Abstract: A method includes, while operating an audio processing device in a use mode, retrieving first direction of arrival (DOA) data corresponding to a first audio output device from a memory of the audio processing device and generating a first null beam directed toward the first audio output device based on the first DOA data. The method also includes retrieving second DOA data corresponding to a second audio output device from the memory of the audio processing device and generating a second null beam directed toward the second audio output device based on the second DOA data. The first DOA data and the second DOA data are stored in the memory during operation of the audio processing device in a calibration mode.

Abstract: An amplifier circuit and a method of amplification using automatic gain control (AGC) are disclosed. A method for reducing distortions incurred by an audio signal when being rendered by an electronic device is described. The method comprises receiving an input signal; determining signal strength; determining a frequency-dependent AGC filter; wherein the frequency-dependent AGC filter is adapted to selectively attenuate the input signal within a number N of predetermined frequency ranges, according to corresponding N degrees of attenuation; wherein N predetermined frequency ranges depend upon a rendering characteristic of the electronic device; and wherein the N-degrees of attenuation depend upon the signal strength; and attenuating the input signal using the frequency-dependent AGC filter to obtain an output signal for rendering by the electronic device.

Abstract: A method and system is presented for sampling analog signals in a manner that avoids the effects of signal clipping due to a limited dynamic range. A method and device for sampling an analog input using multiple gains, or gain mask, is described. By using different gains during different time quanta, a subset of the sampled points may effectively be attenuated before being sampled and converted to a digital representation. If clipping occurs during the sampling process, the true values of the clipped samples may be interpolated using the amplitudes of the non-clipped samples, which may not have been attenuated. Such interpolation may include constructing and/or solving a constraint optimization problem using linear programming. In one embodiment, such a problem may be constructed and/or solved by using sign information from the clipped samples and/or by imposing a sparsity assumption on the signals during the reconstruction process.

Abstract: Provided is an audio coding apparatus and method that can selectively apply a operation mode of a coding module for stereo or multi-channel representation according to input signal characteristics of each channel, when voice or music signals are transmitted using an audio codec in portable terminals capable of stereo or multi-channel input and output. The audio coding apparatus includes a down-mixer for down-mixing multi-channel audio signals into mono signals; a coder for coding the mono signals; an input channel correlation analyzer for deciding whether to give them stereo effect based on their signal distribution characteristics, and outputting a control signal indicating whether to perform stereo representation process; and a stereo representation unit for performing stereo representation process onto the multi-channel audio signals when the control signal indicating to perform stereo representation process.

Abstract: Systems and methods for acoustic echo cancellation with wireless microphone and speaker systems are described herein. In one embodiment, an acoustic echo canceller is provided that can, among other things, cancel an acoustic echo component of an audio signal that is produced when an audio signal transmitted to a remote speaker on a lossy wireless link is picked up by a remote microphone and transmitted back to a base station on a lossy wireless link.

Abstract: The spirit of the present invention is to vary the step size parameter in accordance with the error signal and the output acoustic signal, wherein the filter is easy to implement, nonparametric VSS-NLMS algorithm which employs the mean-square error and the estimated system noise power to control the step-size update. The new nonparametric VSS-NLMS algorithm has been shown to perform with fast convergence rate, good tracking, and low mis-adjustment. In comparison with existing VSS-NLMS algorithms, the proposed algorithm has demonstrated consistently superior performance both in convergence and for final error level relative to published algorithms in application on both simulated data and real speech data.

Abstract: A method of post-processing raw banded gains for applying to an audio signal, an apparatus to generate banded post-processed gains, and a tangible computer-readable storage medium comprising instructions that when executed carry out the method. The raw banded gains are determined by input processing one or more input audio signals. The method includes applying post-processing to the raw banded gains to generate banded post-processed gains, generating a particular post-processed gain for a particular frequency band, including median filtering using raw gain values for frequency bands adjacent to the particular frequency band. One or more properties of the post-processing depend on classification of the one or more input audio signals.

Abstract: An earpiece (100) and acoustic management module (300) for in-ear canal echo suppression control suitable is provided. The earpiece can include an Ambient Sound Microphone (111) to capture ambient sound, an Ear Canal Receiver (125) to deliver audio content to an ear canal, an Ear Canal Microphone (123) configured to capture internal sound, and a processor (121) to generate a voice activity level (622) and suppress an echo of spoken voice in the electronic internal signal, and mix an electronic ambient signal with an electronic internal signal in a ratio dependent on the voice activity level and a background noise level to produce a mixed signal (323) that is delivered to the ear canal (131).

