Date: Wed, 04 Jan 2017 18:47:01 +0000
From: Manfred Mornhinweg <manfred@ludens.cl>
To: amps@contesting.com
Subject: Re: [Amps] Date: Wed, 04 Jan 2017 18:47:01 +0000
From: Manfred Mornhinweg <manfred@ludens.cl>
To: amps@contesting.com
Subject: Re: [Amps] FCC Denies Expert Linears' Request for Waiver of
15 dB Rule
> Most modern transceivers include speech processing. In the pro
> world, we use both peak limiting and compression. Peak limiting being a
> short time constant that simply reduces gain on speech peaks, and
> compression being more of a dynamic gain-riding. Good signal processing
> can sound very good with up to 10 dB of gain reduction, and some systems
> are good for more than that.
The problem with that is that 10dB of control range is far too small to
accommodate the variations in the audio level coming from the
microphone, as the operator moves closer or farther away, and speaks up
or speaks softly. And to maintain the 10dB compression you first need to
have a stable audio signal. So, in order to achieve that 10dB
compression, you need to place this compressor after an automatic gain
control system, that has a larger control range, and has such a long
decay time (1 second or longer) that it doesn't cause significant
distortion. Good speech processors usually do this.
#### Unless you talk at a constant level, and use a boom-mic, the amount
of RF clipping or RF compression will be a function of how loud you talk.
The easy
fix for all this to 1st use either a noise gate... or a downward expander. A
noise gate is
just a high ratio downward expander. Then follow it up with an audio
compressor.
Then follow that up with your limiter and or distortion cancelled audio
clipper. Add some
eq in there to tweak the audio for a specific application. The noise gate
will kill any back ground
noise when not talking..and just b4 vox drops out..and also between words etc.
Done right,
its al totally transparent.
## wonder how TV commercials are much louder than the program you are trying
to watch ?
Same basic process as described above. If you look at TV commercials on an
audio type
spectrum analyzer, you will see that the average power goes up exactly 6
db..at least in my
town. Peaks actually all drop .5 db during commercials. Good and loud..and
squeaky clean.
They are not using rvs connected diodes to make an audio clipper either.
Darlington connected
transistors used, on e pair for the negative..and another pair for the
positive. You can easily
obtain 3-15 db of peak limiting and clipping with such setups. I typ use
just 6-9 db most of the
time. Louds of clean talk power.... and either no alc, or barely any on the
xcvr.
After such an audio processor, indeed you don't strictly need ALC, as
long as the operator always sets up the TX gain in a correct way, so
that all amplifier stages are kept out of saturation. But with any band
change this gain setting will be different. Often it will also change
with frequency changes inside the same band, and what's worse, the gain
of most amplifiers changes with temperature, so the operator will have
to watch the output and readjust the TX gain rather frequently, to stay
at the optimum output level. That's quite inconvenient, and so we use
ALC to perform that task automatically.
### IF the PO of the xcvr stays put, I dont have to tweak anything, when
changing bands. Drive level to the xcvr remains constant. I bypassed the
yaesu audio mic jack completely..and instead feed the line level audio from
the
rack gear into a 20 db pad, then into a balanced to unbalanced jensen audio
transformer,
then coupled to analog BM with a 220 uf panasonic su type cap, non
polarized. Cap
has to be there, otherwise the analog BM will completely unbalance itself.
The manufacturers build ALC into the transmitters, as a
non-user-defeatable feature, because they have good reason to suspect
that most hams will not properly set the TX gain by hand all the time.
Even more so in case of radio operators in other services, who don't
have any technical knowledge at all!
### 9 vdc through a 50 K pot..then into the ALC input on the xcvr will
tweak the RF PO of the xcvr dead on. No need to worry about alc time constants
or developing ALC voltages... after the house has left the barn. You have
already
developed your own alc voltage externally with a 9 vdc source and adjustable
50k pot.
### The above 9 vdc + 50k pot method has one limitation. On a normal xcvr,
with NO
external audio compression, if you whisper into the mic, you will get hardly
any PO.
If you talk normal into the mic..or even scream into the mic, the PO of the
xcvr will be
fixed at what ever level you tweaked it for. It wont budge. If a boomset is
used, its a
non issue.
> Audio processing done entirely at baseband creates artifacts at
> baseband, but those baseband components won't get past the TX passband
> filter.
Yes. Very true. But the same is true for RF speech processing, as long
as it's done before that filter!
> W4TV has noted, however, that some rigs, notably Yaesu and ICOM,
> do part of their processing at RF, and can splatter pretty badly.
If they splatter badly, it's because of some other reason. RF speech
processing, done before a good filter, cannot create more splatter than
audio processing. A good RF speech processing scheme needs a first
sideband filter, then the clipping and compressing, and then a second
sideband filter to remove the out-of-band artifacts. The advantage of RF
speech processing, relative to audio speech processing, is that fewer
artifacts fall inside the passband. So, RF speech processing should
result in a cleaner signal, having less distortion within the passband,
and no more crud outside the passband than AF speech processing causes.
