Abstract: A speakerphone calibration system (system) is provided herein, comprising: a calibration unit adapted to generate at least one test signal, and further adapted to determine at least one calibration factor in response to at least one test signal, and wherein a first calibration factor characterizes a speakerphone system under test in regard to mechanical vibrations generated in the speakerphone system under test, the mechanical vibrations caused by a first test signal.

Abstract: Embodiments herein enable fast and easy interconnectivity among multimedia accessories including mobile devices and other devices. There is only limited space on mobile devices yet there are numerous input connectors. The standard TRRS audio jack is one such input that has and remains common, primarily because it is the accepted standard for audio input; namely, headphones and earpieces for listening purposes. Embodiments herein describe an intelligent switch to that audio jack that permits for additional backward and forward compatibility. It transparently allows a user to insert analog or digital audio devices, such as earphones, without the need to manually reconfigure device settings. The device herein automatically converts between input connector types using the same input convention present on their existing mobile devices. Other embodiments are disclosed.

Abstract: A system configured to perform cascade echo cancellation processing to improve a performance when reference signals are asymmetric (e.g., dominant reference signal(s) overshadow weak reference signal(s)). The system may perform cascade echo cancellation processing to separately adapt filter coefficients between the dominant reference signal(s) and the weak reference signal(s). For example, the system may use a dominant reference signal to process a microphone audio signal and generate a residual audio signal, using the residual audio signal to adapt first filter coefficient values corresponding to the dominant reference signal. Separately, the system may use a weak reference signal to process the residual audio signal and generate an output audio signal, using the output audio signal to adapt second filter coefficient values corresponding to the weak reference signal.

Abstract: A method and system for real-time auralization is described in which room sounds are reverberated and presented over loudspeakers, thereby augmenting the acoustics of the space. Room microphones are used to capture room sound sources, with their outputs processed in a canceler to remove the synthetic reverberation also present in the room. Doing so gives precise control over the auralization while suppressing feedback. It also allows freedom of movement and creates a more natural acoustic environment for performers or participants in music, theater, gaming, home entertainment, and virtual reality applications. Canceler design methods are described, including techniques for handling varying loudspeaker-microphone transfer functions such as would be present in the context of a performance or installation.

Abstract: A non-transitory computer-readable storage medium may include instructions stored thereon. When executed by at least one processor, the instructions may be configured to cause a computing system to determine that a video system is aiming at a single speaker of a plurality of people, receive audio signals from a plurality of microphones, the received audio signals including audio signals generated by the single speaker, based on determining that the video system is aiming at the single speaker, transmit a monophonic signal, the monophonic signal being based on the received audio signals, determine that the video system is not aiming at the single speaker, and based on the determining that the video system is not aiming at the single speaker, transmit a stereophonic signal, the stereophonic signal being based on the received audio signals.

Abstract: The disclosure relates to an audio signal processing method, device, and computer-readable medium. The method is applied to an electronic equipment that includes multiple audio acquisition devices with distances between the multiple audio acquisition devices meeting a preset distance condition.

Abstract: A method and a device for processing an audio signal in a vehicle are provided. The method includes: obtaining an audio signal by a microphone array; performing echo cancellation on the obtained audio signal, to obtain a first processed signal; and performing beamforming on the first processed signal according to sound zones in which microphones of the microphone array are located, to obtain a second processed signal, wherein the vehicle includes at least two sound zones, and each microphone of the microphone array is located in at least one sound zone. With the beamforming, the requirements for isolation degree between different sound zones is not high, and the sound source of the audio signal can be accurately determined.

Abstract: A system that performs wall detection, range estimation, and/or corner detection to determine a position of a device relative to acoustically reflective surfaces. The device generates output audio using loudspeaker(s), generates microphone audio data using a microphone array, and generates impulse response data for each of the microphones. The device may generate the impulse response data using an acoustic echo cancellation (AEC) component or multi-channel AEC (MC-AEC). The device may detect a peak in the impulse response data and determine a distance to a reflective surface based on the peak. Based on a number of reflected surfaces detected by the device, the device may classify a position of the device within the room, such as whether the device is in a corner, along one wall, or in an open area. By knowing the position relative to the room surfaces, the device may improve sound equalization and other processing.