Abstract: A variable update step size is determined in proportion to a magnitude ratio or magnitude difference between a first residual signal and a second residual signal. The first residual signal is obtained by using adaptive filter coefficient sequence, where the adaptive filter coefficient sequence has been obtained in previous operations of the adaptive equalizer. The second residual signal is obtained by using a prior update adaptive filter coefficient sequence, where the prior update adaptive filter coefficient sequence is obtained by performing a coefficient update with an arbitrary prior update step size on the adaptive filter coefficient sequence having been obtained in previous operations of the adaptive equalizer.

Abstract: A method for processing audio signals is provided comprising outputting an audio signal; receiving the output audio signal via a first receiving path as a first received audio signal; receiving the output audio signal via a second receiving path as a second received audio signal; determining an echo suppression gain based on the first received audio signal and the second received audio signal; and filtering echo suppression of the audio signal based on the first received audio signal and the echo suppression gain.

Abstract: An audio processing device includes: a directivity adjustment unit adjusting directivity and sharpness thereof in audio picked up by plural microphones picking up audio; and a howling suppression adjustment unit adjusting intensity of suppressing howling of audio picked up by the plural microphones, wherein the directivity adjustment unit adjusts the directivity and sharpness thereof in preference to the howling suppression of audio performed by the howling suppression adjustment unit.

Abstract: Methods and systems for echo modulation are described. In one embodiment, intensities of a plurality of values in multiple windows of an audio signal may be obtained. The windows may be subject to a periodic boundary condition. A plurality of echo values may be calculated for each of the respective windows. The audio signal may be altered in one or more of the windows using a windowing function and echo values. Additional methods and systems are disclosed.

Abstract: Methods and apparatus for signal processing are disclosed. Source separation can be performed to extract source signals from mixtures of source signals and perform acoustic echo cancellation. Independent component analysis may be used to perform the source separation in conjunction with acoustic echo cancellation on the time-frequency domain mixed signals to generate at least one estimated source signal corresponding to at least one of the original source signals. It is emphasized that this abstract is provided to comply with the rules requiring an abstract that will allow a searcher or other reader to quickly ascertain the subject matter of the technical disclosure. It is submitted with the understanding that it will not be used to interpret or limit the scope or meaning of the claims.

Abstract: Methods and systems for cancelation of table noise in a speaker system used for video or audio conferencing are disclosed. Table noise is cancelled by using a vertical microphone array to distinguish the tilt angle of sound received by a microphone. If the sound is close to horizontal, the audio is muted. If the sound is above a given angle from horizontal, it is not muted, as this indicates a person speaking. This eliminates paper rustling, keyboard clicks and the like.

Abstract: A howling suppression device includes a subtractor which subtracts a pseudo feedback signal from an input signal; an adaptive filter which produces a pseudo feedback signal for a next input signal; and a coefficient update control unit which controls an update rate of a filter coefficient of the adaptive filter and includes: a level calculation unit which calculates a signal level of the input signal; a signal-rising-edge detection unit which detects a rising-edge point; a reverberation section detection unit which detects a reverberation section; and an update rate control unit which sets the update rate to a first rate in the reverberation section and to a second rate in other sections. The adaptive filter updates the filter coefficient at the update rate set by the update rate control unit.

Abstract: A communication system having echo-cancelling mechanism is provided. The communication system communicates with a remote communication device. The communication system comprises a computer system end, a digital playback device, a sound-receiving module, a digital-to-analog (D/A) converter and a digital processing module. The computer system end receives a digital signal from the remote communication device. The digital playback device plays the digital signal to generate a sound signal. The sound-receiving module generates an analog audio signal comprising the sound signal. The D/A converter receives a digital sound signal of the digital signal and converts the digital sound signal to an analog sound signal. The digital processing module performs an echo-cancelling process on the analog audio signal according to the analog sound signal.

