[Amps] Amps Digest, Vol 169, Issue 51

Paul Gerhardt phgerhardt at gmail.com
Tue Jan 10 17:19:45 EST 2017


Ten Tec 422 Centurion Issue?  Anybody good with these?

Will not go to transmit.  Forward Power reads nearly full scale in
standby.  Green LEDs show full power all on.

Shorted RCA plug in PTT or Key will not switch it to transmit in either PTT
or QSK.

Is this a QSK board issue or something else.

Ten Tec used to sell a 5 transistor 2 flip flop rebuild kit for the QSK
board I will order the parts from DigiKey if necessary.

Any thoughts from the group?

TIA
K3PG

On Tue, Jan 10, 2017 at 12:53 PM, <amps-request at contesting.com> wrote:

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> Today's Topics:
>
>    1. Audio processing- Part 1 (Jim Thomson)
>    2. Alpha Service (David B. Ritchie)
>    3. Re: Alpha Service (Charlie Young)
>    4. Re: Alpha Service (Jim Brown)
>    5. Audio processing-Part 2 (Jim Thomson)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Tue, 10 Jan 2017 09:21:14 -0800
> From: "Jim Thomson" <jim.thom at telus.net>
> To: <amps at contesting.com>
> Subject: [Amps] Audio processing- Part 1
> Message-ID: <6AB3A7FB2EDC4F889589FE3B933F3982 at JimPC>
> Content-Type: text/plain;       charset="iso-8859-1"
>
>
>
>
> Date: Wed, 04 Jan 2017 18:47:01 +0000
> From: Manfred Mornhinweg <manfred at ludens.cl>
> To: amps at contesting.com
> Subject: Re: [Amps] Date: Wed, 04 Jan 2017 18:47:01 +0000
> From: Manfred Mornhinweg <manfred at ludens.cl>
> To: amps at 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
>
>
>
>
>
>
> ------------------------------
>
> Message: 2
> Date: Tue, 10 Jan 2017 09:19:29 -0800
> From: "David B. Ritchie" <dbritchie at gmail.com>
> To: <amps at contesting.com>
> Subject: [Amps] Alpha Service
> Message-ID: <105f01d26b65$b3cfe1a0$1b6fa4e0$@gmail.com>
> Content-Type: text/plain;       charset="us-ascii"
>
> Can anyone recommend a good way to get Alpha (or whatever they are now) to
> repair a 9500 - or alternatively recommend a third party who is good?  I
> cannot seem to get them to respond to emails.
>
> Thank you.
>
> David W6DR
>
>
>
>
>
>
>
> ------------------------------
>
> Message: 3
> Date: Tue, 10 Jan 2017 17:27:45 +0000
> From: Charlie Young <weeksmgr at hotmail.com>
> To: "David B. Ritchie" <dbritchie at gmail.com>, "amps at contesting.com"
>         <amps at contesting.com>
> Subject: Re: [Amps] Alpha Service
> Message-ID:
>         <DM5PR1201MB02176180E91755CCBFEB41D1D8670 at DM5PR1201MB0217.
> namprd12.prod.outlook.com>
>
> Content-Type: text/plain; charset="iso-8859-1"
>
> David, Brad Focken K0HM recently contacted me and said he would repair
> Alpha amplifiers.   He is one of the designers of the 9500.
>
>
> bfocken at msn.com
>
>
> 73 Charlie N8RR
>
>
> ________________________________
> From: Amps <amps-bounces at contesting.com> on behalf of David B. Ritchie <
> dbritchie at gmail.com>
> Sent: Tuesday, January 10, 2017 11:19 AM
> To: amps at contesting.com
> Subject: [Amps] Alpha Service
>
> Can anyone recommend a good way to get Alpha (or whatever they are now) to
> repair a 9500 - or alternatively recommend a third party who is good?  I
> cannot seem to get them to respond to emails.
>
> Thank you.
>
> David W6DR
>
>
>
>
>
> _______________________________________________
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>
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> http://lists.contesting.com/mailman/listinfo/amps>
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>
>
>
>
>
> ------------------------------
>
> Message: 4
> Date: Tue, 10 Jan 2017 09:50:13 -0800
> From: Jim Brown <jim at audiosystemsgroup.com>
> To: amps at contesting.com
> Subject: Re: [Amps] Alpha Service
> Message-ID:
>         <d2ad07c4-bbd7-5743-70ce-b4262ca1e51d at audiosystemsgroup.com>
