Abstract: A handheld ultrasound device may comprise components configured to provide decreased size, weight, complexity, and power consumption. The handheld ultrasound device may comprise a beamformer configured to implement and compress a flag table in place of a delay table. These improvements can decrease the amount of memory used to generate ultrasound images, which can decrease the size, weight, and power consumption of the handheld ultrasound device. Ultrasound image data on a handheld imaging probe can be compressed on the handheld imaging probe prior to transmission from the probe in order to decrease the amount of data transmitted from the probe. The compressed data may comprise compressed pixels to maintain spatial image resolution. The compression circuitry may comprise an amount of memory related to a dynamic range of the compressed data that is independent of the dynamic range of the input data, which can decrease memory, power consumption, and latencies.

Abstract: Systems and methods for three-dimensional beamforming disclosed herein include, among other features (i) generating a set of received-sound beams by applying a plurality of sets of beamforming coefficients stored in a tangible memory of the network device to sound received via a microphone array of the network device, wherein each received-sound beam corresponds to a separate direction relative to the microphone array, (ii) identifying a subset of the received-sound beams comprising speech content, (iii) for each received-sound beam in the subset of the received-sound beams comprising speech content, determining whether the speech content comprises a wake word, (iv) selecting one final received-sound beam from the received-sound beams in the subset of the received-sound beams determined to comprise a wake word; and (v) causing the selected one final received-sound beam to be processed to identify a voice command.

Abstract: A wireless communication apparatus includes a first antenna circuit that forms a beam selected from among a plurality of first beams through a beamforming process performed for communication using a millimeter wave band and a monitor that displays a first pattern indicating a radiation direction of the selected beam.

Abstract: Systems and methods for three-dimensional beamforming disclosed herein include, among other features (i) generating a set of received-sound beams by applying a plurality of sets of beamforming coefficients stored in a tangible memory of the network device to sound received via a microphone array of the network device, wherein each received-sound beam corresponds to a separate direction relative to the microphone array, (ii) identifying a subset of the received-sound beams comprising speech content, (iii) for each received-sound beam in the subset of the received-sound beams comprising speech content, determining whether the speech content comprises a wake word, (iv) selecting one final received-sound beam from the received-sound beams in the subset of the received-sound beams determined to comprise a wake word; and (v) causing the selected one final received-sound beam to be processed to identify a voice command.

Abstract: Systems and methods for three-dimensional beamforming disclosed herein include, among other features (i) generating a set of received-sound beams by applying a plurality of sets of beamforming coefficients stored in a tangible memory of the network device to sound received via a microphone array of the network device, wherein each received-sound beam corresponds to a separate direction relative to the microphone array, (ii) identifying a subset of the received-sound beams comprising speech content, (iii) for each received-sound beam in the subset of the received-sound beams comprising speech content, determining whether the speech content comprises a wake word, (iv) selecting one final received-sound beam from the received-sound beams in the subset of the received-sound beams determined to comprise a wake word; and (v) causing the selected one final received-sound beam to be processed to identify a voice command.

Abstract: A method includes receiving a first microphone array processing signal (associated with a frequency band that includes a plurality of sub-bands) and receiving a second microphone array processing signal associated with the frequency band. The method includes generating a first output corresponding to a first sub-band based on the first microphone array processing signal and generating a second output corresponding to the first sub-band based on the second microphone array processing signal. The method includes generating a third output corresponding to a second sub-band based on the first microphone array processing signal and generating a fourth output corresponding to the second sub-band based on the second microphone array processing signal. The method includes performing a first set of microphone mixing operations to generate an adaptive mixer output for the first sub-band and performing a different set of microphone mixing operations to generate another adaptive mixer output for the second sub-band.

