Abstract: A sound determination apparatus receives acoustic signals by a plurality of sound receiving units, and generates frames having a predetermined time length. The sound determination apparatus performs FFT on the acoustic signals in frame units, and converts the acoustic signals to a phase spectrum and amplitude spectrum, which are signals on a frequency axis, then calculates the difference at each frequency between the respective acoustic signals as a phase difference, and selects frequencies to be the target of processing. The sound determination apparatus calculates the percentage of frequencies at which the absolute values of the phase differences of the selected frequencies are equal to or greater than a first threshold value, and determines that the acoustic signal coming from the nearest sound source is included in the frame when the calculated percentage is equal to or less than a second threshold value.

Abstract: A unidirectional active microphone that includes a diaphragm, coil, and piezoelectric component. An electrical circuit provides amplification to the coil. The piezoelectric component produces unidirectional coupling between coil and diaphragm. The microphone is unidirectional in that the amplification is provided solely to the coil and is not transferred to the diaphragm.

Abstract: An audio system includes a passive radiator that is attached to one end of a lever arm. The other end of the lever arm is attached to a mass that serves to move out of phase of the passive radiator to cancel mechanical vibrations of the passive radiator, but without significantly affecting audio output. The lever arm is attached to a mechanical ground, which may be the enclosure on which the passive radiator is mounted. A system may use multiple lever arms to reduce rocking of the passive radiator.

Abstract: An electroacoustic transducer for converting electrical input signals into acoustic output signals includes a diaphragm adapted to move relative to a housing in response to the electrical input signals to produce the acoustic output signals, a displacement sensor, a bias voltage source, and an amplifier. The displacement sensor includes a first electrode adhered to the diaphragm and a second electrode on a first surface of the housing located proximate to the first electrode. A capacitance between the first electrode and the second electrode varies with a displacement of the diaphragm relative to the housing. The bias voltage source is coupled to at least one of the first electrode or the second electrode and provides a fixed charge to the electrode to which it is attached. The amplifier amplifies a change in voltage between the first and second electrodes to produce an output voltage between first and second signal outputs.

Abstract: An electrical apparatus to sense current through a load includes a first input terminal having a first input voltage relative to a reference, a second input terminal having a second input voltage relative to the reference, a first load terminal of the load having a first load voltage relative to the reference, a second load terminal of the load having a second load voltage relative to the reference, a first current sensing element connected between the first input terminal and the first load terminal and a second current sensing element connected between the second input terminal and the second load terminal. A first sense voltage is determined by a relationship between the first input voltage and the second load voltage and a second sense voltage is determined by a relationship between the second input voltage and the first load voltage.

Abstract: An apparatus for enhancing a multichannel audio signal comprising at least two channels configured to: estimate a value representing a direction of arrival associated with a first audio signal from at least a first channel and a second audio signal from at least a second channel of at least two channels of a multichannel audio signal; determine a scaling factor dependent on the direction of arrival associated with the first audio signal and the second audio signal; and apply the scaling factor to a parameter associated with a difference in audio signal levels between the first audio signal and the second audio sign.

Abstract: A method for controlling adaptation of a feedback suppression filter in a hearing aid comprises the step of suspending the adaptation of the feedback suppression filter when the level of a hearing aid reference signal is below a predetermined threshold. The invention further provides a hearing aid with a feedback suppression filter and with means for suspending adaptation of the feedback suppression filter.

Abstract: A signal processing device includes: a detection unit generating a digital detection signal corresponding to motion of a diaphragm of a speaker to output the digital detection signal; a gain adjustment unit generating a digital feedback signal by multiplying the outputted digital detection signal by again coefficient to output the generated digital feedback signal; a combining unit combining the outputted digital feedback signal with a digital audio signal; a storage unit storing plural gain coefficients; and a control unit performing control so that a given gain coefficient is selected from the plural gain coefficients and that the selected gain coefficient is used for the multiplication.

