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The Personal 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:

A saxophonist 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".

A 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!

So what happens in digital audio? How is digital different than analog?

First the original sound is converted to analog audio voltage fluctuations by the microphone(s).

Instead of 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 resolution.

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.

Here's the story in pictures:
 

The audio 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 recorder.

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.

The digital 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.

These 'line 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").

If stereo playback is desired, then two Digital-to-Analog Converters must be used, as are found in a typical CD player or DAT recorder.

If 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).

A typical 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.

The analog 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).

stereo LINE 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.

digital 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.

8 LINE 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.

We're all 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!

The analog 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.

USING DIGITAL INPUT/OUTPUT

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.

Unlike before, 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.

This time, 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?

CONCLUSION

Every change 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

S/PDIF 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.
 

AES/EBU 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."
 

The Alesis ADAT Lightpipe is an 8-channel digital audio transmission system that uses the same TOSlink optical connectors as optical S/PDIF.
 

Tascam's T-DIF is an 8-channel digital audio transmission system that uses multi-conductor cables, usually with computer-style D-sub connectors.

MADI is a newer format that allows simultaneous transmission of up to 64 channels of synchronized digital audio.

The above 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 website and read through his excellent library of articles on digital audio.