Incorporate Audio Multimedia Presentations

Incorporate Audio into Multimedia Presentations
ICPMM44CAICPMM44CAICPMM44CAICPMM44CAICPMM44CA
Part 1 - Audio TheoryPart 1 - Audio TheoryPart 1 - Audio TheoryPart 1 - Audio TheoryPart 1 - Audio Theory
Education and Training

Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty Ltd
ICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORY
Incorporate Audio into Multimedia Presentations
Welcome to the learning resource for Incorporate Audio into Multimedia
Presentations from QANTM Australia CMC Pty Ltd. This learning resource
covers or exceeds the competency in the following qualification/s:
CUF30601 Certificate III in Multimedia
CUF40801 Certificate IV in Multimedia
ICA40499 Certificate IV in Information Technology (Multimedia)
CUF50401 Diploma of Screen (Animation)
Assessment
Those students enrolled in this unit need to complete the assessment items
as per the Assessment Criteria Sheet.
Activities (non-assessable)
There are no activities for this unit.

Unit Description
This unit describes the competency required to edit, combine and incorporate
audio into multimedia presentations.
Suggested Hardware
To successfully complete this course students will need access to a personal
computer with soundcard, keyboard, mouse, microphone, headphones and/or
speakers and an internet connection.
Suggested Software
• Sound Forge 6
• Internet Explorer Version 4 or higher or other suitable browser
• Microsoft Word or other suitable word processor

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Multimedia PresentationsMultimedia PresentationsMultimedia PresentationsMultimedia PresentationsMultimedia Presentations
KEY FEATURESKEY FEATURESKEY FEATURESKEY FEATURESKEY FEATURES
This unit is designed to provide you with the knowledge in order to
correctly identify and understand the terms and features associated with
digital audio. You will also learn how to create, edit and manipulate digital
audio which is an important aspect in multimedia presentations.
• Introduction to the structure and the features of analog and
digital audio
• Audio file formats and compression
• Audio hardware
• Types of microphones and recording tips
• How to use Audio Encoder and Decoder Converters
OVERVIEW OFOVERVIEW OFOVERVIEW OFOVERVIEW OFOVERVIEW OF
THE UNITTHE UNITTHE UNITTHE UNITTHE UNIT
I C P M MI C P M MI C P M MI C P M MI C P M M 44C44C44C44C44CAAAAA
U N I TU N I TU N I TU N I TU N I T
Identify and describe
formats of digital
a u d i o
Use digital audio
s o f t w a r e
Edit digital audio
Construct a digital
audio track

Menu
Part 1 - Audio Theory
Introduction to Sound .............................................................................. 8
Analog Audio .......................................................................................... 8
Digitising Audio...................................................................................... 10
Sampling and Sample Rate ..................................................................... 10
Quantisation and Bit Depth ..................................................................... 11
Bit-Rates ............................................................................................. 12
The Nyquist Theorem ............................................................................ 13
Digital Audio Formats............................................................................. 16
Common Audio File Formats .................................................................. 17
Streaming and Non-Streaming Audio ....................................................... 20
Shockwave Audio ................................................................................. 21
Audio Compression ............................................................................... 22
Hardware Considerations ....................................................................... 23
How Microphones Work ......................................................................... 27
Dynamic and Condenser Microphones ...................................................... 27
Categories of Microphones ..................................................................... 29
Tips When Using Microphones ................................................................. 31
Part 2 - Sound Forge
Introduction to Sound Forge 6 .................................................................. 5
Starting Sound Forge from a Shortcut ....................................................... 5

The Design Window of Sound Forge 6.....................................................6
The Work Area........................................................................................ 6
The Toolbars........................................................................................... 6
ACID Loop Creation Toolbar...................................................................... 7
Status or Selection Toolbar ....................................................................... 8
Status Bar.............................................................................................. 9
Types of Microphones ............................................................................ 10
Tips on How to Use a Microphone for Voice Recording ............................... 11
Use Digital Audio Software .................................................................. 12
How to Record Your Own Voice............................................................... 12
DC Offset ............................................................................................. 15
Applying DC Offset ................................................................................ 15
Editing Audio ....................................................................................... 16
How to Copy/Paste a Waveform ............................................................. 16
How to Cut a Waveform ........................................................................ 17
How to Delete a Waveform .................................................................... 17
How to Convert Mono to Stereo ............................................................. 18
How to Convert Stereo to Mono ............................................................. 18
Converting Mono to Stereo .................................................................... 19
Converting Stereo to Mono .................................................................... 20
How to Copy and Paste Waveform into Only One Channel .......................... 21
How to Copy Waveform From One File to Another .................................... 22
How to Copy and Paste Special............................................................... 24
Paste Special in a Mono Wave File ........................................................... 24
How to Insert Silence ............................................................................ 25

How to Reverse Sound .......................................................................... 26
How to Use Normalise ........................................................................... 26
How to Fade Sound ............................................................................... 28
How to Use Graphic EQ ......................................................................... 30
How to Resample Files........................................................................... 31
How to Convert to 8-Bit ........................................................................ 32
How to Apply Effects to Waveform .......................................................... 33
Pitch Bend Effect ................................................................................... 34
How to Use Encoder and Decoder Audio Converters .................................. 35
Convert Track Files to WAV Files Using Soundforge .................................... 35
Adding Effects ....................................................................................... 37
Convert WAV Files to MP3 Files ............................................................... 37
Convert MP3 Files to WAV Files ............................................................... 38
Video and Audio in Sound Forge .............................................................. 40
How to Attach Audio to a Video Track ...................................................... 41

