OCT :2 0 1991
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`5825898
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`, N
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`rJ
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`ISSUE FEE IN ALE
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`""""' NO.
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`T"'"
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`Label
`Area
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`RTL898_1020-0001 Realtek 898 Ex. 1020
`(FACE) TG
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`..•
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`CONTENTS
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`-fIIo:,Md
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`r-I{P14
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`14. :!~ _ .:r0
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`13-
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`Glint OCT 2 01998
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`_ __ ,_31. _________ ..:.....-______ _
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`__ ~~\.---32. ____ ..:.....-____ _
`RTL898_1020-0002
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`(FRONT)
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`

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`POSITION
`CLASS4AER
`EXAMINER
`TYP:tST
`VERIFIER
`CORPSCORR.
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`-
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`INDEX OF CLAIMS
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`RTL898_1020-0003
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`(LEFT INSIDE)
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`RTL898_1020-0005
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`RTL898_1020—0006
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`RTL898_1020-0006
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`RTL898_1020-0006
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`

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`BAR CODE LABEL
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`U.S. PATENT APPLICATION
`
`SERIAL NUMBER
`
`08/672,899
`
`FILING DATE
`
`ClASS
`
`GROUP ART UNIT
`
`06/27/96
`
`3JJ
`
`2502
`
`JOSBPH MARASH, HAIPA, ISRAEL •
`
`•• CONTINUING DATA··.········.····.····
`VBRIFI!D
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`•• FOREIGN/PCT APPLICATIONS ••••••••••••
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`••••• SMALL ENTITY·····
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`ILX
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`14
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`PENNIB AND EDMONDS
`1155 AVBNUB OF THB AMERICAS
`NEW YORk NY 10036-2711
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`$899.00
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`8797-003
`
`SYSTBM AND KBTHOD FOR ADAPTIVE INTERFERENCE CANCBLLING
`
`This is to certitv that annexed hereto is a true copy from the records of the United States
`Patent and Tradamark Office of the application whICh is identified abolla.
`BV authority of lhe
`COMMISSIONER OF PATENTS AND TRADEMARKS
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`"""
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`Certifying Officer
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`RTL898_1020-0007
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`08/ 672899
`u.s. DEPARTMENT OF COMMERCE
`PATENT AND TRADEMARK OFFICE
`FEE RECORD SHEET
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`260 VC 16-1150 07/10/96 086]2899
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`PTO-1556
`(5/87)
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`RTL898_1020-0008
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`.. '~. . en
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`• 08/ 672899
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`\ ttbllHfE & EDMONDS
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`New Volt, N.Y. 10036-2111
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`AlTORNEY DOCKET NO . ....I!871J9!J7~-OO1&l3'-______ ____ . _____ Date June 2]. 1926
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`Assistant Commissioner for Plilents
`W'/.~~~~202J1
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`Th .... 'U()wi.a utility patenl application is enclosed for tiling:
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`SYSTEM AND METHOD FOR ADAPTIVE INTERFERENCE CANCELLING
`Pages of Specification 32 + ABSTRACT
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`Sheets of Drawin&s 14
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`This form is not tor use with re-issuc, design or plant pah:nt applications.
`
`RTL898_1020-0009
`
`

`
`15
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`BACKGROUND OF DIE llIVEIITION
`The present invention relates generally to 8ign~1
`processing, and more specifically to an adaptive signal
`system and method for reducing interference in a
`signal.
`There are many instances where it is desirable to have a
`capable of receiving an information signal from a
`particular signal source where the environment includes
`10 sources of interference signals at locations different from
`that of the signal source . one Buch instance is the use of
`microphones to record a particular party' s speech in a room
`where there are other parties speaking simultaneously,
`causing interference in the received aignals.
