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.

Abstract: In one embodiment, a recording optimizer included in a recorder optimizes the fidelity of output signals. In operation, the recorder receives and records two sets of input signals—one set of input signals received via microphones included in the recorder and another set of input signals received as line-inputs from a front of house console mixer. The recording optimizer analyzes the recorded signals to identify discrepancies, such as unamplified instruments that are underrepresented in the line-inputs. The recording optimizer then performs compensation operations that adjust one or more recorded signals to mitigate the discrepancies. Subsequently, the recording optimizer combines the compensated recorded signals, generating output signals that leverage the strengths of both sets of input signals to accurately represent the experienced sound.

Abstract: An array of microphones placed on a mobile robot provides multiple channels of audio signals. A received set of audio signals is called an audio segment, which is divided into multiple frames. A phase analysis is performed on a frame of the signals from each pair of microphones. If both microphones are in an active state during the frame, a candidate angle is generated for each such pair of microphones. The result is a list of candidate angles for the frame. This list is processed to select a final candidate angle for the frame. The list of candidate angles is tracked over time to assist in the process of selecting the final candidate angle for an audio segment.

Abstract: At a microphone array, a soundfield is detected to produce a set of microphone signals each from a corresponding microphone in the microphone array. The set of microphone signals represents the soundfield. The detected soundfield is decomposed into a set of sub-soundfield signals based on the set of microphone signals. Each sub-soundfield signal is processed, such that each sub-soundfield signal is separately dereverberated to remove reverberation therefrom, to produce a set of processed sub-soundfield signals. The set of processed sub-sound field signals are mixed into a mixed output signal.

Abstract: Described is a cognitive blind source separator (CBSS). The CBSS includes a delay embedding module that receives a mixture signal (the mixture signal being a time-series of data points from one or more mixtures of source signals) and time-lags the signal to generate a delay embedded mixture signal. The delay embedded mixture signal is then linearly mapped into a reservoir to create a high-dimensional state-space representation of the mixture signal. The state-space representations are then linearly mapped to one or more output nodes in an output layer to generate pre-filtered signals. The pre-filtered signals are passed through a bank of adaptable finite impulse response (FIR) filters to generate separate source signals that collectively formed the mixture signal.

Abstract: Reducing feedback in an input signal that contains a signal component based on an output signal from a proximate output. The input signal is separated into a plurality of frequency bands by band pass filters. The power of signal in each band is determined, and the band signal with the greatest power is selected. That band's signal is sampled at a sampling rate, and at regular intervals one of the samples is selected. Blind signal separation is used to estimate signal sources from the selected samples. The estimated signals are compared to the output signal, and the estimated signal most similar to the output signal is subtracted from the input signal.

Abstract: Presented is a method and associated system for suppression of linear and nonlinear echo. The method includes dividing an input signal into several frequency bands in each of a several of time frames. The input signal may include an echo signal. The method further includes multiplying the input signal in each of the several frequency bands by a corresponding echo suppression signal. Calculating the corresponding echo suppression signal may include estimating a power of the echo signal in a particular frequency band as a sum of several component echo powers, each of the several component echo powers due to an excitation from a far-end signal in a corresponding one of the several frequency bands. Calculating the corresponding echo suppression signal may further include subtracting the power of the echo signal in the particular frequency band from a power of the input signal in the particular frequency band.

Abstract: An embodiment of the invention provides a noise cancellation method for an electronic device. The method comprises: receiving an audio signal; applying a Fast Fourier Transform operation on the audio signal to generate a sound spectrum; acquiring a first spectrum corresponding to a noise and a second spectrum corresponding to a human voice signal from the sound spectrum; estimating a center frequency according to the first spectrum and the second spectrum; and applying a high pass filtering operation to the sound spectrum according to the center frequency.

Abstract: An apparatus for computing filter coefficients for an adaptive filter is disclosed. The adaptive filter is used for filtering a microphone signal so as to suppress an echo due to a loudspeaker signal. The apparatus has: an echo decay modeling means for modeling a decay behavior of an acoustic environment and for providing a corresponding echo decay parameter; and computing means for computing the filter coefficients of the adaptive filter on the basis of the echo decay parameter. A corresponding method has: providing echo decay parameters determined by means of an echo decay modeling means; and computing the filter coefficients of the adaptive filter on the basis of the echo decay parameters.

