Abstract: The loudspeaker device according to the present invention comprises a loudspeaker; a feedforward processing section for performing feedforward processing on an electric signal to be inputted to the loudspeaker based on a preset filter coefficient so that non-linear distortion which occurs from the loudspeaker is removed; and a feedback processing section for detecting vibration of the loudspeaker, and performing feedback processing on an electric signal concerning the vibration with respect to the electric signal to be inputted to the loudspeaker. The feedback processing section performs feedback processing on the electric signal concerning the vibration so that the non-linear distortion which occurs from the loudspeaker is removed and so that a frequency characteristic concerning the vibration of the loudspeaker becomes a predetermined frequency characteristic.

Abstract: In an audio amplification apparatus connected to a microphone and a speaker, an audio amplification circuit amplifies an audio signal and inputs the amplified audio signal to the speaker. A howling canceller has an adaptive filter which is set with a filter coefficient based on the audio signal input to the speaker and a residual signal so as to simulate a feedback transmission path from the speaker to the microphone such that the adaptive filter processes the audio signal to produce a simulation signal. The residual signal is obtained by subtracting the simulation signal from an input audio signal inputted from the microphone and fed to the audio amplification circuit. An internal sound source generates the audio signal and inputs the audio signal to the speaker.

Abstract: A method and system is provided for eliminating acoustical feedback in a system. The method determines a parameter for at least one notch filter, adjusting the notch filter based on the parameter, processing the digital signals through the notch filter, testing at the effect of the notch filter in the system, and removing the notch filter if the notch filter is not effective. Also disclosed is a method and system of selecting candidate frequencies which might be feedback, as opposed to other wanted sound frequencies. The selection method sampling the digital signals, converting the time domain digital signal samples by a fast Fourier transform algorithm into the frequency domain, using a ballistics approach to find prominences in the frequency spectrum, and testing the sizes of the prominences.

Abstract: A system for canceling acoustic feedback includes an input for receiving a digital audio signal and a processor configured to detect acoustic feedback signal in the digital audio signal and to determine the frequency of the feedback signal The system also includes a plurality of bandpass filters for attenuating the feedback signal. The processor is further configured to: select a bandpass filter from among the plurality of bandpass filters. The selected bandpass filter comprises a response characteristic that attenuates parts of the signal at the frequency of acoustic feedback signal.

Abstract: Method and apparatus for determination of gain margin of a hearing assistance device under test. In varying examples, the impulse response for multiple levels can be taken and used to arrive at a gain margin. The method and apparatus, in various examples, process critical portions of the resulting data for efficient processing and to increase accuracy of measurements. The method and apparatus performing a plurality of measurements to determine impulse responses and to derive gain margin as a function of frequency therefrom. The present subject matter includes principles which may are adapted for use within a hearing assistance device using a single white noise stimulus, according to one example. The principles set forth herein can be applied to occluding and non-occluding hearing device embodiments. Additional method and apparatus can be found in the specification and as provided by the attached claims and their equivalents.

Abstract: A multiple channel steered spatialized signal is generated from a signal input modified according to respective spatialization gain functions to generate a plurality of audio channels. An echo cancellation signal is applied to a return path using a combined spatialization and echo path estimate. The estimate is derived from the gain functions applied to the respective channels. When the gain functions applied in the respective channels are changed, for instance to represent a different apparent position of the sound source, a new estimate of the echo paths is generated, based on a previous estimate of the echo path and on the new gain functions.

Abstract: A signal processing system comprises a microphone (20), a subtractor (22) arranged to receive an output of the microphone (20), an amplifier G arranged to receive an output of the subtractor (22), a rear loudspeaker (24) arranged to receive an output of the amplifier G, a front loudspeaker (26) arranged to receive an output of the amplifier G, and one or more summers (28) interposed between the amplifier G and a loudspeaker (24, 26), the or each summer (28) arranged to add an audio signal m[k] to the signal s[k] received from the amplifier G.

