1548results about "Noise generation" patented technology

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 reduces annoyance by managing the acoustic noise produced by a device. During operation, the system receives a set of acoustic characteristics for noise-producing components within the device. The system then uses these acoustic characteristics to estimate the acoustic noise being generated by each of these noise-producing components. Next, the system aggregates this set of acoustic noise estimates to produce an aggregate estimate for the acoustic noise produced by the device. The system then analyzes this aggregate estimate using an acoustic annoyance model to determine the acoustic annoyance level. The system then adjusts a setting in the device to manage the acoustic annoyance level produced by the device.

An active headset system is provided with compatibility with existing socket configurations in a way that minimizes the number of electrical connections. The number of connections between the active headset and the remote noise cancellation circuitry is reduced through the use of a common contact, having a controlled impedance, that serves as an input connection to corresponding terminals of the two earphones of the active headset. According to another aspect of the present invention, the transients associated with plugging in or unplugging stereo jack plugs or up to a seven pin connector into an active headset may be overcome by a transient detector in the noise cancellation circuitry. Yet another aspect of the present invention concerns the powering of the noise cancellation circuitry whether the noise cancellation circuitry is placed inside the active headset or at least partially internal to a remote external device.

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.

A method and system use an alternative sensor signal received from a sensor other than an air conduction microphone to estimate a clean speech value. The estimation uses either the alternative sensor signal alone, or in conjunction with the air conduction microphone signal. The clean speech value is estimated without using a model trained from noisy training data collected from an air conduction microphone. Under one embodiment, correction vectors are added to a vector formed from the alternative sensor signal in order to form a filter, which is applied to the air conductive microphone signal to produce the clean speech estimate. In other embodiments, the pitch of a speech signal is determined from the alternative sensor signal and is used to decompose an air conduction microphone signal. The decomposed signal is then used to determine a clean signal estimate.

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.

Systems and methods for providing single microphone noise suppression fallback are provided. In exemplary embodiments, primary and secondary acoustic signals are received. A single microphone noise estimate may be generated based on the primary acoustic signal, while a dual microphone noise estimate may be generated based on the primary and secondary acoustic signals. A combined noise estimate based on the single and dual microphone noise estimates is then determined. Using the combined noise estimate, a gain mask may be generated and applied to the primary acoustic signal to generate a noise suppressed signal. Subsequently, the noise suppressed signal may be output.

A system and method are provided for smoothing the in-room acoustic magnitude response of an audio reproduction system. An in-room acoustic magnitude response analysis is performed to determine a room resonance induced peak associated with an audio signal. A replica of the audio signal is filtered at the room resonance induced peak. The filtered replica signal is added with the audio signal. Through this, smoothing of the room resonance induced peak may be achieved, such that a subjective impression of transient response and dynamics of the audio signal are preserved.

An active noise cancellation system that reduces, at a listening position, power of a noise signal radiated from a noise source to the listening position. The system includes an adaptive filter, at least one acoustic actuator and a signal processing device. The adaptive filter receives a reference signal representing the noise signal, and provides a compensation signal. The at least one acoustic actuator radiates the compensation signal to the listening position. The signal processing device evaluates and assesses the stability of the adaptive filter.

Active noise control system and method for controlling an acoustic noise generated by a noise source at a listening location, in which system and method sound is picked up in the surroundings of the listening location by a sound sensor; an electrical noise signal which corresponds to the acoustic noise of the noise source is generated and filtered adaptively in accordance with control signals. The adaptively filtered noise signal is irradiated into the surroundings of the listening location by a sound reproduction device, where a secondary path transfer function extends between the sound reproduction device and sound sensor. The noise signal is filtered with a transfer function that models the secondary path transfer function. The signals which provided by the sound sensor after first filtering serve as control signals for the adaptive filtering.

Ear protecting device with a sealing section for acoustically sealing the meatus of a human, includes a sound generator with a sound outlet for being directed toward the user meatus; an inner microphone with a sound inlet from the meatus, arranged for measuring the resulting sound field in the meatus; connected to an electronics unit including a sound analyser coupled to the inner microphone, for analyzing sound characteristics of the resulting sound field in the meatus, producing analyzed sound characteristics; storing part in the electronics unit for storing measured predetermined sound characteristics of a properly functioning ear protecting device; a comparing part in the electronics unit for comparing the inner microphone analyzed sound characteristics with the stored measured predetermined sound characteristics; indicating part coupled to the comparing part for being activated if the analyzed sound characteristics differ significantly from the predetermined sound characteristics.

