Tweaking DSP for Stage Monitors: Tips & Tricks To Maximize Potential

Some problems cannot be completely cured, but with some DSP horsepower and modern test equipment, they can be minimized

November 17, 2009, by Bruce Main

bruce dsp

In the days when digital signal processing (DSP) first stalked the arena, it was the guys at front of house that had all the fun. They would ignore their comm light for long periods of time while staring intently at asymmetrical crossovers on their laptop displays.

But now that DSP is ubquitous, the guys at the other end of the snake are beginning to experience the joys of audio in the digital domain.

The real beauty of DSP in monitorland is in optimizing wedges and side fills so that little or no equalization is needed to minimize feed back.

You might have a rack full of graphics, but the show will start out with all of them set flat. These units will only be used for on-the-fly feedback reduction or to ”personalize” a mix.

And with DSP, a preset can be designed for each model in your inventory, particularly useful if you have multiple wedge types. So let’s get on to the tweaking part.

Very few monitor cabinets are inherently “flat”. There are compromises made due to factors like box size versus low frequency driver selection, box size versus horn selection, low frequency driver dispersion versus. crossover frequency, etc.

Your favorite microphones also have frequency anomalies, especially in off axis response. Combine the two and things can get quite complicated.

Many of these problems cannot be completely cured, but with some DSP horsepower and modern test equipment, they can be minimized.

The first step is to achieve flat on-axis response from your monitors. I would highly recommend that if you don’t already own (Gold-Line) TEF, (Rational Acoustics) Smaart, or at least a real-time analyzer (RTA) then beg, borrow or steal some audio analysis equipment for this portion of the procedure.

Measured with TEF, a high-frequency device before equalization. (click to enlarge)

At an AES convention a few years ago, an equipment manufacturer that I have a relationship with set up a loudspeaker, an EQ and a pink noise source in their booth. Passersby were invited to try to equalize the pink noise by ear to achieve a flat response from the loudspeaker.

One person was spot on, but most of the results were pretty scary, even though the participants were all audio professionals. We need accuracy of +/-1 or 2 dB to give us a truly flat baseline to work from.

Some may have the ears to accomplish this, but most of us don’t. If you’re using an RTA, be sure to get one that displays in increments as small as 1 dB.

Again with TEF, the data of a high-frequency device after equalization. (click to enlarge)

Measuring the low frequency response of a loudspeaker minus the room’s acoustic contribution is a somewhat tricky proposition. A 40 Hz wavelength is approximately 28 feet long.

In order to measure those frequencies properly with a time windowed measurement system the window has to be long enough to contain at least one full wavelength.

Unfortunately that means that it is long enough to contain room reflections that contaminate the measurement. At higher frequencies this is not a problem because the wavelengths become short enough for the windowing to provide anechoic measurements.

Because real-time analyzers are time blind they include room reflections at all frequencies.

Testing Methods
There are a couple of ways to deal with this. One is to do your testing outdoors far enough from any buildings to minimize reflected energy.

My favorite approach comes from Don Keele, one of the really smart guys in our industry. He places the measurement microphone about one inch from the center dome of the woofer, takes a measurement, then places the mic one inch from the port, takes a measurement, and then sums the two responses.

The signal-to-noise ratio of the measurement is improved greatly because of inverse square law gains that result from being so close to the source.

A word of caution here: if the device under test has a maximum output of 120 dB at one meter, it’s output at one inch will approach 136 dB. This may be enough to do bad things to your expensive measurement microphone.

So start out at a fairly low volume level and work your way up. This test will give you a good idea of what is going on from 200 Hz on down.

The first parameter I set is a high-pass filter to prevent signals that the box is not capable of reproducing from wasting power and potentially damaging components.

Then correct any large EQ anomalies with parametric filters. The microphone can be moved to normal listening distance for this and all subsequent tests.

Make sure that the distance is at least three times the longest dimension of the box under test. This puts us in the far field. I do this test with no crossover engaged for the low-frequency device.

If you test from 200 Hz to, say, 5 kHz, you get a good idea of the total low frequency response curve of the box.

When you’re done, a configured system should look something like this on the DSP software. (click to enlarge)

Next, look at the upper range of the device’s response curve. There will be an obvious point where the amplitude drops off or the speaker gets into breakup modes represented on the test display by narrow notches and/or peaks.

