[net.audio] more on speaker phasing

keithe@teklabs.UUCP (08/03/83)

	Some time ago I was playing around with my "audio system" (in
quotes 'cause you hi-buck boys would laugh at it); the speaker systems
are three-way, and at the time I was tri-amping with active crossovers
(obviously) ahead of the amplifiers.  I had made the mistake of
building 2nd order filters, and later realized the (alleged) problems
that can occur: at the crossover frequency the adjacent drivers will be
180 degrees out of phase (+90 for one and -90 for the other) - one
speaker is pushing while the other is pulling. But before I corrected
the problem (I went to third order filters) I decided to play around
with it for a while. I connected a phase-reversal switch in the
midrange circuit of one of the speaker systems and (with the other
speaker turned down/off) I played around with the effect.

	The conclusion I drew was that they sounded different, but I
could not determine which way sounded better. "Better" depended
ENTIRELY on the source material. Male vocal or female vocal wanted
different switch selections. Instrumental? Well, how do you want it to
sound? Rock sounded best if I flipped the switch back and forth as fast
as I could. (Not really.) Without the REAL LIVE SOURCE to compare to -
and where do I get one of those in my living room, anyway? - how can I
tell? (By the way - that's a rhetorical question. Please clog neither
my mail box nor the net with replies :-) )

(Sidebar follows for one paragraph):
	Some time ago I heard about a guy who had developed some
digitization/storage techniques he was making available. He had come up
with some clever ways of reducing the number of bits required to store
x amount of speech. One of the tricks is that he took parts of speech
and phase-shifted the various parts of the spectrum so that the
time-domain representation of the signal was symmetrical around some
mid-point. That way he could "record" just half of the waveform and
play it by reading out the waveform forward to the midpoint and then
"backward" to the beginning. It was still easy to recognize any given
voice - fidelity was very acceptable (remember this was demonstrated
for voice recording now).  The reason it works is because the ear is
(apparently) very tolerant of phase diffferences. What counts is time
delays, not phase delays. You can't have the flute coming out before
the tuba; your brain sez "Hey! Something' is wrong here!"

(Opinion follows - one more paragraph)
	I still say that the best thing you can do to spruce up a
modest sound system is to get a multi-band equalizer, and make the
sound come out the way you like it. It's your system, your ears - have
fun and enjoy. Don't let anyone force you into the "that's what it's
SUPPOSED to sound like" syndrome.

(And, the signature...) 
keith ericson
    or 
keithe o'teklabs 
(see the signatures debate in net.general if
you're wondering what that's all about)