Abstract: A crosstalk canceller for reducing acoustic crosstalk at a time of audio playback is derived by forming a channel frequency response for a nominated playback geometry, and decomposing the channel frequency response to derive a decomposition element such as a singular value decomposition matrix. A value of the decomposition element, such as the smallest singular value, is then adjusted to reduce spectral coloration, and filter coefficients of a crosstalk cancellation filter are derived from the adjusted decomposition element.

Abstract: A method may include determining a desired speech estimate originating from a speech acceptance direction range while reducing a level of interfering noise, determining an interfering noise estimate originating from a noise rejection direction range while reducing a level of desired speech, calculating a ratio of the desired speech estimate to the interfering noise estimate, dynamically computing a set of thresholds based on the speech acceptance direction range, noise rejection direction range, a background noise level, and a noise type, estimating a power spectral density of background noise arriving from the noise rejection direction range, calculating a frequency-dependent gain function based on the power spectral density of background noise and thresholds, and applying the frequency-dependent gain function to at least one microphone signal generated by the plurality of microphones to reduce noise arriving from the noise rejection direction while preserving desired speech arriving from the speech acceptan

Abstract: A sound control module is configured to determine N magnitudes for outputting a predetermined engine sound at N predetermined harmonics of a base frequency, respectively, where N is an integer greater than one. An adjustment module is configured to determine N magnitude adjustments for the N predetermined harmonics, respectively, based on a position of a compression ratio (CR) actuator configured to vary a CR of an engine. The sound control module is further configured to determine N adjusted magnitudes for the N predetermined harmonics based on: the N magnitudes for the N predetermined harmonics, respectively; and the N magnitude adjustments for the N predetermined harmonics, respectively. An audio driver module is configured to apply power to at least one speaker of the vehicle and output the predetermined engine sound based on the N adjusted magnitudes at N frequencies, respectively, corresponding to the N predetermined harmonics of the base frequency.

Abstract: A computing device senses speech or other user input. One or more physical variables pertaining to a user of the computing device are also sensed, and respective signals are analyzed or compared to the user input. The analysis determines if the user input is likely that of an authorized user, and assigns a confidence metric to that determination. The computing device may then perform actions corresponding to the speech or user input content in accordance with the determination.

Abstract: Methods, apparatus, systems and articles of manufacture to implement selective suppression of audio emitted from an audio source are disclosed. Example apparatus disclosed herein include a speaker, an audio output driver in communication with the speaker, memory including computer readable instructions, and a processor. In some disclosed example apparatus, the processor is to execute the instructions to perform operations including obtaining reference audio data corresponding to an audio signal to be output subsequently by the audio source and including identification data identifying the audio source, and in response to an input selection corresponding to the identification data, generating a suppression signal to provide to the audio output driver, the suppression signal based on the reference audio data and a time delay between a first time at which the reference audio data was received and a second time at which the audio signal is to be output by the audio source.

Abstract: In accordance with embodiments of the present disclosure, a processing circuit may implement an adaptive filter, a first signal injection portion which injects a first additional signal into a first frequency range content source audio signal, and a second signal injection portion which injects a second additional signal into a second frequency range content source audio signal, wherein the first additional signal and the second additional signal are substantially different.

Abstract: The technology described in this document can be embodied in a computer-implemented method that includes receiving, by one or more processing devices, a first plurality of values representing a set of coefficients of an adaptive filter disposed in an active noise cancellation system. The method also includes accessing one or more estimates of instantaneous phase values associated with a transfer function representing an effect of a secondary path of the active noise cancellation system, and updating the first plurality of values based on the one or more estimates of the instantaneous phase values to generate a set of updated coefficients for the adaptive filter. The method further includes programming the adaptive filter with the set of updated coefficients to affect operation of the adaptive filter.

Abstract: A time heuristic audio control system, comprises a receiver for receiving time-based data from a personal computing device and a memory storing one or more sets processing parameters comprising instructions for processing the ambient sound based upon the time-based data. The system further includes a processor coupled to the memory and the receiver configured to adjust the ambient sound as directed by a selected set of processing parameters retrieved from the memory to create adjusted audio, the selected set of processing parameters retrieved based upon the time-based data and at least one speaker for outputting the adjusted audio.