Abstract: A multichannel acoustic echo reduction system is described herein. The system includes an acoustic echo canceller (AEC) component having a fixed filter for each respective combination of loudspeaker and microphone signals and having an adaptive filter for each microphone signal. For each microphone signal, the AEC component modifies the microphone signal to reduce contributions from the outputs of the loudspeakers based at least in part on the respective adaptive filter associated with the microphone signal and the set of fixed filters associated with the respective microphone signal.

Abstract: An apparatus for suppressing feedback in an environment where a microphone and a loudspeaker are located, comprises a means for embedding a test signal into a loudspeaker signal, a microphone signal or a modified microphone signal, preferably by using a psychoacoustic masking threshold by using a pseudo-noise test signal, a means for determining a characteristic of a transmission channel in the environment between the loudspeaker and the microphone by using the embedded test signal and the microphone signal, a filter for filtering the loudspeaker signal to obtain a filtered loudspeaker signal, wherein the filter is adaptable to be adapted with regard to its filter characteristic to the characteristic of the transmission channel by the means for determining, as well as a means for subtracting the filtered loudspeaker signal from the microphone signal to obtain the modified microphone signal, in which the feedback is reduced due to the loudspeaker signal.

Abstract: An on-channel repeater has a receiving antenna for receiving an RF signal and a transmitting antenna for transmitting on the same frequency as the input signal. An amplification path between the antennas provides substantially linear processing and includes a combiner, a decorrelating delay and a power amplifier. A filter estimator receives a reference signal and the combiner output and generates a plurality of control coefficients which are applied to an adaptive filter which filters the reference signal accordingly and applies its output to the combiner. The reference signal is derived not from the output of the decorrelating delay but rather from the output antenna after the power amplifier. Where there is an adjacent transmitter generating potentially interfering signals, the transmitter output is coupled to the same transmitter antenna.

Abstract: A decorrelation method for improving feedback cancellation utilizes a small frequency shifting ratio, on the order of 0.3 percent. Frequency shifting is applied only to the high frequency portion of the signal, which is shifted alternately upward and downward.

Abstract: A signal processing system improves signal quality by accurately locating and eliminating a feedback signal in an input signal, such as an audio signal. The signal processing system interpolates between frequency sample points to obtain a more accurate identification of a feedback signal frequency. A less intrusive filter reduces or eliminates the identified frequency signal frequency without excessive adverse effects on adjacent frequencies in the input signal.

Abstract: The invention relates to a listening system comprising a first input transducer for converting an input sound to an electrical input signal, the electrical input signal comprising a direct part and an acoustic feedback part, an output transducer for converting an electrical output signal to an output sound, a forward path being defined between the input and output transducer and comprising a signal processing unit, a feedback cancellation system for estimating acoustic feedback comprising an adaptive FBC filter arranged in parallel to the forward path, the adaptive FBC filter comprising a variable FBC filter part and an FBC update algorithm part for updating the variable FBC filter part, the FBC update algorithm part comprising first and second FBC algorithm input signals influenced by the electrical input and output signals, respectively, the first and second FBC update algorithm input signal paths comprising first and second variable filters, respectively, the listening system further comprising an electric

Abstract: An interference signal removing apparatus of a radio frequency (RF) receiver includes a low noise amplification unit which performs low noise amplification, a feedback processing unit which removes a necessary signal in a desired band from a signal output from the low noise amplification unit, and performs feedback of the signal from which the necessary signal is removed, and a signal processing unit which transmits a processed RF signal by synthesizing an input RF signal and the feedback signal to the noise amplification unit.

Abstract: A voice recognition system includes an adaptive filter and a subtractor. The adaptive filter generates a simulated talk-back voice y(n) by setting a filter coefficient simulating a transfer system in which an input voice corresponding to a voice command and a talk-back voice output from a speaker are input into a microphone and by filtering a talk-back voice x(n). The subtractor extracts the input voice by subtracting the simulated talk-back voice y(n) from mixed sound input into the microphone. With this configuration, the talk-back voice is attenuated from the mixed sound including the input voice and the talk-back voice input tedinto the microphone, and then, the mixed sound is supplied to a voice recognition engine. Accordingly, the user can input his/her voice during a talk-back operation without the need to interrupt it by pressing a speech button every time the user wishes to input the voice. The voice recognition operation time can be thus reduced.

