COURSE: DIPLOMA IN BROADCAST JOURNALISM

UNIT: SOUND PRODUCTION

LESSON 1: INTRODUCTION TO SOUND PRODUCTION
LECTURER: EUNICE WAITIKI
OBJECTIVES: At the end of the lesson the student should:

 Understand what sound is

 Understand the different types of waves

 Learn the different factors that determine the quality of the sound

 Learn the different hardware resources in sound production

WHAT IS SOUND?

Sounds are waves of air. We hear sounds because our ears are sensitive to these waves. Sound is a
response of the ear to small changes in the air pressure.

Vibrations transmitted through solid, liquid, or gas are capable of being detached by human organs of
hearing. One of the easiest types of sound waves to understand is a short, sudden sound like a hand
clap. When hands clap together, they create a Pressurized Wave of air which moves at about 340 meters
per second (“the speed of sound”). When this wave reaches ones ear, it pushes on the eardrum slightly,
causing it to vibrate and the individual hears the clap.

A hand clap is a short event that causes a single wave of air that quickly dies out. The image above
shows the shape of the wave ("waveform") for a typical hand clap. Other sound waves are longer
events. A ringing bell serves as a good example of this. When a bell rings, after the initial strike, the
sound comes from the ongoing "ringing" of the bell. While the bell rings, it vibrates at a particular speed
("frequency") and this causes the nearby air to vibrate at the same speed. This causes waves of air to
travel from the bell, again, at the speed of sound. Pressure waves from continuous vibration look more
like this:

Both of these types of waves are called sound waves or acoustic waves
DIGITAL RECORDING AND PLAYBACK

A microphone consists of a small membrane which vibrates when it meets these acoustic waves. The
microphone translates movements of the membrane into electrical signals. Basically, a microphone
converts acoustic waves into electrical waves. If one examined the shape of an electrical wave from a
microphone, that person would notice that it looks very similar to the shape of the original sound wave.

The following is the electrical wave created by the microphone (notice the measurement for
"Current"):

One could say that the shape of the electrical wave is analogous to ("similar to") the shape of the
original sound wave. This is why these electrical waves that represent sound waves are called analog
waves.

The main device used in digital recording is an Analog-to-Digital Converter (ADC). The ADC measures the
voltage of an electrical wave thousands of times per second. It then uses these measurements to create
a map of an electrical wave:

Each dot in the figure above represents one audio sample. The more samples per second, the more
accurate the mapping of the electrical wave.

Playback of digital audio uses a Digital-to-Analog Converter (DAC). This takes the samples and converts
them back into an electrical wave. In a computer, this electrical wave is sent to a sound card's
headphone or speaker sockets and the speakers recreate the original sound wave by vibrating their
diaphrams.

A computer's sound card comes with an Analog-to-Digital Converter (ADC) for recording, and a

Digital-to-Analog Converter (DAC) for playing audio. Your operating system (Windows, Mac OS X, Linux,
etc.) talks to the sound card to actually handle the recording and playback, and audio applications talk to
your operating system so that you can play sound files, capture sounds to a file, edit them, and mix
multiple tracks while playing, etc.

QUALITY

Two factors determine the quality of a digital recording:

Sample rate: this is the rate which the ADC records samples or a DAC plays them back. Sample Rate is
measured in Hertz (Hz), or samples per second. An audio CD has a sample rate of 44,100 Hz (often
written as "44 KHz" for short).

Sample format or sample size: Essentially, this is the number of digits in the digital representation of
each sample. Think of the sample rate as the horizontal precision, and the sample format as the vertical
precision. An audio CD has a precision of 16 bits.
Higher sampling rates allow a digital recording to accurately record higher frequencies.

Higher sample sizes allow for more dynamic range—better reproduction of loud and soft sounds.

AUDIO FILE FORMATS

There are two main types of audio files on a computer:

PCM stands for Pulse Code Modulation. This is just a fancy name for the technique where each number
in the digital audio file represents exactly one sample in the waveform. Common examples of PCM files
are WAV files, AIFF files, and Sound Designer II files.

Compressed audio files are the other type. Modern compressed audio files use sophisticated
psychoacoustic algorithms to represent the essential frequencies of the audio signal in far less space
than PCM files. Examples include MP3 (MPEG I, layer 3), Ogg Vorbis, and WMA (Windows Media Audio).
Creating one of these files sacrifices some quality in order to use less disk space.

HARDWARE RESOURCES

RECORDERS

Although digital recorders are increasingly the norm, either an analogue or a digital recorder will do the
job. Whichever you use, when you input your recording into your computer it will be converted to digital
sound, which you can then edit with free digital sound-editing software. You can use a minidisk
recorder, any type of digital recorder (DAT or hard disc), a professional-grade analogue recorder, or a
simple “walkman”- style cassette recorder.

If you plan to pitch your piece to on-air radio broadcasters, you should not use a mini-cassette recorder,
because the sound they produce is not of broadcast quality. Two important factors that distinguish
recorders from each other is the presence of a time counter and the ability to adjust sound levels.
Neither are necessities, but both are extremely helpful.

Note that you’ll need a special converter to get your analogue sound into digital format to edit on the
computer.

MOBILE PHONES

Once you have captured your pictures, video and sound on your mobile phone you need to get them
onto your computer in order to incorporate them into your organization’s campaign communications or
your blog post.

Mobile phones typically record sounds using a file format called .AMR, which is primarily designed for
phones and should be transferred onto a computer, and converted for editing. Once the sound files are
on the computer, they can be converted, using a freeware tool like Mobile AMR converter
(http://tiny.cc/ UoB23), into the .WAV or FLAC format, which can then be edited on the computer using
a sound editing tool, such as Audacity (see p. x), or any other audio editing application you already have
access to.

