Abstract: An engine sound processing apparatus includes an engine sound signal generating section that generates an engine sound signal indicating an engine sound generated in an engine room of a vehicle, a signal processing section that includes a reverberation effect applying portion which applies a reverberation effect to the engine sound signal and a mixing portion which mixes the engine sound signal to which the reverberation effect is not applied and the engine sound signal to which the reverberation effect is applied, a reverberation effect controlling section that instructs the reverberation effect applying portion to apply the reverberation effect to the engine sound signal, a mixing balance setting section that sets a mixing balance in the mixing portion, and a speaker that outputs a sound on the basis of the engine sound signal being output from the signal processing section.

Abstract: Techniques for multi-channel acoustic echo cancellation include adaptive filtering. An adaptive filter can use a lattice predictor of order M coupled to an adaptive LMS/Newton filter of length N, wherein M<N. The lattice predictor can provide decorrelation of the input to the LMS/Newton filter and can provide faster convergence for the LMS/Newton filter. Efficient operation of the LMS/Newton filter can also be provided by using output from the lattice predictor to provide low complexity update of weights for the LMS/Newton filter.

Abstract: A signal processing system detects reverberation. The system may suppress the reverberation and improve signal quality. The system analyzes frequency bands of an input signal to determine whether reverberation characteristics are present. When reverberation is detected, the system may attenuate the reverberant frequency band to reduce or eliminate the reverberation.

Abstract: According to one embodiment, a signal processor includes: a plurality of loudspeakers configured to reproduce sound of a plurality of channels; a plurality of microphones configured to pick up sound of a plurality of channels; a detector configured to detect a user who is present in a direction of a space from which the microphones pick up the sound, and output directional characteristic information indicating a relative direction of the user to the loudspeakers; and a signal processor configured to switch contents of processing to reduce a disturbance signal included in a picked-up sound signal of the sound picked up by the microphones from the picked-up sound signal based on the relative direction indicated by the directional characteristic information.

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: According to one embodiment, a sound corrector includes a signal outputter, a response signal, a frequency specifier, a coefficient specifier, a filter, and an outputter. The signal outputter outputs a measurement signal to measure acoustical properties of an object to be measured. The response signal receiver receives a response signal from the object in response to the measurement signal. The frequency specifier specifies a resonant frequency at a resonance peak from the response signal. The coefficient specifier specifies a correction coefficient of a correction filter for reducing the resonant frequency based on the specified resonant frequency. The filter performs filtering on a signal to be output to the object using the correction filter with the correction coefficient. The outputter outputs the signal having undergone the filtering to the object.

Abstract: Near-end equipment for a communication channel with far-end equipment. The near-end equipment includes at least one loudspeaker, at least two microphones, a beamformer, and an echo canceller. The communication channel may be in one of a number of communication states including Near-End Only state, Far-End Only state, and Double-Talk state. In one embodiment, when the echo canceller determines that the communication channel is in either the Far-End Only state or the Double-Talk state, the beamformer is configured to generate a nearfield beampattern signal that directs a null towards a loudspeaker. When the echo canceller detects the Near-End Only state, the beamformer is configured to generate a farfield beampattern signal that optimizes reception of acoustic signals from the near-end audio source. Using different beamformer processing for different communication states allows echo cancellation processing to be more successful at reducing echo in the signal transmitted to the far-end equipment.

Abstract: A linear full duplex system and method for acoustic echo cancellation is disclosed. In one embodiment, a method includes calculating a residual echo after subtraction of an echo estimate from a near end signal associated with a communication, refining a far end and a residual signal associated with the communication, updating, based on the far end signal, the echo estimate of an echo associated with the communication, subtracting the echo from the near end signal based on the echo estimate to cancel the echo associated with the communication, updating, based on the refined far end and refined residual signal, the adaptive filter module used for echo estimation, and detecting a steady state and, during the steady state, dynamically detecting internal substates and switching between the internal substates and detecting and managing double talk associated with the communication.

Abstract: An embodiment of an acoustic echo cancellation system is disclosed. The system comprises an echo cancellation unit, a second filter and a subtraction unit. The echo cancellation unit comprises a first attenuator, a first filter and a first subtractor. The first attenuator has a first down-scaling factor for attenuating a first signal. The first filter generates a first echo signal estimate based on the attenuated first signal. The first subtractor generates a third signal by subtracting the first echo signal estimate from a second signal. The second filter generates a second echo signal estimate based on the first signal. The subtraction unit subtracts the second echo signal estimate from the third signal.