`If one knows the exact characteristics of the
`interference, one can use a fixed-weight filter to suppress
`it. But it is often difficult to predict the exact
`characteristics of the interference because they may vary
`according to changes in the interference sources, the
`20 background noise, acoustic environment, orientation of the
`sensor with respect to the signal source, the transmission
`paths from the signal source to the sensor, and many other
`factors . Therefore, in order to suppress such interference,
`an adaptive system that can change its own parameters in
`25 response to a changing environment is needed.
`An adaptive filter is an adaptive system' that can change
`its own filtering characteristics in order to produce a
`desired response. Typically, the filter weights defining the
`characteristics of an adaptive filter are continuously
`30 updated so that the difference between a signal representing
`a desired response and an output signal of the adaptive
`filter is minimized .
`The use of adaptive filters for reducing interference in
`a received signal has been known in the art as adaptive noise
`3S cancelling .
`It is based on the idea of cancelling a noise
`component of a received signal from the direction of a signal
`source by sampling the noise independently of the source
`
`RTL898_1020-0010
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`

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`•
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`•
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`signal and modifying the sampled noise to approximate the
`noise component in the received signal using an adaptive
`filter. For a seminal article on adaptive noise cancelling,
`see B. Widrowet al .• Adaptive Noise Cancelling: Principles
`5 and Applications, Proe. IEEE 63:1692-1716, 1975.
`A basic configuration for adaptive noise cancelling has
`a primary input received by a microphone directed to a
`desired signal source and a reference input received
`independently by another microphone directed to a noise
`10 source. The primary input contains both a source signal
`component originating from the signal source and a noise
`component originating from the noise source. The noise
`component is different from the reference input representing
`the noise source itself because the noise signal must travel
`15 from the noise source to the signal source in order to be
`included as the noise component.
`The noise component, however, is likely to have some
`correlation with the reference input because both of them
`originate from the same noise source. Thus, a filter can be
`20 used to filter the reference input to generate a cancelling
`signal approximating the noise component. The adaptive
`filter does this dynamically by generating an output signal
`which is the difference between the primary input and the
`cancelling signal, and by adjusting its filter weights to
`25 minimize the mean-square value of the output signal. When
`the filter weights settle, the output signal effectively
`replicates the source signal substantially free of the noise
`component because the cancelling signal closely tracks the
`noise component .
`Adaptive noise cancelling can be combined with
`beamforming, a known technique of using an array of sensors
`to improve reception of signals coming from a specific
`direction. A beamformer is a spatial filter that generates a
`single channel from multiple channels received through
`35 multiple sensors by filtering the individual multiple
`channels and combining them in such a way as to extract
`signals coming from a specific direction . Thus, a beamformer
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`30
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`- 2 -
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`RTL898_1020-0011
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`

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`•
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`can change the direction of receiving sensitivity without
`physically moving the array of sensors. For details on
`beamforming, see B.D. Van Veen and K.M. Buckley, Beamforming:
`A Versatile Approach to Spatial Filtering, IEEE ASSP Mag.
`55(2), 4-24.
`Since the beamformer can effectively be pointed in many
`directions without physically moving its sensors, the
`beamformer can be combined with adaptive noise cancelling to
`form an adaptive beamformer that can suppress specific
`10 directional interference rather than general background
`noise. The beamformer can provide the primary input by
`spatially filtering input signals from an array of sensors so
`that its output represents a signal received in the direction
`of a signal source. Similarly, the beamformer can provide
`15 the reference input by spatially filtering the sensor signals
`so that the output represents a signal received in the
`direction of interference sources. For a seminal article on
`adaptive beamformers, see L.J. Griffiths & C.W. Jim, An
`Alternative Approach to Linearly Constrained Adaptive
`20 Beamforming. IEEE Trans. Ant. Prop. AP-30:27-34, 198 2 .
`One problem with a conventional adaptive beamformer is
`that its output characteristics change depending on input
`frequencies and sensor directions with respect to
`interference sources. This is due to the sensitivity of a
`25 beamformer to different input frequencies and sensor
`directions. A uniform output behavior of a system over all
`input frequencies of interest and over all sensor directions
`is clearly desirable in a directional microphone system where
`faithful reproduction of a sound signal is required
`30 regardless of where the microphones are located.