There are various ways to get sounds from your phone to your computer:

1. Bluetooth Wi-Fi Data cable

Bluetooth is a technology which allows two handsets or a handset and a computer within close
proximity of each other to transfer information to each other. Most Bluetooth technology works over a
range of approximately 10 metres. Although newer variants can reach further, up to 100 metres, it’s
most likely that you will use Bluetooth to transfer data off your phone while sitting next to the computer
with the phone. To connect your phone and your computer via Bluetooth you should follow the
instructions on your computer about ‘pairing’ a device via Bluetooth. You have to make sure that
Bluetooth is switched on, on both devices and follow the instructions. If you are transferring data this
way, always remember to switch Bluetooth off when you are finished.

2. A MICROPHONE

It’s best to have an external microphone for recording so you can put it as close to the sound source as
possible. Any standard microphone, uni or Omni-directional, will do. Many recorders have a built in
microphone that is often more than sufficient for non-broadcast quality recording but may not produce
clear enough sound for radio play. If you have to use a recorder with a built-in microphone, be sure to
hold the recorder as close to source of the sound as possible – if it is an interview, hold it relatively close
to the person’s mouth, but be aware that too high an input will create distortion.

3. HEADPHONES

You will need a set of headphones to check sound levels as you record. The headphones enable you to
hear the sound exactly as it is being recorded, and therefore exactly as the audience will hear it. It’s a
good idea to record a minute or so of sound in situ before you start and listen back to it on headphones
to check for problems such as noise, distortion or insufficient level.

COURSE: DIPLOMA IN BROADCAST JOURNALISM

UNIT: SOUND PRODUCTION
LESSON 2: INTRODUCTION TO SOUND PRODUCTION
LECTURER: EUNICE WAITIKI

OBJECTIVES: At the end of the lesson the student should:

 Understand the steps to a great audio

STEPS TO A GREAT AUDIO

Follow the steps below to achieve fantastic audio.

STEP 1 – PLANNING

Creating great audio is not only about sound levels. Planning what you want to produce is the first vital
step, and you should revisit your plan throughout the process of making the audio, to make sure that
you are still working towards what you had planned.

Your plan should answer the following questions: Who is this for? And what is it trying to achieve?
Identify your audience. What is the key message? What do you want listeners to learn / feel / do? What
are the barriers to this audience hearing this message?

STEP 2 – CHOOSING A FORMAT

Choose the right format for your audience and message. Here are some common formats for you to
choose from:

Panel discussion – an interviewer with two or more contributors

Phone-in – recording of contributors on the phone; usually used in studio-based context. Note: be
careful about legal restrictions about recording people in the phone, this is illegal in some countries,
even if the person has given permission.

Single interview – with an interviewee and interviewer.

Feature – with voice, background sound, narration and other elements mixed together. Dramatic – this
is a broad category and can include theatre, music and other entertaining formats.

Informative/documentary – a piece that primarily conveys information, in the same way as a public
service announcement or advert provides educational information. Endorsement – using a well-known
person to convey a message, such as a leader or a celebrity.

STEP 3 – CHOOSING A STYLE

Choose a style for your audio piece that suits your audience and your message.

 Formal or informal – do you want to use humor and familiarity as tools to reach your audience,
or do you want to convey information by invoking authoritative sources and “experts”? The
most obvious example of the formal style is a news item, in which the emphasis is put on the
authority of the information.
  With a narrator or without – do you want to let the voice of your contributor(s) be the whole
audio piece, as many ‘oral history’ productions do, or do you want to incorporate a “presenter”
voice to draw the pieces together for the audience?

STEP 4 – SETTING UP THE RECORDING

Whether you are doing an interview or capturing raw sound, you need to take time to test the sound
levels before you actually start recording. Background sound, such as the hum of an air conditioner,
might not have been noticeable before you started recording, but once you have your headphones on it
can suddenly sound very loud.

Some background sound can add to the atmosphere, but some can be purely distracting. If the noise is a
problem, ask it to be switched of or silenced, or if necessary move to another location. There is nothing
worse (and it happens a lot) than to come back with unusable recordings simply because the person
making the recording felt too awkward to do anything about it at the time.

If you are doing an interview, take the time to test your contributor’s voice for loudness and clarity, and
make any necessary changes – such as adjusting the sound levels, repositioning the microphone, or
changing the seating arrangement or general environment.

You can also use this test period or “sound-check” as a way to break the ice – people are often nervous
about being recorded and uncomfortable speaking into a microphone, but you can take steps to ensure
that they are as relaxed as possible. Welcome them, perhaps make a joke, and then tell them that you
will ask a few ‘trailer’ questions that won’t be recorded. ‘What did you have for breakfast?’ is a standard
first question to break the ice, and also to test voice levels.

For some sorts of interviews, you may want to prepare the interviewee(s) in advance by discussing what
sort of questions you are going to ask. Especially if the recording is being played live, or if you hope to
use the interview without much editing, this is time well spent.

STEP 5 – REVIEWING YOUR MATERIAL

One of the most important steps in producing audio is to listen back to your recording and make notes
or a full transcript of what was said and where the good sounds are located. If you do this in shorthand,
it is called a “log.” This step takes time, and a frequent mistake made by audio producers at all levels of
experience is hasty logging. This can result in a great deal of wasted time. Time spent reviewing and
logging your content is time well spent.

A log can take a number of forms depending on what works for you, but at minimum, be sure to record
the time of each new paragraph or new sound (make sure to start your playback at

00’00”), and then additionally the time when there is a good bit of speech or background sound. Note
the start time, the first few words, the last few words, and the end time for each section that you like.

For example:
INTRO (00’20”): “I believe the most important aspect is ……

OUT: …………………everyone should know this”. (00’50”)

If your recorder does not have a counter button, you can use a stopwatch to capture these times. You
might also want to write notes to yourself such as “overview” or “part 3 – significance” to help you
remember what part of your story each particular sound connects to. If you set up your log as a table,
you can make a column for such notes, and if you do a fuller transcription you can just insert them in the
text with a consistent flag. However you choose to do it, think of this step as identifying the building
blocks that you are later going to go back to when you edit or mix.

