Abstract: An active noise reduction system includes a controller configured to acquire a plurality of noise signals output from a plurality of noise microphones, select a plurality of reference signals and an error signal from among the plurality of noise signals, the plurality of reference signals corresponding to a noise, the error signal corresponding to an error between the noise and a canceling sound, generate a control signal from the plurality of reference signals by using a plurality of control filters, and adaptively update the plurality of control filters by using a plurality of acoustic transmission filters. The plurality of acoustic transmission filters includes at least one adaptive update filter configured to be adaptively updated, and at least one non-adaptive update filter configured to be updated based on an update value of the adaptive update filter.

Abstract: An apparatus for generating at least one distance estimate to at least one sound source within a sound scene comprising the least one sound source, the apparatus configured to: receive at least two audio signals from a microphone array located within the sound scene; receive at least one further audio signal associated with the at least one sound source; determine at least one portion of the at least two audio signals from a microphone array corresponding to the at least one further audio signal associated with the at least one sound source; determine a distance estimate to the at least one sound source based on the at least one portion of the at least two audio signals from a microphone array corresponding to the at least one further audio signal associated with the at least one sound source.

Abstract: An adaptive feedback canceller of an ear-wearable device has an adaptive foreground filter that inserts a feedback cancellation signal into a digitized input signal to produce an error signal. An instability detector of the device is configured to extract wo or more features from the error signal. The instability detector has a machine learning module that determines instability in the error signal based on the two or more features. The instability module changes the adaptive foreground filter in response to determining the instability. The change causes the adaptive foreground filter to have a faster adaptation to perturbations in the error signal compared to a previously used step size.

Abstract: In the method according to the invention for active noise suppression, a transfer function for a secondary path between a loudspeaker and an error microphone is measured (20). Based on the measured transfer function for the secondary path, a transfer function for a primary path between a reference microphone and the error microphone is estimated (21). Based on the estimated transfer function for the primary path, filter coefficients for filtering to generate the cancellation signal are then determined (22).

Abstract: A hearing aid adapted for being worn by a user at or in an ear of the user comprises a) at least one input transducer for converting sound in an environment around the user to at least one electric input signal representing said sound; b) an output transducer for converting a processed output signal provided in dependence of said at least one electric input signal to stimuli perceivable to the user as sound; c) a feedback control system comprising an adaptive filer, the feedback control system being configured to provide an adaptively determined estimate (h*(n)) of a current feedback path (h(n)) from said output transducer to said at least one input transducer in dependence of c1) said at least one electric input signal, c2) said processed output signal, and c3) an adaptive algorithm.

Abstract: An acoustic feedback control adaptive method, the input signal being a function of a captured signal and an estimation of an acoustic feedback, the method including the following steps: —determining an impulse response (RI) of a filter (A) according to a partition of time blocks (b0, . . . bi, . . . , bNb), according to the following steps of: —for each sub-block (h1,i, h2,i, . . . hj,i, . . . hNi,i) of each block of the impulse response (RI), calculating a frequency transform (F1,i, F2,i, . . . Fj,i, . . . FNi,i); —repeating the following steps of: —applying the filter (A) to the output signal (u) using the frequency transform (F1,i, F2,i, . . . Fj,i, . . . FNi,i) of each sub-block (h1,i, h2,i, . . . hj,i, . . . hNi,i); —updating the frequency transform (F1,i, F2,i, . . . Fj,i, . . . FNi,i) of each sub-block (h1,i, h2,i, . . . hj,i, . . . hNi,i) as a function of the output signal and the input signal based on the same partition as that used in the step of applying the filter (A).

Abstract: Apparatuses, systems, and techniques to perform federated training of neural networks while maintaining control over dissemination of local models of neural networks from which aspects of local training data might be extracted. In at least one embodiment, a neural network is trained on local training data and a local model is provided to be aggregated with other local models into a global model that is in turn used for further local model training, wherein a provided local model or training is adjusted to reduce an ability to extract aspects of local training data therefrom.

Abstract: An image capture device with beamforming for wind noise optimized microphone placements is described. The image capture device includes a front facing microphone configured to capture an audio signal. The front facing microphone co-located with at least one optical component. The image capture device further includes at least one non-front facing microphone configured to capture an audio signal. The image capture device further includes a processor configured to generate a forward facing beam using the audio signal captured by the front facing microphone and the audio signal captured by the at least one non-front facing microphone, generate an omni beam using the audio signal captured by the at least one non-front facing microphone, and output an audio signal based on the forward facing beam and the omni beam.