Introduction to Sound
Before we can begin to understand and analyse the specific features of digital
audio, we need to consider some fundamental questions; what is sound and
how does the human ear perceive variations in pitch, volume and scale?
In order for sound to exist, there needs to be a medium, like air or water or a
solid object, that acts as the transmitter, a motion or disturbance in this
medium that creates waves and some kind of receiver that detects these
variations (see Figure 1). Without these three essential parts, sound cannot
exist. Therefore, for humans to hear sound, pressure waves or changes in air
pressure must be generated by some physical vibrating object, such as
musical instruments or vocal chords. Then these changes are perceived via a
diaphragm, the eardrum, which in turn converts these pressure waves into
electrical signals, which are interpreted by the brain. This type of sound
production is referred to as analog audio.
Analog Audio
An Analog Audio signal can be graphically represented as a waveform (see
Figure 2). A waveform is made up of peaks and troughs that are a visual
representation of wavelength, period, amplitude (volume level) and
frequency (pitch).
The horizontal distance between two successive points on the wave is
referred to as the wavelength and is the length of one cycle of a wave. A
cycle is the distance between two peaks. The period of the wave refers to the
amount of time that it takes for a wave to travel one wavelength (see Figure
6 on the next page).
Amplitude is half the distance from the highest to the lowest point in a wave.
If the distance is large the volume level is comparatively loud and alternately, if
the wave is small, then the volume level is low.
The number of cycles per second is referred to as frequency (see Figure 3).
http://www.atpm.com/6.02/digitalaudio.shtml
Figure 1
Figure 2: Graphic representation of the waveform
Figure 3:
The greater the
amplitude the
higher the volume.
1 second
Trough
cycle
Peak
Amplitude
1 second
!
!
Frequency
!
!Frequency
!
!

Frequency is measured in hertz (Hz), a unit of measure named after Heinrich
Hertz, a German physicist and indicates the number of cycles per second that
pass a specified location in the waveform.
Frequency directly relates to the sound’s pitch. The pitch or key is how the
brain interprets the frequency of the sound created and the higher the
frequency, or the faster the sound vibrations occur, the higher the pitch. If the
vibrations are slower, then the frequency is low.
Analog signals are continuous and flexible; able to change at varying rates
and size which means that analog audio is relatively unconstrained and unlim-
ited. The flexible nature of analog sound, though seemingly positive, is in fact
its biggest disadvantage, as it is therefore more susceptible to the effects of
extreme changes in audio that cause degradation of sound like distortion and
noise.
Originally sourced from:
http://www.library.thinkquest.org/19537/Physics3.html
Figure 5
Figure 4
Figure 6
Figures 4 , 5 and 6 provide a visual guide to the structure of a waveform.

Digitising Audio
Computers cannot understand analog information. In order for an analog
signal to be understood by a digital device (such as a computer CD or DVD
player), it first needs to be digitized or converted into a digital signal by a
device called an Analog to Digital Converter or ADC (see Figure 7). Then for
the human ear to be able to hear a digital signal, it needs to be converted back
to an analog signal. This is achieved using a Digital to Analog Converter or
DAC.
The conversion of an analog signal to a digital one requires two separate
processes: Sampling and Quantisation.
Sampling and Sample Rate
The analog signal is sampled, or 'measured' and assigned a numerical value,
which the computer can understand and store.
The number of times per second that the computer samples the analog signal
is called its Sample Rate or Sampling Frequency. While the basic unit used to
measure frequency or cycles per second is hertz, when sampling audio it is
generally measured in thousands of cycles per second or kilohertz (kHz).
An audio CD, for example, generally has a sampling rate of 44.1 kHz that is
forty four thousand one hundred Hertz (or samples per second), while the AM
radio has a sample rate of 11.025 kHz or eleven thousand and twenty five
Hertz. The more samples taken, the higher the quality of the digital audio signal
produced.
Play the two provided examples, 44100Hz.wav
and 11025Hz.wav to hear the difference in sound
quality between the two different sample rates.
Low sampling rates, below 40 kHz, can result in a static distortion caused by
data loss. This is referred to as Aliasing (see Figure 8). Aliasing can cause
digitally reconstructed sound to playback poorly. To avoid the aliasing effect
Figure 7: Converting to Digital
Analogue
sound
Analogue to
Digital
converter
Digital
reproduction
!
Samples
Figure 8: Aliasing
This sound wave is 100th of a second long.
It has been sampled at the very low rate of 160hz.
The red dots are the samples taken.
The green line indicates the the sound wave the
samples would generate.
As you can see the aliasing is severe in this case.
The new sound file would not sound like the original.
44100Hz.wav 11025Hz.wav
Sound wave
Continious Discontinious

