No one can deny that today’s digital technology allows the discerning home recordist to make very high-quality recordings. Affordable MDMs and digital mixers are helping to level the playing field for home studios and small, independent facilities. Why is it, then, that some studios consistently crank out mixes that sound just a tad clearer, a little smoother, and a bit bigger than those produced by the masses? You might chalk it up to better analog gear, such as mics and preamps. Maybe the studio’s acoustics have something to do with it, or perhaps someone with really good ears is mixing the projects. These factors can all have a profound influence on the sound of a recording. But what if you have already got great gear, experienced ears, and a fine-sounding room, and you’re still not capturing that golden sound? What else will deliver that last bit of quality to your digital recordings? If you want the very best quality possible, there are certain practices that you must follow.

Higher is Better

The cardinal rule for digital recording is to deliver the highest possible audio levels to your A/D and D/A converters without clipping. Converters are the single greatest source of distortion in digital audio. (By distortion, I don’t mean the familiar-sounding overdrive effect but rather any aberration from the original analog waveform.)

To keep distortion at an absolute minimum, your levels should be as hot as possible. This uses the full bit resolution (see sidebar, "Digital Definitions") of the converters for the smoothest and most accurate translation of the analog waveform into the digital domain. By the way, this rule applies to the converters in digital reverbs, delays, and other outboard gear the same as it does to the converters in MDMs and digital mixers.

For the same reason, it’s often a good idea to compress a track before it hits an A/D converter rather than afterward in the digital domain. By compressing a track and applying make-up gain to raise its overall level, you are presenting a hotter average signal to the A/D converter and using the full bit resolution the converter can offer. This should give you smoother-sounding, more detailed tracks.

This assumes that you know ahead of time that the track will need compression at some point. Of course, whether or not (and when) you compress a signal should always be a musical decision, not a technical one. Obviously, if your analog compressor isn’t very good and the digital compressors in your workstation or digital mixer sound great, you’ll want to use the latter.

Good Noise?

Once inside the digital domain, precious bits can be lost due to truncation. For example, when the 20-bit converters on a digital mixer, such as the Yamaha 02R, feed a 16-bit recorder, such as an ADAT or DA-88, the "bottom" four bits are lopped off. Similarly, when a mixer’s 24-bit digital output feeds a 16-bit R-DAT’s digital inputs, eight bits are lost. In both cases, the audio suffers a phenomenon called requantizing distortion—the stuff that leads naysayers to proclaim that digital is cold and edgy.

To preserve some of the audio detail contained in the bits that would have otherwise been thrown away and to avoid harsh-sounding requantizing distortion, digital mixers, converters, and other devices often allow you to add dither to the signal. Dither, in its most basic form, is low-level broadband noise that prevents truncation-derived distortion and is designed to preserve detail at low recording levels. In return, it adds a very tiny amount of noise that sounds like hiss. (Because digital signals typically have dither added during the A/D conversion process, adding dither to a digital recording is often referred to as redithering.) Not all dither is broadband, however; there are many different types of dither, each having its own spectral content. (For a more complete discussion of dither, see "Square One: Dithering Heights" in the December 1996 issue of EM.)

When you add dither to a 20-bit signal, before it is recorded to a 16-bit MDM or DAT deck, the effect is heard mostly on signals below -40 dBFS (40 dB below "Full Scale," or digital zero), for example, on fade ins and fade outs. Reverb tails fade smoothly rather than cutting off, and percussive sounds, like drum hits and finger-picked guitar, might be more clearly defined.

I say "might be" because, in reality, sometimes a redithered recording will sound less clearly defined than a truncated version—especially if the quality of the dithering is poor. I’m actually not a big fan of dithering; I prefer Apogee’s UV22 process instead. (I must admit, though, that I haven’t heard every type of noise-shaped dither currently available. Some people, for example, are big fans of Sony’s Super Bit Mapping.) As with dithering, UV22 adds noise to the signal. However, the noise in UV22 is confined to a frequency band centered around 22 kHz (hence the name UV22); therefore it is essentially inaudible.

About eighteen months ago, for my own personal edification, I set up a 3-way test in which I compared various recordings made at -40 dBFS. I listened to how the recordings sounded when processed with UV22, with dither added, and with simple truncation. The UV22 process made reverbs sound more airy. Acoustic guitars and cymbals sounded cleaner and clearer. The stereo positioning of all the elements in the mix was also, by far, the most solid. The truncated recording sounded choked and fuzzy in comparison. The redithered recording, significantly obscured by hiss, was the least pristine of all.

