90 degree phase shifter
 

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

Hmmmm, couldn't find online info for ya, but a bunch of good bibliography
on the subject :-)

http://www.home.earthlink.net/~christrask/pshift.html

--
Gregg
*It's probably useful, even if it can't be SPICE'd*
http://geek.scorpiorising.ca

Hmmmm, couldn't find online info for ya, but a bunch of good bibliography
on the subject :-)

http://www.home.earthlink.net/~christrask/pshift.html

--
Gregg
*It's probably useful, even if it can't be SPICE'd*
http://geek.scorpiorising.ca

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.

alternatively, you can try using the approach of rick campbell in
using op-amp based all-pass network. these require fewer components
and only 0.01uf capacitors. the resistors can be easily measured on a
digital VOM for precise value and soldered into the circuit. search
the net for R2 direct conversion receiver.

unless you are planning some portable work, an exciting way to do this
is to make your computer do the audio processing work. If you feed the
left and right channels with I and Q, the rest of the receiver can be
implemented in software.

this kind of a receiver can be really simple. an rf amplifier followed
by two singly balanced diode mixers, each followed by a single stage
audio amplifier that will directly feed left and right channels of the
sound card. the rest is software. if you find dsp software to be
messy, you can probably download the freely available software from
www.flex-radio.com and use the ssb modules.

- farhan

Alan Peake wrote in message . ..
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

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.

alternatively, you can try using the approach of rick campbell in
using op-amp based all-pass network. these require fewer components
and only 0.01uf capacitors. the resistors can be easily measured on a
digital VOM for precise value and soldered into the circuit. search
the net for R2 direct conversion receiver.

unless you are planning some portable work, an exciting way to do this
is to make your computer do the audio processing work. If you feed the
left and right channels with I and Q, the rest of the receiver can be
implemented in software.

this kind of a receiver can be really simple. an rf amplifier followed
by two singly balanced diode mixers, each followed by a single stage
audio amplifier that will directly feed left and right channels of the
sound card. the rest is software. if you find dsp software to be
messy, you can probably download the freely available software from
www.flex-radio.com and use the ssb modules.

- farhan

Alan Peake wrote in message . ..
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

Hi,

I'm looking for the element values of the 2Q4 phase shifter as
described in the 1992 ARRL Handbook.

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

Hi,

I'm looking for the element values of the 2Q4 phase shifter as
described in the 1992 ARRL Handbook.

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

Pin 1 - 2 : 680pF in series with 487k

Pin 2 - 3 : 430pF in parallel with 770k

Pin 5 - 6 : 680pF in series with 125k

Pin 6 - 7 : 430pF in parallel with 198k

Inputs across 1/5 and 3/7 with the quadrature outputs from 2
and 6.

Bama website has the schematic of the B&W transmitter


73 Gary N4AST

Pin 1 - 2 : 680pF in series with 487k

Pin 2 - 3 : 430pF in parallel with 770k

Pin 5 - 6 : 680pF in series with 125k

Pin 6 - 7 : 430pF in parallel with 198k

Inputs across 1/5 and 3/7 with the quadrature outputs from 2
and 6.

Bama website has the schematic of the B&W transmitter


73 Gary N4AST

"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

"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

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!).

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!).

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

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

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

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

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

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

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

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

JGBOYLES wrote:
Pin 1 - 2 : 680pF in series with 487k

Pin 2 - 3 : 430pF in parallel with 770k

Pin 5 - 6 : 680pF in series with 125k

Pin 6 - 7 : 430pF in parallel with 198k

Inputs across 1/5 and 3/7 with the quadrature outputs from 2
and 6.

Bama website has the schematic of the B&W transmitter


73 Gary N4AST
Thanks Gary,
Alan

JGBOYLES wrote:
Pin 1 - 2 : 680pF in series with 487k

Pin 2 - 3 : 430pF in parallel with 770k

Pin 5 - 6 : 680pF in series with 125k

Pin 6 - 7 : 430pF in parallel with 198k

Inputs across 1/5 and 3/7 with the quadrature outputs from 2
and 6.

Bama website has the schematic of the B&W transmitter


73 Gary N4AST
Thanks Gary,
Alan

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.

How about doing an FFT then the inverse which IIRC gives orthogonal outputs?
Alan

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.

How about doing an FFT then the inverse which IIRC gives orthogonal outputs?
Alan

Ashhar Farhan wrote:

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.

