Jim,
You are confusing speech processing, which is generally done at base
band (that is, on the audio signal before it is applied as modulation),
with ALC, which is control feedback between an RF amplifier and its
driver.
No, I'm not confusing them. Maybe I wasn't clear enough, by using the
term "ALC" in its generic meaning (automatic level control), rather than
specifically over the RF stages. In this sense, speech processing is a
combination of ALC (often with two different time constants) and filtering.
To avoid further misunderstanding in this regard, from here on I will
use "ALC" in the sense you use it (and radio manufacturers too), as an
automatic gain control loop sensing the RF output and acting typically
on one or a few low-level TX IF stages; and I will treat all gain
control loops at AF without using the "ALC" acronym.
> 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.
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.
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!
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.
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.
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.
ALC doesn't have to cause splatter and key clicks. It's _bad_ ALC, or
improperly used ALC, that does. Typically setting the mic gain far too
high and having the ALC throttle back the gain by 30dB or so, and
doing this on a poorly designed radio that has an ALC with a very slow
attack time.
Most amplifier manufacturers disagree with you. As long ago as 1980, Ten
Tec's manual for their Titan amp advised against using ALC, and the
manual for most power amps includes that advice. Perhaps it's the need
for that careful matching of time constants that forms part of the basis
for their advice.
Yes, I think that's the reason. Unfortunately there is no established
standard for the ALC interface between transceivers and power
amplifiers, except that ALC voltage is usually negative relative to
ground. For the ALC loop to work correctly, the entire loop needs to be
optimized. If a radio takes ALC input and then processed it with some
frequency/gain/delay curve, and the amplifier also has such a circuit
between its RF output sensor and its ALC output, we are in trouble.
Since most of the gain is inside the radio, it would be logical for an
amplifier to provide a fast, linear sample of the amplitude at its
active devices to the ALC connection, without any further processing.
The radio should do the rest, but that woud require the operator to
configure an ALC threshold in the radio. The other way around is that
this threshold is set in the amplifier, and then radios have to provide
a linear gain reduction according to ALC voltage. The problem here is
how fast a radio can react, which depends on its design.
"Fast" means that its attack speed must be fast enough to react well to
the whole significant spectrum of an SSB envelope, and that extends up
to many times the highest audio frequency. A bandwidth of 50kHz or so is
probably adequate for an ALC interface. Decay time instead has to be
slow enough to avoid significant distortion.
If the amplifier has a peak detector with a slow decay time feeding the
ALC ouput, and the radio's ALC input has its own slow decay time, then
indeed it's better not using that signal, as the loop might easily
become unstable! Again, this is a problem of improper implementation,
not of the principle.
I'm just checking the schematics of some amps. For example, the ALC
output of the Ameritron AL-811 is linear, has a fast attack, and a decay
time constant of 1ms, which is pretty slow. It's taken from the RF
output, so that it doesn't really reflect whether or not the tubes are
being driven into clipping - it depends on exactly how the amplifier was
tuned! So I would say that this ALC output is pretty useless, as it
gives no information about saturation and clipping.
The Drake L4B provides an ALC signal derived from its grounded-grid
tubes' cathodes. The signal provided is non-linear, having a threshold
at which it starts. It has a fast attack time, but its decay time
depends totally on the input impedance of the radio's ALC input, varying
from 0.3ms to infinite! Such a response is usable only with radios
designed specifically to use this signal.
The Heathkit SB200 uses a similar setup, with a threshold, but being
grid-driven it takes the sample from the tubes' grids. Its ALC decay
time constant also depends totally on the radio's ALC input impedance,
and starts from a shorter value than the L4B. It has two-stage low pass
filtering, which adds phase delay of fast signals. So its usability
again depends on the radio used to drive it.
The National NCL-2000 has a cleverly thought out system, that's derived
from the grid bias and basically starts delivering an output when the
grids are being driven a certain amount into conduction. So it does have
a threshold like the two above. Its attack is fast, it's decay is
totally dependent on the radio's ALC input impedance - it might vary
from zero to infinity! So, again, this isn't usable with every radio.
The Yaesu FL-2100 uses basically the same setup as the SB-200, just with
single stage low pass filtering and with a time constant varying from
zero to infinite depending on the radio's input impedance.
Those are the well-known amps for which I happen to have the schematics
on file, and clearly none of it has a really well implemented ALC
output. And the style implemented by Ameritron is totally different to
that of the other models cited here: The Ameritron has linear ALC
output, while the others have an internal threshold. This means that
radios that are ALC-compatible with the AL-811 will not be
ALC-compatible with the others, and vice-versa!
No wonder that hams find it hard to use that sort of ALC. But still, the
concept of overall ALC from the final amp's active devices to the
low-level RF stages is a good one - if properly implemented. We need a
standard in that regard, so that manufacturers of transceivers and
amplifiers can provide ALC inputs and outputs that are truly compatible.
> Most amp manufacturers also provide ALC output ONLY so
that its omission won't be a buying obstacle for those customers who
think they need it.
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?
Maybe I should investigate the ALC input behaviour of a few common HF
radios... Maybe in a future post...
The pot that Vic was talking about sets the MAX output level.
