13.

0 Digital Audio

Lesson Introduction

In previous lesson, you have learned about digital video production and editing.
During this week you will learn about digital audio technologies related to
multimedia product development. The learner will be able to will learn the
fundamental concepts and technologies behind the digital audio. Further, the
learner will be able to obtain the necessary skills for audio editing and publishing.

Learning Outcomes:
After completion of this lesson, the learner will be able to create digital
audio clips and integrate them to videos and animations.
▪ Describe theoretical aspects of digital audio and sound
▪ Apply audio effects
▪ Edit and compress audio files
▪ Integrate audio with other multimedia assets
Lesson Outline:
▪ Introduction to Analog and Digital Audio Basics
▪ Audio data capacity and attributes
▪ Capture and Reproduction Techniques
▪ Audio Filtering and Encoding Techniques
▪ Audio Editing
▪ Sound Effects
▪ Mixing audio and video
13.1 Introduction to Audio and Digital Audio Basics

Audio media refers to the use of sounds in multimedia communication. The audio
integration can be used as an efficient media to transfer information to the
audience. Other than transferring content or vocal information, it may also
helpful to attract the audience while setting their mind suitable to the event.
Audio content can take the form of symbolic sound, vocal sound and music.

Audio or sound is becoming important part of multimedia products as it adds an

extra dimension to the multimedia application. At the basic level, sound provides
a notice or a warning that something is happening. This is a feature commonly
found in desktop applications.

Sound is a physical phenomenon produced by the vibration of matter and

transmitted as waves. However, the perception of sound by human beings is a
very complex process. It involves three elements which are the source which
emits sound, the medium through which the sound propagates and the detector
which receives and interprets the sound.

In general, sound is perceived by the human based on loudness or volume of the

sound. The sound depends on two things which are energy and a transport
medium. As an example, energy from a trumpet is converted into pressure or
vibration that is transmitted by the air. Through air, the vibration is detected by
human ear drum which sensed and that is how human can hear sound. The
vibration of the ear drum can be represented by a time signal or a wave. This
generates an analog sound.

The digital sound can be obtained through a process called digitizing where this
wave can be converted into digital signal. The main two steps involved in the
digitization process are sampling of signal and quantization of signal. Normally,
Sound is generated as a continuous signal, that is an analog signal. To generate
audio data in a computer, this signal needs to be digitized.
Sampling: Sampling is the slicing of the signal at discrete time intervals. The
frequency of sampling or the number of times the sampling is carried out is
determined by the Nyquist theorem. It emphasizes that the perfect reproduction
of the signal is possible when sampling is done more than the twice of highest
frequency of the signal been sampled. Then the output is discrete set of samples.

Quantization: Quantization refers to the process of taking the amplitude values of

the signal at above time sample intervals. Quantization converts actual amplitude
sample values, to discrete set of values. The number of discrete value levels
depends on the number of bits allocated per sample.

Figure 13.1: Audio digitization and reproduction

In the above figure 13.1, it illustrates the digitization process and the storage of
sound signal generated from the microphone through ADC (Analog to digital
converter) process (Digitization).
According to the above diagram, Recording and Digitizing sound is done through
an analogue-to-digital converter (ADC) converts the analogue sound signal into
digital samples. A digital signal processor (DSP) processes the sample, e.g.
filtering, modulation, compression, and so on. The play back of sound is done by a
digital signal processor by processing the samples, e.g. decompression,
demodulation, and so on. A digital-to-analogue converter (DAC) converts the
digital samples into sound signal.
13.2. Audio data capacity and attributes

Sound waves can be characterized by the several attributes. Those are Frequency,
sound resolution, number of channels, Period, Amplitude, Bandwidth, Pitch,
Loudness and Dynamics.

Frequency is the rate which sound is stored in samples per second or Hertz (Hz). It
means for one second how many sound information can be stored. This is also
known as sampling rate, the higher the frequency, the clearer and sharper the
sound. It measures the quality of the overall sound. The unit is Herts (Hz) or
kiloHertz (kHz).

Infra-sound 0 – 20 Hz
Human hearing range 20 – 20 kHz
Ultrasound 20 kHz – 1 GHz
Hypersound 1 GHz – 10 THz

Sound resolution refers to number of bits used to represent each sample. Sound
resolution actually determines the quality of each sample which will give a better
precision in sound. Normally, sound resolution is based on 8-bit, 16-bit or 32-bit.

Sound channel is either mono or stereo recording. Stereo recordings are more
lifelike and realistic sound (3D localization) because humans have two ears. Mono
recordings are fine but tend to sound a bit “flat” and uninteresting.

Period is the interval at which a periodic signal repeat regularly. Frequency

measures a physical property of a wave. It is the reciprocal value of period f = 1/P.

Pitch is a perception of sound by human beings. It measures how ‘high’ is the

sound as it is perceived by a listener. It is an auditory sensation in which a listener
assigns musical tones to relative positions on a musical scale based primarily on
their perception of the frequency.
The other important perceptual quality is loudness or volume. Amplitude is the
measure of sound levels. For a digital sound, amplitude is the sample value. The
reason that sounds have different loudness is that they carry different amount of
power. The unit of power is watt. The intensity of sound is the amount of power
transmitted through an area of 1 m2 oriented perpendicular to the propagation
direction of the sound. If the intensity of a sound is 1watt/m2, you may start feel
the sound. The ear may be damaged. This is known as the threshold of feeling. If
the intensity is 10-12watt/m2, you may just be able to hear it. This is known as the
threshold of hearing.
The relative intensity of two different sounds is measured using the unit Bel or
more commonly deciBel (dB). It is defined by relative intensity in
dB = 10 log(I2/I1), Very often, it will compare a sound with the threshold of hearing
(10-12/m2).