Spatial noise suppression for audio signals involves generating a ratio of powers of difference and sum signals of audio signals from two microphones and then performing noise suppression processing, e.g., on the sum signal where the suppression is limited based on the power ratio. In certain embodiments, at least one of the signal powers is filtered (e.g., the sum signal power is equalized) prior to generating the power ratio. In a subband implementation, sum and difference signal powers and corresponding the power ratio are generated for different audio signal subbands, and the noise suppression processing is performed independently for each different subband based on the corresponding subband power ratio, where the amount of suppression is derived independently for each subband from the corresponding subband power ratio. In an adaptive filtering implementation, at least one of the audio signals can be adaptively filtered to allow for array self-calibration and modal-angle variability.

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.

Speakers deployed in a space and divided into groups associated with different zones produce a mix of sounds that create internal noise pollution and, combined with external noise, an unpleasant environment for occupants. The present invention contemplates sound systems and methods for creating personalized sound zones to address these and related problems.

An active control of an unwanted noise signal at a listening site radiated by a noise source uses a reference signal that has an amplitude and / or frequency such that it is masked for a human listener at the listening site by the unwanted noise signal and / or a wanted signal present at the listening site in order to adapt for the time-varying secondary path in a real time manner such that a user doesn't fell disturbed by an additional artificial noise source.

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.

Improved vocal signals are obtained in headsets and similar devices by including a microphone inside a chamber formed at least in part by the wearer's ear. This second microphone provides a reduced noise input signal. The reduced noise signal is corrected by input from another microphone, located outside the chamber. This correction can include echo cancellation, spectral shaping, frequency extension, and the like.

There is provided a noise cancellation system, comprising: an input for a digital signal, the digital signal having a first sample rate; a digital filter, connected to the input to receive the digital signal; a decimator, connected to the input to receive the digital signal and to generate a decimated signal at a second sample rate lower than the first sample rate; and a processor. The processor comprises: an emulation of the digital filter, connected to receive the decimated signal and to generate an emulated filter output; and a control circuit, for generating a control signal on the basis of the emulated filter output. The control signal is applied to the digital filter to control a filter characteristic thereof.

A method and system for enhancing a target sound signal from multiple sound signals is provided. An array of an arbitrary number of sound sensors positioned in an arbitrary configuration receives the sound signals from multiple disparate sources. The sound signals comprise the target sound signal from a target sound source, and ambient noise signals. A sound source localization unit, an adaptive beamforming unit, and a noise reduction unit are in operative communication with the array of sound sensors. The sound source localization unit estimates a spatial location of the target sound signal from the received sound signals. The adaptive beamforming unit performs adaptive beamforming by steering a directivity pattern of the array of sound sensors in a direction of the spatial location of the target sound signal, thereby enhancing the target sound signal and partially suppressing the ambient noise signals, which are further suppressed by the noise reduction unit.

Provided is an earphone, including an earphone body and an earphone cap. The earphone body includes an earphone shell forming a sound cavity and disposing a sound-guiding pipe. The sound-guiding pipe disposes at least one through hole. The earphone cap is rotatably mounted on the earphone body and includes a main body and a barrel. The barrel is located on the inside of the main body and arranged to encompass the sound-guiding pipe. The barrel disposes at least one opening, which is arranged to be aligned or misaligned with the part or whole of the through hole. The earphone of the present invention can make the opening be aligned or misaligned with and the part or whole of the through hole by rotating the earphone cap, thereby varying the pressure of the sound cavity to attain the object of adjusting sound effect.

An array of microphones utilizes two sets of two microphones for noise suppression. A primary microphone and secondary microphone of the three microphones may be positioned closely spaced to each other to provide acoustic signals used to achieve noise cancellation. A tertiary microphone may be spaced with respect to either the primary microphone or the secondary microphone in a spread-microphone configuration for deriving level cues from audio signals provided by tertiary and the primary or secondary microphone. Signals from two microphones may be used rather than three microphones. The level cues are expressed via an inter-microphone level difference (ILD) which is used to determine one or more cluster tracking control signals. The ILD based cluster tracking signals are used to control the adaptation of null-processing noise cancellation modules. A noise cancelled primary acoustic signal and ILD based cluster tracking control signals are used during post filtering to adaptively generate a mask to be applied against a speech estimate signal.

A virtual speaker demonstration system is disclosed that permits a retail outlet to use a reference speaker to demonstrate the performance of multiple different demonstration speakers. A user interface permits a user to select a demonstration speaker and signal processing is performed so that the output from the reference speaker simulates the output of the selected demonstration speaker. The invention provides benefits to all three of the consumer, the retailer, and the manufacturer. The consumer can listen to and compare multiple demonstration speakers easily and conveniently from the same reference speaker. The retailer to use a single (or few) reference speaker to demonstrate the performance of multiple demonstration speakers, saving costs and space. The manufacturer to be able to display and demonstrate to consumers a broader range of the manufacturer's product line.