Abstract: The disclosure relates to combined active noise cancellation and noise compensation in a headphone. An audio processing device includes a selector and a noise compensation unit. The selector can select one of a plurality of first transfer functions based on at least one feature of at least one of an external noise and a content audio signal representing a sound to be reproduced through the headphone. The noise compensation unit can compute a second audio signal by applying the selected first transfer function to a first audio signal, and derive gains for the noise compensation at least based on the second audio signal. The at least one feature can be used to distinguish at least two of the first transfer functions. Each of the first transfer functions can transform the first audio signal to the second audio signal which is assumed as representing a version of the sound represented by the first audio signal, which arrives at an eardrum of a listener wearing the headphone.

Abstract: A system for audio processing comprising an initial background statistical model system configured to generate an initial background statistical model using a predetermined sample size of audio data. A parameter computation system configured to generate parametric data for the audio data including cepstral and energy parameters. A background statistics computation system configured to generate preliminary background statistics for determining whether speech has been detected. A first speech detection system configured to determine whether speech was present in the initial sample of audio data. An adaptive background statistical model system configured to provide an adaptive background statistical model for use in continuous processing of audio data for speech detection. A parameter computation system configured to calculate cepstral parameters, energy parameters and other suitable parameters for speech detection.

Abstract: A noise filter includes a filter device and a filter retainer. The filter device includes a device unit having at least one capacitor provided therein, an input side lead wire extending from the device unit, an output side lead wire extending from the device unit, and a ground connection portion provided on the device unit. The filter device forms the noise filter that uses a mutual inductance between the input side lead wire and the output side lead wire. The filter retainer has a holding structure configured to maintain an arrangement of the input side lead wire and the output side lead wire. The arrangement forms an overlapping section in which the lead wires are closely opposed to each other. The noise filter may be attached to a harness.

Abstract: A method and apparatus are for determining one or more background noise characteristics, determining one or more incoming audio characteristics; and generating a combined audio signal comprising an active noise cancellation (ANC) component and a modified incoming audio (MIA) component. The ANC component is determined based on at least one of the one or more incoming audio characteristics and the background noise characteristics. Each of a limit of the ANC component and a limit of the MIA component is dynamically controlled to be less than or equal to a system limit, wherein a limit of the combined signal is approximately at the system limit.

Abstract: A transmitter/receiver apparatus including: a transmitter/receiver terminal; a switching amplifier that includes a low-side switching element connected between a ground terminal and a pulse output terminal and a high-side switching element connected between the pulse output terminal and a power supply terminal and that outputs a pulse signal from the pulse output terminal; a filter that passes therethrough and outputs as a transmitted signal a predetermined frequency component of the pulse signal from a transmitter terminal; and a transmit/receive switch unit that switches the connection status between the transmitter/receiver terminal and the transmitter terminal and also switches the connection status between the transmitter/receiver terminal and a receiver terminal.

Abstract: Provided is a noise reduction apparatus, a noise reduction method, and a program for generating voice that is easy to hear while reducing sudden sound. The noise reduction apparatus includes a first microphone configured to collect mainly voice and output the voice as a first signal, a second microphone configured to collect mainly sound other than the voice and output the sound as a second signal, a sudden sound detecting unit configured to detect sudden sound in the first signal and the second signal, a sudden sound information storage unit configured to store the sudden sound as one or a plurality of reference signals, and an adder configured to remove the sudden sound from the first signal based on the reference signal stored in the sudden sound information storage unit.

Abstract: A method that is accomplished in a vehicle engine harmonic modification system. A non-zero target engine harmonic signal that is representative of a target engine harmonic sound level in the vehicle cabin is provided. The target engine harmonic signal is used in an operation of the engine harmonic modification system so as to modify the level of engine harmonic sound in the vehicle cabin, to bring the engine harmonic sound level in the vehicle cabin closer to the target engine harmonic sound level.

Abstract: A method for masking a vehicle noise includes setting a noise-generated driving region in which the vehicle noise is generated while a vehicle is running; measuring the vehicle noise in the noise-generated driving region; determining a masking sound which cancels the vehicle noise based on the measured vehicle noise; and outputting the determined masking sound into an interior of the vehicle.

Abstract: An integrated circuit for digital signal routing. The integrated circuit has analog and digital inputs and outputs, including digital interfaces for connection to other integrated circuits. Inputs, including the digital interfaces, act as data sources. Outputs, including the digital interfaces, act as data destinations. The integrated circuit also includes signal processing blocks, which can act as data sources and data destinations. Signal routing is achieved by means of a multiply-accumulate block, which takes data from one or more data source and, after any required scaling, generates output data for a data destination. Data from a data source is buffered for an entire period of a data sample clock so that the multiply-accumulate block can retrieve the data at any point in the period, and output data of the multiply-accumulate block is buffered for an entire period of the data sample clock so that the data destination can retrieve the data at any point in the period.

