Although
different animals, AM-FM audio and processing techniques can be
applied to ssb
transmission to achieve some improvement in the audio quality.
There are
"hi-fi" amateur products available from Bob Heil, W2IHY, and others,
but
standard professional equipment can be had for about the same
money. I'm currently using a Shure SM-58 microphone ($99) , a
dbx microphone processor ($175), and a Dod 1/3rd octave equalizer
($99.)
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In the
pursuit of optimized ssb audio quality, I think most hams start out
looking in the wrong place. The most important factors are ssb
generation and amplification (essentially choice of rig), and optimal
drive level adjustment which I personally can't do without seeing the
rf envelope on an oscilloscope. In surfing some "hi-fi" ssb and
am websites, I see some hams have
virtual recording studios grafted onto their hamshack but no scope to
get maximum peak
output without clipping. I can only presume that they are unfamililiar
with its use. The scope is a very useful tool.
Two-tone testing
with a spectrum analyzer measures the non-linearity of the
amplification, which causes audio distortion,
but isn't practical for most hams. I can tell you
that the 870, with its dsp generation, is among the best I've seen in
this respect. The other major quality element (perhaps the one
which most people perceive as high quality) - limited
primarily by the rig is the audio frequency
response. Most
transmitters generate double sideband using a balanced
modulator, then pass only usb or lsb using a crystal filter, the width
of which determines the maximum audio bandwidth (frequency
response).
Typically this is quantified at the -3 db
points, for example at 300 hz
and 2600 hz it may be 3 db below the center response.
Although certainly adequate for basic voice intelligibility, its far
from natural
sounding. The 870, using dsp generation with front-panel variable
filtering, can pass lower than 200 hz and greater than 3000 hz at the
-3db points, which, assuming a receiver is used which has appropriately
wide IF bandwidth and audio response, makes a noticeable
improvement.
Transmitter frequency
response can be measured with an audio tone
generator (with an appropriate resistor pad reducing it to mic
level) slowly sweeping the 50hz - 5khz audio range at about
30% transmitter output
power, while noting the wattmeter. When 15% power is shown, the
-3 db point is indicated.
Once the transmitter and microphone frequency response is known and
charted, the equalizer can be set to correct the shortcoming.
Initially, maximally flat response should be the goal, beyond
that personal preferences can modify that somewhat. Maximum
intelligibility will allways be had with boosted presence (treble), but
some of the "hi-fi" guys seem to like bass boost.
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One
benefit of ssb, aside from its enhanced modulation efficiency when
compared to am, is to communicate using the smallest practical
bandwidth - the receiver hears less noise that way, and highest signal-to-noise transmission is
achieved through a noisy
medium. There are major advantages to be gained by reducing
the
dynamic range of the transmitter audio through compression and
clipping, which we'll discuss later. But when band conditions
permit good signal to noise, its remarkable what a
seemingly minor frequency response increase can do for the
"naturalness" of the audio.
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Starting
with the choice of microphone, understand that once we achieve a
relatively modest level of mic quality, there is not much more to be
gained
for ssb voice reproduction. I selected a plain professional
microphone with a dynamic (as opposed to condenser) element, a cardioid
directional pattern (to minimize off axis pickup of blowers and other
noise), and finally one that had some natural boost of the lower
treble,
or "presence" band frequencies - those which add a certain "crispness"
and are proven to enhance intelligibility. Although the
microphone's frequency response extends to 10 KHz, we only need
response to just above 3000 Hz.
Because the room acoustics of most hamshacks are terrible, instead of
working the microphone six inchs to two feet away, we'll maximize
source signal-to-noise by working it at two to six inches.
All cardioid microphones except "variable-d" models exhibit
proximity effect which increase bass response dramatically when worked
close. At two to six inches this effect is relatively modest for
our purposes. Its $99.
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Just
because a microphone looks conventional and costs just $99, don't sell
it short. Its balanced, low-impedance dynamic cardioid, and
optimized for voice intelligibility - somewhat rolled bass and a
presence peak at 3 khz - just right for ssb. If you want the
"hi-fi" sound with natural bass, you might reduce the presence peak a
little with the EQ - out of the box it will be boosted 10 db.
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Professional
microphone processors can cost more than your hamshack but there is one
that is economical and does a wonderful job. Typically this type
of unit consists of a preamp to bring mic level (-60 dbm) up to line
level (~ 0dbm), and then add certain enhancements. Dynamic range
compression is found on most units as is some form of bass/treble
(equalization) control. Another common feature is downward
expansion gate
which is sort of a subtle vox - when speech pauses, it reduces
gain slightly to lessen apparent background noise. De-esssing is
also a common process (This dynamically reduces treble when excessive
sss are present.) The dbx 286A does these very well and includes
input and output RF filtering at
$175.
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The Dod
equalizer ($99) has 31 bands of individual adjustment which are
particularly
useful for eliminating the frequencies which we cannot transmit, and
therefore do not wish to be acted upon by dynamics processing.
The dbx conveniently provides for sidechain eq
insertion. I set it to cut everything below about 100 hz, and
everything above 4000 hz. If my transmitter had -3 db point at
300
and 2600, I would boost those frequencies some to oppose the
filter rolloff, which would widen it some.
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I
alternate between the 870's internal audio processing which is actually
very good (but nowhere close to flat frequency response, as they tailor
the response to split the voice formants, processing thenm seperately)
and using just the agc with no clipping. The AP makes
for louder, denser audio, but the frequency response is
somewhat peculiar. While normally any clipping of the waveform is
taboo in pro audio circles, the reality of HF communications is that it
can gain you better apparent signal to noise through higher average
transmitted power, at the tradeoff of slight peak distortion.
With AM modulation, broadcasters generally use
asymmetrical clipping, set to brickwall downward (negative) modulation
at 95% to avoid splatter, and clipping the positive (louder!)
excursions at 125%. Most ham transmitters are pretty poor
positive modulators.
FM broadcasters, even those with formats that don't allow aggressive
processing, use some amount of clipping to prevent
overmodulation. At the time of this writing, I' currently
building an adjustable clipper as an adjunct for the dbx286A.
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No audio discussion would be complete without
opinion. The current state of audio equipment merchandising
amazes me; in no other segment of electronics can you find more
psychoacoustic doublespeak and snake oil. I estimate that
over half the money spent on "high-end" equipment is entirely
wasted on hyped, and often misleading features and
specs. A perfect example is the retro-fad use of tubes in low
level audio stages. One motherboard manufacturer even boasts about
their "tube" output amplifier for its built-in sound card output
stage. There are those who swear by their tube "preee"
(pre-amplifier). Having done proof-of-performance testing on tube
broadcast equipment in the 70's, and since then many generations of
solid-state equipment, let me bear witness to the fact that, with all
other things
equal,
modern solid-state devices excel at audio, and tubes can only come
close at
best, and are awful at
worst. One argument heard is that tubes clip less abruptly,
therefore producing distortion which is less harsh sounding; (This from
the guitar amplifier crowd.) Certainly this is true, but the argument
is ridiculous because when properly operated, one should never reach
clipping in a normal amplifier - if you do this by overdriving a stage,
then you're
intentionally
introducing distortion, the prevention of which is what our best
efforts are all about. Tubes are
microphonic, noisier, are in a constant state of deterioration, and due
to the high impedances involved, necessarily require transformers which
add low-frequency distortion. The only
advantage I'll cede is that when
the thermonuclear device detonates, indeed the tube gear may not be as
vulnerable to damage by the electromagnetic pulse. And people pay extra for these features.
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