Abstract: A device for room geometry analysis comprising: a plurality of segments (101-106) built of acoustic metamaterials, each segment (?, ?) acting as a waveguide with a unique transfer function (B(?, ?, ?)); and a processor configured to calculate delays (?(?, ?)) and respective directions (c(?, ?)) of mirror sound sources (721-725) by decomposing a sound signal (?) obtained from a microphone (110) based on the transfer functions (B(?, ?, ?)) of the segments (101-106) and based on a calibration signal (tsp(t)) emitted by a speaker (420).

Abstract: Audio signal processing techniques are described which are employed within a circuit of a speech dereverberation system. The amount of data or number of samples input to a reverberation coefficient determination unit is determined, taking into account information about the background noise in the acoustic space and information about energy of reverberant sound in the acoustic space.

Abstract: Audio systems and methods are provided that receive a playback signal and produce an acoustic signal based upon the playback signal, and include microphone signal(s) for capturing and processing user voice signals. An echo reference signal is based upon the playback signal, and an echo canceler reduces echo components from the microphone signal(s). Functionality of the echo canceler is modified, such as by freezing an adaptive filter, in response to a non-linear condition in the audio playback, or a likelihood of such a non-linear condition.

Abstract: A real-time audio signal processing system includes an audio signal processor configured to process audio signals using a modified generalized eigenvalue (GEV) beamforming technique to generate an enhanced target audio output signal. The digital signal processor includes a sub-band decomposition circuitry configured to decompose the audio signal into sub-band frames in the frequency domain and a target activity detector configured to detect whether a target audio is present in the sub-band frames. Based on information related to the sub-band frames and the determination of whether the target audio is present in the sub-band frames, the digital signal processor is configured to use the modified GEV technique to estimate the relative transfer function (RTF) of the target audio source, and generate a filter based on the estimated RTF. The filter may then be applied to the audio signals to generate the enhanced audio output signal.

Abstract: A source device can transmit initial streaming content to a playback device (e.g., wireless ear buds) using first settings and measure playback performance of the content at a plurality of times. The measured performance values can relate to a quality of communication of the initial streaming content between the source device and the playback device, e.g., relating to packet loss, retransmission rates and patterns, fluctuations in a playback (jitter) buffer, and/or other values. The measured performance values can be used to determine one or more second settings to be used for a playback of subsequent streaming content between the source device and the playback device. In this manner, each source device can account for variations in communication behavior specific to a user (e.g., due to differences in body type as electromagnetic waves travel through the body when a source device is in a pocket).

Abstract: An apparatus of reducing feedback noise in an acoustic system, the apparatus comprising: a first input for receiving a first signal derived from a first microphone associated with a first channel, the first signal comprising a first set of frequency sub-bands; a second input for receiving a second signal derived from a second microphone associated with a second channel, the second signal comprising second set of frequency sub-bands, the first and second sets of frequency sub-bands having matching frequency ranges, each frequency sub-band of the first and second sets of frequency sub-bands having a frequency of greater than a threshold frequency; and one or more processors configured to: determining feedback at a first speaker associated with the first channel; and responsive to determining feedback, mix each of the first set of frequency sub-bands with a corresponding one of the second set of frequency sub-bands to generate a mixed output signal comprising a mixed set of frequency sub-bands; wherein the mixin

Abstract: A speech signal processing method is performed at a terminal device, including: obtaining a recorded signal and a to-be-output speech signal, the recorded signal including a noise signal and an echo signal; calculating a loop transfer function according to the recorded signal and the speech signal; calculating a power spectrum of the echo signal and a power spectrum of the noise signal according to the recorded signal, the speech signal, and the loop transfer function; calculating a frequency weighted coefficient according to the two power spectra of the echo signal and the noise signal; adjusting a frequency amplitude of the speech signal based on the frequency weighted coefficient; and outputting the adjusted speech signal to a speaker electrically coupled to the terminal device. As such, the frequency amplitude of the speech signal is automatically adjusted according to the relative frequency distribution of a noise signal and the speech signal.

Abstract: Techniques for simulating a microphone array and generating synthetic audio data to analyze the microphone array geometry. This reduces the development cost of new microphone arrays by enabling an evaluation of performance metrics (False Rejection Rate (FRR), Word Error Rate (WER), etc.) without building device hardware or collecting data. To generate the synthetic audio data, the system performs acoustic modeling to determine a room impulse response associated with a prototype device (e.g., potential microphone array) in a room. The acoustic modeling is based on two parameters—a device response (information about acoustics and geometry of the prototype device) and a room response (information about acoustics and geometry of the room). The device response can be simulated based on the microphone array geometry, and the room response can be determined using a specialized microphone and a plane wave decomposition algorithm.

