-level mixers have fewer frequencies that can
be targeted and adjusted with broad strokes, while high-end
desks have EQ areas that can be used as a scalpel.

• Dynamics

High-end mixers commonly have a “Dynamics” area, either on each channel or in the master section. Common
dynamic effects included in this area are compressors and gates. When a switch is engaged to activate the
dynamics area, the compressor will make the voice or instrument sound more even (loud parts become softer,
soft parts become louder). The gate will open to allow any signal to pass through the channel, then close to keep
out background noise and bleed.

• Fader

The fader is the device that raises or

lowers the amount of audible signal
from the channel. A fader is actually a
resistor: as it is lowered, it increases
resistance. This is why the numbers
start high at the bottom of the fader bank
on each channel and decrease to the
zero point (usually located about two-
thirds of the way to the top).
These numbers indicate a general idea
of loudness (more advanced users will
recognize these numbers as the logarithmic decibel scale). The most important number on the fader bank is zero.
The zero point on the fader is referred to as “unity,” meaning there is no resistance applied to the channel’s signal.
Any fader position above zero is amplifying the signal as opposed to varying the amount of resistance as it is
moved up or down below zero.
 • Group Faders

Group faders on an audio mixer are used to control multiple channel faders at once. A common use for a group
fader is to raise or lower all of the faders that contain drum signals together at once — the collective mix of the
drum faders can be raised or lowered using a group fader (also referred to as a subgroup). If included in a console’s
construction, there may be anywhere from two to eight group faders.
In many cases, these group faders are actually voltage-controlled amplifiers, meaning there is no audio passing
through them. These are commonly referred to as VCA Group Faders.

• Auxes

Mixers have a dedicated area for auxiliaries (auxes) used to send a copy of the channel’s signal to another
destination. In the studio environment, auxes are used for creating a headphone mix or adding a time-based effect
(such as reverb or delay). On a live console, auxes are used for time-based effects, but also may be used to send
sounds to in-ear monitors or a monitor wedge on stage.
These copies can be taken before or after the fader, referred to as “pre-fader” or “post-fader” auxes. A pre-fader
aux sends the same amount of level to the new destination regardless of the fader’s position. A post-fader aux
maintains the “wet-to-dry ratio.” This means each post-fader aux has its own level control and the effected signal
can be raised and lowered with the fader.
With this in mind, application of these auxes becomes easy. Pre-fader auxes are used for headphone mixes so
adjustments to the fader are not heard in the performer’s headphone mix. Post-fader auxes are used for time-
based effects so the amount of the effect (reverb, etc.) stays consistent with respect to the fader level.

• Bus Assignment

A bus is no more than a path on which signal can travel. This is relatable to any bus driving down the street —
and every bus needs a destination. (Here in Los Angeles, the buses stop before they drive into the ocean…most
of the time.) The bus assignment area on a mixer is the same concept: we are sending signal down a path that
leads to a destination. Common destinations are external pieces of gear, audio subgroups or an audio interface.
The most common bus assignment is the stereo bus. The stereo bus is a two-channel mix (left and right) of all the
faders on the console. It can also be thought of as the sum of all the faders. Most consoles require a button or
switch to be engaged in order to send the channel to the stereo bus (and therefore be audible in the mix).

• Track Busses & The Routing Matrix

Track busses are a collection of paths for our signal to travel. If the mixer is equipped with track busses, there are
commonly anywhere from four to sixteen located on the top of each channel. The busses can be routed to an
audio interface for recording, other channels for summing or to a piece of outboard gear (such as a compressor).
More advanced consoles contain anywhere from twenty-four to forty-eight busses and additional routing options.
If so, this area is referred to as “the routing matrix.” A routing matrix possesses the same functionality as the
track busses but expanded routing and destination options are available.
 • Pan

The purpose of the pan pot (short for potentiometer) is to pan a channel’s signal left or right across the stereo bus.
An in-line console will have two pan pots, each assigned to a designated path. A split monitor desk will have one
pan pot, normally located above the fader.
Panning information can also be translated to track busses or the routing matrix: if more than one bus is selected
on the channel and the fader has been assigned to those busses, the stereo image of the signal will be retained.

• The Master Section

The master section provides areas for global adjustments to the channels or modes of the console. It is typically
located in the middle of the desk. Some manufacturers refer to the master section as the “Centre Section.”
Common adjustments made in the master section are toggling which level of signal the channels are accepting,
master levels for auxes and busses, toggling the control room source (what is playing through the speakers),
changing the control room level (volume of playback) and speaker selection. There also may be a foldback area
(used for configuring headphone mixes) and stereo effect returns (a destination for the return of outboard
equipment).