Abstract: A system includes a CPU; an accelerator; a comparing unit that compares a first value that is based on a first processing time period elapsing until the CPU completes a first process and a second processing time period elapsing until the accelerator completes the first process, and a second value that is based on a state of use of a battery driving the CPU and the accelerator; and a selecting unit that selects any one among the CPU and the accelerator, based on a result of comparison by the comparing unit.

Abstract: The present invention provides a soundproof housing for an earset, comprising: a housing main body, coupled to the inside of a front surface case having a protrusion portion inserted into the ear, and provided with a speaker accommodation groove and a microphone accommodation groove; a speaker output hole, penetratingly formed in the speaker accommodation groove so as to communicate with the front surface case, and adjacent to the output end of the speaker; and a microphone input hole, formed in a recessed manner inside the microphone accommodation groove so as to communicate with the front surface case, and adjacent to the input end of the microphone, wherein the housing main body is protrudingly formed as a long protrusion toward the inside of the protrusion portion of the front surface case, so as to be tightly coupled with the inside of the protrusion portion of the front surface case.

Abstract: In accordance with an embodiment of the present invention, a noise/interference reduction method for speech enhancement processing includes selecting one of the microphones as a main microphone wherein the signal from the main microphone is used as a target signal, the selection of the main microphone is adaptive for mono output case, and the selection of the main microphone is fixed for stereo output case. The noise/interference component signal is estimated by subtracting voice component signal from a first microphone input signal wherein the voice component signal is evaluated as a first replica signal produced by passing a second microphone input signal through a first adaptive filter. A noise/interference reduced signal is output by subtracting a second replica signal from the target signal, wherein the second replica signal is produced by passing the estimated noise/interference component signal through a second adaptive filter.

Abstract: The present application a digital self-interference residual cancellation method that adjusts a magnitude of a sampled transmit signal based on compared magnitude and phases associated with tones. The digital self-interference residual cancellation method may follow an analog carrier cancellation stage where the digital self-interference residual cancellation is based on a determination of the channel circuit response used to control an infinite impulse response filter which can compensate using both poles and zeroes.

Abstract: An audio processing unit having a first and second input for receiving output signals of a microphone with a first and a second physically symmetrically structured microphone capsule. The audio processing unit further has a first filter unit and a first delay unit which is coupled to the first input, and a second filter unit and a second delay unit which is coupled to the second input. The audio processing unit further has an adding unit for adding signals from the first and second filter units and a control unit for influencing the filter parameters of the first and second filters and/or the delay times of the first and second delay units depending on an amplitude of an audio signal received via the first and/or second input. A directivity factor of the output signal of the microphone is controlled depending on the amplitude of the output signal of the microphone.

Abstract: Broadly speaking, the embodiments disclosed herein describe replacing a current hearing aid profile stored in a hearing aid. In one embodiment, the hearing aid profile is updated by sending a hearing aid profile update request to a hearing aid profile service, receiving the updated hearing aid profile from the hearing aid profile service, and replacing the current hearing aid profile in the hearing aid with the updated hearing aid profile.

Abstract: Methods and systems are provided for controlling bone conduction, in which a bone conduction element may be used to output acoustic signals when it is in contact with a user. A bone conduction sensor may also be made in contact with the user, and used to obtain feedback relating to the outputting of the acoustic signals via the bone conduction element. The outputting of the acoustic signals may then be adaptively controlled based on processing of the feedback. The adaptive controlling may comprise adjusting components and/or functions related to or used in the outputting of the acoustic signals. For example, the adaptive controlling may comprise adjusting gain, frequency response, and/or equalization associated with a drive amplifier driving the bone conduction element.

Abstract: A sound gathering system is disclosed herein and includes a plurality of microphones each configured to sample sound coming from a sound source. A plurality of processors are arranged in a processor chain. Each processor is coupled to at least one of the microphones and is configured to store sound samples received from the at least one microphone to a memory. A controller is terminally connected to the processor chain via a first processor. The controller is configured to calculate at least one time delay for each microphone, wherein the at least one time delay for each microphone is provided to the processor coupled thereto and is used by the processor to determine a memory position from which to begin reading sound samples.

