Did you know that approximately 40,000 new songs are
added to Spotify's libraries every day? Streaming and downloading music has
never been easier.
Do you know how audio files work on your computer?
Knowing what they are composed of and how they work can help you to know how
much disc space or internet bandwidth you need.
Why not take a few minutes to read our in-depth
article to find out how music files work on your computer.
The Start of the Digital Era
You likely use digital files every day in the form
of mp3 files or even tracks stored on compact discs. you
likely switch between digital formats on a regular basis or even convert
digital audio files yourself using tools such as the ones on this page: https://setapp.com/how-to/convert-m4a-to-mp3-on-mac.
However, these technologies are actually the
digital form of technologies that have been used for a long time.
Radio signals, grooves on a vinyl record were all
precursors to the digital music age that we enjoy today. The major change came
with the introduction of the computer and in particular the digital converter.
How do digital converters work? Let's find out.
Digital Conversion
In short, a digital converter takes an audio signal
and takes tens of thousands of photographs of it. In doing this, it is
basically converting it into binary code, or a language that the computer can
understand and manipulate.
Although it takes a lot of these pictures, it is
not a true replication of the original as there is space in between the
snapshots. However, it is of a quality high enough to ensure that the human ear
cannot notice this.
The higher the quality the more information it
provides to the computer. This is especially important if you plan to
manipulate or edit the track later. The quality of the conversion depends on
the sample rate. What is this?
Sample Rate
To continue the illustration, the digital converter
takes photographs of the audio waveform. Each one of these photographs is
actually known as a "sample". The higher the frequency of these
samples, also known as the sample rate, the higher the quality of the
conversion.
While audio engineers use different sample rates
for different purposes, commonly used sample rates include:
- 44.1
kHz (Equivalent to CD Audio)
- 48
kHz
- 88.2
kHz
- 96
kHz
- 192
kHz
You will notice that 44.1 kHz is the lowest rate
permissible. Why is it limited to this I hear you ask? It is because of the
Nyquist-Shannon Sampling Theorem.
The Nyquist-Shannon Sampling
Theorem
To correctly record audio, a digital converter must
be able to record samples from the full spectrum of human hearing. The spectrum
of human hearing is generally considered to be 20Hz – 20kHz.
To be able to accurately represent this, you will
need to take at least two photographs or samples from each waveform. One at the
top and one at the bottom.
Since the human hearing range is 20 Hz - 20 kHz, to
take two samples will require a sample rate of at least 40 kHz. In reality,
audio engineers use a slightly higher rate of 44.1 kHz.
This is the limitation placed on the sampling of
audio tracks. So the higher the sample rate the better for everyone right? Not
exactly.
The Impact of High Sample Rates
High sample rates do provide a better quality of
sound and more information for the computer to work with. However, this brings
some side-effects that can be considered as drawbacks in certain situations.
The overall size of the file is considerably
larger. This means that the speed that the computer can work will be slower.
You will need a larger disk space to hold tracks with a high sample rate. To
really benefit from tracks with a high sample rate, you should use high-end
speakers.
Audio Compression
Once you have created a digital copy of your audio,
you then need to decide what format you will use to store and transport it. To
convert the file to a type that most media players can work with you, it may
need to undergo compression.
In general terms, there are two types of
compression that are used. Lossless compression involves
bringing the file into a condition where you can play it on media players, but
with little or no data loss.
However, engineers perform most audio compression
using "lossy" compression. This means that in order to achieve a
particular file type and size, some file quality is permitted. Using this form
of compression it is possible to shrink audio files in mp3 or AAC format to
1/10th of their original size.
Does Compression Degrade Quality?
If an engineer lowers the size of the file this
much, surely the listener will be able to tell the difference? Actually, most
people do not notice, and using standard audio equipment it is debatable
whether anyone can tell the difference.
This is because engineers employ auditory masking.
An engineer in a studio using high-end speakers will likely know the difference
between an original audio file and a highly compressed mp3 file.
However, if the engineers have processed the file
according to industry standards, it will meet the expectation of the average
listener. In short, they will not know the difference. Further, the equipment
that the majority of people use will not indicate any missing quality.
The result is that unless you are a true
audiophile, you will likely have been listening to audio tracks for many years
without knowing that a higher quality exists somewhere in the world.
Everything You Needed to Know
About Audio Files and Much More
Whether you streaming downloading or creating audio
files it pays to learn a little about the technology behind them. By doing this
you can calculate how much disk space you need and your internet bandwidth
requirements.
If you would like to know more about audio files
and other types of technology, keep reading our blog.
