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