Abstract: An echo suppression system and method, and a computer-readable storage medium that is configured with instructions that when executed carry out echo suppression. Each of the system and the method includes the elements of a linear echo suppressor having a reference signal path, with a nonlinearity introduced in the reference signal path to introduce energy in spectral bands. Unlike an echo canceller, the echo suppression system and method are relatively robust to errors in the introduced nonlinearity.

Abstract: A method for processing sound that includes, generating one or more noise component estimates relating to an electrical representation of the sound and generating an associated confidence measure for the one or more noise component estimates. The method further comprises processing, based on the confidence measure, the sound.

Abstract: A model application unit calculates linear prediction coefficients of a multi-step linear prediction model by using discrete acoustic signals. Then, a late reverberation predictor calculates linear prediction values obtained by substituting the linear prediction coefficients and the discrete acoustic signals into linear prediction term of the multi-step linear prediction model, as predicted late reverberations. Next, a frequency domain converter converts the discrete acoustic signals to discrete acoustic signals in the frequency domain and also converts the predicted late reverberations to predicted late reverberations in the frequency domain. A late reverberation eliminator calculates relative values between the amplitude spectra of the discrete acoustic signals expressed in the frequency domain and the amplitude spectra of the predicted late reverberations expressed in the frequency domain, and provides the relative values as predicted amplitude spectra of a dereverberation signal.

Abstract: A method and system for clear signal capture comprehend several individual aspects that address specific problems in improved ways. In addition, the method and system also comprehend a hands-free implementation that is a practical solution to a very complex problem. Individual aspects comprehended related to echo and noise reduction, and divergence control.

Abstract: Provided is a dereverberation system or the like which copes with an arbitrary condition flexibly and is capable of recognizing a sound or a sound source signal. According to the dereverberation system, an inverse filter (h) is set by using a pseudo-inverse matrix (R+) of a non-square matrix (R) as a correlation matrix of input signals (x). On the basis of the inverse filter (h) and an estimated correlation matrix (R^) generated according to a window function (w), an error cost (J(h) between a correlation value of the input signals (x) and output signals (y) and a desired correlation value (d) is calculated. On the basis of the error cost (J(h)), the inverse filter (h) is adaptively updated according to a gradient method.

Abstract: Disclosed is a noise suppression technology for suppressing various types of noise including unknown noise without storing a large number of noise information in advance. Specifically disclosed is an auxiliary device connectable to an information processing apparatus. The information processing apparatus is provided with: means for suppressing noise in a noisy signal, generating the noise information, and suppressing the noise in the noisy signal by using the generated noise information; and noise information generation means for updating the noise information on the basis of the result of suppression of the noise in the noisy signal. The auxiliary device is provided with a mechanism unit for generating noise replica and a mechanism control unit for controlling the mechanism unit so that the noise replica occurs at a timing at which the noise suppression means performs a noise suppression process.

Abstract: A communication end device of a two-way communication system is shown. The device includes an audio signal capture device for capturing local audio to be transmitted to another end device, an audio signal rendering device for playing remote audio received from the other end device, and buffers for buffering the captured and rendered audio signals. The device also includes an audio echo canceller operating to predict echo from the rendered audio signal at a calculated relative offset in the captured audio signal based on an adaptive filter, and subtract the predicted echo from the signal transmitted to the other end device The calculated relative offset that is used by the audio echo canceller for a current signal sample is adjusted if a difference between it and an adjusted relative offset of a preceding sample exceeds a threshold value.

Abstract: An oscillation-echo preventing circuit has a microphone/speaker unit and a voltage-canceling circuit for canceling voltages of audio receive signals. The microphone/speaker unit has a main body, at least two microphones, and a speaker or an earphone. The microphone seals a first inside space from an outside space. The microphone seals the first inside space from a second inside space. The speaker or the earphone seals the first inside space from the outside space. The voltage-canceling circuit cancels out the voltages of audio receive signals coming from the microphones, respectively, generating an output of minimum magnitude. Thus, the circuit can sufficiently suppress oscillation and echoing.

