Abstract: Technology described in this document can be embodied in a method that includes receiving a first input signal captured by at least a first feedforward microphone associated with an active noise reduction (ANR) device, receiving a second input signal captured by at least a second feedforward microphone associated with the ANR device, processing the first input signal using a first filter disposed in a first ANR signal flow path to generate a first output signal for an acoustic transducer of the ANR device, processing the second input signal using a second filter disposed in a second ANR signal flow path to generate a second output signal for the acoustic transducer, in which the second filter is different from the first filter, and generating a combined signal for the acoustic transducer based on combining the first output signal with the second output signal.

Abstract: [Object] To allow a listener to listen to ambient sounds of the external environment in an appropriate manner while wearing a head mounted acoustic device.

Abstract: An improved system and method for reducing the ambient noise experienced by a user listening to an earpiece without the use of a microphone is disclosed. An “ambient noise signal” created by the sound pressure wave of the ambient noise acting on the earpiece transducer is obtained. In some embodiments, the ambient noise signal is inverted and fed back, and the inverted signal is added to the intended audio signal being sent to the earpiece so that the ambient noise is cancelled. In other embodiments, a processor receives the ambient noise signal and predicts the modification to the intended audio signal needed to counteract the ambient noise. The ambient noise signal may be obtained by comparing the actual signal across the earpiece transducer to the intended audio signal, or by detecting variations in the current across the transducer from the current generated to drive the transducer.

Abstract: An audio system is configured to present a modified audio experience that reduces the degradation of a target audio experience presented to a user by the audio system. The audio system includes an acoustic sensor array, a controller, and a playback device array. To generate the modified audio experience, the acoustic sensor array receives the sound waves from one or more non-target audio source(s) causing the degradation, identifies the audio source(s), determines the spatial location of the audio source(s), determines the type of the audio source(s) and generates audio instructions that, when executed by the playback device array, present the modified audio experience to the user. The modified audio experience may perform active noise cancelling, ambient sound masking, and/or neutral sound masking to compensate for the sound waves received from non-target audio sources. The audio system may be part of a headset that can produce an artificial reality environment.

Abstract: A headset includes a first sound receiving circuit, a second sound receiving circuit, an adaptive circuit, a first synthesis circuit, and a second synthesis circuit. The adaptive circuit is configured to: obtain a first direction of arrival and a second direction of arrival according to a first sound signal and a second sound signal; obtain a first conversion function and a second conversion function according to the first direction of arrival and the second direction of arrival; obtain a first feed forward audio signal according to the first conversion function and the first sound signal; and obtain a second feed forward audio signal according to the second conversion function and the second sound signal.

Abstract: The specification and drawings present a use of multiple microphones for increasing acoustic sensing capabilities by processing acoustic signals from the multiple microphones in outdoor luminaire mounted surveillance/sensor systems. For example, various embodiments presented herein describe signal processing means to utilize stereo/multiple microphones in a luminaire (such as an outdoor roadway luminaire) to provide enhanced information regarding the surroundings of the luminaire. The multiple microphone luminaire sensor processing system can provide a more environmentally robust and sensitive approach which can be, for example, resistant to environmental noise such as a wind noise, as well as capable of isolating specific sounds from the surroundings, e.g., in specific directions.

Abstract: Systems and methods for optimizing network microphone devices using noise classification are disclosed herein. In one example, individual microphones of a network microphone device (NMD) detect sound. The sound data is analyzed to detect a trigger event such as a wake word. Metadata associated with the sound data is captured in a lookback buffer of the NMD. After detecting the trigger event, the metadata is analyzed to classify noise in the sound data. Based on the classified noise, at least one performance parameter of the NMD is modified.

Abstract: A system configured to perform power normalization for voice enhancement. The system may identify active intervals corresponding to voice activity and may selectively amplify the active intervals in order to generate output audio data at a near uniform loudness. The system may determine a variable gain for each of the active intervals based on a desired output loudness and a flatness value, which indicates how much a signal envelope is to be modified. For example, a low flatness value corresponds to no modification, with peak active interval values corresponding to the desired output loudness and lower active intervals being lower than the desired output loudness. In contrast, a high flatness value corresponds to extensive modification, with peak active interval values and lower active interval values both corresponding to the desired output loudness. Thus, individual words may share the same peak power level.

