Abstract: Adaptive noise cancellation systems and methods comprise a reference sensor operable to sense environmental noise and generate a corresponding reference signal, an error sensor operable to sense noise in a noise cancellation zone and generate a corresponding error signal, a noise cancellation filter operable to receive the reference signal and generate an anti-noise signal to cancel the environmental noise in the cancellation zone, an adaptation module operable to receive the reference signal and the error signal and adaptively adjust the anti-noise signal, and a transient activity detection module operable to receive the reference signal, detect a transient noise event and selectively disable the adaptation module during the detected transient noise event.

Abstract: Two subsystems, each including a microphone, a speaker, a canceling sound-generating adder, an error-computing adder, and two adaptive filters and two auxiliary filters that accept two noises as input, are provided in correspondence with two cancellation positions. Each canceling sound-generating adder adds together the outputs from the adaptive filters and outputs the result to the speaker of each subsystem. Each error-computing adder adds together the output from the microphone of the subsystem and the output from the auxiliary filter of the subsystem, and the result is treated as the error of the adaptive filters of each subsystem. A transfer function is learned in advance and set in each auxiliary filter such that each error computed by each error-computing adder becomes zero (0) when a transfer function in which each noise is canceled at each cancellation position in a predetermined standard acoustic environment is set in each adaptive filter.

Abstract: A signal processing apparatus includes a signal processor configured to: generate a first noise reduction signal using an adaptive filter based on a signal output from a first input apparatus, and cause the generated first noise reduction signal to be output by a first output apparatus; and generate a second noise reduction signal using a feedback filter based on a signal output from a second input apparatus, and cause the generated second noise reduction signal to be output by a second output apparatus. The feedback filter has a fixed feedback coefficient. The first input apparatus is located at a first noise-cancellation target location in an environment, and the second input apparatus is located at a second noise-cancellation target location in the environment, which may be different from the first noise-cancellation target location.

Abstract: Techniques associated with conforming local and remote media characteristics data to target media presentation profiles are described, including a media characteristic configuration module comprising an audio characteristic manager configured to modify an audio portion of media content data received from a source to conform the audio portion to a target audio characteristic indicated in a target configuration profile, and to modify an audio portion of other media content data received from another source to conform the audio portion to the target audio characteristic indicated in the target configuration profile, a communication facility configured to receive media content data from different sources using different data communication protocols, and a storage device configured to store target configuration profiles and data associated with media content.

Abstract: The present invention provides an audio adjustment method and associated audio adjustment device for active noise cancellation. The audio adjustment method includes: broadcasting a single tone having a frequency fk; generating M sets of filtering coefficients regarding the frequency fk, wherein each set of filtering coefficients within the M sets of filtering coefficients includes a combination of an amplitude and a phase, and the M sets of filtering coefficients are different from one another; determining an mth set of filtering coefficients from the M sets of filtering coefficients to minimize energy corresponding to the frequency fk; and adjusting the single tone with the mth set of filtering coefficients to obtain an adjusted single tone corresponding to the frequency fk.

Abstract: From a first microphone, first microphone signals are received that represent first sound waves. From a second microphone, second microphone signals are received that represent second sound waves. In response to the first microphone signals, analog processing is performed to estimate noise in the first sound waves, and first analog signals are generated for cancelling at least some of the estimated noise in the first sound waves. In response to the second microphone signals, digital processing is performed to estimate noise in the second sound waves, and digital information is generated for cancelling at least some of the estimated noise in the second sound waves. The digital information is converted into second analog signals that represent the digital information. The first and second analog signals are combined into third analog signals for cancelling at least some of the estimated noise in the first and second sound waves.

Abstract: An apparatus includes a hybrid adaptive active noise control unit (HAANCU) configured to provide an anti-noise signal to an ear speaker from a reference noise signal of a reference microphone and an error signal of an error microphone, a decimator configured to decimate the reference noise signal and error signal, an adaptive hybrid ANC training unit (AHANCTU) including at least one noise cancellation filter and a filter configured to provide a feedback signal to the at least one noise cancellation, which trains parameters of the AHANCTU based on the decimated reference noise signal, the decimated error signal, and the feedback signal. The apparatus further includes a rate conversion unit configured to up-sample the parameters and update the HAANCU with the up-sampled parameters.