`Another problem with adaptive beamforming is "signal
`leakage ll
`• Adaptive noise cancelling is based on an
`assumption that the reference input representing noise
`sources is uncorrelated with the source signal component in
`35 the primary input, meaning that the reference input should
`not contain the source signal. But this "signal free"
`reference input assumption is violated in any real
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`3 -
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`RTL898_1020-0012
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`

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`•
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`•
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`environment. Any mismatch in the microphones (amplitude or
`phase) or their related analog front end, any reverberation
`caused by the surroundings or a mechanical structure, and
`even any mechanical coupling in the physical microphone
`5 structure will likely cause "signal leakage" from the signal
`source into the reference input.
`If there is any correlation
`between the reference input and the source signal component
`in the primary input, the adaptation process by the adaptive
`filter causes cancellation of the source signal component,
`10 resulting in distortion and degradation in performance.
`It is also important to confine the adaptation process
`to the case where there is at least some directional
`interference to be eliminated. Since nondirectional noise,
`such as wind noise or vibration noise induced by the
`15 mechanical structure of the system, is typically uncorrelated
`with the noise component of the received signal, the adaptive
`filter cannot generate a cancelling signal approximating the
`noise component.
`Prior art suggests inhibiting the adaptation process of
`20 an adaptive filter when the signal-to-noise ratio (SNR) is
`high based on the observation that a strong source signal
`tends to leak into the reference input . For example, U.S.
`Pat. No. 4 , 956,867 describes the use of cross-correlation
`between two sensors to inhibit the adaptation process when
`25 the SNR is high.
`But the ' prior art approach fails to consider the effect
`of directional interference because the SNR-based approach
`considers only nondirectional noise , Since nondirectional
`noise is not correlated to the noise component of the
`30 received signal, the adaptation process searches in vain for
`new filter weights, which often results in cancelling the
`source signal component of the received signal.
`The prior art approach also fails to consider signal
`In
`leakage when the source signal is of a narrow bandwidth .
`35 a directional microphone application, the source signal often
`contains a narrow band signal, such as speech signal, with
`its power spectral density concentrated in a narrow frequency
`
`- 4 -
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`RTL898_1020-0013
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`

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`•
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`•
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`range. When signal leakage occurs due to a strong narrow
`band signal, the prior art approach may not inhibit the
`adaptation process because the overall signal strength of
`such narrow band signal may not high enough. The source
`5 signal component of the received signal is cancelled as a
`result, and if the source signal is a speech signal,
`degradation in speech intelligibility occurs.
`Therefore, there exists a need for an adaptive system
`that can suppress directional interference in a received
`10 signal with a uniform frequency behavior over a wide angular
`distribution of interference sources.
`
`SUMMARy OF THE INVENTION
`Accordingly, it is an object of the present invention to
`15 suppress interference in a received signal using an adaptive
`filter for processing inputs from an array of sensors.
`Another object of the invention is to limit the
`adaptation process of such adaptive filter to the case where
`there is at least some directional interference to be
`20 eliminated.
`A further object of the invention is to control the
`adaptation procesB to prevent signal leakage for narrow band
`signals.
`Another object is to produce an output with a uniform
`25 frequency behavior in all directions from the sensor array.
`These and other objects are achieved in accordance with
`the present invention, which uses a system for processing
`digital data representing signals received from an array of
`sensors. The system includes a main channel matrix unit for
`30 generating a main channel representing signals received in
`the direction of a signal source where the main channel has a
`source signal component and an interference signal component.