Abstract: The present technology relates to a locally silenced sound field forming apparatus and method capable of controlling a silenced area in a depth direction. The locally silenced sound field forming apparatus includes a first speaker array that outputs a sound on the basis of a first speaker drive signal to form a predetermined sound field, and a second speaker array arranged at a position different from the position of the first speaker array and that outputs a sound on the basis of a second speaker drive signal to form a sound field that cancels the predetermined sound field. The present technology can be applied to a locally silenced sound field forming apparatus.

Abstract: A non-linearity evaluation circuit for use with a signal generator having at least a partly non-linear operation. The non-linearity evaluation circuit may include a detection unit operable to detect a given amplitude attribute in a target signal generated by the signal generator, a time position of the amplitude attribute in the target signal defining a time location of a snapshot time window relative to the target signal, a part of the target signal occupying the snapshot time window being a corresponding signal snapshot, and a presence of the given amplitude attribute indicating that the signal snapshot includes noise due to the non-linear operation of the signal generator. The non-linearity evaluation circuit may further include a controller operable to analyse the signal snapshot rather than a larger part of the target signal and to evaluate the non-linear characteristics of the operation of the signal generator based on the analysis.

Abstract: An echo cancellation system that synchronizes output audio data with input audio data in a heterogeneous system. The system may append a most recent outgoing audio data frame to an incoming audio data frame to synchronize a receiving buffer with a transmission buffer. By synchronizing the receiving buffer with the transmission buffer, the incoming audio data frames may be associated with corresponding outgoing audio data frames. After synchronizing the incoming audio data frames and the outgoing audio data frames, the system may perform Acoustic Echo Cancellation by removing the outgoing audio data frames from the incoming audio data frames.

Abstract: A computationally implemented system and method that is designed to, but is not limited to: electronically providing audio output information to one or more portions of a portable electronic device to be outputted from said portable electronic device via one or more acoustic ultrasonic signals; and electronically outputting, said one or more acoustic ultrasonic signals to be demodulated into one or more acoustic audio signals containing one or more portions of said audio output information at one or more locations spaced from said portable electronic device based at least in part according to said one or more acoustic ultrasonic signals and based at least in part according to one or more portable electronic device ultrasonic emitter arrangements. In addition to the foregoing, other method aspects are described in the claims, drawings, and text forming a part of the present disclosure.

Abstract: A signal processing apparatus, for processing sounds collected in an environment where a target sound and an interfering sound are mixed in order to estimate a diffuse interfering sound accurately, is provided. The signal processing apparatus includes phase difference calculating means and generating means. The phase difference calculating means calculates a phase difference between the first input signal and the second input signal. The first input signal is generated based on the first input sound which is input in the environment where the target sound and the interfering sound are mixed. The second input signal is generated based on the second input sound which is input in the environment. The generating means generates an estimated interfering sound signal, based on the phase difference and the first input signal.

Abstract: A beamformer system includes an in-ear audio device, such as an earbud, that has three microphones. Two microphones may be disposed on an external face of the audio-device; one microphone may be disposed in or near the ear canal of a user. Data from two microphones is phase-adjusted and combined; a reference signal is generated by phase-adjusting and removing data from the two microphones from data from a third microphone. The combined data is filtered using the reference signal to remove any residual echo, and the resulting data may be used for communications, speech processing, or other uses.

Abstract: A system, method and storage medium for noise cancellation in a vehicle includes determining a waveform of a first sound wave at a first location, calculating another waveform of the first sound wave at a second location of an operator based on the waveform of the first sound wave at the first location and a first distance between the first location and the second location, generating at least one control signal based on the determined another waveform of the first sound wave at the second location, and generating a second sound wave based on the at least one control signal. A waveform of the second sound wave are formed to cancel out the first sound wave at the second location. The first sound wave is generated by a noise source.

Abstract: A controller of a collaboration endpoint generates a primary audio signal for an ultrasonic source audio signal produced by a source audio speaker, a reference audio signal for the ultrasonic source audio signal, and, based on the reference audio signal, a predicted signal that is predictive of the primary audio signal. The controller produces a prediction error of the predicted signal by comparing the primary audio signal with the predicted signal and determines whether the prediction error is indicative of a motion of one or more persons near the collaboration endpoint.

