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Transcript of [IEEE 2013 26th Symposium on Integrated Circuits and Systems Design (SBCCI) - Curitiba, Brazil...
Hybrid Filter for High-Power Converter Systems
Guilherme H. K. MartiniProgress Rail Services
Curitiba, Brazil
Joao Alberto FabroDepartment of Informatics
Graduate School of Applied Computing (PPGCA)
Federal University of Technology - Parana
Curitiba, Brazil
Abstract---This paper presents a digital hybrid filter whichcan be applied to power electronics control systems. The filtersuppresses high-frequency noises while still providing a faststep response, its weighting algorithm is based on an open-loopcriteria that analyses the discrepancy between an Infinite ImpulseResponse (IIR) and a Moving Average (MA) filter. The filterperformance is compared to classical implementations througha step response and a Signal-to-Noise Ratio (SNR) analysis. Thecomputational cost is also evaluated to verify its use on hard real-time systems. To evaluate the proposed filter, experiments weremade on a microcontroller-based high-power frequency inverterwith hard-time constraints, ensuring its applicability.
Index Terms---Hybrid Filters, Embedded Software, DigitalSignal Processing, Power Electronics, Switching converters.
I. INTRODUCTION
Control systems that make use of high-power switching
drivers are very dependent on continuous voltage signals.
Those signals usually inform to the controller the conditions of
currents, voltages and temperatures of the operating equipment.
The information carried by the signals is critical to the correct
operation of the system and due to various noise sources those
signals are usually degraded. This degradation can be caused
by the driver switching output itself, by its switching power
supply, by its proximity to the load, and many other factors
that, depending on the application, are inevitable.
In order to keep the system operation adequate to its needs,
a filtering method is required. The ideal filter would attenuate
all high-frequency components while still having a good step
response. This would allow the system to respond quickly
to changes and to work with stability on its steady-state.
Frequency-domain based filters would give an exceptional
attenuation by using a Fast-Fourier Transform (FFT) to easily
detect the noise components but wouldn’t fit the tight time
requirements of the switching converters control loops, hence
the use of a time-domain based filter is required.
The direct use of moving average (MA) filters would give
a good white noise suppression, but, when the order of the
filter increases in an attempt to get a better attenuation, the
step response delay becomes larger. The MA filter is also
problematic when the signal contains narrowband noise, which
is exactly the one generated by the switching components.
Finite impulse response (FIR) filters would also need to have
too many delay stages to attenuate narrowband noise, thus
having the same disadvantage.
A low-pass infinite impulse response (IIR) filter would
deliver an even better noise attenuation but the step response
would be a drawback. As the step response time and the
filtering attenuation are trade-offs, a change on its frequency
response is needed to allow the system to have a better step
response.
Time constraints and the previous knowledge of the noise
characteristics are the main bottlenecks to the development
of such hybrid methods. Recent papers show that hybrid and
adaptive filtering methods are being used in a wide range
of applications. For example, an adaptive weighted moving
average (WMA) algorithm is proposed in [1] to clean the
channel noise generated in a wireless sensor network. Statistical
filters are applied allowing the change of the filter coefficients
depending on the noise spectrum in [2]. Warm-up periods of
MA filters are evaluated in [3], indicating that certain time-
critical applications must be developed with special care. In
the same area of application, [4] presents a FIR filter for delay-
constrained applications showing that a flexible filter can have
a better step response than the traditional moving average filter.
Hybrid filters can be used to suppress the noise generated
by non-linear loads on power applications, as presented in
[5], that uses passive and active elements that form a hybrid
filter to damp undesired harmonics in a shunt compensation
application. Harmonics elimination over voltage signals on
power applications using adaptive and hybrid filters can also
be verified on [6] and [7]. Examples of adaptive filtering being
applied in high-power applications can also be found in [8], [9]
and [10]. Implementations that have a prediction step phase
such as the Kalman filter are also common [11] but those
are not suited for this work because of time constraints of
the application. Predictive filters have a higher computational
cost generated by the mathematical models which attempts to
reproduce the plant behavior.