Abstract: A method of increasing the intelligibility of an audio broadcast in an at least partially enclosed space from at least one amplified audio source. An input microphone receives an incident audio wavefront at a first position in the at least partially enclosed space. An active noise control system is employed to generate a cancelling audio wavefront having a magnitude substantially equal to the magnitude of incident audio wavefront and a phase substantially opposite to the phase of the incident audio wavefront. The cancelling audio wavefront is broadcast at a second position in the at least partially enclosed space adjacent to a reflective surface of the at least partially enclosed space so as to attenuate the incident audio wavefront substantially at or near the reflective surface in order to reduce reverberations of the incident audio wavefront. In this manner, reverberations which could reduce the intelligibility of the audio broadcast to an audience is reduced.

Abstract: Technologies are generally described for a system for controlling an audio signal based on a proximity of, e.g., a user's hand to at least one component of an audio control system. In some examples, an audio control system may include a filter configured to provide an echo signal and a control decision unit configured to provide a control signal based on the echo signal.

Abstract: An apparatus includes first and second microphone arrangements, arranged to output first and second signals respectively and is operable in a first mode and a second mode. In the first mode, an output signal is generated based on the second signal and a third signal, where the second signal and, optionally, the first signal, can be used to compensate for ambient noise, for example, for noise cancellation when a telephone call is relayed through a speaker. In the second mode, an output signal is generated based on the first and second signals. In this manner, the combination of the first and second microphone arrangements provides a directional sensitivity that can pick up sound from a remote source, for example, in an audio or video recording session. The apparatus may include a sensor to allow automatic switching between one or more of modes, directional sensitivity patterns and types of recording session.

Abstract: A loudspeaker drive circuit has a microphone which forms part of an acoustic echo cancellation system. An input signal is processed before application to a loudspeaker driver, and the processing is controlled in dependence on the echo cancellation system performance, such as to control the extent to which the loudspeaker is driven into a non-linear operating region. In this way, the linearity can be controlled so as to provide an excursion limit, without needing a model of the loudspeaker or additional dedicated sensors.

Abstract: An ambient information notification apparatus includes: an interior sound acquisition device acquiring an interior sound in a compartment of the vehicle; an ambient information presentation sound generator generating an ambient information presentation sound; and an ambient information presentation sound output device outputting the ambient information presentation sound. The ambient information presentation sound satisfies that a sound pressure level of the ambient information presentation sound is higher than the interior sound in a predetermined frequency band, and is lower than or equal to the interior sound in other frequency band, and that the ambient information presentation sound is provided by stereophony, in which a sound image localization direction approximately directs to a virtual sound source.

Abstract: An echo canceler includes a signal-to-echo ratio calculating unit 103 and a residual echo suppressing unit 105. The signal-to-echo ratio calculating unit 103 computes a signal-to-echo ratio SE(n) indicating a ratio of an echo component to a received signal x(n) from a first residual signal and a second residual signal. The first residual signal is obtained using a filter coefficient sequence of an update filter 102, which is obtained up to the previous operation. The second residual signal is obtained using an updated filter coefficient sequence that undergoes a coefficient update which is performed, using an arbitrary update step size ?(n), on the filter coefficient sequence of the update filter 102, which is obtained up to the previous operation. The residual echo suppressing unit 105 suppresses the echo component contained in the microphone input signal in accordance with the signal-to-echo ratio the signal-to-echo ratio calculating unit 103 computes.

Abstract: A method and arrangement for cancelling echoes in a communication terminal (400) during a voice call with an opposite party. The communication terminal (400) comprises an echo cancelling unit (416,600) adapted to produce an estimate of echo information in received speech signals coming from a microphone in the communication terminal (400), and subtract the estimate from the received microphone signals, before being transmitted to the opposite party. The estimate is produced based on down-sampled received microphone signals and down-sampled received speech signals from the opposite party. The down-sampled speech signals from the opposite party are filtered in a digital filter (606), and the output signals from the filter (606) are up-sampled to form the estimate.

Abstract: A robust noise reduction system may concurrently reduce noise and echo components in an acoustic signal while limiting the level of speech distortion. The system may receive acoustic signals from two or more microphones in a close-talk, hand-held or other configuration. The received acoustic signals are transformed to frequency domain sub-band signals and echo and noise components may be subtracted from the sub-band signals. Features in the acoustic sub-band signals are identified and used to generate a multiplicative mask. The multiplicative mask is applied to the noise subtracted sub-band signals and the sub-band signals are reconstructed in the time domain.