sampling needs to occur at a high enough rate to ensure that the sound's
fidelity is maintained, or anti-aliasing needs to be applied when the audio is
being sampled. An anti-alias filter can ensure that nothing above half the
desired sampling rate can enter the digital stream ie any frequencies above the
desired frequency are blocked. Be aware that using anti-alias filters may, in
turn, introduce further unwanted noise.
Analog audio is a continuous sound wave that develops and changes over
time. After it is converted to digital audio it is discontinuous, as it is now
made up of thousands of samples per second.
Quantisation and Bit Depth
Once an analog signal has been sampled, it is then assigned a numeric value in
a process called Quantisation. The number of bits used per sample defines
the available number of values.
Bit is short for binary digit. Computers are based on a binary numbering
system that uses two numbers; 0 and 1. This differs from the more familiar
decimal numbering system that uses 10 numbers.
This two number system means each additional bit doubles the number of
values available - a 1-bit sample has 2 possible values; 0 and 1 and a 2-bit
sample has 4 possible values; 0 and 0, 1 and 0, 0 and 1, 1 and 1 and so on
(See Figure 9).
These binary values are defined as its Resolution or Bit Depth. This method
of measurement is used throughout digital technologies. You may already be
familiar with bit depth in digital graphics, where a 1-bit image is black and
white, a web safe or greyscale image is 8-bit and an RGB image has one byte
or 8-bits allocated for each of the three colours and is 24-bits in total.
Typically, audio recordings have a bit depth of either 8 or 16-bit and even 24-
bit on some systems. An 8-bit sample will allow 256 values, whereas a 16-bit
sample will allow 65 536 values. The greater the bit depth, the more accurate
the sound reproduction and the better the sound quality.
Bit Depth Possible values
1
2
3
4
5
6
7
8
2
4
8
16
32
64
128
256
Adding one bit doubles the
number of values available.
Figure 9: Bit values
Bit: “A fundamental unit of infor-
mation having just
two possible values,
as either of the binary digits
0 or 1.”
Source: The American Heritage®
Dictionary of the English Language,
Fourth Edition (2000), by Houghton
Mifflin Company

An audio Dynamic Range is the difference between the
lowest and highest points of a wave and is measured in
decibels (dB). The larger the dynamic range the greater
the risk of distorted sound. Audio files with a large
dymamic range tend to require greater bit depth to
maintain sound quality
An 8-bit sample, with 256 values can recreate a dynamic
range of 48 dB (decibels), which is equivalent to AM radio,
whereas a 16-bit sample can recreate a dynamic range of
96 dB, which is the equivalent of CD audio quality.
The dynamic range of the average human ear is
approximately 0 to 96 dB (120dB is the pain threshold),
so it is no coincidence that the standard bit depth for CD
quality audio is 16-bit.
Bit-Rates
The number of bits used per second to represent an
audio recording is defined as Bit-Rate. In digital audio
bit-rates are defined in thousands of bits per second
(kbps).
The bit-rate is directly associated with a digital audio file’s
size and sound quality. Lower bit-rates produce smaller file
sizes but inferior sound quality. Higher bit-rates produce
larger files but are of a better sound quality. An
uncompressed audio track's bit-rate and approximate file
size can be calculated using the following formulas:
When calculating bit-rate it is important to remember that:
• 8 bits = 1 byte
• 1024 bytes = 1 Kb or a Kilobyte
• 1024 kilobytes = 1 Mb or a Megabyte
(see Figure 9 to calculate file size.)
Calculating an uncompressed CD track's bit rate
Calculating its file size
Sampling Rate x Bit Depth x Number of channels = Bit Rate
(in KHz)
44.1 KHz x 16 bits x 2 = 1 411.2kbps
Sampling
Rate
(in KHz)
= file sizex Bit Depth x Number of
channels
Length in
seconds
Bits to
Bytes
x /
44.1 KHz
10,584,000
bytes or
over 10 MB
8 =/
60
seconds2 xx 16 bits x
Figure 9
The amount of 1024 is often
rounded down to 1000 if
strict accuracy is not required.

The Nyquist Theorem
Audible Frequency refers to the range of frequencies that are detectable by
the average human ear. There is a direct correlation between the sample rate
and the highest audible frequency perceived by the ear. The relationship
between sample rate and the highest audible frequency is referred to as the
Nyquist Theorem. The Nyquist Theorem, named after Harry Nyquist, a Bell
engineer who worked on the speed of telegraphs in the 1920s, is a principle
that is used to determine the correct sampling rate for a sound.
Essentially, the Nyquist Theorem states that a sound needs to be sampled at a
rate that is at least twice its highest frequency in order to maintain its fidelity
or sound quality. Therefore, a sample taken at 44.1kHz will contain twice the
information of a sample taken at 22,050 kHz. Put simply, this means that the
highest audible frequency in a digital sample will be exactly half the sampling
frequency.
http://www.csunix1.lvc.edu/~snyder/2ch11.html
Figure 10: The Nyquist Theorem rules states that a
waveform must be sampled twice. The positive peak and
the negative peak must both be captured in order to get a
true picture of the waveform.
Average human hearing, at best, covers a range from 20 Hz (low) to
20 kHz (high), so a sample rate of 44.1 kHz should theoretically cover
most audio needs. It is also the standard for CD audio, which requires
near optimum sound quality. Therefore, the higher the sample rate,
the better the quality of sound that is reproduced. However, this also
means that the higher the sample rate, the greater amount of audio
data produced and consequently the larger the file size. This means
that there is a direct correlation between the sample rate, the
quality of sound and the file size of the audio file.
An example of how this affects the quality of digital audio is illustrated
by the example provided in Figure 10. A music track that has an
optimum frequency of approximately 20 kHz, the highest audible
frequency perceived by the average human ear, needs to be sampled
at 44.1 kHz in order to maintain CD quality sound fidelity. However, if
the same track is sampled at a rate lower than 44.1 kHz eg 30 kHz,
then according to the Nyquist Theorem, the range between 15 kHz
and 20 kHz will be lost and therefore the sound quality will deteriorate.
The reason for sampling below the recommended rate of the Nyquist
theorem, would be where the sample rate is determined by the