The development of a simple hybrid filter that has a good
step response while still attenuating high-frequency noises
is the main purpose of this work. Every stage of the filter
is presented in a general structure allowing its use in any
application that has the same requirements. The computational
cost is shown and compared to traditional techniques to allow
its applicability in hard real-time applications. Empirical data is
retrieved from an application of the filter in a microcontroller-
based AC-AC (alternate current) frequency inverter to analyze
978-1-4799-1132-5/13/$31.00 c© 2013 IEEE
978-1-4799-1132-5/13/$31.00 ©2013 IEEE
3ϕ
Input
Full Bridge
Diode
Rectifier
μC-Based
Control
Board
IGBT-Based
3ϕ Source
Inverter
DC Bus
Output
toLoad
System
Set-Points
Output to
Actuator
Feedback
Sensor Data
Feedback Sensor Data
Fig. 1. 3-phase AC-AC frequency inverter structure.
its final performance. Those frequency-inverters are already
being used on a locomotive fleet running on the southeastern
part of the United States. They drive traction motor blowers
and air compressors, generate 220V AC to peripherals on
each locomotive and produce power to the cabin heater/air
conditioning system.
The second section of this paper provides a brief description
of the equipment in which the filter was applied and the filter
design structure along with every limiting factor that affected
its conception. The third section describes the implementation
of the filter and presents its results when applied to the
frequency inverter. A comparison with other classical filter
implementations is also made in this section. The fourth section
concludes the paper.
II. FILTER DESIGN REQUIREMENTS EVALUATION AND
CONCEPTION
The filter design method was defined to meet the application
constraints. Those constraints are usually the same or very
similar for every power converter equipment: fast control loops
that make complex algorithms very cumbersome; inevitable
noise generated by the equipment itself; the need to have a good
steady-state regulation to increase the equipment efficiency;
and a good step response to allow the control to respond well to
unexpected plant variations, increasing the equipment lifetime,
security and reliability.
The testing equipment is a 3-phase AC-AC frequency
inverter consisting of a full rectifying bridge, a microcontroller-
based control board and a IGBT-based (Insulated Gate Bipolar
Transistor) three-phase voltage source inverter as seen in figure
1. A group of sensors and transducers are also included in the
equipment allowing a closed loop control that is performed by
a RISC (Reduced Instruction Set Computer) microcontroller
running at 88Mhz.
The process control block consists of a plant, that in most
applications of the equipment is an AC induction motor, a
control transfer-function and a filter transfer-function. The filter
transfer-function samples analog signals from the plant, filters
them and sends its information to the control transfer-function.
The main scope of this work is to deliver a reliable feedback to
the control by developing this filter transfer-function. The filter
block is evaluated only by its inputs and outputs, verifying
the gain that this block provides to the whole process. The
filter’s transfer-function is easily modifiable because it can
be developed by software after the analog-to-digital converter
(ADC) acquisition in the microcontroller, allowing progressive
improvements without any change on its physical structure
(which is not possible to the equipments that are already running
on field).
The control-loop time is constant and set to 1 millisecond
allowing its actuation to meet the equipment application
requirements. Every new control loop requires new information
from five different sensors to verify the actual output state and
calculate the new system actuator output. The five cited sensors
measure the following signals: The DC input voltage after the
rectifying bridge, the three AC output currents and the IGBTs
heat sink temperature. Besides the heat sink temperature, all
measurements are time-critical and need to be updated before
the execution of every new control iteration. Knowing this
constraint, it is set that the time window given to the filter
stage to produce the sensor outputs is up to a quarter of the
control-loop time, i.e. 250 microseconds, thus 50 microseconds
per sensor. The temperature sensor is sampled with the same
frequency of the other sensors for a better measurement as it is
a low-power input. The remaining time of the loop is used to
perform the control loop, update the actuators outputs, sample
the analog inputs, receive its set-points through industrial
communications protocols and inform its actual state through
the same industrial network.