Abstract: An audio power management system manages operation of audio devices in an audio system. The audio power management system includes a parameter computer, a threshold comparator and a limiter. Audio signals generated with the audio system may be provided to the audio power management system. Based on a measured actual parameter of the audio signal, such as a real-time actual voltage and/or a real-time actual current, the parameter computer can derive estimated operational characteristics of audio devices, such as a loudspeaker included in the audio system. The threshold comparator may use the estimated operational characteristics to develop a threshold and manage operation of one of more devices in the audio system by monitoring the measured actual parameter, and selectively directing the limiter to adjust the audio signal, or another device in the audio system to protect or optimize performance.

Abstract: A signal processing device includes: a detection unit generating a digital detection signal corresponding to motion of a diaphragm of a speaker to output the digital detection signal; a gain adjustment unit generating a digital feedback signal by multiplying the outputted digital detection signal by again coefficient to output the generated digital feedback signal; a combining unit combining the outputted digital feedback signal with a digital audio signal; a storage unit storing plural gain coefficients; and a control unit performing control so that a given gain coefficient is selected from the plural gain coefficients and that the selected gain coefficient is used for the multiplication.

Abstract: A method, apparatus, and computer-readable storage medium for processing a sound signal is provided. The method includes receiving first reference data associated with a positional relationship between reference locations on a first device, receiving second reference data associated with a positional relationship between reference locations on a second device, receiving a reference transfer characteristic, wherein the reference transfer characteristic is based on the first and second reference data, determining, by a processor, an actual transfer characteristic based on acoustic data resulting from a test signal, and calculating, by the processor, a correction coefficient based on a difference between the reference transfer characteristic and the actual transfer characteristic.

Abstract: A listening device for processing an input sound to an output sound, includes an input transducer for converting an input sound to an electric input signal, an output transducer for converting a processed electric output signal to an output sound, a forward path being defined between the input transducer and the output transducer and including a signal processing unit for processing an input signal in a number of frequency bands and an SBS unit for performing spectral band substitution from one frequency band to another and providing an SBS-processed output signal, and an LG-estimator unit for estimating loop gain in each frequency band thereby identifying plus-bands having an estimated loop gain according to a plus-criterion and minus-bands having an estimated loop gain according to a minus-criterion.

Abstract: A method of controlling a loudspeaker output comprises deriving an admittance function over time from the voice coil voltage and current. In combination with a delta function, the force factor of the loudspeaker and the blocked electrical impedance, the input-voltage-to-excursion transfer function over time is obtained. This is used to control audio processing for the loudspeaker thereby to implement loudspeaker protection and/or acoustic signal processing; The invention provides a modelling and control approach which is not based on a parametric model. As a consequence, it does not require prior knowledge regarding the enclosure (e.g. closed or vented box) and can cope with complex designs of the enclosure.

Abstract: Method and apparatus for entrainment containment in digital filters using output phase modulation. Phase change is gradually introduced into the acoustic feedback canceller loop to avoid entrainment of the feedback canceller filter. Various embodiments employing different output phase modulation approaches are set forth and time and frequency domain examples are provided. Additional method and apparatus can be found in the specification and as provided by the attached claims and their equivalents.

Abstract: Devices and methods are disclosed that allow for selective acoustic near-field nulls for microphone arrays. One embodiment may take the form of an electronic device including a speaker and a microphone array. The microphone array may include a first microphone positioned a first distance from the speaker and a second microphone positioned a second distance from the speaker. The first and second microphones are configured to receive an acoustic signal. The microphone array further includes a complex vector filter coupled to the second microphone. The complex vector filter is applied to an output signal of the second microphone to generate an acoustic sensitivity pattern for the array that provides an acoustic null at the location of the speaker.

Abstract: Systems, methods, and other embodiments associated with selective notch filtering for howling suppression are described. According to one embodiment, an apparatus includes a howling detector that detects howling by performing a time domain analysis of speech signals to identify a speech signal that may be exhibiting howling and performing frequency domain analysis to confirm that the speech signal is exhibiting howling. The apparatus also includes a suppression selector configured to select a suppression technique, a signal processor configured to process the speech signal according to a selected suppression technique. The apparatus outputs, without suppression-related processing, speech signals that are not identified based on the time domain analysis.

