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Old June 14th 04, 02:14 PM
Hans Summers
 
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"Ashhar Farhan" wrote in message
om...
alan,

the polyphase network used by hans
(http://www.hanssummers.com/radio/polyphase/index.htm) is a very good
idea if you _have_ to use passive components. although hans has used
1% tolerance components, you can get away with even lesser tolerence
and use ordinary capacitors with 1% resistors.


Thanks Farhan. I actually used 0.1% resistors, and matched the capacitors
also to 0.1% by adding parallel capacitance. Of course, all that could
change with temperature due to the different parallel capacitances drifting
different amounts.

I believe the passive polyphase network to be superior to the active phase
shifting networks which use op-amps. For any given level of component
tolerance, a passive network will give much better opposite sideband
suppression (I recall seeing 10-20dB reported somewhere but can't provide
references). Provided attention is paid to the values used, the network can
be made lossless which overcomes any concerns about gain distribution and
harming the overall receiver dynamic range.

So the way I tend to put it, is if you _have_ to use active components ...

73 de Hans G0UPL
http://www.HansSummers.com


  #12   Report Post  
Old June 14th 04, 11:41 PM
Ken Scharf
 
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Alan Peake wrote:
Hi all,
I'm looking for the element values of the 2Q4 phase shifter as
described in the 1992 ARRL Handbook.
Alternatively, can anyone recommend an active (analog) all-pass that
would give the same or better results. I have some precision capacitors
so that's no problem.
Thanks,
Alan
VK2TWB

These days the way to do it would be to use a DSP. If this is for
a receiver the DSP would have two A/D converters and a single D/A
unit. The DSP would combine the two quadature signals and apply
bandwidth filtering. The quadature RF drive to the mixer would
come from a DDS circuit with both sine and cosine waveform
outputs (AD9853/54).

For a transmitter the DSP would have a single A/D and two D/A
stages, and would split the audio into two quadature signals
after applying bandwidth limiting and compression filtering.

Having said this, I wish I knew how to write the required DSP
software! However I'm sure there are some reading this list
with the required skill (talent?). (My software experience
lies in other areas, such as embedded controllers, NOT
math with imaginary numbers!).

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Old June 14th 04, 11:41 PM
Ken Scharf
 
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Alan Peake wrote:
Hi all,
I'm looking for the element values of the 2Q4 phase shifter as
described in the 1992 ARRL Handbook.
Alternatively, can anyone recommend an active (analog) all-pass that
would give the same or better results. I have some precision capacitors
so that's no problem.
Thanks,
Alan
VK2TWB

These days the way to do it would be to use a DSP. If this is for
a receiver the DSP would have two A/D converters and a single D/A
unit. The DSP would combine the two quadature signals and apply
bandwidth filtering. The quadature RF drive to the mixer would
come from a DDS circuit with both sine and cosine waveform
outputs (AD9853/54).

For a transmitter the DSP would have a single A/D and two D/A
stages, and would split the audio into two quadature signals
after applying bandwidth limiting and compression filtering.

Having said this, I wish I knew how to write the required DSP
software! However I'm sure there are some reading this list
with the required skill (talent?). (My software experience
lies in other areas, such as embedded controllers, NOT
math with imaginary numbers!).

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Old June 15th 04, 04:33 AM
Ashhar Farhan
 
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Alan Peake wrote:

Having said this, I wish I knew how to write the required DSP
software! However I'm sure there are some reading this list
with the required skill (talent?). (My software experience
lies in other areas, such as embedded controllers, NOT
math with imaginary numbers!).


alan,

you don't really have to know a lot of DSP to play around with this
particular beast. very simply, you collect the audio samples in a
first-in first-out buffer of about 250 slots. Each time a new sample
is added at one end, a sample is retired at the other end.

each of the 90 degree phase shift-ed samples is generated by simpy
multiplying all the samples in the pipe with a individual 'magic'
constants and adding them all up together. pretty basic stuff as far
as programming goes. the magic constants are themselves quite complex
to calculated, but that work has alread been done for you. The CD
accompanying EMRFD has those constants in a text file under the DSP
folder.

it is really simple. all the controls are soft and you can play with a
bunch of things.

if you were considering the analog route, i think polyphase approach
is simply the best : it is simple and without any tune-up and the
results are on par with the best DSP can offer.