Use parametric filters to flatten the response as much as possible within the useable range of the device. Choose an upper crossover point for the woofer that filters out the nasty modes and only utilizes the relatively flat part of the speaker response.

The high-frequency test is next. I prefer doing this test with no crossover engaged, however, the sweep frequency must be started at a high enough frequency to avoid damaging the driver.

Check the manufacturer’s recommendation for the lowest suggested crossover point and start your sweep there. If full range pink noise is being used as the test signal, start with the crossover engaged to protect the driver.

Using the parametric filters in your DSP of choice, correct for frequency response anomalies.

If the box is using a constant directivity horn you may need to use a shelving filter to increase the high frequency output above 2 to 3 kHz. Get the response as flat as you can across the full frequency spectrum.

Remember, if you leave a 3 dB peak in the response and it happens to coincide with a 3 dB peak in the vocal mic response, it will cost you 6 dB of headroom.

You should discover an overlapping frequency range where the low- and high-frequency devices are behaving in a fairly linear fashion. The crossover can be set anywhere within that region. As a general rule, if the horn is small, set the crossover towards the high end of the overlap zone.

Larger horn mouths provide pattern control down to lower frequencies, so if the horn is larger, you can set the crossover point lower while maintaining good directivity from the device.

Next, the levels and time alignment between the low- and high -requency sections should be set. With the mic on-axis and centered between the horn and woofer, do a full range sweep. Set the crossover outputs so the average volume level is the same across the entire frequency spectrum.

Then look at the frequency and phase response at the crossover frequency. Pretty ugly, eh?

Using an impulse response or ETC measurement, look at the arrival times for the two devices. Set the alignment delay on the DSP to eliminate the time arrival offset.

Now look at the frequency and phase again. Better? You will need to do a little fine tuning to get the flattest possible phase line.

If you’re using an RTA, this part is harder. Try inverting the polarity of the high output on the crossover. You should see a notch at the crossover frequency. Adjust the delay until the notch is at its deepest. Reset the high polarity back to normal.

If the time offset is correct the notch should disappear. If your RTA has a 1/12th octave mode, it will be easier to see.

Some real time analyzers have a loudspeaker timing analysis feature as well. Using asymmetrical crossover slopes can produce better (or worse) off-axis response.

Experiment with this if you have time, but this magazine isn’t long enough to cover all the possible permutations. Program in some brick wall limiters just before the little red lights on the amps start to dance.

Less From Two Than One?
Save these settings as a preset in your DSP of choice and repeat the process for each type of monitor wedge in your inventory as well as side fills and drum monitor rigs. You can also use these settings as a basis for multiple wedge setups.

But remember that when you use multiple cabinets of any sort, comb filtering will occur because of the time arrival differences. These peaks and notches are non-minimum phase. That means that they are not “EQ-able”. (Is that a word? It is now!)

Because of this, sometimes it’s possible to get less output with two wedges than with one.

But riders being what they are, go ahead and do a preset for dual wedge setups. The crossover and time alignment settings will remain the same, but you may get some summing in the low frequencies.

Use the RTA to check the frequency response because a TEF sweep will be too frightening to look at. If you need a preset for absolute maximum output with a particular vocal mic, try putting it on a stand exactly as if you were setting up for a show. Plug the mic into the test microphone input on the test rig.

The response will be a combination of the speaker under test and the off axis response of the microphone. EQ the response to be as flat as possible. It may not sound pretty, but it will get loud. (At least until the singer cups the microphone, sealing off the back of the cartridge, turning it into an omni).

Voila! A look at the final measurement result, courtesy of TEF. (click to enlarge)

You may also want to do presets for full range response or one with a higher frequency on the low-cut filter for vocal only. Sometimes it is beneficial to attenuate the lows for an acoustic set to avoid exciting acoustic guitars or pianos. Save a preset and switch back and forth at the appropriate times.

With the current crop of DSP devices, it’s not uncommon to find configurations like four input, eight output that work perfectly for either four two-way mixes or two three-way side fills and a cue wedge. Look for routing flexibility and the ability to store lots of presets.

But most of all, listen to the units. Audio quality varies as much with digital equipment as with analog.

It’s too easy to buy this type of device based on a laundry list of features and functions when the most important thing is great sound.

Bruce Main has been a systems engineer and FOH mixer on and off for more than 30 years. He has also built, owned and operated recording studios and designed and installed sound systems. Courtesy ProSoundWeb.

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