Abstract: A system and method for Acoustic Echo Cancellation. The system and method include a subband affine projection filter and a variable step size controller configured to cancel an estimated echo from a near-end signal. The system and method also include a divergence detector adapted to reset the subband affine projection filter in response to determining a divergence is occurring. Additionally, the system and method include a double talk detector adapted to transmit a signal to mask an output signal when double talk is detected.

Abstract: A microphone-array-based speech recognition system using a blind source separation (BBS) and a target speech extraction method in the system are provided. The speech recognition system performs an independent component analysis (ICA) to separate mixed signals input through a plurality of microphone into sound-source signals, extracts one target speech spoken for speech recognition from the separated sound-source signals by using a Gaussian mixture model (GMM) or a hidden Markov Model (HMM), and automatically recognizes a desired speech from the extracted target speech. Accordingly, it is possible to obtain a high speech recognition rate even in a noise environment.

Abstract: An audio processing apparatus includes first and second audio pickup units. The second audio pickup unit includes an audio resistor provided to cover a sound receiving portion to suppress external wind introduction while passing an external audio. A first filter attenuates a signal having a frequency lower than a first cutoff frequency of the output signal of a first A/D converter. A second filter attenuates a signal having a frequency higher than a second cutoff frequency of the output signal of a second A/D converter. A third filter is provided between the first audio pickup unit and the first A/D converter to attenuate a signal having a frequency lower than a third cutoff frequency for suppressing the wind noise.

Abstract: An autobias vehicular microphone system (300) includes a microphone (301) uses an amplifier (306) for amplifying an output of the microphone. A first feedback path (308) provides an amplifier output signal to the amplifier input for providing amplifier linearity and a second feedback path (305) is used for providing bias to an voltage reference (303). The voltage reference (303) operates to provide an autobias to the amplifier (306) based upon amplifier loading. Thus, a DC feedback loop works as an average voltage sensing circuit operating to center the amplifier (306) to an operating point near one half its supply voltage. By allowing the bias point to vary, a constant clip level can be maintained depending on varying load conditions of electronic devices (307, 309, 311) using the microphone (301).

Abstract: In one embodiment, an acoustic echo control (AEC) module receives an outgoing signal and an incoming signal, which, at various times, contains acoustic echo corresponding to the outgoing signal. The AEC module has a delay estimation block that estimates, in the time domain, the echo delay using an adaptive filtering technique. This delay estimation is used to align samples of the incoming signal having acoustic echo with the corresponding samples of the outgoing signal from which the acoustic echo originated. The AEC module determines whether or not samples of the incoming signal contain acoustic echo based on the aligned outgoing signal, and the determinations are applied to a hangover counter. The AEC module then suppresses acoustic echo in the incoming signal and adds comfort noise to the incoming signal. The amount of echo suppression performed is gradually increased or decreased based on comparisons of the counter to a hangover threshold.

Abstract: The enhancements provided herein are designed to allow an acoustic echo suppressor algorithm to operate with significant echo path delay between an audio block Rin and audio block Sin. In one illustration, a single reference sample or envelope value can be taken from the received audio block Rin. Through an algorithm, echo signals can be suppressed and/or masked from audio block Sin by delaying the previously captured reference sample and aligning the delayed reference sample with multiple samples of the audio block Sin. Instead of each individual reference sample being matched with an individual sample of the audio block Sin, a single captured and delayed reference sample or envelope value can be matched with multiple samples of audio block Sin. The algorithm presented improves voice quality on a telephone call while reducing buffer requirements.

Abstract: Processing multi-channel audio streams using one or more arrangements of single-channel components. Components that only process the near-end, or capture stream, such as noise suppression (NS) components, are limited in how they can be suitably arranged for processing multi-channel streams. However, components that process the near-end stream using one or more inputs from the far-end, or render stream, such as acoustic echo cancellation (AEC) and automatic gain control (AGC) components, are arranged in one or more of the ways suitable for use with multiple channels.

Abstract: A unified microphone pre-processing system includes a plurality of microphones arranged within a vehicle passenger compartment, a processing circuit or system configured to receive signals from one or more of the plurality of microphones, and the processing circuit configured to enhance the received signals for use by at least two of a telephony processing application, an automatic speech recognition processing application, and a noise cancellation processing application. The method includes receiving signals from one or more of a plurality of microphones arranged within a vehicle passenger compartment, and enhancing the received signals for use by at least two of a telephony processing application, an automatic speech recognition processing application, and a noise cancellation processing application. A computer readable medium containing executable instructions to cause a processor to perform a method in accordance with an embodiment of the invention is also described.

