PREVIOUS | MAIN PAGE
Computer Based Music Studio
(Really) Basic Concepts
of Digital Audio
Here's some background info to help you understand
what all this digital audio mumbo-jumbo is about...
As you know, computers can only
work with binary data, or "0's and 1's". The zeroes and ones represent
two states, either "off" or "on". This is like having lots of tiny
switches that form a sort of super-fast Morse Code, which a computer
uses to represent real world events (such as musical sounds) in what is
known as binary code.
The audio we
hear from our stereos and home entertainment systems is 'analog audio'.
This means that oscillating voltages are used to represent the original
sounds. Here's how this works:
plays a note in a smoky basement jazz club. The vibrating air coming
from the horn moves the air in the smoky room, and your eardrums vibrate
back and forth along with the vibration of the air molecules. We
experience these vibrations as "sound".
microphone and an
analog tape recorder are set up in the room. The saxophone vibrates the
air around it, setting up a series of pressure changes that radiate
through the air in the room. When these pressure changes reach the
microphone's diaphragm, it shakes back and forth with the vibrations,
much like the tympanic membranes in our ears. The microphone "hears"
these vibrations and converts them into electrical voltages that are an
"analogy" of the air pressure changes that made the original sounds. The
tape recorder's record head then stores these electrical voltages
("analog audio signal") on magnetic tape as magnetic fluctuations.
After the set
is over, we take the tape recorder home and hook it up to our stereo
system. Now we can play the recording back. We play the tape, the
magnetic fluctuations on the analog tape are converted to electrical
voltage changes (analog audio signal) by the tape playback head and the
resulting voltages are sent to our stereo amplifier. The amplifier
changes those fluctuating voltages into current
fluctuations which move our stereo speakers back and forth, far and fast
enough to create disturbances in the air of our listening room that are
almost exactly the same as the original vibrations caused by the
saxophone playing in the jazz club. That's High Fidelity analog audio!
And now, Digital Audio...
happens in digital audio? How is digital different than
original sound is converted to analog audio voltage fluctuations by the
using an analog tape deck, we are now going to use a digital recorder.
Let's use a DAT recorder as our example. The analog audio voltage
fluctuations are fed to a circuit called the Analog-to-Digital
Converter that changes the incoming voltages to digital
"snapshots", 44,100 times a second. Each "snapshot" consists of 16
zeroes and/or ones. Each combination of zeroes and/or ones represents a
different signal voltage. Using sixteen 0's and 1's in each "sample",
one of 65,536 different voltage levels can be described by each sample.
A DAT or CD uses a "sampling rate' of 44,100 samples per second
(44.1kHz). This means that 2,890,137,600 different analog audio voltage
levels can be described each second -- and you're right, that's a lot.
But some say that capturing audio with 16 bits, 44,100 times a second
may not be enough to accurately describe what our ears can hear, so
that's why there is now a push on to record everything in 24 bits,
96,000 times a second ("24/96 resolution"). The latest digital recorders
(such as the new Pro Tools HD system) capture audio at 24 bit, 192kHz
When we want
to actually hear the digital audio, the audio data has to go through a
Digital-to-Analog Converter, which changes the binary
code samples to analog voltage fluctuations that are then sent to a
power amp and on to the speakers, which shake the air molecules in our
listening room enough for us to hear a reasonably accurate reproduction
of the original sound.
How does Digital Recording work?
story in pictures:
captured with a microphone is sent to the inputs of the
Analog-to-Digital Converter, which can be in a soundcard as pictured, or
in a DAT, ADAT, hard disk recorder, MiniDisc or any other digital
The audio is
now stored as binary data (0's and 1's) on a hard disk or magnetic tape.
If a stereo
recording is desired, two Analog-to-Digital Converters will be used. If
an eight-channel recording is desired (like in an Alesis ADAT or Tascam
DA-38 multitrack digital tape recorder) then eight Analog-to-Digital
Converters must be used.
How does Digital Playback work?
audio captured on the hard disk or tape is sent to the Digital-to-Analog
Converter, which converts the digital audio data back to the fluctuating
voltages that make up analog audio.
level' voltages are sent to a power amplifier which turns the signal
voltages into current fluctuations strong enough to move a speaker cone
back and forth. The speaker cone moves the air in a similar way to what
the original sound did, and we hear the recording. The closer the
speaker's fluctuations are to the original sound, the "higher the
fidelity" to the original sound ("HiFi").
playback is desired, then two Digital-to-Analog Converters must be used,
as are found in a typical CD player or DAT recorder.
multichannel playback is desired, as in Dolby Digital Surround (which
uses six channels of audio), then one D-to-A converter (DAC) must be
used for each channel of audio (six DACs are used in this case).
