DIGITAL SIGNAL PROCESSING IN ANALOG-DIGITAL AND DIGITALANALOG CONVERSION PROCESSES

Robert Wade Lowdermilk and fredric j. harris Communications Systems and Signal Processing Institute San Diego State University San Diego, California

ABSTRACT Signal Processing has been used for many years to realize enhanced performance and reduced cost of the ADC and DAC processes. The Sigma-Delta converter is a well-known example of the application of DSP to realize enhanced performance conversion. The sigma delta converter can be used as an A-to-D, as a D-to-A, as a D-to-D as well as a signal conditioner for other processing tasks. The signal processing which accompanies the sigma delta modulator performs a multi-rate filtering task, which entails a bandwidth reduction and a sample rate change. INTRODUCTION Digital signal processing (DSP) is concerned with the digital representation of signals and the use of digital signal processing platforms to analyze, modify, or extract information from signals. Many signals of interest are analog, meaning that they vary continuously in amplitude and in time. To take advantage of digital signal processing techniques, the signals must be converted to a sampled data signal and then quantized. A sampled and quantized signal is called a digital signal. Often we desire an analog representation of the digital data obtained after processing. To reconstruct the analog representation the data must be converted to a continuous time signal that has a reconstruction error related to the number of bits representing the data.

Analog-to-Digital

Analog Anti-Alias Filter

Automatic Gain Control

Sample and Hold

Quantize

Binary Pulse Code

Quant error

Figure 1. Signal Conditioning Tasks for Analog-to-Digital Converters

A process called analog-to-digital conversion (ADC) accomplishes the transformation of analog signals into a digital format. This process maps samples of a continuous amplitude waveform into a finite set of amplitudes. Conversely, a process called digital-to-analog conversion (DAC) accomplishes the transformation of digitally represented waveforms 1

into a continuous time signal. This process maps a discrete-time (sampled) waveform to a continuous time analog signal. A typical ADC or DAC is characterized, to the first order, by two parameters, the converter sample rate and the number of bits per sample. The converter sample rate is related to the Nyquist frequency of the input signal, and is typically 2.2 times the highest expected frequency of the input signal. The number of bits required to represent the waveform is related to the desired fidelity in signal reconstruction and is application dependent. The dynamic range of a converter is 6 dB per bit. For example, an 8-bit ADC running at 22 MHz would be capable of supporting a Nyquist frequency of 10 MHz and has a dynamic range of approximately 48 dB. Fig. 1 shows the signal conditioning required to operate a conventional ADC. The signal conditioning of analog signals starts with a bandwidth limiting operation performed by an analog filter designed to prevent spectral aliasing. The anti-aliasing filter is a recursive filter defined primarily by its poles. It tends to have a steep transition band edge that is always accompanied by severe group delay distortion. Thus the process of band-limiting the input signal by the anti-aliasing filter will significantly distort the time function within the band of interest while we reject the band not of interest. After the filtering, the signal amplitude is controlled by an automatic gain control loop and then is sampled and held by the sampling system. The ADC then quantizes the sampled data and the qua ntized level is converted to a binary representation for ease of transport and storage. For the DAC we reverse the process, which is shown in Fig. 2. We first convert the binary data samples into discrete levels. The discrete levels are converted to DC levels valid till the arrival of the next sample. The DC level is held by a sample-and-hold that protects the output levels from bit-race conditions at the sample boundaries. We say we de-glitch the converter. The voltage levels are controlled by a gain control system and finally an analog filter filters the signal. Here the analog filter serves to finish rejecting the residue spectral replicates. When we were collecting data we were eliminating the out-of-band energy. Here we are doing the same thing.