Abstract: Various embodiments of the present invention are directed to methods for dereverberation of audio generated in a room. In one aspect, a method for dereverberating reverberant digital signals comprises transforming a reverberant digital signal from the time domain into Fourier domain signals using a computing device, each Fourier domain signal corresponding to a subband. For each subband of the Fourier domain signal, the method computes autoregressive model coefficients of the reverberation with the current and previous magnitudes of the Fourier digital signal, and inverse filters the magnitude of the Fourier domain signal using the computing device, based on the autoregressive model coefficients and previous magnitudes of the Fourier digital signal. The method includes inverse transforming the Fourier domain signals with filtered magnitudes into an approximate dereverberated digital signal.

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: Harmonic distortion residual echo suppression (HDRES) technique embodiments are presented which act to suppress the residual echo remaining after a near-end microphone signal has undergone AEC, including harmonic distortion in the signal that was caused by the speaker audio signal playback. In general, an AEC module is employed which suppresses some parts of the speaker audio signal found in a near-end microphone signal and generates an AEC output signal. A HDRES module then inputs the AEC output signal and the speaker audio signal, and suppresses at least a portion of a residual part of the speaker audio signal that was left unsuppressed by the AEC module. This includes at least a portion of the harmonic distortion exhibited in the AEC output signal.

Abstract: An echo component of a first signal received at an audio input device is removed. A second signal is output from an audio output device. The echo component in the first signal is the result of the second signal traversing an echo path. The characteristics of the first and second signals are compared, and if the first signal only comprises the echo, an estimate of the echo path is determined by comparing the first and second signals. The echo path estimate is applied to the first signal to determine an equalised first signal, which is is compared with the second signal to determine an estimate of the echo component. The echo component from the first signal is removed in dependence on the estimate of the echo component.

Abstract: An echo canceller (1399) with adaptive non-linearity is disclosed. In an embodiment, an incoming signal (1301) coming in from the far end is passed to a probe signal adder, which may add a probe signal to the incoming signal and may perform other signal conditioning before passing the signal to a playback device (1304). A recording device (1310) picks up a part of the signal generated by the playback device and also picks up other sounds/physical phenomena from its environment. An echo remover creates an estimate of the signal picked up by the recording device from its environment alone without the signal generated by the playback device. The echo remover creates this estimate by using the signal going towards the playback device and the signal recorded by the recording device. A linear filter estimator (1342) generates an estimate of the linear filter section of the environment, which may be used by the echo remover.

Abstract: A transceiver includes an analog front end (AFE) device and first, second, third, and fourth transmitters. A digital signal processor (DSP) receives digital receive signal and digital transmit signals and generates a shared error control signal and first, second, third, and fourth individual error control signals. An echo canceller system generates an estimated echo signal. First, second, and third NEXT canceller systems generate first, second, and third estimated NEXT signals, respectively, each based on the second analog receive signal and respective ones of the analog transmit signals from the second, third, and fourth transmitters during the first mode and each based on the shared error control signal, respective ones of the second, third, and fourth individual error control signals, and the respective ones of the analog transmit signals from the second, third, and fourth transmitters during the second mode.

Abstract: Method, user terminal and computer program product for controlling audio signals at the user device during a communication session between the user device and a remote node, in which a primary audio signal is received at audio input means of the user device for transmission to the remote node in the communication session. It is determined whether the user device is operating in (i) a first mode in which secondary audio signals output from the user device are likely to disturb the primary audio signal received at the audio input means, or (ii) a second mode in which secondary audio signals output from the user device are not likely to disturb the primary audio signal received at the audio input means.

Abstract: A system and method for providing microphonic isolation on a transmission line. The transmission line has a first part and a second part. The first part of transmission line carries a data signal and a microphonic signal. The microphonic signal has frequencies that include those in a range of substantially 20 Hz to substantially 20 kHz. The system includes an isolation apparatus. The isolation apparatus has an input in electrical communication with a first part of the transmission line, an output in electrical communication with the second part of the transmission line, and a filter in electrical communication with the input and the output. The filter is arranged to substantially remove the microphonic signal received at the input from first part of transmission line and pass the data signal to the output.