Abstract: An audio signal processing method and apparatus for adaptively adjusting a decorrelator. The method comprises obtaining a control parameter and calculating mean and variation of the control parameter. Ratio of the variation and mean of the control parameter is calculated, and a decorrelation parameter is calculated based on the said ratio. The decorrelation parameter is then provided to a decorrelator.

Abstract: A method for detecting one or more objects adjacent to a vehicle includes capturing a image of an object adjacent to the vehicle; determining a driving area in the image; cutting the driving area to form an identification window; selecting an identification area in the identification window; accessing a plurality of object image data in a memory to compare the plurality of object image data with the identification area; and identifying a specific object in the identification area by the processor.

Abstract: Sound capturing which includes applying a far-field microphone functionality to a multiplicity of first microphone signals to provide a first output signal, and applying a less directional microphone functionality to one or more second microphone signals to provide a second output signal.

Abstract: An audio processing device comprises a) at least one input unit for providing a time-frequency representation Y(k,n) of an electric input signal representing sound consisting of target speech and noise signal components, where k and n are frequency band and time frame indices, respectively, b) a noise reduction system configured to b1) determine a first signal to noise ratio estimate ?(k,n) of said electric input signal, and to b2) determine a second signal to noise signal ratio estimate ?(k,n) of said electric input signal from said first signal to noise ratio estimate ?(k,n) based on a recursive algorithm providing non-linear smoothing, and wherein a determination of said one or more bias and/or smoothing parameters comprises the use of supervised learning, e.g. one or more neural networks. The invention may be used in audio processing devices, such as hearing aids, headsets, ear phones, active ear protection systems, handsfree telephone systems, mobile telephones, etc.

Abstract: Audio (noise source) is output as a cancellation sound through a variable filter and a first filter, and transmitted to the second filter. A subtractor subtracts an output of a second filter from an output of a microphone, and an adaptive algorithm execution unit updates a transfer function of the variable filter so that the subtracted result becomes zero (0). A transfer function A for the first filter is a transfer function which can cancel noise at a position of a user's ear by setting, as the cancellation sound, a sound obtained by applying the transfer function A to audio at the time of learning, and a transfer function B for the second filter is a transfer function which can eliminate, for the cancellation sound, a difference between a sound obtained by applying the transfer function B to audio and the output of the microphone.

Abstract: A conference system is provided that includes a microphone array unit having a plurality of microphone capsules arranged in or on a board mountable on or in a ceiling of a conference room. The microphone array unit has a steerable beam and a maximum detection angle range. The conference system comprises a processing unit which is configured to receive the output signals of the microphone capsules and to steer the beam based on the received output signal of the microphone array unit. The processing unit is configured to control the microphone array to limit the detection angle range to exclude at least one predetermined exclusion sector in which a noise source is located.

Abstract: An electronic device or method for adjusting a gain on a voice operated control system can include one or more processors and a memory having computer instructions. The instructions, when executed by the one or more processors causes the one or more processors to perform the operations of receiving a first microphone signal, receiving a second microphone signal, updating a slow time weighted ratio of the filtered first and second signals, and updating a fast time weighted ratio of the filtered first and second signals. The one or more processors can further perform the operations of calculating an absolute difference between the fast time weighted ratio and the slow time weighted ratio, comparing the absolute difference with a threshold, and increasing the gain when the absolute difference is greater than the threshold. Other embodiments are disclosed.

Abstract: A method for noise attenuation includes comparing a frequency of noise generated by a plurality of noise sources with a plurality of frequencies associated with a pre-determined noise spectrum and stored in a computer system, identifying one or more noise sources that generate the noise based on the comparison of the frequency of the noise and the plurality of frequencies stored in the computer system; generating anti-noise corresponding to the noise; and outputting the anti-noise using one or more noise mitigation devices associated with the one or more noise sources.