Keep in mind that these tests were done at very low recording levels. At higher levels (peaks up around 0 dBFS), the dither imparted a subtle veil to the mix; truncation made vocals, sax, and harmonica a tad edgy; and the UV22 process added both an airy clarity and an analog-like smoothness to the overall sound.

Unfortunately, UV22 processing is currently offered only on expensive recording gear, such as the Apogee AD-1000 and AD-8000 A/D converters, the Millennia Media HV-3C stereo mic preamp/converter, and the Z-Systems Z-Q1 digital equalizer. Pro Tools users can take advantage of UV22 with Apogee’s MasterTools TDM plug-in (see Fig. 1).

UV22’s effect, like that of dither, is quite subtle and would be lost on most untrained ears, especially when heard on a crummy home stereo system. You should try to audition UV22 for yourself to see if you can justify the expense. Many mastering houses can provide UV22 processing for your project if you don’t have the bucks to buy the gear yourself.

Whether you use dither or UV22 when recording to an MDM or R-DAT, make sure you match the source (e.g., digital mixer) and the destination word lengths to avoid truncation. For example, the Yamaha 02R allows you to select the word length of the digital data to be sent out from its tape buses. If you’re recording digitally to most MDMs or DAT machines, you should choose a 16-bit word length. On the other hand, if you’re recording to the new Alesis M20, you should choose the 20-bit word length for the tape buses.

Avoiding the Jitters

There’s a common myth in our industry that making a digital copy of a track or DAT master will always result in an exact replica with zero degradation. Contrary to popular opinion, however, the way you move data around inside the digital domain can have a noticeable effect on the way your music sounds. This is due to a phenomenon known as jitter.

To understand jitter, it’s helpful to take a look at the importance of synchronizing digital audio bitstreams. When one piece of digital gear sends data to another piece of digital gear, the two pieces of equipment must share the same stable clock. The clock sets the sampling frequency of both devices to be exactly the same. Although two devices may be ostensibly set to 44.1 kHz, their sampling frequencies will drift and become slightly different with respect to each other if they are not synchronized to the same clock. Because the potential for clock drift is compounded when a signal is routed through multiple digital devices, it’s best to slave all digital devices in your studio to a single external master clock.

If the clock is not rock solid, its timing inaccuracies will cause some audio bits to arrive early or late at the receiving device, introducing audible artifacts into your tracks. This is not because the actual values of the bits change; rather, it is because the arrival timing of those bits—the quantizing intervals—drifts. This is loosely analogous to a MIDI sequencer failing to snap notes exactly to a grid when quantizing a track at 100 percent strength—the note values remain the same, but the timing isn’t totally locked in. When this happens in digital audio, mild distortion occurs.

Although the clock-recovery circuits in high-end D/A converters can correct these timing anomalies, many of the converters offered as standard fare on cost-effective digital gear just can’t cope with the problem. For most personal-studio owners, jitter is an unfortunate fact of life.

What does jitter sound like? That depends on how jittery the signal is (that is, how wide the timing variations are). If the jitter is very low, the effect is virtually inaudible. When jitter is audible, it manifests itself in a number of subtle ways depending on the spectra of the jitter itself. The frequency components of jitter can vary widely and can, therefore, modulate the incoming signal in different ways, causing a variety of subtle effects. Because of its chameleonic nature, jitter is something you must train your ears to recognize. Here are some things to listen for.

Jitter is most obvious on stereo tracks (including mixes), where phase anomalies are heard more readily. In most cases, the high-end detail of your mix will suffer. For example, the "ping" of a cymbal hit will be less defined and will lose some of its silvery sweetness. Flatpicking on an acoustic guitar will sound duller, harsher, or lacking in complex overtones.

Clarity in the low midrange often suffers; reverbs become more flattened or 2-dimensional, and you can’t hear as far into the mix. The mix will sound more like it’s coming from two speakers on a flat plane rather than occupying a 3-dimensional space. Subtle sweetener parts that are tucked back in the mix will be a tad harder to hear due to masking. Soundstage localization (the exact pan position of each element within the stereo field) will become a little more vague: the lead vocal might sound somewhat nebulous instead of smack dab in the middle of the speakers.

If the jitter is severe enough, the mix will actually collapse inward slightly from the speakers, resulting in a narrower stereo image. Sometimes, jitter will even rob a mix of a little bottom-end warmth, causing guitars and drums to sound slightly glassy or harsh. The bass guitar and kick drum might not sound as tight and focused as they should.

How serious a problem is jitter? Some gear is more jittery than others. The higher the quality of your digital audio equipment, the less jitter it will introduce into the bitstream. Although the effects of jitter are usually quite subtle, even with budget gear, there’s no amount of EQ, panning, or effects processing that can prevent or undo the damage. Fortunately, there are easy ways to keep jitter to a minimum, so why not get the best out of your gear? Most of the following tips won’t cost you a dime.