Abstract: A method of decoding an audio signal is disclosed, The present invention includes the steps of receiving the audio signal having a plurality of channel signals including an ambient component signal and a source component signal, extracting the ambient component signal and the source component signal of each of the channels based on correlation between the channel signals, modifying the ambient component signal using surround effect information, and generating the audio signal including a plurality of channels using the modified ambient component signal and the source component signal. Accordingly, in an apparatus for decoding an audio signal and method thereof according to the present invention, an ambient component signal is extracted and modified based on correlation and the modified ambient and source component signals are outputted using different signal output units, respectively. Therefore, the present invention enhances a stereo effect of the audio signal.

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: The invention is directed to echo cancellation for a microphone system. An exemplary microphone system comprises a first transistor, wherein a gate terminal of the first transistor is connected to a ground terminal via a microphone electret element, the microphone electret element being associated with a capacitance and a voltage, the microphone electret element reverse biasing the first transistor; and a second transistor in parallel with the first transistor, wherein a gate terminal of the second transistor is connected to the ground terminal via a capacitor, the capacitance of the capacitor being selected to suppress at least a portion of a common mode signal, and wherein the gate terminal of the second transistor is not connected to the microphone electret element. The common mode signal comprises the echo, which may be the output of a speaker system that is received as input to the microphone system.

Abstract: A method, system, and computer program for enhancing a signal are presented. The signal is received, and energy estimates of the signal may be determined. At least one characteristic of the signal may be inferred based on the energy estimates. A mask may be generated based, in part, on the at least one characteristic. In turn, the mask may be applied to the signal to produce an enhanced signal, which may be outputted.

Abstract: A system that utilizes closed-form solutions to perform echo cancellation is described. The system includes a filter, filter parameter determination logic and a combiner. The filter is configured to process a far-end audio signal in accordance with one or more filter parameters to generate an estimated echo signal. The filter parameter determination logic is configured to update estimated statistics associated with the far-end audio signal and a microphone signal based on instantaneous statistics associated with the far-end audio signal and the microphone signal, and calculate the one or more filter parameters based upon the updated estimated statistics. The combiner is configured to generate an estimated near-end audio signal by subtracting the estimated echo signal from the microphone signal.

Abstract: A two-stage structure for performing non-linear echo cancellation is described in which a first echo canceller is used to attenuate linear echo components of a microphone signal and a second echo canceller is used to attenuate non-linear echo components of the output signal generated by the first echo canceller. One or both of the echo cancellers may be implemented using closed-form solutions, including a closed form solution for a hybrid method in the frequency domain.

Abstract: Different sampling rates between a playout unit and a capture unit are compensated for via a system, method and computer program product. The playout unit receives samples from a computational unit, and the capture unit sends samples to the computational unit. A playout FIFO buffer operates in a playout time domain, and a capture FIFO buffer operates in a capture time domain. The computational unit is synchronized to a common clock. A first relationship is calculated between the common clock and a playout fifo buffer read pointer, and a second relationship is calculated between the common clock and a capture FIFO buffer write pointer. For each sample in the playout time domain a corresponding sample in the samples from said computational unit is found and sent to the playout FIFO buffer. For each sample in the common clock time domain the corresponding sample in the capture time domain is found and sent to the computational unit.

Abstract: A conference bridge of a conference system having a plurality of conference stations, the bridge comprising: reception means for receiving dual-channel audio signals coming from the conference station; determination means for determining at least one processing function per conference station of the conference system; application means for applying respective weighing functions to the received dual-channel signals; build-up means for building up one hybrid dual-channel signal for forwarding per conference station by means for summing a portion of the process dual-channel audio signals; and forwarding means for forwarding the respective hybrid dual-channel audio signal to each of the various conference stations of the conference system. A method of forwarding a spatialized audio scene and implemented by a conference bridge is also disclosed.

Abstract: Embodiments of the present invention exploit redundancy of succeeding FFT spectra and use this redundancy for computing interpolated temporal supporting points. An analysis filter bank converts overlapped sequences of an audio (ex. loudspeaker) signal from a time domain to a frequency domain to obtain a time series of short-time loudspeaker spectra. An interpolator temporally interpolates this time series. The interpolation is fed to an echo canceller, which computes an estimated echo spectrum. A microphone analysis filter bank converts overlapped sequences of an audio microphone signal from the time domain to the frequency domain to obtain a time series of short-time microphone spectra. The estimated echo spectrum is subtracted from the microphone spectrum. Further signal enhancement (filtration) may be applied. A synthesis filter bank converts the filtered microphone spectra to the time domain to generate an echo compensated audio microphone signal.

