Abstract: An echo canceller is proposed that has an echo path change detector (10) capable of reliably distinguishing an echo path change from the condition of double talk. The echo path change detector is adapted to sample filter parameters from an FIR adaptive filter in the echo canceller, and to detect a change in intensity in a pattern formed by the sampled filter parameters. An echo path change is signalled when the degree of said pattern intensity change exceeds a predetermined level. The invention resides in the understanding that the parameters or coefficients of the adaptive FIR filter react differently to a condition of double talk or echo path change. Specifically, when these parameters are sampled and represented as an intensity pattern, a significant change in this intensity pattern over time can indicate an echo path change.

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: An earpiece (100) and a method (300) personalized voice operable control can include capturing (302) an ambient sound from an Ambient Sound Microphone (111) to produce an electronic ambient signal (426), delivering (304) audio content (402) to an ear canal (131) by way of an Ear Canal Receiver (125) to produce an acoustic audio content (404), capturing (306) in the ear canal an internal sound (402) from an Ear Canal Microphone (123) to produce an electronic internal signal (410), wherein the electronic internal signal includes an echo of the acoustic audio content and a spoken voice generated by a wearer of the earpiece, detecting (312) the spoken voice in the electronic internal signal in the presence of the echo, and controlling (314) a voice operation of the earpiece when the spoken voice is detected.

Abstract: An apparatus is provided for suppressing an echo signal included in a measured signal corresponding to a measured sound. In the apparatus, the measured signal and a reference signal in a time domain are transformed into a frequency domain, and calculated for obtaining each value of a ratio and a correlation between the measured signal and the reference signal in the frequency domain. With executing a comparison of the values of the ratio and the correlation, a coefficient is derived, where a product of the coefficient and the measured sound in the frequency domain gives an estimated value of the echo signal. The echo in the measured signal is suppressed with subtracting the estimation of the echo signal from the measured signal, respectively in the frequency domain.

Abstract: A system (1) is described which is suited for suppressing audio distortion. The system comprises echo cancelling means g1, g2 coupled between an audio output (4) and a distorted desired audio sensing microphone array (3), and a filter arrangement (7) coupled to the echo cancelling means g1, g2 and/or the microphone array (3). The filter arrangement (7) includes filter coefficients representing at least a part of the audio distortion, such as reverberation. The system also comprises an at least partly mirrored circuit arrangement g?, 7? for copying thereby simulated audio distortion representative filter coefficient values into the filter coefficients of said filter arrangement (7). Such copied values can be then used for suppressing reverberation in a distorted desired signal by the filter arrangement (7).

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: 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: An acoustic echo cancellation device (1) for canceling an echo (e) in a microphone signal (z) in response to a first input signal (s) and a second input signal (m) comprises: —a first combination unit (13) arranged for combining the first input signal (s) with the second input signal (m) into an aggregate input signal (x), —an adaptive filter unit (11) arranged for filtering the aggregate input signal (x) so as to produce an aggregate echo cancellation signal (y), —a second combination unit (14) arranged for combining the aggregate echo cancellation signal (y) with the microphone signal (z) so as to produce a residual signal (r), —an additional filter unit (12) arranged for filtering either the first input signal (s) or the second input signal (m) so as to produce a first partial echo cancellation signal (ys) or a second partial echo cancellation signal (ym) respectively, and—a post-processor unit (15) arranged for suppressing remaining echo components in the residual signal (r).

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: A method of detecting double talk condition in hands free communication devices is disclosed. In general, the method in accordance with the teachings of this invention detects double talk conditions based on inherent frequency response differences between the transducers used and acoustical effect on the spectrum of the returned echo signal. An input signal from a far-end talker and an input signal from the output from an echo canceler are received by the detector. K spectral subbands are created for each input signal. From this K subbands q subbands are selected based on inherent frequency differences between the far-end transducer and a near-end transducer. The spectral echo residual power is estimated at each subband. The estimated spectral echo power and the output signal from the echo canceler for a selected subband are compared to a predetermined threshold. Based on this comparison, it is determined whether double talk conditions exist based on the comparison.

