Down sampling, bandpass sampling theorem, downconversion

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SUMMARY

The discussion centers on demodulating AM signals using an ADC and the bandpass sampling theorem. The theorem states that for a bandpass signal with frequency f0 and bandwidth B, a sampling frequency greater than 2B is sufficient for signal reproduction, even if the sampling frequency is less than f0. It is confirmed that the ADC's bandwidth must be at least f0 to prevent signal attenuation. Additionally, the discussion highlights the importance of understanding the sampling aperture in relation to the instantaneous nature of the delta function used in theoretical formulations.

PREREQUISITES
  • Understanding of bandpass sampling theorem
  • Familiarity with analog-to-digital converters (ADCs)
  • Knowledge of signal processing concepts, particularly in demodulation
  • Basic principles of Digital Signal Processing (DSP)
NEXT STEPS
  • Study the implications of the bandpass sampling theorem in practical applications
  • Learn about the design and specifications of ADCs, focusing on bandwidth requirements
  • Explore Digital Signal Processing (DSP) textbooks for deeper insights into sampling and filtration
  • Investigate techniques for improving signal-to-noise ratio (SNR) in bandpass sampling
USEFUL FOR

This discussion is beneficial for electrical engineers, signal processing students, and professionals involved in demodulating AM signals or working with ADCs in communication systems.

FrankJ777
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I'm trying to demodulate simple AM by using an ADC and the bandpass sampling theorem (as I understand it.) The way I understand the theorem is that by sampling a bandpass signal of frequency f_{0} and bandwidth B, where f_{0} >> B, as long as I use a sampling frequency of >2B I can reproduce the signal even though the sampling frequency is much less than f_{0}.
From what I've read, to accomplish the bandwidth of the ADC must be at least f_{0}. I'm not sure exactly what this means. Does it only mean the input to the ADC must not attenuate a signal of f_{0}? Or are they referring to the sampling aperture. I see the formula use to demonstrate the concept is m[n] = M(t)δ (t-nTs). But i realize that δ is instantaneous where as a ADC is not, so I'm wondering if the width of my sample needs to be around 1/f_{0}?

If anyone can lend some insight I'd appreciate the help.
Thanks
 
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FrankJ777 said:
I'm trying to demodulate simple AM by using an ADC and the bandpass sampling theorem (as I understand it.) The way I understand the theorem is that by sampling a bandpass signal of frequency f_{0} and bandwidth B, where f_{0} >> B, as long as I use a sampling frequency of >2B I can reproduce the signal even though the sampling frequency is much less than f_{0}.
From what I've read, to accomplish the bandwidth of the ADC must be at least f_{0}. I'm not sure exactly what this means. Does it only mean the input to the ADC must not attenuate a signal of f_{0}?
Yes, that is correct.
FrankJ777 said:
Or are they referring to the sampling aperture. I see the formula use to demonstrate the concept is m[n] = M(t)δ (t-nTs). But i realize that δ is instantaneous where as a ADC is not, so I'm wondering if the width of my sample needs to be around 1/f_{0}?
I don't understand your question.

Bandpass sampling is somewhat sophisticated, and requires different filtration than regular baseband sampling to achieve a good SNR. I fear that a couple of short answers on this forum won't give you a deep enough understanding of sampling, filtration and DSP. Have you had a course in DSP? Do you have a DSP text to read?
 
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