transmission technology. For example, telephone wires and the bandwidth
allocated to radio transmission, where low data rates and storage space are
considered over sound quality.
Sampling rates are directly linked to the desired sound quality produced,
therefore, different audio types and delivery methods require different
sampling rates (see Figure 11). Many applications do not require a wide
frequency range to produce an ‘acceptable’ level of sound quality. The highest
audible frequency in the human voice is approximately 10 kHz which is
equivalent to a sample rate of 20 kHz. Telephone systems, however, rely on
the fact that even with the highest audible frequency of 4kHz (a sample rate
of 8kHz), the human voice is perfectly understood.
Sampling rates for radio broadcasts are also confined within frequencies that
suit the required quality of the sound produced. AM radio has been broadcast
since the early 1900s and in the 1920s it was allocated to a specific
frequency. Due to the limited technology of the period, in relation to the
capabilities of radio and electronics, the frequencies for AM radio were
therefore relatively low. Edwin Armstrong developed FM radio in the 1930s.
His intention was to produce high fidelity and static free broadcasts, therefore
requiring higher frequencies. Although FM radio was available earlier, it was not
really popular until the 1960s.
The sampling rate used for CD is 44.1kHz or 44 100 samples per second. This
relates directly to the Nyquist Theorem whereby in order to produce high
quality sound, the sample rate must be at least twice the maximum audible
frequency signal. So for a CD to produce audio up to a maximum frequency of
20 kHz, which is the upper limit of human hearing, then it requires a sampling
rate of 40Khz. The standard sample rate for CD, however, is set at 44.1kHz.
Quality Sampling Rate
Telephone
AM Radio
FM Radio
CD
DAT (Digital Audio
Tape)
8kHz
11.025 kHz
22.050 kHz
44.1 kHz
48 kHz
Figure 11: Some common sampling rates

Digital Audio Tape (DAT)
Developed in the late 1980s the DAT is still used by some sound recording
studios for both direct recording or data backup. They resemble small cassette
tapes in appearance but they have the capacity to record up to 2 hours of
audio. The DAT recording process is similar to cassette recording but the qual-
ity of recording can be compared to CD quality or higher, with 3 possible sam-
pling rates; 32kHz, 44.1kHz and 48kHz. DAT Recording is also discussed later
in the Hardware Considerations section of the notes.
Stereo and Mono
Audio is typically recorded in either Mono or Stereo.
A stereo signal is recorded using two channels and when played through
headphones will produce different sounds in each speaker. This allows for a
more realistic sound because it mimics the way that humans hear, therefore
giving us a sense of space.
Mono signals, on the other hand, have identical sounds in each speaker and
this creates a more unnatural sound - ‘flat’ sound. This is a major
consideration when digitising audio, in that it will take twice as much space to
store a stereo signal compared to mono signal.

Digital Audio Formats
An audio file consists of two main components; a header and the
audio data. The header stores information in relation to
Resolution, Sampling Rate and Compression Type.
Sometimes, a wrapper is also used which adds information
about things such as license management information or
streaming capabilities (see Figure 12).
Digital audio files can be found in a huge variety of file formats but basically
these files can be divided into two main categories:
1. Self–Describing
2. RAW
Self-Describing formats are usually recognised by their file extension. The
extension, which is part of the file name, will refer to the type and structure of
the audio data within the file and it instructs the user and the computer in
relation to how to deal with the sound information.
RAW formats are files that are not compressed. They rely on the sound
software to correctly interpret the sound file by reading the data or code of
the header component.
File formats are used for different purposes and they vary in terms of file sizes
created. Therefore, when choosing an audio file format, its function and
eventual context need to be considered. This is particularly important when
working with audio files for the web.
http://www.teamcombooks.com/mp3handbook/12.htm
Figure 12

Common Audio File Formats
1. Wave File Format (.wav)
This is a Windows’ native file format for the storage of digital audio data. Due
to the popularity of Windows, it is one of the most widely supported audio file
formats on the PC. WAV files are usually coded using the PCM – Pulse Code
Modulation format. PCM is a digital scheme for translating analog data. WAV
files are uncompressed and therefore have large file sizes. It is a RAW format
that is often used for archiving or storage. The audio data within the wave file
format is stored in a chunk, which consists of two sub-chunks; a fat chunk
that stores the data format and a data chunk that contains the actual sample
data.
The WAV format supports a variety of bit depths and sample rates as well as
supporting both mono and stereo signals.
2. Audio Interchange File Format (.AIFF)
This is an audio file format that is a standard audio format used on Macintosh
systems, although it can be used on other platforms. Like the WAV file format,
the audio data within an AIFF file format uses the Pulse Code Modulation
method of storing data in a number of different types of chunks. This is a
Binary file format that is quite flexible, as it allows for the storage of both
mono and stereo sampled sounds. It also supports a variety of bit depths,
sample rates and channels of audio.
3. MPEG – Encoded Audio (.MP3)
MPEG audio is a standard technology that allows compression of an audio file
to between one-fifth and one-hundredth of its original size without significant
loss to sound quality. The MPEG audio group includes MP2, MP3 and AAC
(MPEG-2 Advanced Audio Coding).
The most common, however, is MPEG 2 Layer 3, which has the file extension
MP3. MP3 compression makes it possible to transmit music and sound over
the Internet in minutes and can be downloaded and then played by an MP3
Player. There are several free MP3 Players, but many are not streaming and if

