>On 17.11.17 20.54, Peter Mairhofer wrote:
>> I am fighting this problem for two days now: I want to simulate a
>> passive mixer (50% and 25% Duty Cycle) which is basically a switch and
>> hence can be modeled by multiplying the input with a rectangular
>> waveform (or less harmonics).
>>
>> Suppose I have a clean sinusoidal LO x0 = sin(2*pi*100MHz*t).
>>
>> If I have sign(x0) (or alternatively Zero-Order-Hold with 2 samples
per
>> period) then I get a rectangular waveform containing all odd-order
>> harmonics (3rd, 5th, 7th, ...).
>>
>> Now if I do Zero-Order-Hold with 4 samples per period, theory tells me
I
>> should get rid of the 3rd harmonic, i.e. I should have 1st, 5th, 9th,
...
>>
>> Now consider this sinple simulation setup: https://snag.gy/5YpQ8X.jpg
>> with the expected time domain outputs: https://snag.gy/MRkcNi.jpg
>>
>> However, if I take the FFT of it, it can be clearly seen that BOTH
>> out0/out1 contain all the harmonics (including 3rd):
>> https://snag.gy/q7mMt0.jpg
>>
>> How is that possible?
>
>You are missing the reconstruction filter.
>Using a sample and hold function puts many rectangular shapes into your 
>wave from. They have many harmonics.

No.
Unfortunately not. Re-quoting from my original post:

> * I tried adding steep Butterworth filter (and getting rid of the
> transoients) because I thought it's because of implicit harmonic
folding
> when sampling. No change

Again, the time-domain and frequency domain plots for this case:

https://snag.gy/DvNVkd.jpg
https://snag.gy/RfOpkP.jpg

Now I also provide the waveforms:
https://www.dropbox.com/s/i8y89lvkq3i97kt/data.mat?dl=0

And the DFT code:

N1 = 15;
N2 = 25;

O0 = fft(out0(N1*1000+1:N2*1000))+eps;
O1 = fft(out1(N1*1000+1:N2*1000))+eps;
O2 = fft(out2(N1*1000+1:N2*1000))+eps;

f = linspace(0, 100e6*1000, length(O0));

plot(f, dB20([ O0 , O1 , O2 ]));


Thanks.




---------------------------------------
Posted through http://www.DSPRelated.com

>Marcel Mueller  <news.5.maazl@spamgourmet.org> wrote:
>
>>> However, if I take the FFT of it, it can be clearly seen that BOTH
>>> out0/out1 contain all the harmonics (including 3rd):
>>> https://snag.gy/q7mMt0.jpg
>
>>> How is that possible?
>
>>You are missing the reconstruction filter.
>>Using a sample and hold function puts many rectangular shapes into your

>>wave from. They have many harmonics.

As replied above: No, I added the filter:

> * I tried adding steep Butterworth filter (and getting rid of the
> transoients) because I thought it's because of implicit harmonic
folding
> when sampling. No change

I actually tried all different kinds of filters.

>Equivalently, an FFT is not a spectral estimate it is a transform.
>You need a window/filter to get a meaningful spectrum... lots of 
>discussions on computing power spectral density here in the past.

I do not even want a spectral estimate - I want the transform. Hence I
excercise much care that I take an exact even number of cycles and when
using the filter to skip initial transient.

The transform also tells me that the third coefficient has to be zero.

Using a circuit simulator (Cadence spectre) to do the same thing, the
result is as expected: Third harmonic zero. With spectre I also use the
DFT, in the same way. It only shows up if I want to simulate the same
thing in Simulink (or MATLAB using zero-order-hold upsampling).

---------------------------------------
Posted through http://www.DSPRelated.com

Marcel Mueller  <news.5.maazl@spamgourmet.org> wrote:

>> However, if I take the FFT of it, it can be clearly seen that BOTH
>> out0/out1 contain all the harmonics (including 3rd):
>> https://snag.gy/q7mMt0.jpg

>> How is that possible?

>You are missing the reconstruction filter.
>Using a sample and hold function puts many rectangular shapes into your 
>wave from. They have many harmonics.

Equivalently, an FFT is not a spectral estimate it is a transform.
You need a window/filter to get a meaningful spectrum... lots of 
discussions on computing power spectral density here in the past.

Steve

On 17.11.17 20.54, Peter Mairhofer wrote:
> I am fighting this problem for two days now: I want to simulate a
> passive mixer (50% and 25% Duty Cycle) which is basically a switch and
> hence can be modeled by multiplying the input with a rectangular
> waveform (or less harmonics).
>
> Suppose I have a clean sinusoidal LO x0 = sin(2*pi*100MHz*t).
>
> If I have sign(x0) (or alternatively Zero-Order-Hold with 2 samples per
> period) then I get a rectangular waveform containing all odd-order
> harmonics (3rd, 5th, 7th, ...).
>
> Now if I do Zero-Order-Hold with 4 samples per period, theory tells me I
> should get rid of the 3rd harmonic, i.e. I should have 1st, 5th, 9th, ...
>
> Now consider this sinple simulation setup: https://snag.gy/5YpQ8X.jpg
> with the expected time domain outputs: https://snag.gy/MRkcNi.jpg
>
> However, if I take the FFT of it, it can be clearly seen that BOTH
> out0/out1 contain all the harmonics (including 3rd):
> https://snag.gy/q7mMt0.jpg
>
> How is that possible?

You are missing the reconstruction filter.
Using a sample and hold function puts many rectangular shapes into your 
wave from. They have many harmonics.


Marcel

Hi,

I am fighting this problem for two days now: I want to simulate a
passive mixer (50% and 25% Duty Cycle) which is basically a switch and
hence can be modeled by multiplying the input with a rectangular
waveform (or less harmonics).

Suppose I have a clean sinusoidal LO x0 = sin(2*pi*100MHz*t).

If I have sign(x0) (or alternatively Zero-Order-Hold with 2 samples per
period) then I get a rectangular waveform containing all odd-order
harmonics (3rd, 5th, 7th, ...).

Now if I do Zero-Order-Hold with 4 samples per period, theory tells me I
should get rid of the 3rd harmonic, i.e. I should have 1st, 5th, 9th, ...

Now consider this sinple simulation setup: https://snag.gy/5YpQ8X.jpg
with the expected time domain outputs: https://snag.gy/MRkcNi.jpg

However, if I take the FFT of it, it can be clearly seen that BOTH
out0/out1 contain all the harmonics (including 3rd):
https://snag.gy/q7mMt0.jpg

How is that possible?

* I tried increasing the sampling ratio and number of periods. No changes
* I tried adding steep Butterworth filter (and getting rid of the
transoients) because I thought it's because of implicit harmonic folding
when sampling. No change
* I tried fixed and variable step solvers in Simulink (always sampling
the output signals 1000x higher) with arbitrary accuracy. No change.

Thanks,
Peter