Abstract: An improved method and system for varying an amount of mechanical coupling in a speakerphone is disclosed. Solutions and implementations provided vary the amount of mechanical coupling between one or more speakers and one or more microphones of the speakerphone to generate high-quality sounds. Implementations include receiving an input signal, sending a copy of the input signal to a first speaker, performing a signal transformation on the input signal to produce a transformed input signal, and transmitting the transformed input signal to a second speaker, where the first speaker generates a first vibration force in response to the input signal, and the second speaker generates a second vibration force in response to the transformed input signal, the second vibration force being in an opposite direction to that of the first vibration force and offsetting at least part of the first vibration force.

Abstract: An audio system provides for spatial enhancement, crosstalk processing, and crosstalk compensation of an input audio signal. The crosstalk compensation compensates for spectral defects caused by the application of the crosstalk processing to a spatially enhanced signal. The crosstalk compensation may be performed prior to the crosstalk processing, after the crosstalk processing, or in parallel with the crosstalk processing. The crosstalk compensation includes applying filters to the mid and side components of the left and right input channels to compensate for spectral defects from crosstalk processing of the audio signal. The crosstalk processing may include crosstalk simulation or crosstalk cancellation. In some embodiments, the crosstalk compensation may be integrated with a subband spatial processing that spatially enhances the audio signal.

Abstract: Techniques are provided for pre-processing enhancement of a speech signal. A methodology implementing the techniques according to an embodiment includes performing de-reverberation processing on signals received from an array of microphones, the signals comprising speech and noise. The method also includes generating time-frequency masks (TFMs) for each of the signals. The TFMs indicate the probability that a time-frequency component of the signal associated with that TFM element includes speech. The TFM generation is based on application of a recurrent neural network to the signals. The method further includes generating steering vectors based on speech covariance matrices and noise covariance matrices. The TFMs are employed to filter speech components of the signals, for calculation of the speech covariance, and noise components of the signals for calculation of the noise covariance.

Abstract: An apparatus includes a beamformer, an echo suppression control unit, and a residual echo cancellation unit. The beamformer is configured to pass desired portions of audio signals and to suppress undesired portions of the audio signals. The beamformer includes a speech blocking filter to prevent suppression of near-end desired talker speech in the audio signals and an echo suppression filter to suppress echo in the audio signals. An echo suppression control unit is coupled to the beamformer and receives signals and determines whether to dynamically adapt the speech blocking filter or to dynamically adapt the echo suppression filter. The speech blocking filter remains unchanged during dynamic adaptation of the echo suppression filter, and the echo suppression filter remains unchanged during dynamic adaptation of the speech blocking filter. The residual echo cancellation unit is coupled to the beamformer and receives output audio signals from the beamformer and further suppresses residual echo.

Abstract: A system, article, and method of acoustic dereverberation factoring the actual non-ideal acoustic environment.

Abstract: A dereverberation device includes an input instantaneous value calculation unit configured to calculate an input instantaneous value based on an input signal; a reverberation estimation unit configured to calculate a moving average of the input instantaneous value as a reverberation component; a gain calculation unit configured to calculate, with the input instantaneous value and the reverberation component, a first gain as a basic gain for the input signal; a gain suppression control unit configured to calculate, according to a ratio between the input instantaneous value and the reverberation component, a second gain changing within a range between a predetermined lower limit and a predetermined upper limit, thereby outputting a larger one of the first gain or the second gain as a third gain; and a gain processing unit configured to multiply the input signal by the third gain.

Abstract: A method of operating a playback device includes receiving source audio content that includes a first and second channel stream of audio. The method also includes playing back, via a first and second speaker driver of the playback device, the first and second channel streams of audio, thereby producing a first and second channel audio output. A captured stream of audio is received by a microphone of the playback device, and portions of the captured stream of audio correspond to the first and second channel audio outputs. The first and second channel streams of audio are combined into a compound audio signal, and acoustic echo cancellation is performed on the compound audio signal to produce an acoustic echo cancellation output, which is then applied to the captured stream of audio to increase the signal-to noise ratio of the captured stream of audio.

Abstract: Processes and devices for equalizing an audio system that is adapted to use a loudspeaker to transduce test audio signals into test sounds. The processes and devices can involve the use of infrared signals to convey information in one or both directions between the audio system and a portable computer device that captures test sounds, calculates audio parameters that can be used in the equalization process, and transmits these audio parameters back to the audio system for its use in equalizing audio signals that are played by the audio system.

