Abstract: Methods and transducers for producing acoustical signals having a spiral wavefront with omnidirectional magnitude and a phase that varies with angle and transducers for producing broadband omnidirectional reference signals for underwater navigation and communication.

Abstract: A signal processing apparatus includes a source separation module for producing respective separation signals corresponding to a plurality of sound sources by applying an ICA (Independent Component Analysis) to observation signals produced based on mixture signals from the sound sources, which are taken by source separation microphones, to thereby execute a separation process of the mixture signals, and a signal projection-back module for receiving observation signals of projection-back target microphones and the separation signals produced by the source separation module, and for producing projection-back signals as respective separation signals corresponding to the sound sources, which are taken by the projection-back target microphones. The signal projection-back module produces the projection-back signals by receiving the observation signals of the projection-back target microphones which differ from the source separation microphones.

Abstract: A device for measuring, documenting, and issuing citations for noise violations comprises a mobile device for measuring the decibel sound level from a sound source, for measuring the distance from the device to the source, for capturing an image or video of the sound source, and determining whether the sound source exceeds a user-predetermined level. The device may be in the form of a hand-held gun-shaped device wherein the user actuates the trigger to record an image or video of the sound source, the decibel sound level, and the distance between the device and the source. After the data is collected, the sound-measuring device is attached to a download component. The download component accepts data from the sound-measuring component and allows the user to input additional data regarding the noise violation. The user may then use the download component to print a citation, which may include an image of the sound source.

Abstract: A method and apparatus for detecting contact between an underground electric wire conduit and an underground utility pipeline in which a detectable signal is applied to an electric line wire disposed within the underground electric wire conduit and the underground utility pipeline is monitored for a presence of the detectable signal. When there is a contact between the underground electric wire conduit and the underground utility pipeline, the signal applied to the electric line wire is detected by a suitable signal detector disposed on the surface of the utility pipeline and/or within the utility pipeline.

Abstract: A top drive assembly may comprise a drive motor that provides rotational torque to a drill string for driving the drill string into the earth. A sensor and a transmitter may be located on a section of the drill string or on a device that can be incorporated into a drill string. The sensor may take measurements of the drill string that is rotating during the drilling operation. If the sensor is located near the top drive assembly, the sensor may take measurements of an upper portion of the rotating drill string during the drilling operation. The transmitter may wirelessly transmit the sensor information in real-time to a coordinator or main radio. The transmitter may also be located near the top drive assembly and during the drilling operation, may transmit from the rotating drill string while located in a position above the earth's surface.

Abstract: Systems and methods for parallel and distributed processing of audio signals produced by a microphone array are described. In one aspect, a distributed signal processing system includes an array of microphones and an array of processors. Each processor is connected to one of the microphones and is connected to at least two other processors, enabling communication between adjacent connected processors. The system also includes a computing device connected to each of the processors. Each microphone detects a sound and generates an audio signal, and each processor is configured to receive and process the audio signal sent from a connected microphone and audio signals sent from at least one of the adjacent processors to produce a data stream that is sent to the computing device.

Abstract: A method for designing a sound waveguide surface is described. The method includes the steps of forming a parametric model of the sound waveguide surface where the parametric model has at least one input parameter and then simulating a sound field that is formed by the sound waveguide surface. A frequency dependent spatial distribution measure is then determined for the sound field associated with the sound waveguide surface and the at least one input parameter is varied to change the sound waveguide surface to adjust the value of the frequency dependent spatial distribution measure.

Abstract: This invention describes a method for efficient beamforming for generalized sidelobe canceling using complementary noise separation filtering for generating a noise reference for adaptation performance of an adaptive interference canceller (AIC). The adaptive filter provides noise estimates to be subtracted from the desired signal path providing further noise reduction in the system output. More specifically, the present invention relates to a multi-microphone beamforming system similar to a generalized sidelobe canceller (GSC) structure, but the difference with the conventional GSC method is that the complementary filter used for desired signal blocking can be realized with a simple subtraction without compromising the beam steering flexibility of the polynomial beamforming filter front end using the desired target signal and the complementary background noise estimate signal, respectively, with the complexity of one complementary filter and one sum beamformer.