Abstract: In a howling detection circuit, an audio signal voltage generator generates an audio signal voltage based on a level of an input audio signal. A capacitor is chargeable and dischargeable by the audio signal voltage and develops a monitor voltage. An indication circuit is activated depending on the monitor voltage to provide an alert indicating that a howling noise is contained in the input audio signal. A discharging element discharges the capacitor according to a first time constant when the audio signal voltage is lower the monitor voltage, thereby lowering the monitor voltage. A charging element charges the capacitor according to a second time constant being set greater than the first time constant when the audio signal voltage is higher than the monitor voltage, thereby raising the monitor voltage so as to activate the indication circuit.

Abstract: An input sound processor compares power at each frequency component of an input sound with a reference value, and sets multiplication points indicating frequency components at which the total power of the input sound is to be determined. A product-sum operation is performed at the multiplication points on the power at each frequency component and the square amplitude of each filter coefficient indicating the transfer characteristic from a loudspeaker to a microphone to estimate the total power of the input sound at the position of the microphone.

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

Abstract: A method for providing an embellishment representation of a noise information is discloses.

Abstract: Disclosed is a hearing aid, where at least part of the hearing aid is adapted to be inserted into an ear canal of a user such that one end of the hearing aid and the tympanic membrane in the ear canal define a residual space of the ear canal, the hearing aid comprising an ambient space input transducer adapted to convert ambient sound to an electric sound signal, when the ambient sound reaches the ear of a user from an ambient space, a processor connected to said ambient space input transducer and adapted to process said electric sound signal, an output transducer connected to said processor and adapted to convert said processed electric sound signal to a sound pressure in the residual space of the ear canal, and wherein the hearing aid further comprises a residual space input transducer connected to said processor and arranged to predominantly receive said sound pressure in the residual space, and wherein the residual space input transducer is adapted to convert said sound pressure to an electric residual sp

Abstract: A control unit gives acoustic environment instruction data to a picked up sound directionality control unit and an adaptive filter. According to this, the picked up sound directionality control unit generates a picked up sound signal constituted by a predetermined picked up sound directionality. The adaptive filter detects the picked up sound directionality from the acoustic environment instruction data, and reads out the filter parameter corresponding to this picked up sound directionality, from a memory. The adaptive filter sets a delay coefficient and a filter coefficient of an FIR filter, and generates a pseudo echo signal by an impulse response with respect to the received sound signal. Based on an error signal obtained by subtracting the pseudo echo signal from the picked up sound signal by an adder, the adaptive filter sets a more optimum filter parameter, and generates the next pseudo echo signal.

Abstract: An adaptive beamforming apparatus and method includes a fixed beamformer that compensates for time delays of M noise-containing speech signals input via a microphone array having M microphones (M is an integer greater than or equal to 2), and generates a sum signal of the M compensated noise-containing speech signals; and a multi-channel signal separator that extracts pure noise components from the M compensated noise-containing speech signals using M adaptive blocking filters that are connected to M adaptive canceling filters in a feedback structure and extracts pure speech components from the added signal using the M adaptive canceling filters that are connected to the M adaptive blocking filters in the feedback structure.

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: Optimal head related transfer function spatial configurations designed to maximize speech intelligibility in multi-talker speech displays by spatially separating competing speech channels combined with a method of normalizing the relative levels of the different talkers in a multi-talker speech display that improves overall performance even in conventional multi-talker spatial configurations.

Abstract: A hearing assistance system having adjustable bulk delay for cancellation of a time varying acoustic feedback path. The hearing assistance system including an FIR filter, coefficient update module, and delay rules module for programmable adaptive filtering. The hearing assistance system adjustable for continuous bulk delay adjustments. The hearing assistance system providing a number of coefficient update routines, including, but not limited to an LMS coefficient update process and a normalized LMS coefficient update process.

Abstract: Acoustical signals impinging on an input converter are converted into a first electric signal by a controllably variable transfer characteristic. The transfer characteristic is dependent on the angle at which the acoustical signals impinge on the input converter. The first electric signal is processed and a resulting signal is applied to an output converter. Feedback to be suppressed is compensated by a feedback compensating signal, which is generated in dependency of the resulting signal and is fed back by a feedback signal path upstream of the processing. The electric feedback compensating signal is fed back to and superimposed upon the first electric signal. An adaptation rate of the converting into a first electric signal by a controllably variable transfer characteristic is controlled in dependency of the loop gain along the feedback signal path.