Yes, and that was the one I meant when I wrote that many hams around
here tweak that pot to get 150W from their 100W radios, and don't even
start to understand that this causes intense splatter and is a nuisance
to everyone else on the band.
There's a
front panel control to set the operating power. I set it for about 50W
of drive for my Titan.
Actually I do the same with my TS-450SAT and my NCL-2000. This one
requires around 30W drive for full output. The TS-450 has a
slow-attacking ALC, so it has severe power overshoot at the attacks. So
this isn't a really good solution, but I have no better one, short of
modifying the radio. I just try to avoid excessive splattering by
keeping the mic gain low enough to just start activating the radio's ALC
on voice peaks.
And this adjustment simply sets the trigger level for ALC. Nothing
else. Let's face it: SSB transmitters control the output power by
means of ALC. It's the best method found to date, as far as I know, at
least for voice transmission. Instead in digimodes, including CW, it's
better to adjust the drive to stay just below the ALC activation
level. With voice you can't really do this, because the voice level
changes too much.
I completely disagree with you, Manfred. Audio processing has been
widely used in broadcasting for at least 60 years. The AM station where
I worked as a student was using it in the early '60s. That is done
entirely at AF, NOT at RF.
But there you are talking about a very different animal: AM
broadcasting! In AM you cannot use ALC to control the peak output of a
linear amplifier string, because that would modulate the carrier level
too. Instead the normal approach is to drive the final amplifier into
saturation, and use automatic gain control for the audio signal so as to
keep the modulation at 100%, or whatever level the station uses. Indeed
this has to be done at AF, as there is no RF stage in an AM transmitter
where you could control the modulation level separately from the carrier
level! In SSB, instead, it's just as valid to apply automatic gain
control to audio stages, as it is doing so to IF or RF stages. Except
for possible dynamic range limitations in mixers and amplifier stages,
the end result will be the same. Note that here I'm referring to slowly
changing automatic gain adjustment, no faster than the syllabic rate.
Any faster changes of gain must be done before the last (or only) narrow
filter, to keep distortion products confined to the passband.
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.
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! :-)
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.
I've adjusted my K3 to roll off everything below 500 Hz and
provide about 10 dB of compression on audio peaks.
That's the same I did in my PQD-5 QRP transceiver, which I built for
backpacking and mountaineering when I was 21 years old. The compression
was rather crude, being implemented in a saturating balanced modulator!
This combination of low frequency roll-off and compression provides
about the best intelligibility per power, and that's what QRP is all
about. And it's valid too at QRO, when conditions are rough.
It's not an error, Jim! While I agree with you on the fact that many
radios have rather high phase noise and poor ALC by design, in the
case I described the phase noise was far higher than normal even for a
rather bad radio, so it must have have been a fault, not a "feature".
Yes, it IS a design error. Any rig that is broader than necessary for
the mode being transmitted is badly designed or badly operated.
I have to insist: When a radio works correctly, with acceptable phase
noise, for 20 years or so, and then suddenly the phase noise shoots
through the roof and becomes totally unacceptable, then that's a fault,
not a design error. And for all I know, that was what happened in the
particular case I described.
Quite
common with the el-cheapo "do everything from 160M to 440 MHz for $1K US
boxes,
Those are built to a price, obviously. As long as enough hams prefer to
buy radios loaded with multiple features, while not understanding their
technical specs, the manufacturers will serve this market and make their
business. They will make radios full of features, bells and whistles,
and they will make radios that try to cover DC to daylight in all modes,
and they will sacrifice performance as much as they have to in order to
meet a pre-determined price limit. And many hams will keep buying them.
Take the example of the country where I happen to live, Chile. The
average ham here is maybe one tenth as wealthy as the average ham in the
US, or in western Europe. So these hams buy only the cheapest radios
available. Add to this that radios are more expensive here than in the
US - the cheapest way to get a new radio is still to order it from a
store in the US, paying the US price plus international shipping plus
around 30% in Chilean taxes, so that your $1000 radio turns into a $1500
radio when a Chilean ham gets it. If you buy that same radio in a
Chilean store, it costs $1800. Buying a new HF transceiver is thus a big
effort for many hams here, even when it's the absolutely cheapest HF
transceiver available on the world market. That explains why the TS-120
was so popular here in 1980's, and remains popular even today, and why
the FT-450D and FT-857D are the most popular new transceivers at this
time. And Chile isn't really bad off in this regard, compared to many
other countries! You know that a country has a bad ratio between average
income and radio prices, when you notice that it has a very small ham
population and that the few hams there are mostly either foreigners or
adept homebrewers!
So we shouldn't criticize inexpensive HF radios for having marginal
specs. Only those hams who can build and sell better radios, at that
same price or lower, have a right to criticize the manufacturers who are
currently building those "cheap" radios!
but also with the high priced rigs from some mfrs.
That's indeed more objectionable. It seems that there is a significant
market for extra big and expensive radios that look very fancy, have
some added bells and whistles, but inside have the same technology, at
the same quality, as much cheaper radios. I don't like such rigs - but
as long as there is a sufficently large market, there will be companies
serving it.
At the end it's "to each his own", and that's what makes this world so
colorful and interesting!
Manfred
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