`The system includes a reference channel matrix unit for
`generating at least one reference channel where each
`35 reference channel represents signals received in directions
`other than that of the signal source. The system uses
`adaptive filters for generating cancelling signals
`
`- 5
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`RTL898_1020-0014
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`•
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`10
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`approximating the interference signal component of the main
`channel and a difference unit for generating a digital output
`signal by subtracting the cancelling signals from the main
`channel. Each adaptive filter has weight updating means for
`5 finding new filter weights based on the output signal. The
`system includes weight constraining means for truncating the
`new filter weight values to predetermined threshold values
`when each of the new filter weight value exceeds the
`corresponding threshold value.
`The system may further include at least one decolorizing
`filter for generating a flat-frequency reference channel.
`The system may further include inhibiting means for
`estimating the power of the main channel and the power of the
`reference channels and for generating an inhibit signal to
`15 the weight updating means based on normalized power
`difference between the main channel and the reference
`channels.
`The system produces an output substantially free of
`directional interference with a uniform frequency behavior in
`20 all directions from the system.
`The objects are also achieved in accordance with the
`present invention using a method, which can readily be
`implemented in a program controlling a commercially available
`nsp processor.
`
`2S
`
`30
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`BRIEF DESCRIPTION OF THE DRAWINGS
`The objects, features and advantages of the present
`invention will be more readily apparent from the following
`detailed description of the invention in which:
`~~ is a block diagram of an overall system;
`~G~ is a block diagram of a sampling unit;
`,
`FIG ) .3 is a block diagram of an alternative embodiment
`of a s~pling unit;
`.fIG:~~ is a schematic depiction of tapped delay lines
`35 used in~ main channel matrix and a reference matrix unit;
`FIG.'S is a schematic depiction of a main channel matrix
`unit;
`
`- 6 -
`
`RTL898_1020-0015
`
`

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`•
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`•
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`FI~ is a schematic depiction of a reference channel
`
`5
`
`,
`matrix ~it;
`FI?~ is a schematic depiction of a decolorizing
`filter; ~ ,
`F'~G'~ is a schematic depiction of an inhibiting unit
`based on directional interference;
`FIG.
`is a schematic depiction of a frequency-selective
`con8tr~t adaptive filter;
`FI~.~O is a block diagram of a frequency-selective
`10 weight-~n~traint unit;
`FIG.~l is a flow chart depicting the operation of a
`program that can be used to implement the invention.
`
`15
`
`PETAlLED DESCRIPTION OF THE INVENTION
`FIG. 1 is a block diagram of a system in accordance with
`a preferred embodiment of the present invention. The system
`illustrated haa a sensor array 1, a sampling unit 2, a main
`channel matrix unit 3, a reference channel matrix unit 4, a
`set of decolorizing filters 5, a set of frequency-selective
`20 constrained adaptive filters 6, a delay 7, a difference unit
`8, an inhibiting unit 9, and an output O/A unit 10.
`Sensor array 1, having individual sensors la-ld,
`receives signals from a signal source on-axis from the system
`and from interference sources located off-axis from the
`2S system.
`The sensor array is connected to sampling unit 2
`for sampling the received signals, having individual sampling
`elements, 2a-2d, where each element is connected to the
`corresponding individual sensor to produce digital signals
`11.
`
`30
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`35
`
`The outputs of sampling unit 2 are connected to main
`channel matrix unit 3 producing a main channel 12
`representing signals received in the direction of a source.
`The main channel contains both a source signal component and
`an interference signal component.
`The outputs of sampling unit 2 are also connected
`reference channel matrix unit 4, which generates reference
`channels 13 representing signals received from directions
`
`- 7 -
`
`RTL898_1020-0016
`
`

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`•
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`•
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`10
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`other that of the signal source. ThuB, the reference
`channels represent interference signals.
`The reference channels are filtered through decolorizing
`filters 5. which generate flat-frequency reference channels
`5 14 having a frequency spectrum whose magnitude is
`substantially flat over a frequency range of interest. Flat (cid:173)
`frequency reference channels 14 are fed into the set of
`frequency-selective constraint adaptive filters 6, which
`generate cancelling signals 15.