Digital-to-Analog

Binary Pulse Code

Sample and Hold

Automatic Gain Control

Analog Anti-Alias Filter

Glitch error

Figure 2. Signal Conditioning Tasks for Digital-to-Analog Converters

Another method of amplitude quantization that has gained great popularity is a class of conversion techniques called sigma-delta modulation. This method is based on significantly over-sampling the waveform relative to the Nyquist bandwidth with a low-bit quantizer and taking advantage of signal processing techniques to trade excess sample rate for enhanced dynamic range. This method applies to analog-to-digital converters (ADC), digital-to-analog-converters(DAC), and digital-to-digital converters(DDC). This paper will focus on the signal processing and what can we do with DSP to improve performance. We will not address the ADC realization. That process is more of an analog task that relies heavily on semiconductor technology as opposed to DSP. The emphasis of this paper will be the DAC and the DDC. USE OF OVER-SAMPLING AND FILTERING IN ADC Fig. 3 illustrates the motivation for the transformation from the conventional analog-to-digital process to an oversampled method. As seen in the figure, we use an analog anti-aliasing filter to limit the bandwidth and then sample the filter output at typically 10-20 percent excess sample rate relative to Nyquist. The process forms digital data that represents the band-limited analog input. Nyquist is defined as at least the two-sided bandwidth of the input signal. In practice Nyquist is equal to the two-sided bandwidth plus the transition bandwidth of the antialiasing filter. The transition bandwidth represents overhead spectrum aliased and not available for useful processing. To minimize this overhead we tend to specify analog filters with sharp transition bandwidths for the anti-aliasing task. The sharp transition band in the analog filter implies group delay distortion. This distortion corrupts the fidelity of the signal waveshape. We respond to this distortion by selecting analog filters with a very broad transition bandwidth. This has the advantage of reducing the cost of the filter (to match the reduced performance) while reducing the

16-Bits ANALOG ANTI-ALIAS FL E IT R BW=f0 ANALOG to DIGITAL C N E T R O V RE DIGITAL SIGNAL P O ES R CS

f=0.2f0

fS =2.2f0 15-Bits 16-Bits DIGITAL ANTI-ALIAS FL E IT R (1/4-BW) BW=f0 fS =8.8f0 1-Bit DIGITAL SIGNAL P O ES R CS 4:1

ANALOG ANTI-ALIAS FL E IT R BW=f0

ANALOG to DIGITAL C N E T R O V RE

f=8.8f0

f=0.2f0
12-Bits DIGITAL ANTI-ALIAS FL E IT R (1/16-BW) BW=f 0 16-Bits DIGITAL ANTI-ALIAS FL E IT R (1/4-BW) 16:1 BW=f0 DIGITAL SIGNAL P O ES R CS 4:1

ANALOG ANTI-ALIAS FL E IT R BW=f0

ANALOG SIGMADELTA MODULATOR

f=138.8f 0

fS =140.8f0

f=33.2f 0

f=0.2f0

Figure 3. Use of Over-sampling and Filtering in the Analog-to-Digital Conversion Process

distortion caused by the filter. We must operate then ADC at a much higher rate relative to Nyquist because the rate is the 2-sided bandwidth plus the transition that is now considerably wider. Once we have a digital representation of the signal we can reduce the bandwidth by using a digital filter. We can build digital filters with linear phase, which would eliminate the phase distortion that would have occurred had filtered the signal in the analog domain. A second benefit in the use of the digital filter is that it can correct for the residual distortion in gain and phase of the analog filter while reducing the bandwidth and lowering the sample rate. After bandwidth reduction, we reduce the sample rate commensurate with the bandwidth reduction. This rate is the data rate we had selected for the original single stage sampling of the analog signal. The advantage here is that we have the desired sample rate without the group-delay distortion. We have used oversampling to reduce the cost of the analog process while obtaining a non-distorting version of the filtering process in the digital world. We can extend this concept by massively over-sampling the data, shown in the third block diagram of the figure. We present the analog signal to an inexpensive analog filter that has an even wider transition bandwidth. We then use an analog sigma-delta modulator to convert the analog signal into a binary data stream. The sigma-delta modulator typically outputs one bit per sample that represents the input signal by a pulse width modulated sequence. The sequence preserves the fidelity of the original analog signal. We use digital filters to reject the modulation noise of the sigma-delta modulator. Since we are at a very high data rate we seek a very inexpensive filter. The filter commonly applied to this task is a Hogenauer filter; an efficient filter that requires no multiplies. We typically use the first stage of filtering to reduce the bandwidth and sample rate 16to-1. We then use a high quality filter to finish the anti-aliasing task and correct the spectral droop created by the analog filter and the first stage Hoegenauer filter. After bandwidth reduction the sample rate is again reduced 4-to-1. This is the form of the standard sigma-delta modulator used in the commercial audio community. USE OF OVER-SAMPLING AND FILTERING IN DAC. We now consider converting digital data to its analog counterpart. We simply reverse the processing steps shown in Fig. 3. This is illustrated in Fig. 4. Digital data is presented to a DAC that generates samples at the Nyquist rate (twice the bandwidth plus the transition bandwidth). The spectral replicates at integer multiples of the sample rate are suppressed but not totally rejected by the spectral zeros of the sinx/x response. The sinx/x is the frequency response of the zero-order hold, the time response of the DAC. The analog filter must finish the task of rejecting the spectral replicates. Of course if the analog filter has a sharp transition width we distort the reconstructed signal.

