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SNR in Data converters

The ideal ADC quantizes its input with the practical result of adding noise to the
input signal. This noise is called as the quantization noise. Quantization noise is
the effective noise added to a signal after passing through an ADC. This chapter
discusses how to determine the actual signal to noise ratio (SNR) of a data
conversion system and topologies for improving the data conversion systems

SNR ideal =

Improving SNR using Averaging

Quantization noise can be reduced to a certain extent by averaging the input
signal. The input signal is passed through two parallel paths which contain an
ADC and a DAC. The output through both these paths is averaged. In general
averaging K samples results in an RMS quantization noise voltage of

Then SNRideal = 6.02N + 1.76 + 10 log K

From the above equation it can be deduced that averaging two samples
causes the SNRideal to increase by 3 dB or the effective resolution of the data
converter to increase by 0.5 bits.
Increased resolution, Ninc =
The averager can be thought of as a filter. Averaging results in an attenuation of
some of the input signal frequencies and the average of the input signal goes to
zero when the input signal frequency is

. These observations can be

supported by taking the magnitude and phase response of the averager.

Let the input signal be x(nTs). The clock signal is passed through an
inverter and given to the second path which is the delayed signal
Both these are added to produce an output signal


Taking z transforms

Decimating filters for ADC

It was observed that while averaging in a data converter the input signal
bandwidth B has to be equal to

. To lower power dissipation and to simplify

the circuitry, the rate at which these samples are generated is lowered.
new = 2B =
This reduction in sampling frequency is called decimation or down
In the time domain the input and output of the decimating filter is

The decimation filter will take K samples add them and the result in divided by K.
Taking the z transform of Eq.


shows one circuit to implement the above Eq. and is called the

accumulate and dump circuit.


Latches L1 are used to accumulate the K samples and Latches L2 are used to dump
the sum, hence the name accumulate and dump. First the set of latches L1 are reset.
The sampling clock is used to clock L1 K times until the sum of K inputs is
accumulated. The accumulated sum is dumped into L2. At the same time L1 starts
the process of accumulating the next set of K samples. To find the frequency
response of the circuit Z is set to

The frequency response of the accumulate and dump for K=2 and K=4 are shown
in Fig. These are called sinc filters for obvious reasons.


Averaging filter :
To implement averaging filter on the chip, the transfer function is split into
the numerator and denominator.

This implies there are L differentiators and L integrators. Fig. shows the
block diagram of a digital integrator and digital differentiator.


Band pass and high pass sinc filters.

A high pass filter can be generated by cancelling a comb filter zero at fs/2.
The filter response shifts to fs/2. For K = 8 H (z) =

. The high pass filter

frequency response is as shown in Fig.

Interpolating filters for DAC

Interpolation or up sampling or introduction of samples between adjacent

DAC inputs is used to attain a large effective output resolution. Fig.

shows the block diagram of a DAC that uses interpolation.

Digital in

analog out



Fig. 6.19 Block diagram of a DAC with interpolator.

The interpolator introduces additional samples in between input samples.
For example samples may be introduced after every (k-1) samples. If the
frequency of the input samples is 2B then the frequency of the samples coming out
of the interpolator is
If y (nTs) is the output of the interpolator and x [ki Ts] the input to the
interpolator then

Taking the z transform


The input signal to the interpolator is digital. This input band limited to B is
connected to a set of latches clocked at 2B. The output of these latches is
connected to the digital filter which is clocked at 8B. The signal is then given to
the second stage interpolator. The amplitude of the spectrum reduces after passing
through the second stage interpolator. The word size and world rate increases. The
reconstruction filter RCF attenuates the unwanted spectral contents.
Summary :
This chapter characterized a system using ADCs and DACs in terms of the
signal to noise ratio (SNR). The data converter performance can be measured by
ENOB, spurious free dynamic range and signal to noise plus distortion ratio. The
effect of clock jitter was presented. Clock jitter is the variation in the period of the
clock signal around the ideal value. To reduce the quantization noise voltage
averaging is used.
Not only averaging but decimation is employed in decimating filter. To
implement averaging filters on the chip, the transfer function is split into L
differentiators and L integrators.
From the magnitude and frequency response of the differentiators or comb
filters it was observed that by cancelling zeros on the unit circle yielded different
types of filters. A digital resonator was employed to cancel zeros. The
interpolation filter was used to up sample the input signals.
The interpolation and decimation fitters are used in DACs and ADCs. One
application of this is in the digital audio field where different frequencies are
required for broad casting, for compact discs and audio tapes. Both up sampling
and down sampling are employed.