Systems and methods for a partial sum digital fir filter

Electrical computers: arithmetic processing and calculating – Electrical digital calculating computer – Particular function performed

Reexamination Certificate

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Reexamination Certificate

active

06718355

ABSTRACT:

BACKGROUND OF THE INVENTION
1. Field of the Invention
The present invention relates generally to digital filters and, more particularly, relates to a partial sum digital FIR filter.
2. Background
In electrical systems, signals of interest are often corrupted by electrical noise and other forms of interference at unwanted frequencies. When this occurs, it is necessary to attenuate or eliminate the corrupting signals from the signal of interest. Electrical filters are used for this purpose.
FIG. 1
a
depicts a basic filter
100
. A desired signal
102
and an undesired signal
104
are combined to produce a signal
106
that is input to filter
100
. Filter
100
suppresses the undesired signal portion of signal
106
and passes only the desired signal portion
108
.
FIGS. 1
b
-
1
d
illustrate some of the basic forms filters can take.
FIG. 1
b
depicts a low pass filter
110
. A desired low frequency signal
112
is combined with an undesired high frequency signal
114
to produce a signal
116
that is input to low pass filter
110
. Low pass filter
110
passes only those signals having a frequency below a given cutoff frequency. Hence, filter
110
suppresses the undesired high frequency portion of combined signal
116
and passes only the desired low frequency signal
118
.
FIG. 1
c
shows a high pass filter
120
. An undesired low frequency signal
122
is combined with a desired high frequency signal
124
to produce a signal
126
that is input to high pass filter
120
. High pass filter
120
passes only those signals having a frequency above a given cutoff. Hence, filter
120
suppresses the undesired low frequency portion of combined signal
126
and passes only the desired high frequency signal
128
.
FIG. 1
d
portrays a bandpass filter
130
. Only those signals whose frequencies fall within a given frequency band are passed by filter
130
; signals whose frequencies are above or below the passband are attenuated. In this case, three signals
132
,
134
and
136
are combined to produce a signal
138
that is input to bandpass filter
130
. Signal
132
has a frequency below the passband of filter
130
; signal
134
has a frequency within the passband of filter
130
; and signal
136
has a frequency above the passband of filter
130
. Hence, filter
130
suppresses the high and low frequency portions
132
and
134
of signal
138
and outputs a signal
139
containing only the signal
134
within the filter passband.
The filters of
FIGS. 1
a
-
1
d
have traditionally been implemented using analog electric components such as resistors, capacitors and operational amplifiers. The signal being filtered is an electrical voltage or current corresponding to a physical quantity, such as a sound or video signal. Analog filters, while useful for many purposes, have several associated drawbacks. Since an analog filter is implemented by analog circuitry components, it can be changed only by redesigning or replacing the circuitry components. Moreover, the performance of analog filters, particularly those implemented with analog components, is typically temperature-dependent and subject to drift.
Advances in digital technology have led to the design and implementation of digital filters. Computer programs and/or processors can filter digital signals in the same way that analog filters filter analog signals. A digital filter is typically implemented as software or code running on a processor or controller to perform numerical calculations on sampled values of a signal. The processor may be a general-purpose microprocessor such as a PC microprocessor, or it may be a specialized DSP (Digital Signal Processor) chip. The high computational speeds possible with DSPs make them ideal candidates for implementation of digital filters. Alternatively, a digital filter can be implemented in digital logic hardware. Such hardware configurations can be dedicated or configurable, such as with a PAL or GAL. Digital filters have several advantages over analog filters. As software or code residing in the processor's memory determines the operation of a digital filter, the filter is programmable and can be easily changed or redesigned without changes in the circuitry or hardware. Digital filters are also more stable than analog filters and do not generally suffer from time and temperature induced variations.
A commonly used type of digital filter is a finite impulse response (FIR) filter. Each output of a FIR filter is the sum of a finite number of weighted samples of the input sequence. FIR filters are sometimes also referred to as feed-forward or non-recursive filters since all paths lead forward from the input to the output; no part of the filter output is fed back to the filter input. A common application of digital FIR filters is in video encoders for filtering or scaling input video signals from sources such as computers into forms suitable for display on devices such as television monitors.
A block diagram of a conventional FIR filter
150
is set forth in
FIG. 2
a
. Filter
150
, as do most FIR filters, comprises three basic building blocks: unit delays (
152
,
154
,
156
); multipliers (
162
,
164
,
166
,
168
); and adders (
172
,
174
,
176
). Input samples x[n] supplied to filter
150
are advanced through unit delays
152
,
154
and
156
. Typically, each unit delay imposes a delay of one clock cycle. Hence, on a clock cycle where the input sample is x[n], delay
152
outputs the previous input sample x[n−1]; delay
154
outputs the second previous input sample x[n−2]; and delay
156
outputs the third previous input sample x[n−3]. In a hardware or DSP implementation of a unit delay, an input sample value is stored in memory for one clock cycle and then released to the output. In filter
150
, for example, delays
152
,
154
and
156
may be implemented as three memory cells configured as a shift register.
The input sample x[n] and delayed samples x[n−1] . . . x[n−3] are input to multipliers
162
-
168
and multiplied by filter coefficients a[0] . . . a[3]. A FIR filter is said to comprise a “tap” for each input sample, i.e., x[n] . . . x[n−3], and each filter coefficient is referred to as a “tap coefficient”. Even if some the tap coefficients are “0”, the filter is still said to comprise the same number of taps as there are input signals. Hence, filter
150
is a 4-tap filter even if some of coefficients a[0] . . . a[3] are “0”. The values of filter coefficients a[0]-a[3] determine the response and characteristics of filter
150
, that is, the coefficient values determine the filter gain and frequency response. In a video encoder implementation, the coefficient values will determine the scaling and filtering performed on input data samples. From a mathematical standpoint, computation of the filter coefficients is often the most difficult and complex part of filter design. Several approaches for coefficient generation are familiar to those of ordinary skill in the art. Examples include the Fourier Series method, the Frequency Sampling method and the Remez Exchange Method.
The multiplied or weighted outputs of multipliers
162
. . .
168
are summed by adders
172
. . .
176
. The outputy[n] of filter
150
, then, can be expressed as:
y[n]=a[
0
]x[n]+a[
1
]x[n−
1
]+a[
2
]x[n−
2
]+a[
3
]x[n−
3].
More generally speaking, the output y[n] of an M+1 tap filter can be expressed as:
y

[
n
]
=

k
=
0
M

a
k

x

[
n
-
k
]
.
Hence, each output sample can be described as the current weighted input sample added to a predetermined number of previous and variously weighted input samples.
FIG. 2
b
illustrates a filter
180
that is an alternative implementation of filter
150
of
FIG. 2
a
. Filter
180
is a 4-ta

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