Sampling theorem with noise

In summary, Shannon's sampling theorem states that the sampling rate should be at least twice the signal bandwidth, but ideally equal to the bandwidth. Oversampling is unnecessary but can be used for noise reduction in repetitive signals. However, for properly designed sampling systems, oversampling does not improve the SNR. Oversampling may improve the accuracy or resolution in spectral analysis, but this is a separate topic from SNR. A mathematical proof for the improvement of accuracy in spectral analysis through oversampling can be found in books on signals and systems or through the application of the Wiener-Khinchin theorem. It is important to separate and understand the different concepts related to sampling, noise reduction, and spectral analysis in order to pose a well-posed question.
  • #1
sergetsier
1
0
Sampling theorem says that we should sample at least at a double rate than the bandwidth of the signal : Fs>=2B.
Ideally Fs==2B e.i. oversampling is unnecessary.
However, there's always noise present in the signal. Hartley's theorem connects the SNR with the bandwidth and the sampling rate for a binary digital signal : http://en.wikipedia.org/wiki/Shannon–Hartley_theorem
But I deal with an analog signal which is similar to an audio signal. What is the relation between the sampling rate and the SNR?
Wiki about oversampling:
"Noise reduction/cancellation. If multiple samples are taken of the same quantity with a different (and uncorrelated) random noise added to each sample, then averaging N samples reduces the noise variance (or noise power) by a factor of 1/N"
I cann't find the proof of it.
To precise my question, I have an oversampling signal acquiring system and the oversampling normally improves the accuracy in spectrum calculation. Matlab simulation confirms it.
I'd like to have a mathematical proof of this.
Thank you
 
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  • #2
You have rather indiscriminantly jumbled together a bunch of separate concepts in this post. First, Shannon's channel capacity theorem usually has little to do with sampling. The theoretical maximum information transfer rate over a noisy channel is approached in practice by error correction, Viterbi and turbo coding, block interleaving, and other tools available to the communications engineer.

Second, signal averaging is most commonly applied to repetitive signals with independent noise. A classic example is trying to detect a weak radar echo return that is below the radar receiver's thermal noise floor. If the returns from many transmit pulses are averaged, the echo signal power adds faster than the noise power and the SNR increases. The math behind should be found in books on signals and systems (I'd look in Oppenheim and Willsky, or Root and Davenport). Here is a wiki page
http://en.wikipedia.org/wiki/Signal_averaging"

Are you suggesting that oversampling improves the SNR? If so, then this is not supported by theory for a properly designed sampling system. By Shannon's sampling theorem, the signal bandwidth must be limited to less than Fs/2 by an anti-aliasing filter in front of the sampler. This filter also limits the noise bandwidth, with the result that samples taken at a sampling rate greater than Fs/2 are no longer independent. Say you oversample by a factor of b then average every b samples to produce one "improved" sample. Both signal and noise are correlated over the b samples, so the SNR of the improved samples is not appreciably better. This may be demonstrated mathematically by applying the Wiener-Khinchin theorem that relates a signal's autocorrelation to its power spectrum. This theorem should be in the same books, or see Bracewell's The Fourier Transform and Its Applications (he calls it the autocorrelation theorem).

On the other hand, you later say that "oversampling normally improves the accuracy in spectrum calculation." Spectral analysis is an entirely different topic, unrelated to the ones above dealing with SNR. Do you really mean accuracy, by the way, or resolution? If you take some time to learn about, digest and separate the different topics you have mixed up here, and come back with a single well-posed question, we'll be better able to help you.
 
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1. What is the Sampling Theorem?

The Sampling Theorem is a mathematical concept that states that a continuous signal can be accurately represented by a discrete sequence of samples, as long as the sampling rate is at least twice the highest frequency component of the signal.

2. How does noise affect the Sampling Theorem?

Noise, or any unwanted interference in the signal, can cause errors in the sampling process. If the noise is within the frequency range of the signal, it can be mistakenly sampled and lead to distortion of the reconstructed signal.

3. How is the Sampling Theorem used in practical applications?

The Sampling Theorem is used in various fields such as digital signal processing, communication systems, and data compression. It allows for the accurate representation of analog signals in digital form, making it possible to process and transmit them efficiently.

4. What are the limitations of the Sampling Theorem?

The Sampling Theorem assumes that the signal is band-limited, meaning that it has no frequency components above a certain limit. In real-world scenarios, this is not always the case, and the Sampling Theorem may not hold. It also requires a sufficiently high sampling rate, which can be costly and impractical in some situations.

5. How is the Nyquist-Shannon sampling rate related to the Sampling Theorem?

The Nyquist-Shannon sampling rate is the minimum rate required to satisfy the Sampling Theorem. It states that the sampling rate must be at least twice the highest frequency component of the signal to avoid aliasing and accurately reconstruct the original signal.

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