[TenTec] Digital Speech Processing & Pegasus

[Robert & Linda McGraw K4TAX]

While somewhat old technology today, I developed and patented a multiband
audio processor for broadcast applications many years ago.

The basic processing system consisted of a method to divide the audio
spectrum (20Hz - 25KHz) into 5 bands.  Each different band was passed
through a separate compressor/limiter.  Each used different attack and
release times for each band as the compliment of audio was different.  Then
the 5 "processed" signals were combined back into one signal.  For the 2
channel stereo version the same method was applied for each the Left and the
Right channels.  Then a mono sum signal was derived ( L+R) and a mono
difference signal (L - R) was derived.  These 2 signals were sent to the
transmitter where the (L+R) was added to the (L-R) netting a Left channel
and a Right channel signal.

The system allowed each station to "tailor" their sound and their "loudness"
by adjusting the respective levels of the 5 bands.  Made for a loud station
or a sweet sound or one with heavy bottom as one might imagine.  Just what
the competitive market demanded.

Today, on the ham rig, I use a 5 band EQ followed by a Behringer Compressor
Pro on the Paragon.  Most agree, it sounds great.

73
Bob K4TAX

----- Original Message -----
From: "Paul Christensen" <w9ac@arrl.net>
To: <tentec@contesting.com>
Sent: Tuesday, February 19, 2002 2:09 PM
Subject: Re: [TenTec] Digital Speech Processing & Pegasus


> > Langevin used to make a "distributed" clipper where the audio went
> > through a many element low pass filter and at each shunt element of the
> > filter there were clipping diodes. This rounded the corners. VOA liked
> > to use it, probably still does.
>
> Professional broadcast engineers have been using similar techniques for
more than twenty years, beginning with devices like the
> Dorrough DAP310, Inovonics 230/231-MAP-II, and later with Orban's line of
multiband Optimods...and particularly with the Gregg Labs
> multiband processors.
>
> The idea is that the audio spectrum is split via a multiband cross-over
network,  The resultant, distributed output from each
> cross-over is then directed to an independent compressor/limiter/clipper
circuit.  Typical processors contain from two to six audio
> bands. As the audio is deliberately clipped in the audio domain, it is
immediately proceeded by a low-pass filter whose sole
> function is to filter all the harmonics which are generated by the
clipping.  This process can be repeated at the final, combined
> output of all spectral bands and the audio is low-pass filtered one final
time (e.g., 15 kHz for FM stereo to protect the stereo
> pilot and stereo baseband components).  Why do all this?  Loudness, of
course.  The trick in the broadcasting world is to compress,
> limit and clip to the greatest extent possible without introducing overly
objectionable artifacts and maintain some reasonable
> degree of fidelity.  Many AM/FM broadcast and shortwave stations have
abused these audio processing systems for years in an attempt
> to garner more audience share.  But are they attracting more listeners or
turning them away from the artifacts?  That has always
> been the $64K question.
>
> -Paul, W9AC
>
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