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: A method and apparatus for polarity detection. The method includes applying a band-pass filter to an impulse response of a loudspeaker, applying an exponential weighting to the band-pass filtered impulse response, wherein the exponential decay parameter is related to the higher corner frequency of the band-pass filter, finding the maximum peak in a waveform of sampled impulse responses, and detecting the connection polarity of the maximum peak as the polarity of the peak.

Abstract: A method and apparatus for receiving and playing a signal in a radio receiver to suppress microphonic feedback are provided by alternately pitch shifting a received audio signal. The pitch of the received audio signal is alternately shifted up and then down, repeatedly over successive intervals of the audio signal, to produce a pitch swing signal which is then played over a speaker. The alternating pitch shifting prevents the buildup of regenerative feedback normally caused by acoustic vibrations coupling into the radio receiver.

Abstract: A method of speech enhancement in a room (10) includes the steps of capturing audio signals from a speaker's voice by a microphone (12), estimating an ambient noise level in the room from the captured audio signals, processing the captured audio signals by an audio signal processing unit (20), estimating a reverberation level, determining the gain to be applied to the captured audio signals by the audio signal processing unit according to a comparison between the estimated ambient noise level and the estimated reverberation level, and generating sound according to the processed audio signals by a loudspeaker arrangement (24) located in the room, wherein the reverberation level is the level of reverberant components of the sound generated by the loudspeaker arrangement.

Abstract: There is provided an audio amplifier circuit (200), comprising: an input, for receiving an input signal; an amplifier (205) for receiving a modified input signal and outputting an amplified signal; and a feedback loop (245) comprising digital circuitry for feeding back the amplified signal and combining a feedback signal with the input signal to generate the modified input signal. The feedback loop (245) comprises: a low-pass filter (220) for filtering the amplified signal; a comparator (225) for comparing the filtered signal with a threshold value and outputting a comparison signal; and an integrator (230) for integrating the comparison signal and outputting the feedback signal. According to another embodiment, the feedback loop comprises an integrator and a sigma-delta modulator.

Abstract: An acoustic apparatus which outputs test signals from a plurality of speakers configuring a multichannel reproduction system and adjusts acoustic characteristics by picking up response signals from the speakers using a microphone, includes a test signal storage unit storing test signals output from the speakers; a response signal storage unit storing response signals from the speakers picked up by the microphone; a parameter calculation unit calculating acoustic adjustment parameters including at least a polarity reversal parameter and a phase filter parameter based on the response signals from the speakers stored in the response signal storage unit; and an acoustic adjustment parameter storage unit storing the acoustic adjustment parameters calculated by the parameter calculation unit. The parameter calculation unit calculates the polarity reversal parameter using a low-pass component of the response signals.

Abstract: A method of modeling the frequency-dependent input-voltage-to-excursion transfer function of a loudspeaker, comprises, for a plurality of measurement frequencies, measuring a voltage and current and deriving an impedance at the measurement frequency. A frequency-dependent impedance function is derived. By additionally using the blocked electrical impedance and a force factor for the loudspeaker, a frequency-dependent input-voltage-to-excursion transfer function can be calculated. The invention provides a modeling approach which is not based on a parametric model, but computes the transfer functions for a set of frequencies separately. As a consequence, it does not require prior knowledge regarding the enclosure (e.g. closed or vented box) and can cope with complex designs of the enclosure.

Abstract: A signal processing apparatus includes: one or more detection means for detecting movement of a diaphragm of a speaker in correspondence with feedback methods that are different feedback methods; analog-to-digital conversion means for converting one or more detection signals acquired by the detection means into a digital form; feedback signal generating means for generating feedback signals corresponding to the feedback methods using the digital detection signals; synthesis means for combining an audio signal to be output as a driving signal of the speaker with the feedback signals; correction equalizer means for setting an equalizing characteristic to allow a sound reproduced by the speaker to have a target frequency characteristic by changing the digital audio signal; feedback operation setting means for setting feedback methods in which a feedback operation up to combining the audio signal with the feedback signal is performed and the feedback operation is not performed equalizing characteristic changing a