Your first line of defense against jitter is to use the most stable clock available as your word-clock master for the entire system. A high-end studio might slave its digital mixer, converters, DAW, and MDMs to a dedicated master audio-sync box, such as the Aardvark AardSync II. Some outboard converters, such as the Apogee AD-1000, are noted for having extremely low jitter and work well as a master word-clock source for your other gear. But you don’t necessarily need to buy any expensive toys to improve your synchronization in a modest setup.

Every piece of digital audio gear, regardless of the price tag, has its own internal clock. If your equipment has word-clock I/O, try synchronizing your system first from one piece of gear (e.g., using the mixer as the word-clock master) and then the other (using your MDM as the word-clock master). See whether one setup sounds better than the other one does.

It is usually a good idea to synchronize your system using a word-clock feed that is independent of the digital audio bitstream. For example, I typically slave my digital mixer to the word-clock output of my Alesis BRC rather than to the clock embedded in the fiber-optic output of my master ADAT. The theory is that more jitter will be introduced if you force the receiving device’s clock-recovery circuitry to extract the clock from a bitstream full of audio data, which it sees as noise. By feeding a master clock to all slaved devices via a dedicated line, you can theoretically keep jitter to a minimum. However, some devices put out horribly noisy word clock, so you should always try synchronizing your system in all possible ways to see what sounds the best.

The length of your digital audio cables also influences jitter. The longer the cable, the higher the jitter. Using a 1-meter cable will make your signal sound better—typically, a tad warmer, smoother, and more detailed—than using a 5-meter cable. Generally speaking, anything longer than five meters should be avoided.

Notice that I said "digital audio cables." Using standard microphone cables for AES/EBU lines or standard coax cables for S/PDIF lines will give you inferior results for two reasons: First, both the AES/EBU and the S/PDIF spec require cabling to have a specific impedance. The wrong impedance will increase jitter. Second, analog audio (and the cables it uses) has a bandwidth in the thousands of hertz. Digital audio, with its clock signals, has a bandwidth in the millions of hertz. If you don’t want to screw up the data going from here to there in a digital system, use cables, such as Apogee’s Wyde Eye cables, that have the necessary bandwidth and impedance for digital audio.

While we’re on the subject of cables, make absolutely sure that any delicate fiber-optic cables in your studio are well protected. A break, kink, or even a sharp bend in a fiber-optic cable spells data loss, dropouts, and distortion. To protect my six 5-meter Alesis fiber-optic cables on their journey between three ADATs and a mixer, I tie them together very (and I mean very) loosely with twisty ties and sheath the entire bundle in Snakeskin from American Recorder Technologies. Snakeskin is a flexible, smooth, springy, tube-shaped material that feels a lot like, well, snake skin. You can unroll it to lay cables inside, and it springs back to a tube shape when released. It resists impact, won’t snag on other gear, can be cut to length, and comes in different diameters. It’s a bit pricey, but hey, so are damaged fiber-optic cables!

One last tip on avoiding data corruption: when copying a master tape from one R-DAT to another, use the shortest length digital cable possible, and go straight from one deck to the other (see Fig. 2). If both decks are patched to a digital mixer, avoid the convenience of running the audio through the mixer. The benefit of using a short "straight wire" path between R-DATs is extremely subtle, but it is perceptible. In my own personal blindfold tests, clones made with a direct connection via a 1-meter Wyde Eye cable sounded slightly warmer and smoother, with silkier highs and tighter stereo imaging, compared to clones made with 5-meter Wyde Eye cables routing the signal through my Yamaha 02R digital console.

The exact explanation for these results is hard to pin down. Some industry experts claim that jitter can affect a digital-domain DAT recording by being incorporated onto the control track, which serves as a clock for the audio samples. Others say this is rubbish and the real culprit is that many digital devices actually change the data they are supposed to pass through unaltered. Whatever the cause, it makes no sense to route your precious music through anything unnecessary. Bite the bullet and repatch for the shortest, most direct signal path.

Know Thy Converters!

Most of the time, the tracks on an MDM are recorded at their optimal 48 kHz sample rate. That poses a problem at mixdown when a digital mixer must deliver CD-compatible 44.1 kHz audio to a mixdown deck, such as a DAT recorder. In that case, you have two choices: use a sample-rate converter to convert from 48 kHz to 44.1 kHz (thereby staying in the digital domain), or go through D/A/D conversion (mixer analog outputs to DAT analog inputs). Which is better? That depends on the converters you have on hand.

