Jun 1, 2008 12:00 PM, By Bob Hodas

CONSIDERING DIGITAL EQUALIZERS FOR STUDIO MONITORS

More and more, the recording studio environment is becoming one of plug-ins replacing outboard gear. I know several engineers who are mixing inside the box and forgoing the use of mixing consoles. As an extension of that, many of my room-tuning clients are considering how to use equalizers with their DAWs for tuning their control room monitors. Some follow the standard of using either digital or analog hardware EQs for their monitors, just as they would with a mixing console. Some have taken the approach of buying control room monitors that have digital equalization built into their amplifiers. Some are assigning plug-in equalizers to the output bus and tuning the room using an analyzer, and some are considering plug-in-automated equalization systems.

In this article, I'll examine alternatives for digital equalization of your control room, and play a bit of the devil's advocate by providing examples of issues to watch out for if you rely on an automatic EQ system.

KNOW YOUR OPTIONS

At the most basic level, you need to consider the consequences of moving audio from a DAW into a digital equalizer and back out into an analog amplifier. If you use the equalizer's analog inputs, then you're looking at three conversions to get to your monitors. For my money, the smart way to go would be to come out of the DAW digital out and into the digital input of your outboard equalizer. This way, you only have one digital conversion in the chain. In either case, sampling rate should be your primary concern; aim for a minimum sampling rate of 96kHz/24-bit. Even at this sampling rate, you are restricting the signal path going to your monitors.

Some people see a digital EQ's inherent processing delay as a negative, but I really don't think this is much of a problem. The amount of delay incurred in these conversions and processing hardly amounts to much more than the effect of moving your speakers back 1.5 to 2.5 feet. This delay may even be of some benefit in reconciling video delay induced from using plasma screens.

Using digital equalizers allows more room-tuning flexibility than with analog. In most cases, you have the advantage of more bands of parametric equalization to use in the process of room tuning. Some equalizers allow you to set delays and compression, and incorporate crossovers — all in the digital domain. You also have the ability to store presets, which, for instance, let you have a curve for broadcast productions, a Dolby X-curve for film mixing or a personality curve for mixing rock 'n' roll.

Digital EQs can come as part of self-powered speakers, as stand-alone boxes or as plug-ins for DAWs. They are either controlled by the user (in which case, the room gets tuned by some person with an analyzer) or by a computer, what I will term automated tuning systems. The computer systems that I feel most comfortable with give the user some ability to tweak the automated tuning, allowing for some human decisions about the final outcome.

The pursuit of automated equalization seems to be a trend that is growing in this industry. For example, even a certain speaker company — which in the past has basically accused me of being “Satan's spawn” because I used equalizers to tune its speaker systems in studios — has come out with a self-powered speaker that also contains auto-equalization.

BEFORE YOU EQ

It is important to understand what constitutes good equalization technique before turning your process over to a computer. Equalizing speakers in a room should be the icing on the cake; it should be the last thing that happens in a studio, not the first. Seventy percent of getting a good room/speaker interaction is physically finding that critical physical placement of the speakers and the listener. Simply setting up a pair of speakers in a room and believing that you can force them into alignment with equalization is a huge mistake.

Remember that some aspects of the speaker/room interface must be treated acoustically, not with equalization. One example would be first-order reflections: The frequency response of first-order reflections mixed in with the speaker's direct signal changes as the listener moves around the console. If fixed with equalization, then a dip in one listening position may become a peak in another listening position. Equalizing bass response also requires some specific care. If you try to fill in a large hole in the bass response with equalization, you'll wind up cutting a lot of headroom out of your speaker system, resulting in distortion. Ideally, bass response should be within ±6dB acoustically before applying equalization.

Some auto-alignment systems solve the problem of equalizing first-order reflections by simply restricting the upper limits of the equalization that will be applied. This upper limit, depending on the manufacturer, seems to run from about 200 Hz to about 400 Hz. While audio purists believe you should never equalize high frequencies, this philosophy can restrict the system; for example, the system certainly cannot apply a Dolby X-curve if one is required. It is also important to know the resolution of the filters in the low frequencies. If the filter resolution is only a half- or third-octave, then you're not going to get a very accurate room tuning, no matter how sophisticated the analysis computer.

Learn which types of filters are used in the equalizer you're going to apply to your system. It is important to understand that room resonances are minimum-phase phenomena. This means that to truly correct a room resonance, you must use a second-order minimum-phase filter. A linear-phase filter is good for correcting frequency linearity inside of a speaker box or perhaps to contour an instrument or voice in a track, but will not truly correct a room/speaker problem.

Automatic EQ systems work by analyzing samples of noise played through speakers, as picked up by reference mics placed at various points in the room. Here's where the systems vary. Some systems require several microphones spread throughout the listening position. Some use just one microphone and take samples from only one position in the room. Others use one microphone but sample several positions throughout the room, while still others test in real time as the user moves a microphone throughout the listening position. While most people seem to think it's good to get a frequency response average within the mix position, there doesn't seem to be a single concept as to what the best methodology is.