Abstract: In at least one embodiment, a computer-program product embodied in a non-transitory computer readable medium that is programmed to perform selective active noise cancellation (ANC) for a vehicle is provided. The computer-program product comprising instructions to determine an amount of noise present in a first zone and a second zone and to selectively drive only at least one first loudspeaker in the first zone to generate a first cancellation field to cancel road noise and/or engine noise if the amount of noise in the first zone is greater than the amount of noise in the second zone. The computer-program product comprising instructions to selectively drive only at least one second loudspeaker in the second zone to generate the second cancellation field to cancel road noise and/or engine noise if the amount of noise in the second zone is greater than the amount of noise in the first zone.

Abstract: The technology described this document can be embodied in a method that includes receiving an error signal captured using a microphone, the error signal representing a difference between the sinusoidal component of a noise signal and an output of an acoustic transducer. The output of the acoustic transducer is configured to reduce the effects of the sinusoidal component of the noise signal. The method includes processing the error signal to compensate for effects due to a signal path between the acoustic transducer and the microphone, and determining a current estimate of one or more first parameters of the error signal. Based on such parameters, a current estimate of a time-varying step size associated with an adaptive process is determined, and based on the current estimate of the time-varying step size, a driver signal configured to change the output of the acoustic transducer is generated.

Abstract: The overall performance of an ANC system may be improved by configuring the ANC system to perform adaption in the frequency domain. The ANC systems may be configured to update an algorithm of an adaptive filter based, at least in part, on the first input signal, the second input signal, and a feedback signal that is based on an output of the adaptive filter. Updating may include changing parameters of the algorithm based on a SDR based, at least in part, on the first input signal. Updating may also include normalizing a step size and processing at least full band information for the input signal in a frequency domain to generate coefficient values for the algorithm. Updating may also include applying a frequency domain magnitude constraint on adaptive filter coefficients.

Abstract: A microphone array is disclosed. In an embodiment a microphone array includes a power supply circuit arrangement, at least one microphone circuit and a voltage detector. The power supply arrangement includes a switched-mode power supply circuit configured to generate at least two different levels of output voltage. The at least one microphone circuit includes a microphone transducer, and audio amplifier, a switchable voltage regulator circuit supplying the amplifier. The voltage detector is configured to switch the output voltage of the voltage regulator circuit depending on the voltage supplied by the switched-mode power supply circuit.

Abstract: An apparatus which allows smooth conversation with improved sound quality to be performed independently of the magnitude of noise or the like in a vehicle. A transmission system is provided, which transmits a sound signal from a first microphone provided corresponding to a first position to a speaker provided corresponding to a second position. A transmission apparatus, a transmission method, and a program include: an evaluating unit which evaluates at least one of a level of a direct sound transmitted from the first position to the second position without intervention of the first microphone and the speaker, a noise level, and an operating state of a noise source; and a delay setting unit which sets a transmission delay from the first microphone to the speaker based on an evaluation result of the evaluating unit.

Abstract: A system and method for disabling adaptation in an adaptive feedforward control system at low speeds. The method includes generating a noise signal representative of undesired noise detected by a noise sensor of a vehicle and generating a noise-cancellation signal via a controller within the vehicle. Residual noise resulting from the combination of the acoustic energy of the noise-cancellation signal and the undesired noise is detected by a reference sensor, which generates an error signal based on the residual noise. The error signal and a speed signal from a speed sensor on the vehicle are transmitted to an adaptive processing module for the generation of a filter update signal. The adaptive processing module selectively permits or prevents the filter update signal to adapt filter coefficients of a filter when the speed signal is within a set of conditions.

Abstract: An apparatus includes a sensor module configured for receiving sensed information indicative of a sensed signal. The sensed signal includes a source signal component and a source noise component. The apparatus also includes a reference module configured for reference information indicative of a reference signal. The reference signal also includes a reference noise component. The apparatus also includes a filter module configured as a fixed lag Kalman smoother. The filter module is configured for adaptively filtering the reference signal to generate an estimate of the source noise component. The apparatus also includes a processing module configured for calculating an output signal based on the sensed signal and the estimate of the source noise component. The apparatus also includes an interface module configured for transmitting an indication of the output signal. The filter module is further configured for, based on the output signal, tuning the Kalman smoother.

