4561 results about "Sound pressure" patented technology

A pile driving control apparatus for a pile driving system includes a hydraulic control system that controls a throttle of a pile driving hammer, and thereby controls an impact velocity of the hammer with a pile. A controller provides a control signal to the hydraulic control system. Based on the control signal, the hydraulic control system controls an impact velocity of the hammer during a subsequent hammer stroke. The controller may determine one or more control parameters such as sound pressure at a sound control location during a hammer stroke, vibration at a vibration control location during a hammer stroke, an impact force imparted to the pile during a hammer stroke, and / or actual pile capacity of the pile, and provide to the hydraulic control system a control signal based on the determined control parameter(s).

In a mobile device, a bone conduction or vibration sensor is used to detect the user's speech and the resulting output is used as the source for a low power Voice Trigger (VT) circuit that can activate the Automatic Speech Recognition (ASR) of the host device. This invention is applicable to mobile devices such as wearable computers with head mounted display, mobile phones and wireless headsets and headphones which use speech recognition for the entering of input commands and control. The speech sensor can be a bone conduction microphone used to detect sound vibrations in the skull, or a vibration sensor, used to detect sound pressure vibrations from the user's speech. This VT circuit can be independent of any audio components of the host device and can therefore be designed to consume ultra-low power. Hence, this VT circuit can be active when the host device is in a sleeping state and can be used to wake the host device on detection of speech from the user. This VT circuit will be resistant to outside noise and react solely to the user's voice.

Apparatus includes generator generating reference signal based on noise emitted from sound source, detector detecting level of reference signal and change in level, unit comparing change with threshold-value range and produce compared result, filter filtering reference signal, adaptive filter having variable filter coefficient, unit updating filter coefficient according to change of level of reference signal for obtaining an updated filter coefficient, unit stopping updating of filter coefficient in response to compared result when change falls outside threshold-value range, unit storing updated filter coefficient each time filter coefficient is updated, generator generating control signal using stored filter coefficient, unit generating control sound based on control signal, microphone detecting synthesis sound pressure of control sound and noise to produce an error signal, and unit setting stored filter coefficient to more accurate coefficient than stored filter coefficient based on error signal, and signal acquired by filtering control signal through filter.

A system is provided for configuring an audio system for a given space. The system may statistically analyze potential configurations of the audio system to configure the audio system. The potential configurations may include positions of the loudspeakers, numbers of loudspeakers, types of loudspeakers, listening positions, correction factors, filters, or any combination thereof. The statistical analysis may indicate at least one metric of the potential configuration including indicating consistency of predicted transfer functions, flatness of the predicted transfer functions, differences in overall sound pressure level from seat to seat for the predicted transfer functions, efficiency of the predicted transfer functions, or the output of predicted transfer functions. The system also provides a methodology for selecting loudspeaker locations, the number of loudspeakers, the types of loudspeakers, correction factors, listening positions, crossover filters or a combination of these schemes in an audio system that has a single listening position or multiple listening positions.

A method for monitoring and reporting sound pressure level exposure for a user of a first communication device (104) is implemented in one embodiment when the device measures a sound pressure level (SPL) of the surrounding environment. The device stores at least the SPL measurement in a memory, producing an SPL exposure record, and displays a visual representation of the SPL exposure record on a display screen (212). In another embodiment, the SPL is measured by a second communication device (102) and combined with a known SPL for an output audio transducer (306) of the second device, producing a user sound exposure level. The user sound exposure level is transmitted to the first communication device. The user is notified when the user sound exposure level exceeds a predetermined threshold. A server (112) may also be used to track SPLs over time and recommend corrective action when exposure limits are exceeded.

It becomes possible to obtain high sound pressure in a high frequency domain by a capacitive ultrasonic transducer which comprises a membrane on which one electrode is formed, a cavity constructed in its backface, and a substrate on which these are mounted and supported and on whose surface an electrode is provided, on a surface in an ultrasonic transmission and reception side, characterized in that the membrane comprises two or more layers, and at least one layer of them comprises a high dielectric constant film.

An active noise cancellation helmet includes a detection unit which detects noise in a helmet body, and a sound outputting unit which outputs a sound for canceling the noise detected by the detection unit. A control signal is generated by processing an output signal of the detection unit through computation. The control signal is amplified by an amplification unit, and applied to the sound outputting unit. A ratio of sound pressures in different frequency ranges is determined on the basis of the output signal of the detection unit. A gain of the amplification unit is adjusted on the basis of the sound pressure ratio so as to approximate a spectrum of the output signal of the detection unit to a predetermined target spectrum.