Abstract: A method of estimating the reverberations in an acoustic signal (y) comprises the steps of determining the frequency spectrum (Y) of the signal (y), providing a first parameter (?) indicative of the decay of the reverberations part (r) of the signal over time, and providing a second parameter (?) indicative of the amplitude of the direct part (d) of the signal relative to the reverberations part (r). An estimated frequency spectrum ({hacek over (R)}) of the reverberations signal (r) is produced using the frequency spectrum (Y) of a previous frame, the first parameter (?), and the second parameter (?). The second parameter (?). The second parameter (?) is preferably inversely proportional to the early-to-late ratio of the signal (y).

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 system used in a loudspeaker-room-microphone environment includes a microphone signal partitioner that divides a signal from a microphone into one or more divided portions. A reverberation energy estimator estimates reverberation energy in some of the divided portions of the microphone signal based on a loudspeaker signal. The estimated reverberation energy is processed to generate a dereverberated output signal.

Abstract: An echo detector includes means (34) for forming a set of distance measures between pitch estimates of a first signal and pitch estimates of a second signal at predetermined delays with respect to the first signal. A selector (36) selects a distance measure from the set corresponding to the highest similarity between the first and second signals.

Abstract: Various embodiments of the present invention are directed to adaptive real-time, acoustic echo cancellation methods and systems. One method embodiment of the present invention is directed to reducing acoustic echoes in microphone-digital signals transmitted from a first location to a second location. The first location includes a plurality of loudspeakers and microphones, each microphone produces one of the microphone-digital signals including sounds produced at the first location and acoustic echoes produced by the loudspeakers. The method includes determining approximate impulse responses, each of which corresponds to an echo path between the microphones and the loudspeakers. The method includes determining a plurality of approximate acoustic echoes, each approximate acoustic echo corresponds to convolving a digital signal played by one of the loudspeakers with a number of the approximate impulse responses.

Abstract: An echo canceller used for hands-free communication systems in which hands-free communication is performed by using a speaker and a microphone is disclosed. The echo canceller includes a step size control unit calculating a step size value in an adaptive filter and an adaptive filter unit estimating an echo component of a feedback path from an input signal to the feedback path by adaptively identifying an impulse response of the feedback path formed by an acoustical coupling and the like of the speaker and the microphone, and subtracting the echo component from an output signal from the feedback path, in which the step size control unit calculates a step size value by using an echo reduction amount defined based on the ratio between the output signal from the feedback path and a residual signal and outputs the value to the adaptive filter unit.

Abstract: An electricity echo cancellation device applied at a terminal includes: an input buffer memory module, a network echo delay computation module and an adaptive filtering module. The adaptive filtering module includes an adaptive filter, a subtracter and a dual-ended voice detection module. An electricity echo cancellation method includes: calculating a network echo delay according to relevant information of an RTCP packet transmitted from the network; and dynamically adjusting a terminal input signal to be adaptively filtered according to the network echo delay. The present invention ensures the electricity echo cancellation effect at the final user end on the whole, and improves the effectiveness of electricity echo cancellation. Meanwhile method of the present invention can be realized with software, thus avoiding influences of hardware memory restricts on the echo cancellation effect. In addition, the present invention only needs a single-point deployment, and thus the cost is saved.

Abstract: According to one embodiment, a signal processing apparatus includes a speaker configured to output the received input signal on which a delay detection signal which has a frequency component of an inaudible frequency on a received input signal is superposed to an acoustic space, an extracting section configured to extract the delay detection signal from the sending input signal outputted from microphone configured to collect sound in the acoustic space a calculating section configured to calculate a delay time between the received input signal and an acoustic echo component contained in the sending input signal, a delay section configured to delay the received input signal by a time corresponding to the delay time and generate a delayed received input signal, and an echo suppression processing section configured to suppress the acoustic echo component contained in the sending input signal by use of the delayed received input signal.

Abstract: An audio signal is processed by transforming the signal into a frequency domain representation having a plurality of frequency subbands. A decorrelated signal is derived from the frequency domain representation using a phase rotation.