We respond by raising the sample rate of the digital data prior to the DAC. As shown in the middle block diagram, we up-sample the data 1-to-4 with a digital-interpolating filter. This results in increased separation of the spectral replicates in the frequency domain. We now of course must operate the DAC at the higher rate. The analog filter following the DAC is less expensive because the filter can have a wider transition bandwidth to achieve the attenuation required to suppress the residues of the spectral copies. All compact disk players use this method of signal reconstruction. CD players usually boast 4-to-1 oversampled outputs.

16-Bits DIGITAL SIGNAL PROCESS DIGITAL to ANALOG CONVERTER ANALOG LOW-PASS FL E IT R

B = f0 W fS =2.2f0 16-Bits DIGITAL LOW-PASS FL E IT R

f=0.2f 0
16-Bits DIGITAL-to ANALOG CONVERTER

DIGITAL SIGNAL PROCESS

ANALOG LOW-PASSS FL E IT R

1:4

(1/4-BW) B = f0 W

f=0.2f0
16-Bits DIGITAL SIGNAL PROCESS DIGITAL LOW-PASS FL E IT R (1/4-BW) B = f0 W

B = f0 W fS =8.8f0 16-Bits DIGITAL LOW-PASS FL E IT R (1/16-BW) B = f0 W DIGITAL SIGMADLA ET MODULATOR DIGITAL-to ANALOG CONVERTER

f=8.8f0
1-Bit ANALOG LOW-PASS FL E IT R

1:4

1:16

f=0.2f0

f=33.2f0

B = f0 W fS =140.8f 0

f=8.8f 0

Figure 4. Use of Over-sampling and Filtering in the Digital-to-Analog Conversion Process

This trade of excess sample rate for reduced cost analog filtering can be extended by further increases in sample rate. After raising the digital data sample rate 1-to-4 we continue and use the digital Hogenauer filter to raise the sample rate again by 1-to-16. We now have massively over-sampled data, which we convert to a one-bit binary sequence by using a digital sigma-delta modulator. This is a digital-to-digital sigma-delta modulator. We now have a one-bit representation of the base-band signal with all the fidelity of the base-band signal. A one-bit DAC and an inexpensive low-pass filter converts the output of the sigma-delta modulator to an analog signal. The filter is required to eliminate the modulation noise. This is the DAC structure in CD players that advertise one-bit mash converters. These are examples of how over-sampling and filtering allows us to take advantage of the high correlation to convert a large number of bits into a smaller number of bits to represent a signal over a fraction of the sampled bandwidth.

QUANTIZATION ERROR To facilitate later discussions the concept of quantization error needs to be defined. A quantizer is defined as a device that converts a higher resolution signal into discrete amplitude levels of lower resolution. A quantizer is normally considered a memoryless non-linearity, which means each conversion is performed independently of other conversions. Taking the difference between the input and the output of a quantizer results in the error between the input signal and the output signal. This error is referred to as the quantization noise. We model this error as additive white noise. Fig. 5 shows the quantization models, the transfer function and the corresponding quantization error for a memoryless nonlinearity. A characteristic that distinguishes the sigma-delta modulator from other quantizers is that it has memory. Having memory means that it uses the correlation from sample to sample, which is achieved by over-sampling, to represent the data with a smaller number of bits per sample.
^ x ^ q=x - x
x

Quantization Error Memoryless Transfer Function

Q(X)

