THE FACTS ABOUT eSSB^© AND IMD

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.  John-NU9N 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 re-aligned 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 mentioned that I had seen this before when I wanted to just verify what the rig was doing, and was told that I was flat wrong.  See for yourself....  There's one more thing I'd like you to notice here.  When you look at my eSSB spectra plot made using my PC & sound card, the signal is about 4.2 kHz wide, going from the -6db point on the high side, to the equal level on the low side.  The spectrum analyzer pretty much confirms that!  It's good to see some consistency here!  The occupied BW shows 99% of all the power within 4.9 kHz.  Please notice that the analyzer I borrowed did NOT have HP's Narrow BW option.  Therefore, you must subtract the 300 Hz IF BW of the analyzer from the result, giving us 4.6 kHz.  That's pretty good as well.  Anytime you see the Occupied BW measurement, you must subtract the analyzer's IF BW from the value shown.

I also ran a test comparing my eSSB signal to a "standard" SSB signal (at left), which in the case of my rig is ~2.7 kHz wide.  You can see that the eSSB signal is wider (which is of course the idea!), but it isn't significantly more offensive than the 2.7k signal.  Both of these were done with speech.  I'd also like to show some other images for reference.  I was sent a couple links from Jim (Jan)-SM2EKM when the subject of this came up.  He has traces done on the Yaesu FT-1000 and the Collins KWM-2.  I have the same images on my site HERE and HERE, respectively.  The KWM-2 is known for being very clean, so it could be used as a very good performing reference.  The point of showing these are twofold.  One is that it's a secondary source to show that what I'm seeing is roughly what has been found elsewhere.  And two, my rig is at least on par with a different one, even in eSSB mode.  I'd like you to notice the noise level at the edges of the screen referenced to the highest level of the test signals on all of these traces.  Also, most of the images I have are done at 20 kHz, edge to edge, like Jim's are.  I do have some that were done at 25 kHz, but that's because I hadn't paid close attention to that at first.  What I want to point out is that my rig is ~60-65dB down from the peak at +/- 10 kHz, just like Jim's FT-1000 (remember, that's at max output -- I usually run much less power).  The KWM-2 is ~65-70dB down, clearly the better performer.  Also notice that he doesn't have the signal centered in the analyzer's display, so one side is a bit better than the other.  I attempted to center the signal in the display, evening out the sides.

What I set out to do was to prove or disprove some of the claims made buy a few out there that eSSB is substantially worse at producing IMD than standard SSB.  The claim was made without any real measurements (in other words, pure speculation!), so I wanted to see for myself!  It is clear that the IMD claim is being used falsely or at least blown way out of proportion, and there are much worse signals out there that AREN'T eSSB!!  Grossly overdriven rigs and amplifiers produce far more IMD than a well controlled eSSB signal.  A vast majority of eSSB users drive their rigs so that there is NO ALC DEFLECTION, into amplifiers that are capable of producing much more than legal limit, even when being driven with less than the rig's maximum power output.  In my case, I drive a Henry 3K-A which is capable of about 2.5 kW PEP output due to its hefty power supply (almost 4 kV @ nearly 1A).  This is what I was referring to earlier about only running about 80-90W -- that's all I need to get legal limit.  This is pretty typical.  Most run an 8877 or pair of 3cx800s, which do 1500W out with well under 100W drive, as well as no ALC.  So is eSSB wider than standard SSB?  Of course!  That's why it sounds better (if you have a receiver capable of listening to it all)!  Is eSSB substantially worse than standard SSB?  I don't think so.  Especially when you consider that the rigs are driven to much less than their maximum output power capabilities, and without showing any ALC deflection.  Judge for yourself.

To see all images referenced, go HERE.

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 John-NU9N'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?

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

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 lower some of the boominess in the midrange, and increase some of the high end of our speech up to the highest frequency the radio will pass.  This brings us closer to something the radio can handle.  Additionally, we can tailor it for good signal ragchewing or for "DX style" high noise, high interference conditions.  Also, we can use compression to decrease the difference between the low and high volume parts of speech.  When adjusted properly, it allows for a more consistent level to feed to the rig by reducing the peaks and increasing the valleys which means more overall talk power.  Related to this is peak limiting which keeps loud spikes from entering the rig causing clipping and distortion.  Also related is noise gating or downward expansion.  Gating simply blocks anything below a certain level from going to the next stage.  When done properly and in the right environment, it can be fairly transparent.  However, gating can chop the first few milliseconds off words or sentences and can be annoying.  Therefore, I prefer downward expansion.  Sort of opposite to compression, it takes everything below a certain level and reduces it even further.  Background noise is reduced to almost nothing and other ambient noises are also reduced in level bringing your voice literally out of the noise.

These steps so far go a long way toward making your voice clean, clear, and easily heard by the distant end.  However, doing all that can make your voice sound flat and unnatural.  There are still a couple things we can do with processing to fix this.  One is to add harmonics, the other is some type of effect to give your voice a more life-like sound.  What we're trying to do is eliminate the effect that can be heard if you've ever gone into a special sound-proof room called an anechoic chamber.  These are typically found in acoustic labs that basically eliminate any kind of room reflections so that whatever is under test is the only source of acoustic energy.  Walking into one of these and closing the door is an eerie feeling.  You hear nothing reflecting off the walls or floor and even your own breathing is only heard from inside your head.  Your speech sounds unnatural because we're used to hearing ourselves both internally, and from external room reflections.  Although that's an extreme case, there is an element of this when we've done the processing above.  Also, harmonic processing can fill in the holes (like the one I have from 4.2 to 7.5 kHz).  However, both of these must be done carefully so that the only way to tell it's there is to turn it off.  Too much room effect can be distracting and too many harmonics can sound harsh or over processed.  Either of these cases are not what we're after.  The idea is to increase our intelligibility, not reduce it!

Okay, so once all these pieces are wired together, there is the daunting task of getting everything adjusted properly.  This can be a never ending process as you become more and more critical and aware of how you sound.  This is the reason the non-eSSB'er happens across a group and hears a lot of back & forth about turn this adjustment up or that one down, etc.  Also, recording & playback are heard frequently in eSSB circles.  All that can turn the non-eSSB'er off at best or at worst for some reason prompts a few to intentionally QRM or jam the QSO.  I don't know that there's any way to fix that, but I hope that through learning and awareness, eSSB can achieve more of an acceptance.

So to answer the question at the top of this piece, it's partially because our radios aren't meant to handle the dynamic range of natural speech, and the built-in generic processing isn't really good enough to do exactly what we need at all times.  And we can go a long way towards creating much more intelligible signals when conditions are difficult (which is not necessarily "eSSB" as John-NU9N defines it, but a definite improvement over stock, one-size-fits-all audio for those conditions), as well as a much more warm, personal sound when conditions are good.  The latter is generally the "eSSB" as defined by John, where the "e" stands for "Extended" not enhanced.  He explains the differences HERE.  So my hope is to proliferate the idea that eSSB is about good audio, good Amateur practice, and above all, adds to Amateur Radio positively.

LINKS