Abstract: An apparatus comprising: an audio signal analyzer configured to analyze at least one audio signal to determine at least one audio component with an associated orientation parameter; a reference definer configured to define at least one of: a reference orientation for an apparatus; and a reference position for the apparatus; a directional determiner configured to determine a direction value based on the reference orientation/position for the apparatus and at least one of: an orientation of the apparatus; a position of the apparatus; an orientation of a further apparatus co-operating with the apparatus; and a position of the further apparatus; and a directional processor configured to process at least one associated directional parameter for the at least one audio component dependent on the direction value.

Abstract: A video conference endpoint determines a position of a best audio pick-up region for placement of a sound source relative to a microphone having a receive pattern configured to capture sound signals from the best region. The endpoint captures an image of a scene that encompasses the best region and displays the image of the scene. The endpoint generates an image representative of the best region and displays the generated image representative of the best region as an overlay of the scene image.

Abstract: A microphone device for an automobile has a microphone capsule for detecting ambient noise and a housing in which the microphone capsule is arranged. The microphone device is intended to have a small installation depth and to be capable of being easily installed in an automobile. An electrical connection interface which includes externally accessible connection contacts for making electrical contact with the microphone capsule located in the housing, is provided in the microphone device. The connection contacts are integrated into a mechanical plugging apparatus for plugging together with a complementary plugging part of an electronic device of the automobile. The connection interface of the microphone device is configured in a cableless fashion here in that the plugging apparatus is embodied as a part of the housing.

Abstract: A method operates a hearing aid system and the hearing aid system has at least two hearing aid devices, between which a signal path is provided, and with at least one signal processing unit, which is provided to process audio signals. In the method the signal processing apparatus filters first audio signals with a filter predetermined for a defined spatial direction, from which a useful signal arrives, so that second audio signals are generated, in which the components of the useful signal in the second audio signals are equalized to a greater degree than in the first audio signals. The second audio signals are then filtered with an adaptive filter, so that third audio signals are generated, in which the components of the useful signal are equalized to an even greater degree than in the second audio signals.

Abstract: A system and method for measuring the frequency response of a system under test using a single swept-frequency chirp signal. A tapered chirp-frequency test signal is created with a bandwidth defined by first and second frequencies. The test signal is routed to a calibration path, and the output of the calibration path is routed to a digitizer. The output of the calibration path is digitized, and a Fourier transform of the calibration path output is generated. The test signal is then routed to a test system, and the output of the test system is coupled to the digitizer. The output of the test system is digitized, and a Fourier transform of the test system output is generated. A normalized frequency-domain representation of the test system created by dividing the Fourier transform of the test system output by the Fourier transform of the calibration path output.

Abstract: A distance sensor for vehicles is provided which includes a sensor device and an electric connector for an electrical connection between the sensor device and a mating connector. The connector includes at least one first terminal pin and at least one second terminal pin. The first terminal pin has a first section leading to the sensor device and a second section extending toward an inlet opening of the connector. The second terminal pin has a first section leading to the sensor device and at least two branched second sections extending toward the inlet opening of the connector. Specifically, the second terminal pin is shaped to provide two-pin plugs, thereby permitting the size of the connector to be reduced, which facilitates ease of installation of the distance sensor to, for example, a bumper of the automotive vehicle.

Abstract: A method of combining at least two audio signals for generating an enhanced system output signal is described.

Abstract: A method and system is described for reducing unwanted noise components/interfering targets detected through an ambiguous beam-steer direction, such as the ambiguous ‘back-lobe’ of a sensor array. A pressure sensor array 52 and a plurality of pressure gradient sub-arrays 541-N are beamformed individually at processing stages 56, 581-N to derive signals indicative of directional responses, one from each group or sub-group. The signals are scaled and weighted (60, 620-N) and processed to define a cardioid beam with nulls pointing in the ambiguous beam-steer direction.

