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filtering 40 kHz signal

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BkraM

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Hi All,

I'm working on a ultrasonic 40 kHz sonar.
The 40 kHz signal is generated from an NE602 mixer that mixes a 10 and 50 kHz signal.
the output of this (unbalanced) mixer contains many multipals that I'd like to filter out.

I'd like to have a bandpass filter for 38 to 42 kHz that gives -20 dB at 30 and 50 kHz.
After some reading into filters, I'm starting to believe that an analog filter with this performance will start to be quite complex (12th order).
Is there another way to achieve this result?

THanks,
 
'Switched capacitor filter' perhaps?
Why not generate the 40kHz directly?
 
I've taken the sensors from a HR-S04 module ;)
I'm building an acoustic imaging prototype from them (synthetic aperture sonar)
the system is working, but needs some signal conditioning to improve image quality.
 
You seem to be making this VERY over-complicated :D
I've taken the sensors from a HR-S04 module ;)
I'm building an acoustic imaging prototype from them (synthetic aperture sonar)
the system is working, but needs some signal conditioning to improve image quality.

As others have mentioned, it seems bizarre not generating the 40KHz directly - you're making it far too complicated by not doing so.
 
I'm not generating the 40kHz directly as I need a chirp signal from my pc
Do you have a broadband transducer? From what I read, most 40kHz transducers have a poor response away from their resonant frequency.
 
transducers indeed have a very poor bandwidth, 4 kHz is about all I can squeeze out of them (in the roll off band).
Should be enough to give a resolution of a couple of cm.
 
I'm not generating the 40kHz directly as I need a chirp signal from my pc, which has a too low sample rate

Synthetic aperture systems need precise phase control. Are you trying to do this on a Windows PC? If so, I recommend an external driver system that gets parameters from the PC but drives the transducers directly.

ak
 
The transducers are driven by a separate amp, after the filtering stage.
The received signal is downconverted again to a 10 kHz signal and logged by the windows PC at 96 kHz
 
If the carrer for the down conversion mixer is not a true sinewave, you might not be gaining much versus direct A/D on the 40 KHz signal. Also, all sound cards have AC coupled inputs. Is this ok for the signal parameters you want to capture?

ak
 
the LO signal is (as good as it gets) a true sine wave.
Why would i loose detail if i used a square wave? If i filter out the harmonics, the information in the signal would be the same?

I'm not sure what you mean by AC coupled inputs

(I'm on a learning curve, so any suggestions are welcome ;))

Thanks,
 
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