Abstract: A speech intelligibility enhancing system for enhancing speech, the system comprising: a speech input for receiving speech to be enhanced; an enhanced speech output to output the enhanced speech; and a processor configured to convert speech received from the speech input to enhanced speech to be output by the enhanced speech output, the processor being configured to: i) extract a frame of the speech received from the speech input; ii) calculate a measure of the frame importance; iii) estimate a contribution due to late reverberation to the frame power of the speech when reverbed; iv) calculate a prescribed frame power, the prescribed frame power being a function of the power of the extracted frame, the measure of the frame importance and the contribution due to late reverberation, the function being configured to decrease the ratio of the prescribed frame power to the power of the extracted frame as the contribution due to late reverberation increases above a critical value, {tilde over (l)}; and v) apply

Abstract: Disclosed methods and systems are directed to determining a best microphone pair and segmenting sound signals. The methods and systems may include receiving a collection of sound signals comprising speech from one or more audio sources (e.g., meeting participants) and/or background noise. The methods and systems may include calculating a TDOA and determining, based on the TDOA and via robust statistics, the best pair of microphones. The methods and systems may also include segmenting sound signals from multiple sources.

Abstract: A technique for controlling a loudspeaker system with an artificial neural network includes filtering, with a deconvolution filter, a measured system response of a loudspeaker and a reverberant environment in which the loudspeaker is disposed to generate a filtered response, wherein the measured system response corresponds to an audio input signal applied to the loudspeaker while the loudspeaker is disposed in the reverberant environment. The techniques further include generating, via a neural network model, an initial neural network output based on the audio input signal, comparing the initial neural network output to the filtered response to determine an error value, and generating, via the neural network model, an updated neural network output based on the audio input signal and the error value.

Abstract: Disclosed are an electronic device and a method of processing an audio signal by the electronic device. The electronic device includes: a processor functionally connected to a speaker and a microphone; and a memory electrically connected to the processor. The memory includes instructions to cause the processor, when executed, to output a first audio signal through the speaker; identify that a second audio signal detected by the microphone corresponds to the first audio signal when the first audio signal is outputted through the speaker; and control output of the first audio signal based on the identification. Further, various other embodiments may be possible.

Abstract: A method for auralizing a multi-microphone device. Path information for one or more sound paths using dimensions and room reflection coefficients of a simulated room for one of a plurality of microphones included in a multi-microphone device is determined. An array-related transfer functions (ARTFs) for the one of the plurality of microphones is retrieved. The auralized impulse response for the one of the plurality of microphones is generated based at least on the retrieved ARTFs and the determined path information.

Abstract: Described is a cognitive signal processor (CSP) for signal denoising. In operation, the CSP receives a noisy signal as a time-series of data points from a mixture of both noise and one or more desired waveform signals. The noisy signal is linearly mapped to reservoir states of a dynamical reservoir. A high-dimensional state-space representation is then generated of the noisy signal by combining the noisy signal with the reservoir states. A delay-embedded state signal is generated from the reservoir states. The reservoir states are denoised by removing noise from each reservoir state signal, resulting in a real-time denoised spectrogram of the noisy signal. A denoised waveform signal is generated combining the denoised reservoir states. Additionally, the signal denoising process is implemented in software or digital hardware by converting the state-space representation of the dynamical reservoir to a system of delay difference equations and then applying a linear basis approximation.

Abstract: Described is a system for signal denoising. The system linearly maps a noisy input signal into a high-dimensional reservoir, where the noisy input signal is a time-series of data points from a mixture of waveforms. A high-dimensional state-space representation of the mixture of waveforms is created by combining the noisy input signal with reservoir states. A delay embedded state signal is generated from the reservoir states, and a denoised spectrogram of the noisy input signal is generated.

Abstract: Acoustic echo cancellation systems and methods are provided that can cancel and suppress acoustic echo from the output of a mixer that has mixed audio signals from a plurality of acoustic sources, such as microphones. The microphones may have captured speech and sound from a remote location or far end, such as in a conferencing environment. The acoustic echo cancellation may generate an echo-cancelled mixed audio signal based on a mixed audio signal from a mixer, information gathered from the audio signal from each of the plurality of acoustic sources, and a remote audio signal. The systems and methods may be computationally efficient and resource-friendly.