they are streaming, they use different, often incompatible, methods of
achieving the playback. MP3 files can be compressed at different rates but the
greater the compression the lower the sound quality. MP3 technology uses a
lossy compression method, which filters out all noise that is not detectable
to the human ear. This means that any ‘unnecessary’ information is deleted in
the compression process, which results in a file that is a fraction of the original
WAV file but the quality remains virtually the same. The main disadvantage of
MPEG compression in software, is that it can be a really slow process.
4. Real Audio (.RA, .RM)
Real Audio is a proprietary form of streaming audio (described later) for the
web from Progressive Networks’ RealAudio that uses an adaptive
compression technology that creates extremely compact files compared to
most other audio formats. The resulting bit rate can be optimised for delivery
for various low-to-medium connection speeds. Real Audio either requires a
Real Audio server or the use of metafiles, otherwise the files won’t
download and play.
Real Audio is a good choice for longer audio clip sounds because it lets you
listen to them in ‘real-time’ from your Web browser and the sound quality of
the high bandwidth compressions is good. Real Audio players can be included
with a web browser or can be downloaded from the web.
5. MIDI – Music Instrument Digital Interface
MIDI, or Musical Instrument Digital Interface, is not an actual audio file
format but rather a music definition language and communications code that
contains instructions to perform particular commands. Rather than
representing musical sound directly, MIDI files transmit information about how
music is produced. MIDI is a serial data language, composed of MIDI
messages, often called events, that transmit information about pitch,
volume and note duration to MIDI-Compatible sound cards and
synthesizers.
New Audio File Formats
AAC
Keeper of the format: the MPEG group that
includes Dolby, Fraunhofer (FhG), AT&T, Sony, and
Nokia
Size: Smaller than MP3
Extension: *.aac, *.m4a
Writer of the format: QuickTime 6x supports
AAC. Other encoders such as Real Networks are
starting to support AAC.
File size: AAC files are approximately 50% smaller
than MP3 files.
Sound quality: AAC files have a quality better
than MP3.
Comments: AAC files are based on MPEG 4 and
have a better compression and higher quality than
MP3. Apple wants AAC to become the industry
standard audio format.
WMA
Keeper of the format: Microsoft
Size: Smaller than MP3
Extension: *.wma
Writer of the format: Various encoders will write
the WMA format.
File size: WMA files are approximately 50%
smaller than MP3 files.
Sound quality: WMA files have a quality better
than MP3.
Comments: Better compression and higher
quality than MP3. Microsoft wants WMA to become
the industry standard audio format.

Messages transmitted include:
• Start playing (Note ON)
• Stop playing (Note OFF)
• Patch change (eg change to instrument #25 - nylon string guitar)
• Controller change (eg change controller Volume to value from 0 to 127)
It was initially developed to allow sequencers to control synthesisers. Older
synthesisers were Monophonic, that is, they were only able to play one note
at a time. Sequencers could control those synthesisers by voltage and a
trigger or gate signal that told you if a key was up or down. Contemporary
synthesisers are Polyphonic, enabling them to play many notes at once,
which is more complex. A single voltage was not enough to define several
keys so the only solution was to develop a special language; the Midi. It has
much smaller file sizes than other audio file formats, as it only contains player
information and not the actual direct sound. The positives of the MIDI are its
small file size but the disadvantage is the lack of direct sound control.
To play MIDI files you need two things:
• Either a MIDI plug-in or a MIDI helper application and
• A MIDI device, which can take the form of a soundcard,
an external MIDI playback box or MIDI keyboard, or a software-
based MIDI device, such as the set of MIDI sounds that comes with
the current version of QuickTime.
These are the most common audio file formats in the current market but in
the past, computers that had sound capabilities developed their own
proprietary file formats.
The following is a list of same of the current proprietary file formats:
• .SFR – Sonic Foundry
• .SWA – Shockwave
• .SMP – Turtle Beach

Streaming and Non-Streaming Audio
Audio files, by their very nature, are data intensive, which can result in
large file sizes; particularly if the audio track requires high sound quality
and needs to be more than a few seconds or minutes in length. These
considerations become of paramount importance when an audio file is
incorporated into a web page. Depending on the type and size of the
audio file, a user may experience a long delay between clicking on an
audio link and hearing the sound. This is because the entire audio file
needs to be downloaded before it can be played. An audio file embeded
into a webpage eg a sound effect will be dowloaded into the browser’s
cache. With other audio files the user will be asked where to save the file
on their hard drive. This method of downloading a complete sound file
and subsequently playing it, is referred to as Non-Streaming Audio.
An audio technology called Streaming alleviates this delay in sound
delivery and allows the user to hear the sound immediately or with only a
slight pause. It also prevents users from saving copies of the file to their
computer.
Streaming audio uses a buffering system whereby a buffer space in the
form of a temporary file is created in RAM or Virtual Memory and the
audio data is transferred to this when the user clicks on an audio link.
Within seconds, the buffer becomes full and the audio begins to play.
Once this portion of information is used, more audio data is downloaded
while the sound is playing. Audio data in the buffer is continually
overwritten until the file has finished playing.
The smoothness of playback of the audio file is directly linked to the ratio
between data download rate and the data rate required for playback. If
the audio data can be transferred as quickly as it is used, then the file
with play smoothly. Another factor that determines the quality of the
streamed sound is the user’s machine and the mode of data transfer. The
faster the user’s modem, the fewer ‘glitches’ will occur during playback of
a streamed file (see Figure 13).
Figure 13: Streaming audio
http://www.cit.cornell.edu/atc/materials/streaming/
definition.shtml
The Principle of Streaming
(A snapshot in time)
Time
The portion in the bufferThe portion you are viewing
The portion on your
hard drive at one time
The entire streaming audio or video