Abstract: Disclosed is a tuning sound (TS) feedback apparatus. The tuning sound (TS) feedback apparatus includes: a TS pick-up casing which is disposed and fixed inside or near the hearing organ of a sound engineer who manages tuning operations for sound output from a sound output terminal of a sound device having an equalizer; a sound/electric conversion element which is included in the TS pick-up casing, picks up the TS output from the sound output terminal of the sound device, and converts the picked-up TS into an electrical signal; and a TS output module which is included in the TS pick-up casing, converts the TS converted into the electrical signal by the sound/electric conversion element into wired or wireless communication data to be used for external communication, and transmits the data to the sound device.

Abstract: A system is provided to protect a loudspeaker (144) by controlling a level of an applied audio signal. A control signal is generated by applying an input audio signal (115) to the collective operations of a gain control system (100). The gain control system (100) uses the input audio signal (115) in conjunction with at least one parameter to derive an estimated stress associated with the loudspeaker (144). The estimated stress is compared with a protection threshold stress (127). If the protection threshold stress is exceeded, a gain applied by a gain component (134) is selectively adjusted to modify the input audio signal (115). The resulting gain-controlled audio signal (116) is employed to drive the loudspeaker (144).

Abstract: An embodiment of an apparatus for computing control information for a suppression filter for filtering a second audio signal to suppress an echo based on a first audio signal includes a computer having a value determiner for determining at least one energy-related value for a band-pass signal of at least two temporally successive data blocks of at least one signal of a group of signals. The computer further includes a mean value determiner for determining at least one mean value of the at least one determined energy-related value for the band-pass signal. The computer further includes a modifier for modifying the at least one energy-related value for the band-pass signal on the basis of the determined mean value for the band-pass signal. The computer further includes a control information computer for computing the control information for the suppression filter on the basis of the at least one modified energy-related value.

Abstract: The present invention relates to the field of signal processing technologies, and provides an audio signal playback system and an electronic device. The system includes: two or more loudspeakers, configured to separately play audio signals in different frequency bands; a filter unit, configured to filter an input audio signal, amplify the filtered audio signal, and output the amplified audio signal to the two or more loudspeakers; and a feedback control unit, configured to collect the audio signals played by the two or more loudspeakers, measure power values of the audio signals in different frequency bands, and adjust, according to an measurement result, a amplification factor of the filter unit for the filtered audio signal in a corresponding frequency band.

Abstract: A system and method for performing efficient directed sound and/or data to a user utilizing position information. Various aspects may, for example, comprise determining position associated with one or more recipients of audio signals, determining one or more audio signal parameters based, at least in part, on such determined position information, and generating audio signals based on such determined audio signal parameters. For example, direction, timing, phasing and/or magnitude of such audio signals may be adapted based on a dynamic recipient positional environment.

Abstract: A new acoustic echo suppressor and method for acoustic echo suppression is described herein. Exemplary embodiments of the acoustic echo suppressor use one linear regression model for each subband. The linear regression model for each subband may operate on the squared magnitude of the input samples as well as corresponding cross-products. In this way, accurate and robust estimates of the echo signal in each subband can be obtained, thereby providing good echo reduction while keeping the signal distortion low.

Abstract: In reverberant environments, reflected waves including an echoic sound and a muffled sound affect and disable recognition of sound arrival directions. As a result, the subjective clearness of the sounds deteriorates. In order to enhance the clearness of a reproduced sound in a reverberant environment, a pre-processing filter unit corrects an input sound signal portion having a frequency band relating to human auditory recognition on a sound wave arrival direction, and speakers reproduce the sound signal. The correction involves attenuating an input sound signal in the frequency band portion, based on the relationship between the frequencies of the input sound signal and the magnitude of influence to the recognition of the sound wave arrival direction. This attenuation is achieved by filtering using filter coefficients that are set by a first filter characteristic setting unit using hearing characteristic parameters that are set by a hearing characteristic setting unit.

