From: "Jeff Edmonson" <[EMAIL PROTECTED]>
...the 'rack audio' SSB guys who are adding pre-emphasis, tone-tailored
audio
into their SSB rig, AFTER the Balanced modulator...
Jeff,
I'm not exactly sure what you mean. You put the audio INTO the balanced
modualtor and rf (in the form of a DSB signal) comes out. Immediately
following the balanced modulator is a bandpass filter to suppress one of the
sidebands. No matter how wide the response of the audio fed into the
balance modulator, the filter will limit the bandwidth of the SSB signal,
except for distortion products generated in the amplifiers that follow.
Maybe the "hi-fi SSB" ops are replacing their bandpass filters with wider
ones to achieve a wider audio frequency response in the SSB signal. The
problem with that is that the skirt slope of a wider filter is not as
steep, so at lower audio frequencies, they lose sideband suppression.
However, I have seen some phasing type SSB circuits using digital techniques
in the audio phase shift nework, that can result in real hi-fi SSB with good
sideband suppression down to 50 cps or so.
We need to make a distinction between "wide" signals resulting from wide
audio frequency response, versus spurious sideband products resulting from
distortion. Whether AM or SSB, most "broad" signals result from distortion
products (splatter), not the frequency response of the audio itself. I
doubt if a clean hi-fi AM or SSB signal would get much attention from Riley.
It's the guys who don't know what they are doing, and generate garbage way
beyond the normal passband of the signal that are causing the problem.
At the Dayton FCC forum, this topic came up, and both Riley and Bill Cross
seemed to indicate that the FCC was not contemplating specific bandwidth
limits, because that would hamper experimentation. They said the rules are
intentionally vague in order to allow the maximum flexibility for
experimentation. But the rules do call for "good engineering practice" and
they could use that to go after someone who repeatedly causes harmful
interference with splatter from a distorted signal.
If activity on a band is light, for example during the daytime on 75 or 160,
or on 10m when there is no skip propagation, I see no reason why a ham
shouldn't run hi-fi double-sideband AM with audio response 20-20,000 cps if
he so wishes. But it wouldn't be good amateur practice to run the same
signal when the band is heavily occupied. It's a matter of common sense and
consideration, not more restrictive FCC regulations.
On my signal, with the 3400 cps cutoff, the pre-emphasis curve with the
rising response compensates for the loss of highs. Normally, with flat
response, if the highs are cut off at 4000 cps or less, you need to cut the
bass somewhere around 200 cps, or else the audio will sound bassy. There
has to be a balance in frequency response. I have found that with the
pre-emphasis, many report my signal as "broadcast quality", completely
unaware that I am cutting off the treble at such a low frequency. The upper
midrange boost balances out the flat low frequency response.
With a "bassy" signal, there is a difference between too much bass and not
enough highs. "Tinny" audio is usually the rusult of not enough bass, not
too much treble. Many times I have heard ham signals that lacked any high
frequency response above 2000 cps, and they would get "bassy audio" reports,
so what they would do was cut the bass by reducing the values of some
coupling capacitors, and the rusult was extremely restricted audio, perhaps
600-2000 cps, and it sounded weak, unintelligible, like a tin-can telephone.
The secret is to strike a balance between the highs and lows. For good
intelligibility, the lows need to be flat down to 200 cps or below, and the
highs up to at least 3000 cps, with a proper response curve to strike a
tonal balance.
A SSB signal should be approximately 1/2 the bandwidth of an AM signal with
the same audio. The pro-SSB advocates who claim a SSB signal is 1/3 the
bandwidth of AM need to review their arithmetic: one sideband is, by
definition, one-half as wide as two sidebands. The "1/3 bandwidth" signals
can be achieved only by pinching the frequncy response of the SSB audio to
the point of compromising intelligibility. You hear plenty of SSB signals
like that. Next time, listen closely, and try to note how much of the audio
you actually hear, and how much is missing while your brain subconsciously
fills in what is left out. Of course, most ham QSO's are trivial enough
that it isn't hard to do, and in the case of DX contacts "you're 59 in ..."
the vocabulary of conversation is so limited that it is easy to guess the
missing parts, especially if phonetics are used.
We are not aware of how much information we miss even in normal
conversation. When I lived in France many years ago, I knew a girl who was
a real George Harrison fan. She had a record of one of his songs, and had
difficulty understanding the words. She asked me to write them down in
English for her. I thought it would be easy, since I had heard that song
many times and thought I understood every word. So I listened to the record
and tried to write out the lyrics word by word. I was amazed at how often
I hadn't a clue exactly what the word was, due to slurred speech or the
loudness of the instruments in the background. Casually listening to the
song, my brain subconsciously filled in the holes in the text, and left me
with the impression that I understood perfectly. We all do that every time
we listen to anyone speak. Just try to transcribe word for word what is
said in a recorded conversation or speech.
That's the only reason anyone can understand so-called "communications
quality" voice restricted to 600-2700 cps or so with a 2.1 kc/s SSB filter.
Don K4KYV
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