STEP 6 – EDITING & “PACKAGING” CONTENT

Once you have your building blocks identified, you can go back and start putting your piece together.
This may entail recording additional clips of narration to bridge certain themes. Even if your piece has
only three sections – for example: a three sentence intro, a two-minute interview, and a conclusion
pointing to where listeners can learn more – you still want to have identified these three pieces and
thought through how they are going to fit together. At this stage, it is important to refer back to the
priorities you identified in Step 1, in order to keep yourself on track.

STEP 7 – USAGE & DISTRIBUTION

If you create an interesting and engaging audio piece, you can make it available to radio stations as well
as to online distributors – for example advocacy websites. The Internet enables online audio to be used
and accessed around the world, usually at no extra cost to the distributor or user. This makes it a
powerful media and advocacy tool that is difficult to block or censor. An audio piece can have a long
shelf life, particularly if it is not dated by a reference to a time or event.

The ability to reuse an audio piece is a strength of this kind of resource. Audio work can be archived in
an online audio database, and it can be repeated on radio shows in new and different configurations. In
order to successfully distribute content to both online or on-air sources, advance research and
relationship-building work is necessary.

STEP 8 – EVALUATION

It can be a major challenge to evaluate the success or impact of an audio piece. You can obtain data
about who listened online from programs that tally website hits and downloads, and radio stations also
have tools to assess audience size. But evaluating the impact and effectiveness of the content of your
piece requires focus groups, questionnaires and other methods that are applied to groups of listeners, if
these can be identified and such data collection arranged. You could prearrange for a number of people
to listen to the audio and give you their feedback, or ask for feedback at the end of the piece, providing
a web contact.
How Does Sound Travel?

Before we discuss how sound travels, it’s important to understand what a medium is and how it
affects sound. We know that sound can travel through gases, liquids, and solids. But how do these
affect its movement? Sound moves most quickly through solids, because its molecules are densely
packed together. This enables sound waves to rapidly transfer vibrations from one molecule to
another. Sound moves similarly through water, but its velocity is over four times faster than it is in air.
The velocity of sound waves moving through air can be further reduced by high wind speeds that
dissipate the sound wave’s energy.

The speed of sound is dependent on the type of medium the sound waves travel through. In dry air at
20°C, the speed of sound is 343 m/s! In room temperature seawater, sound waves travel at about
1531 m/s! When physicists observe a disturbance that expands faster than the local speed of sound,
it’s called a shockwave. When supersonic aircraft fly overhead, a local shockwave can be observed!
Generally, sound waves travel faster in warmer conditions. As the ocean warms from global climate,
how do you think this will affect the speed of sound waves in the ocean?

When a sound wave is produced, it moves forward through the medium, creating compressions and
rarefactions. As the sound wave comes in contact with air particles, it vibrates them to create
alternating patterns of bunched and expanded areas. Imagine a slinky moving down a staircase. When
falling down a stair, the slinky’s motion begins by expanding. As the first ring expands forward, it pulls
the rings behind it forward, causing a compression wave. This push and pull chain reaction causes
each ring of the slinky’s coil to be displaced from its original position, gradually transporting the
original energy from the first coil to the last. The compressions and rarefactions of sound waves are
similar to the slinky’s pushing and pulling of its coils.

Sound waves lose energy as they travel through a medium, which explains why you cannot hear
people talking far away, but can hear them whispering nearby. As sound waves move through space,
they are reflected by mediums, such as walls, pillars, and rocks. This sound reflection is better known
as an echo. If you’ve ever been inside a cave or canyon, you’ve probably heard your echo carry much
farther than usual. This is due to the large rock walls reflecting your sound off one another.
Types of Waves

So what type of wave is sound? Sound waves fall into three categories: longitudinal waves, mechanical
waves, and pressure waves. Keep reading to find out what qualifies them as such.

Longitudinal Sound Waves

A longitudinal wave is a wave in which the motion of the medium’s particles is parallel to the direction of
the energy transport. If you push a slinky back and forth, the coils move in a parallel fashion (back and
forth). Similarly, when a tuning fork is struck, the direction of the sound wave is parallel to the motion of
the air particles.

Mechanical Sound Waves

A sound wave moves through air by displacing air particles in a chain reaction. As one particle is
displaced from its equilibrium position, it pushes or pulls on neighboring molecules, causing them to be
displaced from their equilibrium. As particles continue to displace one another with mechanical
vibrations, the disturbance is transported throughout the medium. These particle-to-particle,
mechanical vibrations of sound conductance qualify sound waves as mechanical waves. Sound energy,
or energy associated with the vibrations created by a vibrating source, requires a medium to travel,
which makes sound energy a mechanical wave.

Pressure Sound Waves

Because sound waves consist of compressions and rarefactions, their regions fluctuate between low and
high-pressure patterns. For this reason, sound waves are considered to be pressure waves. For example,
as the human ear receives sound waves from the surrounding environment, it detects rarefactions as
low-pressure periods and compressions as high-pressure periods.
Transverse Waves

Transverse waves move with oscillations that are perpendicular to the direction of the wave. Sound
waves are not transverse waves because their oscillations are parallel to the direction of the energy
transport. Among the most common examples of transverse waves are ocean waves. A more tangible
example can be demonstrated by wiggling one side of a string up and down, while the other end is
anchored. Still a little confused? Check out the visual comparison of transverse and longitudinal waves
below.