Abstract: The present embodiments generally relate to enabling participants in an online gathering with networked audio to use a cancelling auralizer at their respective locations to create a common acoustic space or set of acoustic spaces shared among subgroups of participants. For example, there are a set of network connected nodes, and the nodes can contain speakers and microphones, as well as participants and node mixing-processing blocks. The node mixing-processing blocks generate and manipulate signals for playback over the node loudspeakers and for distribution to and from the network. This processing can include cancellation of loudspeaker signals from the microphone signals and auralization of signals according to control parameters that are developed locally and from the network.

Abstract: A system and method for determining the position and orientation of a wearable audio device, for example, methods and systems for determining the position, orientation, and/or height of a wearable audio device using acoustic beacons. In some examples, the determined position, orientation, and/or height can be utilized to correct for drift experienced by an inertial measurement unit (IMU). In other examples, the drift may cause am externalized or virtualize audio source, generated within a known environment, to move or drift relative to the known locations of physical audio sources within the environment. Thus, the systems and methods described herein can be utilized to correct for drift in the position of a virtual audio source with respect to the wearable audio device by first determining its own absolute position and orientation within the environment.

Abstract: A noise canceling audio in/out device includes an audible in/out transducer operable to convert an audible noise signal to a noise signal and convert an audio transmit (TX) signal to an audible output signal. A transducer signal of the audible in/out transducer generated by the audio TX signal is affected by the noise signal. The noise canceling audio in/out device further includes a noise canceling circuit operable to convert a digital TX signal to a TX reference signal, compare the transducer signal with the TX reference signal to produce an analog comparison signal, where the analog comparison signal includes a representation of the audio TX signal and the noise signal, and regulate the transducer signal to substantially match the TX reference signal to remove the effect of the noise signal on the transducer signal.

Abstract: Methods and systems are provided for an automatic noise control system. Automatic noise control includes evaluating an amplitude of an acceleration acting on an acceleration sensor and generating a reference signal representative of the amplitude of the acceleration, the acceleration being representative of unwanted noise sound generated by a noise source, filtering the reference signal with a noise control transfer function to generate an anti-noise signal, and converting with a loudspeaker the anti-noise signal into anti-noise sound.

Abstract: An adaptive active noise control (ANC) system includes an ANC circuit and a control circuit. The ANC circuit generates an anti-noise signal for noise reduction, wherein the ANC circuit includes at least one adaptive filter. The control circuit receives a first input signal derived from a reference signal output by a reference microphone that picks up ambient noise, receives a second input signal derived from an error signal output by an error microphone that picks up remnant noise resulting from the noise reduction, and performs a transfer function variation detection based on the first input signal and the second input signal to control the at least one adaptive filter.

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 noise controlling method includes generating a reference signal representing a primary noise, generating a secondary noise in response to a control signal for cancelling the primary noise, generating an error signal representing a superposition of the primary and secondary noises at a position, generating an additional reference signal, secondary noise, or additional error signal, and generating the control signal for generating the secondary noise using adaptive subband filtering based on the reference and error signals, the generating the control signal including decomposing the reference signal and the error signal into a subband reference and error signal for each subband, updating subband adaptive filters for a subband based on the subband reference signal and the subband error signal, updating a fullband adaptive filter based on the updated subband adaptive filter, and generating the control signal by filtering the reference signal by the updated fullband adaptive filter.

Abstract: A public address system includes audio inputs from a moderator/presenter and from one or more participants. In an embodiment, the moderator speaks first and then selects participants to speak, utilizing audio captured by participant devices. A central signal processor is configured to receive the audio inputs and to utilize a configured acoustic model to provide for acoustic echo cancellation (AEC) and feedback control (FBC) during various phases of a presentation or conference. Audio signals from the presenter and/or participants, that have been processed to remove echo, are utilized as reference signals during various phases of the audio presentation that utilize the acoustic model for either AEC or FBC. The system utilizes the knowledge that the best learning occurs during the far talking state to learn the echo path in the canceler mode (AEC) vs. the feedback mode (FBC), which usually can only train in a double-talking mode.

Abstract: An adaptive active noise control (ANC) system includes an ANC circuit and a control circuit. The ANC circuit generates an anti-noise signal, and includes at least one adaptive filter. The control circuit receives a first input signal derived from a reference signal output by a reference microphone that picks up ambient noise, receives a second input signal derived from an error signal output by an error microphone that picks up remnant noise resulting from noise reduction, and performs a comparison operation based on a first characteristic value of the first input signal and a second characteristic value of the second input signal to control the at least one adaptive filter.