Abstract: A method of enhancing an audio signal includes the steps of: a) receiving a primary audio input signal, b) receiving a detected audio signal which comprises: A) an echo component derived from play-out of the primary audio input signal and B) a noise component, and c) estimating from the primary audio input signal and the detected audio signal: 1) a set of frequency-specific lower bound gains, such that each frequency-specific lower bound gain, when applied to a respective frequency of the primary audio input signal, would cause the noise component to just mask the echo component at that respective frequency and 2) a set of frequency-specific upper bound gains, such that each frequency-specific upper bound gain, when applied to a respective frequency of the primary audio input signal, would cause the echo component to just mask the noise component at that respective frequency; d) estimating a set of frequency-specific gains in such a way that each frequency-specific gain falls between the respective frequency-

Abstract: A system of signal processing an input signal in a hearing aid to avoid entrainment, the hearing aid including a receiver and a microphone, the method comprising using a transform domain adaptive filter including two or more eigenvalues to measure an acoustic feedback path from the receiver to the microphone, analyzing a measure of eigenvalue spread against a predetermined threshold for indication of entrainment of the transform domain adaptive feedback cancellation filter, and upon indication of entrainment of the transform domain adaptive feedback cancellation filter, modulating the adaptation of the transform domain adaptive feedback cancellation filter.

Abstract: According to one embodiment, an acoustic correction apparatus includes an input module, a calculator, a divider, a converter, an extractor, a synthesizer, and a generator. The input module receives an audio signal propagated through a sound field. The calculator calculates an impulse response from the audio signal. The divider divides the impulse response into first and second impulse responses. The converter converts the first and second impulse responses into first and second frequency spectrums. The extractor specifies an amplitude component of the first frequency spectrum with a peak relatively higher than that of the amplitude component of the first frequency spectrum, and extracts the peak as a resonance component. The synthesizer synthesizes a first property and a second property for attenuating the resonance component. The generator generates a correction filter for performing correction to obtain the synthesized property.

Abstract: A method is provided which is suitable to cope with non-linear echo paths during acoustic echo cancellation in speakerphones. Non-linear paths occur particularly in hands-free operation of, e.g., a mobile phone, due to driving the amplifier and loudspeaker in the non-linear range. The idea is to combine the commonly known one microphone approach of linear acoustic echo cancellation using an adaptive filter and a post-processor together with a multiple microphone approach using beam forming which separately removes the non-linear part of the echo.

Abstract: A method of echo cancellation, particularly for use in high definition applications, splits an input signal and reference signal into M single-sided sub-band. The subbanded signals are downsampled at a downsampling rate N, where N?M, adaptively filtered, and recombined to produce an output signal. The sub-bands are preferably oversampled such that N<M. The use of oversampling and single-sided sub-banding reduces complexity and avoids aliasing problems.

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: An audio processing device includes a reverb property estimating unit that estimates a reverb property at each frequency on the basis of a first audio signal and a second audio signal representing sounds of the first audio signal output by an audio output unit and collected by an audio input unit, a gain calculating unit that determines an attenuating ratio for a component of the first audio signal at each frequency such that the larger the reverb property at the frequency is, the larger the attenuating ratio for the component at the frequency becomes, and a correcting unit that attenuates the first audio signal at the each frequency in accordance with the attenuating ratio determined for each frequency.

Abstract: An acoustic processing apparatus is provided. The apparatus includes a pre-processing component, a filter and a first signal processing component. The pre-processing component compensates a non-linearity of a reference signal to generate an input signal. The filter coupled to the pre-processing component, the filter executes filtering on the input signal to generate an output signal. The first signal processing component, coupled to the pre-processing component, the reference signal obtains a gain from the first signal processing component to generate a first signal, and the first signal processing component passes the gain to the pre-processing component.

Abstract: Various embodiments of the present invention are directed to adaptive methods for reducing acoustic echoes in multichannel audio communication systems. Acoustic echo cancellation methods determine approximate impulse responses characterizing each echo path between loudspeakers and microphones within a room and improve performance based on previously determined impulse responses. In particular, the methods adapt to changes in the room by inferring approximate impulse responses that lie within a model of an impulse response space. Over time the method improves performance by evolving the model into a more accurate space from which to select subsequent approximate impulse responses.

