DIGITAL SOUND

Analogue vs digital signals

Analogue Sound vs Digital Sound

Sound (e.g. music, speech) is an air pressure (analogue signal) that is

sensed by our ears.
A transducer such as microphone converts continuously varying
sound pressure into a continuous varying electrical signal.
To change into Digital signal, the electrical signal is converted into
sequence of numerical value proportional to the strength of the
signal and the numerical values are stored and processed.
The digital sound can be converted back into analogue form using a
DAC.

Conversion from analogue signal to

digital and vice-versa

WHY DIGITAL SOUND?

Analog transmission is not particularly efficient. When analog
signals become weak because of transmission loss, it is hard to
separate the complex analog structure from the structure of
random transmission noise. If you amplify analog signals, it also
amplifies noise, and eventually analog connections become too
noisy to use.
Digital signals, having only "one-bit" and "zero-bit" states, are more
easily separated from noise. They can be amplified without
corruption. Digital coding is more immune to noise corruption on
long-distance connections.

PULSE CODE MODULATION

PCM is a process for coding sampled analogue signals by recording the
height of each sample in a binary electrical equivalent
Level
(rounded)

Time

Binary
Representat
ion

000

001

100

110

111

110

011

001

000

More PCM

Sampling and quantization of a signal (red) for 4-bit PCM

PCM STEPS

PAM
1. Samples are taken of the analogue signal at fixed and regular intervals of

time. The sampling rate or sampling frequency is at least twice the highest
frequency in analogue signal (Nyquist criterion). This process is known as PAM
(Pulse Amplitude Modulation)

PCM Quantization
2 . To produce PCM data, the PAM samples are
quantized.

PCM ENCODING
Finally, the height of each
PCM pulse is encoded in n
bits to produce the digital
output. The output from
PCM encoder is a sequence
of fixed height pulses. The
train of pulses may then be
stored in memory in groups ,
where each group consists
of 4 bits.

Why Sampling?
Sampling Rate - The number of samples taken per second. It is
expressed in hertz (Hz).
Hertz (Hz) - the SI unit of frequency defined as the number of
cycles per second of a periodic phenomenon
The discrete approximations (in red) can be used to recreate the
original sound (grey). However, due to limitations in the number
of samples we take we are often unable to truly represent a
sound wave, though we can get close enough for the human ear
not to notice the difference.

Sampling Rate (1)

To create digital music that sounds close to the real thing you need to look at the
analogue sound waves and try to represent them digitally. This requires you to try
to replicate the analogue (and continuous) waves as discrete values.
Lower sampling rates decrease the quality of the sound (very distant from the one
being recorded).
Higher sampling rates increase the quality of the sound recording but require
more storage space than lower sampling rates.

Sampling rate (2)

Sampling resolution
Sampling resolution - the number of bits assigned to each
sample
Different sounds can have different volumes. The sampling
resolution allows you to set the range of volumes
storable for each sample. If you have a low sampling
resolution then the range of volumes will be very limited, if
you have a high sampling resolution then the file size may
become unfeasible.
The sampling resolution for a CD is 16 bits used per sample.

Sampling Rate and Resolution

File sizes
Bit rate - the number of bits required to store 1 second of
sound
To work out the size of a sound sample requires the following
equation:
File Size = Sample Rate * Sample Resolution * Length of
sound
This is the same as saying:
File Size = Bit Rate * Length of sound

Calculation of file size

Sample Rate = 8,000Hz
Sample Resolution = 16 bit
Length of Sound = 30 seconds
Therefore the total file size would be:
8,000 * 16 * 30 = 3 840 000 Bits = 480 000 Bytes

DIGITAL AUDIO FILE FORMATS

Features of Audio tools

Editing Tools : Format Converter, Mixer, Equalizer,
Frequency Tuner, Track Splitter, Edits Multiple Tracks
Filters: Band Pass, High/Low Pass, Silence Reduction,
Noise Reduction, Band Stop
Effects: Invert, Pitch, Reverb, Echo, Cross Fading,
Silence, Delay, Stretch, Noise, Trim, Resample,Reduce
Vocals
Recording Ability: Line-In, Streaming Audio, MIDI Files

END