Abstract: A noise-cancellation system includes a noise-cancellation filter in communication with at least one speaker, the noise-cancellation filter generating a noise-cancellation signal that, when actuated by the at least one speaker, cancels noise within at least one cancellation zone; and an amplifier disposed between the noise-cancellation filter and the speaker, the amplifier applying a first scaling gain to the noise-cancellation signal and outputting an scaled noise-cancellation signal, the scaled noise-cancellation signal being a linear reduction of the noise-cancellation signal when the first scaling gain is less than unity, wherein the first scaling gain is set to less than unity in response to a signal representative of the noise-cancellation signal exceeding a threshold.

Abstract: A vehicle-implemented, adaptive noise-cancellation system responsive to entertainment audio is provided. The noise-cancellation system uses reference signal from a reference sensor, such as an accelerometer, to generate a noise-cancellation signal to destructively interfere with road noise in the vehicle cabin. A first set of entertainment audio thresholds triggers the system to enable or disable adaptation of an adaptive filter of the noise-cancellation system. A second set of entertainment audio thresholds triggers the system to enable, attenuate, or disable the noise-cancellation signal. As the entertainment audio increases, the system first disables the adaptation of the adaptive filter, then attenuates the noise-cancellation signal, then completely disables the noise-cancellation signal.

Abstract: Described herein are system and method embodiments for adaptive noise control for headphones, specifically for open-ear headphones. A leakage detection module in an ambient sound control (ASC) circuit implements leakage detection to determine a leakage mode. Based on the determined leakage mode, an ASC profile may create, select or modify an ASC profile for the ASC circuit to operate. Pilot tone, ambient noise, or audio playback may be used respectively or in combination for leakage detection. Experimental results show that embodiments of adaptive ASC approach may achieve improved performance compared to a default ASC, especially under loose fitting of an earphone.

Abstract: A monitoring system can include an earpiece, a database with stored earpiece characteristics data, and a microphone to receive a plurality of signals where each signal of the plurality of signals can represents a respective sound pressure level of sound pressure values over a time duration. The system can also include a processor and a memory coupled processor, the memory having computer instructions which when executed by the processor causes the processor to perform the operations. The operations can include determining exposure time duration when a signal of the plurality of signals exceeds a sound pressure level threshold value, retrieving a subset of data of the stored earpiece characteristics data, and modifying an acoustic output of the earpiece in accordance with the subset of data of the stored earpiece characteristics data.

Abstract: Methods and systems are provided for automatic noise control. Automatic noise control includes controlling a shadow noise control transfer function based on a shadow error signal and a filtered or unfiltered reference signal, generating the shadow error signal based on a filtered or unfiltered shadow anti-noise signal and an error signal, and substituting the noise control transfer function by the shadow noise control transfer function if the shadow error signal is smaller than the error signal.

Abstract: The present invention provides an improved noise separation hybrid type ANC system, which includes a reference audio receiving device, an error audio receiving device, an audio output device, and an audio processing device. The audio processing device includes a feed-forward noise cancellation filter module, a feedback noise cancellation filter module, a mixer, a noise shaper, a first infinite impulse response filter, and a second infinite impulse response filter. When the noise bandwidth detector detects irregular noise, it adjusts the coefficient of the first infinite impulse response filter to set it as a low-pass filter; when regular noise is detected, the coefficient of the second infinite impulse response filter is adjusted to set it as a band-pass filter.

Abstract: An urban air mobility (UAM) noise reduction system and method are provided, where the UAM noise reduction system includes a UAM configured to detect and to provide rotation per minute (RPM) information of a propeller and location coordinate information, and a noise canceling device configured to predict a noise canceling sound wave amplitude on the basis of the RPM information of the propeller and the location coordinate information received through the UAM and to output a noise canceling sound wave corresponding to the predicted noise canceling sound wave amplitude to the UAM.

Abstract: An apparatus for generating at least one distance estimate to at least one sound source within a sound scene comprising the least one sound source, the apparatus configured to: receive at least two audio signals from a microphone array located within the sound scene; receive at least one further audio signal associated with the at least one sound source; determine at least one portion of the at least two audio signals from a microphone array corresponding to the at least one further audio signal associated with the at least one sound source; determine a distance estimate to the at least one sound source based on the at least one portion of the at least two audio signals from a microphone array corresponding to the at least one further audio signal associated with the at least one sound source.