Abstract: The invention is directed to a single channel mask estimation method capable of improving reverberant speech identification for CI users. The method is based on the energy of the reverberant signal and the residual signal computed from linear prediction (LP) analysis. The mask is estimated by comparing the energy ratio of the two signals at different frequency bins with an adaptive threshold. As the threshold is updated for each frame of speech based on the energy ratios of the reverberant and LP residual signals computed from previous frames, it is amenable for real-time implementation. It can thus be used as a specialized (for reverberant environments) sound coding strategy used for cochlear implant applications.

Abstract: An echo cancellation device includes: a full-band echo canceller that generates a pseudo-echo signal; a downsample processor that downsamples a received signal and extracts a low-band component delayed by a delay amount D1; a delay controller that delays the low-band component by a delay amount D2; a delay controller that delays an output signal of the delay controller by a delay amount D3; a low-band echo canceller that generates a pseudo-echo signal delayed by a delay amount D1+D2; and an upsample processor that upsamples the low-band pseudo-echo signal to generate a full-band pseudo-echo signal delayed by the delay amount 2D1+D2. The delay controllers control the delay amounts D2 and D3 such that a tap length LA satisfies a condition of LA?2D1+D2=D2+D3, the tap length LA indicating a response time of the adaptive filter in the full-band echo canceller.

Abstract: A method for suppressing the late reverberation of an audio signal. A plurality of prediction vectors are calculated. A plurality of observation vectors from the modulus of the complex time-frequency transform of an input signal is generated. A plurality of synthesis dictionaries from the plurality of observation vectors are constructed. A late reverberation spectrum from the plurality of synthesis dictionaries and the plurality of prediction vectors are estimated. A plurality of observation vectors are filtered to eliminate the late reverberation spectrum and obtain a dereverberated signal modulus.

Abstract: A method for echo reduction by an electronic device is described. The method includes nulling at least one speaker. The method also includes mixing a set of runtime audio signals based on a set of acoustic paths to determine a reference signal. The method also includes receiving at least one composite audio signal that is based on the set of runtime audio signals. The method further includes reducing echo in the at least one composite audio signal based on the reference signal.

Abstract: A sound processing device includes a first calculation unit configured to calculate a suppression gain of noise by using respective input signals input from a plurality of microphones; an integration unit configured to obtain an integration gain by using a suppression gain of an acoustic echo and the suppression gain of the noise; an application unit configured to apply the integration gain to one input signal among the plurality of input signals; and a second calculation unit configured to calculate the suppression gain of the acoustic echo by using signals to which the integration gain is applied, output signals that are output to a replay device, and the one input signal.

Abstract: A method and system for acoustic echo cancellation stores received far-end data in a first buffer. When the far-end data in the first buffer exceeds a predefined length, the stored far-end data is used to calculate echo estimate data. The echo estimate data is stored in a second buffer. Whenever microphone data is received the error data is calculated independent of echo estimate data availability. In particular, subsequent to sufficient echo estimate data being stored in the second buffer and responsive to the reception of the microphone data, the error data is calculated by subtracting, from the microphone data, corresponding echo estimate data stored in the second buffer.

Abstract: A sound processing device includes: a nonlinear processing unit that outputs a plurality of sound signals including sound sources existing in predetermined areas by performing a nonlinear process for a plurality of observed signals that are generated by a plurality of sound sources and are observed by a plurality of sensors; a signal selecting unit that selects a sound signal including a specific sound source from among the plurality of sound signals output by the nonlinear processing unit and the observed signal including the plurality of sound sources; and a sound separating unit that separates a sound signal including the specific sound source that is selected by the signal selecting unit from the observed signal selected by the signal selecting unit.

Abstract: In accordance with embodiments of the present disclosure, an audio processing circuit for use in an audio device may perform non-linear acoustic echo cancellation by predicting a displacement associated with an audio speaker, wherein such prediction takes into account a nonlinear response of the audio speaker with a mathematical model that calculates the predicted displacement of the audio speaker as a function of a current signal associated with the audio speaker using a time-varying difference equation, wherein coefficients of the difference equation are based on a set of physical parameters of the audio speaker. From the predicted displacement, the processing circuit may calculate a predicted acoustic output of the audio speaker, which may be used to generate a reference signal to an acoustic echo canceller.