Properties of Sound

What makes music different from noise? A bird’s call is generally more melodic than a car alarm. And,
we can usually tell the difference between ambulance and police sirens - but how do we do this? We use
the four properties of sound: pitch, dynamics (loudness or softness), tone color, and duration.
In music, we alter the four properties of sound to make repeating patterns. Duration is the length of
time a musical sound lasts. When you strum a guitar, the duration of the sound is stopped when you
quiet the strings. Pitch is the relative highness or lowness that is heard in a sound, and is determined by
the frequency of sound vibrations. Faster vibrations produce a higher pitch than slower vibrations. The
thicker strings of the guitar produce slower vibrations, creating a deeper pitch, while the thinner strings
produce faster vibrations and a higher pitch. A sound with a definite pitch, or specific frequency, is called
a tone. Tones have specific frequencies that reach the ear at equal time intervals, such as 320 cycles per
second. When two tones have different pitches, they sound dissimilar, and the difference between their
pitches is called an interval. Musicians frequently use an interval called an octave, which allows two
tones of varying pitches to share a similar sound. Dynamics refers to a sound’s degree of loudness or
softness and is related to the amplitude of the vibration that produces the sound. The harder a guitar
string is plucked, the louder the sound will be. Tone color, or timbre, describes the overall feel of an
instrument’s produced sound. If we were to describe a trumpet’s tone color, we may refer to it as bright
or brilliant. When we consider a cello, we may say it has a rich tone color. Each instrument offers its own
tone color, and new tone colors can be created by layering instruments together. Furthermore, modern
music styles like EDM have introduced new tone styles, which were unavailable prior to digital music
creation.

Characteristics of Sound Waves

There are five main characteristics of sound waves: wavelength, amplitude, frequency, time period, and
velocity. The wavelength of a sound wave indicates the distance that wave travels before it repeats
itself. The wavelength itself is a longitudinal wave that shows the compressions and rarefactions of the
sound wave. The amplitude of a wave defines the maximum displacement of the particles disturbed by
the sound wave as it passes through a medium. A large amplitude indicates a large sound wave. The
frequency of a sound wave indicates the number of sound waves produced each second. Low-frequency
sounds produce sound waves less often than high-frequency sounds. The time period of a sound wave is
the amount of time required to create a complete wave cycle. Each vibration from the sound source
produces a wave’s worth of sound. Each complete wave cycle begins with a trough and ends at the start
of the next trough. Lastly, the velocity of a sound wave tells us how fast the wave is moving and is
expressed as meters per second.

Units of Sound

When we measure sound, there are four different measurement units available to us. The first unit is
called the decibel (dB). The decibel is a logarithmic ratio of the sound pressure compared to a reference
pressure. The next most frequently used unit is the hertz (Hz). The hertz is a measure of sound
frequency. Hertz and decibels are widely used to describe and measure sounds, but phon and sone are
also used. A sone is the perceived loudness of a sound and a phon is the unit of loudness for pure tones.
Additionally, the phon refers to subjective loudness, while the sone is the perceived loudness.
Sound Wave Graphs Explained

Sound waves can be described by graphing either displacement or density. Displacement-time graphs
represent how far the particles are from their original places, and indicates which direction they’ve
moved. Particles that show up on the zero line in a particle displacement graph didn’t move at all from
their normal position. These seemingly motionless particles experience more compressions and
rarefactions than other particles. Since pressure and density are related, a pressure versus time graph
will display the same information as a density versus time graph. These graphs indicate where the
particles are compressed and where they are very expanded. Unlike displacement graphs, particles
along the zero line in a density graph are never squished or pulled apart. Instead, they are the particles t

hat move back and forth the most.

Sound Pressure

Sound pressure describes the local pressure deviation from the ambient atmospheric pressure as a
sound wave travels. It’s important to recognize that sound pressure and air pressure are not the same
concept. Overall, the speed of sound is not influenced by air pressure. As sound waves pass from the
sound source through the air, they alter the pressure experienced by air particles.

COURSE: JOURNALISM

UNIT: SOUND PRODUCTION

LECTURER: EUNICE WAITIKI

LESSON 5: DIGITAL RECORDING AND PROCESSING

OBJECTIVES: At the e end of the lesson, the student should:

v Understand the digital recording and processing

v Understand the digital audio process

DIGITAL RECORDING AND PROCESSING

The digitizing and recording of sound and video is a sophisticated and powerful modern technology, but
its origins go back to mechanical calculating machines. These work on the simple principle that any
calculation is possible using just two digits, ‘zero’ and ‘one’. This technology has enabled the
transformation of cumbersome mechanical calculators into lightning-fast electronic calculators and
computers, all driven by ‘chips’.

The power and speed of the electronic silicon chip has grown exponentially since the 1970s, but it is only
comparatively recently that it has developed enough storage capacity to successfully serve the video
and audio post production industries.

The whole digital revolution in communications, both audio and video, is dependent on devices called
analogue-to-digital converters (often abbreviated as A/D converters or ADCs). These are devices
designed to take in an analogue signal one end and to deliver a digital version at the other. They work
on the principle of breaking down a waveform into two distinct parts, one part recording the
waveform’s position in time as it passes a point and the other recording the waveform’s amplitude –
how loud it is at that point. These time and amplitude readings are repeated and recorded very regularly
and very accurately thousands of times a second. The time element, the speed with which the sound is
sampled, is called sampling. The recording of amplitude is called quantization.

Digital signals take one of two possible values when recorded: on or off. These become equally
saturated points of recorded data – one positive and one negative. Noise within the system can be
completely ignored providing, of course, that it doesn’t interfere with the two values recorded. The
fundamental unit of digital information is the ‘bit’ (binary digit), whose state is traditionally represented
by mathematicians as ‘1’ and ‘0’, or by engineers as ‘Hi’ and ‘Lo’. The two states can be physically
represented in a wide variety of ways, from voltage peaks in wires to pits in plastic discs (e.g. DVDs).