- farhan
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Old June 15th 04, 04:33 AM
Ashhar Farhan
 
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Alan Peake wrote:

Having said this, I wish I knew how to write the required DSP
software! However I'm sure there are some reading this list
with the required skill (talent?). (My software experience
lies in other areas, such as embedded controllers, NOT
math with imaginary numbers!).


alan,

you don't really have to know a lot of DSP to play around with this
particular beast. very simply, you collect the audio samples in a
first-in first-out buffer of about 250 slots. Each time a new sample
is added at one end, a sample is retired at the other end.

each of the 90 degree phase shift-ed samples is generated by simpy
multiplying all the samples in the pipe with a individual 'magic'
constants and adding them all up together. pretty basic stuff as far
as programming goes. the magic constants are themselves quite complex
to calculated, but that work has alread been done for you. The CD
accompanying EMRFD has those constants in a text file under the DSP
folder.

it is really simple. all the controls are soft and you can play with a
bunch of things.

if you were considering the analog route, i think polyphase approach
is simply the best : it is simple and without any tune-up and the
results are on par with the best DSP can offer.

- farhan


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Old June 15th 04, 06:30 AM
Alan Peake
 
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Thanks for the suggestions. I'm reluctant to use the computer as I'm on
solar power and can't use the machine for too long, particularly on
cloudy days in winter - such as today.
I do have some DSP stuff though - ADSP2100- but as it's fixed point, I'm
not sure if it has the precision needed.
I really want to use this an exciter and while filters are probably
easier, the phasing method always struck me as more "elegant".
73s
Alan

  #17   Report Post  
Old June 15th 04, 06:30 AM
Alan Peake
 
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Thanks for the suggestions. I'm reluctant to use the computer as I'm on
solar power and can't use the machine for too long, particularly on
cloudy days in winter - such as today.
I do have some DSP stuff though - ADSP2100- but as it's fixed point, I'm
not sure if it has the precision needed.
I really want to use this an exciter and while filters are probably
easier, the phasing method always struck me as more "elegant".
73s
Alan

  #18   Report Post  
Old June 15th 04, 06:32 AM
Alan Peake
 
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Thanks Farhan. I actually used 0.1% resistors, and matched the capacitors
also to 0.1% by adding parallel capacitance. Of course, all that could
change with temperature due to the different parallel capacitances drifting
different amounts.


This may be why the 2Q4 network is enclosed in a can. The can may also
be filled with something like thermal grease to keep all the components
at the same temperature.
Alan

  #19   Report Post  
Old June 15th 04, 06:32 AM
Alan Peake
 
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Thanks Farhan. I actually used 0.1% resistors, and matched the capacitors
also to 0.1% by adding parallel capacitance. Of course, all that could
change with temperature due to the different parallel capacitances drifting
different amounts.


This may be why the 2Q4 network is enclosed in a can. The can may also
be filled with something like thermal grease to keep all the components
at the same temperature.
Alan

  #20   Report Post  
Old June 15th 04, 06:33 AM
Alan Peake
 
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The 2Q4 was an 8-pin plug-in (octal?) and this is how it is shown in
the 51SB-B sideband generator schematic.

Pin 1 - 680pF - 487k - Pin 2 - 770k||430pF - Pin 3

Pin 5 - 680pF - 125k - Pin 6 - 198k||430pF - Pin 7

Pins 1 & 5 were strapped and fed with one side of a balanced, band-
limited audio input and 3 & 7 (also strapped) with the other. Phase-
shifted outputs were then taken from 2 & 6. I guess that 4 or 8 could
have been a grounded shell.

I haven't worked it out but wouldn't be surprised if these are not
just Wein Bridge values for a certain frequency.


Cheers - Joe


Many thanks Joe - I'll put those values into my
simulator and see what comes out.
Cheers,
Alan

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