> Content-Type: text/plain; charset=windows-1252; format=flowed
>
> On Tue,1/10/2017 9:19 AM, David B. Ritchie wrote:
> > Can anyone recommend a good way to get Alpha (or whatever they are now)
> to
> > repair a 9500 - or alternatively recommend a third party who is good?  I
> > cannot seem to get them to respond to emails.
> I understand that their email response is slow due to limited staffing,
> but that good people are repairing amps. Array Solutions is also
> repairing Alpha.
>
> 73, Jim K9YC
>
>
> ------------------------------
>
> Message: 5
> Date: Tue, 10 Jan 2017 09:52:46 -0800
> From: "Jim Thomson" <jim.thom at telus.net>
> To: <amps at contesting.com>
> Subject: [Amps] Audio processing-Part 2
> Message-ID: <4FEC6D4E240246F89C7C3D8CAC7FD8CF at JimPC>
> Content-Type: text/plain;       charset="iso-8859-1"
>
> manfred sez...
>
>
> 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, one for each sideband.
>
>
>
>
>
> 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  TX 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  RX
> BW station
> cant copy.
>
>  manfred sez...
>
>
> 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 most 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 had to 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.
>  You can easily
> go from Q3  to Q5.  I recorded a fellow  doing the above tests on my sony
> rack mount
> mini disc setup, and it still astounds me to this day.  Then the above
> tests  were done
> again, but   TX po reduced such that signals were right at the noise
> level.  Same deal, but
> even more pronounced effect.   Then a bunch of similar tests done with dx
> stations that
> were also essb equipped.   We tried everything from  300-2700.....all the
> way up to
> 300-5800 hz.   Intelligibility  increases every time the top end was
> increased.  Turns out we just
> re-invented the wheel, as the BW  vs  intelligibility is well  documented
> in  tech references, like
> SSB systems and circuits.... written by collins engineers.   They have
> already graphed it all out,
> with supporting documentation.   They used standardized tests, which also
> included using
> non sensical  random word groups.
>
> ##   BTW, the audio eq setup used for say a 300-3000 hz tx BW is not even
> close to what is
> used on say a 300-4100 hz setup.   Its not just the extra top end either.
> The  lower and middle portion
> is tweaked quire a bit differently.  Hence when a RX station listens to
> wide band ssb.... on his narrow band
> RX, it sounds bad.
>
> 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 on each HF band.   Typ what happens is even if I tx
> with a top end
> cut off at say 3900 hz, I will RX  with a top end of 5800 hz....so I can
> listen to both
> wide..and  wider ssb stations.   Not to harp on the subject, but some of
> us have experimented
> with audio processing on RX.   Everything from gentle downward expansion
> on RX, to  extensive
> digital eq-ing.   I also tried adding artificial harmonics to the RX
> audio, to widen the BW, then
> eq-ing the added harmonics...with great success.    I 1st tried that stunt
> when listening to a KH6
> calling cq in a contest..on 15M...just as the band was about to pack it
> in.   That was beyond an eye opener.
>
> Jim   VE7RF
>
>
>
> Manfred
>
>
> ------------------------------
>
> Subject: Digest Footer
>
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>
> ------------------------------
>
> End of Amps Digest, Vol 169, Issue 51
> *************************************
>



-- 
Paul Gerhardt
K3PG
http://pgerhardt.blogspot.com
QRP ARCI 6674
FP 274
Zombie 661
"You must do this work with love or you fail." -- John Muir, from How to
Keep Your Volkswagen Alive


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