THE FACTS ABOUT eSSB AND IMD

In various mediums (on the air, forums, reflectors, etc.), the subject of eSSB comes up
periodically. You often see the same people posting the same tired words about how it's wasteful,
rude, sounds bad, etc., etc., etc... However, there are a few out there that claim that eSSB creates
terrible IMD Products that can be heard up and down the bands. I will not go into what odd-order
IMD products are and how they are created here -- that's beyond the scope of this. Suffice to say
that they are products created when signals are amplified, and are worse when the amplifier
(especially a non-Class A amp) is driven into compression. For our purposes, when you begin to
get ALC deflection, you are approaching that level and IMD products shoot up rapidly. Also, bear
in mind that rigs are designed differently, and ALC is implemented in different ways. Also keep in
mind that rigs can be mis-aligned with terrible results (i.e. twisting adjustments to get 150W out of
100W rigs). Using poorly designed amplifiers (like the type often found on the CB band) with
improper bias or poor bias regulation also severely degrades IMD performance. And all too
common these days, over-driving and over-compression are ways to make your neighbors on the
bands unhappy.
While it is true that a wider signal (of ANY type) is also wider IMD-wise, it can be shown that
properly done, eSSB is no more offensive on the bands than standard SSB. There are a few
things to consider: How is this measured? Under what conditions? What drive level? What output
level? There are about as many opinions about this as there are people to measure it! I borrowed
an HP 8591E spectrum analyzer for my experimentation. It is said that a picture is worth a
thousand words, and that's the beauty of a spectrum analyzer! You can infer that you are fairly
clean by using an oscilloscope, an RF sampler, and a detector, but if you really want to know
what's going on, you can't beat a spectrum analyzer! Another way is to use a second receiver
(preferably the high-quality commercial grade type) and tune off-channel. And by "off-channel", I
mean away from the occupied BW of the intended signal, but not so far away that you're beyond
the IMD products. A few kHz should do it. Also, knowing the exact BW of the receiver is
imperative. It must be selective enough to reject the main envelope, and still be able to listen the
intended BW effectively. You want to put the receiver's BW next to the signal being produced but
not overlap the BW, like THIS. Ideally, you would have things set up so the signal under test is
attenuated enough to not swamp the test receiver. If the input to the test receiver can be set so
that the S-Meter is about an S9, things should be okay. Now, it is well known that S-Meters are
quite inaccurate. The purpose is not to get absolute values, but an idea of how the transmitter is
doing. Also, it is a fair representation of a real world scenario (minus typical band noise, which
would mask much of the IMD products anyway, unless signals are extremely strong between the
transmitter & receiver) and something many Amateurs would be able to repeat themselves. JohnNU9N has done some experimentation like this (it can be found HERE). He also has an MP3 of it.
It's worth repeating, though: The BEST WAY to measure this is a spectrum analyzer!
Now to describe what I have done... I took a quick look at my rig typical
of how I operate it (about 80-90W PEP output). I thought, "Wow! This
is actually better than I thought!". Probably because of all the rhetoric
I've been hearing about this. So I decided to run the tests at the rig's
max power level, and even mis-tuned the ALC to allow the rig to put out
more power than it is supposed to be capable of. The TS-950SDX is
specified as a 150W rig, which it does easily on all bands. The final
amplifier uses a pair of MRF-150 TMOS FETs, which are actually
capable of 150W EACH!! As an example of this, the Ameritron ALS-600 uses four (4) of these for
600W PEP output. Therefore, a pair at 150W (or even more, as I've found) should be pretty
clean. I was able to get in excess of 200W out of this rig by disabling the ALC. Don't worry, I realigned it after I was finished with these tests. The image to the left is what I found as a first pass
(click it to see a larger image). It is my rig being driven to max output with white noise, with a
saved trace of eSSB overlaid. As can be seen, things don't look too bad. One thing I'd like to point
out is that a statement commonly made is "eSSB is much worse than white noise at producing
IMD". As can be seen from the test I did, that is not the case at all. White noise is worse than
eSSB. That has been found each and every time I did the comparison, regardless of power level
or ALC deflection. Here are some examples of White Noise and eSSB alone. I had previously