Abstract: An audio system comprises an ultrasound sensor array (105) which has a plurality of ultrasound sensor elements, and an audio band array (101) comprising a plurality of audio band elements. The same array of wideband audio transducers may be used for both the ultrasound sensor array (105) and the audio band array (101). An estimator (107) generates a presence characteristic of a user in response to ultrasound signals received from the ultrasound sensor array. The presence characteristic may specifically comprise a position estimate for the user. An audio array circuit (103) generates a directional response for the audio band array (101) by applying weights to individual audio band signals for the audio band elements. A weight circuit (109) determines the weights in response to the presence characteristic. The system may provide improved adaptation of the directivity of the audio band array (101) and specifically does not require the sound source in the audio band to be active for adaptation.

Abstract: Provided is a radio communication device for performing radio communication with another radio communication device includes a control unit that controls to prepare for data loss during radio communication of transmission data and a transmission unit that transmits the transmission data by radio according to the control of the control unit. One of the radio communication device and the other radio communication device estimates a distance from the other based on a field intensity of a radio signal which is judged to satisfy a certain requirement regarding noise component among received radio signals received from the other of the radio communication device and the other radio communication device. The control unit performs a control of a content according to the distance estimation result.

Abstract: A fault monitoring system for an electrical substation includes one or more arrays of acoustic sensors placed around a substation boundary and configured to measure acoustic pressure waves. A processing circuitry in the fault monitoring system localizes and identifies faults in the substation and includes a sound source localization module to identify an area of a sound source based on the acoustic pressure waves. A beamforming module of processing circuitry provides enhanced acoustic pressure waves by eliminating background noise and interfering noise from the acoustic pressure waves and a component signature classification module in the processing circuitry compares enhanced acoustic pressure waves with component signatures to classify the acoustic pressure waves into various events to detect faults in the electrical substation.

Abstract: A system locates a speaker in a room containing a loudspeaker and a microphone array. The loudspeaker transmits a sound that is partly reflected by a speaker. The microphone array detects the reflected sound and converts the sound into a microphone array, the speaker's distance from the microphone array, or both, based on the characteristics of the microphone signals.

Abstract: There is provided a sound source characteristic determining device (10) capable of being applied in an environmental where the type of a sound source is unknown. The device includes a plurality of beamformers (21-1 to 21-M) used when a sound source signal generated from a sound source at an arbitrary position in a space is inputted to a plurality of microphones (14-1 to 14-N), for weighting the acoustic signal detected by each of the microphones by using a function for correcting the difference of the sound source signals generated between the microphones and outputting a totaled signal. Each of the beamformers (21-1 to 21-M) contains a function having a unit directivity characteristic corresponding to one arbitrary direction in the space and is arranged for each of the directions corresponding to an arbitrary position in the space and the unit directivity characteristic.

Abstract: A distance sensor for vehicles is provided which includes a sensor device and an electric connector for an electrical connection between the sensor device and a mating connector. The connector includes at least one first terminal pin and at least one second terminal pin. The first terminal pin has a first section leading to the sensor device and a second section extending toward an inlet opening of the connector. The second terminal pin has a first section leading to the sensor device and at least two branched second sections extending toward the inlet opening of the connector. Specifically, the second terminal pin is shaped to provide two-pin plugs, thereby permitting the size of the connector to be reduced, which facilitates ease of installation of the distance sensor to, for example, a bumper of the automotive vehicle.

Abstract: Compressive Sensing (CS) is an emerging area which uses a relatively small number of non-traditional samples in the form of randomized projections to reconstruct sparse or com-pressible signals. Direction-of-arrival (DOA) estimation is performed with an array of sensors using CS. Using random projections of the sensor data, along with a full waveform recording on one reference sensor, a sparse angle space scenario can be reconstructed, giving the number of sources and their DOA's. Signal processing algorithms are also developed and described herein for randomly deployable wireless sensor arrays that are severely constrained in communication bandwidth. There is a focus on the acoustic bearing estimation problem and it is shown that when the target bearings are modeled as a sparse vector in the angle space, functions of the low dimensional random projections of the microphone signals can be used to determine multiple source bearings as a solution of an 1]-norm minimization problem.

