Using a Behringer DSP8024 for Room EQ

I have a Behringer DSP8024 Ultra-Curve Pro audio processor on the output of my computer.

Behringer DSP8024 audio processor

I picked up this relatively ancient unit for £50 about 15 years ago (it cost about $500 back in 2001!), and they can still be found on eBay, along with later models like the DEQ2496, and related hardware like Focusrite’s (discontinued) VRMBox. It provides many different audio processing functions, including:

  • Stereo 31-band ⅓-octave graphic equaliser
  • Real-time stereo 31-band spectrum analyser
  • Stereo 6-band parametric equaliser
  • Delay up to 2.5 sec
  • Noise gate
  • Automatic “feedback destroyer”
  • Accurate level meter with selectable scales
  • “Brick wall” limiter for output protection
  • Automatic room equalization using microphone input and internal noise generators

It’s this last feature that is the most useful, combining the analyser with the graphic equaliser. Room equalisation (EQ) can correct a lot of acoustic deficiencies in a room. The shape, composition, and contents of a room, and non-linearities in your speakers and audio interface all contribute to how audio sounds within it. Ideally you want to minimise these effects so as to hear as true a signal as possible. It’s a good idea to apply corrective EQ after adding simple physical acoustic controls (e.g. absorber panels, diffusers, and bass traps, or just old duvets and cushions). Room EQ gets some criticism from audiophiles because it can be very hit & miss and can’t address bigger issues, but it can work very well if you listen from a single location in your room (e.g. in front of your desk).

To measure the room equalisation accurately, you need a microphone with a flat (or at least well-documented) frequency response; I use a t.bone MM-1 for this.

The t.bone MM-1 measurement microphone

The equalisation process works like this, starting from flat EQ (no alteration):

  • Output pink noise from the unit through the speakers
  • Analyse what it sounds like through the microphone, from your usual listening position
  • Alter the equalisation towards a flat response
  • Iterate over this process until the overall response is as flat as possible

This process is loud and quite unpleasant, so leave the room or stick on some closed headphones while it’s busy! It takes a minute or so, and you can hear the change in characteristic of the noise playing through the speakers, and see the changes in EQ on the screen of the unit during the process. After it’s done you can save the EQ curves, and switch the EQ in and out to A/B the config. The difference is pretty noticeable, particularly at the low end where most room-related acoustic problems tend to be; overall it’s like having a major speaker upgrade! One benefit I really notice is when switching between my corrected speakers and a decent pair of monitoring headphones – the audio really doesn’t change in character; there’s no significant tonal shift between the two.

Some people have noted problems with “digital noise” when using this unit, particularly at low volume levels. I suffered from this for a long time, but then realised what caused it and solved the problem. If you have a volume control that is before the processor, you will end up with a small signal going through the analogue to digital converters (ADCs), effectively throwing away much of their available resolution, and you’ll get a lot of quantization noise as a result. The best way to hear this deliberately is turn the input level down, and the output level up, then play something smooth and quiet. It will sound horrible, gritty and noisy, you can really “hear the bits”. This isn’t a problem unique to this unit – any ADC provided with insufficient signal will suffer the same problem.

You want to maximise the use of ADC resolution by giving it a full-range signal to convert. So if you have an audio interface before it, make sure it’s turned up full, and if you have any software level control (e.g. macOS system volume), make sure that’s turned up full too, so you’re always sending a full-volume signal. This way the converters will always use their full 24-bit resolution and the quantization noise will be so small you won’t hear it (it’s impossible to remove completely). However, you still want to control your output level. There are 2 ways to do this: alter the level on your monitors (which can be inconvenient as volume controls on active monitors are often on each speaker separately, and often hidden around the back) or use a passive volume control between the equaliser and the speakers. I use a Mackie Big Knob Passive for this.

A Mackie “Big Knob” passive volume controller.

Passive volume controls have no power supply (so no noise or extra cables), and can only turn a signal down, not up. It’s analogue, so there are no DACs or ADCs, just simple passive components. Ideally when it’s turned up full, it should be indistinguishable (acoustically speaking) from a length of cable.

Controlling level directly on the speakers (or on a separate amplifier if you have one) is possibly better than this approach, but usually less convenient. If you want to be able to run your speakers at full volume via the passive volume control, you need to have the monitors turned up full, and this often means you’ll get significant analogue noise (hiss) from the speakers when you’re listening at lower volume, however, that’s generally less unpleasant and not the problem we’re addressing here.

All of this attention to correct signal levels throughout an audio signal path is part of a wider concept known as gain staging, and occurs in many other places in audio recording, processing, mixing, mastering, etc.

It is possible to do all this processing in software using systems like REW or RoomEq, or even to go further and emulate other listening environments, famous studios or speakers, but I quite like having all this externalised and independent of software, and it also means that it can be applied to external inputs too, if you’re playing an instrument directly through a mixer. The “big knob” also provides a very convenient single control for output level, along with other features such as mute/dim and speaker and input switching.

NASA Space Sounds for EXS-24

I saw that NASA released a load of audio clips from various historic space missions – from Sputnik to the final flight of Atlantis, via the moon! Space sounds have long been used musical contexts – SpaceOddity, Telstar, Pulsar, Lemon Jelly’s “Space Walk” to name but a few. I felt I had to make these more musically useful that the ‘ringtone’ MP3s available on NASA’s site, so I wrapped them up as a library for the EXS-24 sampler (appears in Apple’s Logic and Logic Express DAWs). The sounds will work straight away in Logic, but the sounds are accessible in the archive as AIFF files so you can easily convert them to other formats. I split up the sounds into the same historical categories as on the NASA site so you’re not loading up all the samples at once. Keyboard mapping isn’t anything particular (white notes starting at C1), but I did clean up the samples a little and edited down some shorter clips of the more familiar or musical sounds (“Houston, we have a problem”, “The Eagle has landed”, “That’s one small step” etc).