How do I hook all this up to my PC?
stereo soundcard has a pair of Analog-to-Digital Converters (ADC's) and
a pair of Digital-to-Analog Converters (DAC's). The LINE IN of the
soundcard is an analog input, and the LINE OUT is an analog output. The
converters are on the soundcard itself.
audio comes in the LINE IN of the soundcard and is digitized in the
soundcard's ADC. The audio data travels through the PCI bus to the CPU
and is then stored on the hard drive as a digital audio file (a .WAV on
a PC, or as a Sound Designer 2 file or AIFF file on a Mac).
IN -> stereo ADC -> digital audio file on hard disk
To play back that digital audio file, the CPU sends the audio
data through the PCI bus to the soundcard, where its DAC converts the
audio to analog voltages and sends it out through the LINE OUT jack.
audio file on hard disk -> stereo DAC -> stereo LINE OUT
An ADAT or other 8 channel digital recorder is basically like
four stereo soundcards all synchronized together.
INPUTs -> 8 DACs -> digital audio data
digital audio data -> 8 DACs -> 8 LINE OUTs
One of the nicest
features of the ADAT format is the Alesis ADAT Lightpipe digital audio
interface. A single ADAT Lightpipe interface on a soundcard can take in
or send out eight channels of synchronized digital audio. Most digital
mixing boards have at least one ADAT Lightpipe interface, so you can
send eight or more microphones into a digital mixer, send eight separate
channels of audio to your ADAT interface-equipped PC or Mac, and record
those eight separate tracks to your hard drive(s).
Later, you can
open those eight tracks of synchronized digital audio files in a
multitrack digital audio editor and mix down on the computer, while
adding effects, muting bad notes or cutting out unwanted sections.
Once you have
the mix the way you like it, you can record the mix to a stereo audio
file on your hard drive. Then you can apply final tweaking and
enhancements to the stereo files and put them in a playlist for making
an audio CD layout. Finally, you can use a good CD burning program to
burn an audio CD in your computer's CD-Recordable drive.
The audio has
remained in the digital domain all through this process, ever since it
was first sent to the Analog-to-Digital Converters. The only time the
audio needs to be sent to a Digital to Analog Converter for playback
(except for monitoring during mixing, which doesn't get saved) is when
the listener puts the final CD in his or her CD player.
familiar with plugging the outputs from a CD player into the CD inputs
on a stereo receiver. These are analog audio connections, the same as
the signals from a record player, AM/FM tuner or cassette deck. Digital
audio is a bit different, because it is analog audio converted to binary
data. While in binary form, the audio data can be transmitted, processed
and edited with almost no addition of unwanted artifacts like noise and
distortion. "Digital I/O" is the transmitter/receiver and cabling that
allows digital audio data to be transmitted from one device to another.
The best-known digital I/O format is probably S/PDIF, which is used for
digital connections between consumer digital audio devices like CD
players and MiniDisc recorders. It's also used to send the digital audio
data from DVD players to the Dolby Digital or DTS decoders used in
surround sound playback systems. Digital audio data must be converted to
analog form in order for it to be heard through amplifiers and speakers
(in other words, it must be processed by a Digital to Analog Converter).
USING ANALOG INPUT/OUTPUT
When you plug
your microphone into a MIC IN jack on your mixer, you are plugging an
analog audio device (the mic) into an analog audio input (MIC IN), whose
signal is routed to the mixer's level, EQ and effects controls (all
analog) and finally to its MAIN OUT, which is connected to the
soundcard's LINE IN, then to the soundcard's Analog to Digital
Converter, where the sound is 'digitized'. Let's say you save this
digital audio to your computer's hard disk in a file. The audio is now
in binary (digital) form, called 'digital audio data'.