Abstract: A method for adjusting a vehicle seat includes activating a pump at a selected speed from a number of selectable speeds to inflate a first bladder and adjusting an inner chamber within a sound dampening assembly that is fluidically coupled between the first bladder and the pump. The inner chamber is adjusted to cancel at least one primary frequency generated by the pump that varies with the selectable speeds thereof.

Abstract: The present technology relates to a sound processing apparatus and a sound processing system for enabling more stable localization of a sound image. A virtual speaker is assumed to exist on the lower side among the sides of a tetragon having its corners formed with four speakers surrounding a target sound image position on a spherical plane. Three-dimensional VBAP is performed with respect to the virtual speaker and the two speakers located at the upper right and the upper left, to calculate gains of the two speakers at the upper right and the upper left and the virtual speaker, the gains being to be used for fixing a sound image at the target sound image position. Further, two-dimensional VBAP is performed with respect to the lower right and lower left speakers, to calculate gains of the lower right and lower left speakers, the gains being to be used for fixing a sound image at the position of the virtual speaker.

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

Abstract: An improved system and method for reducing the ambient noise experienced by a user listening to an earpiece without the use of a microphone is disclosed. An “ambient noise signal” created by the sound pressure wave of the ambient noise acting on the earpiece transducer is obtained. In some embodiments, the ambient noise signal is inverted and fed back, and the inverted signal is added to the intended audio signal being sent to the earpiece so that the ambient noise is cancelled. In other embodiments, a processor receives the ambient noise signal and predicts the modification to the intended audio signal needed to counteract the ambient noise. The ambient noise signal may be obtained by comparing the actual signal across the earpiece transducer to the intended audio signal, or by detecting variations in the current across the transducer from the current generated to drive the transducer.

Abstract: A system and method for reducing or eliminating undesirable acoustic artifacts when a vehicle is struck by road debris. The method includes generating a noise signal representative of a first acceleration detected by an accelerometer of a vehicle caused by a disturbance 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 disturbance is detected by a reference sensor, which generates a reference sensor signal based on the residual noise. The reference sensor signal is transmitted to an adaptive processing module to adapt filter coefficients. A second acceleration is detected by the accelerometer and a level detector calculates the absolute value of a derivative of the second acceleration. If the absolute value exceeds a threshold value, adjustment of the filter coefficients is prevented.

Abstract: Systems and methods for active noise cancellation are provided. An example method includes receiving at least two reference signals associated with at least two reference positions. Each of the at least two reference signals includes at least one captured acoustic sound representing an unwanted noise. The reference signals are filtered by individual filters to obtain filtered signals. The filtered signals are combined to obtain a feedforward signal. The feedforward signal is played back to reduce the unwanted noise at a pre-determined space location. The individual filters are determined based on linear combinations of at least two transfer functions. Each of the at least two transfer functions is associated with one of the reference positions. In certain embodiments, the at least two reference signals are captured by at least two feedforward microphones.

Abstract: An electronic device is provided. The electronic device includes an audio module including a plurality of audio reception units and a plurality of audio output units and a processor electrically connected to the audio module and configured to receive sound via the plurality of audio reception units, generate antiphase signals based on waveforms of the received sound, determine directions in which to emit the antiphase signals, based on locations of the plurality of audio reception units, and emit the antiphase signals via the plurality of audio output units.

Abstract: A planar element is provided for the active compensation of noise from different directions in an interior room. The planar element is equipped with anti-noise modules, each having a microphone and an electronic circuit having a combination of filters for adjusting phase and amplitude and an electronically controllable amplifier and a loudspeaker for active noise cancellation. The planar element is equipped over the width thereof with a plurality of anti-noise modules on each of a plurality of different longitudinally extending lines and is to be placed, for example, in front of a window having bounding walls. The anti-noise modules can be inconspicuously integrated into the planar element and jointly form longitudinally extending chains of anti-noise modules. The loudspeakers emit a canceling sound with respect to the noise, which canceling sound first passes through the planar element and subsequently largely cancels the noise.