Abstract: A desired signal is extracted with a higher accuracy from a mixed signal wherein a plurality of signals are mixed. At the time of extracting a first signal from a first mixed signal and a second mixed signal, said first mixed signal and second mixed signal having the first signal and second signal mixed therein, an estimate value of the first signal in the past is obtained as a first estimate value, and an estimate value of the second signal in the past is obtained as a second estimate value. Then, a first isolation signal is generated by subtracting the second estimate value from the first mixed signal, and a second isolation signal is generated by subtracting the first estimate value from the second mixed signal. Then, the signal generated using the first isolation signal and the second isolation signal is outputted as the first signal.

Abstract: A system and method for estimating a reverberation time is provided. The method includes estimating at least one room response of an audio capture environment with an acoustic echo canceller and generating an estimate of the reverberation time of the audio capture environment based on the at least one room response from the acoustic echo canceller.

Abstract: A method and system for acoustic echo cancellation varies a step size of an adaptive filter in an acoustic echo canceller. Far-end data is received and echo estimate data is calculated using the received far-end data. Microphone data is received and error data is calculated using the received microphone data and the echo estimate data. A first average of the microphone data and a second average of the error data are computed over a predefined number of samples. An echo leakage is estimated using the first average and the second average wherein the echo leakage indicates an extent to which the far-end data is present in the error data, and the step size of the adaptive filter is varied based on the echo leakage and a maximum allowed step size.

Abstract: An audio device may be configured to produce output audio and to capture input audio for speech recognition. In some cases, a second device may also be used to capture input audio to improve isolation of input audio with respect to the output audio. In addition, acoustic echo cancellation (AEC) may be used to remove components of output audio from input signals of the first and second devices. AEC may be implemented by an adaptive filter based on dynamically optimized filter coefficients. The filter coefficients may be analyzed to detect situations in which the first and second devices are too close to each other, and the user may then be prompted to increase the distance between the two devices.

Abstract: According to one embodiment, an apparatus for discriminating speech/non-speech of a first acoustic signal includes a weight assignment unit, a feature extraction unit, and a speech/non-speech discrimination unit. The first acoustic signal includes a user's speech and a reproduced sound. The reproduced sound is a system sound having a plurality of channels reproduced from a plurality of speakers. The weight assignment unit is configured to assign a weight to each frequency band based on the system sound. The feature extraction unit is configured to extract a feature from a second acoustic signal based on the weight of each frequency band. The second acoustic signal is the first acoustic signal in which the reproduced sound is suppressed. The speech/non-speech discrimination unit is configured to discriminate speech/non-speech of the first acoustic signal based on the feature.

Abstract: The present invention discloses an echo elimination device and method for a miniature hands-free voice communication system. The system comprises a receiver, a primary transmitter and an auxiliary transmitter, a distance from the primary transmitter to the receiver being greater than that from the auxiliary transmitter to the receiver. The device comprises an array echo elimination unit, a self-adaptive echo elimination unit and a residual echo elimination unit, which are structurally cascaded in turn.

Abstract: A controller for the conference session receives at least one audio signal to generate a speaker signal. The controller correlates the speaker signal with network timing information and generates speaker timing information. The controller transmits the correlated speaker signal and timing information to a mobile device participating in the conference session. The mobile device generates an echo cancelled microphone signal from a microphone of the mobile device, and transmits the echo cancelled signal back to the controller. The controller also receives array microphone signals associated with an array of microphones at known positions in the room. The controller removes acoustic echo from the array microphone signals, and estimates a relative location of the mobile device. The controller dynamically selects as audio output corresponding to the mobile device location either (a) the array microphone signal, or (b) the echo cancelled microphone signal from the mobile device.

Abstract: Residual echo that remains after an echo cancellation process may interfere with speech recognition. If near-end speech is detected in an audio input signal, a controller may attenuate the audio playback signal. Too much attenuation may disturb playback, whereas too little attenuation may not improve speech recognition. Accordingly, features are disclosed for attenuating an audio playback signal based at least in part on residual echo level.

Abstract: A method for mobile radio communication between a communication partner device and a mobile communication device includes conducting a communication, at least temporarily, via at least two different mobile radio connections in two different mobile radio networks; evaluating the reception quality of receive signals of each mobile radio connection in the mobile communication device; selecting the mobile radio connection with the best reception quality for the mobile radio communication, at least in a receiving direction; deriving different audio signals from the receive signals of the different mobile radio connections; and feeding the derived different audio signals to a delay circuit, which aligns transit time differences and/or the amplitude differences of the derived different audio signals with respect to one another.

