aaWhen we want to digitise an analogue audio signal, we feed the analogue signal

through a
digitaltoanalogue
convrter (DAC). What the DAC does is measure the voltage of the
analogue signal at very precise intervals. The number of times it does this per
seond is
referred to as the Sampling Rate. You may have heard of the Red Book Standard, w
hich is
what audio CD's use. This specifies a sampling rate of 44,100 samples per second
. This
means that 44,100 times every SECOND, the DAC measures the voltage of the analog
ue
signal, converts that value into a binary number, and passes that number onto th
e
computer so that the computer can store that information ready for later retriev
al. The
Sampling Rate used must always be AT LEAST double the highest frequency we wish
to
hear at playback (google "nyquist theorem" for more). Because adult humans can h
ear up
to roughly 20,000 Hz (it's different for everyone), we need a Sampling Rate of a
t least
40,000 samples per seond in order to play back a digital signal of 20,000 Hz.
So, that's a brief introduction to sample rate.
But there's another element to the sampling process, and that is the Bit Resolut
ion (many
people mistakenly refer to it as bit rate, but that is not entirely accurate, as
you shall see
later in this essay). The Bit Resolution is usually something like 16 bit, 24 bi
t, or 32 bit.
Other resolutions are possible, but those 3 are the most common in use today. Bu
t what
does Bit Resolution do exactly? In short, it determines how many possible values
can exist
in the system to represent the AMPLITUDE of the signal we are digitizing. In a 1
6 bit
recording, we can have 2^16 values to represent all amplitudes between the noise
floor
('digital silence' for want of a better label) and the loudest possible value (w
hat is referred
to as 0dBFS, or "zero decibels full scale"). So, in a 16 bit digital recording,
we can have
65,536 different values to represent the amplitude of our incoming analogue sign
al. You
might think that that should be plenty of resolution, but actually, it's not eno
ugh. Hence
why we now have 24 and 32 bit systems which respetively, offer over 16 million a
nd 4
billion possible values to represent our analogue signal.
OK, hopefully, I haven't lost you yet!
So, to digitize a signal, we need both a Sampling Rate (how often we sample the
signal)
and a Bit Resolution (how accurately we describe the amplitude of the signal).
When we record to a file stored in .wav (or .aiff if you're on a Mac... it's the
same format,
just different info in the file header), we are storing every value accurately,
exactly as the
DAC described it. However, over the last 20 years or so (from the mid 90's to th
e mid
teenies), the file sizes generated by storing audio data in this format were lar

ge. Too large

for being able to carry vast quantities of digital audio with you, too large for
transferring
audio files across the internet, and so on. With a 44.1kHz sampling rate and 16
bit
resolution, 1 minute of stereo audio requires a little over 10MB of hard disk sp
ace.
Of course, as of this writing in 2016, those issues are becoming less and less o
f a concern,
as internet speeds get faster, and hard drives contine to grow in capacity at st
upifying
rates.
But, that wasn't always the case.
So, some smart people working in psychoacoustics started to develop algorithms l
ike mp3
encoding. What these algorithms are designed to do is to analyse the frequency c
ontent
and harmonic structure of a .wav/.aiff file, determine which parts of the inform
ation are
surplus to requirements, throw them out, and store what's left.
Say what??
Surplus to requirements?
What is he talking about?
ALL of my audio signal is important!
Yes, you are welcome to believe that. But it has been shown that because of a ph
enomenon
known as 'masking' (two different sounds occuring at the same time, but one sign
al
temporarily hides, or masks, the other), we really can do some tricky things wit
h the
encoding where we discard that part of the audio signal which our brains won't m
iss. Now,
there is no onesizefitsall
when it comes to encoding, and that's why we can encode our
mp3 files with minimal removal of content (and thus ending up with files which a
re only a
Bruce Williams, Professional audio engineer
615 Views Most Viewed Writer in MP3 (audio encoding format)
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