^ X

q + q X + ^ X

Quantization Noise Difference Between Output and Input

Linear Model Additive White Noise

Figure 5. Quantization Error Model for Memoryless Conversion Process

QUANTIZER SIGNAL-TO NOISE RATIO Typically we are interested in the error caused by the quantization process. Fig. 6 illustrates the concept of the signal-to-noise ratio for the case of a uniform quantizer. As shown in the figure, we can compute the mean squared error and the mean squared signal energy, allowing us to compute the signal-to-noise ratio. The first curve shows the probability density function of the quantization error, the x-axis representing the amplitude of the error bound between plus and minus 1/2 quantile. Since the area under the curve equals one, the height of the function is simply 1/q, where q 6

equals one quantile. Similarly, the second curve represents the probability density function of the quantized signal, the x-axis being the amplitude of the signal bound between the most negative and maximum count of the quantile, +Mq, where M is the maximum count. The signal-to-noise ratio can be easily computed by taking the ratio of the corresponding variances. This represents the dynamic range of the signal relative to the noise. The derivation is shown in Fig. 6. We can see from the figure that in a sense the narrower the noise is compared to the signal, the more levels there must be to represent the signal, hence more bits are required. With a memoryless quantizer, the only way to improve quantizer SNR is to use more bits. The rule is simple: if we want a better signal-to-noise ratio (with a memoryless quantizer) we must spend additional bits. There is a second option. If we cant increase the number of bits, generate correlated samples and use filtering to convert them into more bits. We can average correlated samples to improve the SNR. Correlated signals occur as a result of over-sampling, and that is the focus of this paper.

fe (e)

1 q

fs(s)
1 2Mq e S
-Mq M q

-q/2 q/2

2 q =

1 q

+q/ 2

/2 q

e2de =

1 e3 + q / 2 q2 = q 3 q/2 12

2 s =

q 1 +M 2 1 s3 + Mq q2 q2 s ds = = M 2 = 22b q 2Mq M 2 Mq 3 Mq 12 12

SNR LIN =

s2 2 2 b q 2 / 2 = = 22 b 2 q q2 / 2

SNR dB = 10 log10 (SNR LIN ) SNR dB = 10 log10 (2 2b ) = 20 log10 ( 2) = 6b dB

Figure 6. Maximum Signal-to-Noise Ratio for a Uniform Quantizer

OVER-SAMPLING, FILTERING AND DOWN-SAMPLING: Let us conduct a thought experiment and examine a signal in the time and frequency domain. This is illustrated in Fig. 7. First we quantize a signal and examine the difference between the input and output of the quantizer. The difference between the input and the output is the qua ntization error and we know its energy is q2 /12. If we take the Fourier transform of the error signal and integrate the spectral noise, we will again measure q2 /12. This is Parseval's Theorem, which says, that if you integrate the squared time signal or squared spectral function, you get the same energy. With this background we are ready conduct our experiment.

X(t)

X(n/4)

Xq(n/4)

14-bit ADC 4fs

1/4-BW Digital Filter

q1 n ) (

yq(n/4)

yq(n)

4:1

E q1 (n)}= { 12
2

q(n) 2

E q2 (n)}= {

(q/2) 12

f0 q
2

f
12

f
f0 (q/2) 12
2

4fS

f
f0 (q/2) 12
2

f
f0 4fS

4fS

f0

fS

Figure 7. 4-to-1 Over-sampling, Filtering, and Down-sampling. Quantizing Noise in Time and Frequency.

We are going to over-sample the data by 1-to-4 and again form the difference between the input and output of the sampler. The converter does not care what the sample rate is, it just knows that the difference between the input and the output is the quantization noise. Therefore, when we square and average the quantization error we still measure q2 /12. When we examine the error signals' spectrum in the frequency domain, we observe that you now have four times the spectral span that we originally had because we now have four times the sample rate. We gather up all the noise energy that we observe over this larger spectral span and we, once again, measure q2 /12. In this thought experiment, we have a larger spectral span over which we have distributed the same amount of spectral energy. Thus the noise power spectral density must be lower to maintain the same amount of noise spread over a larger bandwidth.