Abstract: The disclosure includes an acoustic processing network comprising a Digital Signal Processor (DSP) operating at a first frequency and a Real-Time Acoustic Processor (RAP) operating at a second frequency higher than the first frequency. The DSP receives a noise signal from at least one microphone. The DSP then generates a noise filter based on the noise signal. The RAP receives the noise signal from the microphone and the noise filter from the DSP. The RAP then generates an anti-noise signal based on the noise signal and the noise filter for use in Active Noise Cancellation (ANC).

Abstract: A method for acoustic echo cancellation is disclosed herein. A microphone receives a second acoustic signal from a near-end environment, the second acoustic signal including a delayed version of the first acoustic signal from a far-end environment. A processor models a relationship between the first acoustic signal and the second acoustic signal using an adaptive filter. The adaptive filter uses sampling points of the first acoustic signal and the second acoustic signal along a timeline as inputs. The processor identifies a sampling point among the sampling points, wherein weight values of the adaptive filter associated with the identified sampling point experience a significant increase (e.g., 50% increase). The identified sampling point along the timeline represents an estimated delay between the first acoustic signal and the second acoustic signal. The processor further removes the delayed version of the first acoustic signal from the second acoustic signal based on the estimated delay.

Abstract: A device for the acquisition and conditioning of a sound signal generated by a source of an internal combustion engine and comprising a box-shaped support body having a first measuring assembly and a second measuring assembly; wherein the first measuring assembly is provided with at least two first microphones which lie on a first plane and are configured for the detection of a first acoustic signal generated by the source along a first provenance direction; and wherein the second measuring assembly is provided with at least a second microphone arranged on a second plane configured for the detection of a second acoustic signal generated by the source along a second provenance direction different from the first provenance direction.

Abstract: An active noise cancellation system comprising an active noise cancellation circuit connected to a microphone arranged to sense environmental noise, the active noise cancellation circuit comprising: —an analog-to-digital converter (ADC) arranged to convert the sensed environmental noise to a digital environmental noise signal, —a prediction filter configured for predicting a plurality D of inverted digital environmental noise samples and generating a digital inverted environmental noise signal, —a digital-to-analog converter (DAC) to convert the digital inverted environmental noise signal to an analog inverted environmental noise signal for cancelling the environmental noise.

Abstract: An apparatus and a method for controlling a linear compressor, and a linear compressor operable with high power and low noise are provided. The apparatus may include a reference operating frequency determiner that determines a reference operating frequency at which a linear motor is operated, and an actual operating frequency determiner that determines an actual operating frequency as an arbitrary value included in a predetermined numerical value range in a vertical direction around the reference operating frequency. A correction signal may be determined by the actual operating frequency.

Abstract: A method of equalising sound in a headset comprising an internal microphone configured to generate a first audio signal, an external microphone configured to generate a second audio signal, a speaker, and one or more processors coupled between the speaker the external microphone, and the internal microphone, the method comprising: while the headset is worn by a user: determining a first audio transfer function between the first audio signal and the second audio signal in the presence of sound at the external microphone; and determining a second audio transfer function between a speaker input signal and the first audio signal with the speaker being driven by the speaker input signal; determining an electrical transfer function of the one or more processors; determining a closed-ear transfer function based on the first audio transfer function, the second audio transfer function and the electrical transfer function; and equalising the first audio signal based on a comparison between the closed-ear transfer funct

Abstract: According to an example aspect of the present invention, there is provided a Method for regularizing the inversion of a stereo headphone transfer function for headphone equalization, characterized by using the equation for the equalization: H S ? ? 1 - 1 ? ( ? ) = H * ? ( ? ) ? H ? ( ? ) ? 2 + ? ^ ? ( ? ) ? D ? ( ? ) = H * ? ( ? ) ? H ? ( ? ) ? 2 + [ ? ? ( ? ) + ? 2 ? ( ? ) ] ? D ? ( ? ) . in which equation HSI?1(?) is sigma inversion H*(?) is complex conjugate of a response D(?) is a delay filter introduced to produce a causal inverse HRI?1(?) H*(?) a response ?(?) is headphone reproduction bandwidth, ?(?) an estimation of the regularization needed inside that bandwidth.