`In the mean time, main channel 12 is delayed through
`delay 7 so that it i8 synchronized with cancelling signals
`15. Difference unit 8 then subtracts cancelling signals 15
`from the delayed main channel to generate an digital output
`signal 16, which is converted by D/A unit 10 into analog
`15 form. Digital output signal 15 is fed back to the adaptive
`filters to update the filter weights of the adaptive filters.
`Flat-frequency reference channels 14 are fed to
`inhibiting unit 9, which estimates the power of each flat(cid:173)
`frequency reference channel as well as the power of the main
`20 channel and generates an inhibit signal 19 to prevent signal
`leakage.
`FIG. 2 depicts a preferred embodiment of the sampling
`unit. A sensor array 21, having sensor elements 21a-21d, is
`connected to an analog front end 22, having amplifier
`25 elements 22a-22d, where each amplifier element is connected
`to the output of the corresponding sensor element.
`In a
`directional microphone application, each sensor can be either
`a directional or omnidirectional microphone. The analog
`front end amplifies the received analog sensor signals to
`30 match the input requirement of the sampling elements. The
`outputs from the analog front ends are connected to a set of
`delta-sigma AID converters, 23 , where each converter samples
`and digitizes the amplified analog signals. The delta-sigma
`sampling is a well-known AID technique using both
`3S oversampling and digital filtering. For details on delta(cid:173)
`sigma AID sampling, see Crystal Semiconductor Corporation,
`Application Note: Delta-Sigma Techniques, 1989.
`
`8 -
`
`RTL898_1020-0017
`
`

`
`•
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`•
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`FIG. 3 shows an alternative embodiment of the sampling
`unit. A sensor array 31, having sensor elements 31a-31d, is
`c onnected to an ampl i fier 32, having amplifier elements 32a -
`32d, where each amplifier element amplifies the received
`5 signals from the corresponding sensor element. The outputs
`of the amplifier are connected to a sample & hold (S/ H) unit
`33 having sample & hold elements 33a-33d, where each 5/H
`element samples the amplified analog signal from the
`corresponding amplifier element to produce a discrete signal .
`10 The outputs from the S/ H unit are multiplexed into a single
`signal through a multiplexor 34 . The output of the
`multiplexor is connected to a conventional AI D converter 35
`to produce a digital signal.
`FIG . 4 is a schematic depiction of tapped delay lines
`15 used in the main channel matrix unit and the reference
`channel matrix in accordance with a preferred embodiment of
`the present invention . The tapped delay line used here is
`defined as a nonrecursive digital filter, also known in the
`art as a transversal filter, a finite impulse response filter
`20 or an FIR filter. The illustrated embodiment has 4 tapped
`delay lines, 40a-40d. Each tapped delay line includes delay
`elements 41, multipliers 42 and adders 43. Digital signals,
`44a - 44d, are fed into the set of tapped delay lines 40a-40d.
`Delayed signals through delay elements 41 are multiplied by
`25 filter coefficients, Fij , 45 and added to produce outputs,
`46a-46d .
`The n-th sample of an output from the i-th tapped delay
`line, Yi (n), can then be expressed as:
`t ltj•o F1,j Xi (n - j ) , where k is the length of the
`Yi (n)
`30 filter, and Xi (n) is the n-th sample of an input to the i-th
`tapped delay line .
`FIG . 5 depicts the main channel matrix unit for
`generating a main channel in accordance with a preferred
`embodiment of the present invention. The unit has tapped
`35 delay lines, SOa-SOd, as an input section taking inputs Sla (cid:173)
`Sld from the sampling unit.
`Its output section includes
`multipliers, 52a - 52d, where each multiplier is connected to
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`the corresponding tapped delay line and an adder 53, which
`The unit
`Bums all output signals from the multipliers.