A common method used to reduce white noise from analog
inputs is to oversample them. Using a moving average stage
before the actual filter with an oversampled input would fit
easily in the time constraints cited above and would result
in a very good final result if not for one shortcoming: The
ADC conversion must be done when the output IGBTs are
not switching states to avoid spurious samples caused by
electromagnetic interference. The switching pattern used for
switching the output allows the microcontroller to sample
all analog inputs every 1 millisecond in noise-free moments,
this would be enough to change the control loop input in
every iteration but it wouldn’t be enough to oversample the
inputs. Changing the switching pattern would provide a larger
sampling time but would also result in a poorer output quality
and controllability. This leads to the conclusion that changing
the sampling time would not be a good approach to the problem.
The step response time is another constraint that must be
observed. Differently from the time constraint, it is hard to
estimate what is the maximum allowable settling time. The
closer that the filter responds to the step, the better would be
the control adaption. On the other hand, a fast step response
filter would probably have a poor noise attenuation. The same
happens to the noise attenuation criteria, there is no optimal
or desired value for it. With that defined, it was stipulated that
the evaluation of the filter step response and noise attenuation
would be made by comparing its values with a moving average
filter of order 20. From previous experiments it is possible to
know that the MA filter of order 20 provides a satisfactory
Stage IIIStage IIStage I
ADC
Spurious
Samples
Rejection
20th order
MA Filter
5th order
Chebyshev
Filter
Weighting
Stage
ADCSampleInput
OutputtoControl
Fig. 2. Block Diagram of the proposed filter.
step response but poor noise attenuation, specially narrowband
noise.
After defining every limitation of the filter application, it
is possible to describe the filter structure. The block diagram
of the filter is shown in figure 2. It is composed of three
consecutive stages, where the first one requests ADC samples
of every analog input of the system. The main function of the
first stage is to reject spurious samples from the ADC that are
caused by electromagnetic interference. This type of noise is
present even when sampling is synchronized with the switching
instants. The first step of this stage is to double-sample every
input, meaning that whenever a sample is requested, two
instantly consecutive ADC conversions of the same signal
are made. For example, when sampling an input at 1KHz,
two ADC conversions are made one right after another. The
time between those two samples is determined only by the
ADC conversion delay, that is around 8μs. After around 1
millisecond, two consecutive ADC conversions are once again
performed. After this sampling step, both samples are compared
and their difference is calculated. If the difference is below a
stipulated value, the mean value of the two samples is directed
to the stage output. If the difference is above that value, the last
valid conversion is placed as the stage output. This mechanism
can be seen as a high-frequency narrowband noise attenuator
that works between the two samples that were acquired, being
a simplified oversampling method that performs well due to
the low probability of occurrence of two consecutive spurious
samples. The output results can be seen in the next section
where this stage has been proven useful while still fitting the
time requirements.
The second stage consists of two parallel classical filters
with different transfer functions. One of them is a moving-
average filter that has 20 delay stages. The other one is a
Chebyshev-IIR low-pass filter of the 5th order. The purpose
of the two filters is to provide the third stage two different
inputs with different characteristics of the same signal. The
moving average filter responds better to step responses while
still benefits from the narrowband attenuation of the first filter.
The IIR filter provides a very slow step response but provides a
very good noise attenuation when the signal is steady. The 20-
delay MA filter was chosen due to its previous implementation
at the same application. By inspection of data retrieved from
the frequency inverter, its was possible to verify that its step
response was adequate. Its noise suppression capability was
not optimal, but still significant while on transients. The IIR
filter was designed to deliver the largest possible attenuation at
stopband frequencies, because of that the Chebyshev topology
was chosen. The order was determined by its execution delays:
Bigger orders increased execution time but did not deliver a
much better noise attenuation.
The third and last stage is the weighting part of the filter.
Using both filters from the second stage as inputs, a weighted
average of both is provided in its output. The weighting method
varies depending on the difference of the two inputs. If the
MA filter is providing a value that is too different from the IIR
filter, the weights favors the moving average aiming to deliver
a better step response. If not, the filter favors the IIR filter
to maximize noise attenuation. Besides the weighting process,
the triggering of the change of weights can only happen when
some threshold values are surpassed. This trigger mechanism
prevents MA filter from being favored over the IIR filter when
the output of the MA filter is noisy, allowing the weighting
algorithm to differ a step from random noise. The fine tuning
of the third stage is done empirically to ensure its applicability
on the frequency inverter.