Digital media can be transferred from electrical wires to magnetic tape, to plastic CD, and back again
with perfect fidelity, so long as the same sampling rate is used throughout. If the speed is only
fractionally out, the system will not work properly – there will be synchronization ‘timing errors’ and
possibly no signal reproduced at all.

The digital audio process

In audio, digital sampling above ‘40kHz’ with quantization above ‘16 bits’ gives high-fidelity recordings,
equal to and better than analogue recordings. Providing the analogue-to-digital conversion to the
recorder and then the returning digital-to-analogue conversion is of high quality, the reproduction will
be near perfect. But audio digital processing is a demanding technology. In analogue recording, the
frequency range of the audio equipment need only reach 20 000 cycles a second, for this is the highest
frequency we can hear. But in digital recording the analogue signal is immediately converted into a fast
stream of code, recorded as a stream of offs and on, instantly requiring a frequency range 30 times
greater than our analogue system. In fact, digital audio requires a bandwidth similar to that required for
video recordings – indeed, the first commercial digital audio recordings were made on modified video
recorders.

Before processing, the analogue audio is cleaned of signals that are beyond the range of the digital
recording system. This is because the rate at which the signal is to be sampled has to be twice as high as
the highest frequency to be recorded. This means the minimum sampling rate has to be at least 40 kHz,
otherwise there could be interference between the digital signal and the audio signal. Such processing is
known as anti-aliasing filtering.

The incoming signal is now ready to be sampled (in kHz) at regular intervals, and the magnitude
determined.

The samples are then ‘quantified’ to give each a quantity value from a set of spaced values (bit values).
This means the original signal is now represented by a string of numbers.

Sampling rate and pulse code modulation

The sampling rate is normally greater than twice the highest frequency to be recorded and it essentially
determines the frequency response of a system. For successful audio recording, the sampling rate must
exceed 40 kHz. Audio CDs, for example, have a sampling rate of 44.1 kHz. The sampling must take place
at precise regular intervals, controlled by an accurate quartz crystal clock, similar to the one found in
watches. This is the fundamental principle of pulse code modulation (PCM), where the audio waveform
is sampled in regular steps. The distance between the steps is dependent on the speed of the sampling.
As the sampling rate increases, so does the number of samples. The more samples there are, the higher
the quality. However, this also means that the amount of data recorded increases, which will require an
increase in storage capacity. Inconsistencies in the sampling, such as timing errors, will induce sound
glitches and drop-outs in the reproduced sound. This becomes particularly critical if two or more sounds
are being combined together.

When choosing a working sampling rate for a project, it is necessary to take into account the local studio
standard and the end application. Music recording studios use the 44.1 kHz CD standard.

However, in broadcasting, 48 kHz is the standard sampling rate. In actuality, most people will not notice
any quality difference if a sampling rate is above 48 kHz or not, although 96 kHz is now supported on
some professional audio systems.

Quantizing level

Quantization helps determine the dynamic range of a recording. Digital audio is recorded in what is
known as linear pulse code modulation (LPCM). In this system, there is a 6 dB reduction in noise as each
bit is added to the word length. In practice, a 16-bit recording is capable of a dynamic range of around
90 dB. It is restricted by ‘noise’ or dither, which is added to the recordings to remove the complete
silence that ‘falls away’ when no signal whatsoever is being recorded (the effect is similar to using a
noise gate, as described in Chapter 16). The minimum quantizing rate for professional results is 16 bits.
In practice, at 20 bits, noise level is below audibility. Although many digital audio workstations and
mixing desks support 24-bit recording, they employ longer bit words internally than they offer to the
external world. This allows a system to mix many channels of sound together without an apparent
reduction in dynamic range. Although the summing of digital channels does increase total noise, it will
not be apparent in the audio at the output of the device. In any project the word length should be at
least the same or better than that of the final transmitted or delivered material. This ensures

COURSE: JOURNALISM

UNIT: SOUND PRODUCTION

LECTURER: EUNICE WAITIKI

LESSON 6: MICROPHONE POSITION

At the end of the lesson, the student should:

 Understand the different microphone position

Microphone position

On-camera microphones

Built-in camera mics on camcorders are very often omnidirectional, which means that sound is recorded
in an unfocused way, together with any extraneous noise. Professional cameras can be fitted on mic. In
either case, the mic is, of course, tied to the position of the camera and all audio will be recorded from
the same perspective as the picture. This may mean that audio quality will vary from shot to shot, posing
a continuity problem when the material is edited together, and in wider shots it may be difficult to
isolate foreground speech from the background noise. One solution is to record close dialogue with a
personal mic and mix this with the output of the camera mic, which will record the natural acoustic of a
location more successfully.

The advantage of using an on-camera mic is that, in some situations, such as in covert or observational
filming, it seems less intrusive than fitting the subject with a personal mic or following them round with
a boom. On productions where crew levels are an issue, this option means that the camera operator can
record both sound and picture. In fast turnaround situations such as news gathering, the audio can be
routed directly into the camera inputs. However, the disadvantage of this is that the camera's Automatic
Gain Control must be used to control recording levels, which may result in an audible 'pumping' effect. A
better alternative is to route the audio to the camera inputs through a small dual input mixer, which can
be attached to a belt for ease of use. Once the shoot is complete, the location tapes can be sent directly
to the edit suite.
Off-camera microphones

For most professional TV and film productions, microphones will be sited off camera in the optimum
position for sound quality, regardless of the camera's position. In order to 'cover' a scene or location, a
number of mic types may be used in a single set-up. However, the position of each microphone in
relation to the subject is also a critical factor in providing the editing department with usable sound. The
main techniques used in film and TV can be summarized as follows.

