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Matt Shacklady

Mr Gooch

03/04/2015

Digital Audio Theory (DAC)


An analogue to digital converter converts an incoming electrical pressure sound
wave (essentially a changing pattern of electrical pressure/voltage) into binary in
order that the audio signal may be recorded, processed, edited, replayed and
stored by a digital audio device or computer.

The process of re-creating an analogue audio wave from digital PCM data
(Sample rate and Bit Depth) is handled by a digital to analogue converter (DAC).
This involves the process known as successive approximations and is of
interest to digital audio designers.

Page 1 of 5

Matt Shacklady

Mr Gooch

03/04/2015

How do ADCs Work?


Audio analogue to digital converters work by repeatedly measuring the
amplitude of an incoming electrical pressure soundwave (Voltage) and outputting
these measurements as a long list of binary bytes. In this way a mathematical
picture of the shape of the wave is created.

There are 2 important parameters which control the quality of the audio
conversion process:

Sample rate- The number of measurements of amplitude per second


Bit Depth- Accuracy of each measurement of amplitude (Resolution of the
sample rate)

The sample rate is the number of samples taken per second. For example, CD
quality sample rate is expressed as 44.1 khz meaning simply that the converter
takes 44,100 measurements of amplitude per second.
Once set sample rate does not vary during recording, although different audio
files recorded at different sample rates may be used together in a multi-track
system if the software permits it.

Higher sample rates produce better quality recordings but also


bigger file sizes which demand greater space on storage devices and
also faster processers.
Lower sample rates produce poorer quality but also smaller file
sizes which demand less storage and CPUs will transfer over the internet
faster.

Page 2 of 5

Matt Shacklady

Mr Gooch

03/04/2015

Nyquist Theory
A simple rule has been created that determines the appropriate sample rates for
different sound:
The sample rate should be little over twice the amount of the highest
audio frequency to be recorded if poor sound quality is to be avoided
Because humans can hear audio frequencies as high as 20KHz a minimum
sample rate of 44.1KHz was decided upon.
Human audio spectrum= 20Hz to 20,000Hz. Therefore Highest audio
frequency= 20,000 Hz therefore. 20,000 X 2= 40,000 + A little bit
more = 44,100 samples per second.
However, increasing the sample rate above 44.1 KHz does not dramatically
improve the sound. Increasing bit depth has a greater impact.
If the sample rate is set too low then a type off distortion called aliasing will be
audible in the signal when it is converted back to analogue by a DAC.

ADCs therefore employ a low pass filter before the converters to remove any
harmonics from the sound wave which are above the highest frequency that the
sample rate can accommodate. Thus, an ant aliasing filter in the CD recorder will
remove any harmonics above 20KHz from a sound wave before it is converted
and recorded.

Page 3 of 5

Matt Shacklady

Mr Gooch

Page 4 of 5

03/04/2015

Matt Shacklady

Mr Gooch

03/04/2015

Bit Depth
In digital audio bit depth determines the accuracy of each sample measurement.
In audio files, higher bit depths provide a converter with more accurate ruler
(higher bit resolution) to measure amplitude with, thereby producing more
accurate measurements. In audio quality terms, more accurate measurements
mean less distortion of the true shape of the sound wave.
For example, in an 8 bit sampling system, each measurement is recorded as an 8
bit binary byte. Between 00000000 and 11111111 there are 256 possible values.
This means that each sample measurement of amplitude will be recorded as one
of these numbers.

Quantisation
A Ruler with 256 divisions is NOT very accurate. If when a measurement is
taken, the amplitude of the wave does not fall exactly on one of these points,
then the measurement must be rounded up or down to the next nearest point.
This process is called quantisation and results in a distorted recording of the
true shape of the wave.
A measurement which has been rounded up or down is known as a
quantisation error and produces quantisation distortion. At loud signal
levels quantisation errors manifests themselves as noise, but at low signal levels
they can manifest themselves as unwanted audible distortion.

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