Because the signal is over-sampled, the signal's spectrum is located over a small fraction of the spectral span. We now use a digital filter to reject the noise in the out-of-band spectral regions. In doing so we reject noise occupying of the bandwidth, and we eliminate of the noise power. We only have of the original noise, so now instead of having q2 /12, we have (q/2)2 /12. By over-sampling by 4-to-1 and filtering, the amplitude of the quantization noise has been reduced by a factor of 2. What this has shown us is that we can convert excess sample rate to improved quantization SNR by averaging to obtain and bandwidth reduction. We want to find a faster exchange of excess sample rate for bit precision. The current rule is that every time we double the sample rate and have more spectral span to spread out the noise, we improve the SNR by 3 dB or bit. We want to do much more than bit per doubling. We now invoke an additional technique that takes advantage of the high data correlation obtained by over-sampling. We use the correlation to predict the signal and only quantize the prediction error rather than the signal itself. If we do a good job of predicting, the prediction error will be small, hence the quantization can proceed with a reduced number of bits while maintaining a specified quantization SNR. CONVERTING A PREDICTOR-CORRECTOR DIFFERENTIAL QUANTIZER TO A DELTA MODULATOR: The delta modulator represents early uses of prediction to obtain high SNR with a low precision quantizer. The delta modulator took advantage of the fact that if we are significantly over-sampled; we can use the last sample of the signal to predict the next sample of the signal. We only have to measure and quantize the difference between the prediction and the actual data, and that difference represents the signal to be delivered be a data path to a reconstruction filter. Fig. 8 shows the evolution of the predictor-correlator architecture to the delta modulator. As seen in the figure, we have an integrator estimating the next input sample of signal by the last sample. We form and quantize the difference, which we define as the prediction error. We correct the estimate with the quantized error, resulting in a quantized estimate of the signal. The quantized errors represent the information content of the sampled signal. It is these errors that are transmitted to the reconstruction filter at the receiver. Note that the input to the delta modulator is a discrete time sample that has not yet bee amplitude quantized. The receiver shown in the top block diagram of the figure contains the same predictor as the transmitter. The receiver uses the predictor and the past data to predict the next sample in the same way the transmitter does. The prediction has the same error as the transmitter. The transmitter has measured the error and tells the to correct the prediction by the measured and quantized error. Only the errors have to move across the data path. The quantization noise is distributed over the extra spectral span obtained by the oversampling process. We reject the noise in that extra bandwidth to have to improve SNR. We eliminate the extra bandwidth with a digital low-pass filter. We perform the filtering to reject the out-of-band noise energy. As an example, if we have over-sampled 20:1,

then the signal resides in the lower 5 percent of the total bandwidth and the rest of the bandwidth contains noise only. Therefore by eliminating 95 percent of the noise bandwidth we obtain a significant reduction of the noise power.

q(n) x(n) e(n)

~ Q ~

e(n)

e(n)

x(n)

^ x(n)
Predictor H(Z)

x(n)

^ x(n)

Predictor H(Z)

q(n) x(n) e(n)

~ Q ~
Z
-1

e(n)

e(n)

x(n)

x(n)

LPF
x(n)

^x(n)
H(Z)

^x(n)

-1

H(Z)

q(n) x(n) e(n)

~ Q

e(n)

e(n)

~
Z Z-1

x(n)

~ ~

x(n)

LPF

1 Z-1

Figure 8. Converting a Predictor-Corrector Differential Quantizer to a Delta Modulator

CONVERTING DELTA MODULATOR INTO A SIGMA-DELTA MODULATOR The delta modulator receiver is receiving a sequence of +1 that it accumulates to form an output equal to the input of the delta modulator at the transmitter. The quantization noise in the reconstruction occupies the total bandwidth of the system, and the filtering that follows eliminates the noise realizing the desired improvement in the signal to noise ratio. We now impose the requirement that we dont want uniform noise nor noise with spectral density peaking in the neighborhood of the input signal spectrum. Particularly we want to keep the noise out-of-band. We will process the noise to reduce the noise power spectral density within the bandwidth of the signal. Fig. 9 shows the process we are going to discuss.