Mic with fish pole

This is one of the most common techniques used in location work involving a sound crew, mainly
because it achieves good-quality sound and allows the subject freedom of movement without being
hampered by mic cables or fixed mic placement. The mic is mounted on a long boom or fish pole, which
enables the mic to be held just out of shot. This technique requires a dedicated crew member known as
a boom swinger to follow the subject around whilst keeping them on mic. To record speech, the mic is
best angled down towards the subject from overhead, a technique which results in the most natural
sounding dialogue. Some ambient fx will also be picked up, resulting in a recording that reflects the
acoustic properties of the location in proportion to the dialogue. Audio 'perspective' can be suggested
by lifting the mic higher for a wide shot, resulting in a thinner sound. A more intimate sound can be
achieved by holding the mic closer to the speaker for a close-up. In some shots it may be necessary to
mic from below, although in doing so, it may be harder to keep the mic out of shot. This technique tends
to make for a boomier recording which reflects the mic's proximity to the speaker's chest. It may also
mean that fx such as footsteps are more prominent in the recording than might be desirable. To reduce
wind and handling noise, the mic should be suspended in a shock mount and housed in a windshield.

Fixed booms

In situations where the subject is not likely to move around much, a fixed mic stand can be used. This
technique is limited to productions such as music and entertainment shows, simply because in order to
be close enough to the subject, both the stand and mic will be in shot.

Fisher booms

The Fisher boom is a large platform boom designed for use within the studio. It can be suspended over a
set, out of shot, and needs two operators: one to physically move the platform, and one to control the
angle and placement of the mic over the set during takes. The advantage of these booms is that they
can reach right into the back of a set without getting into shot, but are only suitable for large studios
that have the floor space to accommodate the platform.

Spot/plant mics

Boom and personal mics may be hidden within the set/location itself. For example, in a restaurant scene
the mic may be hidden within a table lamp or a vase of flowers. In choosing where to plant a mic, it is
important to work out exactly where the performers will deliver their lines, and ensure the mic is
directed at them. It is also important not to hide the mic in a prop that a performer is likely to pick up or
move!

Slung mics

Mics can be suspended over the subject, a technique that is particularly useful for recording static
groups of people such as studio audiences, orchestras and choirs. Its disadvantage is that the mic(s)
cannot be adjusted during a take.

Hand-held mics

Hand-held mics are limited to news and other productions where it is acceptable (and sometimes even
part of the convention) to have a mic in shot. An omnidirectional mic is often used and this can be
moved around by a presenter without the need to remain on axis in order to obtain a good record- ing.
How good the recording actually is will depend on the presenter's mic skills rather than the sound crew.

Personal mic worn on body

In some productions, personal mics are worn visibly in shot, on the lapel or pinned to the chest. Where
line mics are used, the cable is hidden within clothing, with the mic connector led down to the ankle.

Most problems start when the mic has to be hidden on the performer's person, perhaps in the costume
or even in their hair. The costume can be made to conceal the mic without actually abrading the mic in
any way. Because they are so susceptible to wind noise, these mics are often buried deep within the
costume, at the expense of HF response. Dialogue recorded in this way can sound dull and muffled, and
will often need to be brightened in post-production. Clothing rustle and wind noise are the main reasons
why dialogue recorded with personal mics is often ADR'd, and the recordist may resort to tricks such as
using loops of blue tack over the mic capsule to avoid contact with the costume.

Radio mics worn on body

Again, satisfactory concealment of the mic capsule without interference from clothing is the main issue.
However, the battery pack and the transmitter antenna must also be concealed in a way that is
considerate both to performance and movement. Unfortunately, once a take has started, the sound
recordist has little control over the quality of audio recorded.

Using multiple microphones

Apart from the simplest set-ups, such as a news piece to camera, most film and TV productions will use
of a number of mics to cover a single set-up. This is done for two reasons:

1. A scene or location may involve a number of actors, all of whom need to be on mic. In film or TV
drama rehearsals, a set of marks (marked out by tape on the floor) should be established for a scene so
that the actor(s) can replicate the same movements around the set on every take. These marks can be
used to plan out the best mic positions, so that the actor effectively walks from mic to mic. In this
scenario it is important that directional mics are used, and spaced correctly to avoid a sound being
picked up by more than one mic. Where this does happen, the sound reaches each mic at a slightly
different time. When the two mic outputs are mixed together (see below), they will be out of phase and
phase cancellation occurs, causing the recording to sound oddly reverberant, and ren- dering the sound
unusable in its mixed form. To minimize the chances of this happening, mics should be placed as close as
possible to the subject. A useful rule of thumb is to measure the dis-tance between the subject and the
first mic. The next nearest mic should be placed at least three times this distance away from the first.
Where a number of radio mics are used, each should be set to operate on a different frequency.

2. The sound recordist may choose to record a scene using more than one different type of mic. This
gives the sound editors some choice as to which they will use in the mix. For example, radio mics may be
used on the establishing wide shot of a scene, where it is difficult to conceal a boom. On the subsequent
medium and close-up angles, the recordist may switch to a boom mic/fish pole to obtain a better quality
recording. This, however, will pose a problem in post-production as the quality of the two mics will differ
hugely and will not easily blend together in the mix. The wide shot may be marked down for ADR
anyway, using a boom mic to match the rest of the scene, but the sound recordist may give the editors
the whole scene recorded on both radio mics and boom for continuity. It is important that the recordist
uses matching mics as far as possible, so that the various mic sources can be blended together in the
final mix to sound as if they come from a single source.

COURSE: JOUNALISM

UNIT: SOUND PRODUCTION

LESSON 8: SOUND ELEMENTS IN BROADCAST PRODUCTION

LECTURER: EUNICE WAITIKI

OBJECTIVES: At the end of the lesson the student should:

 Understand the different elements in broadcast production

Sound Elements in Broadcast Production

Words alone cannot tell the story a scriptwriter or the director of a radio/TV programme wants to pass
across to the audience. He needs other elements like music, sound effects, voice quality among others.
Here are ways by which music, sound effects and voice quality can add meaning to an overall
production.