Abstract: A method for acquisition and processing of marine seismic signals to extract up-going and down-going wave-fields from a seismic energy source includes deploying at least two marine seismic energy sources at different depths in a body of water. These seismic energy sources are actuated with known time delays that are varied from shot record to shot record. Seismic signals from sources deployed at different depths are recorded simultaneously. Seismic energy corresponding to each of the sources is extracted from the recorded seismic signals. Up-going and down-going wave-fields are extracted from the sources deployed at different depths using the extracted seismic energy therefrom. A method includes the separated up-going and down-going wave-fields are propagated to a water surface or a common reference, the up-going or the down-going wave-field is 180 degree phase shifted, and the signals from these modified up-going and down-going wave-fields are summed.

Abstract: A guiding sound generating apparatus which generates a guiding sound for guidance from a present location to a destination, the apparatus having a sound source, a variable sound-source attribute unit that changes a sound source attribute depending on a position or a positional change of a present location with respect to a destination, and an attribute-application unit that applies the sound-source attribute generated from the variable sound-source attribute unit to the sound source and generates a guiding sound where a sound-source attribute is changed by the position or positional change of the present location with respect to the destination.

Abstract: A system and method for performing audio source localization that uses minimum variance distortionless response (MVDR) beams or other super-directive beams that are more suitable for reducing noise, interference and/or acoustic coupling than the delay-and-sum beams conventionally used in performing audio source localization based on a Steered Response Power (SRP) approach. Since super-directive beams such as MVDR beams do not provide good properties for performing conventional SRP-based audio source localization, an embodiment of the present invention utilizes a different metric than maximum response power in order to determine which of a plurality of beams provides the best response for audio source localization. This metric involves identifying which beam provides the smallest measure of distortion with respect to a reference power or reference response.

Abstract: Mapping coherent/incoherent acoustic sources as determined from a phased microphone array. A linear configuration of equations and unknowns are formed by accounting for a reciprocal influence of one or more cross-beamforming characteristics thereof at varying grid locations among the plurality of grid locations. An equation derived from the linear configuration of equations and unknowns can then be iteratively determined. The equation can be attained by the solution requirement of a constraint equivalent to the physical assumption that the coherent sources have only in phase coherence. The size of the problem may then be reduced using zoning methods.

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 novel enhanced beamforming technique that improves beamforming operations by incorporating a model for the directional gains of the sensors, such as microphones, and provides means of estimating these gains. The technique forms estimates of the relative magnitude responses of the sensors (e.g., microphones) based on the data received at the array and includes those in the beamforming computations.

Abstract: A handsfree communication system includes microphones, a beamformer, and filters. The microphones are spaced apart and are capable of receiving acoustic signals. The beamformer compensates for propagation delays between the direct and reflected acoustic signals. The filters are configured to a predetermined susceptibility level. The filter process the output of the beamformer to enhance the quality of the received signals.

Abstract: A novel beamforming post-processor technique with enhanced noise suppression capability. The present beam forming post-processor technique is a non-linear post-processing technique for sensor arrays (e.g., microphone arrays) which improves the directivity and signal separation capabilities. The technique works in so-called instantaneous direction of arrival space, estimates the probability for sound coming from a given incident angle or look-up direction and applies a time-varying, gain based, spatio-temporal filter for suppressing sounds coming from directions other than the sound source direction resulting in minimal artifacts and musical noise.

Abstract: A novel adaptive beamforming technique with enhanced noise suppression capability. The technique incorporates the sound-source presence probability into an adaptive blocking matrix. In one embodiment the sound-source presence probability is estimated based on the instantaneous direction of arrival of the input signals and voice activity detection. The technique guarantees robustness to steering vector errors without imposing ad hoc constraints on the adaptive filter coefficients. It can provide good suppression performance for both directional interference signals as well as isotropic ambient noise.