Flow rate measurement system includes two measurement regions 14,16 located an average axial distance .DELTA.X apart along the pipe 12, the first measurement region 14 having two unsteady pressure sensors 18,20, located a distance X.sub.1 apart, and the second measurement region 16, having two other unsteady pressure sensors 22,24, located a distance X.sub.2 apart, each capable of measuring the unsteady pressure in the pipe 12. Signals from each pair of pressure sensors 18,20 and 22,24 are differenced by summers 44,54, respectively, to form spatial wavelength filters 33,35, respectively. Each spatial filter 33,35 filters out acoustic pressure disturbances P.sub.acoustic and other long wavelength pressure disturbances in the pipe 12 and passes short-wavelength low-frequency vortical pressure disturbances P.sub.vortical associated with the vortical flow field 15. The spatial filters 33,35 provide signals P.sub.as1,P.sub.as2 to band pass filters 46,56 that filter out high frequency signals. The P.sub.vortical -dominated filtered signals P.sub.asf1,P.sub.asf2 from the two regions 14,16 are cross-correlated by Cross-Correlation Logic 50 to determine a time delay .tau. between the two sensing locations 14,16 which is divided into the distance .DELTA.X to obtain a convection velocity U.sub.c(t) that is related to an average flow rate of the fluid (i.e., one or more liquids and / or gases) flowing in the pipe 12. The invention may also be configured to detect the velocity of any desired inhomogeneous pressure field in the flow. The invention may also be combined with an instrument, an opto-electronic converter and a controller in an industrial process control system.

At least one parameter of at least one fluid in a pipe is measured using a spatial array of acoustic pressure sensors placed at predetermined axial locations along the pipe 12. The pressure sensors provide acoustic pressure signals, which are provided to a signal processing system that determines the speed of sound amix of the fluid (or mixture) in the pipe 12 using acoustic spatial array signal processing techniques. Numerous spatial array processing techniques may be employed to determine the speed of sound amix. The speed of sound amix is provided to another logic system that 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 amix. The signal processing system may also determine the Mach number Mx of the fluid. The acoustic pressure signals measured are lower frequency (and longer wavelength) signals than those used for ultrasonic flow meters, and thus are 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 present invention relates to a hearing aid logging data and learning from these data. The hearing aid (10, 100) comprises an input unit (12) converting an acoustic environment to an electric signal; an output unit (16) converting an processed electric signal to a sound pressure; a signal processing unit (14) interconnecting the input and output unit, and generating the processed electric signal from the electric signal according to a setting; a user interface (18) converting user interaction to a control signal thereby controlling the setting; and finally a memory unit (20) comprising a control section storing a set of control parameters associated with the acoustic environment, and a data logger section receiving data from the input unit (12), the signal processing unit (14), and the user interface (18); and wherein said signal processing unit (14) configures the setting according to the set of control parameters and comprises a learning controller adapted to adjust the set of control parameters according to the data in the data logging section.

A method for using an earpiece (800) in a work environment is provided. The earpiece (800) attenuates sound from the work environment to the user's ear. The earpiece (800) includes an ear canal microphone (820) for measuring a sound pressure level in an ear canal of the user. Sound pressure levels are measured periodically while in the work environment. Each measured sound pressure levels is stored in memory (127) of the earpiece with time and location information. The sound pressure level information is downloaded to a database (1704) when the earpiece is removed from the user ear for recharging. The sound pressure level information is analyzed and any potential noise compliance issues in the work environment are identified.

Disclosed herein is a touch panel display apparatus including: a display panel; a movable panel unit including a touch panel and being flat plate-like in shape; a movable support section for supporting the movable panel unit on a support structure so as to permit the movable panel unit to be moved along a plane parallel to the panel surface of the movable panel unit; an actuator configured to drive the movable panel unit to vibrate; an actuator drive control section for outputting an actuator drive signal to drivingly control the actuator; an acoustic pressure generating member configured to generate an acoustic pressure by being driven by the actuator to vibrate; and a superposing section for superposing an acoustic signal on the actuator drive signal outputted by the actuator drive control section.

An apparatus and method provides control over the relative level of audio and background sound for an audio device such as a headset or headphones. An input sound transducer can be provided and configured to convert ambient acoustical pressure into an electrical background signal representing the background sound. Noise cancellation circuitry create a cancellation signal that is the inverse of the electrical background signal. A summing component is coupled to the noise cancellation component and combines the cancellation signal with an audio signal representing audio program content and provides the combined electrical signal to an output sound transducer. A controller is coupled to the noise cancellation component and is configured to control the level of a cancellation signal relative to the audio signal, thereby controlling the mix of cancellation signal and audio signal reaching the output sound transducer.