Abstract: There is provided an information processing device, an information processing method, and a program which are capable of inputting information input by a plurality of users to targets manipulated by the users even in a situation in which the plurality of users perform manipulations at the same time, the information processing device including: an acquiring unit configured to identify and acquire input information from each of a plurality of target regions serving as acquisition sources of the input information, for each of the target regions; and a control unit configured to associate each of at least one or more applications with any of the plurality of target regions, and allocate the input information acquired from the target region to the application associated with the target region.

Abstract: A noise cancellation system with harmonic filtering for a vehicle audio system may include at least one input sensor configured to transmit reference signals, at least one input sensor configured to transmit a narrowband input signal and a broadband input signal, each of the narrowband input signal and broadband input signal including harmonic noise. The system may include a processor being programmed to receive the reference signals, the reference signals including a narrowband reference signal and a broadband reference signal, apply an adaptive filter to each of the reference signals, apply a secondary path to the input signals to generate antinoise signals, sum the antinoise signals broadcast over the secondary path and the primary noise signals to generate an error signal at the output sensor, and apply an adaptive filter to the error signal to remove harmonic noise to prevent cancelation of the harmonic noise.

Abstract: A headset device includes a first earpiece configured to receive a reference sound and to generate a first reference audio signal based on the reference sound. The headset device further includes a second earpiece configured to receive the reference sound and to generate a second reference audio signal based on the reference sound. The headset device further includes a controller coupled to the first earpiece and to the second earpiece. The controller is configured to generate a first signal and a second signal based on a phase relationship between the first reference audio signal and the second reference audio signal. The controller is further configured to output the first signal to the first earpiece and output the second signal to the second earpiece.

Abstract: An electronics with cascaded adaptive filters for attenuating noise in a feedback path of a flow controller is provided. The electronics includes a signal processor configured to receive a flow signal from a flow sensor, the flow sensor being configured to measure a flow rate of the pulsating flow, generate a noise reference signal from the flow signal and generate a flow rate signal using the noise reference signal. The electronics also includes a controller communicatively coupled to the signal processor, the controller being configured to generate a flow rate control signal using the flow rate signal, and a signal generator communicatively coupled to the controller. The signal generator is configured to receive the flow rate control signal, generate a valve signal based on the flow rate control signal, and provide the valve signal to a valve to control the flow rate of the pulsating flow.

Abstract: An electronic device includes a processor. The processor is configured to examine the sound level of an audio signal obtained from an external object for a preset time; change, if the frequency with which the sound level becomes higher than or equal to a specified level satisfies a preset condition, a reference level, at which the gain value for audio signal amplification is changed in accordance with the amplitude change of the audio signal, for at least one section of the audio signal satisfying the preset condition; and amplify the at least one section of the audio signal according to the changed reference level.

Abstract: Systems and methods for selectively ignoring an occurrence of a wakeword within audio input data is provided herein. In some embodiments, a wakeword may be detected to have been uttered by an individual within a modified time window, which may account for hardware delays and echoing offsets. The detected wakeword that occurs during this modified time window may, in some embodiments, correspond to a word included within audio that is outputted by a voice activated electronic device. This may cause the voice activated electronic device to activate itself, stopping the audio from being outputted. By identifying when these occurrences of the wakeword within outputted audio are going to happen, the voice activated electronic device may selectively determine when to ignore the wakeword, and furthermore, when not to ignore the wakeword.