Abstract: A method for processing an audio signal received through a microphone array coupled to an interfacing device is provided. The method is processing at least in part by a computing device that communicates with the interfacing device. The method includes receiving a signal at the microphone array and applying adaptive beam-forming to the signal to yield an enhanced source component of the signal. Also, an inverse beam-forming is applied to the signal to yield an enhanced noise component of the signal. The method combines the enhanced source component and the enhanced noise component to produce a noise reduced signal, where the noise reduced signal is a target voice signal. Then, monitoring an acoustic set-up associated with the audio signal as a background process using the adaptive beam-forming inverse beam-forming to track the target signal component, and periodically setting a calibration of the monitored acoustic set-up.

Abstract: Methods and systems for acquiring spectral and velocity information with a multi-dimensional array are provided. For example, a dedicated receive aperture is formed at a multi-dimensional array for steered continuous wave imaging. Other elements not within the dedicated receive aperture are used for transmitting continuous waves or transmitting and receiving pulsed waveforms in other modes of imaging. As another example, switches or other structures are provided for selecting between a plurality of possible apertures for a steered continuous wave aperture. The selection is performed in response to a configuration of an ultrasound system, such as selection of a focal location or steer direction. The aperture is then used for either transmit or receive operations of steered continuous wave imaging. As yet another example, at least part of the steered continuous wave beamformer is provided within a transducer assembly.

Abstract: The adaptive beamformer unit (191) comprises: a filtered sum beamformer (107) arranged to process input audio signals (u 1, u2) from an array of respective microphones (101, 103), and arranged to yield as an output a first audio signal (z) predominantly corresponding to sound from a desired audio source (160) by filtering with a first adaptive filter (f1(-t)) a first one of the input audio signals (u1) and with a second adaptive filter (f2(-t)) a second one of the input audio signals (u2), the coefficients of the first filter (f1(-t)) and the second filter (f2(-t)) being adaptable with a first step size (a1) and a second step size ((x2) respectively; noise measure derivation means (111) arranged to derive from the input audio signals (u1, u2) a first noise measure (x1) and a second noise measure (x2); and an updating unit (192) arranged to determine the first and second step size (a1, (x2) with an equation comprising in a denominator the first noise measure (x1) for the first step size (a1), respectively the

Abstract: A system using beamforming techniques in conjunction with an active or passive acoustic buoy field, where the buoy field has a plurality of buoys, each buoy employing at least one sensor attached to and extending substantially downward from that buoy so as to form a planar or conformal array. Each array buoy uses highly accurate GPS tracking devices to locate that buoy's array sensor position relative to all other buoy arrays in the known buoy field. This accurate positional data is used in conjunction with the sensor data from each depth to beamform a planar or a volumetric array.

Abstract: A system and method for performing audio source localization that uses minimum variance distortionless response (MVDR) beams or other super-directive beams that are more suitable for reducing noise, interference and/or acoustic coupling than the delay-and-sum beams conventionally used in performing audio source localization based on a Steered Response Power (SRP) approach. Since super-directive beams such as MVDR beams do not provide good properties for performing conventional SRP-based audio source localization, an embodiment of the present invention utilizes a different metric than maximum response power in order to determine which of a plurality of beams provides the best response for audio source localization. This metric involves identifying which beam provides the smallest measure of distortion with respect to a reference power or reference response.

Abstract: The present invention, as typically embodied, represents a novel methodology for effecting linear processing of output signals that are received from one or more acoustic vector sensors. First, as pertains to each modal beam, the modal beam amplitude response bmn, is calculated as the matrix product of a data vector d and a modal weighting vector wmn, wherein the weighting vector wmn is uniquely defined in terms of three different linear modal weighting vector equations corresponding to wmnx, wmny, and wmnz, respectively. Second, as pertains to all of the modal beams, the directional beam amplitude response bdirMN is calculated as the sum of all of the individual modal beam amplitude responses bmn. Because the inventive processing methodology is linear in nature (as distinguished from non-linear, e.g., quadratic, in nature), inventive practice is highly effective for performing quantitative acoustic measurements of sound fields.