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 howling suppression device that can reduce quality deterioration of processed sound includes: a delay unit delaying the input signal to output the delayed input signal as a reference signal; a signal separation unit including an adaptive filter extracting a periodic signal component from the reference signal by adaptively updating a filter coefficient; a howling detection unit detecting an occurrence of howling using at least a signal of the periodic signal component output from the adaptive filter; and a howling suppression unit. The howling suppression unit includes: a suppression filter obtaining the updated filter coefficient from the adaptive filter with timing when the howling detection unit detects the occurrence of the howling, to extract the periodic signal component from the reference signal based on the filter coefficient; and a subtractor subtracting the periodic signal component from the input signal so as to output a signal obtained by the subtraction.

Abstract: Systems and methods for envelope-based acoustic echo cancellation in a communication device are provided. In exemplary embodiments, a primary acoustic signal is received via a microphone of the communication device, and a far-end signal is received via a receiver. Frequency analysis is performed on the primary acoustic signal and the far-end acoustic signal to obtain frequency sub-bands. An echo mask based on magnitude envelopes of the primary and far-end acoustic signals for each frequency sub-band is generated. A noise mask based on at least the primary acoustic signal for each frequency sub-band may also be generated. A combination of the echo mask and noise mask may then be applied to the primary acoustic signal to generate a masked signal. The masked signal is then output.

Abstract: A wideband acoustic echo cancellation apparatus with an adaptive tail length in an embedded system, and a wideband acoustic echo cancellation method are provided, and the wideband acoustic echo cancellation apparatus may include a delay length calculating unit to calculate a delay length of an echo path, using a near-end signal and a far-end signal, an adaptive filter implementing unit to implement an adaptive filter based on the calculated delay length, using selected coefficients, and an error calculating unit to search for three intervals having a largest impulse response value from all intervals of a tail of the adaptive filter, and to calculate an error during an interval in which the selected coefficients are used.

Abstract: Mobile communication devices, having multiple speakers and/or microphones to perform a number of audio functions, for use with mobile devices, are provided. The microphones may be housed within the communication device housing. To compensate for the unwanted signal feedback between the speakers and microphones, acoustic echo cancellation may be implemented to determine the proper distance and relative location between the speakers and microphones. Acoustic echo cancellation removes the echo from voice communications to improve the quality of the sound. The removal of the unwanted signals captured by the microphones may be accomplished by characterizing the audio signal paths from the speakers to the microphones (speaker-to-microphone path distance profile), including the distance and relative location between the speakers and microphones. The optimal distance and relative location between the speakers and microphones is provided to the user to optimize performance.

Abstract: An apparatus includes a non-adaptive filter, an adaptive filter, and a controller. The non-adaptive filter may have non-adaptive filter coefficients and be configured to develop a non-adaptive error signal as a function of the non-adaptive filter coefficients. The adaptive filter may have adaptive filter coefficients and be configured to develop an adaptive error signal as a function of the adaptive filter coefficients. The controller may be configured to monitor a quality of the non-adaptive and adaptive error signals and perform one or more of a full coefficient update, a partial coefficient update and a fractional coefficient update of the non-adaptive filter coefficients based on a comparison of the quality of the adaptive error signal to a determined current best-attained performance measurement.

Abstract: A sound source model storage section stores a sound source model that represents an audio signal emitted from a sound source in the form of a probability density function. An observation signal, which is obtained by collecting the audio signal, is converted into a plurality of frequency-specific observation signals each corresponding to one of a plurality of frequency bands. Then, a dereverberation filter corresponding to each frequency band is estimated by using the frequency-specific observation signal for the frequency band on the basis of the sound source model and a reverberation model that represents a relationship for each frequency band among the audio signal, the observation signal and the dereverberation filter. A frequency-specific target signal corresponding to each frequency band is determined by applying the dereverberation filter for the frequency band to the frequency-specific observation signal for the frequency band, and the resulting frequency-specific target signals are integrated.

Abstract: A filter coefficient setting device for setting a filter coefficient of an echo prevention device including a first FIR filter, and a second FIR filter, comprises: a filter coefficient initial setting portion configured to set a predetermined filter coefficient for the first and second FIR filters when the echo prevention device is started.