Field Programmable Gate Array Implementation of 14 Bit Sigma-Delta Analog To Digital Converter
Field Programmable Gate Array Implementation of 14 Bit Sigma-Delta Analog To Digital Converter
=
=
(1)
Where,
( ) 1 t =
, 0 t =
0, elsewhere
Most ADCs can be classified into two groups according to
the sampling rate criteria. Nyquist rate converters, such
as a successive approximation register (SAR), double A
practical A/D converter transforms x(t) into a discrete-
time digital signal, x*(t), where each sample is expressed
Field Programmable Gate Array
Implementation of 14 bit Sigma-Delta Analog
to Digital Converter
Sukhmeet Kaur
1
, Parminder Singh Jassal
2
1
M.Tech Student , Department of Electronics and Communication Engineering, Yadavindra College of Engineering, Punjabi
University Guru kashi Campus , Talwandi Sabo 151 302 , Punjab (India)
2
Assistant Prof., Department of Electronics and Communication Engineering, Yadavindra College of Engineering, Punjabi
University Guru kashi Campus , Talwandi Sabo 151 302 , Punjab (India)
I nt ernat i onal Journal of E mergi ng Trends & Technol ogy i n Comput er Sci ence (I JE TTCS)
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Volume 1, Issue 2, July August 2012 ISSN 2278-6856
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with finite precision. Each sample is approximated by a
digital code, i.e., x(t) is transformed into a sequence of
finite precision or quantized samples x(n). Integration,
and oversampling converters, sample analog signals
which have maximum frequencies slightly less than the
Nyquist frequency, f
N
= fs/ 2, where fs is the sampling
frequency [2]. Meanwhile, oversampling converters
perform the sampling process at a much higher rate, f
N
<< Fs, where Fs denotes the input sampling rate.
Figure 2: Generalized ADC process [8]
Fig. 3 illustrates the conventional A/D conversion process
that transforms an analog input signal x(t) into a
sequence of digital codes x(n) at a sampling rate of fs =
1/T, where T denotes the sampling interval.
Figure 3: Conventional ADC process [8]
( 2 )/
1
( ) ( ) ( )
j n t T
n n
x t t nT x t e
T
= =
=
(2)
also
2 ( 2 )/
1 1
*( ) ( ) ( )
s
j f nt j n t T
n n
x t x t e x t e
T T
= =
= =
(3)
Eqn. 2 states that the act of sampling (i.e., the sampling
function):
( ) ( )
n
x t t nT
=
is equivalent to modulating the input signal by carrier
signals having frequencies at 0, fs, 2fs,. . .. In other
words, the sampled signal can be expressed in the
frequency domain as the summation of the original signal
component and signals frequency modulated by integer
multiples of the sampling frequency Thus, input signals
above the Nyquist frequency, f
n
, cannot be properly
converted and they also create new signals in the base-
band, which were not present in the original signal. This
non-linear phenomenon is a signal distortion frequently
referred to as aliasing [6]. The distortion can only be
prevented by properly lowpass filtering called the anti-
aliasing filter. In addition to an anti-aliasing filter, a
sample and hold circuit is required. Although the analog
signal is continuously changing, the output of the sample
and hold circuitry must be constant between samples so
the signal can be quantized properly. This allows the
converter enough time to compare the sampled analog
signal to a set of reference levels that are usually
generated internally [7]. If the output of the sample-and-
hold circuit varies during T, it can limit the performance
of the A/D converter subsystem.Each of these reference
levels is assigned a digital code. The process of
converting an analog signal into a finite range number
system introduces an error signal that depends on how the
signal is being approximated. This quantization error is
on the order of one least-significant-bit in amplitude, and
it is quite small compared to full-amplitude signals.
However, as the input signal gets smaller, the
quantization error becomes a larger portion of the total
signal. When the input signal is sampled to obtain the
sequence x (n), each value is encoded using finite word
lengths of B-bits including the sign bit. Assuming the
sequence is scaled such that ( ) 1 x n s for fractional
number representation, the pertinent dynamic range is 2.
Since the encoder employs B-bits, the number of levels
available for quantizing x (n) is 2
B
. The interval
between successive levels, q, is therefore given by:
1
1
2
B
q
=
(4)
which is called the quantization step size. Mean square
value of quantization error can be calculated as [5]:
/ 2
2 2
2 2 2
( )/2
1 2
12 3
q
B
e
q
q
E e e de
q
= = = =
}
(5)
where E denotes statistical expectation
3. - A/D CONVERTER
Fig. 4 shows the block diagram of a - A/D converter.