Let's also say
that you have some effects boxes you want to use, and you want to add
these both while recording basic tracks and as added effects on the
recording after it's saved on your computer. Well, if those effects are
digital, the audio coming from the mixer or your soundcard's LINE OUT is
in analog format, and each pass through the effects boxes will add an
analog-to-digital conversion (A/D conversion). So let's say you want to
add reverb to the audio file you've saved. It's already gone through one
A/D conversion, and now it will go through a digital-to-analog
conversion (D/A conversion) to go out to the effects, where it will go
through an A/D conversion, effects will be added, then back through a
D/A conversion, into the soundcard's LINE IN, where it will go through
yet another A/D conversion, so that your new file with added effects can
be recorded to the hard drive. Phew!
audio has traveled out the analog MAIN OUT jacks of the mixer into the
LINE IN jacks of the soundcard, where an A/D conversion took place. Then
you added outboard effects, which added a D/A conversion, an A/D
conversion and another D/A and A/D conversion before being recorded
again. The sound of all those analog and digital circuits have been
recorded into our digital audio file, along with the original sounds. If
the analog circuits and A/D and D/A converters are all of the highest
quality, then there isn't a problem. But we used a cheap mixer and a
computer soundcard, didn't we? And I'll bet you bought that digital
reverb on sale from Guitar Center, didn't you? These usually are made of
the cheapest available parts, and may not sound so great. But to be
fair, you paid what, $120 for the mixer, $99 for the digital reverb and
$15 for that sound card, right?
So now we burn
an audio CD-R from our recording. The audio on that disc has made
several passes through our cheap analog mixer, cheap soundcard and cheap
effects unit, and has gone through several A/D and D/A conversions (some
cheap) already, correct?
When we play
the CD-R disc in a stereo CD player, the audio data goes through a
Digital to Analog (D/A) conversion, and out the LINE OUT jacks of the CD
player, then on to the amplifier and speakers, where you are finally
able to hear the sound. You'll probably notice a flat, sort of
one-dimensional quality to the sound of many homemade music productions
made with PC-based systems. All those cheap analog and digital audio
circuits can really take a toll on ultimate sound quality.
Now let's say
we want to keep things a little tidier and keep the audio in digital
format for as long as possible. Let's start again from the microphone.
The same as
before, the microphone is plugged into the mixer's MIC IN jack. But this
time, we're using a digital mixer. The analog audio travels from the MIC
IN straight to our digital mixer's A/D converter (which should be of
much better quality than the one in the PC soundcard), and the mixer
handles the audio in digital format from there. Any effects are now
"digital signal processing" (DSP) effects, built into the digital mixer.
Now let's also say that our computer is equipped with a digital audio
input in addition to the more common analog LINE IN.
instead of the analog audio going to the soundcard's Analog to Digital
Converters, this time the analog audio was converted to digital audio
data in the digital mixer, and will be sent straight out the digital
mixer's digital output, and then into the computer's digital input. The
digital audio is then recorded onto the hard disk, and then onto the
CD-R. Only one analog-to-digital conversion has taken place, right at
the digital mixer's input. Only one digital-to-analog conversion takes
place, in the CD player.
after the audio was digitized right at the mixer's input, it stayed
digital all the way to the final playback (the CD player). Even adding
digital effects would not necessitate the recording of any more
A/D or D/A conversions, even if those effects were added in the computer
as software plugins. (While monitoring playback, the audio will travel
through the mixer's D/A converters, but you won't be recording this.)
Much more neat and clean, no?
of state from analog to digital and back again causes some degradation
of the audio signal...and that is why Digital I/O is a good thing when
working with digital audio -- the number of A to D and D to A
conversions is kept to a minimum. Also, the quality of those A/D and D/A
converters is of critical importance for recording high fidelity music.
All this translates to cleaner sound.
DIGITAL I/O FORMATS
comes in two formats, Coaxial (Electrical) and
Optical. Coax S/PDIF uses regular RCA jacks, with the digital
audio data being sent as a stream of electrical impulses. Optical S/PDIF
uses a TOSlink optical transmitter and receiver, where the electrical
impulses are converted to light and sent over a thin fiber-optic cable.
is an electrically balanced version of stereo audio transmission.
Although similar to coaxial S/PDIF, it is designed for transmission over
longer lengths of cable, and uses three-pin XLR (Cannon) connectors.
Multiple pairs of AES/EBU can be used in parallel to achieve multitrack
digital audio capability. The multiple pairs will usually be
synchronized to a single "word clock."
ADAT Lightpipe is an 8-channel digital audio
transmission system that uses the same TOSlink optical connectors as
T-DIF is an 8-channel digital audio transmission system
that uses multi-conductor cables, usually with computer-style D-sub
is a newer format that allows simultaneous transmission of up to 64
channels of synchronized digital audio.
is a simplified explanation, to make it easy to understand the basic
process without wading through a lot of jargon and tech-talk. Once you
get serious about digital recording, you may want to do some more
in-depth research on how this stuff works. If so, be sure to check out
Bob Katz's Digital Domain
and read through his excellent library of articles on digital audio.