First off, I would think that the TT final amp can take quite a bit more
peak output and still stay within its linear range. Certainly an occasional
transient peak exceeding 100 Watts by 1 dB wouldn't "break the bank" and
cause splatter. That being said, the average power output under a specific
peak output restriction could be increased by a couple of the methods you
mentioned.
Split spectrum processing is certainly one way to increase average power,
and this has been done as part of the old Alpha Vomax scheme, also it is
used in the TS-870 processing, as well as in commercial broadcasting, both
AM and FM where high average power is very much desired.
The comparisons that have been made with the Orions are usually against rigs
that are in common use that show ALC level on an analog or digital meter of
some sort. These rigs achieve their higher average to peak ratio by
fast-acting "syllabic rate" ALC time constants. I recall that Collins had a
Service Bulletin for the KWM-380 that increased ALC time constants to
increase "talk power".
Clipping with appropriate post-clipping filtering is also a useful way to
bring up the average. Setting the SP carefully on the O2 seems to bring up
the average level before distortion becomes noticeable, but you can't get
very 'deep' before it gets dirty-sounding. It's OK for DX and contests,
especially when the bass is curtailed, but not for every-day rag-chew.
The current thought in broadcasting is to use all of the above methods.
Usually a wide-band AGC, followed by 3 to 6 spectral bands of AGC, followed
by 3 to 6 spectral bands of peak limiting, followed by one or two stages of
either very fast wide band peak limiting or hard clipping or both. All of
those processes are DSP functions. That pretty well describes the
characteristics of the best Orban and Omnia brands of processors. Obviously
overkill for ham radio!
I think the problem could be fixed by changing the time constant on the ALC
when in the SSB mode. The big question that still looms is whether this is
a purely analog hardware function, or if it is controlled in any way the
firmware.
Ron N6IE
www.N6IE.com
(Formerly N6AHA)
----- Original Message -----
From: "Dr. Gerald N. Johnson" <geraldj@storm.weather.net>
To: <tentec@contesting.com>
Sent: Thursday, December 21, 2006 12:06 PM
Subject: Re: [TenTec] Orion 1 & 2 "Talk Power" problem
> On Thu, 2006-12-21 at 10:19 -0800, Ron Castro wrote:
>> I have had a couple of people contact me off-list a problem with the v.
>> 2.x Orions regarding the ability to drive external amplifiers to normal
>> average power levels in SSB. I have referred to this as the "Talk Power"
>> problem. A few weeks ago, I looked at the SSB wave form under normal
>> speech on a spectrum analyzer and found that very few voice peaks
>> approached the output level reached when measuring CW output for the same
>> power-level setting. This appears to be a result of an extremely slow
>> release time in the ALC circuit. This may be great for preventing CW
>> key-clicks, but looks like very low average power output in SSB.
>>
>> I'm wondering if the ALC is controlled by the CPU or if this is a purely
>> hardware-related function. Looking at the schematic (A5 ALC.pdf in the
>> O2 schematics), it looks to me like R9 and C5 set the timing, and if my
>> math is right, that combination (470 k and 4.7 mf) should result in an
>> 8.7 dB "release" over a period of 2.2 seconds. If the control is
>> actually digital or combined with a digital override, then the time
>> constants can be completely different. I sounds to me like very fast
>> attack and very slow release.
>>
>> Has anyone else noticed this problem and what thoughts do you have on it?
>> Also, does anyone know what parameter R-19 in the schematic adjusts?
>>
>> Ron N6IE
>> www.N6IE.com
>> (Formerly N6AHA)
>
> Average power may NOT be talk power. The fundamental problems come from
> two things: 1. the PA has a peak power limit that if you try to exceed
> that you get splatter. ALC works to prevent that over driving condition.
> Without being able to control the gains of stages after the D/A it
> should be an analog function based on PA output or PA drive. 2. Summing
> a gaggle of RF tones of different frequency, every once in a while the
> vector peak amplitude to be far taller than the RMS sums. This is true
> of voice or multi carrier cell phone base stations. That peak comes when
> all the positive or negative peaks of the tones line up. The PA has to
> have a peak rating far above the average power. But on CW or RTTY or
> PSK-31 there is only one tone or carrier and then the PA can run at full
> peak power capability, if it has enough power supply and cooling.
>
> I have thought that the relative phase responses of microphones and
> filters can be a contributor to strong or weak talk power, but I think
> now that may not be the case. Rather that voice characteristics, and
> amplitude response ripples in the microphone and filter serve to either
> make one voice frequency component stronger than all the rest so that
> the vector sums don't add to the peak so much which gives more talk
> power, or the three characteristics make there be two or three
> frequencies with about equal amplitude which is the worst case that has
> high peak to average ratio. I know that on my of my radios using an EV
> 664 gets me a higher average power on SSB. Though those in the
> neighborhood listening on 2m FM with handhelds having 3/4" diameter
> speakers grumble about the bass tones pushing the speaker beyond its
> limits. The effects the SSB filter ripple has on voice components is
> sensitive to the carrier offset also.
>
> Are there ways to improve the talk power? Maybe. One might shape the
> audio response curve to enhance one particular part of the voice
> spectrum, trying to turn it into an amplitude modulated tone with just
> one carrier frequency and then let the harmonics rattle through at
> reduced amplitude to allow better understating.
>
> One might split the audio/RF spectrum and use a separate PA for each
> frequency component so that the peak happened in the antenna, but not in
> the individual PA.
>
> One might use a fast acting ALC to limit the peaks but let the average
> power come up. Though the RF output may not have a significantly
> different characteristic than clipping the RF to limit the peaks.
>
> One might use a peak reading wattmeter and burn the average reading
> meter and accept the peak to average ratio governed by the voice,
> microphone, filter, PA and PA ALC characteristics and make contacts.
> Then it would handy if the receiver S-meter was peak reading, not
> average reading on the other end of a path.
>
> One could use a sinewave tone generator in place of the vocal chords and
> modulate the amplitude of that single tone and so eliminate most of the
> other components of the voice that contribute to the vector sum peaks. I
> suppose a DSP could do that too. It doesn't sound natural, but to the
> person without vocal chords, any sound is better than silence.
>
> I have thought that SSB might be synthesized by analyzing the audio
> waveform each half cycle, and from that half cycle picking an amplitude
> and RF frequency so the output of the transmitter had never more than
> one tone. I invented this scheme back about 1974 and wrote it down in my
> engineering notebook and had an ISU professor understand and witness it,
> but in those days it was impossible to accomplish the task with the
> available computer and synthesizer techniques. Since I've not yet tried
> it, I don't know how natural it might sound or if it would even be
> understandable at all. But it might set the peak to average ratio at RF
> about the same as at audio, IF the transients from switching synthesizer
> frequency and amplitude each half cycle of audio didn't take over.
> --
> 73, Jerry, K0CQ,
> All content copyright Dr. Gerald N. Johnson, electrical engineer
>
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