Pre-cancelling mixer products (DAC sample selection)

H

Hobbyist

Guest
I'm working on a hobby project where I need to get a "phase locked"
pair of sinewaves (25Mhz and 27Mhz) from point A to point B. I've
considered several approaches and I've concluded that the most
appropriate way is going to be "analog modulating" an LED with a
signal generated by a DAC. For a first test I'm thinking 1350 samples
(i.e. 25 * 27 * 2) fed to a 54 MSPS converter. I'm picturing a modest
bandpass filter (26Mhz +/- 1Mhz) after the DAC but basicly the DAC
output will directly modulate the current flow through the transmitter
LED.

My problem comes in generating the sample values. I generated a plot
of a simple sin25MHz( t ) + sin27Mhz( t ) for each sample point. The
resulting "mixer products" were obvious. My hobby level books
(magazines mostly) say they are (f1 + f2), (f1 - f2), (f1 + 2f2), etc.
My first question is: Are these "mixer products" new information which
can be removed from my DAC output or are they inherently required if
I'm going to want to recover the individual source signals at the
receiver?

If I can remove the "mixer products", then I was thinking it would be
a lot easier if I didn't generate them in the first place. I thought
an easy way to do this (for the first few "influential" products at
least) would be to add additional "compliment" waves to the initial
suming process so that each product would be "cancelled" out of the
sample stream. Does this make sense? Could anyone provide (or direct
me to) a simple description of how to determine the correct phase and
amplitude of the "compliment" wave for each "mixer product"? (It's not
at all clear from my initial plot)

Thanks for any and all comments or suggestions,
Hobbyist
 
Wow, you are getting pretty involved for a hobyest. First of all you left the
2 pi out of the sin wave descripton, it should read sin25MHz*2*pi (t) +
sin27MHz*2*pi (t) the resulting spectrum will be (f1 + f2), (f1 - f2) the
additional harmonics only occur with non sinosidioal wave forms e.g. square
wave.
No you cannot remove both these mixer products and still expect to be able to
recover the original waveform.


"Hobbyist" <no@thank.you> wrote in message
news:8cnnqvccn0fq552oc48vofgg2ucefl1cmq@4ax.com...
I'm working on a hobby project where I need to get a "phase locked"
pair of sinewaves (25Mhz and 27Mhz) from point A to point B. I've
considered several approaches and I've concluded that the most
appropriate way is going to be "analog modulating" an LED with a
signal generated by a DAC. For a first test I'm thinking 1350 samples
(i.e. 25 * 27 * 2) fed to a 54 MSPS converter. I'm picturing a modest
bandpass filter (26Mhz +/- 1Mhz) after the DAC but basicly the DAC
output will directly modulate the current flow through the transmitter
LED.

My problem comes in generating the sample values. I generated a plot
of a simple sin25MHz( t ) + sin27Mhz( t ) for each sample point. The
resulting "mixer products" were obvious. My hobby level books
(magazines mostly) say they are (f1 + f2), (f1 - f2), (f1 + 2f2), etc.
My first question is: Are these "mixer products" new information which
can be removed from my DAC output or are they inherently required if
I'm going to want to recover the individual source signals at the
receiver?

If I can remove the "mixer products", then I was thinking it would be
a lot easier if I didn't generate them in the first place. I thought
an easy way to do this (for the first few "influential" products at
least) would be to add additional "compliment" waves to the initial
suming process so that each product would be "cancelled" out of the
sample stream. Does this make sense? Could anyone provide (or direct
me to) a simple description of how to determine the correct phase and
amplitude of the "compliment" wave for each "mixer product"? (It's not
at all clear from my initial plot)

Thanks for any and all comments or suggestions,
Hobbyist
 

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