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 sound processing apparatus includes a receiving module configured to receive audio signals of one or more sounds acquired by a personal sound device, a processing module configured to use a sound processing model to perform: classification processing in which a type of a scenario where a user of the personal sound device is located is determined based on the audio signals; identification processing in which each of the one or more sounds is determined as a desired sound or an undesired sound based on the determined type of the scenario, and filtering processing in which filtering configuration is performed based on a result of the identification processing. The audio signals are filtered based on the filtering configuration, and an output module is configured to output the filtered audio signals, so as to provide same to the user.

Abstract: A noise cancellation circuit includes: a first filter circuit for filtering a first input signal according to a first filter coefficient to generate a first filtered signal; a signal processing circuit for generating a feedback signal according to a second input signal and an audio signal; a second filter circuit for filtering the feedback signal according to a second filter coefficient to generate a second filtered signal; a first multiplication circuit for multiplying the first filtered signal by a first scale to generate a first intermediate signal; a second multiplication circuit for multiplying the second filtered signal by a second scale to generate a second intermediate signal; a first adder circuit for adding the first intermediate signal to the second intermediate signal to generate a noise cancellation signal; and a second adder circuit for adding the noise cancellation signal to the audio signal to generate an output signal.

Abstract: An integrated circuit for implementing at least a portion of a personal audio device may include an output for providing a signal to a transducer including both a source audio signal for playback to a listener and an anti-noise signal for countering the effects of ambient audio sounds in an acoustic output of the transducer, a reference microphone input for receiving a reference microphone signal indicative of the ambient audio sounds, an error microphone input for receiving an error microphone signal indicative of the output of the transducer and the ambient audio sounds at the transducer, and a processing circuit configured to implement an adaptive infinite impulse response filter having a response that generates the anti-noise signal to reduce the presence of the ambient audio sounds at the error microphone and implement a coefficient control block that shapes the response of the adaptive infinite impulse response filter in conformity with the error microphone signal by generating coefficients that determi

Abstract: A system configured to perform distributed echo cancellation processing to attenuate feedback echo from occurring when two devices are acoustically coupled during a communication session. To reduce the feedback echo, one of the devices is configured as a hub device and receives microphone signals, synchronizes the microphone signals, and generates a mixed microphone signal. To enable distributed echo cancellation, the system includes bidirectional feedback link(s) between the hub device and each device synchronized with the hub device. For example, a first bidirectional feedback link sends a microphone signal from a second device to the hub device and sends the mixed microphone signal from the hub device to the second device, which the second device uses to perform echo cancellation. In addition, a second bidirectional feedback link sends a playback signal from the hub device to the second device and sends the output of echo cancellation back to the hub device.

Abstract: Systems and methods of providing improved directional noise suppression in an electronic device implement a technique that specifies a direction or speaker of interest, determines the directions corresponding to speakers not lying in the direction of interest, beam forms the reception pattern of the device microphone array to focus in the direction of interest and suppresses signals from the other directions, creating beam formed reception data. A spatial mask is generated as a function of direction relative to the direction of interest. The spatial mask emphasizes audio reception in the direction of interest and attenuates audio reception in the other directions. The beam formed reception data is then multiplied by the spatial mask to generate an audio signal with directional noise suppression.

Abstract: A first cancellation signal output from a first speaker cancels noise at a first cancellation point, which is a typical position of the right ear of a user, together with a second cancellation signal output from a second speaker. In addition, the second cancellation signal output from the second speaker cancels noise at a second cancellation point, which is a typical position of the left ear of the user, together with the first cancellation signal output from the first speaker. The first speaker and the second speaker are arranged side by side on a second line segment, which passes through the midpoint of a first line segment connecting the first cancellation point and the second cancellation point to each other and is perpendicular to the first line segment, and a range where the relationship between noise and the first cancellation signal and the second cancellation signal is the same as that at the cancellation point is extended.

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 conversation support device includes: a speaker; a microphone; a noise source acquisition unit that acquires a noise signal indicating noise; a first calculator that calculates a transfer characteristic of a secondary path between the speaker and the microphone; an echo cancellation unit that cancels an echo by using the transfer characteristic of the secondary path; a second calculator that calculates a coefficient of an adaptive filter, based on the transfer characteristic of the secondary path and the noise signal; and an active noise cancellation controller that generates a noise cancelling signal by using the coefficient of the adaptive filter and the noise signal. The noise cancelling signal is for controlling cancellation of the noise.