Abstract: Methods, systems, and media for ambient background noise modification are provided. In some implementations, the method comprises: identifying at least one noise present in an environment of a user having a user device, an activity the user is currently engaged in, and a physical or emotional state of the user; determining a target ambient noise to be produced in the environment based at least in part on the identified noise, the activity the user is currently engaged in, and the physical or emotional state of the user; identifying at least one device associated with the user device to be used to produce the target ambient noise; determining sound outputs corresponding to each of the one or more identified devices, wherein a combination of the sound outputs produces an approximation of one or more characteristics of the target ambient noise; and causing the one or more identified devices to produce the determined sound outputs.

Abstract: Personal audio systems and methods are disclosed. A personal audio system includes a processor to generate a personal audio stream by processing the digitized ambient audio in accordance with an active processing parameter set, an event detector to detect a trigger event other than manual selection by a user of the personal audio system, and a controller configured to, in response to detection of the trigger event, select a new processing parameter set and instruct the processor to begin generating the personal audio stream by processing the digitized ambient audio in accordance with the new processing parameter set.

Abstract: A method in a communications device includes the following operations. During a call, a process automatically detects that the device has moved from an at-the-ear position to an away-from-the-ear position. Based on the detection, a noise suppressor that operates upon an uplink signal for the call is signaled to change its noise suppression performance. Other embodiments are also described and claimed.

Abstract: According to one embodiment, a noise reduction system for reducing noise including impact noise repetitively generated at a time interval includes the following elements. The error signal generator generates an error signal based on the noise being detected. The delay signal generator has a time delay characteristic and delays a signal, which is generated based on the error signal, to generate a delay signal, the time delay characteristic being determined based on an imaging sequence or pre-scanning by the MRI device and corresponding to the time interval. The control filter generates the first control signal from the delay signal. The loudspeaker unit includes at least one pair of a first filter and a control loudspeaker and a transmission unit.

Abstract: An audio output device controls the application of active noise cancellation (ANC) levels based on a current status of the device. The audio output device includes a communication interface configured to provide a wired or wireless connection to a playback device, a signal detector configured to detect an audio signal provided through the communication interface, an ANC module configured to apply a first-level ANC to the audio signal, and a control unit configured to determine a level of the ANC to be applied by the ANC module. Upon determining that the audio signal is not provided based on information provided from at least one of the communication interface or the signal detector, the control unit instructs the ANC module to perform a second-level ANC.

Abstract: A noise reduction device includes a plurality of noise microphones, noise controller, and control speakers. Noise controller generates a control-sound signal to reduce a noise, at a control center of a control space, with the noise being detected by the plurality of noise microphones. Noise microphone disposed at a shorter distance from the control center than distance “d” indicated by Relational Expression (1) is smaller in number than noise microphones disposed at longer distances than distance “d” where “?” is a wavelength corresponding to control upper-limit frequency “f” in noise microphones, “d0” is a distance from the control center to control speakers, “t” is a control delay time in control speakers, and “v” is a sound speed.

Abstract: An adaptive noise canceling (ANC) circuit adaptively generates an anti-noise signal from a reference microphone signal that is injected into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone proximate the speaker provides an error signal. A secondary path estimating adaptive filter estimates the electro-acoustical path from the noise canceling circuit through the transducer so that source audio can be removed from the error signal. Tones in the source audio, such as remote ringtones, present in downlink audio during initiation of a telephone call, are detected by a tone detector using accumulated tone persistence and non-silence hangover counting, and adaptation of the secondary path estimating adaptive filter is halted to prevent adapting to the tones. Adaptation of the adaptive filters is then sequenced so any disruption of the secondary path adaptive filter response is removed before allowing the anti-noise generating filter to adapt.

Abstract: The technology described herein can be embodied in a computer implemented method that includes detecting, by one or more processing devices, onset of an unstable condition in an active noise control system. The method also includes obtaining, responsive to detecting the onset of the unstable condition, updated filter coefficients for a system-identification filter configured to represent a transfer function of a secondary path of the active noise control system. The updated filter coefficients are generated using a set of multiple subband adaptive filters, wherein filter coefficients of each subband adaptive filter in the set are configured to adapt to changes in a corresponding portion of a frequency range associated with potential unstable conditions in the active noise control system. The method also includes programming the system identification filter with the updated coefficients to affect operation of the active noise control system.