• Patch Bay

A patch bay is a device located next to an audio

mixer that has a series of jacks meant for moving
signal from one place to another on the console.
The jacks are organized by row and offer
increased flexibility.
Additionally, they serve an organizational
purpose. Cables from the mixer and outboard
equipment can be wired directly to the patch bay
so signal can be routed easily without having to
connect cables directly to a device. Patch bays
are most common in the studio environment but
can be used in other situations as well.
Flipping the polarity on one of the channels allows us to hear the sum frequency response of both channels and
verify there are no phase issues detracting from the sound. Make sure to use whichever polarity position produces
a full, thick sounding result.

• Phantom Power

A Phantom power button is present on each channel of most every mixer. Phantom power is required for
condenser microphones: it adds 48 volts of direct current via the XLR cable used to wire the microphone to the
mixer. The phantom power button may be labeled “48,” “+48,” “+48v” or “phantom power.”
 • Polarity

A button on most mid-level and high-end mixers is a polarity flip, commonly using the Ø symbol. Flipping the
polarity on a channel changes the phase relationship. Phase is defined as a time relationship between two
waveforms.
An inverse phase relationship can cause frequency builds and cancellations, which will negatively affect the
sound. Anytime there is more than one mic on the same source (e.g. top and bottom snare), we must flip the
polarity on one of the channels in order to check the phase relationship.
Flipping the polarity on one of the channels allows us to hear the sum frequency response of both channels and
verify there are no phase issues detracting from the sound. Make sure to use whichever polarity position produces
a full, thick sounding result.

• PAD

Many mixers have a button labeled only with a number rating, such as “-10” or “-10dB.” This is the PAD for the
channel. PAD is an acronym for “Passive Attenuation Device,” which softens the sensitivity of the capacitor
inside of the mic, allowing louder signals to pass through the channel without distortion. The number rating (eg
-20dB) is the strength of the PAD, measured in decibels. The higher the decibel rating, the stronger the PAD.

• Filter

High-pass filters are available as a button on the channels of most mixers. The high pass filter will dramatically
soften the low-end frequency response of the channel and is useful for decreasing or removing rumble from a
signal. Low-end mixers may only have a symbol that looks like this:
Moderate and high-end consoles typically have a numerical value next to the symbol. The numerical value
represents the frequency at which the filter begins.

• Meters

Various styles of meters exist on mixers and for

audio production in general. On a mixer, the
collective area of the meters across all of the
channels is referred to as the meter bridge. Three
common meters that may be found on a mixer
are VU, Peak, and RMS.
VU meters display the level of perceived
loudness on a channel or the stereo bus, with
signals far across the zero point likely to distort.
Peak meters, which are most familiar in modern
times, indicate the loudest part of a signal at any
instant. RMS meters display the average loudness of a channel or the stereo bus, indicating the dynamic range of
the signal when compared to the zero point.
 • Digital vs. Analog

Audio mixers are manufactured in either analog or digital form. Each of these console types has its own set of
advantages and disadvantages, which are directly related to one another.
The biggest advantage digital consoles hold over an analog mixer is instant “recall.” This means a mix (or setup)
can be reloaded to the exact parameters from when it was last saved. Every knob, switch, button, and fader will
snap to its saved position. Analog consoles have to be recalled manually, meaning each knob, button, switch, and
fader has to be documented and returned to its original position by hand. This can be a time-consuming and
tedious process.
The main advantage an analog desk holds over a digital mixer is “summing.” Summing is the process of
combining signals from all the channels using analog circuitry: iron, wires, faders and electrical components.
Analog summing adds a desirable dimension to signal and is very familiar to listeners.
Digital mixers do not sum signal in the same way: they are merely processing the signal using digital, binary code
(1’s and 0’s). The sonic texture of analog summing is so popular that devices called “summing mixers” have
become very popular in home and small recording studios.
A summing mixer can be thought of as an analog console minus the channels and features. They most commonly
take the form of a unit that has sixteen analog inputs. In some summing mixers, each input has its own volume
and pan pots. Signal passes through the analog circuitry and is summed to a physical stereo bus in the same
fashion as an analog console, adding a desirable quality to the vocals and/or instruments.