The 1-bit digital output from the modulator is supplied to
a digital decimation filter which yields a more accurate
representation of the input signal at the output sampling
rate of fs. In the figure is a first-order - modulator. It
consists of an analog difference node, an integrator, a 1-
bit quantizer (A/D converter), and a 1-bit D/A converter
in a feed- back structure. The modulator output has only
1-bit (two levels) of information, i.e., 1 or -1. The
modulator output y(n) is converted to x(t) by a 1-bit D/A
converter. The input to the integrator in the modulator is
the difference between the input signal x(t) and the
quantized output value y(n) converted back to the
I nt ernat i onal Journal of E mergi ng Trends & Technol ogy i n Comput er Sci ence (I JE TTCS)
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predicted analog signal, x(t). Provided that the D/A
converter is perfect, and neglecting signal delays, this
difference between the input signal x(t) and the fed back
signal x(t) at the integrator input is equal to the
quantization error. This error is summed up in the
integrator and then quantized by the 1-bit A/D converter.
Figure 4: Block Diagram of Sigma delta A/D converter
[8]
Although the quantization error at every sampling
instance is large due to the coarse nature of the two level
quantizer, the action of the - modulator loop is to
generate a 1 output which can be averaged over several
input sample periods to produce a very precise result. The
averaging is performed by the decimation filter which
follows the modulator.
4. FPGA IMPLEMENTATION OF - ADC
- ADC has been implemented using System Generator
tool of Xilinx, which reduces the design cycle. The
snapshot of the ADC has been shown in Fig 5.
Figure 5: Snapshot of the Set-Up
Fig. 6 shows the block diagram of ADC. ADC employs a
sequential binary search to arrive at the appropriate
output sample. The sample rate of the ADC is computed
as
( )
( ) ( )
( )
1
2 1 1
clk
sr
MSBI
mst
f
ADC
F MSBI
+
=
+ +
Samples/sec (6)
Where is the most significant bit of the DAC input
and is filter settle time constant.This approach
makes the ADC unsuitable for applications requiring a
high sampling rate. The 'ADC
Out
Register' captures the
reference shifter output after a sample has been computed,
then produces the ADC output sample. The 'ADC
Out
Register' captures the reference shifter output after a
sample has been computed, then produces the ADC
output sample. The 'ADC
Out
Register' captures the
reference shifter output after a sample has been computed,
then produces the ADC output sample. The 'ADC
Out
Register' captures the reference shifter output after a
sample has been computed, then produces the ADC
output sample.
4
sampl e
3
ADC_Sampl ed
2
ADCout
1
DAC_dri ver
z
-1
a
b
a = b
z
-1
rel1
z
-1
and
l 2
z
-1
and
l 1
z
-1
[a:b]
a
b
a = b
AgtR
shif t
mask
ref _shif ter
Reference
Shifter
shif t mask
Mask
Shi fter
[shi ft]
[shi ft] l oad
di n
en
--
FSTM
Counter
++
DAC Sampl e
Counter
0
0
d
en
q z
-1
ADCOut
Regi ster
0
'c1'
1
AgtR
Figure 6: Block Diagram of ADC
1
DAC_Out
d
rst
q z
-1
r1
[a:b]
a
b
a + b
Sigma Adder
d
rst
q z
- 1
a
b
a + b
Delta
Adder
0
0
0
hi
l o
}
hi
l o
}
1
DAC_In
Figure 7: Block Diagram of DAC
Fig. 7 shows block diagram of DAC. The Delta Adder
computes the difference between the current DAC input
and the current DAC output. The two concatenators
create a 16-bit output with the DAC output copied in the
two MSB positions. This effectively creates a difference
from the Delta Adder when the DAC output is 1. The
Sigma Adder accumulates the differences produced by the
Delta Adder by using 'r
1
' to storing the adder output on
each successive cycle. The MSB of the 'r
1
' output is sliced
off and provides the DAC output. The pulse string is
I nt ernat i onal Journal of E mergi ng Trends & Technol ogy i n Comput er Sci ence (I JE TTCS)
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registered before it is driven on the DAC output port. The
proposed ADC has been verified by converting its output
back to analog domain by using DAC. Table 1 shows the
comparison of ADC input and output of DAC. The results
show that the proposed design is quite accurate in
converting the analog data into digital data. The small
error shown in Table 1 may be a due to inaccuracy of
DAC.
Figure 8: Internal View of Top Level RTL Schematic of
Proposed ADC
The summary of the resources utilized has been shown in
table 2.
Table 2: Resource Utilization for Spartan 3E xc3s100e-
4vq 100Device
Device Utilization Summary (estimated values)
Logic Utilization Used Available Utilization
Number of Slices 38 960 3%
Number of Slice Flip Flops 55 1920 2%
Number of 4 input LUTs 45 1920 2%
Number of bonded IOBs 3 66 4%
Number of GCLKs 1 24 4%
5. CONCLUSION
This paper presents the design and implementation of
proposed - ADC. The proposed design has been
implemented on Spartan 3E xc3s100e-4vq 100 FPGA
Device and only consumes 38 number of slices, number of
55 Slice Flip Flops, 45 number of 4 input LUTs, 3
number of bonded IOBs and 1 number of GCLKs out of
available 960, 1920, 1920, 66 and 24 respectively. The
proposed design has also been verified for its accuracy.
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