Abstract: The invention is directed to a handsfree system for use in a vehicle comprising a microphone array with at least two microphones and a signal processing means wherein the signal processing means comprises the superdirective beamformer with fixed superdirective filters.

Abstract: This system has a plurality of microphones oriented in several directions and mounted in a support so that each one is flush with or disposed on a rigid surface, the various microphones being connected to a signal processing device that processes the signals coming from the microphones to provide a noise source hologram, namely a distribution of the acoustic pressures or intensities at various points on a given surface.

Abstract: A beamsteerer for broadband energy source location, comprising an array of sensors, each for generating a signal vector within one of a plurality of sectors in a domain of interest, and a beamformer for receiving and multiplying each signal vector by a set of optimal weight vectors to generate a plurality of beampatterns, each of the beampatterns being characterized by a null having high angular resolution, and detecting the source by selecting a maximum steering index.

Abstract: The invention concerns a method for reproducing spatial impression of existing spaces in multichannel or binaural listening. It consists of following steps/phases: a) Recording of sound or impulse response of a room using multiple microphones, b) Time- and frequency-dependent processing of impulse responses or recorded sound, c) Processing of sound to multichannel loudspeaker setup in order to reproduce spatial properties of sound as they were in recording room, and (alternative to c), d) Processing of impulse response to multichannel loudspeaker setup, and convolution between rendered responses and an arbitrary monophonic sound signal to introduce the spatial properties of the measurement room to the multichannel reproduction of the arbitrary sound signal, and is applied in sound studio technology, audio broadcasting, and in audio reproduction.

Abstract: A method and system for mapping acoustic sources determined from a phased microphone array. A plurality of microphones are arranged in an optimized grid pattern including a plurality of grid locations thereof. A linear configuration of N equations and N unknowns can be formed by accounting for a reciprocal influence of one or more beamforming characteristics thereof at varying grid locations among the plurality of grid locations. A full-rank equation derived from the linear configuration of N equations and N unknowns can then be iteratively determined. A full-rank can be attained by the solution requirement of the positivity constraint equivalent to the physical assumption of statically independent noise sources at each N location.

Abstract: Systems and methods for detection and analysis of amplitude modulation of underwater sound employ a product of a time delayed first electrical signal with a second electrical signal to generate a summed-product signal. The time delayed first electrical signal and the second electrical signal have an amplitude modulation indicative of characteristics of a vessel propeller. The summed-product signal is analyzed to detect a vessel and to determine the characteristics of the vessel propeller. In some arrangements a plurality of summed-product signals are analyzed to also determine a depth of the detected vessel.

Abstract: Estimating the proximity of an audio source is accomplished by transforming audio signals from a plurality of sensors to frequency domain. The amplitudes of the transformed audio signals are then determined. The proximity of the audio source is determined based on a comparison of the frequency domain amplitudes. This estimation permits a device to differentiate between relatively distant audio sources and audio sources at close proximity to the device. The technique can be applied to mobile handsets, such as cellular phones or PDAs, hands-free headsets, and other audio input devices. Devices taking advantage of this “close proximity” detection are better able to suppress background noise and deliver an improved user experience.

Abstract: Systems for analyzing acoustic waves are provided. An exemplary system includes a housing, multiple acoustic passageways and an acoustic sensor. The housing defines an interior cavity. The multiple acoustic passageways communicate acoustically between the interior cavity and an exterior of the housing. Each of the acoustic passageways has an inlet port and an outlet port, with each outlet port being located within the housing to direct a portion of an acoustic wave to the interior cavity. The acoustic sensor is mounted within the interior cavity and is operative to receive portions of an acoustic wave directed to the interior cavity by the acoustic passageways. The acoustic sensor also is operative to provide information such as direction of arrival information corresponding to a composite waveform formed by acoustic interference, within the interior cavity, of the portions of the acoustic wave. Methods and other systems also are provided.

Abstract: The invention is directed to a method comprising receiving input signals emanating from at least two microphone arrays each comprising at least two microphones, processing the input signals of each microphone array by a beamformer to determine temporal and spatial information about the input signals of each microphone array.