Abstract: An oscillation/echo canceller system (1) comprising a hollow main body (5) having an insert section (4) provided with an opening (3) and adapted to be inserted into the canal of the ear, a microphone (10) and a speaker (11), the microphone (10) and the speaker (11) being arranged in the main body. The speaker is arranged with its sound emitting hole directed to the opening of the insert section and provided with an even number of sound emitting canals (12) formed between the sound emitting hole and the opening and having identical lengths and identical inner diameters, the even number being equal to two or even number times of two. The microphone is arranged more remotely from the opening of the insert section than the sound emitting hole of the speaker and the microphone has a sound collecting canal (15) for collecting sounds from the opening that is made of a material incapable of directly collecting sounds from the sound emitting canals of the speaker.

Abstract: A telephony system equipped with an echo cancellation module is disclosed. A hybrid interface circuit outputs an outbound signal to a wall jack while receiving an inbound signal from the wall jack. The inbound signal may contain a line echo of the outbound signal caused by the impedance mismatches in the hybrid interface circuit. A line echo canceller containing an adaptive filter is used to cancel the line echo in the inbound signal based on the learned line echo path characteristics captured in the training period. During a brief period after the telephony system is activated, the line echo canceller enters into a calibration mode wherein its adaptive filter is trained to learn the line echo path characteristics. During the calibration mode, the line echo canceller generates a calibration signal as the outbound signal and receives the line echo from the hybrid interface circuit to perform learning of the line echo path characteristics.

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: A system and a method for canceling an echo is disclosed. In accordance with the system and the method, a plurality of independently and variably delayed adaptive algorithm blocks are selectively applied to a delayed feedback signal to generate a plurality of echo components in parallel, thereby canceling the echo component from an input signal.

Abstract: The ability to combine multiple audio signals captured from the microphones in a microphone array is frequently used in beamforming systems. Typically, beamforming involves processing the output audio signals of the microphone array in such a way as to make the microphone array act as a highly directional microphone. In other words, beamforming provides a “listening beam” which points to a particular sound source while often filtering out other sounds. A “generic beamformer,” as described herein automatically designs a set of beams (i.e., beamforming) that cover a desired angular space range within a prescribed search area. Beam design is a function of microphone geometry and operational characteristics, and also of noise models of the environment around the microphone array. One advantage of the generic beamformer is that it is applicable to any microphone array geometry and microphone type.

Abstract: A method of providing an audio signal to an audio output device may include receiving a first audio signal generated by a microphone located in a physical environment; processing the first audio signal at least to provide echo cancellation to obtain an echo-canceled first audio signal; generating a livening signal based on the echo-canceled first audio signal; and providing the generated livening signal to an audio output device located in the physical environment.

Abstract: An oscillation-echo preventing circuit according to the present invention has a microphone/speaker unit (12) and a voltage-canceling circuit (13) for canceling voltages of audio receive signals. The microphone/speaker unit (12) has a main body, at least two microphones MIC1 and MIC2, and a speaker SP1 or an earphone. The microphone MIC1 seals a first inside space from an outside space. The microphone MIC2 seals the first inside space from a second inside space. The speaker SP1 or the earphone seals the first inside space from the outside space. The voltage-canceling circuit (13) 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: Acoustic echo control and noise suppression is an important part of any “handsfree” telecommunication system, such as telephony or audio or video conferencing systems. Bandwidth and computational complexity constraints have prevented that stereo or multi-channel telecommunication systems have been widely applied. The advantages are very low complexity, high robustness, scalability to multi-channel audio without a need for loudspeaker signal distortion, and efficient integration of echo and noise control in the same algorithm.

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.

Abstract: Signal detectors are described herein. By way of example, a system for detecting signals can include a microphone signal detector, a loudspeaker signal detector, a signal discriminator and a decision component. When the microphone signal detector detects the presence of a microphone signal, the loudspeaker signal detector detects the presence of a loudspeaker signal and the signal discriminator determines that near-end speech dominates loudspeaker echo, the decision component can confirm the presence of doubletalk. When the microphone signal detector detects the presence of a microphone signal and the signal discriminator determines that near-end speech dominates loudspeaker echo, the decision component confirms the presence of near-end signal.

