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A Road Map Of Future Loudspeaker Optimization

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By Adam Shulman April 9, 2012

An understanding of the sound reinforcement system optimization process as described in my previous article (here) leads to what may be a startling conclusion: with relatively little effort, this technology could be leveraged immediately.

A number of products are currently on the market that begin to leverage this technology. These range from stand-alone beam-steering column loudspeakers from companies such as Renkus-Heinz, Tannoy and EAW to higher-power arrayable products such as Martin Audio MLA, Duran Audio Target, and Renkuz-Heinz IC Squared.

On the most basic level, the requirements for an optimized system include:

An accurate loudspeaker model. Nearly every legitimate loudspeaker manufacturer has the means of generating basic complex (in the mathematical sense) polar data. The question of accuracy arises when one looks in detail at the mid- and low-frequency parts of the spectrum, and the effect of boundaries (either in the measurement environment, or due to adjacent loudspeakers in an array).

The virtue of various measurement and calculation techniques is not within the scope of this article but generally speaking, the more accurate the model is, the closer the predicted result will be to reality.

A direct interface between the optimization program and the sound system DSP. The results of the optimization algorithm will at least initially include both mechanical (splay angle, etc.) and electronic parameters.

To successfully implement the latter, the optimization program should have direct access to the loudspeaker digital signal processing and amplification on a per-transducer basis. That is, the best performance is gained when each transducer (or at least transducer type or pass-band) can be adjusted independently.

Adding Precision

What type of systems can this apply to? In practical terms, any that satisfies these criteria! In the case of existing arrayed loudspeakers (line or otherwise), the manufacturer prediction tool, in addition to calculating optimized splay angles, would also compute different DSP parameters for each amplifier channel and download these directly into the system processing.

The possibilities for improvement will depend on the resolution of the processing (how many components are on each channel), processing power available (number of FIR filter taps), and accuracy of the loudspeaker model. Assuming that these logistical challenges are overcome, the performance advantages could be substantial, while requiring little more than a software update from the user. Upgrades could be strategically marketed, breathing new life into even aging systems (“Presenting model XX loudspeaker, now with optimization technology!”).

Frequency responses sampled on audience for uniformly driven array. Courtesy Martin Audio (click to enlarge)

Because manually gain- and equalization-shading arrays is a common practice and acknowledged to improve consistency through the audience, it is not unreasonable to expect that much more precise implementation of these techniques would produce far better results, even without any physical component change. Gone would be the days of turning down the loudspeakers at the bottom of an array, or boosting the HF for those at the top!

Some manufacturers provide a rudimentary version of this, whereby “array compensation” (i.e. mid- and low-frequency reduction to compensate for coupling) or ‘atmospheric’ correction (i.e. high-frequency boost to compensate for air attenuation) is provided based on some predetermined functions and assumptions. However, this generally does not factor in mechanical articulation or the audience area geometry. Though not a trivial effort to implement, the jump between this functionality and a truly ‘optimized’ system is not so far.

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