`generates a main channel 54, as a weighted Bum of outputs
`from all multipliers. The filter weights 55a-55d can be any
`5 combination of fractions as long as their sum is 1. For
`example, if 4 microphones are used, the embodiment may use
`the filter weights of 1/4 in order to take into account of
`the contribution of each microphone.
`The unit acts as a beamformer, a spatial filter which
`10 filters a signal coming in all directions to produce a signal
`coming in a specific direction without physically moving the
`sensor array. The coefficients of the tapped delay lines and
`the filter weights are set in such a way that the received
`signals are spatially filtered to maximize the sensitivity
`15 toward the signal source.
`Since some interference signals find their way to reach
`the signal source due to many factors such as the
`reverberation of a room, main channel 54 representing the
`received signal in the direction of the signal source
`20 contains not only a source signal component, but also an
`interference signal component.
`FIG. 6 depicts the reference channel matrix unit for
`generating reference matrix channels in accordance with a
`preferred embodiment of the present invention. It has tapped
`25 delay lines, 60a-60d, as an input section taking inputs 61a-
`61d from the sampling unit. The same tapped delay lines as
`that of FIG. 4 may be used, in which case the tapped delay
`lines may be shared by the main and reference channel matrix
`units.
`Its output section includes multipliers, 62a-62d. 63a-
`63d, 64a-64d and adders 65a-65c, where each multiplier is
`connected to the corresponding tapped delay line and adder.
`The unit acts as a beamformer which generates the reference
`channels 66a-66c representing signals arriving off-axis from
`3S the signal source by obtaining the weighted differences of
`certain combinations of outputs from the tapped delay lines.
`The filter weight combinations can be any numbers as long as
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`their sum of filter weights for combining a given reference
`channel is O. For example, the illustrated embodiment may
`(Wll, W12, W13, W14) =
`use a filter weight combination,
`(0.25, 0.25, 0.25, -0.75), in order to combine signals 61a-
`5 61d to produce reference channel 66a.
`The net effect is placing a null (low sensitivity) in
`the receiving gain of the beamformer toward the signal
`source. As a result, the reference channels represent
`interference signals in directions other than that of the
`10 signal source.
`In other words, the unit nsteers" the input
`digital data to obtain interference signals without
`physically moving the sensor array.
`FIG. 7 is a schematic depiction of the decolorizing
`filter in accordance with a preferred embodiment of the
`15 present invention.
`It is a tapped delay line including delay
`elements 71, multipliers 72 and adders 73. A reference
`channel 74 is fed into the tapped delay line. Delayed
`signals are multiplied by filter coefficients, Ft , 75 and
`added to produce an output 76. The filter coefficients are
`20 set in such a way that the filter amplifies the low-magnitude
`frequency components of an input signal to obtain an output
`signal having a substantially flat frequency spectrum.
`As mentioned before in the background section, the
`output of a conventional adaptive beamformer suffers a non-
`2S uniform frequency behavior. This is because the reference
`channels do not have a flat frequency spectrum. The
`receiving sensitivity of a beamformer toward a particular
`angular direction is often described in terms of a gain
`curve. As mentioned before, the reference channel is
`30 obtained by placing a null in the gain curve (making the
`sensor array insensitive) in the direction of the signal
`source . The resulting gain curve has a lower gain for lower
`frequency signals than higher frequency signals. Since the
`reference channel is modified to generate a cancelling
`35 signal, a non-flat frequency spectrum of the reference
`channel is translated to a non-uniform frequency behavior in
`the system output.
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`The decolorizing filter is a fixed-coefficient filter
`which flattens the frequency spectrum of the reference
`channel (thus ndecolorizing" the reference channel) by
`boosting the low frequency portion of the reference channel.
`S By adding the decolorizing filters to all outputs of the
`reference channel matrix unit, a substantial ly flat frequency
`response in all directions is obtained.
