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Oversampling

The principle of oversampling is simple: run ADC and DAC (Analog digital converters) at a sampling rate f e well beyond Nyquist, say at f e = β f 0 where f 0 > 2 B is a sufficient sampling rate.

We call β > 1 the oversampling rate (OSR); it is usually an integer. For audio CD, β = 4 (data is sampled at f e = 176'100 samples/sec but stored on the disc at f 0 = 44'100 samples/sec).

To understand the benefits of oversampling, one has to recall the typical schema [link] of digital signal processing (DSP). First, in the ADC and DAC parts oversampling reduces the losses ofquality due to working too close at Nyquist rate: sampling and reconstruction occur with a cutoff-frequency f e far from the Nyquist rate 2 B allowing for simple filter design or even omission of filtering. Second, as an additional benefit, oversampling reducesnoise created by quantification, sampling or other sources of errors.

Note: down- and upsampling before and after processing (DSP: e.g.: de-noising, detection, enhancement, storage, transmission) allowsfor efficient computation and/or transmission at low rate.

b a n d - l i m i t A D C M D S P M D A C s m o o t h

Noise reduction under oversampling

In the context of oversampling, the signal has been sampled (and thereby quantized) at f e = β f 0 , with f 0 > 2 B and OSR β > > 1 . Thus, we may decimate the quantized signal by a factor β and still keep the full signal quality. Decimation consists of digital low-pass filtering atcutoff frequency 1 / ( 2 β ) (which corresponds to f 0 / 2 ), followed by downsampling.

Low-pass filtering and downsampling will not change the signal since f 0 > 2 B , and thus decimation will not change the signal power (cpre. [link] , [link] , [link] magenta). However, low-pass filtering will reduce the power of the noise ; the factor of reduction is β if an ideal filter is used. We offer several arguments for this fact.

Spectral picture of noise reduction (ideal filter). During the first step, low-pass filtering, the high frequencies of the power spectrum of the noise are cut away:

S ( f ) = P · Rect ( f ) 1 2 β i d e a l f i l t e r s i n c S ideal ( f ) = P · Rect ( f β )

After low-pass filtering, the noise is correctly band-limited (at a loss of power); consequently,downsampling will not change the power any further.

Spectral picture of noise reduction (digital filter). When using a digital filter b = ( b 1 , . . . , b q ) with energy E b = b 1 2 + . . . + b q 2 to filter the samples ε n , the spectrum will change through filtering roughly as

S ( f ) = P · Rect ( f ) 1 2 β d i g i t a l f i l t e r b S digital ( f ) P · E b · β · Rect ( f β )

Here, we used that the DFT of a well-designed digital filter b is roughly a rectangle of width 1 / β , meaning that a fraction 1 / β of the samples b ^ k at the center take some constant value λ , the others are zero. It is easy to verify that λ = E b β in order for the energy of b to be E b (see Comment 8 ). Consequently, the portion 1 / β of the K samples ε ^ n will be multiplied by λ , the others will be set to zero. Using [link] we see that S ( k / K ) is multiplied by λ 2 = E b β for - K / ( 2 β ) k K / ( 2 β ) , and set to zero otherwise.

Comment 8 Let us denote by q the number of samples of the DFT of b . For a well designed digital filter there are roughly q / β samples equal to some constant λ , the remaining samples are zero. Since the DFT increases energy by factor equal to the sample size q , we get

q E b = b ˆ 1 2 + . . . + b ˆ q 2 = λ 2 q / β
λ = E b β .

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Source:  OpenStax, Sampling rate conversion. OpenStax CNX. Sep 05, 2013 Download for free at http://legacy.cnx.org/content/col11529/1.2
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