Abstract: A method for multifunctional processing of signals in frequency subbands performs subband decomposition and signal processing in two stages. A fullband signal is first splitted, with downsampling, into wide frequency subband (WFS) signals. Processing algorithms not requiring a high frequency resolution but benefiting from downsampling (such as subband acoustic echo cancellation), are applied to the WFS signals by wide subband processing blocks. Processed WFS signals are splitted, preferably without downsampling, into groups of narrow frequency subband (NFS) signals. The NFS signals are processed using processing algorithms (noise suppression, etc.) requiring a higher resolution. Processed NFS signals are synthesized into processed WFS signals, which are recombined into an output signal. Two-stage processing makes it possible to optimize signal processing, while keeping computational costs at low level and avoiding undesirable time delays.

Abstract: One embodiment of the present invention includes an acoustic echo removal system. A transmit signal and a receive signal each having high and low frequency portions, are propagated between a microphone and at least one voice processor. A first acoustic echo removal portion determines and provides a first variable attenuation gain to the low-frequency portion of the transmit signal at a first sample frequency and provides a second variable attenuation gain to the low-frequency portion of the receive signal at the first sample frequency. A second acoustic echo removal portion provides the first variable attenuation gain to the high-frequency portion of the transmit signal at a second sample frequency and provides the second variable attenuation gain to both the high-frequency portion of the receive signal and a copy of the low-frequency portion of the receive signal at the second sample frequency.

Abstract: Coefficients of an adaptive filter representative of an acoustic channel between an emitted acoustic signal and a microphone signal are determined and smoothed in time. An echo is then estimated by filtering the emitted acoustic signal with the smoothed coefficients. Properties of the estimated echo and of the microphone signal are estimated. The echo cancellation filter is controlled as a function of a comparison between the properties of the estimated echo and those of the microphone signal so as to take into account the potential presence of a signal other than an echo signal in the microphone signal.

Abstract: This invention proposed an Echo Suppressor which can efficiently suppress both echoes and background noise without introducing “choppiness”. The Echo Suppressor System includes said two adaptive gains Gr(RSR) and Gn(NSR), said one adaptive zeros-filter A1(z) and said one adaptive poles-filter A2(z); wherein, thr gain Gr(RSR) is controlled by RSR (Residual echo level to Signal level Ratio); the gain Gn(NSR) is controlled by NSR (Noise signal level to current Signal (Tx) level Ratio); the filter A1(z) is converted from LSF1 obtained from the first modification of LSFTx (Line Spectral Frequencies of Tx signal); the filter A2(z) is converted from LSF2 obtained from the second modification of LSFTx.

Abstract: An echo reduction system includes a signal analysis filter that converts an input into sub-band signals. A signal down-sampling circuit down-samples the sub-band signals at a first down-sampling rate. An echo analysis filter converts a loudspeaker signal into echo sub-band signal that are further processed by an echo down-sampling circuit. The circuit down-samples the echo sub-band signals at a second down-sampling rate to generate down-sampled echo sub-band signals. An echo compensation filter folds the down-sampled echo sub-band signals with an estimated impulse response of a loudspeaker-room-input system. A second echo down-sampling circuit down-samples the folded down-sampled echo sub-band signals at a third down-sampling rate to generate estimated echo sub-band signals. The first down-sampling rate is equal to the product of the second and third down-sampling rates.

Abstract: According to one embodiment, An information processing apparatus includes a first signal input unit configure to receive a first signal, a second signal input unit configure to receive a signal, a first control unit configure to acquire system resources, a second control unit configure to select, in accordance with information of the system resources acquired by the first control unit, a processing method for suppressing at least one of echo and noise of the second signal input from the second signal input unit containing the echo due to the first signal input from the first signal input unit, a third control unit configure to generate an output signal by suppressing at least one of the echo and the noise from the second signal by the processing method selected by the second control unit, and a signal output unit configure to output the output signal generated by the third control unit.

Abstract: The invention is a method and apparatus for frequency-domain adaptive filtering that has broad applications such as to equalizers, but is particularly suitable for use in acoustic echo cancellation circuits for stereophonic and other multiple channel teleconferencing systems. The method and apparatus utilizes a frequency-domain recursive least squares criterion that minimizes the error signal in the frequency-domain. In order to reduce the complexity of the algorithm, a constraint is removed resulting in an unconstrained frequency-domain recursive least mean squares method and apparatus. A method and apparatus for selecting an optimal adaptation step for the UFLMS is disclosed. The method and apparatus is generalized to the multiple channel case and exploits the cross-power spectra among all of the channels.