For example, when I routed a mix through the 02R stereo bus’ excellent 20-bit D/A converters and then back into the digital domain using the Panasonic SV-3700’s A/D converters, I heard a small but significant decrease in the stereo width, midrange clarity, and high-end detail of the mix. Routing the same (automated) mix through a Z-Systems Z-Link+ sample-rate converter, I lost only about half as much width, clarity, and detail as going through the D/A/D conversions—a major improvement. On the other hand, using the 02R’s D/A converters in conjunction with Apogee AD-1000 A/D converters sounded the best of all. The lesson? Quality, not function, should be the main criterion for which gear you use. (Don’t assume that a 20-bit converter will automatically sound better than an 18-bit converter, either. Specs can lie.) Compare the sound of all the converters at your disposal, and choose the most flattering signal path.

Of course, you can avoid sample-rate conversion by recording at 44.1 kHz throughout the system, including on your MDM. But unfortunately, if you are using an Alesis BRC or other synchronizer whose time-code frame rate is sample-locked to 48 kHz (meaning that the synchronizer will spit out one time code frame for every x number of samples), recording at 44.1 kHz on the MDM will cause the real-time frame rate to slow down by a corresponding amount (approximately 8 percent) from the rate indicated by its format (e.g., 30 fps time code will actually slow down to 27.56 frames per second). This can pose problems for the automation systems on some mixers. The Yamaha 02R mixer, for example, is especially finicky about the rate at which it receives time-code frames and can freak out if the frame rate is more than a few percent slower or faster than its time-code format indicates. In this type of system, you are better off recording at 48 kHz on the MDM and living with sample-rate conversion rather than creating time-code problems for the automation system at mixdown.

The Path to Perfection

The key to making high-quality digital audio recordings is to get into the digital domain as soon as possible (allowing for artistic analog preprocessing, such as compression), keep your levels just below clipping, use the shortest digital audio cables available, and avoid routing your signal through unnecessary equipment.

Whether you use redithering or UV22 is a personal choice. In either case, remember to choose the most stable clock source to synchronize all of the digital gear in your studio, and use a dedicated, high-grade word-clock I/O whenever available. If you can afford the expense, using a dedicated low-jitter master clock (such as the AardSync II) to run your entire studio will yield the best results. This assumes that all your digital gear can lock to an external clock, which is an important feature to keep in mind when considering a new purchase.

Aside from feeding your converters healthy levels, which can make a huge difference in audio quality, most of the suggestions in this article, taken individually, will produce only subtle improvements. Analog techniques, such as proper microphone choice and placement, will typically have a far greater impact on your projects. But little things do add up. If you’re serious about digital recording, you can squeeze a little more quality from the gear you already own.

Michael Cooper is a producer, engineer, and owner of Michael Cooper Recording in Eugene, Oregon. Special thanks to Erik Lovell of Aardvark, Andy Moorer of Sonic Solutions, Richard Elen of Apogee Electronics, Gary Hall of d-House, and Mike Rockwell of Digidesign for sharing their vast knowledge on the subject of digital audio.

DIGITAL DEFINITIONS

A/D converter Analog-to-digital converter, a device that digitizes an analog waveform.

bit depth See bit resolution.

bit resolution The number of bits per sample that a digital device (such as an A/D converter, or a multitrack recorder) uses to convert or store data. The greater the number of bits in a digital sample, the more accurate the digitized description of the instantaneous value of the audio waveform. Also called bit depth or word length.

D/A converter Digital-to-analog converter, a device that converts digital data into an analog waveform.

dither Random data, or noise, added to a digital signal for the purpose of moving data from the least-significant bits to "higher" bits in a digital word. Dither is added to preserve high-resolution detail and to reduce requantizing distortion. Virtually all modern A/D converters add dither to the analog signals they process; therefore, when you add dither to a digital signal, you are actually redithering, or dithering again.

jitter Timing inaccuracies in the transmission and reception of a digital bitstream, which cause distortion.

redithering See dither.

requantizing distortion Distortion caused by shortening the word length of a sample as a result of truncation. Distortion occurs because the intervals at which the waveform are quantized become larger, changing the original waveform.

truncation The shortening of a digital word, wherein data belonging to the least significant bits is lost.

UV22 A proprietary process developed by Apogee Electronics that adds a high-frequency, narrow-band signal just below half the sampling frequency (the Nyquist frequency): around 22 kHz for 44.1 kHz systems such as DAT and CD.

word clock The timing signal that is used in a multidevice digital-audio system to synchronize the sampling frequency at which the system’s component devices operate. To avoid data loss and distortion, all digital devices in a system must be slaved to a single word-clock master so that their sampling frequencies will be exactly the same.

word length See bit resolution.