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 method for fitting a hearing instrument comprises placing the hearing instrument in situ includes receiving an audiogram of the user, determining a target gain for the hearing instrument as a function of the audiogram, exposing a reference sensor located adjacent the hearing instrument to an external speech signal, measuring an external sound pressure level (SPL) via the reference sensor, exposing a probe sensor coupled to the inside of the ear to the output of the hearing instrument while the hearing instrument is in situ, measuring an internal sound pressure level (“SPL”) inside the ear of the user via the probe sensor, determining an offset gain as a function of the external SPL, the target gain and the internal SPL, and automatically adjusting a gain of the hearing instrument according to the offset gain. A system for automatically fitting a hearing impaired person with a digital hearing aid in situ includes a digital hearing aid, a reference volume sensor, a probe sensor, a sound mapping module and a parameter control module.

An active noise control system prevents a continuous muffled sound from being generated as an abnormal sound under a high sound pressure from a speaker when a microphone as a sound detector is covered, and reduces noise immediately when the microphone is uncovered. A first threshold value as an upper limit value and a second threshold value as a lower limit value are provided for the filter coefficient of an adaptive notch filter. When the filter coefficient is greater than the first threshold value, a control sound is faded out according to a forgetting process. When the filter coefficient is smaller than the second threshold value, an adaptive control process is resumed. Even if the microphone is covered, the filter coefficient does not exceed the first threshold value as the upper limit value.

A method and apparatus for processing a composite acoustic signal to reconstruct an acoustic signal that substantially matches a selected one of a plurality of sources. A plurality of microphones positioned at different spatial locations detect. variations in sound pressure level resulting from the activity of a plurality of acoustic sources at different locations. The outputs of the microphones are sampled and digitized, and the resulting digital waveform from each microphone is provided as an input to a corresponding filter bank. The outputs of the filter banks are input to a comparison unit. A comparison control unit generates "signature" information that characterizes each source with respect to the microphones. The comparison unit receives "signature" information of a selected source from the comparison control unit and provides an output to a synthesizer unit which produces a synthesized digital waveform for the selected source. Optionally, the synthesized digital waveform is input to a digital-to-analog (D / A) converter to generate an analog signal of the reconstructed source.

Loudspeaker systems for handheld devices (2400, 2500) such as cellular telephones (100, 1400, 1600, 1900) are provided. The loudspeaker systems include at least one omnidirectional monopole loudspeaker (404, 1504, 1802, 1918, 1920) and at least one dipole loudspeaker (402, 1502, 1804, 1914, 1916). Filtering is used to compensate for differences between electric-to-acoustic signal transfer functions of the loudspeakers in order to achieve destructive interference of emitted audio in one direction and reinforcement in another direction. Various loudspeaker systems that are reconfigurable from directional mode, to omnidirectional mode are provided. Filtering is also used to add a frequency dependent phase to one signal in order to alter a direction of maximum sound pressure level. Two monopole-dipole pairs can be used to produce a stereo effect. The monopole and dipole loudspeakers can be housed in a detachable loudspeaker accessory (200).

To efficiently detect a standing wave generated in a room, a standing-wave detection apparatus for detecting a standing wave in a predetermined space, comprises: a sound-receiving unit adapted to receive a sound generated from a sound source arranged in the predetermined space; a storage unit adapted to store time series sound pressure level data acquired by the sound-receiving unit during movement along a path in the predetermined space; an adjustment unit adapted to adjust the time series sound pressure level data stored in the storage unit, based on an adjustment curve determined using a lower envelope of the time series sound pressure level data stored in the storage unit; and a detection unit adapted to detect an existence position of a standing wave in the predetermined space based on the adjusted time series sound pressure level data.

A flow measuring system is provided that provides at least one of a compensated mass flow rate measurement and a compensated density measurement. The flow measuring system includes a gas volume fraction meter in combination with a coriolis meter. The GVF meter measures acoustic pressures propagating through the fluids to measure the speed of sound αmix propagating through the fluid to calculate at least gas volume fraction of the fluid and / or the reduced natural frequency. For determining an improved density for the coriolis meter, the calculated gas volume fraction and / or reduced frequency is provided to a processing unit. The improved density is determined using analytically derived or empirically derived density calibration models (or formulas derived therefore), which is a function of the measured natural frequency and at least one of the determined GVF, reduced frequency and speed of sound, or any combination thereof. The gas volume fraction (GVF) meter may include a sensing device having a plurality of strain-based or pressure sensors spaced axially along the pipe for measuring the acoustic pressures propagating through the flow.