Abstract: In a first system signal processing unit, an adaptive filter generates a noise cancel sound, a first system selector selects an output of a first system auxiliary filter corresponding to a noise cancel position matching a detected position of a right ear of a user from a plurality of first system auxiliary filters corresponding to different noise cancel positions, and a first system subtractor subtracts the selected output from an output of a first microphone and outputs the subtracted result as an error signal to a first system adaptive filter and a second system adaptive filter of a second system signal processing unit. The noise cancel positions are arranged at predetermined intervals in a space where the user can move the right ear due to turning and side bending of the head within a predetermined range in the up-down and front-back directions.

Abstract: An audio system that includes an IMU or accelerometer in physical proximity to a microphone and a driver. The signal generated by the IMU or accelerometer may be used to filter the audio signal generated by the microphone with respect to mechanically coupling between the driver outputting sound and the microphone via a shared structure. In some cases, an accelerometer may be incorporated into a package of the microphone to provide analog-based filtering prior to converting the audio signal of the microphone to a digital format.

Abstract: The present subject matter can improve robustness of performance of acoustic feedback cancellation in the presence of strong acoustic disturbances. In various embodiments, an optimization criterion determined to enhance robustness of an adaptive feedback canceller in an audio device against disturbances in an incoming audio signal can be applied such that the adaptive feedback canceller remains in a converged state in response to presence of the disturbances.

Abstract: A method of automated feedforward filter design comprising designing a feedforward filter for a system implementing active noise cancelling is described. The method includes designing the feedforward filter by determining a filter transfer function of the feedforward filter. Optionally, the filter transfer function is determined using a least square method. The method also includes determining the filter transfer function by defining a target transfer function of the feedforward filter and applying the least square method using the target transfer function to determine a filter expression for the filter transfer function. Optionally, the least square method is a weighted least square method.

Abstract: An image capture device with beamforming for wind noise optimized microphone placements is described. The image capture device includes a front facing microphone configured to capture an audio signal. The front facing microphone co-located with at least one optical component. The image capture device further includes at least one non-front facing microphone configured to capture an audio signal. The image capture device further includes a processor configured to generate a forward facing beam using the audio signal captured by the front facing microphone and the audio signal captured by the at least one non-front facing microphone, generate an omni beam using the audio signal captured by the at least one non-front facing microphone, and output an audio signal based on the forward facing beam and the omni beam.

Abstract: Some demonstrative embodiments include apparatuses, systems and methods of sound control. For example, an apparatus may be configured to process one or more audio inputs to be heard in one or more personal sound zones, and a plurality of monitoring inputs, wherein the plurality of monitoring inputs represent acoustic sound at a plurality of predefined monitoring sensing locations, which are defined within the one or more personal sound zones; determine a sound control pattern based on the one or more audio inputs, and the plurality of monitoring inputs, the sound control pattern comprising a plurality of sound control signals configured to drive a respective plurality of acoustic transducers such that the one or more audio inputs are to be heard in the one or more personal sound zones; and output the plurality of sound control signals to the plurality of acoustic transducers.

Abstract: A data processing device includes an internal volume for housing devices and an isolation test system. The isolation test system makes a determination that an electromagnetic interference suppression state of the data processing device is to be determined; in response to the determination: suppresses generation of electromagnetic radiation by the devices; while the generation of the electromagnetic radiation by the devices is suppressed: identifies a quantity of second electromagnetic radiation that propagated through a boundary of the internal volume; makes a second determination, based at least in part on the quantity, that the electromagnetic interference suppression state of the data processing device is an electromagnetic interference suppression compromised state; and initiates performance of an action set to remediate the electromagnetic interference suppression compromised state of the data processing device based on the second determination.

Abstract: Some demonstrative embodiments include apparatuses, systems and methods of sound control. For example, an apparatus may be configured to process one or more audio inputs to be heard in one or more personal sound zones, and a plurality of monitoring inputs, wherein the plurality of monitoring inputs represent acoustic sound at a plurality of predefined monitoring sensing locations, which are defined within the one or more personal sound zones; determine a sound control pattern based on the one or more audio inputs, and the plurality of monitoring inputs, the sound control pattern comprising a plurality of sound control signals configured to drive a respective plurality of acoustic transducers such that the one or more audio inputs are to be heard in the one or more personal sound zones; and output the plurality of sound control signals to the plurality of acoustic transducers.