The original sounds are mostly mono with low bandwidth, resolution and sample rate, but many are supplied as stereo 44.1KHz 16-bit files, so I’ve converted them all to that as EXS-24 doesn’t seem to like mixing sample rates in one instrument.

So, go ahead and download the NASA sample library! (70Mb zip)

Obviously I have no rights to these samples; NASA is encouraging people to download and use them at will, and I assume it’s being published under their open-source license.

I wrote this entry a while ago but forgot to post it, duh.

Speaker & room calibration

I was lucky enough to pick up a Behringer Ultracurve Pro DSP8024 for a mere £50 on eBay recently. It turned out to have a buggy OS version (1.2), and Behringer very kindly sent me a replacement EPROM with a new 1.3 OS on it, which works just fine. I now have it installed between my Soundcraft mixer and my Wharfedale active monitors. I used its “Auto-Q” calibration routine and put up with some quite loud pink noise to calculate a room correction curve. Because it knows the spectrum it’s generating, it assumes that what it gets back has been altered by the combination of speakers, room and microphone, so it can calculate an eq map to compensate for it. It’s quite fun to watch as it has a nice big LCD screen to display the 31 1/3 octave bands – the initial spectrum is fairly peaky, but as it iteratively applies corrections you see (and hear) it flattening out. It’s also very obvious that my monitors don’t put out much below 50Hz (it analyses down to 20Hz), but that’s to be expected from a moderately sized box with a 6.5″ driver. The results are really pretty good, sounds lovely and smooth, but the real surprise is when you’ve been listening to it for a while and you switch out the EQ – it’s really quite a shock to hear the uncorrected version. Lots of purists don’t like room correction by EQ, saying it’s better to fix the room in the first place, and also that EQ calculated like this is highly dependent on the listening position (which it is). I have a lots of bookshelves facing my speakers; they make fantastic diffusers, and I have some Universal Acoustics absorber tiles on the sloping ceiling above my listening position. The longest room mode will be fairly undamped (I’m not about to start hanging duvets around the walls!), but the resulting EQ is below 6db in either direction across the whole range – I’ve heard of rooms with 24db peaks! Anyway, after all that, it sounds lovely, and I’m happy!


I’ve had a little Behringer UB802 mixer for some time and found it very frustrating to use as a front-end to my computer system. There’s nothing particular wrong with it (clean audio, good functions, simple, reliable), it’s just not very well suited to the job, mainly because its routing is not flexible enough to be used for audio input at the same time as output. One big problem I had was that the mic I use for audio input (mainly for Skype) routed to my speakers too; a recipe for feedback and poor input level.
In order to resolve all this, I’d been contemplating a Soundcraft Compact 10 as it seemed to have much better routing options. Last week I managed to pick up the smaller Compact 4 for a mere £30 on eBay (I figured I could live without the extra size and inputs for that price). What a difference! It achieves its wonders by having an additional “recording” mix buss. It also has separate routing for monitoring. Each channel has a button that when pressed, routes that input to the recording buss and removes it from the main mix. Similarly there is a monitor button that routes it to the monitor buss. if neither are pressed, it goes to the main mix, which you can conveniently route back to the monitor mix too. Multiple mix busses are par for the course on bigger mixers, but almost nobody does it for small mixers, yet there are tons of n:2 small mixers around that are used in this role, suggesting there are a lot of frustrated users that don’t know there is a way out. As Soundcraft’s manual says:

Why don’t other manufacturers design consoles like this ?
a) Because they are out of touch ?
b) Because they are not very innovative ?
c) Because they don’t have the experience ?
d) Because they don’t listen to their users ?
Who knows 🙂

The upshot of all this is that I can route the microphone to the computer’s audio inputs without having it also appearing on the mix bus going to the monitors. As far as I can tell, this routing flexibility makes the Soundcraft about the only small mixer that’s actually designed for this role. Most seem to gloss over this routing problem, or not want to “confuse” users with the concept of an additional mix buss. As an added bonus, it has two headphone outputs that are independent of the mix output, so I can turn my speakers down without turning down my headphones. The Behringer tries to do this, but only by giving you main and monitor mix levels, but as far as I can see you never really need the main mix outputs, only the monitor mix. The only real workaround for simple n:2 mixers is to have a separate mixer for input and output, which is quite a reasonable proposition when you see the price of things like the Behringer Xenyx 502, but I’m much happier having it in one box. Behringer make bigger mixers that have more busses (I think the cheapest is the Xenyx 1222FX), but they are also bigger, more complex and more expensive – overkill for my application.
It’s also interesting to contrast the marketing. Behringer describes the 802 as having 8 inputs, which is technically true – 2 mono, 2 stereo, and a stereo return – but in reality that’s only 5 independent inputs (total of 10 input channels). Soundcraft go the other way – the Compact 4 has 5 independent inputs at a push, but you can actually squeeze 8 channels into it in total, and you can actually get 16 channels into the Compact 10. British understatement at work?
The Behringer is still a great little mixer, and I’ll miss its diminutive size, the aux send, and a couple of extra inputs (which, now I’m firmly in software-synth land, isn’t really a problem any more). Anyone want to buy my UB802?