When I was a beginner live sound engineer, I was taught that if my gain is set too high, my mics would run too hot and increase my chances of feedback. I experienced this in practice and would double-check my gain when I got feedback and it would usually fix the problem (given there weren't any physical proximity issues).

However, reading online, it seems like the vast majority of advice doesn't focus on gain adjustment but rather removing the trouble frequency using EQ. Gain structure is mentioned in one line at the end. This makes me wonder if what I was taught is correct and why it seems everyone jumps to EQ immediately.

Why isn't this advice given more often? Surely setting the correct gain is an important factor in ensuring minimal feedback? Or doesn't it matter as much as I think it does?

2 Answers 2


I'd say your basic tenet is sound, in so much as if it's too loud it will feed back - however, if there is a clearly-identifiable frequency at which the feedback occurs [assuming by that time you have no option of moving mic or speakers etc], then the next step would be to cut that frequency, allowing you to retain the overall apparent volume.

Another thing to consider is that on an old analogue desk, the gain structure could be pushed hotter & hotter through the entire signal path, so judicious use of the input gain would prevent distortion on that journey; preventing unwanted harmonics from exacerbating the tendency to feed back.

A modern rig essentially only has one component that can contribute to distortion, the ip amp itself; after that, once in the digital domain there will be brick-wall limiting to prevent clipping. So, as long as you've got headroom on the input, that should be the end of that possibility, for harmonic distortion to play a part.
On the other hand, hitting the limiters heavily would bring the noise floor & potential for feedback up with it, so it's not a free pass.

Many modern feedback reducers will actively chase potential feedback frequencies & attenuate them on the fly, using very narrow notches.
See http://www.sweetwater.com/c475--Feedback_Reduction for a whole slew of the things - some of them leave you wondering why you'd still book an engineer… unless you're an engineer ;-)

  • Thanks a lot for your response. This question came up as a result of this discussion on reddit. I have an electrical engineering/signal processing background and am trying to consolidate my understanding of what the DSP theory is with my experience as an (ameteur) live sound engineer. My current thought is that the technology has changed so my understanding of the electronics is out of date (since everything is digital). Any feedback you have on this would be appreciated. :)
    – user11426
    Dec 1, 2014 at 6:41
  • Not necessarily - the entire gain down volume up argument is sound, as it's avoiding adding THD at the ip amp. That would equally apply analog or digital. The mention of compressors etc is valid too. Nothing on that thread went against what I wrote, it's just expressed differently.
    – Tetsujin
    Dec 1, 2014 at 6:46
  • Okay, thanks. I just wanted to check that I wasn't talking complete nonsense :)
    – user11426
    Dec 1, 2014 at 6:50
  • Not at all - just that feedback has many contributing factors, so you really have to be aware of them all; gain structure, distortion, compression - that's aside from physical placement, acoustic treatment, reflective paths, resonant flooring & all the other things you may have little control over & just have to try work around.
    – Tetsujin
    Dec 1, 2014 at 6:52

The basic tenet is sound, the difference is experience. Feedback occurs when a particularly frequency of a sound is loud enough out of a sound source (such as a monitor) that it is picked up by a microphone that is feeding that sound source. This results in a loop that increases that frequency and turns it in to a constant tone. To prevent this, we must somehow break the loop by reducing the amount of sound that the microphone hears from the sound source such that the sound source doesn't make the same sound louder than the previous time.

Your gains should never be higher than you need them. The purpose of a gain is to amplify the signal to ensure that you have sufficient signal strength to move through the board without inducing a high noise floor in the signal. It has an ideal value that an experienced sound tech will set it to based on input levels.

With an inexperienced technician however, it is common to turn the gain too high. Additionally, monitors or mains may be too high for the channel. The gain is a quick and easy catch all that will fix the problem in the case where the signal level itself was set too high by a newer tech with less experience to tweak things carefully.

As you gain more experience though, you start setting gains based on the needed signal level. In these cases, you don't want to cut gains because the point of the gain has nothing to do with the output of your speakers. Instead, you want to focus on localizing and correcting the problem.

So now that we are past gain, there is a second problem that can occur, the signal can flat out be too loud in a monitor (or even worse, the main) resulting in a feedback loop. An engineer that isn't experienced with doing monitor mixes may have the monitors louder than they need to (for example, by mixing additively rather than subtractively.) This can result in unneeded feedback, but again, as the tech develops their experience, they will get better at only pushing monitors as loud as they need to.

Given that an experienced tech will already have the gain and monitor levels set where they NEED to be, we are left without the option of reducing them further. The gain and monitor levels, by default, should be at the lowest possible levels that maintain the needed strength, both from a signal quality and from a hearing safety perspective.

If you are already at your minimum level and are getting feedback, then one of two things must change. Either the mic or sound source needs to be moved to prevent the problem (or the type of mic changed), or the EQ of the channel or sound source must be altered to prevent the problematic feedback.

Provided it won't interfere with the presentation, the best bet is likely going to be to actually EQ the sound source that is causing the interference (provided it isn't actually an error in the channel EQ) as this will avoid impacting the sound of the channel outside of the problem area. (ie, the sound in the mains will be proper, even if it is off a bit in the monitor.) In some cases, this may not hold true though as the impact of that frequency being altered on other channels in the monitor may not be acceptable. Additionally, in a two board setup (where a dedicated monitor engineer has monitor specific EQs), then the channel EQ is the best place to go straight to.

Altering the EQ, whether on the channel or for the sound source, results in the particular frequency that was being looped to be reduced and the feedback cycle broken with a minimum amount of impact on the overall quality.

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