Abstract: Methods and systems for acquiring spectral and velocity information with a multi-dimensional array are provided. For example, a dedicated receive aperture is formed at a multi-dimensional array for steered continuous wave imaging. Other elements not within the dedicated receive aperture are used for transmitting continuous waves or transmitting and receiving pulsed waveforms in other modes of imaging. As another example, switches or other structures are provided for selecting between a plurality of possible apertures for a steered continuous wave aperture. The selection is performed in response to a configuration of an ultrasound system, such as selection of a focal location or steer direction. The aperture is then used for either transmit or receive operations of steered continuous wave imaging. As yet another example, at least part of the steered continuous wave beamformer is provided within a transducer assembly.

Abstract: A method for reducing noise associated with an audio signal received through a microphone sensor array is provided. The method initiates with enhancing a target signal component of the audio signal through a first filter. Simultaneously, the target signal component is blocked by a second filter. Then, the output of the first filter and the output of the second filter are combined in a manner to reduce noise without distorting the target signal. Next, an acoustic set-up associated with the audio signal is periodically monitored. Then, a value of the first filter and a value of the second filter are both calibrated based upon the acoustic set-up. A system capable of isolating a target audio signal from multiple noise sources, a video game controller, and an integrated circuit configured to isolate a target audio signal are included.

Abstract: Methods and ultrasonic sensor systems are presented for sensing distance in which an acoustic output signal is directed toward a target and acoustic input signals are received from the direction of the target, and the received signal is compared with a selected one of a plurality of different receiver threshold curves to identify the acoustic wave travel time to the target and back while differentiating between the target and acoustic waves reflected more than once and waves reflecting off obstructions between the sensor and the target, where the user selectable receiver threshold curves are tailored for different target and/or obstruction conditions.

Abstract: Frequency-steered acoustic arrays transmitting and/or receiving multiple, angularly dispersed acoustic beams are used to generate 2D and 3D images. Input pulses to the arrays are generally non-linear, frequency-modulated pulses. Frequency-steered acoustic arrays may be provided in one-dimensional linear and two dimensional planar and curvilinear configurations, may be operated as single order or multiple order arrays, may employ periodic or non-periodic transducer element spacing, and may be mechanically scanned to generate 2D and 3D volumetric data. Multiple imaging fields of view may generated in different directions by switching the polarity of phase-shifted array transducer elements. Multiple frequency-steered arrays arranged in an X-configuration provide a wide, contiguous field of view and multiple frequency steered arrays arranged in a T-configuration provide orthogonally oriented fields of view.

Abstract: An autonomous sonar system and method provide an arrangement capable of beamforming in three dimensions, detecting loud targets, adaptively beamforming in three dimensions to avoid the loud targets, detecting quiet targets, localizing the loud or quiet targets in range, bearing, and depth, detecting modulation of noise associated with propellers of the loud or quiet targets, generating three dimensional tracks of the loud or quiet targets in bearing, range and depth, making classification of the loud or quiet targets, assigning probabilities to the classifications, and generating classification reports according to the classifications for communication to a receiving station, all without human assistance.

Abstract: The invention as disclosed is of a combined acoustic pressure and acoustic vector sensor array, where multiple acoustic pressure sensors are integrated with an acoustic vector sensor in a towed array as a means of resolving the left-right ambiguity of the multiple acoustic pressure sensors.

Abstract: Systems and methods for detection and analysis of amplitude modulation of underwater sound employ a product of a time delayed first electrical signal with a second electrical signal to generate a summed-product signal. The time delayed first electrical signal and the second electrical signal have an amplitude modulation indicative of characteristics of a vessel propeller. The summed-product signal is analyzed to detect a vessel and to determine the characteristics of the vessel propeller. In some arrangements a plurality of summed-product signals are analyzed to also determine a depth of the detected vessel.