Abstract: A vehicle includes an engine and/or powertrain producing noise that is audible in a passenger compartment of the vehicle when the engine and/or powertrain is running. An active noise control arrangement includes a first loudspeaker disposed within a passenger compartment of the vehicle. A digital signal processor receives audio data and transmits an audio signal to the first loudspeaker dependent upon the audio data. A microphone is disposed within the passenger compartment and converts the sound from the first loudspeaker and the noise within the passenger compartment into a microphone signal. The microphone signal is transmitted to the digital signal processor, and the digital signal processor modifies the audio signal such that the audio signal attenuates the noise in the passenger compartment.

Abstract: A system and method for active noise compensation in a vehicle is provided. The system has at least one apparatus for detecting a noise, a computer unit for calculating a compensation sound, at least one apparatus for generating a compensation sound which interferes with the noise, and a device for detecting at least one driver-specific parameter. The compensation sound is calculated depending on the at least one driver-specific parameter. Also provided is a motorcycle having a system for active noise compensation.

Abstract: A method and a system for calibrating a surround sound system are disclosed. The calibration system can provide a graphical user interface for display comprising a visual representation of the room hosting a multichannel surround sound system. The graphical user interface can permit user input of gestures to place or make changes to the placement of icons representing one or more loudspeakers and a listener. The calibration system can estimate the positions of the one or more loudspeakers or the listener based on the placement of the icons in the model room. A spatial calibration based on the estimated positions can then be performed such that the multichannel surround sound system can render sound scenes more accurately.

Abstract: The disclosure includes a headset comprising one or more earphones including one or more sensing components. The headset also includes one or more voice microphones to record a voice signal for voice transmission. The headset also includes a signal processor coupled to the earphones and the voice microphones. The signal processor is configured to employ the sensing components to determine a wearing position of the headset. The signal processor then selects a signal model for noise cancellation. The signal model is selected from a plurality of signal models based on the determined wearing position. The signal processor also applies the selected signal model to mitigate noise from the voice signal prior to voice transmission.

Abstract: Methods, apparatus, systems and articles of manufacture to implement selective suppression of audio emitted from an audio source are disclosed. Example methods disclosed herein include obtaining, at a portable device, data transmitted at a first time by a transmitter associated with an audio source, the data including reference audio data corresponding to a first audio signal to be output by the audio source at a second time after the first time, the data further specifying a time delay corresponding to a difference between the second time and the first time. Disclosed example methods also include generating, based on the reference audio data and the time delay, a suppression audio signal to provide to an audio output driver in communication with a speaker of the portable device to suppress the first audio signal when the first audio signal is output by the audio source at the second time.

Abstract: Method and system for active reduction of a predefined audio acoustic signal (AAAS), also referred to as “noise”, in a quiet zone, without interfering undefined acoustic noise signals within as well as outside the quiet zone, by generating accurate antiphase AAAS signal. The accuracy of the generated antiphase AAAS is obtained by employing a unique synchronization signal(s) (SYNC) which is generated and combined with the predefined AAAS. The combined signal is electrically transmitted (referred to as the “electric channel”) to a processing “quieting component”. Simultaneously, the generated SYNC signal is acoustically broadcasted near the predefined AAAS and merges with it. A microphone in the quiet zone receives the merged acoustic signals that arrive via the air (referred to as the “acoustical channel”) to the quiet zone and a receiver in the quieting component receives the combined electrical AAAS and SYNC signal that arrive wire or wireless to the quiet zone.

Abstract: Multi-sensor signal optimization is facilitated for speech communication. A sensor component including acoustic sensors can be configured to detect sound and generate, based on the sound, first sound information associated with a first sensor of the acoustic sensors and second sound information associated with a second sensor of the acoustic sensors. Further, an audio processing component can be configured to generate filtered sound information based on the first sound information, the second sound information, and a spatial filter associated with the acoustic sensors, determine noise levels for the first sound information, the second sound information, and the filtered sound information, and generate output sound information based on a selection of one of the noise levels or a weighted combination of the noise levels.