Abstract: An active road-noise control system and method include using a sensor arrangement to generate a first sense signal representative of at least one acceleration, motion and/or vibration that occurs at a first position on a vehicle body and a second sense signal representative of sound that occurs at a second position within the vehicle body; they also provide a noise-reducing signal by processing the sense signals according to a first or a second mode of operation. They include the generation of noise-reducing sound within the vehicle body at the second position from the noise reducing signal and the operational state of the sensor arrangement; the first sense signal and the second sense signal are processed in the first mode when the sensor arrangement is in a proper operational state and in the second mode when a malfunction of the sensor arrangement has been detected.

Abstract: A system includes an Active Noise Control (ANC) module unit configured to be installed within an air intake or exhaust of a power generation unit. The ANC module unit includes an ANC housing shaped to fit within the air intake or exhaust, an ANC core configured to be secured within the ANC housing, which includes a microphone configured to detect a sound generated by the power generation unit, a control board configured to control the noise-canceling sound based on the sound signal from the microphone and a set of pre-determined noise reduction transfer functions, and a first speaker configured to deliver a first noise-canceling sound to the air intake or exhaust.

Abstract: An audio processing device comprises a) at least one input unit for providing a time-frequency representation Y(k,n) of an electric input signal representing sound consisting of target speech and noise signal components, where k and n are frequency band and time frame indices, respectively, b) a noise reduction system configured to: determine a first signal to noise ratio estimate ?(k,n) of said electric input signal, and determine a second signal to noise signal ratio estimate ?(k,n) of said electric input signal from said first signal to noise ratio estimate ?(k,n) based on a recursive algorithm providing non-linear smoothing, and wherein parameters of said smoothing are determined in dependence of the first and/or the second signal to noise ratio estimates corresponding to a multitude of frequency band indices. The invention may be used in hearing aids, headsets, ear phones, active ear protection systems, handsfree telephone systems, mobile telephones, etc.

Abstract: A method embodiment includes combining a control signal of a voltage regulator circuit of an apparatus with pseudo-random noise, and using the control signal to provide an output voltage signal as attenuated by a power supply rejection ratio (PSRR) of an analog mixed-signal (AMS) circuit of the apparatus. The method further includes self-testing the AMS circuit by cross-correlating a signal indicative of the output voltage signal from the AMS circuit with the pseudo-random noise and, in response, assessing the results of the cross-correlation relative to a known threshold indicative of a performance level of the AMS circuit.

Abstract: A sound signal processing apparatus includes a sound acquisition unit, a processing unit, an output unit, and a control unit. The sound acquisition unit is configured to acquire sound data generated by collecting a sound from a sound-collecting target space. The processing unit is configured to execute, on the sound data acquired by the sound acquisition unit, signal processing corresponding to each of a plurality of areas included in the sound-collecting target space. The output unit is configured to output processed data generated based on the signal processing executed by the processing unit. The control unit configured to perform control such that an amount of processing performed by the processing unit does not exceed a predetermined amount of processing.

Abstract: A road noise cancellation (RNC) system may include a noise controller for detecting non-stationary events, such as objects striking vibration sensors, based on sensor signals having a spectral character significantly different from steady-state road noise. Upon detection of an object strike, the RNC system may be deactivated or certain speakers may be muted. Alternatively, the RNC system may modify the sensor signals to mask the non-stationary event, thereby preventing the RNC system from generating anti-noise based on the object strike.

Abstract: A noise reducing comprises a first microphone that picks up noise signal at a first location and that is electrically coupled to a first microphone output path; a loudspeaker that is electrically coupled to a loudspeaker input path and that radiates noise reducing sound at a second location; a second microphone that picks up residual noise from the noise and the noise reducing sound at a third location and that is electrically coupled to a second microphone output path; a first active noise reducing filter that is connected between the first microphone output path and the loudspeaker input path; and a second active noise reducing filter that is connected between the second microphone output path and the loudspeaker input path; in which the first active noise reduction filter is a shelving or equalization filter or comprises at least one shelving or equalization filter or both.

Abstract: Techniques are provided for vector noise cancellation. Different value combinations for a plurality of weighting factors may be established for a plurality of selection regions. Each value combination for the plurality of weighting factors may correspond to a different combination of a set of input signals. One or more characteristics of input signals may be used to select a particular selection region. A particular value combination of the set of weighting factors may be chosen to attenuate or amplify the input signals to generate one or more output signals.