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Let us pre-condition the delta modulator input signal so that it is more correlated by ave raging with an integrator. It will be necessary to undo the integration process by placing a differentiator on the output of the delta modulator. If we have a linear system the integrator and differentiator will exactly cancel each other. In fact we do not have a linear system because of the quantizer in the delta modulator. To first order, the circuit behaves as a linear model. What we now have is a pre-integrator, a delta modulation, and a postdifferentiator to undo the integration.

x(n) Z Z-1

y(n)

q(n)

X(n)

~ Q

~
Z-1 Z

e(n)

X(n)

e(n)

~
Z Z-1

x(n)

~ ~

x(n)

LPF

1 Z-1

x(n) Z Z-1

y(n)

~
X(n)

q(n)

~ ~
X(n)

e(n) Z Z-1

X(n)

Z-1 Z

x(n)

~ ~

x(n)

Z Z-1 Z
-1

LPF

q(n) x(n)

~
Z Z-1 Z
-1

x(n)

~ Q

X(n)

X(n)

~ ~

x(n)

LPF

Figure 9. Converting a Delta Modulator to a Sigma-Delta Modulator

Remember that the delta modulator receiver consists of an integrator followed by a lowpass filter. The next step is to move the differentiator of the transmitter to the receiver (the reconstruction filter). At the reconstruction filter we now have a differentiator fo llowed by an integrator, which conveniently cancel each other. Now the receiver consists of only a low-pass filter. This represents a significant simplification of the reconstruction process. At the transmitter, we have both legs into the summing junction containing an integrator. We slide both integrators through the summing junction and merge them as a single int egrator inside the loop. We can do this because the sum of two integrators is equivalent to the integration of the sum. The difference now, is that the integrator and the quantizer are at the same spot in the loop. Before this they were in different locations in the loop; one was feed-forward and the other was feedback. They are now both in the feed-forward path. This is the form of the device called a sigma delta modulator, and is shown in the lower block diagram in Fig. 9.

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In the literature this structure is equally likely to be called a sigma-delta modulator as a delta-sigma modulator. The question is, which is correct? Our vote, based on entomology, it sigma-delta. After all, it is an integrator (sigma) followed by a delta modulator. CONVERTING A NOISE FEEDBACK LOOP TO A SIGNAL DELTA MODULATOR An alternate version of deriving a sigma-delta modulator is shown in the top block diagram of Fig. 10, referred to as a noise feedback loop. In this configuration, we set up the problem by observing that the output of the quantizer as the input signal plus quantization noises. We form the difference between the input to the quantizer and the output of the quantizer, which presents to us the noise contributed by the quantizer. This noise is inserted into the feedback path. If we redraw the noise feedback model we see that in fact there are two loops. One is the minor loop that does not include the quantizer. And the other is the major loop, which does include the quantizer. The inner loop can be replaced with an integrator, resulting in the sigma-delta modulator. We have shown that the sigma-delta modulator can be derived from a different model, a noise feedback loop. By properly drawing the noise feedback loop we obtain the sigmadelta modulator. The noise feedback model allows us to show the transfer function and show the advantage of getting the integrator into the feedback path, it is now the noise that is shaped by the loop.

q(n) x(n) y(n)

-q(n-1) Z
-1

-q(n)

x(n)

q(n) y(n)

Z
-1 -1

x(n)

q(n)

x(n) y(n)

q(n)

y(n)

Z-1

Z-1

-1

Figure 10. Converting a Noise Feedback Loop to a Sigma-Delta Modulator

12

TRANSFER FUNCTION OF SIGMA-DELTA MODULATOR We can derive the transfer function of the noise feedback model shown in Fig. 10 by first writing the difference equation of the output y(n) and taking its z-transform. y(n) = x(n) - q(n - 1) + q(n) Y(z) = Z(y(n)) = X(z) - z -1 Q(z) + Q(z) Y(z) = X(z) + (1 - z -1 )Q(z) Y(z) = X(z) + z -1 Q(z) z (EQ 1.0)

Eq. 1.0 shows that the output of the noise feedback loop is equal to the input plus noise filtered by a derivative, and that the derivative has a zero at DC. We can also determine the transfer function of the sigma-delta modulator shown in Fig. 10 by taking the ratio of the forward gain by one minus the loop gain for the input to the output.