Abstract: The determination of a suitable step size for controlling the adaptation of a filter of an acoustic feedback compensator especially for hearing aids is to be improved. A step size controller is therefore provided in which an input signal is analyzed in at least two frequency bands. In doing so, any steep signal edges in the individual frequency bands are detected. The adaptation step size of an adaptive filter is then controlled depending on the number of frequency bands in which steep signal edges are detected, wherein the magnitude of the signal edges can be included in the decision.

Abstract: The invention relates to oscillation detection and, more particularly, concerns a method and apparatus for identifying oscillation in a signal due to feedback, permitting appropriate action to be taken to suppress the oscillation. The method involves using an FFT device or similar to convert a signal at each of a series of successive time windows into the frequency domain, calculating, for each of a plurality of frequency bands, the change in signal phase from a time window to a subsequent time window, and comparing, for some or all of said frequency bands, the results of the calculation step to one or more defined criteria to provide a measure of whether oscillation due to feedback is present in the signal. For additional discrimination, the change in signal amplitude from a time window to a subsequent time window may also be calculated for each of the frequency bands, and the result compared with one or more further defined criteria. The invention has particular application in hearing aid devices.

Abstract: An improved method for adaptively cancelling acoustic feedback in hearing aids and other audio amplification devices. Feedback cancellation is limited to a frequency band that encompasses all unstable frequencies. By limiting the bandwidth of the feedback cancellation signal, the distortion due to the adaptive filter is minimized and limited only to the unstable feedback regions. A relatively simple signal processing algorithm is used to produce highly effective results with minimal signal distortion.

Abstract: The invention concerns a method (500) and system (100) for suppressing receiver audio regeneration. The method (500) includes the steps of receiving a communication signal (502), at a Radio Frequency (RF) unit (102), demodulating the communication signal to an audio signal (504), monitoring a volume level of the audio signal (506), and shifting the pitch of the audio signal when the volume level reaches a predetermined threshold (508), and playing the pitch-shifted audio signal out of a speaker to produce a pitch-shifted acoustic signal (510). The method can shift the pitch of the audio signal to produce a pitch-shifted acoustic signal with signal properties suppressing regeneration of the acoustic signal onto the audio signal at the RF unit. The amount of pitch-shifting can be a function of the volume level.

Abstract: Improved approaches to adaptively suppress interfering noise in a multi-microphone directional system are disclosed. These approaches operate to adapt the direction null for the multi-microphone directional system in accordance with a dominant noise source.

Abstract: The present invention provides method and apparatus that improve processing of acoustic signals by reducing acoustic feedback in an acoustic system. An aspect of the invention is a multi-channel digital feedback reducer (DFR) system that comprises a plurality of channel elements. Each channel element comprises a notch filter configuration having an adaptive notch filter and an operative notch filter. The operative notch filter processes a signal received from an acoustic input device and provides the processed signal to an acoustic output device, in which acoustic feedback between the acoustic input device and the acoustic output device is ameliorated. If acoustic feedback is detected by a channel element, the channel element informs other channel elements of the multi-channel DFR system about the detected feedback to ensure that all channel elements may incorporate the same notch filters. During the notification, the other channel elements may continue searching for feedback on the associated channels.

Abstract: Feedback reduction in hearing aids is simplified and improved in an amplification device that has two separate compensation paths, an acoustic input and an inductive input. In the first compensation path, a first filter device acts to compensate acoustic feedback, and in the second compensation path, a second filter device acts to compensate inductive feedback. The two very different feedbacks can be individually compensated without increased outlay.

Abstract: This invention is an audio system capable of identifying frequencies of feedback signals and filters these feedback signals. Frequency interpolation is utilized on a sampled frequency spectrum signal corresponding to a feedback signal, where the frequency interpolation allows the frequency of the feedback signal to be identified, especially where the frequency lies between samples of the frequency spectrum signal. The frequency identification allows a less intrusive filter, such as a notch filter, to be placed at the determined frequency of the feedback to eliminate the feedback signal. The placement of the notch filter reduces the effect on the audio signal provided by the audio system. The audio system may provide adaptive filtering of multiple feedback signals, using a single filter such as a notch filter.