Abstract: A method of processing ultrasound signals received from a plurality of data channels each associated with a transducer element. A sorted delay data table having sorted delay data is generated that includes a channel identifier, a fractional delay value, and integer delay value. The sorted delay data table clusters together channel groups including a first channel group having data channels with the first fractional delay value and a second channel group with data channels with the second fractional delay value. Control signals are generated based on the sorted delay data that implements data path combining by directing channel data from the first channel group for processing by a first interpolation filter that provides the first fractional delay value and channel data associated with the second channel group for processing by a second interpolation filter that provides the second fractional delay value.

Abstract: A method for identifying position of acoustic source proximate sediments below the bottom of a body of water includes deploying a plurality of arrays of acoustic sensors on the bottom of the body of water. Each array includes a plurality of lines of acoustic sensors disposed in a substantially radial pattern. The arrays have a center to center distance therebetween of about twice a diameter of each of the arrays. Signals are detected from each of the sensors for a selected time period. A direction of the acoustic source with respect to each of the arrays is determined by steering a response of the sensors in each array. A range of the acoustic source is determined using the determined directions.

Abstract: A directional microphone system is disclosed, which comprises circuitry for low pass filtering a first order signal, and circuitry for high pass filtering a second order signal. The system further comprises circuitry for summing the low pass filtered first order signal and the high pass filtered second order signal. A method of determining whether a plurality of microphones have sufficiently matched frequency response characteristics to be used in a multi-order directional microphone array is also disclosed. For a microphone array having at least three microphones, wherein one of the microphones is disposed between the other of the microphones, a method of determining the arrangement of the microphones in the array is also disclosed.

Abstract: A method for controlling beamforming in a device by a user of the device is provided. The method includes presenting current beam directional information via a user interface and receiving updated beam directional information in response to user input. The method also includes altering beamforming direction based upon the received updated beam directional information.

Abstract: A method of processing signals received from an array of sensors includes sampling and digitally converting the received signals. The digitally converted signals are processed to provide an output signal, the processing including filtering the signals using a first adaptive filter that enhances a target signal of the digitally converted signals and a second adaptive filter that suppresses an unwanted signal of the digitally converted signals, and processing the filtered signals in a frequency domain to further suppress the unwanted signal. The digitally converted signals are processed to determine a direction of arrival of the target signal, the processing including filtering the signals using a third adaptive filter.

Abstract: This invention exploits the synchronization properties of coupled, nonlinear oscillators arrays to perform power combining, beam steering, and beam shaping. This architecture utilizes interactions between nonlinear active elements to generate beam patterns. A nonlinear array integrates the signal processing concurrently with the transduction of the signal. This architecture differs fundamentally from passive transducer arrays in three ways: 1) the unit cells are nonlinear, 2) the array purposely couples the unit cells together, and 3) the signal processing (beam steering and shaping) is done via dynamic interactions between unit cells. The architecture extends to both 1- and 2-dimensional arrays.

Abstract: A hearing device comprising at least two microphones, a processor and an output for further processing in a reproducer, the microphones being electrically coupled to the processor, and the output for further processing in a reproducer being connected to the processor. The processor comprises: a summer providing a sum signal of the microphone signals, and to which summer a proportional amplifier is connected, and a summer for providing a difference signal of the microphone signals, to which summer an integrator is connected, and that the proportional amplifier and the integrator are coupled on the output side to a third summer providing a sum signal of the proportional amplifier and the integrator, and that the third summer forms the output for further processing in a reproducer.

Abstract: An advantageous combination of preamplifier electronics and circuits for acoustic beam forming with ultrasound transducer arrays permits integration of the electronics and transducer(s) at close distances in a compact arrangement and is operable from an ultrasound imaging or measurement system using only a minimal number of connecting electric wires. To achieve dynamic focusing of the receive beam with a transducer array, an integrated electronic circuit mounted in close proximity to the transducer array provides preamplifiers for the individual elements and delay circuits that are automatically switched in a predetermined time sequence after transmission of an imaging pulse. When a high voltage pulse is transmitted via the cable, a breakthrough circuit connects the cable to the transducer for transmission of the ultrasound pulse, while in receive mode the low level signal on the transducer is amplified and fed as a higher level signal via the same cable to the imaging or measurement instrument.