Abstract: A sound field controlling device for supplying audio signals to a plurality of speakers provided in a space to form a sound field in the space, includes a measuring unit which measures levels of indirect sounds, which are outputted from the speakers, reflected from a wall surface of the space, and reach a listening position respectively, a reverberation applying unit which generates a reverberation simulation signal for reinforcing the indirect sounds on the basis of the audio signals, and a reverberation balance adjusting unit which controls the level of the reverberation simulation signal and supplies the controlled reverberation simulation signal to the corresponding speakers on the basis of the levels of the indirect sounds outputted from the speakers so that respective synthesized levels of the indirect sounds and the reverberation simulation signal are balanced between the speakers.

Abstract: An acoustic echo cancellation device for generating a pseudo echo signal by filtering an input remote speaker signal based on a plurality of adaptive filters and controlling the adaptive filters to filter the same based on a filter coefficient. The acoustic echo cancellation device generates an error signal by subtracting the pseudo echo signal from a nearby speaker signal, determines a convergence state of the filter coefficient based on the error signal, and sets at least one filter coefficient with a previously used value to stop the operation for calculating new values for the corresponding filter coefficients when the filter coefficient is determined to be converged.

Abstract: A system reduces noise or other external signals that may affect communication. A device converts sound from two or more microphones into an operational signal. Based on one or both signals, a beamformer generates an intermediate signal. Reflected or other undesired signals may be estimated or measured by an echo canceller. Interference may be measured or estimated by processing the echo-reduced signal or estimate by a blocking matrix. An interference canceller may reduce the interference that may modify or disrupt a signal based on the output of the blocking matrix and the intermediate signal.

Abstract: A method and apparatus for joint noise and echo cancellation of a two microphone system subject to cross-talk. The method includes estimating the reference output by removing the cross-talk and the estimated echo from the reference channel, when an echo is detected in the reference echo signal, adapting filters H13 and H23 by NLMS, when the estimated primary output includes speech, adapting filters H12 and H21 by de-correlation, when neither echo nor speech is detected, adapting filter H12 is adapted by NLMS, obtaining the primary output and the reference output by post-filtering of the estimated primary output and the estimated reference output, respectively, and utilizing the primary output and the reference output for canceling the echo and noise of a two microphone system subject to cross-talk.

Abstract: A double-talk detector finds an estimated power value of near end background noise based on a residual signal by a noise estimator; the average power of a transmitter input signal by a transmitter average power calculator; the average power of a receiver input signal by a receiver average power calculator; and an estimated echo path attenuation value through a predetermined echo path attenuation value estimating process based on the estimated power value of the near end background noise, the average power of the transmitter input signal and the average power of the receiver input signal by an attenuation value estimator. The double-talk detector detects a double-talk state based on the estimated echo path attenuation value, the average power of the transmitter input signal and the average power of the receiver input signal by a double-talk determiner to control update of the coefficient of an adaptive filter.

Abstract: A method for generating and/or performing music in real time includes receiving one or more audio signals, receiving one or more virtual instrument trigger signals, and selecting one or more plug-ins and/or one or more virtual instruments. A processing scheme is selected from a set of operations. The received audio signals and instrument trigger signals are processed in real time as a function of the selected plug-ins, virtual instruments and processing scheme, and outputted in real time as music signals.

Abstract: Certain aspects of a method and system for a dual mode subband acoustic echo canceller with integrated noise suppression may include splitting an input signal into a lowband component and a highband component. The subbands of each of the lowband component and the highband component may be processed in order to reduce an echo associated with the input signal and to suppress the noise associated with the input signal.

Abstract: A noise reduction system includes multiple transducers that generate time domain signals. A transforming device transforms the time domain signals into frequency domain signals. A signal mixing device mixes the frequency domain signals according to a mixing ratio. Frequency domain signals are rotated in phase to generate phase rotated signals. A post-processing device attenuates portions of the output based on coherency levels of the signals.

Abstract: A microphone compensation system responds to changes in the characteristics of individual microphones in an array of microphones. The microphone compensation system provides a communication system with consistent performance despite microphone aging, widely varying environmental conditions, and other factors that alter the characteristics of the microphones. Furthermore, lengthy, complex, and costly measurement and analysis phases for determining initial settings for filters in the communication system are eliminated.