`The decol orizing filter in the illustrated embodiment
`uses a tapped delay l ine filter which is the same as a finite
`10 impulse response (FIR) filter, but other kinds of filters
`such as an infinite impul se response (IIR) filter can also be
`used for the decolorizing filter in an alternative
`embodiment.
`FIG. 8 depicts schematically the inhibiting unit in
`15 accordance with a preferred embodiment of the present
`invention .
`It includes power estimation units 81 , 82 which
`estimate the power of a main channel 83 and each reference
`channel 84, respectively.
`A sample power estimation unit 85
`calculates the power of each sample. A multiplier 86
`20 multiplies the power of each sample by a fraction, a, which
`is the reciprocal of the number of samples for a given
`averaging period to obtain an average sample power 87. An
`adder 88 adds the average sample power to t he output of
`another multiplier 89 which multiplies a previously
`25 calculated main channel power average 90 by (l-a) . A new
`main channel power average is obtained by (new sample power)
`x a + (old power average) x (l-a). For example, if a 100-
`sample average is used, a = 0 . 01. The updated power average
`will be (new sample power) x 0 . 01 + (old power average) x
`In this way, the updated power average will be
`30 0.99.
`available at each sampling instant rather than after an
`averaging period. Although the illustrated embodiment shows
`an on-the-fly estimation method of the power average, other
`kinds of power estimation methods can also be used in an
`3S alternative embodiment.
`A multiplier 91 multiplies the main channel power 89
`with a threshold 92 to obtain a normalized main channel power
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`average 93. An adder 94 subtracts reference channel power
`averages 95 from the normalized main channel power a verage 93
`to produce a difference 96.
`If the difference is positive, a
`comparator 97 generates an inhibit signal 98. The inhibit
`5 signal is provided to the adaptive filters to stop the
`adaptation process to prevent signal leakage .
`Although the illustrated embodiment normalizes the main
`channel power average, an alternative embodiment may
`normalize the reference channel power average instead of the
`10 main channel power average. For example, if the threshold 92
`in the illustrated embodiment is 0 . 25, the same effect can be
`obtained in the alternative embodiment by normalizing each
`reference channel power average by mUltiplying it by 4.
`This inhibition approach is different from the prior art
`lS SNR-based inhibition approach mentioned in the background
`section in that it detects the presence of significant
`directional interference which the prior art approach does
`not consider. As a result , the directional - interference(cid:173)
`based inhibition approach stops the adaptation process when
`20 there is no significant direc tional interference to be
`eliminated, whereas the p r ior art approach does not .
`For example, where there is a weak source signal (e . g.
`during speech intermission) and there is almost no
`directional interference except some uncorrelated noise (such
`2S as noise due to wind or mechanical vibrations on the sensor
`structure) , the SNR-based approach would allow the adaptive
`filter to continue adapting due to the small SNR. The
`continued adaptation process is not desirable because there
`is very litt l e directional interference to be e l iminated in
`30 the first p l ace, and the adaptation process searches in vain
`for new fil t er weights to eliminate the uncorrelated noise,
`which often results in cancel l ing the source signal component
`of the received signal.
`By contrast, the directional-interference-based
`3S inhibition mec hanism will inhibit the adaptation proc ess in
`such a case because the strength of directional interference
`as reflected in the reference channe l power average will be
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`smaller than the normalized main channel power average,
`producing a positive normalized power difference. The
`adaptive process is inhibited as a result until there is some
`directional interference to be eliminated.
`FIG. 9 shows the frequency-selective constraint adaptive
`filter together with the difference unit in accordance with a
`preferred embodiment of the present invention. The
`frequency-selective constraint adaptive filter 101 includes a
`finite impulse response (FIR) filter 102, an LMS weight
`10 updating unit 103 and a frequency-selective weight-constraint
`unit 104.
`In an alternative embodiment, an infinite impulse
`response (IIR) filter can be used instead of the FIR filter.
`A flat-frequency reference channel 105 passes through
`FIR filter 102 whose filter weights are