Dynamic range means the change in sound levels. For example, a large orchestra
can reach 130dB at its climax and drop to as low as 30dB at its softest, giving a
range of 100dB.

13.2.1. Calculating Audio data capacity

Using the first three characteristics just described above, you can calculate the
audio file size. The formula to calculate file size is shown below:

C = number of channels (mono – 1, stereo – 2)

S = sampling rate in Hz
T = time in seconds
B = bytes (1 byte for 8-bit, 2 bytes for 16-bit)

File size = C * S * T * B

For example, when calculating a file size for 1 minute, 44.1 kHz, 16-bit, stereo
sound:
C = 2, S = 44100Hz, T = 60s, B = 2 Bytes
File size = 2 * 44100 * 60 * 2 = 10584000 bytes or 10.6 MB
A table of audio sampling rates are given below.

A higher frequency sampling rate means more samples, it gives better quality.
When more samples, the more storage space will be needed.
Higher Frequency -> higher quality -> higher storage space
Sound cards are able to record sound at different sampling rates.

Activity 13.1

Calculate the file size of a CD quality, 2 minutes mono audio file with 16 bits
resolution.

13.3 Capture and Reproduction Techniques

Audio capturing is the process of obtaining a signal from outside the computer.
A common method of audio capture is recording, such as recording the
microphone input to a sound file as shown in the figure 13.1. But, capturing is
not similar with recording, because recording implies that the application
always saves the sound data that's coming in. A capturing application of audio
not necessarily stores the audio. It can do different operation with the data or
process it.

For various sound production, capturing process carryout with various devices,
steps and roles. The capturing devices include various types of microphones
such as omni mic, clip on mics etc. The steps or methods may be live capturing,
one track or multi track capturing can be used. Based on the type of production,
there are various roles such as editors, effect generators, instrument players,
vocalists may involve.

Activity 13.2

Conduct a literature survey and describe the steps involved in multitrack music
production.

13.4 Audio Filtering and Encoding Techniques

Audio filters are used for cleaning the audio signals and pass the wanted audio
signal. In this process unwanted noises will be removed. Filters are used to
clean up a signal, rather than to shape the sound creatively. They only provide
attenuation of unwanted frequencies, and there's no scope to boost any part of
the frequency range. The filters may operate on time domain or frequency
domain. There are two common approaches, one is the high‑pass filter, which
probably the most useful, as it helps to remove unwanted rumbles and other
unwanted sub‑sonic rubbish that microphones tend to capture. A low-pass filter
is one which does not affect low frequencies and rejects high frequencies. The
function giving the gain of a filter at every frequency is called the amplitude
response.

An audio encoding is the process of which audio data is stored and transmitted
with new format of data. Based on the nature of application, there are
guidelines on choosing the best encoding for multimedia application. This
encoding method highly depends on the distribution media and application
domain. Audio data, like all data, is often compressed to make it easier to store
and to transport. Compression within audio encoding may be either lossless or
lossy. Lossless compression can be unpacked to restore the digital data to its
original form. Lossy compression necessarily removes some such information
during compression and decompression and is parameterized to indicate how
much tolerance to give to the compression technique to remove data.

Activity 13.3

Create a list of Lossy and Loss-less audio file formats.

13.5 Audio Editing and Sound Effects

When any production is captured and recorded, it needs to be edited. Here it

can be embedded with special effects. There are editing tools and such a tool
can create sound tracks and digital mixes. There are variety of sound editor
available such as Adobe Audition (commercial), SoundForge (commercial),
Goldwave (shareware), and Audacity(freeware).
The basic sound editing operations that most commonly needed are:
• Trimming
• Splicing and Assembly
• Volume adjustment
• Format conversion
• Resampling and Downsampling
• Fade-ins and Fade-outs
• Equalization
• Time Stretching
• Digital Signal Processing
• Reversing Sound

Trimming: Removing “dead air” or silence space from the front of recording to
reduce file size.

Splicing and Assembly: Cutting and Pasting different recording into one.
Volume adjustment: When combining several recordings into one there is a
good chance that it won’t get a consistent volume level. It is best to use a sound
editor to normalize the combined audio about 80% – 90% of the maximum
level. If the volume is increased too loud, then it will create a distortion.

Format conversion: Saving into different file formats.

Resampling and Down sampling: It will record sounds at higher sampling rates,
then down sample to lower rates by down sampling the file to reduce the file
size.

Fade-ins and Fade-outs: To smooth the beginning and the end of the sound file
by gradually increasing or decreasing volume.

Equalization: Some program offers digital equalization capabilities to modify

the bass, treble or midrange frequency to make the audio sounds better.

Time stretching: Alter the length (in seconds) of a sound file without changing
its pitch.

Reversing sound: Spoken dialog can produce a surreal effect when played
backward.

Digital Signal Processing (Special Effect): To increase pitch, robot voice, echo,
reverb and other special effects.
For obtaining audio editing skills and audio/video mixing capabilities, follow the
adobe audition tutorials given in the following link.
https://helpx.adobe.com/audition/tutorials.html

Activity 13.4

Take an existing audio file and add echo effect using adobe audition or similar
software.