Abstract: The present disclosure relates in a first aspect to a method of superimposing spatial auditory cues to an externally picked-up sound signal in a hearing instrument. The method comprises steps of a generating an external microphone signal by an external microphone arrangement and transmitting the external microphone signal to a wireless receiver of a first hearing instrument via a first wireless communication link. Further steps of the methodology comprise determining response characteristics of a first spatial synthesis filter by correlating the external microphone signal and a first hearing aid microphone signal of the first hearing instrument and filtering the external microphone signal by the first spatial synthesis filter to produce a first synthesized microphone signal comprising first spatial auditory cues.

Abstract: A method for disguising the environment and thus the location in which a telephone is being used includes acquiring a disguised environmental sound from a storage device in response to a selection of a user, controlling a voice acquiring device to acquire voice of the user and actual undisguised environmental noise during a call. An environmental noise signal is removed and replaced by an environment-disguising sound signal. An ultrasonic signal can be generated and synthesized with the transmitted voice signal to indicate mis-use of a communication device notwithstanding the environment-disguising noises.

Abstract: A noise suppression device includes a generator to generate, on basis of phase differences between phases of the signals input from microphones, additional data obtained by rotating the phase differences; an estimator to select one or multiple ranges in association with a direction in which a sound source of a target sound included in the input signals exists at a high probability, and to estimate, on basis of the phase differences and the additional data, a range that is among the selected one or multiple ranges and in which the sound source exists; and an output signal generator configured to generate, on basis of a suppression coefficient set on basis of a result of determination of whether or not the sound source exists in the estimated range, a output signal in which the noise in the input signals is suppressed.

Abstract: An acoustic coupling sensor 200 for a portable electronic device is described which includes an acoustic transmitter 22, a reference signal generator 20 coupled to the acoustic transmitter, and an acoustic sensor 24 wherein in operation, the reference signal generator is operable to transmit the reference signal to the acoustic transmitter, to detect an acoustic level of the transmitted reference signal via the acoustic sensor, and wherein the detected reference signal level corresponds to a value of acoustic coupling between the acoustic transmitter and the acoustic sensor. The acoustic coupling sensor may be used for example in a mobile phone to detect acoustic leakage in order to adapt the acoustic level in the mobile phone during use.

Abstract: Personal audio systems and methods are disclosed. A personal audio system includes a processor to generate a personal audio stream by processing an ambient audio stream in accordance with an active processing parameter set, a circular buffer memory to store a most recent snippet of the ambient audio stream, and an event detector to detect a trigger event. In response to detection of the trigger event, a controller may transmit, to a knowledgebase remote from the personal audio system, the most recent snippet of the ambient audio stream and associated metadata.

Abstract: In one embodiment, a method for training an adaptive filter includes receiving, by a processor from a device, an input signal and a training reference signal and determining a correlation matrix in accordance with the input signal, the training reference signal, and a filter type. The method also includes determining a plurality of coefficients in accordance with the correlation matrix and adjusting the adaptive filter in accordance with the plurality of coefficients.

Abstract: In accordance with an embodiment of the invention, a sound masking system is used to produce a sound masking zone in the vicinity of a defined area, for example in an area surrounding a conference room, an area within a medical office, or an area within a place of worship. A screen of loudspeakers around the defined area is used to create the sound masking zone. An indicator light, outside the defined area and unique to the sound masking zone around the defined area, is used to indicate that the sound masking system is active or ramping up.

Abstract: A method for generating an audio signal for a synthetic noise of a motor vehicle is provided. An audio signal for the synthetic noise of a motor vehicle is also provided. In one step of the method, a signal is provided that represents a noise. Moreover, a fundamental frequency is specified that represents an operating rotational speed of the motor of the vehicle. In another step of the method, several filters are applied to the noise signal. These filters each have a mid-frequency linked to the specified fundamental frequency via an order factor. At least several of the order factors are whole numbers, so that these filters filter the harmonic components of the fundamental frequency. The order factors and the amplitude curves of the filters are selected according to the sound characteristic of the motor vehicle noise that is to be synthesized.

Abstract: An adaptive active noise cancellation apparatus performs a filtering operation in a first digital domain and performs adaptation of the filtering operation in a second digital domain.