A multi-layer armature for a moving armature receiver. The armature includes a first armature layer and a displacement region. The first armature layer includes a first surface and a second armature layer having a second surface positioned adjacent to the first surface. The displacement region provides relative displacement between the armature layers. The multi-layer construction of the armature in combination with the displacement region creates considerable design freedom in choosing armature geometry outside conventional bounds posed by the above-mentioned constraint between armature cross-sectional area and its mechanical stiffness. The design freedom can be applied to achieve numerous performance benefits for the moving armature receiver such as higher electroacoustic conversion efficiency, increased maximum sound pressure output or smaller overall length of the multi-layer armature. The smaller length leads to a smaller size of moving armature receivers which is an important performance metric for moving armature receivers for numerous severely size-constrained applications.

A system for self-testing an acoustic monitor includes a memory that stores a reference acoustic power of a reference acoustic signal; a digital signal processor (DSP) that computes a real-time acoustic power spectrum of background noise in a frequency range of about 20 Hz to about 80 kHz; a microphone including analog signal conditioning circuitry that receives an acoustic test signal at a test frequency and sound pressure level (SPL), the SPL including an attenuation level; and a processor that compares a measured acoustic power at the test frequency with a calculated acoustic power of the acoustic test signal, the calculated acoustic power including the reference acoustic power and the attenuation level; where the acoustic monitor includes the DSP in signal communication with the microphone.

A system is provided for configuring an audio system for a given space. The system may statistically analyze potential configurations of the audio system to configure the audio system. The potential configurations may include positions of the loudspeakers, numbers of loudspeakers, types of loudspeakers, listening positions, correction factors, or any combination thereof. The statistical analysis may indicate at least one metric of the potential configuration including indicating consistency of predicted transfer functions, flatness of the predicted transfer functions, differences in overall sound pressure level from seat to seat for the predicted transfer functions, efficiency of the predicted transfer functions, or the output of predicted transfer functions. The system also provides a methodology for selecting loudspeaker locations, the number of loudspeakers, the types of loudspeakers, correction factors, listening positions, or a combination of these schemes in an audio system that has a single listening position or multiple listening positions.

In a sound source display system, an omnidirectional camera captures an image of a monitoring area. A microphone array collects a voice in the monitoring area. A monitoring monitor displays the image of an imaging area captured by the omnidirectional camera. A sound pressure calculator in a directivity control device calculates a sound pressure indicating a source of a sound in the image of the imaging area using voice data of the voice collected by the microphone array. An output controller in the directivity control device compares the sound pressure and threshold values (first threshold value, second threshold value), and causes sound image information in which the sound pressure is converted into visual information according to the result of comparison, to be displayed on the monitoring monitor so as to be superimposed on the image of the imaging area.

An apparatus 10 is provided that measures the speed of sound propagating in a multiphase mixture to determine parameters, such as mixture quality, particle size, vapor / mass ratio, liquid / vapor ratio, mass flow rate, enthalpy and volumetric flow rate of the flow in a pipe or unconfined space, for example, using acoustic and / or dynamic pressures. The apparatus includes a pair of ultrasonic transducers disposed axially along the pipe for measuring the transit time of an ultrasonic signal to propagate from one ultrasonic transducer to the other ultrasonic transducer. A signal process, responsive to said transit time signal, provides a signal representative of the speed of sound of the mixture. An SOS processing unit then provides an output signal indicative of at least one parameter of the mixture flowing through the pipe. The frequency of the ultrasonic signal is sufficiently low to minimize scatter from particle / liquid within the mixture. The frequency based sound speed is determined utilizing a dispersion model to determine the at least one parameter of the fluid flow and / or mixture.

An audio processor device and method is disclosed which measures and provides information relating to the audio level being applied to the ear of a user. The processor device uses a preset or calibrated sensitivity of the applied earphones in combination with an analysis of the audio stream to provide sound-pressure-level or time-weighted exposure information to the user or limit the output when preset levels have been achieved. Also disclosed is the use of microphones, internal or external, to combine an additional audio stream, typically the ambient environment, into the main audio channel.

A semiconductor device includes a substrate, a semiconductor chip having a diaphragm, which vibrates in response to sound pressure variations, and a circuit chip that is electrically connected to the semiconductor chip so as to control the semiconductor chip, wherein the semiconductor chip is fixed to the surface of the circuit chip whose backside is mounted on the surface of the substrate. Herein, a plurality of connection terminals formed on the backside of the semiconductor chip are electrically connected to a plurality of electrodes running through the circuit chip. A ring-shaped resin sheet is inserted between the semiconductor chip and the circuit chip. The semiconductor chip and the circuit chip vertically joined together are stored in a shield case having a mount member (e.g., a stage) and a cover member, wherein connection terminals of the circuit chip are exposed to the exterior via through holes of the stage.