dave2010
26-06-2013, 12:33
Decades ago I read about circuits which could do frequency shifting, and their use to reduce acoustic feedback. I have been to several meetings of various amateur groups recently where acoustic feedback almost always appears as a problem - you know the sort of thing - "Can you hear me at the back?" Tap, tap, thump on the microphone. "Speak up, we can't hear you" from those at the back of the room who haven't turned their hearing aids on, or plugged into the loop system. More thumps, sometimes above the threshold of pain, then some bright spark boosts the output, and "WOW Schreech ... WeeeeeyooooowwwwwwWWWWWW ..." - absolute ear drum shattering noise as the acoustic feedback builds up rapidly through a wild crescendo.
Maybe anti-feedback circuits don't work perfectly, but I think there should be possibilities to improve things. The circuits I read about did frequency shifting, the theory being that any feedback would not be on the same frequency as the original signal. Clearly frequency shifting could be by a fixed frequency, which would probably make the source sound a bit odd, but maybe not much, or could be by frequency ratios which if done well would preserve harmonic relationships. It might also be desirable to apply adaptive filtering to the signal, as well as some careful limiting in order to give good results. My ideas here are rather speculative.
Although there could be circuits to do this sort of thing, it would probably now be possible to do everything pretty much in software, so diy effort could be spent on software design rather than circuit design.
One other approach, if the positions of the loudspeakers are known, is to put an amp plus additional loudspeaker(s) between the microphone(s) and the main speakers, and to use phase cancellation so that the microphone does not "hear" the sound from the main speakers.
Yet another approach would be to put delays in the line, or phase shift the input slightly - to tune out the main resonances, again which could be done using circuits (analogue) or simply in software.
Possibly a combination of all these techniques would be needed to come up with a really robust and easy to use solution to this problem.
Has anyone tried this kind of thing, and had any success with it?
Maybe anti-feedback circuits don't work perfectly, but I think there should be possibilities to improve things. The circuits I read about did frequency shifting, the theory being that any feedback would not be on the same frequency as the original signal. Clearly frequency shifting could be by a fixed frequency, which would probably make the source sound a bit odd, but maybe not much, or could be by frequency ratios which if done well would preserve harmonic relationships. It might also be desirable to apply adaptive filtering to the signal, as well as some careful limiting in order to give good results. My ideas here are rather speculative.
Although there could be circuits to do this sort of thing, it would probably now be possible to do everything pretty much in software, so diy effort could be spent on software design rather than circuit design.
One other approach, if the positions of the loudspeakers are known, is to put an amp plus additional loudspeaker(s) between the microphone(s) and the main speakers, and to use phase cancellation so that the microphone does not "hear" the sound from the main speakers.
Yet another approach would be to put delays in the line, or phase shift the input slightly - to tune out the main resonances, again which could be done using circuits (analogue) or simply in software.
Possibly a combination of all these techniques would be needed to come up with a really robust and easy to use solution to this problem.
Has anyone tried this kind of thing, and had any success with it?