Abstract: Provided are a de-reverberation control method and apparatus for a device equipped with a microphone. The method includes: reverberation parameters which indicate, at respective moments, reverberation levels of a room environment where the device is located are acquired from an audio signal played by the device; and a de-reverberation mode adopted by the device is dynamically adjusted according to the reverberation levels indicated by the reverberation parameters at different moments and preset correspondences between reverberation levels and de-reverberation modes. By adopting a dynamic de-reverberation mode, the method and the apparatus disclosed herein significantly improve the rate of the recognition of a device for the voice of the user.

Abstract: The present invention relates to a method for audio signal processing in a vehicle. In order to allow simple and reliable echo cancellation for voice recognition during simultaneous reproduction of a multichannel audio source signal in a vehicle, a mono audio signal is generated on the basis of a multichannel audio source signal. The mono audio signal is limited to a frequency range between a prescribed lower frequency and a prescribed upper frequency, for example to a range from 100 Hz to 8 kHz. The limited mono audio signal is output via multiple loudspeakers in the vehicle. An influence of the limited mono audio signal that is output via the multiple loudspeakers on a voice audio signal received in the vehicle via a microphone is compensated for by means of the limited mono audio signal in an echo canceller.

Abstract: The invention pertains to a communication system for use in a railway vehicle, comprising: a first communication network (10) and a second communication network (20), the communication networks (10, 20) using physically separate communication media; and a plurality of communication terminals (100, 200), each connected to both communication networks (10, 20). The communication system is adapted to prioritize communications from the communication terminals (100, 200) over the communication networks (10, 20) according to at least two levels of service. A first communication terminal (100) comprises a first functional module (110) and a second functional module (120), the functional module (110, 120) being functionally equivalent, the first functional module (110) being adapted to interface with the first communication network (10) and the second functional module (120) being adapted to interface with the second communication network (20).

Abstract: A method, computer program product, and computer system for addressing acoustic signal reverberation is provided. Embodiments may include receiving, at one or more microphones, a first audio signal and a reverberation audio signal. Embodiments may further include processing at least one of the first audio signal and the reverberation audio signal. Embodiments may also include limiting a model based reverberation equalizer using a temporal constraint for direct sound distortions, the model based reverberation equalizer configured to generate one or more outputs, based upon, at least in part, at least one of the first audio signal and the reverberation audio signal.

Abstract: An in-ear BLUETOOTH® headset antenna for single-ear and double-ear BLUETOOTH® headset. The antenna includes a radiation unit and a ground unit, both utilizing components that make up the headset. The radiation unit is composed of a horn of the BLUETOOTH® headset and a conductive foil attached to the horn surface. One end of the conductive foil is attached to the surface of the headset horn and the other end is connected to the feed point of the RF circuit antenna of the BLUETOOTH® headset. The ground unit includes copper pouring on a main printed circuit board connected to copper pouring a key printed circuit board by a cable. No additional antennas are required, and the in-ear BLUETOOTH® headset antenna reduces costs, saves space, and improves the radiation efficiency of the antenna due to the increased effective radiation area of the antenna.

Abstract: A method and an apparatus for tuning an FIR filter in an in-band full duplex transceiver. The method for tuning an FIR filter may include: setting attenuation of the FIR filter to be a first value and then estimating input information of the FIR filter; estimating a delta response using the estimated input information of the FIR filter; and updating the attenuation of the FIR filter to a second value using the estimated delta response.

Abstract: A method and an apparatus for tuning an FIR filter in an in-band full duplex transceiver. The method for tuning an FIR filter includes: converting an input signal of the FIR filter into a first signal that is a baseband signal; converting a signal obtained by subtracting an output signal of the FIR filter from the self-transmitted interference signal into a second signal that is the baseband signal; and calculating attenuation of the FIR filter using the first signal and the second signal.

Abstract: Described is a neuromorphic processor for signal denoising and separation. The neuromorphic processor generates delay-embedded mixture signals from an input mixture of pulses. Using a reservoir computer, the delay-embedded mixture signals are mapped to reservoir states of a dynamical reservoir having output layer weights. The output layer weights are adapted based on short-time linear prediction, and a denoised output of the mixture of input signals us generated. The denoised output is filtered through a set of adaptable finite impulse response (FIR) filters to extract a set of separated narrowband pulses.