Abstract: Disclosed herein is an echo canceller for use in a sound reinforcement communication system configured to carry out a sound reinforcement communication by utilizing a speaker and a microphone, the echo canceller including: an adaptive filter section configured to adaptively identify an impulse response of a feedback path formed by an acoustic coupling or the like between the speaker and the microphone to estimate an echo component in the feedback path from an input signal to the feedback path, and subtracting the echo component thus estimated from an output signal from the feedback path; and an echo suppressing section configured to execute echo suppressing processing for an output signal from the adaptive filter section.

Abstract: The invention provides a sound-processing device with automatic howl cancellation. The sound-processing device includes an array microphone, a digital signal processor, a power amplifier, and a loudspeaker. The array microphone includes a plurality of microphones, receiving a sound wave at different locations and converting the sound wave to a plurality of audio signals, wherein the audio signals carry howl induced by a sound wave feedback. The digital signal processor includes a beam forming module and an acoustic echo cancellation module. The beam forming module derives a beam signal from the audio signals to suppress out-of-beam howl, and the acoustic echo cancellation module estimates and eliminates howl carried by the beam signal. The power amplifier then amplifies the beam signal subsequent to eliminating howl. Finally, the loudspeaker converts the amplified beam signal to an amplified sound wave.

Abstract: A signal processing device includes an adaptive filter (5), a noise estimation circuit (10), and a double talk detection circuit (81) and operates so that the double talk detection circuit (81) detects a double talk by using the estimated noise obtained by the noise estimation circuit (10). The signal processing device further includes noise estimation means and detects a double talk by using an estimated noise, a microphone signal, and pseudo-echo. An echo removal method and device detect a double talk by using a reliability coefficient expressed as continuous values between 0 and 1. By using continuous values instead of two values 0 and 1, it is possible to reduce the affect of a detection error.

Abstract: An echo cancellation device (1) comprises a first adaptive filter (13) for producing a first echo cancellation signal (y1), a second adaptive filter (15) for producing a second echo cancellation signal (y2), and a post-processor (18) for suppressing any remaining echo. The first adaptive filter (13) and the second adaptive filter (15) are designed for canceling a first (e.g. direct) part of the echo impulse response and a second (e.g. diffuse) part of the echo impulse response respectively. The device (1) may be utilized in a mobile telephone.

Abstract: The invention is a method and apparatus for frequency-domain adaptive filtering that has broad applications such as to equalizers, but is particularly suitable for use in acoustic echo cancellation circuits for stereophonic and other multiple channel teleconferencing systems. The method and apparatus utilizes a frequency-domain recursive least squares criterion that minimizes the error signal in the frequency-domain. In order to reduce the complexity of the algorithm, a constraint is removed resulting in an unconstrained frequency-domain recursive least mean squares method and apparatus. A method and apparatus for selecting an optimal adaptation step for the UFLMS is disclosed. The method and apparatus is generalized to the multiple channel case and exploits the cross-power spectra among all of the channels.

Abstract: The present invention relates to a method for designing a modal equalizer (5) for a low frequency audible range, typically for frequencies below 200 Hz for a predetermined space (1) (listening room) and location (2) (location in the room) therein, in which method modes to be equalized are determined at least by center frequency and decay rate of each mode, creating a discrete-time description of the determined modes, and determining equalizer filter coefficients on the basis of the discrete-time description of the determined modes, and forming the equalizer (5) by means of a digital filter by defining the filter coefficients on the basis of the properties of the modes.

Abstract: An audio processor of a loud speech communication system including a speaker and a microphone is provided. The audio processor includes: an adaptive filter wherein an amount of update in a learning event is set to an arbitrary value, and a filter coefficient is serially determined corresponding to the set amount of update; a semi-fixed filter adapted to an echo cancellation process of an audio input signal input from the microphone; adaptive filter assessment unit that calculates a length of an update vector based on the filter coefficient determined by the adaptive filter and a length of an update vector based on a filter coefficient set in the semi-fixed filter and that performs assessment of the filter coefficients in accordance with the update vectors; and coefficient specifying unit that sets an optimal filter coefficient among the filter coefficients into the semi-fixed filter in accordance with the result of the assessment of the filter coefficients performed by the adaptive filter assessment unit.

