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To produce high-quality recordings on your computer, it helps to know a little about the theory of sound and how it relates to digital audio.
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- Sound Basics
- Sound into electricity (analogue audio)
- Analogue to digital conversion
- Resolution, sample rates and sound quality
- Digital audio formats
- Advantages of digital audio
- Further Reading
Sound Basics
Sound is transmitted through the air as waves of high and low air pressure
created by a vibrating object such as a guitar string. The number of vibrations
per second is known as the “frequency” of a sound – low frequency sounds sound
low pitched to us, whereas higher frequencies are heard as higher pitches.
Frequency is measured in kHz, and the accepted range of human hearing is of
frequencies between 20kHz and 20,000 kHz – between twenty thousand and twenty
million vibrations per second (although this decreases, particularly at the
high-frequency end, with age or owing to damage caused by excessive loud noise).
Sound into electricity (Analogue Audio)
To record audio, the sound first needs to be converted into an electrical
signal. The process by which this is achieved takes advantage of the “motor”
effect – the fact that when an electric current is run through a magnetic field,
a force is created. So, if a magnet is suspended within a wire coil and a current
is run through the coil, the magnet will move (or, if the magnet is fixed, the
force will be transferred and the coil will move instead). Conversely, therefore,
if a magnet is moved within a wire coil the opposite will occur – an electrical
current will be generated in the wire.
So, to make a simple microphone, a wire coil is attached to a diaphragm
(a flat piece of material that will vibrate like a drum skin) and a magnet is
fixed within the coil. When sound waves hit the diaphragm it vibrates, and the
spring moves at the same frequency as the sound waves. Since it is moving within
the magnetic field created by the magnet, an alternating electrical current is
created that is proportional to the frequency of the sound vibration that hit the
diaphragm.
To make a speaker, the process is reversed – the electrical current from the
microphone is passed through another coil, again attached to a diaphragm and
with a magnet fixed within the coil. The current in the coil within the magnetic
field causes the coil to move, which causes the diaphragm to move – again at the
same frequency as the original sound vibrations that caused the microphone
diaphragm to move. As the diaphragm moves, it will cause the air around it to
vibrate as well, and will transmit sound waves identical (in a perfect world –
in reality there is always a certain amount of electrical interference, resistance
etc. which affects the resulting sound) to the original sound waves.
This is the principle of all analogue audio recording equipment and is the basic
design upon which all microphones and speakers are modelled.
Analogue to Digital Conversion
Analogue audio works according to variations in an alternating electrical
current. The differing voltages and the alternation of the current corresponds
exactly to the original sound wave (the word “analogue” comes from the same root
as “analogy” and “analogous” – referring to things that relate exactly to
something else). However, digital technology works on completely different principles
– data is converted into binary code made up of ones and zeroes. The analogue
medium can express infinite variation in signal levels, whereas digital technology
works on an “on/off” principle by its very nature (think of a digital watch with
changing numbers compared to an analogue clock face with gradually moving hands.)
So, to convert analogue audio into a digital format it must be converted into
binary code that can be read by a computer.
To do this, the electrical current that represents a sound is passed through
an analogue-to-digital converter. This takes readings of the voltage of the
current many thousands of times per second, which it then rounds to a whole number
and converts into a binary value. Conversely, a digital-to-analogue converter
reads the code and converts each of the binary values to an electrical pulse that
makes up a current very similar to the original analogue signal. These two converters
are usually combined into one AD-DA (analogue-digital, digital-analogue) converter.
Resolution, sample rates and sound quality
Sample Rates
To convert an analogue signal into a digital format, it is passed through an
analogue-to-digital converter. This takes readings or “samples” from the electrical
signal many thousands of times a second and converts the readings into binary code.
So, the higher the number of samples taken (the sample rate), the more accurate a
representation of the original analogue source is delivered and the higher the
sound quality. The standard sample rate for CD-quality audio is 44,100 samples
per second, which is written as 44.1kHz. The majority of professional digital
audio devices operate at 96kHz or higher.