A 56 kb modem is the recommended minimum speed for streaming audio.
However, even a fast modem processes data at a relatively slow rate and this
means that the audio data needs to be compressed in order for it to be
channelled through the modem to be played back at an acceptable quality.
Shockwave Audio
One of the leading providers of streaming audio is Macromedia’s Shockwave
for Director, which also includes an animation player. Shockwave Audio,
developed by Macromedia to stream high quality audio over the Internet,
uses very sophisticated mathematical analysis to compress audio so that it
can be represented by relatively few bytes of data. This much smaller data
stream is sent through the user’s modem; it is then uncompressed in the
user’s computer, converted back into audio and then played back through the
speakers. Shockwave audio is scalable, which means that you can select the
quality level to use for the audio playback. A high quality setting, for example,
may be too data intensive to squeeze through a modem in real-time. In this
case, ‘gaps’ may be present in the audio playback.
Streaming audio, like Shockwave, may require a Plug-in Player. A Plug-in is a
program that can be downloaded and installed on a user’s computer in order
to extend the capability of the web browser by allowing a more seamless
integration of many different kinds of file formats into the browser
environment. The web browser automatically recognises plug-ins and their
functions are integrated into the main html file.

Audio Compression
Compression is the reduction in size of data in order to save space or
transmission time. Generally, compression can be divided into two main
categories: Lossy and Lossless compression. The main objective of both
of these compression techniques is to decrease file size, however, this is the
only similarity between these two compression types.
Text documents can be compressed at extremely high percentages of the
original file size eg on average 90% but audio files can only be compressed to
approximately 25 – 55% of the original file size. Although, this compression
percentage may not seem ideal, it is very useful when reducing audio file sizes
that need to be transferred over the internet or for archiving audio files.
Lossless audio compression (eg Monkey’s Audio) is similar in concept to
using a program like WinZip to compress a document or program. The
information within the audio file is minimized in terms of file size, whilst still
maintaining the fidelity of the original data. This means that the compressed
file can be decompressed and still maintain the identical data of the original
file; with no loss to the audio quality.
Lossy audio compression (eg MP3), on the other hand, does not maintain
the identical fidelity of the original audio file and in fact, does not compress all
of the audio data. Lossy compression methods analyse the audio data in the
file and then discards any information that seems ‘unnecessary’ in order to
reduce the file size. This discarded information is not usually discernible by the
human ear and therefore does not alter the ‘perceived’ quality of the audio.
Any compressor will achieve varied ratios of compression depending on the
amount and type of information to be compressed and there are many
different file formats available for both Lossless and Lossy audio compression.
The web is the most obvious location where audio compression becomes of
paramount importance. Speed and efficiency are the two things that the web
relies on in terms of effective data transfer from the Internet pipeline to the
end user’s machine. Therefore, the smaller the file size the faster the data is
transferred.

There are several ways you can reduce the size of an audio file for delivery
on the web. The first and most obvious method would be to consider the length
of the track. There will be a significant difference for example between 1 minute of
recorded audio, as opposed to 40 seconds (see Figure 14). The next
consideration would be the number of channels; does the track need to be in
stereo or could it be converted to a mono recording. By converting the file to only
one channel you have already effectively reduced the file to a half of its original size
and a half of the download time.
Another way to reduce the file size is to change the bit depth from a 16-bit track,
for example to an 8-bit track. The final way to reduce the size of an audio file is to
alter the sample rate. The key in creating digital audio files for the web is to
experiment with the various recording settings, in order to find an effective balance
between sound quality, performance and file size (See Figure 14).
Hardware Considerations
1. Video Capture Cards
A video capture card is used together with a computer to pass frames from the
video to the processor and hard disk. When capturing video, ensure that all
programs not in use are closed, as video capture is one of the most system-
intensive tasks that can be performed on a computer.
Most capture cards include options of recording with a microphone or line
level signal. A Microphone Level Signal is a signal, which has not been amplified
and has a voltage of .001 (one millivolt). Not surprisingly, microphones usually
generate microphone level signals. A Line Level Signal is a preamplifier and has a
voltage of 1.0 (one full volt) generally created by mixing decks, *Video Tape
Recorders (VTR), tape players and DAT players etc. If your capture card has the
option, you will be able to decide which type of signal you are recording. Your
capture card may have two different types of connectors. The microphone input
is usually (except when using Macintosh system microphones) a 3.5 mini jack
stereo connector. The line input is usually a stereo RCA connector or some
times three-pin XLR connector.
Figure 14: File sizes for one minute of audio recorded at
various bit rates
44.1
kHz
22.05 11.025
16-bit
16-bit
mono
8-bit
8-bit
mono
10.01
MB
5.05
MB
2.52
MB
5.05
MB
2.52
MB
1.26
MB
5.05
MB
2.52
MB
1.26
MB
2.52 MB 1.26 MB 630KB
http://www.beta.peachpit.com/ontheweb/audio/chap1.html
* Video Tape Recorders
VTRs are professional recording and playback
machines which use magnetic tape rolls.