MY THOUGHTS ON AMATEUR RADIO AND eSSB*

I use (or have used) a variety of rigs, from "boat anchors" (vintage AM rigs) to the latest & greatest
from Icom, Kenwood, Yaesu, and Ten-Tec. As I have said above, my main HF rig at home is the
TS-950SDX. I have modified it so the receive is fairly flat from ~DC to about 6k on SSB (a BIG
thanks to WZ5Q for all the help identifying the required parts that needed to be changed!!). Of
course, I can still narrow it to below 2.5 kHz if necessary, but I rarely have the need. There is
some roll-off above 3 kHz, so it's not perfectly "flat", but it's quite good, and much better than
stock. The transmit is good from around 20 Hz to about 4.2 kHz. My processed TX audio
effectively uses approximately 40 Hz to about 4.5 kHz, as you can see from the above link. And as
with my receive, I can narrow the TX up as much as necessary. All of the above applies to SSB. I
also tinker with CW and PSK (as well as other soundcard Digi-Wigi modes), and this rig can
narrow down very tightly. I do NOT have any optional filtering installed since it comes stock with
500 Hz, 2.7 kHz, and 6 kHz filters in both the 8.83 MHz and 455 kHz IFs. I can choose whichever
filter I want independent of mode, and when both IFs have a filter of the same BW, it yields a
slightly narrower BW than each by themselves (a benefit of cascading filters). I have not found a
need for any optional filters, especially since I can adjust whatever filtering I have inline at the first
and second IFs and vary the upper and lower slopes. For maximum BW, the 8.83 filter can be
bypassed completely via the front panel, and I have installed a piece of coax across one of the 455
kHz optional filter slots to bypass that one as well. That effectively lets the DSP handle all the RX
filtering, so that the recovered audio contains as much BW as possible (again, that's approximately
0-6 kHz max on SSB) with no analog filter distortion.
If you haven't figured it out by now, I'm into high quality SSB audio also known as "Hi-Fi" SSB or
eSSB. Many feel this facet of Amateur Radio is wasteful or in extreme cases very rude or even
illegal. I can only say that who gets to decide this? The FCC's Part 97 makes no specific
reference to allowed BW for SSB or AM, so it's up to us to decide as long as there is enough
available BW open to support it. eSSB is not for everyone -- I would certainly never claim that to
be the case. Just like contesting, QRP, DX'ing, CW, Satellites, or any other mode isn't for
everyone, neither is eSSB! It most often takes up less space than AM, and definitely less space
than a dirty, overdriven 2.4 kHz SSB signal. So what's the problem, then? For some reason, this
mode has elicited literally hostile actions by more than a few people. I have been in QSO many
times and have had to put up with rude comments, intentional QRM, disgusting bodily function
noises, etc. just because of how I sound! Now how silly does that sound to you? In reality, not too
many can even receive eSSB anyway! Even if you have a rig with wide enough filtering, you have
to take a couple simple steps to hear it all. First, forget using the internal speaker, and in most
cases, even an external speaker plugged directly into the rig. Usually, the best place to pull the
receive audio is out of the accessory jack usually found on the back of the rig. Of course, you can't
drive a speaker with that, so you have to use some kind of amplifier and external speaker. I feed
mine directly to my PC then on to a dbx preamplifier which drives an Alesis 150W amplifier. It
drives a pair of modified Bose 301s and a JBL professional sub-woofer located under my desk.
Not only does this work very well for eSSB, but also for listening to MP3s or streamed radio
stations, watching movies, and gaming.
The bottom line is that many folks seem to judge something that they haven't really heard! Others
appear to not even want to hear it. I have pointed a few people to sites with lots of eSSB samples
and they have come back either claiming to not hear much difference (oh come ON!!) or, they
claim that it sounds terrible because of one thing or another. They have either made their mind up
and are unwilling to change their opinion, or are just plain deaf! I honestly believe that a lot of
people tune to an eSSB QSO with a rig that has no chance of hearing the entire transmitted signal

accurately (or it simply isn't set up to hear it), and decides that eSSB does nothing and just wastes
space. If you have decent speakers on your PC, or at least half decent headphones, go to JohnNU9N's or Mike-WZ5Q's websites and listen to the clips there. John has the clips organized by
BW, so you can hear the subtle but important differences between less than 3 kHz SSB, 6 kHz
SSB, and everything in between. Mike also has a variety of clips of all different types of sounds.
eSSB is about sounding good, learning, experimenting, and most importantly, HAVING FUN! After
all, besides bona fide emergency communications, isn't that why we do what we do?