Does this approximate the Hilbert Transform?
Alan

Ashhar Farhan wrote:

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.

Does this approximate the Hilbert Transform?
Alan

"Alan Peake" wrote in message
...


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


Alan,
Remembering back... there were two common designs. The difference
depended upon the rest of the circuit. I believe it had to do with the load
impedance presented to the network by the rest of the circuit.

--
Steve N, K,9;d, c. i My email has no u's.

"Alan Peake" wrote in message
...


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


Alan,
Remembering back... there were two common designs. The difference
depended upon the rest of the circuit. I believe it had to do with the load
impedance presented to the network by the rest of the circuit.

--
Steve N, K,9;d, c. i My email has no u's.

Alan Peake wrote in message . ..

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.

Does this approximate the Hilbert Transform?

yes it does. theoretically speaking, Hilbert transform is Finite
Impulse Response filter implmented with a specific set of
coefficients. the FIR itself is pretty simple. just an array of
incoming samples. each time a new sample is inserted, you generate a
new output by running a loop through the previous n samples.

pipe has space for n samples at a time.
HilbertTable has n number of coefficients.

for (each incoming sample)
{
add sample to the begining of the pipe, pushing out the oldest
sample from the other end;

ouputSample = 0;

for (i = 0; i n; i++)
outputSample = outputSample + (HilberTable[i] * pipe[i]);

output the sample;
}

This will give you 90 degrees phase shift.

i have written a dsp shell which will read samples from the sound card
and write them back to the sound card. you can get the source code
from http://www.phonestack.com/farhan

- farhan

Alan Peake wrote in message . ..

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.

Does this approximate the Hilbert Transform?

yes it does. theoretically speaking, Hilbert transform is Finite
Impulse Response filter implmented with a specific set of
coefficients. the FIR itself is pretty simple. just an array of
incoming samples. each time a new sample is inserted, you generate a
new output by running a loop through the previous n samples.

pipe has space for n samples at a time.
HilbertTable has n number of coefficients.

for (each incoming sample)
{
add sample to the begining of the pipe, pushing out the oldest
sample from the other end;

ouputSample = 0;

for (i = 0; i n; i++)
outputSample = outputSample + (HilberTable[i] * pipe[i]);

output the sample;
}

This will give you 90 degrees phase shift.

i have written a dsp shell which will read samples from the sound card
and write them back to the sound card. you can get the source code
from http://www.phonestack.com/farhan

- farhan

i have written a dsp shell which will read samples from the sound card
and write them back to the sound card. you can get the source code
from http://www.phonestack.com/farhan

- farhan
OK, thanks for that - I've just downloaded it.
Alan

i have written a dsp shell which will read samples from the sound card
and write them back to the sound card. you can get the source code
from http://www.phonestack.com/farhan

- farhan
OK, thanks for that - I've just downloaded it.
Alan

Alan Peake wrote in message . ..
Ashhar Farhan wrote:

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.

Does this approximate the Hilbert Transform?
Alan

It does if the "magic constants" are right. Consider
the complex frequency response of a Hilbert Transformer
and work out the impulse response. Unlike linear-phase
FIR filters, the impulse response of a Hilbert
Transformer isn't symmetric.

There are whole books on the subject. I have a copy
of _Hilbert Transforms in Signal Processing_ by Hahn,
which goes in to all of this in more than a little
detail. The challenge is turning the theoretical
impulse response (which is infinite) in to something
you can realize on finite hardware.

Laura Halliday VE7LDH "Que les nuages soient notre
Grid: CN89mg pied a terre..."
ICBM: 49 16.05 N 122 56.92 W - Hospital/Shafte

Alan Peake wrote in message . ..
Ashhar Farhan wrote:

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.

Does this approximate the Hilbert Transform?
Alan

It does if the "magic constants" are right. Consider
the complex frequency response of a Hilbert Transformer
and work out the impulse response. Unlike linear-phase
FIR filters, the impulse response of a Hilbert
Transformer isn't symmetric.

There are whole books on the subject. I have a copy
of _Hilbert Transforms in Signal Processing_ by Hahn,
which goes in to all of this in more than a little
detail. The challenge is turning the theoretical
impulse response (which is infinite) in to something
you can realize on finite hardware.

Laura Halliday VE7LDH "Que les nuages soient notre
Grid: CN89mg pied a terre..."
ICBM: 49 16.05 N 122 56.92 W - Hospital/Shafte