Abstract: A method and apparatus for reducing echo in a communication system are provided. A first signal including a voice component associated to a speaker is received. The first signal is processed to derive a harmonic feature of the voice component. A second signal including an echo component correlated to the first signal is also received. The second signal is processed at least in part on the basis of the harmonic feature of the voice component to remove at least in part the echo component such as to derive an echo reduced signal. The echo reduced signal is then released. In specific implementations, the harmonic feature of the voice component is an estimate of the pitch associated to the voice component.

Abstract: Headsets are used in variety of applications to facilitate one- or two-way audio communications between users and/or devices. The present inventor recognized that conventional headsets lack means for successfully integrating more than one audio source, despite their use in proximity to multiple sources of audio signals, such as cell phones, laptops, aircraft radios, and so forth. Accordingly, the present inventors devised one or more devices, circuits, and methods related to connection of at least two audio input signals to a headset. For example, in one embodiment, an active-noise-reduction (ANR) headset includes at least one auxiliary port for connection to an output of at least one device, such as a personal communications, computing, and/or entertainment device. This exemplary headset also includes a primary port for connection to a two-radio or public-address system and circuitry for automatically suppressing or muting the volume of an auxiliary input signal relative to that of a primary input signal.

Abstract: A nonlinear response function of a loudspeaker is determined by an iterative process during which the nonlinear response function and a linear response associated with an echo and microphone are alternately revised.

Abstract: A system and method facilitating signal enhancement utilizing an adaptive filter is provided. The invention includes an adaptive filter that filters an input based upon a plurality of adaptive coefficients and modifies the adaptive coefficients based on a feedback output. A feedback component provides the feedback output based, at least in part, upon a non-linear function of the acoustic reverberation reduced output. Optionally, the system can further include a linear prediction (LP) analyzer and/or a LP synthesis filter. The system can enhance signal(s), for example, to improve the quality of speech that is acquired by a microphone by reducing reverberation. The system utilizes, at least in part, the principle that certain characteristics of reverberated speech are measurably different from corresponding characteristics of clean speech. The system can employ a filter technology (e.g., reverberation reducing) based on a non-linear function, for example, the kurtosis metric.

Abstract: A cabin communication system for improving clarity of a voice spoken within an interior cabin having ambient noise includes an adaptive speech enhancement filter for receiving an audio signal that includes a first component indicative of the spoken voice, a second component indicative of a feedback echo of the spoken voice and a third component indicative of the ambient noise, the speech enhancement filter filtering the audio signal by removing the third component to provide a filtered audio signal, the speech enhancement filter adapting to the audio signal at a first adaptation rate, and an adaptive acoustic echo cancellation system for receiving the filtered audio signal and removing the second component in the filtered audio signal to provide an echo-cancelled audio signal, the echo cancellation signal adapting to the filtered audio signal at a second adaption rate, wherein the first adaptation rate and the second adaptation rate are different from each other so that the speech enhancement filter does not

Abstract: The present invention provides an indication to the user of a voice communication system that an echo at the user's location has been detected by the system. In response to receiving this indication of echo, users may take any of several actions to reduce the echo, such as changing the position of the microphone or changing their own position with respect to the microphone. The need for potentially expensive echo-canceling hardware and software is reduced because the actions to eliminate the echo are performed by the user. The echo detector in the present invention may be implemented as part of a voice terminal, or as part of a communication switch or server that is used in conjunction with a voice terminal.

Abstract: A system and method facilitating signal enhancement utilizing an adaptive filter is provided. The invention includes an adaptive filter that filters an input based upon a plurality of adaptive coefficients and modifies the adaptive coefficients based on a feedback output. A feedback component provides the feedback output based, at least in part, upon a non-linear function of the acoustic reverberation reduced output. Optionally, the system can further include a linear prediction (LP) analyzer and/or a LP synthesis filter. The system can enhance signal(s), for example, to improve the quality of speech that is acquired by a microphone by reducing reverberation. The system utilizes, at least in part, the principle that certain characteristics of reverberated speech are measurably different from corresponding characteristics of clean speech. The system can employ a filter technology (e.g. reverberation reducing) based on a non-linear function, for example, the kurtosis metric.