Abstract: The disclosure includes an acoustic processing network comprising a Digital Signal Processor (DSP) operating at a first frequency and a Real-Time Acoustic Processor (RAP) operating at a second frequency higher than the first frequency. The DSP receives a noise signal from at least one microphone. The DSP then generates a noise filter based on the noise signal. The RAP receives the noise signal from the microphone and the noise filter from the DSP. The RAP then generates an anti-noise signal based on the noise signal and the noise filter for use in Active Noise Cancellation (ANC).

Abstract: The technology described in this document can be embodied in a computer-implemented method that includes receiving a first plurality of values representing a set of current coefficients of an adaptive filter disposed in an active noise cancellation system. The method also includes computing a second plurality of values each of which represents an instantaneous difference between a current coefficient and a corresponding preceding coefficient of the adaptive filter, and estimating, based on the second plurality of values, one or more instantaneous magnitudes of a transfer function that represents an effect of a secondary path of the active noise cancellation system. The method further includes updating the first plurality of values based on estimates of the one or more instantaneous magnitudes to generate a set of updated coefficients for the adaptive filter, and programming the adaptive filter with the set of updated coefficients.

Abstract: An audio processing device comprises a) at least one input unit for providing time-frequency representation Y(k,n) of an electric input signal representing sound consisting of target speech and noise signal components, where k and n are frequency band and time frame indices, respectively, b) a noise detection and/or reduction system configured to b1) determine an a posteriori signal to noise ratio estimate ?(k,n) of said electric input signal, and to b2) determine an a priori target signal to noise signal ratio estimate ?(k,n) of said electric input signal from said a posteriori signal to noise ratio estimate ?(k,n) based on a recursive decision directed algorithm. The application further relates to a method of of estimating an a priori signal to noise ratio. The invention may e.g. be used for the hearing aids, headsets, ear phones, active ear protection systems, handsfree telephone systems, mobile telephones, etc.

Abstract: A noise cancellation device includes a voice receiving module, a distance measuring module, a noise cancellation module and a speaker module. The voice receiving module is configured to receive a noise. The distance measuring module is configured to send a distance measuring signal to an object, and calculate distance information of the object according to a reflected distance measuring signal reflected by the object. The noise cancellation module is connected to the voice receiving module and the distance measuring module, and is configured to generate a reverse phase signal of the noise according to the noise. The speaker module is connected to the noise cancellation module, and is configured to generate an anti-noise according the reverse phase signal, and sound the anti-noise according to the distance information.

Abstract: A device includes a housing and a core configured to be secured within the housing. The core includes a microphone configured to detect a sound signal. The core also includes a control board configured to generate a noise-canceling sound signal based on the sound signal and a set of transfer functions. The core further includes a speaker configured to deliver the noise-canceling sound signal. The device also includes a fastener configured to connect the housing to an air filter of an intake vent of an enclosure of a power generation unit.

Abstract: The present invention relates to a method of noise reduction including the steps of filtering reference signals and representing noise by an adaptive filter comprising adaptive filter coefficients to obtain actuator driving signals, outputting the actuator driving signals by loudspeakers to obtain loudspeaker signals. The method further includes detecting the loudspeaker signals by microphones and filtering the reference signals by estimated transfer functions representing the transfer of the loudspeaker signals output by the loudspeakers to the microphones to obtain filtered reference signals. The method further includes updating the filter coefficients of the adaptive filter based on the filtered reference signals and based on the previously updated filter coefficients of the adaptive filter multiplied by leakage factors.

Abstract: A method (600) of generating a drive signal for a vibratory sensor (5) is provided. The method (600) includes vibrating a vibratory element (104, 510) configured to provide a vibration signal, receiving the vibration signal from the vibratory element (104, 510) with a receiver circuit (134), generating a drive signal that vibrates the vibratory element (104, 510) with a driver circuit (138) coupled to the receiver circuit (134) and the vibratory element (104, 510), and comparing a phase of the generated drive signal with a phase of the vibration signal.