2. Metering and Monitoring
Your capturing software should also allow you to see a
graphic representation of sound levels – it should
display meters. There are different types of meters,
which use a variety of measurements and colour
codes. Regardless of metering systems used, you
should always use the meter to ensure that the
incoming sound does not exceed the recording abilities
of the capture card. Unlike analog systems, which
due to the electrical nature of the signal and the
recording medium, allow for sounds to be recorded at
levels that clip or peak, digital systems don’t allow
for this. Digital recorders can only record levels
within their range capabilities. If the incoming level
exceeds the maximum level, clipping (distortion) will
occur. The result of this is distortion of the digital
sound when played back.
3. Sound Cards and Sound Considerations
A sound card is a peripheral device that attaches to
the motherboard in the computer. This enables the
computer to input, process and deliver sound. Sound
cards may be connected to a number of other
peripheral devices such as:
• Headphones
• Amplified speakers
• An analog input source (microphone, CD player)
• A digital input source (DAT, CD-ROM drive)
• An analog output device (tape deck)
• A digital output device (DAT, CD recordable
CD-R) (see Figure 15)
CD Player, Cas-
sette, VCR etc.
Line In
Microphone
Line Out
Speakers
Head phone
Joystick/Midi
Adapter Plug
Line-out
Line-in
Stereo Amp. etc.
Figure 15: Back of device shown

The core of the sound card is the audio processor chip and the CODECs. In
this context, CODEC is an acronym for COder/DECoder. The audio processor
manipulates the digital sound and depending on its capabilities, is responsible
for converting sample rates between different sound sources or adding sound
effects. Although the audio processors deal with the digital domain, at some
point, unless you have speakers with a digital input, you will need to convert
the sound back into analog.
Similarly, many of the sound sources that you want to input to your computer
will begin as analog and therefore need to be converted into digital. A sound
card therefore needs some way to convert the audio. DACs (digital to analog
converters) and ADCs (analog to digital converters) are required to convert
these audio types and many audio cards have chips that perform both of
these functions. They are also known as CODECs due to their capability to
encode analog to digital and decode digital to analog.
The other factors that can influence the functionality and usability of the sound
card is the Disk Driver, along with the number and type of input and output
connectors (see Figure 16).
4. DAT Recording
DAT (Digital Audio Tape) is used for recording audio on to tape at a
professional level of quality. A DAT drive is a digital tape recorder with rotating
heads similar to those found in a video deck (see Figure 17). Most DAT drives
can record at sample rates of 44.1 kHz, the CD audio standard and 48 kHz.
Recording on DAT is fast and simple. It is as simple as choosing what you
want, setting the levels and pressing record. DAT has become the standard
archiving technology in recording environments for master recordings. Digital
inputs and outputs on professional DAT decks allow the user to transfer
recordings from the DAT tape to an audio workstation for precise editing. The
compact size and low cost of the DAT medium makes it an excellent way to
compile the recordings that are going to be used to create a CD master.
http://www.tweakheadz.com/dat_recorders.htm
Figure 17: DAT recorder designed for hard disk recording,
editing, digital signal processing
Figure 16: RCA Connectors for PC/MAC

5. Mini Disk Players
MiniDisc was developed by Sony in the mid eighties as portable equipment
that combine the storage qualities of CD with the recordabilty of cassettes.
They are very cost effective and run on power or on re-chargeable batteries,
which last for approximately 14 hours of play time.
While CD-ROMs and DVDs use optical technology and floppys and hard drives
use magnetic technology MiniDisc uses a combination of both to record data.
Therefore care should be taken to protect minidisks from strong magnetic
fields. Just like a computer’s hard drive, the audio data is recorded in digitally
and in fragments - this is called Non-Linear recording.
MiniDisc’s use sample rates of 48Khz, 44.1Khz or 36Khz. They uses
compression to enable them to record the equivalent to a full sized CD on to
the 64mm disc. This compression is called ATRAC (Adaptive Transform
Acoustic Coding) incorporates noise reduction and has a compression ratio of
1:5. Similar to MP3 it reduces data by only encoding only frequencies audible
to the human ear
6. Microphones
Computers that have built in microphones are not usually considered to be
high-fidelity devices. When dealing with audio production, the adage ‘garbage
in garbage out’ applies. In essence, nothing can fix poorly recorded sound. If
your audio is going to be compressed, or its sample rate and bit depth are
reduced, then it is very important to record clear, dynamic sounds. Choosing a
good microphone is very important. There are a variety of microphones
available on the market, each offering different sound qualities that are
outlined in the following section, but firstly, let’s discuss how microphones
work.
Figure 18: Sony Mini Disk
http://www.dealtimeshopping.com/DT_a19/
mini_disk_player.htm