Abstract: A method for feedback detection by an electronic device is described. The method includes receiving a first microphone signal by a first microphone. A feedback loop includes the first microphone and a speaker. The method also includes receiving a second microphone signal by a second microphone that is outside of the feedback loop. A first signal based on the first microphone signal and a second signal based on the second microphone signal exhibit a higher correlation in presence of feedback and exhibit a lower correlation in absence of feedback. The method further includes determining a correlation based on the first microphone signal and the second microphone signal. The method additionally includes determining whether feedback is occurring based on the correlation.

Abstract: A method, system, and apparatus for noise cancelation is disclosed herein, which may be used in a wireless unit (WU) and a headset removably connected to the WU. The WU may include a processor, a memory, a user interface, internal microphones and internal speakers. The headset may include microphones and speakers, and be powered by the WU. The WU may receive an ambient noise at microphone(s), including a headset microphone, which may generate a signal based on the ambient noise. The WU processor may then calculate an estimate of the ambient noise based on the signal, calculate an inverse of the estimated ambient noise based on the estimate of the ambient noise, cancel the estimated ambient noise from an audio output signal based on the application of the inverse of the estimated ambient noise, and send the audio output signal to the speakers of the headset or the WU.

Abstract: Methods and systems are provided for removing transient noises (e.g., keyboard “clicks”) detected in an audio signal. An adaptive jitter buffer and packet loss concealer (“JB/PLC”) is deployed on the transmitting side of a system (e.g., a Voice-over-IP (VoIP) system) to remove transient noises detected in an audio signal before the signal is transmitted over a network to the receiving side of the system. By utilizing the adaptation capabilities of the JB/PLC, the methods and systems provided produce better-sounding transient removal than other existing approaches.

Abstract: A sound processing device is provided. The sound processing device includes a microphone array and a post filtering module. The microphone array includes microphones aiming to different directions and configured for receiving sound signals. The post filtering module is configured for receiving the sound signals from the microphone array, filtering the sound signals to generate groups of filtered signals each corresponding to one of the sound signals, wherein each of the filtered signals within a group corresponds to one of different frequency bands, generating band signals each based on a comparison of an intensity of one of the filtered signals that corresponds to the same one of the frequency bands in each group of the filtered signals and a noise intensity correlation between the frequency bands and adding the band signals to generate an output sound signal.

Abstract: A device and method that is configured to control the operation of a motor vehicle cabin active sound management (ASM) system that uses an ASM controller to control operation of one or more sound transducers that transmit audio signals in the cabin so as to alter vehicle powertrain-related sounds in the cabin, wherein the motor vehicle further comprises a mobile telephone system that is configured to receive incoming mobile telephone calls and transmit outgoing mobile telephone calls. The device can include a controller that is configured to receive from the mobile telephone system an indication that a mobile telephone call is incoming or an outgoing call is being initiated. After receiving the indication, the subject controller commands the ASM controller to cause operation of one or more sound transducers such that the level of vehicle powertrain-related sounds in the cabin is reduced.

Abstract: A hearing assistance system streams audio signals from one or more streaming sources to a hearing aid set and enhances the audio signals such that the output sounds transmitted to the hearing aid wearer include a spatialization effect allowing for localization of each of the one more streaming sources. The system determines the position of the hearing aid set relative to each streaming source in real time and introduces the spatialization effect for that streaming source dynamically based on the determined position, such that the hearing aid wearer can experience a natural feeing of the acoustic environment.

Abstract: Embodiments provided herein describe a feedback system that alters the distribution of a sound in a vehicle from a default configuration indicating to the driver that his action is dangerous or inefficient. The feedback system monitors one or more forces or accelerations such as speeding up, slowing down, changing the vehicle's direction, etc. Moreover, the feedback system may have one or more thresholds associated with these accelerations. If the measured acceleration satisfies the threshold, the feedback system alters the distribution of sound to inform the driver that his driving decisions may be dangerous or inefficient (e.g., causing excessive wear and tear on the vehicle). Furthermore, the change in the sound distribution may mimic the effect of the force or acceleration on the driver or vehicle to provide a more subtle or intuitive sonification relative to outputting a distinct audio alert.

Abstract: An apparatus comprises a first transducer configured to detect sound and generate a first signal based on the detected sound. The apparatus also comprises a second transducer configured to detect vibration and/or sound and generate a second signal based on the detected vibrations and/or sound. The second transducer is less acoustically responsive than the first transducer. The apparatus comprises an interface configured to send the first and second signals to a processor configured to modify the first signal on the basis of the second signal.