## traditional RF clipping is used in my FT-1000D..using 2 x filters.
The MK-V has a choice of either analog SSB..or DSP SSB....and
both choices use the same rf compression scheme..which is typ
a fast attack fast decay.
Now if the radios you mean happen to use low quality second sideband
filters, with slopes that aren't steep enough, and with poor stopband
rejection, then of course there will be more splatter. But that's a
problem of cheap implementation, not of the principle.
## The DX engineering RF clippers used on my old drakes used
tiny filters, think they were 4 pole units. Best version was the
rob sherwood unit, that used real 8 pole filters.
With current DSP technology pretty sophisticated and clean speech
processing can be implemented, at very low cost. Good speech processors
in DSP often shift the audio signal to some low RF (which is the same as
passing it through a balanced modulator and sideband filter in the
analog world) and then apply the compression and limiting, precisely to
move most of the artifacts out of the passband and then filter them away.
### Which xcvrs do that ? On paper it would work...provided you could
use a DSP filter to filter out the DSP artifacts. I havent see any version of
a digital DSP based
clipper, only limiters and compressors.. but that is not at low RF freqs...
just 20-20k audio.
I have dsp 6 x band audio compressors, but they are for a different
application.
When I started writing this post, I intended to question that remark -
but now, after having looked closely at all those ALC implementations, I
see your point. Let's be kind and think that amplifier manufacturers
provide ALC outputs tailored to specific radios. Obviously those
manufacturers that make the amps as accessories for their own radios
will tailor them to these radios, but I wonder what the amplifier-only
manufacturers have in mind, when defining their ALC outputs... Maybe
some specific, widely used radio? Or the radio the company's boss
happens to use?
#### You just nailed it, there is no standard, hence the incompatibility
issues.
And broadcasting is very different from ham operation, too. In
broadcasting, you set up the transmitter on one frequency, then usually
run it there, 24/7, for years. Or at least for hours, in shortwave
broadcasting. You have no frequency changes, and no thermal drifts from
circuits heating up and cooling down. In ham radio instead you change
frequency often, you also might change bands every now and then, and
your transmitter stages are all the time changing temperature, due to
the RX/TX cycling. So there is far more need for ALC than in broadcasting.
> VE7RF does extensive audio processing in his station.
### Even if the xcvr in question has overshoots, its still not an issue.
Either develop the ALC voltage externally, or limit the audio...b4 it
gets into the xcvr.
Audio processing in ham stations, using external consoles with
compressors, AGC, equalizers, etc, seems to be all the rage at present.
Some hams indeed produce excellent transmission in that way. Others not
so much. I keep hearing hams with lousy signal quality, bragging about
their studio mikes and all the audio equipment they are running to
process their audio signal. Some even add cathedral-style echo effects,
like those CB operators of 30-40 years ago! :-)
## ESSB has been going on since late 90s. Its a very tricky setup
depending on what gear is used, and how things are configured.
If I know the station on the other end is using narrow band RX, like
300-2700 hz, I have to configure entirely differently vs a station that
has a wider RX BW... like say 100-3900 hz. Wide band ssb will never
sound correct on a narrow band RX, it just doesnt work. The EQ
setup on TX is such that it wont sound right unless the RX BW at the
other end is the same BW..or wider. Even if you play with the shift control
on a narrow band width RX, it still wont sound right. You get either the
bottom,
middle, or top end, but you can never get all of it at once. When a bunch of
wider band
ops all of a sudden all get lousy audio reports from a fellow using narrow
band RX..and
also a 2 inch diam speaker built into the top lid of his icom, we have a bw
issue.
For all intensive purposes we are transmitting a mode that the narrow BW station
cant copy.
The funny thing is that so many hams use this sophisticated audio
processing on HF SSB, where HiFi audio makes little sense, due to all
the usual QRM and QRN and the intrinsic limitations of SSB. It would be
far more logical to strive for excellent audio quality on VHF and UHF in
FM, but that's something no ham in my area does.
## Not true. It works superb on HF..it also works superb on VHF.
On HF, I have not used phonetics since 2001, even on a real noisy
80m band, during the summer months..with ssb signals buried in the noise.
Several of us ran thousands of exhaustive tests on intelligibility vs BW.
The results are jaw dropping to say the least. On some tests the bottom
end was cut off at 300 hz. The top end was cut off initially at 2700 hz,
then
incremented slowly upwards in small steps to aprx 4000 hz. Slight changes in
the digital tx eq hadto be tweaked in as this is all going on.
Once u hit aprx 3600 hz at the top end, the clarity comes alive.
3600-300 = 3300 hz BW.... which is not much wider
than ur typ ham xcvr. With the top end at 3900+ hz, its razor sharp.
Its all fine as long as no QRM, and no contest is on. Its not like there is
loads of wide band ssb signals strewn across each band. They typ congregate
on one or two freqs.
Jim VE7RF
Manfred
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