Music
Music can be incorporated into radio and television programme for a variety of reasons. These include:

 Music can be use an audience along from one feeling to another for instance music can be use
to create an illusion of love, hatred, horror or happiness.

 Music should be used to establish mood and atmosphere but it is important not to over-
emphasize emotional moments.

 It can be employed as a programme identification tune which is otherwise called signature tune.

 In some cases, mu music can be used to bridge two segments of a programme

 In some children and adult programmes music can be used to tell a story.

Music can be used in any radio or television programme; however when using music it must be a song
that the audience is familiar with, in other words it must be interesting and appealing to the audience.

It is important that you do not speak over vocals. The announcer’s voice may clash with the lyrics of the
song playing under if it not carefully executed.

Music is such an evocative tool that it is used in a great many radio production task; unfortunately, it is
also frequently misuses and overused. By indicating that music can be misused or overused, I implying
that the radio or television scriptwriter and director be wary of how and when to use music in any
production. Hausman et al therefore identify what they called the “do use music” and the “do not use
music”.

Do use music

v When you can find a logical reason to do so, use music to create and reinforce a theme.

v When the music has a logical purpose and fits into the format of your station.

Do not use music

v Do not use music strictly as reflect: don’t make it “other stations are using it, I have to use it”,
in “many times you’ll be better off without it”.

v Do not use music indiscriminately. According to them this warning applies specifically to the novice
radio producer who is tempted to use currently popular music within announcements or other
productions, whether or not it serves to reinforce the message.

Sound Effects

Sound effects can come from audio files, pre-recorded tapes or CDs; and they are used to either indicate
location or action. When sound effect is applied to production appropriately it can add meaning to the
message on the one hand but if it is wrongly applied it may make the message appear amateurish.
According to Hausman et al (2010), there are two good reasons for using sound effects which they called
“do use of sound effects”.

Do use of sound effects

v To save time and words

v To inject drama into the programme

Do not use sound effect

v Do not use sound effect just because they are there: According to them, sound effects are excellent
production tool, but if they’re used just for the sake of using them, they are inappropriate and can
detract from the message.

Voice quality

-Most scriptwriters often want to have a say in who delivers the script they have written. Their reason,
though may sound ridiculous, is based on the quality and the inflections the reader can bring into the
script to drive the message home. A well-written script with a powerful message may lose its power if
the presenter failed to inject the right voice quality. Voice quality, according to Hausman et al (2010), is
the overall image that an announcer’s voice projects. In their opinion, the voice of a reader should not
have defects of any sort. They write further on possible factors that may cause distractions in the voice
of a reader:

One of the most common distractions is improper breathing by the announcer. Over-breathy voices,
except when they are a well-known novelty, sound amateurish. Often in experienced announcers can
be heard gasping for air between phrases. Such gasping sounds might not be apparent in everyday
speech, but a mic can be merciless

The way out of the above problem, is to maintain generous breath support instead of trying to talk until
all your breath is expanded. By planning where to take breaths you can hit the right inflections

QUESTION: What are sound effects and how do they influence a successful production

JOURNALISM

SOUND PRODUCTION

LESSON 9: SOUND PRODUCTION IN GENERAL

LECTURER: EUNICE WAITIKI

OBJECTIVES: At the end of the lesson the student should:

 Understand more about sound production

  Understand what happens when there is interference in sound production

 Understand what podcasting is.

SOUND PRODUCTION

 Sound is produced when something vibrates. The vibrating body causes the medium (water, air,
etc.) around it to vibrate. Vibrations in air are called traveling longitudinal waves, which we can
hear.

 Sound waves consist of areas of high and low pressure called compressions and rarefactions,
respectively. Shown in the diagram below is a traveling wave.

 The shaded bar above it represents the varying pressure of the wave. Lighter areas are low
pressure (rarefactions) and darker areas are high pressure (compressions). One wavelength of
the wave is highlighted in red. This pattern repeats indefinitely.

 The wavelength of voice is about one meter long. The wavelength and the speed of the wave
determine the pitch, or frequency of the sound. Wavelength, frequency, and speed are related
by the equation speed = frequency * wavelength.

 Since sound travels at 343 meters per second at standard temperature and pressure (STP),
speed is a constant. Thus, frequency is determined by speed / wavelength. The longer the
wavelength, the lower the pitch. The 'height' of the wave is its amplitude. The amplitude
determines how loud a sound will be. Greater amplitude means the sound will be louder.

Interference

 When two waves meet, there can be two kinds of interference patterns; constructive and
destructive. Constructive interference is when two waveforms are added together. The peaks
add with the peaks, and the troughs add with the troughs, creating a louder sound.

 Destructive interference occurs when two waves are out of phase (the peaks on one line up
with troughs on the other). In this, the peaks cancel out the troughs, creating a diminished
waveform. For example, if two waveforms that are exactly the same are added, the amplitude
doubles, but when two opposite waveforms are added, they cancel out, leaving silence.

 An “audio production” is more commonly found in the Music Production and Film & TV (Post
Production) categories, where the end result of our recording is a tangible, deliverable product.

 When talking about music production, we can think of late nights in the studio, long hours trying
to get that perfect sound. That, however, is the shorter part of the overall production. The next
 step, where we massage the mix with the help of compressors, EQ, and automation can be a
very involved process.

 Having the knowledge on what processors to use, or techniques to try is key to having a high
quality audio production. Signal flow is the most important aspect of audio recording.

 This not only includes how we get sound from the mic to our recording console, and then back
to the speakers. It also includes how we take our recording, process it through our plug-ins and
digital audio workstations.