Abstract: A digital circuit arrangement for an ambient noise-reduction system affording a higher degree of noise reduction than has hitherto been possible. The arrangement converts the analog signals into N-bit digital signals at sample rate f0, and then subjects the converted signals to digital filtering. The value of N in some embodiments is 1 but, in any event, is no greater than 8, and f0 may be 64 times the Nyquist sampling rate but, in any event, is substantially greater than the Nyquist sampling rate. This permits digital processing to be used without incurring group delay problems that rule out the use of conventional digital processing in this context. Furthermore, adjustment of the group delay can readily be achieved, in units of a fraction of a micro-second, providing the ability to fine tune the group delay for feed forward applications.

Abstract: The present disclosure discloses a system for attenuating sound produced by a vehicle. The system comprises a navigation device associated with the vehicle for determining location details of the vehicle, a sound reduction unit provisioned in an exhaust assembly of the vehicle. The system further includes an Electronic Control Unit (ECU) of the vehicle communicatively coupled to the navigation device and the sound reduction unit. The ECU is configured to detect sound reduction location, by comparing the location details with a pre-defined location data and operate the sound reduction unit to attenuate sound produced by the vehicle when the sound reduction location is detected. The system of the present disclosure, attenuates sound intensity of the vehicle to a desired level by considering surrounding conditions of the vehicle, thus the vehicle noise may be automatically attenuated in the sound sensitive areas or locations.

Abstract: A system and method receives one or more captured signals through a captured audio path and produces one or more playback signals through a playback audio path. The system and method executes one or more signal processing functions and measures the delays within the playback audio path and captured audio path during operation of the one or more signal processing functions. The system and method stores the measured delays in a memory and compensates the one or more signal processing functions for the playback delay and the capture delay.

Abstract: A voice input/output device is connected to a smart device having a voice signal input/output function is provided. A digital voice signal output by the smart device is output to an output unit, (e.g., ear receiver or a speaker), in the form of a voice, and a voice signal transmitted by the smart device is extracted and input to the smart device as a digital voice signal to generate a sound by the smart device itself in addition to an externally generated sound that is input again with improved quality. Accordingly, a voice signal output by the smart device is extracted using a earphone type device having microphone mounted thereon and the voice signal is input to the smart device again.

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 extract audio feature data from the most recent snippet of the ambient audio stream and transmit the audio feature data and associated metadata to a knowledgebase remote from the personal audio system.

Abstract: A signal compensation method and an electronic device adapted to the method are provided. The signal compensation method includes storing filter data related to a hearing aid in a storage unit, recording a first sound signal, and compensating for the first sound signal, based on the filter data.

Abstract: Variable sound decomposition masking techniques are described. In one or more implementations, a mask is generated that incorporates a user input as part of the mask, the user input is usable at least in part to define a threshold that is variable based on the user input and configured for use in performing a sound decomposition process. The sound decomposition process is performed using the mask to assign portions of sound data to respective ones of a plurality of sources of the sound data.

Abstract: The present disclosure relates to a noise reduction device and a noise reduction method capable of reducing noise in a more stable and effective manner. Signal processing is performed to generate a reference signal representing a waveform of noise. Moreover, signal processing is performed for an error signal representing a waveform of an error measured by a microphone in accordance with an amplitude-frequency characteristic. Then, a filter coefficient with which the error signal becomes zero is calculated under adaptive algorithm with reference to the reference signal. The reference signal is filtered by using the filter coefficient to obtain a control signal. The control signal is supplied to the corresponding one of the predetermined number of output units. The present technology is applicable to a noise cancelling system equipped in a closed space such as an interior of a vehicle, for example.

Abstract: Data discovery for sensor data in an M2M network uses probabilistic models, such as Gaussian Mixing Models (GMMs) to represent attributes of the sensor data. The parameters of the probabilistic models can be provided to a discovery server (DS) that respond to queries concerning the sensor data. Since the parameters are compressed compared to the attributes of the sensor data itself, this can simplify the distribution of discovery data. A hierarchical arrangement of discovery servers can also be used with multiple levels of discovery servers where higher level discovery servers using more generic probabilistic models.