Y(z) = X(z)

FG1 (z) FG 2 (z) + Q(z) 1 - LG(z) 1 - LG(z)

z 1 z 1 Y(z) = X(z) + Q(z) z 1 z 1 1+ z 1+ z z 1 z 1 Y(z) = X(z) z z -1 + Q(z) 1 (z - 1) + z z (z - 1) + z z 1 z z -1 + Q(z) z z z -1 z (EQ 2.0)

Y(z) = X(z)

Y(z) = X(z) + Q(z)

Comparison of Eq. 1.0 and Eq. 2.0 indicates that the transfer functions of the noise feedback loop and the sigma-delta modulator are identical. Now that we have shaped the noise we now have the interesting property that the signal and noise now occupy (nearly) non-overlapping spectral bands. Because the signal is significantly over-sampled, its 13

spectrum only occupies the bottom couple few percent of the spectral span. The noise, on the other hand, being filtered by the noise transfer function differentiator, exhibits a spectral zero in the neighborhood of DC. Thus the noise contributes very little power to the band occupied by the signal. In a sense, the signal and noise now occupy different bandwidths and can be separated with a low-pass digital filter. This relationship is illustrated in figure 11.

q(n) x(n) + y(n) +

Z Z-1

y(n) = x(n) q(n) - q(n-1) Y(z) = X(z) + Q(z)(1-z )

-1

NOISE TRANSFER FUNCTION = (1-Z-1) =

Z Z-1

x()

Q()NTF()

Figure 11. Transfer Function of Sigma-Delta Modulator

CONCLUDING REMARKS. We have presented a simple entomology for the structure of the sigma-delta modulator. We started with the memoryless quantizer and determined the quantization SNR as a function of the number of bits involved in the quantization process. We found we could improve quantization SNR by over-sampling and averaging, with the averaging having the effect of improving SNR by rejecting out of band noise. We then took advantage of the over-sampling to form a predictor-corrector loop which could maintain a specified SNR with a small number of bits by quantizing prediction errors rather than the signal directly. Finally we reduced prediction error by using an integrator to improve the correlation of the input signal and compensated for the integrating pre-filter with a differentiator post-filter. After a bit of sliding circuit components about, we recognized the structure as the sigma-delta modulator. As a footnote we also exa mined the noise feedback loop and recognized that with a bit of sliding circuit components we once again had the sigma-delta structure.

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BIBLIOGRAPHY 1. Sklar, Bernard, Digital Communications Fundamentals and Applications, Prentice Hall, 1988 2. Ifeachor, Emmanuel C. and Jarvis, Barrie W., Digital Signal Processing A Practical Approach, Addison-Wesley, 1993. 3. Hnatek, Eugene R., A Users Handbook of D/A and A/D Converters, John Wiley and Sons, 1976 4. Meyr, Heinrich; Moeneclaey, Marc and Fechtel, Stefan A., Digital Communication Receivers Synchronization, Channel Estimation, and Signal Processing, John Wiley and Sons, 1998 5. harris, fred, Multirate Digital Signal Processing in Digital Receivers, Keynote Presentation at the Second International Synposium on DSP for Communication Systems, April 26-29, 1994 6. Dick, Chris and harris, fred, Narrow-Band FIR Filtering with FPGAs using SigmaDelta Modulation Encoding, Journal of VSLI Signal Processing, 1996 7. harris, fred; McNight, W.;Speiser, J.; Whitehouse, H., Developments in Techniques for Enhancing the Dynamic Range of Analog to Digital Converters 8. harris, fred, A Wide Dynamic Range A-to-D Converter using a Band Limited Predictor Correlator DPCM Algorithm 9. Dick, Chris and harris, fred, FPGA Signal Processing using Sigma-Delta Modulation, 1999 10. harris, fred, Applications of A-to-D and D-to-A Converters in digital Signal Processing Based Communication Receivers, Keynote Presentation at the Advanced A-D and D-A Conversion Techniques and Their Applications, July 6-8, 1994 11. harris, fred, Digital Signal Processing in Radio Receivers and Transmitters

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