Speakers
Studio Monitor Speakers provide an easy way to hear what’s going to air without headphones. Often, these are
very high-quality speakers so any abnormalities in sound quality can be detected.
Speakers are one of the most common output devices used with computer systems. Some speakers are designed
to work specifically with computers, while others can be hooked up to any type of sound system. Regardless of
their design, the purpose of speakers is to produce audio output that can be heard by the listener.
Speakers are transducers that convert electromagnetic waves into sound waves. The speakers receive audio input
from a device such as a computer or an audio receiver. This input may be either in analog or digital form. Analog
speakers simply amplify the analog electromagnetic waves into sound waves. Since sound waves are produced
in analog form, digital speakers must first convert the digital input to an analog signal, then generate the sound
waves.
The sound produced by speakers is
defined by frequency and amplitude. The
frequency determines how high or low
the pitch of the sound is. For example, a
soprano singer's voice produces high
frequency sound waves, while a bass
guitar or kick drum generates sounds in
the low frequency range. A speaker
system's ability to accurately reproduce
sound frequencies is a good indicator of
how clear the audio will be. Many
speakers include multiple speaker cones
for different frequency ranges, which
helps produce more accurate sounds for
each range. Two-way speakers typically
have a tweeter and a mid-range speaker, while three-way speakers have a tweeter, mid-range speaker, and
subwoofer.
Amplitude, or loudness, is determined by the change in air pressure created by the speakers' sound waves.
Therefore, when you crank up your speakers, you are actually increasing the air pressure of the sound waves they
produce. Since the signal produced by some audio sources is not very high (like a computer's sound card), it may
need to be amplified by the speakers. Therefore, most external computer speakers are amplified, meaning they
use electricity to amplify the signal. Speakers that can amplify the sound input are often called active speakers.
You can usually tell if a speaker is active if it has a volume control or can be plugged into an electrical outlet.
Speakers that don't have any internal amplification are called passive speakers. Since these speakers don't amplify
the audio signal, they require a high level of audio input, which may be produced by an audio amplifier.
Speakers typically come in pairs, which allows them to produce stereo sound. This means the left and right
speakers transmit audio on two completely separate channels. By using two speakers, music sounds much more
natural since our ears are used to hearing sounds from the left and right at the same time. Surround systems may
include four to seven speakers (plus a subwoofer), which creates an even more realistic experience.

Audio Recording
Audio recording is the process by which sound information is captured onto a storage medium like magnetic tape,
optical disc, or solid-state drive (SSD). The captured information, also known as audio, can be used to reproduce
the original sound if it is fed through a playback machine and loudspeaker system.
Following is the basic process for creating an audio recording:
• Sound waves are converted into electricity using a transducer. (Common transducers include
microphones, tonewheels, and pickups.)
• The electronic information produced by the transducer is stored via computer program or—many years
ago—a tape recorder.
• The captured information—audio—is made audible via playback machines and loudspeaker systems.

The first step in the recording process is to convert musical sounds into electricity using microphones and other
transducers. Next, the audio is routed through an analog-to-digital converter and into a DAW for mixing and
processing. Last, the captured music is finalized by the DAW as a single audio file featuring two distinct channels:
a left and a right.
Signal processing, which usually comes after editing, consists of running your captured audio through electronic
components meant to adjust volume level, dynamic character, and frequency response. Common signal
processors include compressors, limiters, equalizers, reverbs, delays, expanders, and gates. If done well, signal
processing will result in a more refined and brilliant character of audio. In fact, signal processors are largely
responsible for the hyper-real and three-dimensional style of audio common to modern pop music.

Guidance:

Parents are requested to read the extract along with the student before beginning the following exercises. Care
must be taken that the student first learns and understands the material, and does not answer the exercise work
by copying the text.

Learning the various subheadings is crucial as it allows for the student to remember the points that must be further
elaborated on.

Students may compare the notes with real life examples around them to better understand the concepts.
Exercises:

Q1. The following are to be treated as 2 mark questions (2 short points per answer or 2 sentences for a single
point):

1. What is an in-line mixer?

2. What is a split-monitor mixer?
3. Provide a brief explanation of an EQ?
4. What is the use of a fader?
5. Explain the process of basic audio recording.

Q2. The following are to be treated as 5 mark questions (5 sub headings with 2 lines of explanation for each, or
one short paragraph containing 5 elaborated key points, or broken into segments of marks adding up to 5):

1. Provide a distinction between digital and analog mixers.

2. Provide an understanding of “Speakers”.

Resources:

(Note: there is no prescribed text book during this period. All resources mentioned re suggestions or aids, but are
clear in their explanation)

Suggested reading material: Mass Communication in India (Keval J. Kumar)