Abstract: An apparatus comprising: at least one first input configured to receive at least one first sensor signal, the at least one first sensor signal having a first modality; at least one second input configured to receive at least one second sensor signal, the at least one second sensor signal having a second modality different from the first modality; and a codec configured to generate at least one meta signal comprising and stored using separate resources the at least one first sensor signal and the at least one second sensor signal.

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: Disclosed is a method for optimizing noise cancellation in a headset, the headset comprising a headphone and a microphone unit comprising at least a first microphone and a second microphone, the method comprising: generating at least a first audio signal from the at least first microphone, where the first audio signal comprises a speech portion from a user of the headset and a noise portion from the surroundings; generating at least a second audio signal from the at least second microphone, where the second audio signal comprises a speech portion from the user of the headset and a noise portion from the surroundings; generating a noise cancelled output by filtering and summing at least a part of the first audio signal and at least a part of the second audio signal, where the filtering is adaptively configured to continually minimize the power of the noise cancelled output, and where the filtering is adaptively configured to continually provide that at least the amplitude spectrum of the speech portion of

Abstract: A Point-Of-Sale (POS) device detects vibrations produced from equipment in its operational environment. The vibrations are identified and one or more filters produced to remove the vibrations when the POS device is used to improve the accuracy of the POS device.

Abstract: The specification and drawings present a use of multiple microphones for increasing acoustic sensing capabilities by processing acoustic signals from the multiple microphones in outdoor luminaire mounted surveillance/sensor systems. For example, various embodiments presented herein describe signal processing means to utilize stereo/multiple microphones in a luminaire (such as an outdoor roadway luminaire) to provide enhanced information regarding the surroundings of the luminaire. The multiple microphone luminaire sensor processing system can provide a more environmentally robust and sensitive approach which can be, for example, resistant to environmental noise such as a wind noise, as well as capable of isolating specific sounds from the surroundings, e.g., in specific directions.

Abstract: A vehicle includes an engine and/or powertrain producing noise that is audible in a passenger compartment of the vehicle when the engine and/or powertrain is running. An active noise control arrangement includes a first loudspeaker disposed within a passenger compartment of the vehicle. A digital signal processor receives audio data and transmits an audio signal to the first loudspeaker dependent upon the audio data. A microphone is disposed within the passenger compartment and converts the sound from the first loudspeaker and the noise within the passenger compartment into a microphone signal. The microphone signal is transmitted to the digital signal processor, and the digital signal processor modifies the audio signal such that the audio signal attenuates the noise in the passenger compartment.

Abstract: A microphone array device includes a first sound reception unit configured to obtain a first sound signal that is input from a first microphone, a second sound reception unit configured to obtain a second sound signal that is input from a second microphone, a noise state evaluation unit configured to compare the first sound signal and the second sound signal and to obtain an evaluation parameter to evaluate an influence of a non-target sound included in the second sound signal on a target sound included in the first sound signal according to a result of the comparison, a subtraction adjustment unit configured to set a suppression amount for the second sound signal based on the evaluation parameter and to generate a third sound signal; and a subtraction unit configured to generate a signal to be output based on the third sound signal and the first sound signal.

Abstract: [Object] To provide a signal processing apparatus which can allow a user to listen to sound assumed by a designer while a difference among individuals is taken into account. [Solution] Provided is a signal processing apparatus including: a characteristics difference calculating unit configured to calculate a difference between reproduction target characteristics of a first sound signal and characteristics of a second sound signal obtained by collecting, at a microphone provided inside a headphone, sound output from a driver which outputs the sound based on a third sound signal obtained by performing signal processing on the first sound signal; and a sound signal processing unit configured to select a parameter to be used at the signal processing based on the difference calculated by the characteristics difference calculating unit, and perform signal processing on the first sound signal.

Abstract: Examples described herein involve identifying one or more error conditions during calibration of one or more playback devices in a playback environment. A microphone of a network device may detect and sample an audio signal while the one or more playback devices in the playback environment plays a calibration tone. A processor of the network device may then receive, from the microphone, a stream of audio data. The audio data may include an audio signal component and a background noise component. As a subset of the audio data is received, the processor may identify based on at least the subset of audio data, the one or more error conditions. The processor may then cause a graphical display to display a graphical representation associated with the identified error condition.

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.