Abstract: A system and method for a directional microphone system is disclosed. The directional microphone system can adaptively track and detect sources of sound information, and can reduce background noise. A first monolithic detection unit for detecting sound information and performing local signal processing on the detected sound information is provided. In the detection unit, an integrated transducer is provided for receiving acoustic waves and for generating sound information representative of the waves. A processor is coupled to the transducer for receiving the sound information and for performing local digital signal processing on the sound information to generate locally processed sound information. A base unit is coupled to the first monolithic detection unit and includes a global processor which receives the locally processed sound information and performs global digital signal processing on the locally processed sound information to generate globally processed sound information.

Abstract: A transducer is constructed of a plurality of transducer elements which are arranged in multiple layers and columns forming as a whole a generally cylindrical shape. A scanning sonar employing the transducer thus constructed can be switched between horizontal scan mode in which a vertically focused beam having a narrow horizontal beam angle (high horizontal directivity) is steered around the transducer using all the transducer elements and vertical scan mode in which a horizontally focused beam having a narrow vertical beam angle (high vertical directivity) is steered in a vertical plane directed in a specified scan azimuth to find out the angle of incidence of a received signal. The horizontally focused beam is formed by using the transducer elements of specific columns centered on the specified scan azimuth. A vertical scan signal obtained in the vertical scan mode is multiplied by a chirp signal.

Abstract: The hybrid adaptive beamformer of the present invention includes a plurality of microphones for receiving sound energy from an external environment and for producing a plurality of microphone outputs from the sound energy. A processor produces a plurality of first order beams based on the microphone outputs. The processor determines an amount of reverberation in the external environment and adaptively produces a second order output beam taking into consideration the determined amount of reverberation. The processor may determine the amount of reverberation based on a comparison of the first order beams. The processor may produce the second order output beam by adaptively combining the plurality of first order beams, as further described below, or by adaptively combining the microphone outputs. The adaptation varies taking into consideration the determined amount of reverberation.

Abstract: A hull-mounted, multiple-frequency sonar system having a primary transduc which provides transmission power for active modes and an added array of miniature hydrophones is provided. The multiple frequency sonar system includes vertical and horizontal arrays of miniature hydrophones connected by a multiplexer to four beamformer components. These beamformers process data received by acoustic intercept signals, passive signals, active fundamental frequency signals, and active second-harmonic frequency signals. Signal processing and display is available from each of the beamformers and an additional mode is provided wherein fundamental and second harmonic frequency data is combined and then displayed.

Abstract: Interference canceling is provided for canceling, from a target signal generated from a target source, an interference signal generated by an interference source. The beam splitter beam-splits the target signal into a plurality of band-limited target signals band-limited frequency bands and beam-splits the interference signal into corresponding band-limited frequency bands. The adaptive filter adaptively filters each band-limited interference signal from each corresponding band-limited target signal. The inhibitor can permit the adaptive filter to adapt or change coefficients when a signal-to-noise ratio of the reference signal exceeds a predetermined threshold, to be determined periodically, over a signal-to-noise ratio of the main signal. The beam selector selects at least one of a plurality of beams for adaptive filtering by the adaptive filter representing a direction from which the main signal is received.

Abstract: An adaptive system and method for reducing interference in a signal received from an array of sensors. Adaptive filters are used to generate cancelling signals that closely approximate the interference present in the received signal. The adaptive filter weights are converted into the frequency domain where the frequency representation values in a selected frequency range are truncated to avoid signal leakage involving narrow band signals. Decolorizing filters are used to produce the cancelling signals having a flat frequency spectrum. Normalized power difference is used limit the operation of the adaptive filters to the case where there is some directional interference to be eliminated.

