598 results about "Acoustic property" patented technology

Systems and methods for assessment of tissue properties, noninvasively, by acquiring data relating to at least one aspect of intrinsic and/or induced tissue displacement, or associated biological responses, are provided. Data relating to tissue displacement and associated biological changes may be acquired by detecting acoustic properties of tissue using ultrasound interrogation pulses, preferably in a scatter or Doppler detection mode. Based on this data, tissue properties are assessed, characterized and monitored. Specific applications for systems and methods of the present invention include non-invasive assessment and monitoring of intracranial pressure (ICP), arterial blood pressure (ABP), CNS autoregulation status, vasospasm, stroke, local edema, infection and vasculitus, as well as diagnosis and monitoring of diseases and conditions that are characterized by physical changes in tissue properties. Methods and systems for localizing physiological condition(s) and/or biological response(s), such as pain, by targeting and selectively probing tissues using the application of focused ultrasound are also provided.

An active noise cancellation system includes a series of features for more effective cancellation, greater reliability, and improved stability. A particular feature adapted for headset systems includes locating a residual microphone radially offset from the center of a sound generator to detect a signal more similar to that incident upon the eardrum of the user. In addition, an open back headset design includes perforations on the side of the headset instead of the back, so that the perforations are less susceptible to inadvertent blockage. The system also includes a mechanism for detecting changes in the acoustic characteristics of the environment that may be caused, for example, by pressure exerted upon the earpieces, and that may destabilize the cancellation system. The system automatically responds to such changes, for example, by reducing the gain or the frequency response of the system to preserve stability. The system further includes other methods for detecting imminent instability and compensating, such as detecting the onset of signals within enhancement frequencies characteristic of the onset of instability, and adjusting the gain or frequency response of the system or suppressing the enhanced signals. The system further includes a mechanism for conserving battery life by turning the system off when sound levels are low, or adjusting the power supply to the system to correspond to the current power requirements of the system.

One embodiment of the present invention provides a system that manages the acoustic noise produced by a device. During operation, the system receives a set of acoustic characteristics for the device. The system then uses these acoustic characteristics to estimate the acoustic noise being generated by the device. Next, the system uses the estimated acoustic noise to adjust a setting in the device to manage the acoustic noise produced by the device.

Methods and apparatus are disclosed for applying acoustic energy to the skin. Acoustic waveguides with elements of varying thickness or shape are disclosed which deliver energy to more than one depth below a surface of the skin substantially simultaneously. The invention is especially useful with devices that focus ultrasound energy by condensing a propagating wavefront. The invention compensates for the mismatch in acoustic properties of the device's waveguide and the biological tissue that typically cause portions of the collapsing wavefront to lag behind other portions and, thereby, limit the focusing capabilities of acoustic treatment devices.

Systems and methods for determining ICP based on parameters that can be measured using non-invasive or minimally invasive techniques are provided, wherein a non-linear relationship is used to determine ICP based on one or more variable inputs. The first variable input relates to one or more properties of a cranial blood vessel and/or blood flow, such as acoustic backscatter from an acoustic transducer having a focus trained on a cranial blood vessel, flow velocity in a cranial blood vessel, and the like. Additional variables, such as arterial blood pressure (ABP), may be used in combination with a first variable input relating to one or more properties of a cranial blood vessel, such as flow velocity of the middle cerebral artery (MCA) to derive ICP using a non-linear relationship. Methods and systems for locating target areas based on their acoustic properties and for acoustic scanning of an area, identification of a target area of interest based on acoustic properties, and automated focusing of an acoustic source and/or detector on a desired target area are also provided. Acoustic transducer assemblies are described.

An acoustic control system for an electronic musical instrument, which can obtain optimal acoustic characteristics irrespective of whether or not a low range speaker is attached to a body thereof. A piano body has high range and mid range speakers. A low range speaker is removable from the body. A stand-excluding switch designates a first speaker-use mode for using the high range and mid range speakers alone, and a stand-including switch designates a both speaker-use mode for using both the high range and mid range speakers and the low range speaker. A ROM stores stand-excluding and stand-including factors for setting acoustic characteristics in the two modes, respectively. A CPU reads one of the factors, which corresponds to the mode designated by the stand-excluding or stand-including switch. A tone generator circuit generates a musical tone signal to be reproduced in one of the modes, based on the read factor.

A method for assaying a component of a localized region of interest in a body comprising: illuminating the region with at least one pulse of radiation having a wavelength at which the radiation is absorbed by the component to generate a change in an acoustic property of the region; transmitting ultrasound so that it is incident on the region; measuring at least one effect of the change on the incident ultrasound; using the measured at least one effect to determine an absorption coefficient for the radiation in the region; and using the determined absorption coefficient to determine concentration of the component in the region.