* The term eSSB is owned and copyrighted by John Anning, NU9N.

WHY PROCESS OUR SIGNALS INTO eSSB (or at all)?

Well, that's a pretty big can of worms, isn't it? A good place to start is understanding how speech
works, how available bandwidth affects it, and what happens when the listener hears it. A Google
search on "Speech Intelligibility" turns up many hits. Another good source of information is the
Polycom Whitepaper on the subject. Here are a few more:
http://www.assta.org/sst/2004/proceedings/papers/sst2004-103.pdf
http://www.utdallas.edu/~loizou/thesis/kalyan_ms_thesis.pdf
http://www.meyersound.com/support/papers/speech/section2.htm
What's important to understand is that the mantra "SSB at 2.4kHz (or less) has been good enough
for 50 years, why change?" is what a lot of the controversy is about. I think the above references
answer that question in great detail, but the short answer is that very narrow bandwidth causes
holes in our speech when it contains energy above or below the transmission equipment's
bandwidth limits. The high frequency parts are important for clarity and articulation. They basically
allow us to determine the difference between the hard consonants within words. When they are
missing, our brains engage and attempt to fill in the missing parts based on context. The lower the
upper bandwidth limit is in frequency, the more ambiguities there are and the more work our brains
have to do. In the simplest forms, it really isn't too much of an issue because casual contacts
heard on the Amateur Bands in a typical day just don't require all that much accuracy, except for
maybe call signs which are usually given phonetically, anyway. Let's face it, it's easy to tell the
difference between Yaesu, Icom, Kenwood, or Ten-Tec and Vertical, Yagi, or Dipole, etc. You get
the idea. The low end adds warmth and personality to our voices, and while not really critical to
understanding, adds comfort and realism.
Long periods of ragchewing can be fatiguing because of the totally random subjects and the
duration. That's where eSSB really shines! If you listen to my audio clip, you can easily tell the
difference between the P in "PR-30" and T in "TS-950". Also, the S in "SDX" is clearly an S and
not an F. Try that with a 300-2.4k band pass and you'll hear a lot of missing parts. My old call was
"KS4KF" and I can tell you first hand, "normal" bandwidths just don't contain enough information to
determine what I said. Obviously, calls are basically random groups of letters and numbers, and
since there is no valid context, most folks heard it as "KS4KS" or "KF4KF" -- I have heard both.
Probably because our brains like repetition. This is clearly why phonetics exist. But we don't talk
that way in daily person to person speech, so it's unnatural to have to do that constantly (especially
with THAT call!!).
Another phenomenon is our natural speech rolls off as frequency increases. On the Amateur HF
Bands, it's easy for those parts of speech to decrease below the ambient noise levels if nothing is
done to compensate. I made a spectral plot of my voice into a flat studio condenser mic. Look at
it HERE. What you see there is a lot of energy in the low frequency area up to about 700 Hz.
After that, it sort of stair-steps down until you get to 10 kHz then it rapidly rolls off. An interesting
thing is happening with my voice around 5.8 kHz -- there is a fairly deep "V" starting around 4 kHz,
coming back up around 7.5 kHz. I have seen other natural voice plots, and most have something
similar, but are different in each individual case. What that tells me is that even if I had a rig
capable of passing frequencies up to 6 kHz, it would do me little good even with an EQ. Granted,
in very good signal conditions it would be possible to hear differences, but normally I would think
the difference between 4 and 6 kHz transmitters in my case would be nil. These speech dynamics
(frequency and level variations) are very difficult if not impossible for the electronics in our radios to
handle accurately. Enter voice processing.
There are several things that need to be done to the natural voice to make it more "friendly" to our
radios and the medium (HF spectrum in this case) in which they operate. There is some of that
done already in terms of the microphones, internal mic preamps, and other circuits in the rigs.
However, it is rarely if ever perfectly tuned to each of our voices. Like I mentioned above, our
natural voices have different characteristics, and while the rigs certainly work as they are, they are
likely not anywhere near optimal. The first thing that we can do is equalize or "EQ" our voices to