Abstract: There is provided a method of controlling a noise cancellation system, the noise cancellation system being for use in a device comprising a speaker for receiving a wanted signal and generating a sound signal therefrom, and the noise cancellation system comprising: a digital filter, for generating a noise cancellation signal from an input signal representative of ambient noise; and an output for applying the noise cancellation signal to the speaker in addition to the wanted signal to generate a sound signal from which the ambient noise has been at least partially cancelled. The method comprises: determining a resonant frequency of the speaker; based on the determined resonant frequency, selecting a set of filter coefficients; and applying the selected set of filter coefficients to the digital filter.

Abstract: A method for reducing a disturbance in an input signal caused by an output signal in a multi-port connector, a multi-port connector circuit for reducing a disturbance in an input signal caused by an output signal, and a mobile device are described.

Abstract: A method for a multi microphone noise reduction in a complex noisy environment is proposed. A left and a right noise power spectral density for a left and a right noise input frame is estimated for computing a diffuse noise gain. A target speech power spectral density is extracted from the noise input frame. A directional noise gain is calculated from the target speech power spectral density and the noise power spectral density. The noisy input frame is filtered by Kalman filtering method. A Kalman based gain is generated from the Kalman filtered noisy frame and the noise power spectral density. A spectral enhancement gain is computed by combining the diffuse noise gain, the directional noise gain, and the Kalman based gain. The method reduces different combinations of diverse background noise and increases speech intelligibility, while guaranteeing to preserve the interaural cues of the target speech and directional background noises.

Abstract: A method of setting a parameter for an audio apparatus including a delay processing unit which applies a delaying process and an amplifying process on an input audio signal in accordance with the set parameter, includes: performing a measuring process including: causing an outputting device to output measurement sound; and measuring a first impulse response at a sound receiving point; performing a specifying process including: analyzing the measured first impulse response; calculating a second impulse response at the sound receiving point when the outputting device outputs sound in case where a parameter is set in the audio apparatus and an audio signal indicative of the measurement sound is input to the audio apparatus; and specifying a parameter in accordance with the calculated second impulse response; and performing a setting process of setting the specified parameter in the delay processing unit.

Abstract: Method and apparatus for entrainment containment in digital filters using output phase modulation. Phase change is gradually introduced into the acoustic feedback canceller loop to avoid entrainment of the feedback canceller filter. Various embodiments employing different output phase modulation approaches are set forth and time and frequency domain examples are provided. Additional method and apparatus can be found in the specification and as provided by the attached claims and their equivalents.

Abstract: A signal processing device includes: a sound adjustment amount calculation unit which calculates a sound adjustment amount for adjusting sound characteristics of each channel to a predetermined sound characteristic for each channel, using a sound signal that is obtained by collecting the outputs of each channel; an evaluation value calculation unit which calculates a coefficient allocation evaluation value for allocating a size of a filter coefficient necessary for the sound adjustment of the respective channels for each channel, based on the sound adjustment amount that is calculated by the sound adjustment amount calculation unit; and a filter coefficient calculation unit which calculates the filter coefficient for each channel using the coefficient allocation evaluation value that is calculated by the evaluation value calculation unit.

Abstract: A method for reducing howling in a communication system containing collocated mobile devices is presented. In a transmitter, an audio signal is received at a microphone. Acoustic feedback is removed from the audio signal and the resulting signal is encoded and transmitted either using direct or trunked mode operation to a receiver. The encoded signal is decoded at the transmitter, in addition to at the receiver, and fed back to an echo canceller with sufficient delay to account for substantially the entirety of a loop delay from encoding of the audio signal to reception of the acoustic feedback at the microphone to enable removal of the acoustic feedback. An estimate of the acoustic feedback is used to initially remove the acoustic feedback, the error being fed back to the processor to adaptively change the signal being subtracted from the audio signal to better reduce the acoustic feedback.

Abstract: A transducer device includes an acoustic transducer, a parameter extractor and a feedback circuit. The parameter extractor is configured to extract an operating parameter from the acoustic transducer. The feedback circuit is configured to generate a correction signal based on a difference between the extracted operating parameter and a corresponding reference parameter. The correction signal is applied to adjust the operating parameter of the acoustic transducer.