Adaptive time/frequency-based audio encoding and decoding apparatuses and methods. The encoding apparatus includes a transformation & mode determination unit to divide an input audio signal into a plurality of frequency-domain signals and to select a time-based encoding mode or a frequency-based encoding mode for each respective frequency-domain signal, an encoding unit to encode each frequency-domain signal in the respective encoding mode, and a bitstream output unit to output encoded data, division information, and encoding mode information for each respective frequency-domain signal. In the apparatuses and methods, acoustic characteristics and a voicing model are simultaneously applied to a frame, which is an audio compression processing unit. As a result, a compression method effective for both music and voice can be produced, and the compression method can be used for mobile terminals that require audio compression at a low bit rate.

An audio correcting apparatus includes a speaker provided on a television apparatus, a microphone provided on a remote controller, an identifying unit which identifies an acoustic characteristic from the speaker to the microphone, and an acoustic characteristic setting unit having the acoustic characteristic. A signal obtained by allowing an audio signal input to the speaker to pass through the acoustic characteristic setting unit, and a signal representing ambient noise are input to an audio-correcting filter and a loudness-compensation-gain calculating unit. Based on both signals, the sound pressure level of sound output from the speaker is corrected so that the sound output from the speaker is clearly heard when reaching the user without being affected by the ambient noise.

A two-dimensional ultrasound phased array transducer includes an acoustic backing, a first circuit, which may be a flexible circuit disposed over the acoustic backing or a ground plane, an acoustically absorptive interface layer disposed over the flexible circuit, and a piezoelectric layer disposed over the interface layer. A matching layer may be disposed over the piezoelectric layer, and a second circuit, which may be a ground plane or a flexible circuit, may be disposed over the matching layer. The piezoelectric layer and the matching layer are diced by forming kerfs extending through these layers and at least partially into the interface layer. Extending the kerfs into the interface layer reduces cross-talk between elements, electrically isolates the elements, and facilitates manufacturing by reducing the precision required in controlling the depth of the cut. The acoustically absorptive interface layer may have acoustic properties similar to the backing material and may be formed of the same material as the backing material. Electrical interconnection between the piezoelectric elements and the first circuit is provided through the interface layer. The electrical connection may be formed by laser drilled vias in the interface layer, coated with gold or another suitable material.

A headphone apparatus the acoustic characteristics of which can be easily changed in accordance with the type of content the user wants to listen to. An adjusting knob provided on the housing of the headphone apparatus is moved, closing an opening made in a baffle. As a result, an acoustic space defined by the inner side of the baffle and an ear pad is more closed, thus increasing the level of low-frequency components of the sound generated by a sound-generating unit. Hence, the acoustic characteristics of the apparatus can be easily changed as the user desires.

The present invention provides a voice coder for voice communication that employs a multi-microphone system as part of an improved approach to enhancing signal quality and improving the signal to noise ratio for such voice communications, where there is a special relationship between the position of a first microphone and a second microphone to provide the communication device with certain advantageous physical and acoustic properties. In addition, the communication device can have certain physical characteristics, and design features. In a two microphone arrangement, the first microphone is located in a location directed toward the speech source, while the second microphone is located in a location that provides a voice signal with significantly lower signal-to-noise ratio (SNR).

In industrial sensing applications at least one parameter of at least one fluid in a pipe 12 is measured using a spatial array of acoustic pressure sensors 14,16,18 placed at predetermined axial locations x1, x2, x3 along the pipe 12. The pressure sensors 14,16,18 provide acoustic pressure signals P₁(t), P₂(t), P₃(t) on lines 20,22,24 which are provided to signal processing logic 60 which determines the speed of sound a_mix of the fluid (or mixture) in the pipe 12 using acoustic spatial array signal processing techniques with the direction of propagation of the acoustic signals along the longitudinal axis of the pipe 12. Numerous spatial array-processing techniques may be employed to determine the speed of sound a_mix. The speed of sound a_mix is provided to logic 48, which calculates the percent composition of the mixture, e.g., water fraction, or any other parameter of the mixture, or fluid, which is related to the sound speed a_mix. The logic 60 may also determine the Mach number Mx of the fluid. The acoustic pressure signals P₁(t), P₂(t), P₃(t) measured are lower frequency (and longer wavelength) signals than those used for ultrasonic flow meters, and thus is more tolerant to inhomogeneities in the flow. No external source is required and thus may operate using passive listening. The invention will work with arbitrary sensor spacing and with as few as two sensors if certain information is known about the acoustic properties of the system. The sensor may also be combined with an instrument, an opto-electronic converter and a controller in an industrial process control system.