With the requirements described and the filter design
structure defined, simulations and practical tests were realized
to verify its performance. The evaluation metrics and its
results are shown in the next section. Also, details about every
parameter used in the whole process along with simulations of
frequency responses, timings and empirical results are given.
III. FILTER IMPLEMENTATION AND RESULTS
For the implementation of the filter, a simulation step was
made previous to its application to allow a faster tuning of
every stage of the filter and to grant its stability before applying
it to the equipment. Three different filters were developed: a
20 delay stages MA, a 32 delay stages low-pass FIR with the
Blackman window function (for comparative purposes) and a
low-pass 5th order Chebyshev IIR filter. As the hybrid filter
uses the MA filter and the IIR in its second stage, no further
development in the simulation step is needed for its validation.
The trade-off between passband ripple and stopband gain of
the Chebyshev topology is assumed to be well suited to this
application. The criteria used to determine the zero and pole
placing of the IIR filter was to minimize the gain in stopband
frequencies.
The MA filter discrete-time transfer function 1 indicates that
that its output gain is always set to 1, exactly like the FIR
filter transfer function 2, that uses the normalized Blackman
window function 3 to produce the same gain.
F (z) =
20∑
n=0
1
21z−n (1)
F (z) =
32∑
n=0
w(n)z−n (2)
TABLE IIIR FILTER COEFFICIENTS
n/p Feedforward (bn) Feedback (ap)
0 3.35×10−8 1
1 1.68×10−7 -4.95
2 3.35×10−7 9.84
3 3.35×10−7 -9.80
4 1.68×10−7 4.90
5 3.35×10−8 -0.98
0 100 200 300 400 500-450
-400
-350
-300
-250
-200
-150
-100
-50
0
Frequency [Hz]
Mag
nit
ud
e [
dB
]
(a)
(b)
(c)
Fig. 3. Frequency response of the modeled filters. (a) IIR response. (b) FIRresponse. (c) MA response.
w(n) =7938
18608−
9240
18608cos(
2πn
31) +
1430
18608sin(
4πn
31) (3)
The IIR filter discrete-time transfer function 4 uses the
feedforward an feedback gains that are described in table I.
By looking at the feedforward gain values it is possible to
conclude that the step response is very slow due to the very
small input gains.
F (z) =
5∑n=0
bnz−n
5∑p=0
apz−p
(4)
With the discrete-time transfer functions modeled it was
possible to determine the filters frequency responses, phase
delays and stability. The frequency response can be seen in
figure 3. All responses are plotted in the same graph so the
magnitude response of every filter for a same frequency can
be easily compared. It is clear that the IIR response is much
more adequate for noise suppression purposes while the MA
would still provide the best step response.
The phase delay of the filters were not evaluated at the
simulation step as their temporal responses are much more
important and are examined further in the development process.
The FIR and MA filters are inherently stable and a pole-zero
plot of the IIR filter would show that every pole is inside
the unit circle of the Z plane, being also stable. The stability
was tested after the truncation of the filter coefficients for
fixed-point use.
0 100 200 300 400 500-450
-400
-350
-300
-250
-200
-150
-100
-50
0
Frequency [Hz]
Mag
nit
ud
e [d
B] (a)
(b)
Fig. 4. Variable frequency response of the proposed filter. (a) IIR filterthreshold. (b) MA filter threshold.
TABLE IIFILTERS ITERATION TIME
Filter Iteration Time
Moving Average 2μs
FIR 9μs
IIR 28μs
Hybrid 48μs
The spurious samples rejection stage of the hybrid filter is
directly validated with empirical data due to its implementation
simplicity. The weighting stage, which is also validated
empirically, has a final output gain that is always 1, so if
the input filters of this stage are stable, the final output is
also stable. The weighted average between both inputs always
calculates the gain of the MA filter that is between 0 and 1. The
IIR gain receives the remainder, 1 minus MA gain. That hybrid
filter algorithm leads to a final frequency response which is
between the MA response and the IIR response, the variation
is represented by the gray area in figure 4. The threshold value
for triggering the weighted average calculation was determined
empirically, it was based on the noise magnitude observation
and is set to permit calculations only when the inputs differ in
more than 5% of the sensor range in 25 consecutive samples.
