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 peak limiters 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 all totally transparent. Any good quality AF compressor will easily
handle 0- 20 + DB
of compression..and then maintain a constant level output. Constant level
output is what drives the
peak limiters / clippers. Drive to peak limiters is then set for whatever you
want, say 6-10 db.
## The compressor can also be configured as an AGC, with a long decay time as
you noted...and also a
slower attack time. This is normally only done when feeding program material,
which may well vary, from
one song to the next, when used for a broadcast application. The AGC will
slowly increase / decrease the level,
so the listener doesnt hear any abrupt level changes.
## 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, one 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 horse 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. The 1000- MK-V doesnt use RF clipping
when in analog SSB mode, nor in DSP SSB mode.
Jim VE7RF
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