Abstract: An acoustic processing apparatus includes a storage, an estimation unit, and a removal unit. The storage stores therein a reference signal indicating a signal obtained by completing removal of reverberation from a first observation signal included in a first processing section. The estimation unit estimates, on the basis of a model representing an observation signal as a signal obtained by adding a signal obtained by applying a reverberation removal filter to an acoustic signal that is input with a delay and the acoustic signal, a filter coefficient of the reverberation removal filter by using a second observation signal and the reference signal. The removal unit determines an output signal indicating a signal obtained by removing reverberation from the second observation signal by using the second observation signal, the reference signal, and the reverberation removal filter having the estimated filter coefficient.

Abstract: Described is a cognitive signal processor that can denoise an input signal that contains a mixture of waveforms over a large bandwidth. Delay-embedded mixture signals are generated from a mixture of input signals. The delay-embedded mixture signals are mapped with a reservoir computer to reservoir states of a dynamical reservoir having output layer weights. The output layer weights are adapted based on short-time linear prediction. Finally, a denoised output of the mixture of input signals is generated.

Abstract: Described is a cognitive signal processor for signal denoising and blind source separation. During operation, the cognitive signal processor receives a mixture signal that comprises a plurality of source signals. A denoised reservoir state signal is generated by mapping the mixture signal to a dynamic reservoir to perform signal denoising. At least one separated source signal is identified by adaptively filtering the denoised reservoir state signal.

Abstract: A method, apparatus and computer program product enhance audio quality during a voice communication session, such as by enhancing audio quality for a remote participant in a meeting. In a method and for each of two or more microphones of a first device at a first location, a target audio signal is generated that has been steered in a direction of a target audio source in order to provide at least partial isolation from a second audio source in the same environment. The method also produces a filtered audio signal based on the target audio source at least from a respective one of the two or more microphones. The method also includes mixing the filtered audio signal from at least the first device to create an audio output signal associated with an audio playback format and causing the audio output signal to be output by a second device.

Abstract: Embodiments of the present invention relate to enhancing sound through reverberation matching. In sonic implementations, a first sound recording recorded in a first environment is received. The first sound recording is decomposed to a first clean signal and a first reverb kernel. A second reverb kernel corresponding with a second sound recording recorded in a second environment is accessed, for example, based on a user indication to enhance the first sound recording to sound as though recorded in the second environment. An enhanced sound recording is generated based on the first clean signal and the second reverb kernel. The enhanced sound recording is a modification of the first sound recording to sound as though recorded in the second environment.

Abstract: A method for estimating an instantaneous phase of dereverberated acoustic signal, the method comprising the following steps: measurement of an acoustic signal reverberated by propagation in a medium, estimation of a one short-term Fourier transform of the reverberated acoustic signal with a window function, calculation of an instantaneous frequency of dereverberated signal from said short-term Fourier transform and from an influencing factor of the medium, said influencing factor being a function of a reverberation time of said medium, determination of an instantaneous phase of dereverberated signal by integrating the instantaneous frequency of dereverberated signal over time.

Abstract: A system and method for controlling noise originating from a source external to a vehicle is disclosed. The method includes determining, by an active noise controller of a vehicle, characteristics of an unwanted noise. The unwanted noise originates from a source external to the vehicle. The method also includes determining an inverted noise based on the characteristics of the unwanted noise. The method also includes projecting the inverted noise. The projected inverted noise destructively interferes with the unwanted noise. The method also includes receiving a residual noise via an error microphone. The error microphone is configured to generate a signal based on the received residual noise.

Abstract: A terminal monitors a working state parameter of a main microphone arranged on the terminal; judges whether the working state parameter satisfies a preset rule; and prompts a user that an audio input signal is abnormal or selects a main microphone satisfying the preset rule as an audio input according to a judgment result.

Abstract: A transfer function calculation unit is configured to calculate a transfer function from a sound source installed in a predetermined target direction to each microphone of a microphone array, and a determination unit is configured to determine whether or not the microphone array is normal on the basis of a difference amount between a transfer function to each microphone and a predetermined ideal transfer function to each microphone.

Abstract: Provided are methods and systems for acoustic keystroke transient cancellation/suppression for user communication devices using a semi-blind adaptive filter model. The methods and systems are designed to overcome existing problems in transient noise suppression by taking into account some less-defective signal as side information on the transients and also accounting for acoustic signal propagation, including the reverberation effects, using dynamic models. The methods and systems take advantage of a synchronous reference microphone embedded in the keyboard of the user device, and utilize an adaptive filtering approach exploiting the knowledge of this keybed microphone signal.