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Properly Setting Sound System Gain Structure

Analyzing each device in relation to the others -- and in relation to the entire signal path...

By Chuck McGregor May 21, 2019

Image courtesy of Alexander Stein

Editor’s note: This piece goes back a few years but it’s full of information still very much relevant today.

Realistically, audio signals at or near the noise floor of a system are not useful because the signal will not be significantly louder than the noise. Therefore, some minimum usable level must be assumed below which the electronic noise is considered objectionable.

A signal-to-noise ratio of 20 dB is considered minimally acceptable for good intelligibility. For a high-quality system, 30 dB would be a better figure to use. Using this value, the range from this minimum signal level (30 dB above the noise floor) to the clipping level is the usable signal range window for the system (also called the dynamic range in my way of thinking).

However, for purposes of this article, the maximum output to noise floor is used as the dynamic range.

Every audio system with more than one electronic component has a “system gain structure”. Gain structuring for a system occurs in the signal processing chain between the mixer or another signal source and the power amplifiers.

One usual scenario is to set all the signal processors to unity gain and turn the amplifier inputs to maximum. Unfortunately as you will see, given the different maximum outputs and noise levels of typical signal processors, this method will may not come close to the best gain structure.

We will be dealing with the signal voltage levels on the interconnecting cables from the output of the mixer (or signal source if there is no mixer) up to the input of the amplifier. For the convenience of using simple numbers, this analysis uses relative dB, as a voltage ratio where dB = 20 x log (V1/V2), and dBu, where 0 dBu = 0.775V. V1 and V2 are simply two voltages.

To set proper gain structure, the interconnections between devices must be constant voltage interfaces. This means an output device’s voltage at any point in time is unaffected by whether or not it is connected to the device(s) it is driving.

This type of interface is characterized by the output impedance of a device being 1/10 or less of its load. For example, if the output impedance is 100 Ohms, the total load it drives must be 1000 Ohms or greater. Virtually all professional audio equipment meets this criterion when a single device drives only one other device.

However when one device drives multiple devices, such as a mixer feeding a number of power amplifiers, this may not be true. In this case a distribution amplifier may be needed to divide the load between its multiple outputs.

The last thing to consider is the power handling of the loudspeaker(s). As long as the amplifier does not exceed the loudspeaker’s power handling capability and the system is operated without clipping, you should never blow a properly manufactured loudspeaker.

The safest criteria to use in selecting an amplifier is the RMS rating of the loudspeaker. In reality, most loudspeakers can handle peak signals in excess of this rating.

A reasonable choice is an amplifier whose rating that is 2 times (+3 dB) the RMS rating of the loudspeaker. The RMS sine wave used to rate amplifiers has an inherent peak power component of 3 dB.

So this all works out to a 6 dB allowance for power peaks over the loudspeaker’s RMS rating. This is a pretty safe figure for the way most professional loudspeakers are rated (pink noise with a 6 dB peak factor) and given the peak to RMS content of most audio signals.

However, sustained sine wave signals from the likes of a synthesizer could exceed the loudspeakers RMS capability by 3 dB without clipping the system. If you expect these kinds of signals and you expect to drive the system to maximum output levels with them, use the loudspeaker’s RMS rating as the power rating for the amplifier.

With these basics in mind, we’re ready to examine how to achieve proper gain structure in detail…

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