Low Sample Rate
Resolution
The resolution of a digital signal is the range of numbers that can be assigned
to each sample. When the sound is passed through an A-D converter, the values
that are “sampled” are rounded to a binary value. So, the greater the number of
decimal points the number can be rounded to, the more accurate a representation
of the actual reading it gives and the higher the sound quality will be. The
number of binary values assigned to each sample is measured in bits (CD quality
is 16-bit, and most professional audio devices operate at 24-bits or higher).
Low bit rates can cause “quantization distortion”, where the sampled values are
rounded up so much that they no longer give an accurate representation of the
sound, and the digital audio signal takes on a crackly, metallic quality. Higher
bit rates are also capable of transmitting a higher dynamic range, resulting in
deeper bass and clearer high frequencies.
Low Resolution
High Resolution
Digital Audio Formats
Digital audio can be stored in many different formats. In professional audio
work, uncompressed audio file formats are used as these generally give the best
sound quality. Some examples of uncompressed formats are:
AU: A file format (abbreviation for "audio") that originated on the Sun and NeXT
computer systems. Not widely used today.
Audio Interchange Format (AIF, AIFF): File format for Macintosh system sounds,
similar to Windows' WAV format.
Compact Disc Digital Audio (CDA): This is format used for encoding music on all
commercial compact discs.
SND: Another file format (abbreviation for "sound") similar to the AU format and
used primarily for Macintosh system sounds.
Waveform Sound Files (WAV): This format (pronounced "wave") produces an exact
copy of the original recording, with zero compression.
A problem with uncompressed digital audio is that the file sizes can be very large,
however. To combat this, a variety of compressed audio formats exist that are mostly
used to transfer audio over the Internet. These use “codecs”
(short for “compression/decompression”). A codec converts an uncompressed digital
audio file into a smaller format, and will also decode the compressed file enabling
it to be played back. Some codecs work by physically removing data from an uncompressed
audio file to make it smaller. In this case the file cannot be converted back into an
uncompressed format from the compressed file. This is known as a “lossy” compression
method. Others allow the original uncompressed file to be retrieved from the compressed
file – these are known as “lossless” compression methods. Lossless compressed files are
generally larger than lossy ones, though.
Advantages of Digital Audio
The debate over whether digital or analogue audio technology sounds better is
long running and will doubtless continue for years to come. In truth, the difference
in sound quality between top-of-the-range digital and top-of-the-range analogue
equipment is almost indistinguishable. However, for the home user, digital
technology has a number of benefits over analogue:
Value for money
Noise resistance
Copyability
Durability
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A simple moving-coil microphone
A simple loudspeaker
At low sample rates, the gaps between each reading of the current are
large, and the wave is not converted very accurately, resulting in lower sound quality.
High Sample Rate
At high sample rates, the gaps between each reading of the current are
small, so a more accurate representation of the wave is produced, resulting in
higher sound quality.
At a low resolution, the readings are heavily rounded, giving an inaccurate
representation of the wave and a lower sound quality.
At a higher resolution, the readings are rounded less and give a more
accurate representation of the wave, resulting in a higher sound quality.
Digital technology has made home recording accessible to a much wider audience.
Computer multitracking software has in many cases removed the need for hardware
mixing consoles and effects, and as such allows a basic recording setup to be
much cheaper and more compact. In particular, software emulations of analogue
hardware (such as vintage synths) are considerably cheaper than the real thing!
Digital recording is immune to much of the electromagnetic interference that
causes noise on analogue recordings. This makes it much more suitable for home
recording, as often home studios operate in less-than-ideal surroundings without
any of the expensive electrical isolation and shielding of professional analogue
studios.
Digital audio can be copied infinitely without any loss of quality, unlike
analogue audio, which loses quality with each reproduction.
Digital media such as CDs are much more durable than their analogue counterparts.
Further Reading
For more information, try some of these titles from our Books & Guides section:
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