If this does not happen, the MA input receives gain 0 and the
IIR input receives gain 1, producing a good noise attenuation
output.
After simulating, the filters were implemented in C, de-
bugged with a random noise generator code, and ported to
the application. With the filters running at the microcontroller,
their iteration times could be verified, being then possible to
evaluate if they fit in the time constraint, which was previously
determined to be 50μs. Table II shows the time consumed by
every filter to calculate its outputs.
The timing verification was made with an oscilloscope.
It measured one microcontroller digital output which was
switched at the beginning and end of every filter execution,
resulting in a square wave. The table data shows that the MA
filter is very time efficient because it can be implemented with
3 operations. The FIR filter needs to recalculate every new filter
gain when a new input is received thus having an increased
Computer
Delta-Connected
Inductive Load
Frequency
Inverter
Human
Interface
3ϕ
Input
3ϕ Output
Set-Point Filtering Data
Fig. 5. Hardware test bench set to acquire empirical data.
iteration time. The IIR filter would have the same iteration time
of a 12th order FIR (being faster than the implemented FIR
filter) if their input/output gain ratio was not so small. That
difference requires some extra operations to keep its precision
when implemented in fixed-point architectures, which is the
case. The hybrid filter iteration time is the sum of all processes
involved in all filter stages: 8μs used to take a second sample
from an analog input, 2μs to calculate the first filter stage,
28μs to iterate the IIR filter of the second stage, 2μs to iterate
the MA filter of the second stage and 8μs to calculate the third
stage. That extra sample time could have been removed from
the calculation as it is not part of the filter. It was decided to
add this time to the hybrid filter because its first stage requires
more data than the other filters, thus this extra-sampling is a
requirement of the filter, being only executed because of the
filter needs.
The timing constraints were met by every filter implementa-
tion. The next step in the development was to retrieve empirical
results of the noise filtering. To perform that task, a test bench
was set both in software and hardware. The software test bench
removed the control loop from the equipment, leaving only all
the filter implementations working in parallel so the filters can
produce a different output from the same input, making it easier
to generate benchmark data. A simple communication protocol
was also left in the code so an external computer could acquire
data from the equipment and control its outputs. The hardware
test bench consists of a computer used to retrieve data and
command the equipment output, the frequency inverter, a 220V
AC voltage source and inductors connected in delta as the load.
Figure 5 shows the set-up built.
The first test evaluated the input and output of the first stage
of the hybrid filter. The data acquisition refers to the voltage
input of the equipment which was sampled when the system
was producing 300A, 200V AC in every phase output. It is
known that the system produces more noise when the output
power is higher, justifying the need to excite the load. Figure
6 shows the gain of the first stage of the filter. It is possible
to determine graphically that the first stage produces a very
good result suppressing spurious samples.
A second test evaluated the final performance of all filters in
terms of step response and noise attenuation. Using the same
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2940
2950
2960
2970
2980
2990
3000
0.00 100.00 200.00 300.00 400.00 500.00 600.00 700.00
Volt
age
[dV
]
Time [ms]
(a)
(b)
Fig. 6. Results of the first stage of the hybrid filter narrowband noise rejection.(a) Filter input. (b) Output of the first stage of the filter. The stage outputindicates that the proposed method suppresses most of the undesired samples,providing a better input signal for the next stage.
analog input as reference for the comparison, the step was
generated by changing the input voltage with a contactor that
opens and closes the power input line of the equipment. The
noise attenuation test was made with the same conditions of
the previous tests, 300A and 200V AC in every phase output.
The results are shown in figures 7 and 8.
The step response chart shows that the hybrid filter takes a
little longer than the MA filter to start converging to its new set-
point, this happens because of the decision process of changing
the weights, which expects the difference between the MA
input and the IIR input to be bigger than 5% in 25 consecutive
samples. This mechanism is used to certify that random noise
won’t make the third stage detect an input step spuriously,
after the step is detected, the hybrid filter reproduces the MA
output just before it reaches the new set-point. As the MA is
a 20-delay filter, it is convenient to use all its settling time to
certify that a step really occurred.