Abstract: A sound processing device includes a noise suppression unit configured to suppress a noise component included in an input sound signal, an auxiliary noise addition unit configured to add auxiliary noise to the input sound signal, whose noise component has been suppressed by the noise suppression unit, to generate an auxiliary noise-added signal, a distortion calculation unit configured to calculate a degree of distortion of the auxiliary noise-added signal, and a control unit configured to control an addition amount by which the auxiliary noise addition unit adds the auxiliary noise based on the degree of distortion calculated by the distortion calculation unit.

Abstract: A method for estimating a distance between a first and a second loudspeaker characterized by playing back a first stereo source signal vector s1 on the first loudspeaker, and playing back a second stereo source signal vector s2 on the second loudspeaker, acquiring a first recorded signal vector x1, using a first microphone arranged adjacent to the first loudspeaker, and acquiring a second recorded signal vector x2 from a second microphone arranged adjacent to the second loudspeaker, wherein x1 and x2 are N-dimensional vectors, setting the distance equal to ?v/f, where v is the speed of sound, f is the sampling frequency, and ? is an estimated sample delay of a source signal played back on one of the loudspeakers and a recording acquired by a microphone at the other loudspeaker.

Abstract: A method of processing digitized microphone signal data in order to detect wind noise. First and second sets of signal samples are obtained simultaneously from two microphones. A first number of samples in the first set which are greater than a first predefined comparison threshold is determined. A second number of samples in the first set which are less than the first predefined comparison threshold is determined. A third number of samples in the second set which are greater than a second predefined comparison threshold is determined. A fourth number of samples in the second set which are less than the second predefined comparison threshold is determined. If the first number and second number differ from the third number and fourth number to an extent which exceeds a predefined detection threshold, e.g. as determined by a Chi-squared test, then an indication that wind noise is present is output.

Abstract: A signal separation system using directional microphone array and a method for providing same. The signal separation system using directional microphone array includes a signal receiving section which receives a mixed signal in which a first audio signal and a second audio signal are mixed using the directional microphone array, the directional microphone array including at least one first microphone oriented in a first direction and at least one second microphone oriented in a direction different from the first direction, and receives a reference signal through the at least one second microphone. An audio signal separating section cancels the second audio signal from the mixed signal received by the signal receiving section, based on the reference signal inputted through the at least one second microphone.

Abstract: A multi-loudspeaker active noise cancelling headphone that can provide stereo effect and improve the performance of ambient noise reduction employs an active noise reduction circuit to collaborate with a plurality of speakers and a design to generate a multilayer phase difference feedback audio signal in the earmuffs of the headphone to improve the performance of noise reduction efficacy and generate stereo effect for music so that through the speakers of a lower cost and in a medium quality, and individual filters and signal compensation music with spatial sense and a sense of depth can be generated.

Abstract: To address issues with present echo gate control, a method and apparatus for more intelligently operating an echo gate is described herein. In particular, the decision of whether to mute an uplink signal, or not, is formulated herein as primarily a perceptual decision based on an appropriate analysis of the perceptual interaction of the current residual echo and the current near-end signal(s). By doing so, the application of muting through an echo gate may be minimized and/or more appropriately engaged. This will lead to fewer dropouts and muting of speech onsets and offsets 1) during periods such as double-talk or 2) during periods of downlink playback in the presence of low near-end signal levels, two cases of particular importance.

Abstract: According to one embodiment, a sound field control apparatus is provided with a control filter and a calculation unit. The filter executes an FIR computation for an acoustic signal using coefficients to output a main signal and control signals. The unit calculates the main coefficient and the coefficients based on Spatial transmission characteristics and a sound increase factor n, to set a composite sound pressure from a main speaker and control speakers to a first area to be n times of a coming sound pressure from only the main speaker, and to set a composite sound pressure from the main speaker and control speakers to a second area to be equal to the coming sound pressure from only the main speaker.

Abstract: In a personal listening device, an ANC system can benefit from a mechanism to detect an off-ear condition, which may be a situation in which the user of the personal listening device has moved an earphone or handset housing away from her ear. A detector may detect such a condition using signals from a touch sensor and/or a vibration sensor that are integrated in the earphone or handset housing, and in response power down the ANC system, or in the case of an adaptive ANC system slow down, or even freeze, the adaptation of one or more adaptive filters. The detector may operate during for example a phone call or during media file playback. Other embodiments are also described.