Abstract: Processing requirements for acoustic echo cancellation in voice communications are significant and are even more so as the bandwidth of the communication increases. Whilst voice communication occupies a relatively narrow band of frequencies the processing requirements and so forth for wideband communication render acoustic echo cancellation difficult to achieve in a cost effective manner. The invention provides for acoustic echo cancellation within wideband communications by dividing the communications into sub-bands and applying acoustic echo cancellation to some sub-bands whilst processing other sub-bands according to the status of the communications. Additional sub-bands are transmitted at either full-duplex or half-duplex.

Abstract: A dereverberation and feedback compensation system reduces the echo received by a first audio device while reducing the speech feedback received from a second audio device. A decorrelation logic decorrelates audio signals from the first audio device. A first processor generates a noise compensation signal based on the decorrelated audio signals and system determined filter coefficients. The second processor generates an enhanced noise correlation signal based on speech signals of a second audio device and the filter coefficients used by the first processor.

Abstract: A method is provided for estimating a reverberation signal component of an acoustic signal detected by a microphone where the acoustic signal is comprised of a direct sound component and a reverberation signal component. A method for dereverberation of an acoustic signal is further provided.

Abstract: An acoustic echo cancellation device for canceling an echo in a microphone signal in response to first and second input signals includes a first combination unit for combining the first and second input signals into an aggregate input signal. The device further includes an adaptive filter unit for filtering the aggregate input signal so as to produce an aggregate echo cancellation signal. A second combination unit combines the aggregate echo cancellation signal with the microphone signal so as to produce a residual signal, and an additional filter unit filters either the first or second input signal so as to produce a first or a second partial echo cancellation signal. A post-processor unit suppresses remaining echo components in the residual signal. The post-processor unit uses at least one partial echo cancellation signal to suppress echo components corresponding with the first input signal to a greater extent than echo components corresponding with the second input signal.

Abstract: The present invention is embodied in a computer-readable program in a computer-readable medium for upgrading a video conference system, the computer-readable program comprising acoustic echo canceling control software having an application programming interface. The acoustic echo canceling control software is implemented on a computer system that operates the video conference system and macros are configured to couple the acoustic echo canceling control software to hardware components of the video conference system and to interface with the application programming interface. The macros are user configurable for providing real time adjustments of echo canceling runtime parameters of the hardware components during a video conference session.

Abstract: An acoustic echo suppression unit according to an embodiment of the present invention includes and input interface for extracting a downmix signal from an input signal, the input signal including the downmix signal and parametric side information, wherein the downmix and the parametric side information together represent a multichannel signal, a calculator for calculating filter coefficients for an adaptive filter, wherein the calculator is adapted to determine the filter coefficients based on the downmix signal and a microphone signal or a signal derived from the microphone signal, and an adaptive filter adapted to filter the microphone signal or the signal derived from the microphone signal based on the filter coefficients to suppress an echo caused by the multichannel signal in the microphone signal.

Abstract: Microphones arranged in an array shape along a longitudinal direction are respectively formed in both the longitudinal side surfaces of a housing 2 with substantially an elongated rectangular parallelepiped shape, and speakers arranged in an array shape along the longitudinal direction are formed in a lower surface. The speaker array forms sound emission beams based on sound emission directivity set according to a conference environment. On the other hand, when the microphone array forms sound collection beams by sound collection signals collected, a talker direction is detected from these beams and an output sound signal corresponding to this direction is formed and also is reflected on setting of the sound emission directivity. Also, when there are plural input sound signals, the sound emission directivity is set according to a use situation of the plural input sound signals.

Abstract: A multi-channel acoustic echo canceller arrangement comprises a microphone (111) providing a microphone signal having contributions from at least two audio sources (107, 109) to be cancelled. An echo canceling circuit (113, 115) performs echo cancellation of the two audio sources (107, 109) based on channel estimates for channels from each of the audio sources (107, 109) to the microphone (111). An estimation circuit (117) generates each of the channel estimates as a combination of a previous channel estimate and a channel estimate update where the combination includes applying a relative weight to the channel estimate update relative to the previous channel estimate. A weight processor 119 varies the relative weight in response to a time value. The arrangement may provide improved echo-cancellation for scenarios wherein the rendering of sound from the audio sources (107, 109) is time varying, such as when time varying decorrelation filters are used.