The FIR filter takes a little longer to converge, which is
justified by its bigger amount of delay stages and for its smaller
gains on the most recent samples that run through the filter. The
IIR filter is slower that the others, taking around 500ms to settle.
It is noticeable that the Chebyshev signal has a ripple during
its settling time, this is due to its 8dB passband attenuation
and is not considered an issue to this application. After the
step, the third stage of the filter gradually returns the gain of
the weighted average to the IIR filter. This happens linearly
on time and only when the difference between the inputs is
smaller than 5%, ensuring that the step phase is over.
The steady state of the filter worked as expected, the gain
of the weighting stage is completely bound to the IIR filter
input, making their responses identical. It is also possible to
verify the problems caused by narrowband noise in the FIR
and MA filters outputs.
Table III quantifies the results shown in figures 7 and 8. The
0
500
1000
1500
2000
2500
3000
3500
10 30 50 70 90 110 130 150 170
Vo
lta
ge [d
V]
Time [ms]
(a)
(b)
(c)
(d)
(e)
(f)
Fig. 7. Step response the filters. (a) Input voltage of the equipment, inputsignal for all filters. (b) MA filter response. (c) FIR filter response. (d) IIRfilter response, reached its new set-point on 500ms. (e) Hybrid filter response.(f) IIR gain at the third stage of the hybrid filter, the range is from 0 to 1000,meaning a weight multiplier that varies from 0 to 1.000. The MA gain is1000-IIR gain.
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0.00 200.00 400.00 600.00 800.00 1000.00 1200.00
Vo
lta
ge
[dV
]
Time [ms] (a)2860
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2920
0.00 200.00 400.00 600.00 800.00 1000.00 1200.00
Vo
lta
ge
[dV
]
Time [ms] (b)
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2890
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0.00 200.00 400.00 600.00 800.00 1000.00 1200.00
Vo
lta
ge
[dV
]
Time [ms] (c)2860
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2890
2900
2910
2920
0.00 200.00 400.00 600.00 800.00 1000.00 1200.00
Vo
lta
ge
[dV
]
Time [ms] (d)
Fig. 8. Noise attenuation results in a noisy environment. The frequencyinverter was producing 100kW in its output. (a) Steady state voltage inputsampled by the equipment. (b) MA filter output. (c) FIR filter output. (d)IIR/Hybrid filters identical outputs.
signal to noise ratio was calculated by dividing the mean signal
value by its standard deviation (5). This is a good approach
to calculate the noise over a DC signal because it provides a
very accurate dispersion measurement, which is dimensionless.
SNR =μ
σ(5)
The step response time was calculated from the moment
that the step occurred until the moment that the signal reaches
97% of its set-point.
IV. CONCLUSION
Simulations and empirical data show that the proposed
hybrid filter is capable of yielding a good noise attenuation
while still providing a satisfactory step response, even in
noisy environments. The proposed approach is suitable for
applications that operate with very strict time limits. Its basic
TABLE IIIFILTERS STEP RESPONSE TIME AND STEADY STATE NOISE ATTENUATION
Filter Step Response Time SNR
Input Signal ---- 140.83
Moving Average 16ms 343.69
FIR 28ms 304.82
IIR 496ms 704.01
Hybrid 16ms 704.01
topology relies on the fact that the weighting stage of the filter
receives an input that responds well to steps and another that
has a good noise attenuation.
The filter was tested on a frequency inverter that already runs
on 6 American locomotives. Each one provides up to 100kW
of AC power (every locomotive has 3 inverters installed)
that drives blower/air compressor motors, generates 200V
AC to auxiliary systems and produces power to the cabin
heater/air conditioning system. They also provide power to
the battery charging equipment that together with the blowers
and compressors are critical to the locomotive operation. It is
expected that this filter will make the inverters safer and more
energy-efficient.
Some further development can be made to improve noise
attenuation depending on the IIR filter used, other applications
can use this filter topology and adapt it to solve problems
related to electromagnetic interference caused by its switching
converters and rotating loads.
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