An apparatus and methods for acoustically analyzing a fluid sample and determining one or more properties of the sample are disclosed by the present invention. The apparatus comprises a chamber, a transmitter positioned within the chamber for transmitting an acoustic signal through the fluid, a reflector movably positioned within the fluid inside the chamber for reflecting the acoustic signal, and a receiver positioned within the chamber for detecting reflections of the acoustic signal. The methods employ the use of a transmitter, a reflector movably positioned within the fluid inside the chamber, and a receiver to characterize the fluid sample based on one or more of its acoustic properties.

In thermal pulse flow measurements a relatively small bolus of flowing fluid is heated or cooled and the time required for the bolus to move downstream a known distance is measured. In many fluids, changing the temperature changes the acoustic transmission properties of the bolus from those of the rest of the fluid, so the bolus can be detected when it intersects an acoustic beam. The use of an acoustic beam or beams, which are usually defined between acoustic transmitting receiving transducers, typically provides a high frequency carrier which is modulated by the change in acoustic properties of the bolus when it passes between the two transducers. When compared to conventional thermal measurements, this acoustic approach provides faster response times and can thus be used for measuring higher flow rates.

Invention relates to a novel method of and system for ranging between an acoustic source carried on an unmanned or autonomous undersea mobile vehicle (UUV or AUV) and preferably a plurality of hydrophone receivers remotely deployed from the vehicle in predetermined patterns, generally suspended from sonobuoys equipped with above-the-sea relay radio transmitting antennas, and with time synchronization provided amongst the source and the receivers, wherein the time delay from the transmissions of the source is measured by utilizing special signal processing, enabling range to be measured in close to real time by determining the product of the sound velocity and the measured time delay, and with the process continually and periodically being repeated throughout the duration of the vehicle run. Given the range, the system may then be used to measure the acoustical properties of and/or receiver system performance in the sea or other water body, such as transmission or propagation loss TL, channel impulse response, bottom geoacoustic properties, source level, receiver sensitivity calibration, sonar operator readiness and sonar receiver performance and the like. Further, in situ measured data can be assimilated with models to enable more accurate prediction of the ocean environment than could be obtained from either individually.

The invention relates to a biopsy device, particularly a biopsy device comprising a shaft with a transducer element for providing information about acoustic properties of a material to be analysed, a system of positioning a biopsy device and a method for positioning a biopsy device. The biopsy device may be adapted to take biopsies of different regions of the 5 human body for excluding or detecting abnormalities as cancerous lesions. The biopsy device may be used to measure acoustic properties of the material while inserting the tip portion of the biopsy device into the material to be analysed. The biopsy device may further allow measurement based on elastography.

One or more acoustic transducers of a device may be adjusted based on automatic detection of device proximity to the user. In a mobile telephone, when the user is using the receiver and holding the telephone against his/her ear, if the telephone detects that the user has moved the telephone further from his/her ear, the telephone will raise the receiver volume. Similarly, if the user is using the speaker, the telephone will adjust the speaker volume as user distance from the telephone changes. In another embodiment the telephone may fade between the receiver and the speaker. Volume is not the only acoustic property that could be adjusted according to user proximity. Frequency response is another property that could be adjusted, such as using appropriate electronic filtering, or by turning on another transducer that is not otherwise being used.

A method and apparatus for automatically controlling the operation of a speech recognition system without requiring unusual or unnatural activity of the speaker by passively determining if received sound is speech of the user before activating the speech recognition system. A video camera and microphone are located in a hand-held device. The video camera records a video image of the speaker's face, i.e., of speech articulators of the user such as the lips and/or mouth. The recorded characteristics of the articulators are analyzed to identify the sound that the articulators would be expected to make, as in "lip reading". A microphone concurrently records the acoustic properties of received sound proximate the user. The recorded acoustic properties of the received sound are then compared to the characteristics of speech that would be expected to be generated by the recorded speech articulators to determine whether they match. If so, then the received sound is identified as having emanated from the user the speech recognition system is operated to perform speech recognition of the received sound.

A method and system for enhancing the frequency response of speech signals are provided. An average speech spectral shape estimate is calculated over time based on the input speech signal. The average speech spectral shape estimate may be calculated in the frequency domain using a first order IIR filtering or “leaky integrators.” Thus, the average speech spectral shape estimate adapts over time to changes in the acoustic characteristics of the voice path or any changes in the electrical audio path that may affect the frequency response of the system. A spectral correction factor may be determined by comparing the average speech spectral shape estimate to a desired target spectral shape. The spectral correction factor may be added (in units of dB) to the spectrum of the input speech signal in order to enhance or adjust the spectrum of the input speech signal toward the desired spectral shape, and an enhanced speech signal re-synthesized from the corrected spectrum.