Abstract: An electrical apparatus includes a frame, a speaker connected to the frame, a digital signal processor in communication with the speaker to receive audio data and control data to control the speaker, the digital signal processor connected to the frame, and a lamp base coupler electrically connected to the speaker and receiver, the lamp base coupler detachably connectable to a power source, when the power source is present. A method of steering the diffused sound field includes, broadcasting at least one calibration audio signal through a plurality of speakers (M) in an audio system, receiving the calibration audio signal in a plurality of microphones spaced apart and positioned about at a listening position, and calculating respective relative speaker placement angles relative to the listening position between each of the plurality of speakers in response to receipt of the calibration audio signal in the plurality of microphones.

Abstract: [Object] To provide an induced signal removing circuit that feeds back induced voltage regarded as electrical signals into the input side of an inductive load to remove the induced voltage from the metal part, the induced voltage appearing even across an insulated metal part in response to signals input to the inductive load.

Abstract: A sound reproducing device has a loudspeaker arranged to produce sound from an audio signal provided by an audio signal source. A microphone is positioned to pick up ambient noise and generate a microphone signal which comprises the noise. An ambient noise cancellation (ANC) system receives the microphone signal from the microphone and generates anti-noise corresponding to the ambient noise in the microphone signal. An automatic polarity adaptation (AAP) system monitors the ANC system and, when a decision criterion is fulfilled, causes a switch in polarity for the generated anti-noise.

Abstract: An active noise cancellation earphone (1) has an acoustic path including a cavity (36) and a pipe (20) leading to the auditory canal (40) which are arranged to form an oscillator in use which has the effect of recovering the open loop system phase characteristics at a selected frequency or frequency range. The earphone (1) also has two parts (5,18) which can be adjusted relative to each other to allow the earphone (1) to be comfortably and correctly positioned in use.

Abstract: The present invention relates to a method and a circuit for testing a tweeter, said tweeter being part of a loudspeaker system, wherein the method includes the steps of: applying a high-frequency voltage signal to one terminal of said tweeter, said high-frequency voltage signal being generated by first electronic means; applying a constant voltage signal to the other terminal of said tweeter, said constant voltage signal being generated by second electronic means; measuring a current Iload that flows through said tweeter into said second electronic means; determining a connect/disconnect state of said tweeter from the value of said current.

Abstract: A method and apparatus for reducing resonance of a loudspeaker by using a model derived from a software simulated loudspeaker are provided. The method includes generating a model to determine resonance characteristics of the loudspeaker based on physical parameters of the audio reproducing device; determining accuracy of the model of the loudspeaker by comparing calculated frequency characteristics of the model of the loudspeaker and measured frequency characteristics of the loudspeaker that is outputting sound; generating a resonance reduction filter based on the model of the loudspeaker according to the determined accuracy; and reducing the resonance characteristics of the loudspeaker by applying the resonance reduction filter to input audio data.

Abstract: Method and apparatus for entrainment containment in digital filters using output phase modulation. Phase change is gradually introduced into the acoustic feedback canceller loop to avoid entrainment of the feedback canceller filter. Various embodiments employing different output phase modulation approaches are set forth and time and frequency domain examples are provided. Additional method and apparatus can be found in the specification and as provided by the attached claims and their equivalents.

Abstract: Embodiments of the present invention provide methods and devices for controlling a command signal applied to a load. In embodiments of the invention, current through and voltage across a load are determined and the values of both are used to generate a hybrid control signal. For example, in some embodiments the hybrid control signal is generated by taking a weighted summation of the current and voltage control signals. In other embodiments, a percentage of the difference between the current and voltage control signals is added to one of the current or voltage control signals to generate the hybrid control signal. In one embodiment, a potentiometer is used to generate the hybrid control 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: An audio system for a vehicle has at least one source of audio signals. A respective directional loudspeaker array is mounted at each seat position and coupled to the at least one source. The at least one source includes a microphone that detects speech from an occupant of the first seat position. Processing circuitry receives signals from the microphone that correspond to the detect speech and drives each second respective loudspeaker array at the other seat positions to radiate acoustic energy corresponding to the detected speech.

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.