A noise canceling and communication system is described. An in-ear device is adapted to fit in the ear canal of a device user. A passive noise reduction element reduces external noise entering the ear canal. An external microphone senses an external acoustic signal outside the ear canal to produce a representative external microphone signal. An internal microphone senses an internal acoustic signal proximal to the tympanic membrane to produce a representative internal microphone signal. One or more internal sound generators produce a noise cancellation signal and an acoustic communication signal, both directed towards the tympanic membrane. A probe tube shapes an acoustic response between the internal sound generator and the internal microphone to be relatively constant over a wide audio frequency band. An electronics module is located externally of the ear canal and in communication with the in-ear device for processing the microphone signals using a hybrid feed forward and feedback active noise reduction algorithm to produce the noise cancellation signal. The noise reduction algorithm includes a modeling component based on a transfer function associated with the internal sound generator and at least one of the microphones to automatically adjust the noise cancellation signal for fit and geometry of the ear canal of the user. The communication component also includes a modeling component based on a transfer function associated with the internal sound generator and at least one of the microphones to automatically adjust the communication signal for fit and geometry of the ear canal of the user and to assure that the communication signal does not interfere with the noise reduction algorithm and that the noise cancellation signal does not interfere with passing of the communication signal.

The loudspeaker device according to the present invention comprises a loudspeaker; a feedforward processing section for performing feedforward processing on an electric signal to be inputted to the loudspeaker based on a preset filter coefficient so that non-linear distortion which occurs from the loudspeaker is removed; and a feedback processing section for detecting vibration of the loudspeaker, and performing feedback processing on an electric signal concerning the vibration with respect to the electric signal to be inputted to the loudspeaker. The feedback processing section performs feedback processing on the electric signal concerning the vibration so that the non-linear distortion which occurs from the loudspeaker is removed and so that a frequency characteristic concerning the vibration of the loudspeaker becomes a predetermined frequency characteristic.

An adaptive signal processing system eliminates noise from input signals while retaining desired signal content, such as speech. The resulting low noise output signal delivers improved clarity and intelligibility. The low noise output signal also improves the performance of subsequent signal processing systems, including speech recognition systems. An adaptive beamformer in the signal processing system consistently updates beamforming signal weights in response to changing microphone signal conditions. The adaptive weights emphasize the contribution of high energy microphone signals to the beamformed output signal. In addition, adaptive noise cancellation logic removes residual noise from the beamformed output signal based on a noise estimate derived from the microphone input signals.

A system and method for noise cancellation is disclosed herein generally having at least one microphone, a processor, and a speaker array. The processor may contain an adaptive filtering feature, to separate desirable sound from undesirable sound. The undesirable sound may be suppressed by the processor by creating a canceling waveform and transmitting the canceling waveform to the user's head via a speaker array. The desirable sound may be enhanced by increasing the amplitude of the waveform and transmitting the amplified waveform to the user's head via a speaker array. The sound transmitted to the user's head by the system may be localized using an image tracking subsystem and speaker array. The image tracking system may track the actual visible image or alternatively a heat (e.g. infrared) image of the user's head.

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.

A sound masking system and method for providing sound masking in networked workstations or offices. The sound masking system comprises a communication network, a plurality of sound masking devices adapted to emit a sound masking signal, and a remote controller. One or more of the sound masking devices includes a controller, and the controller includes an interface for receiving information from the communication network and a component for controlling or adjusting output characteristics associated with the sound masking signal, such as volume. The controller includes a component responsive to a user input for setting or varying output characteristics associated with the sound masking signal.

The present invention relates to a telephony device comprising a near-mouth microphone (M1) for picking up an input acoustic signal including the speaker's voice signal (S1) and an unwanted noise signal (N1,D1), a far-mouth microphone (M2) for picking up an unwanted noise signal (N2,D2) in addition to the near-end speaker's voice signal (S2), said speaker's voice signal being at a lower level than the near-mouth microphone, and an orientation sensor for measuring an orientation indication of said mobile device. The telephony device further comprises an audio processing unit comprising an adaptive beamformer (BF) coupled to the near-mouth and far-mouth microphones, including spatial filters for spatially filtering the input signals (z1,z2) delivered by the two microphones, and a spectral post-processor (SPP) for post-processing the signal delivered by the beam-former so as to separate the desired voice signal from the unwanted noise signal so as to deliver the output signal (y).

The disclosed methods and devices incorporate a novel expandable bubble portion which provides superior fidelity to a listener while minimizing listener fatigue. The expandable bubble portion may be expanded through the transmission of low frequency audio signals or the pumping of a gas to the expandable bubble portion. In addition, embodiments of the acoustic device may be adapted to consistently and comfortably fit to any ear, providing for a variable, impedance matching acoustic seal to both the tympanic membrane and the audio transducer, respectively, while isolating the sound-vibration chamber within the driven bubble. This reduces the effect of gross audio transducer vibration excursions on the tympanic membrane and transmits the audio content in a manner which allows the ear to utilize its full inherent capabilities.