Abstract: In an adaptive array beamformer, a spatial beamforming filter is connected to a sensor array for respectively filtering and summing array signals to produce a first filter output containing a target signal that arrives in a specified direction. First adaptive filters provide transversal-filtering the first filter output to produce a second filter output not containing the target signal, using a first error signal by restraining their tap weight coefficients. The array signals are further coupled to subtractors. Each subtractor detects a difference between the second filter output of the corresponding first adaptive filter and the corresponding sensor signal to derive the first error signal. Second adaptive filters provide transversal-filtering the first error signals of the subtractors to produce third filter outputs, using a second error signal, by restraining their tap weight coefficients.

Abstract: A passive sonar transducer arrangement includes two steerable array sections (10,11) arranged with the transducers of one section (10) spaced apart from one another by a different distance (d1) from the spacing (d2) between the transducers of the other section (11), the spacing for each section being such that aliasing will occur for at least some steering angles (.THETA..sub.s) and such that the alias beams produced by the two sections of the array will not overlap. Cross-correlation means (12-19) are provided to process the signals received from the sonar transducers of the two sections such that the effects of the alias beams are substantially removed.

Abstract: The invention relates to a device for locating noise emitters with an antenna comprising passive sensors. In a preferred variant, the antenna consists of N acoustic buoys each comprising a pair of hydrophone dipoles exhibiting a double-eight directivity diagram. Each buoy comprises a compass providing a signal K.sub.n representing the heading relative to magnetic North. The pairs of signals (a.sub.1 -b.sub.1 to a.sub.N -b.sub.N) are subjected to a rotation by the angle K.sub.n. The signals are next digitized and the number P of noise emitters estimated. The locating of these noise emitters can be performed according to two variants (8, 9).

Abstract: A device for steering a null in a sonar beam pattern resolves first and snd beam signals into four quadrature components. These components are multiplied in multiplying digital to analog converters by binary number factors, derived in accordance with the desired null angle, to provide analog signals that are combined in a summing amplifier to provide analog output corresponding to acoustic energy received by a beam having a null at the desired angle.

Abstract: An array of sonar transmitting and receiving transducers are so physically rranged and their signal voltage so combined as to form beams pointing to various points of the compass. In this invention a signal is formed on a different conductor for each beam and the conductors are wiped in rapid succession to determine the presence of any beam signal. Since any signal may be short-lived the total sampling process must be in milliseconds. Any signal that occurs is converted to digital form and is fed into a first recirculating delay line time compression storage loop. All signals are stored in the first loop. Then the cluster of signals pertaining to one beam is read out to a second recirculating delay line time compression storage loop from when it is read out to a variable frequency super-heterodyne and displayed on a cathode ray tube.

Abstract: A compensating beamformer which requires orders of magnitude fewer calculations that prior art methods. A compensating beamformer is provided which comprises a plurality of sensing elements and a plurality of analog-to-digital converters, for converting incoming analog signals to digital form. Digital signals from at least four such elements are used to compute phase angle information which is combined to form a matrix of input data in the frequency domain. An unweighted steering vector is determined to sample data from the target direction. A corrector matrix is calculated based on input data from sensing elements. That corrector matrix along with its inverse, which is determined recursively, is used in combination with the unweighted steering vector to determine an optimal steering vector. The input data in the frequency domain are then multiplied by the optimal steering vector to obtain signals in the directions of interest.

Abstract: When observing or setting changeable positions of respective movable components of a machine, such as movable saws of a multiple cut sawing machine, each linear position-displacement transducer used has the wave guide housing thereof, made of sufficient length to receive, instead of one permanent magnet, several overall cylindrical electromagnets serving as flux generators. Each electromagnet: has a centered hole, to permit the movement thereof along the wave guide; and has four electromagnets equally radially spaced in a radial plane, with their commonly positioned north poles, forming an overall central north pole, surrounding the center hole, which is formed by a non ferrous tube, and with their radially spaced south poles being adjacent to an encircling soft iron band.