Abstract: A connection apparatus and method for controlling an ultrasound probe are provided. The ultrasound probe includes a first chamber, a second chamber and a sealing member between the first and second chambers. The ultrasound probe further includes a connection member within the second chamber having a rigid portion and a flexible portion, with the rigid portion forming at least part of the sealing member.

Abstract: A connection apparatus and method for controlling an ultrasound probe are provided. The ultrasound probe includes a first chamber, a second chamber, a sealing member between the first and second chambers and a flexible connection member within each of the first and second chambers. The ultrasound probe further comprises a rigid connection interface forming at least part of the sealing member and connecting the flexible connection member in the first chamber with the flexible connection member in the second chamber.

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: Systems of generating and manipulating an ultrasound beam are disclosed. The system uses selective sets of ultrasound elements to generate an ultrasound beam, scanning the beam over a series of ultrasound elements in order to collected echo data covering an area, and generating an image from the resulting data. The scanning process includes shifting the set of ultrasound elements used to form the ultrasound beam by more than one ultrasound element (block-switching) between each step in the scanning process. This is accomplished without loss of image resolution by using area-forming techniques. The block-switching technique enables use of cross-correlation methods during image construction.

Abstract: The ability to combine multiple audio signals captured from the microphones in a microphone array is frequently used in beamforming systems. Typically, beamforming involves processing the output audio signals of the microphone array in such a way as to make the microphone array act as a highly directional microphone. In other words, beamforming provides a “listening beam” which points to a particular sound source while often filtering out other sounds. A “generic beamformer,” as described herein automatically designs a set of beams (i.e., beamforming) that cover a desired angular space range within a prescribed search area. Beam design is a function of microphone geometry and operational characteristics, and also of noise models of the environment around the microphone array. One advantage of the generic beamformer is that it is applicable to any microphone array geometry and microphone type.

Abstract: A software system and method is presented that calculates the sensor directivity at all spatial angles, and stores these values in a two dimensional matrix. These values are then used as additional “weighting” coefficients in the equation for the pressure detected by an array of directive sensors. The azimuthal and polar angles of a particular sensor normal vector are used to “rotate” the sensor directivity matrix to account for the angular orientation of each sensor within the array. The software system receives as input sensor and array geometry, shading functions, sensor shape, array structure baffling, steering, and shading. A sensor can have the shape of a point, line, plane, volume, baffled ring, or circular plane piston. The sensors within the array can either be baffled by the array structure or retain their free field directivity response, and array as a whole can have steering or no steering.

Abstract: A parametric audio system having increased bandwidth for generating airborne audio signals with reduced distortion. The parametric audio system includes a modulator for modulating an ultrasonic carrier signal with a processed audio signal, a driver amplifier for amplifying the modulated carrier signal, and an array of acoustic transducers for projecting the modulated and amplified carrier signal through the air along a selected projection path to regenerate the audio signal. Each of the acoustic transducers in the array is a membrane-type transducer. Further, the acoustic transducer array is a phased array capable of electronically steering, focusing, or shaping one or more audio beams.

Abstract: The present invention relates to a digital beamformer in ultrasound imaging after the process of receiving a digital echo signal. Provided herein is a time multiplexed device and method for beamforming. The device includes an echo data write/read unit, a multi-beam processing unit, a channel data summing unit, and a parameter distributing and synchronizing unit for distributing and synchronizing respective receive parameters to the echo data write/read unit and the multi-beam processing unit through a parameter load bus for each of the beams in the channel.

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: An apparatus for determining a position includes a source which transmits a signal having a source position and a transmission time coded therein. A sensor having a directional beam pattern is positioned at the location of interest. A signal processor steers the directional beam pattern of the sensor in order to determine the direction to the signal source. A sensor processor uses a clock to find a receipt time of the signal. The transmission time and source position is decoded from the signal. The position of interest is calculated from the receipt time, transmission time, direction, and source position. A method is also provided.

Abstract: A parallel GSC structure is provided by which an adaptive process is performed by a plurality of beamformers in parallel in such a way that they present a common response to the equalization signal that varies over time in an optimal manner with respect to the statistics of the steering vectors.