 Post Production has a similar workflow. We begin the process with the script and initial filming,
with production audio. Production audio is recorded on the set, live while filming. However it’s
rarely used in the final recorded product. It is used as a reference guide, but most of the audio in
television and movie productions is recorded separately, after the filming happens.

What is Podcasting?

A Podcast has come to mean any audio programs made available for downloading from a website. If I
make an audio documentary and put it on my web page for download many would call this a Podcast.
However this is not really what Podcasting is. A Podcast is a method of subscribing to audio content.

This is quite unlike the traditional way of downloading content. Usually this process involves browsing to
a site, find what you want, then click on the link to download it.

This is ok if the website has just a couple of audio files to distribute. However if the website makes new
audio available regularly then you have to remember to check the site periodically, find the new
content, and click on the link to download it. This is inefficient and very time consuming.

Podcasting is intended to make exactly this situation more manageable. A Podcast enables you to
subscribe to content so that you are notified of new material when it is made available. The software
that manages this kind of subscription will then conveniently download the content for you without you
needing to continuously revisit the webpage.

JOURNALISM

SOUND PRODUCTION

LESSON 10: SOUND PRODUCTION IN GENERAL

LECTURER: EUNICE WAITIKI

OBJECTIVES: At the end of the lesson the student should:

 Understand more about editing of picture and sound

 Understand the editing process.

Editing picture and sound

The main limitation of tape editing is its lack of flexibility. Unless the length of the new material is
exactly the same as that which is to be removed, then changes can only be made by dubbing the entire
assembly (with the new edit) to another tape. Where analogue tape is used, such as Beta SP, each tape
generation used will mean a gradual loss in picture and sound quality, as well as the usual analogue
problems of tape hiss and drop-outs.

Non-linear editing

The term non-linear editing is used to describe an editing technique that is almost entirely digital. All
source media is first logged and then digitized roll by roll, into the NLE (non-linear editor), where each
shot is held on a series of hard drives as a clip. When all the source media has been digitized and sorted
into bins, the editor can select a clip and load it into the source monitor. The clip is played, and in and
out points are marked. The clip is then edited into position on the timeline, and the process repeated
with the next clip. The timeline is a graphical representation of the edit assembly, showing the video
track positioned above the audio tracks.

On playback, each edited video or audio clip merely refers the system back to the master clip held on
the hard drive and the process is entirely non-destructive. Material can be accessed instantly by clicking
on to the selected point within a source clip or the assembly, eliminating the need to shuttle through
tapes in real time. All tracks can be independently edited, material can be overlaid, inserted or removed,
and different versions of the cut saved quickly and easily. Edits can be further refined using the trim
function, and video/audio effects added.

Some effects which cannot be played in real time will need to be rendered, producing a new clip. This
becomes an issue for audio when plug-ins are used (see below). Once the project is complete, the edit
can be outputted to tape or transferred to another system via OMF or an EDL.

System configuration

Most systems consist of one or two computer monitors, which display the software menus, source bins,
timeline, and the source and assembly windows in which the rushes and the edit can be viewed.

The core of the system is a PC or Mac computer on which the operating software runs, such as Avid or
Final Cut Pro. The internal hard drive is used to store all project data and saved edits. The actual media
(picture and audio files) is stored on an array of external drives. Alternatively, multi-user sys- tems such
as Avid 'News cutter' store media on a shared storage device, which can be accessed by many
workstations via a high-speed network. A system such as Avid 'Unity' can store up to 5 terabytes of
media on an array of drives set up in a RAID configuration, and media is sent and retrieved via a net-
work (LAN or SAN) using both standard Ethernet and Fibre Channel interconnections (see Chapter 1,
'File sharing and networking' section).

NLEs are supplied with a standard pair of speakers and a small mixing desk on which to monitor the
audio output channels. Editing is usually carried out using a keyboard and mouse. In addition to the NLE,
most cutting rooms will be equipped with a VTR (usually Beta SP), which can be used for making digital
cuts, and other equipment such as a Jazz drive, VHS, DAT machine and CD player.

The editing process

Project settings for audio

When starting a new project, there are a number of important audio settings to be selected prior to
digitizing.

File format

There will be an option within the NLE to set a default file format for the project. This will usually be a
choice between AIFF and BWAV. If the project is to be exported to a sound facility, it is best to consult
them as to which format to use. Failing this, BWAV is probably a better choice, as it is now supported by
PC- and Mac-based editing systems, has time stamp capability, and is the designated native file format
for AAF.

Sample rate

This setting controls the rate at which audio is played back in the timeline, and for all professional
applications a sample rate of 44.1 or 48 kHz is used. Sample rates higher than 48 kHz are not currently
supported by picture editing systems. DV camcorders can record audio at 32 kHz and some NLEs offer
this additional setting. Audio recorded through the analogue inputs can be digitized at either 44.1 or 48
kHz, although the latter is slightly higher in quality. If the project is to be exported to a sound facility, it is
important to check what their working sample rate is - most now work at 48 kHz as this com- plies with
broadcast specifications. Where audio is digitized through the digital inputs, the NLE setting should
match the sampling frequency of the original recording to avoid any loss of quality. Audio can be sample
rate converted, so that a CD playing at 44.1 kHz, for example, can be digitized in at 48 kHz. Newer NLEs
can convert non-standard audio in real time during playback.

Audio levels

The input level setting on the NLE is only critical when the audio is being digitized through the analogue
inputs. In this case, the sound recordist's tone should be used to line up the meters displayed in the
audio settings of the NLE. (Line-up tone is usually set at 18 dBfs in Europe and 20 dBfs in North America.)
In the absence of tone, it will be necessary to shuttle through the recording to find the loudest point and
check that it is peaking at just below 0 dBfs.
It is not necessary to set a level for audio that is digitized through the digital inputs, as the original
recording level will be reproduced unaltered. This also applies to audio that is imported or ingested as a
file, such as BWAV rushes from a file-based field recorder.