Abstract: A three-dimensional systolic architecture for beamforming according to convolve signals from an array with sets of weighting coefficients using a one dimensional array of similar modules with finite impulse response filter elements with outputs directed into pipelining register stacks and propagated to the stack boundaries for summing in accumulator arrays and demultiplexing.

Abstract: A signal processing arrangement is connected to a microphone array to form at least one directable beam sound receiver. The directable beam sound receivers are adapted to receive sounds from predetermined locations in a prescribed environment such as auditorium. Signals representative of prescribed sound features received from the plurality of predetermined locations are generated and one or more of the locations is selected responsive to the sound feature signals. A plurality of directable beam sound receivers may be used to concurrently analyze sound features from the predetermined locations. Alternatively, one directable beam sound receiver may be used to scan the predetermined locations so that the sound feature signals therefrom are compared to sound features from a currently selected location.

Abstract: Apparatus and method for discriminating against non-Gaussian noise. Analog signals from an array of sensors are converted to real and imaginary digital representations and processed such that non-Gaussian noise is separated from signals of interest. The processor uses estimates of Kurtosis and quantiles from either past or adjacent frequency components to construct non-linear elements, which are then used to process remaining signal data to improve the signal-to-noise ratio thereof by removing non-Gaussian noise therefrom.

Abstract: The chirp rate in a chirp Z-transform sonar beam former is varied as a function of range to form beams in the near field. The chirp rate may be varied with each range cell under examination or with each group of range cells.

Abstract: A frequency-domain beamformer is used with arrays comprising M rings with a maximum of N elements per ring, the positions of the elements having rotational symmetry. An element need not be present at each position. The outputs of each of the M rings of arrays are connected to the inputs of M sets of lowpass filters, each set comprising N filters. Each filter output is an input to a sample-and-hold circuit.A temporal discrete Fourier transform (DFT) is performed on the output signal from each sample-and-hold circuit. M demultiplexers, one for each of the M sets of DFT circuits, convert their parallel input signals into serial output signals.A memory, or function generator, is used for steering. A circular convolver convolves the outputs of the demultiplexers and the memories, the outputs of the convolvers being summed in an output summer.The beamformer is used with a 3-D array such that it has rotational symmetry, that is, if rotated at some angle, the situation is exactly the same as before the rotation.

Abstract: A beamforming apparatus is provided for processing the outputs of a linear rray of spaced apart receiving elements. The beamforming apparatus includes a surface acoustic wave device which has a pair of transducers mounted on a substrate in a spaced apart relationship. Each transducer is capable of receiving and converting an electrical chirp signal into an acoustic signal for propagation across the surface of the surface acoustical wave device. A plurality of taps are mounted on the substrate in a spaced apart relationship between the pair of transducers for receiving, sharing and converting the acoustic signals back into electric signals. Each tap is adapted to receive a bias voltage. A device is provided for mixing the signal from each tap with a signal from a respective receiving element so as to produce a plurality of mixed output signals, and another device is provided for summing the mixed output signals so as to provide a summed output signal.

Abstract: An apparatus is provided for processing the outputs of a two dimensional nar array of spaced apart receiving elements. The receiving elements are grouped and spaced apart in rows, which may be horizontal and vertical however, the rows may be in other directions provided they lie within a plane. The apparatus includes a plurality of surface acoustic wave (SAW) devices, each SAW device corresponding to a respective horizontal row of receiving elements and being capable of producing a plurality of horizontal phase shift outputs which linearly correspond to the number and horizontal spacing of receiving elements in a respective horizontal row.

Abstract: Discrimination of long range targets from short range targets is provided by generating a vertical dipole beam and a narrow uni-directional horizontal beam which may be steered. Targets in the dipole beam are short range targets whose acoustic signals arrive at angles to the horizontal plane, whereas long range targets are those in the horizontal beam, since acoustic signals from long range targets come in on the horizontal plane.