Abstract: An echo-canceling unit for a simultaneous transmit and receive (STR) system includes at least three phase shifters instead of just two phase shifters in order to provide immunity to phase and/or amplitude imbalances. Each respective phase shifter is coupled to a transmit signal to generate an output signal comprising a selected phase shift with respect to the transmit signal. A weight calculation unit generates a corresponding amplitude-weight signal for the output signal of the phase shifter. A variable attenuator attenuates the output signal of each respective phase shifter based on the corresponding amplitude-weight signal to form an echo-cancelation signal component corresponding to the phase shifter. A first summer then sums the respective echo-cancelation signal components into a received signal containing an echo signal to form an echo-canceled signal. In some embodiments, an information handling system includes a receiver and a transmitter coupled to the echo-canceling unit.

Abstract: A method comprises processing M subband communication signals and N target-cancelled signals in each subband with a set of beamformer coefficients to obtain an inverse target-cancelled covariance matrix of order N in each band; using a target absence signal to obtain an initial estimate of the noise power in a beamformer output signal averaged over recent frames with target absence in each subband; multiplying the initial noise estimate with a noise correction factor to obtain a refined estimate of the power of the beamformer output noise signal component in each subband; processing the refined estimate with the magnitude of the beamformer output to obtain a postfilter gain value in each subband; processing the beamformer output signal with the postfilter gain value to obtain a postfilter output signal in each subband; and processing the postfilter output subband signals to obtain an enhanced beamformed output signal.

Abstract: Features are disclosed for measuring and correcting clock drift and propagation delay in an audio system through one or more waveforms embedded in an audio signal. A first device in communication with a speaker may be configured to obtain an audio signal and insert one or more waveforms into the audio signal. For example, the waveforms may be inserted during an interval of time. A second device in communication with a microphone may be configured to detect sound as an audio input signal. The second device may obtain a spectral representation of the audio input signal and determine a rotation based on the spectral representation at the frequency of at least one of the inserted waveforms. Clock drift may be determined based on the rotation.

Abstract: In an embodiment, a method of processing audio signals at a device includes receiving audio signals at a plurality of microphones of the device; processing at least one of the audio signals received by the plurality of microphones to generate a first characteristic; a beamformer applying beamformer coefficients to the received audio signals, thereby generating a beamformer output; processing the beamformer output to generate a second characteristic. An echo canceller is applied to the beamformer output, thereby suppressing, from the beamformer output, an echo resulting from audio signals output from an audio output. An operating parameter of the echo canceller is determined, using a relationship between the first and second characteristics.

Abstract: An echo cancelling device splits a low-band signal through LPFs having characteristics which do not cause aliasing during downsampling of downsamplers, and splits a high-band signal through HPFs having characteristics which do not cause aliasing during downsampling of downsamplers. The echo canceling device generates a mid-band signal by subtracting the low-band signal and the high-band signal from a pre-split signal by adder-subtractors, and cancels an echo on a band-by-band basis.

Abstract: A reverberation reduction device includes, a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute, calculating reverberation characteristics in response to an impulse response of a path of a sound from an audio output unit to an audio input unit by determining the impulse response from a first audio signal and a second audio signal that represents a sound that the audio input unit has picked up from the first audio signal reproduced by the audio output unit, and estimating a distance from the audio input unit to a sound source in accordance with at least one of a volume and a frequency characteristic of a third audio signal that represents a sound that the audio input unit has picked up from a sound from the sound source; correcting the reverberation characteristics so that the reverberation characteristics.

Abstract: Acoustic echoes in communications systems are distracting and undesirable. Acoustic echoes occur in communications systems where sound produced by a speaker is picked up by a microphone in a communications system. In a stereo playback environment, echo cancellation techniques become more complicated. Echo cancellation can be performed by performing echo cancellation on a center signal, which is the sum of a left channel signal and the right channel signal, or left signal and a difference signal, which is the difference of the right channel signal and the left channel signal. The adaptation rates of the two echo cancellers meet certain constraints to prevent degeneracies in the echo cancellation system.

