Are filters really needed for full range louspeakers?

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Depends.

Certainly a hi-pass so that helper woofers can be used (ie creating a FAST) can bring significant performance improvements.

Adding a hi-pass so that a single driver gets cut off from bass input below its cut-off frequency will help its performance. This needs to be done before the amplifier because passive, speaker level circuits rarely work because of the impedance rise at resonance.

dave
 
Hi 'rosenkrux',
personally, I am enjoying a pair of home made speakers that run without any filters or crossovers or equalisation and I find them to be very, very nice to listen to. It is a back to basics experiment to see what commonly used speaker design and measurement software can produce before modifying with the sort of filters etc one is 'supposed' to use.
I'll not be adding anything at all although though, well maybe just a tiny lift on the bass usng a pre-amp depending on speaker positioning, which I find to be one 'the' most important things to get right before you do anything else.
 
It's true that if you want to listen to high spl's you might want to protect your driver with a high pass filter but if you're content with normal domestic levels within modest rooms then if you are careful with both speaker postion 'and' listening position you'll be surprised what you can get away with as one can make the most of room gain by doing so.
 
hi gmad - are there linear phase highpass plug-in for media players ? how do you accomplish this in your computer system?

You can download a nice Linkwitz-Riley linear phase filter from this page:

Christian's private site - EQs & Filters - Christian's private site

You would need to be able to run VST plugins to use it. Since I can use multiple convolvers in foobar2000, I do all my filtering that way. I actually use foobar to create an impulse response (.wav file) of a filter by running a single sample spike through it and capturing the output.
 
thank you very much - hope I can figure it out. Most of my larger coax and woofer based speakers/toys will handle 200 watt peaks on percussion with ~1/8" or so peak to peak excursion and are tuned high enough where the lower Z peak cuts power input below cutoff - but for other stuff I'd like to highpass in a player. I've got an I3 build to use - its bulky like a breadmaker in a "Steamcastle" case
 
Then there are BSC filters often implemented on small / narrow baffle FR driver systems.

Thanks to various DSP methods, this can be accomplished ahead of amps when deemed necessary. And while any type of filter might "spoil" the sound in some purists' perspective, it beats having a driver ruined by excessive excursion or even DC from defective amps.
 
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hi gmad - are there linear phase highpass plug-in for media players ? how do you accomplish this in your computer system?
Linear phase even order filters are easily made. Use a good Editor like CoolEdit ('Adobe Audition' these days). Make a piece of silence (1 second or so), and set the center sample to 1 (make a Dirac pulse). Filter with one of the 'scientific filter' highpasses like Butterworth, any order from 1..3. Then time reverse the output and filter again, the result will a linear phase filter. When using Butterworth this will result in a Linkwitz-Riley (2nd 4th or 6th order) characterstic. Trim the excess head and tail of this impulse below -80 or -100dB or so, then fade-in/fade-out a section of the remaining head and tail to avoid sharp edges (those would give audible pre- and post echoes if to large). Save and load into a convolver plugin (like the one supplied with FooBar).
 
Linear phase even order filters are easily made. Use a good Editor like CoolEdit ('Adobe Audition' these days). Make a piece of silence (1 second or so), and set the center sample to 1 (make a Dirac pulse). Filter with one of the 'scientific filter' highpasses like Butterworth, any order from 1..3. Then time reverse the output and filter again, the result will a linear phase filter. When using Butterworth this will result in a Linkwitz-Riley (2nd 4th or 6th order) characterstic. Trim the excess head and tail of this impulse below -80 or -100dB or so, then fade-in/fade-out a section of the remaining head and tail to avoid sharp edges (those would give audible pre- and post echoes if to large). Save and load into a convolver plugin (like the one supplied with FooBar).

Yes, this is a good approach. I made a 96db/octave subsonic filter this way inside of Audacity. It's a long filter, but the sound is transparent.
 
i run a ported alpair 10M speaker without any filters in it for about 1 year and a half, often with bass heavy music (like dub, dubstep, UK garage, grime, ...) and never had any damage or distorted sound. It's amped with an marantz ss amp of 45 watts butt never more than half open, and often even only a quarter open. That's loud enough for my living room...

i've run it with a double sub (a scanspeak 10" in a closed enclosure) of a friend for a little homeparty (20ppl in his living room) powered by 2 60 watt PP valve monoblock amps and a passive 2nd order corssover at 95hz with an very expensive studio limiter as protection, and it was loud enough for that (probally arround 100dB rms in total) with probally the best sound i ever had on a party. But that was the only time i used a hpf on it.
 
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