How Microphones Work
Microphones work by converting real sound waves into electrical audio signals.
They have a small light material inside them called a diaphragm. When sound
vibrations travel through the air they reach the diaphragm, which causes it to
vibrate. This in turn causes an electrical current which is sent out to a mixer,
preamplifier or amplifier for use. Microphones are generally classed by how
the diaphragms produce sound.
Dynamic and Condenser Microphones
1. Dynamic
Sometimes called a Moving Coil Microphone, this microphone works on an
electro-magnetic principle – that is, a coil of wire moving within the flux of a
magnetic field to produce a small voltage. Dynamic microphones consist of a
fine coil of wire attached to a pressure sensitive diaphragm. This coil is
suspended in a permanent magnetic field and as sound waves hit the
diaphragm, the coil moves within the magnetic field thereby producing an
electrical signal. The microphone, because it generates its own signal
(voltage), does not require a battery.
Dynamic microphones are not as sensitive as higher-grade condenser
microphones and they have a reputation for being reliable and hardy which is
why they are used frequently in live performances where they can take the
rough handling as well as more powerful sound. They are also relatively
inexpensive and have a ‘warm’ sound quality (see Figure 19).
2. Condenser
This microphone works on a Capacitive or Electro-Static Effect. The
condenser microphone is essentially a capacitor with one of its plates being
movable and the other plate fixed (back plate). Once again, a diaphragm
(which is bonded to the movable plate of the capacitor) is used to sense
changing air pressure - sound waves. As the air pressure changes it impacts
on the diaphragm, the gap (insulator) between the movable capacitor plate
Figure 19: A Dynamic microphone

and the fixed plate changes. This changing insulator alters the capacitive
reactance (Xc) and thereby alters the current flowing through the capacitor.
Condenser microphones require batteries to drive the capacitor circuit.
Condenser microphones have smooth, sound quality and a clarity and
definition not usually found in dynamic microphones. Another advantage is
that they can also be miniaturized, making them especially suitable for clip-on
use (see Figure 20). Both dynamic and condenser microphones can be
designed to be directional or omnidirectional.
3. Phantom Power for Condenser Microphones
A power source is required to produce the charge on to the capacitor of a
Condenser microphone. This power source may be provided by either an
internal battery, a permanent charge on the microphone’s diaphragm or by an
external ‘phantom’ power supply.
Phantom power is the supply of power through the ground cable of an XLR
cable. The voltage of Phantom power supplies ranges from 9 volts up to 48
volts. The power can enter the cable from a number of sources; from a
battery pack which is an alternate source to the mains power; a phantom
power box, which is like an intermediate component between a mixer and a
microphone that just puts a charge on the ground cable, or a mixer that might
have a button that enables the phantom power source through the XLR cable
(see Figure 21).
Other types of microphones include Electret Microphones, Plaintalk
Microphones, Ribbon Microphones, and Carbon Granule Microphones.
These can all be further researched on the Internet.
Figure 20: A condenser microphone
Figure 21: PM4 - Phantom Power Adapter for
Condenser Microphones
http://www.samsontech.com/products/
productpage.cfm?prodID=118&brandID=2

Categories of Microphones
Microphones can be used for many different recording uses but unfortunately
they cannot always pick up sound from different directions. The way in which
a microphone detects sound is known as its pickup pattern. The standards
of the pickup pattern are:
1. Omni-Directional
Omni-Directional Microphones pick up sounds from all directions. They
work well, either pointed away or towards the subject, providing that the
microphone is at equal distance. Other factors that have a bearing on how
well the microphone maintains its omni-directional characteristics, is its
physical size. The body of the microphone blocks the shorter high-frequency
wavelengths that arrive from the rear; the smaller the microphone body
diameter the closer the microphone can come to being truly omni-directional.
(See Figure 22).
Typical Uses: Used for vocals because of their lack of proximity effect,
picking crowd noise at a football match or as lapel microphones for
newsreaders, which allows them to keep looking directly at the camera or
telereader.
Figure 22: An omni-Directional microphone

2. Uni-Directional
Uni-Directional Microphones are best at detecting sounds from one direction
ie directly in front. These microphones are generally long and rod shaped with
grooves on the side. This allows sound coming from the side to either pass
through without reaching the pickup or cancel each other out. (See Figure
23).
A slightly modified pickup pattern is also found in specialised Uni-Directional
microphones. These are the Shotgun and Cardioid (Supercardioid and
Hypercardioid) microphones (See Figures 24 & 25).
• Shotgun Microphones are more directional in that they can pick up
close perspective sound, with less background noise, from a greater distance.
• Cardioid Microphones are less sensitive to sounds from behind,
than they are to the sides and front, which is why they are favoured for stage
use. There are two types of the Cardioid pickup pattern. These are called
Supercardioid and Hypercardioid, which have limited ranges of pickup. (See
Figure 25 on previous page).
Typical Uses: Good for noisy locations to hone in on sounds such as in an
interview at a sports game. Good for drum and
instrument applications.
3. Bi-Directional
Bi-Directional Microphones pick up sounds in two-axis -
from two opposite directions. This is known as the
figure-8 pickup as, when viewed from above, the
pattern resembles a figure-8 (See Figure 26).
Typical Uses: Generally used in interview situations.
Most stereo microphones can be used as bi-directional
devices.
Figure 23: Uni-Directional microphones
Figure 24: A Shotgun microphone
http://www.micsupply.com/festivariandelight.htm
Figure 25: A Cardioid microphone
http://www.aes.harmony-central.com/109AES/Content/
Earthworks/PR/Z30X.html
Figure 26:
A Bi-Directional microphone
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