Abstract: Provided is a signal processing method which reduces a plurality of echoes by receiving a plurality of reception signals and subtracting a pseudo echo generated by a plurality of adaptive filters which input the reception signals from a plurality of echoes generated by the reception signals. At least one of the reception signals is delayed to generate a delayed reception signal. The reception signal and the delayed reception signal are inputted to the adaptive filters to generate a pseudo echo. The frequency of inputting the reception signal and the delayed reception signal to the adaptive filters is controlled in accordance with the sensitivity of a localization change of the reception signals.

Abstract: An audio conferencing system has a base station, a speaker and one or more mobile microphones, and it operates to receive audio information from both a far-end audio source and a near-end audio source and to process the audio information for transmission to a far-end audio conferencing system. At least one of the mobile microphones has a motion detection device and operates to send detected motion information to the base station. The base station operates to control the processing of audio information it receives from the microphone according to the detected microphone motion information.

Abstract: Example apparatus and methods concern performing stereo acoustic echo cancellation using a correlation based filter adaptation control approach and without using stereo de-correlation. An embodiment includes a stereo adaptive filter that produces an echo removed microphone signal from received audio signals. The embodiment includes a mono adaptive filter that produces an echo removed microphone signal from the received audio signals. A correlation detector determines a level of correlation between the received signals and provides a signal to an adaptive filter controller. The adaptive filter controller controls how the stereo adaptive filter and the mono adaptive filter adapt audio echo cancellation as a function of the correlation between the received signals. A signal selector may select for output the signal from either the stereo adaptive filter or the mono adaptive filter based, for example, on the power level of the signals.

Abstract: A voice interaction architecture that compiles multiple audio signals captured at different locations within an environment, determines a time offset between a primary audio signal and other captured audio signals and identifies differences between the primary signal and the other signal(s). Thereafter, the architecture may provide the primary audio signal, an indication of the determined time offset(s) and the identified differences to remote computing resources for further processing. For instance, the architecture may send this information to a network-accessible distributed computing platform that performs beamforming and/or automatic speech recognition (ASR) on the received audio. The distributed computing platform may in turn determine a response to provide based upon the beamforming and/or ASR.

Abstract: A maximum-kurtosis, distortionless response (MKDR) technique and an extension, the maximum-kurtosis, Wiener estimate (MKWE) technique, are provided. In one form, blind estimates of the speech source's channel response are made from the microphone data and MVDR is applied. The source direction is estimated by finding weights that maximize output kurtosis, or the fourth central statistical moment, in the frequency domain. The MKWE approach approximates the Wiener filter by using MKDR-output noise power estimates to compute a Wiener post-filter. These approaches can be extended to block-adaptive versions if the speech source is not quickly moving in space.

Abstract: In an example, time and frequency domain speech enhancement is implemented on a platform having a programmable device, such a PC or a smartphone running an OS. Echo cancellation is done first in time domain to cancel a dominant portion of the echo. Residual echo is cancelled jointly with noise reduction during a subsequent frequency domain stage. The time domain block uses a dual band, shorter length Adaptive Filter for faster convergence. Non-linear residual echo is cancelled based on an echo estimate and an error signal from the adaptive filters. A controller locates regions that had residual echo suppressed and which do not have speech and injects comfort noise. The controller can be full-duplex and operate non-linearly. An AGC selectively amplifies the frequency bins, based on the Gain function used by the residual echo and noise canceller.

Abstract: A sound emission and collection device has a plurality of speakers (11) and a plurality of microphones. The speakers each have a sound emission surface arranged on the side surface of a case (1) so that sound can be emitted in all circumferential directions. The microphones (12) each are arranged with the sound collection direction set in the center direction of the case (1). The microphone (12) and the speaker (11) have directivities opposing to each other to minimize a wraparound sound from the speaker (11) to the microphone (12). Moreover, since the speaker (11) and the microphone (12) are arranged on circumferences of concentric circles, it is possible to obtain a compact configuration.

Abstract: Presented is a method and associated system for suppression of linear and nonlinear echo. The method includes dividing an input signal into several frequency bands in each of a several of time frames. The input signal may include an echo signal. The method further includes multiplying the input signal in each of the several frequency bands by a corresponding echo suppression signal. Calculating the corresponding echo suppression signal may include estimating a power of the echo signal in a particular frequency band as a sum of several component echo powers, each of the several component echo powers due to an excitation from a far-end signal in a corresponding one of the several frequency bands. Calculating the corresponding echo suppression signal may further include subtracting the power of the echo signal in the particular frequency band from a power of the input signal in the particular frequency band.