Abstract: Disclosed herein are methods for operating an apparatus for generating acoustic compensation signals used to compensate acoustic signals from operation of a motor-vehicle engine, comprising the steps: providing an EOC device, which is designed to generate acoustic compensation signals used to compensate acoustic signals that result from the operation of the engine; providing a device configured using the EOC device as a model; determining at least one audio signal to be output into a passenger compartment of a motor vehicle by means of an audio output device, before said audio signal is captured by an audio capturing element associated with the EOC device; applying an evaluation specification to evaluate the at least one audio signal with respect to at least one evaluation criterion; generating evaluation information with respect to the at least one evaluation criterion; controlling the operation of the EOC device on the basis of the evaluation Information.

Abstract: Exemplary road and engine noise control systems and methods include directly picking up road noise from a structural element of a vehicle to generate a first sense signal representative of the road noise, directly picking up engine noise from an engine of the vehicle to generate a second sense signal representative of the engine noise, and combining the first sense signal and the second sense signal to provide a combination signal representing the combination of the first sense signal and the second sense signal. The systems and methods further include broadband active noise control filtering to generate a filtered combination signal from the combination signal, converting the filtered combination signal provided by the active noise control filtering into anti-noise and radiating the anti-noise to a listening position in an interior of the vehicle. The filtered combination signal is configured so that the anti-noise reduces the noise at the listening position.

Abstract: An ear cup housing has several reference microphones, an error microphone and a speaker. A processor drives the speaker for acoustic noise cancellation and transparency, by processing the microphone signals, and performs an oversight process by adjusting the reference microphone signals in response to detecting wind noise events and scratch events. In another aspect, the ear cup housing has an outside face that is joined to an inside face by a perimeter and the reference microphones are on the perimeter. Other aspects are also described and claimed.

Abstract: A monitoring system can include an earpiece, a database with stored earpiece characteristics data, and a microphone to receive a plurality of signals where each signal of the plurality of signals can represents a respective sound pressure level of sound pressure values over a time duration. The system can also include a processor and a memory coupled processor, the memory having computer instructions which when executed by the processor causes the processor to perform the operations. The operations can include determining exposure time duration when a signal of the plurality of signals exceeds a sound pressure level threshold value, retrieving a subset of data of the stored earpiece characteristics data, and modifying an acoustic output of the earpiece in accordance with the subset of data of the stored earpiece characteristics data.

Abstract: Provided is a signal processing apparatus of a plurality of signal processing apparatuses that includes a noise canceling processing unit capable of connecting one or more input portions and one or more output portions, and performs a noise canceling processing. Signal processing apparatuses of the plurality of signal processing apparatuses are connected to one another.

Abstract: An earphone signal processing method includes: a signal picked up by a first microphone of an earphone at a position close to a mouth outside an ear canal, a signal picked up by a second microphone of the earphone at a position away from the mouth outside the ear canal and a signal picked up by a third microphone are acquired, the third microphone being in a cavity formed by the earphone and the ear canal; dual-microphone noise reduction is performed on the signals picked up by the first and second microphones to obtain a first intermediate signal; dual-microphone noise reduction is performed on the signals picked up by the second and third microphones to obtain a second intermediate signal; the first and second intermediate signals are fused to obtain a fused voice signal; and the fused voice signal is output.

Abstract: According to various embodiments, an electronic device includes a plurality of microphones, and a processor electrically connected to the plurality of microphones, wherein the processor may obtain audio signals through the plurality of microphones, estimate a probability of existence of a voice signal included in the obtained audio signals, obtain correlation information between the audio signals based on the probability of existence of the voice signal and/or the obtained audio signals, obtain voice blocking information based on the correlation information or a direction of arrival (DOA) estimation, obtain a first signal among the audio signals based on the audio signals, the correlation information, and the voice blocking information, obtain a second signal including the voice signal among the audio signals, and obtain a noise-removed voice signal by removing the first signal from the second signal. In addition, it is possible to implement various embodiments understood through the disclosure.

Abstract: An active noise control device includes an updated value table operating unit that writes initial values of an initial value table into an updated value table as updated values and writes updated coefficients of a secondary path filter C{circumflex over (?)} updated in a secondary path filter coefficient updating unit during active noise control, into the updated value table as updated values. The secondary path filter coefficient updating unit reads the updated value corresponding to a frequency from the updated value table before updating the coefficients of the secondary path filter C{circumflex over (?)}, and updates the coefficients of the secondary path filter C{circumflex over (?)}, using the read updated values as the pervious values.