Abstract: A howling control apparatus and howling control method for controlling time up to the cancellation of howling suppression according to a howling occurrence situation, thereby eliminating the repetition of howling suppression and cancellation. A howling detecting section (104) detects howling based on the band level and the band level average value and measures time for which no howling occurs, a waiting time setting section (105) decides waiting time to be set this time from time for which no howling occurs and a previous waiting time, and a gain control section (106) causes a gain that is set to a howling suppressing section (107) to be retuned to a normal value during the waiting time.

Abstract: Systems, devices, and methods are provided to inhibit at least one feedback component of an input audio signal by adjusting a feedback-inhibiting filter through a narrowband subaudible signal. The level of the signal may be determined using an audibility model so as to yield its subaudible quality.

Abstract: The invention relates to a method for reducing feedback caused by electromagnetic interference between an induction coil and an output transducer, a voltage supply or an amplifier or a combination thereof, the method comprising: producing by means of a filter an equivalent of an electromagnetic feedback path in a system with a pickup coil and subtraction of the equivalent of the feedback signal from the input signal in order to obtain feedback reduction. The invention further relates to a device for implementing the method, the device comprising an induction coil and an output transducer, voltage supply or an amplifier or a combination thereof. The device further comprises filter means for generating an equivalent of an electromagnetic feedgack signal occurring between the induction coil and an output transducer, a voltage supply or an amplifier or a combination thereof and means for subtracting the equivalent from an input signal in order to obtain feedback reduction.

Abstract: A loudspeaker device comprising a power amplifier (13) to which an input signal is delivered via a subtracter (6), a speaker unit (1) for reproducing output signals delivered from the power amplifier (13), an acoustic pipe (2) coupled in front of the speaker unit for guiding the sound waves, a microphone (4) for detecting an acoustic outputs radiated from the speaker unit, a microphone amplifier (5) for amplifying an acoustic output signals detected by the microphone, and a negative feedback circuit. Output signals of the microphone amplifier are delivered to the subtracter, at the same time the same output signal is connected to a high-pass filter (7) to be delivered to the subtracter. Thus the negative feedback circuit is formed to suppress peaks and dips for providing a flattened sound pressure frequency characteristic.

Abstract: A system and method for adaptively removing feedback in audio systems and, more particularly, hearing aid systems. The audio system comprises an analysis unit, for receiving an input signal and providing N bandpass input signals, and an adaptive feedback cancellation unit for removing a feedback condition in one or more of the N bandpass input signals. The adaptive feedback cancellation unit comprises N sub-units with at least one of the sub-units including: (i) a feedback detector for indicating the presence of the feedback condition in one of the bandpass input signals, (ii) an adaptive feedback canceller for providing an adaptive gain modification factor for adjusting gain when the feedback condition is detected within one of the bandpass input signals, and (iii) a multiplier coupled to the adaptive feedback canceller and the analysis unit for providing a bandpass output signal based on one of the bandpass input signals and the corresponding adaptive gain modification factor.

Abstract: Method and system for canceling echoes and noises in environments with variable acoustic conditions and high feedback, such as the interior of an automotive vehicle, permitting audible communication between the occupants, comprising one/several microphone/s (1) for a speech signal (5) and A/D, D/A converters, amplification and filtering means generating an electric signal (13) reproduced as an acoustic signal (7) by a speaker (3), and an echo canceller (15) using said signal (13) and an electric signal (10) from the microphone integrating the feedback of said signal (7), another speech signal (5) and noise (9) seized by the microphone (1), the system (15) being applied to eliminating said feedback, the proposed method and system providing an additional filtering of an electric output signal (11) of the canceling device (15), after treatment of said signal, comprising a time variant filter that suppresses the residual acoustic echo not cancelled by the system (15) and the noise seized by said microphone/s (1).

Abstract: Stereo echo cancellation is necessary to overcome the objections observed by, for example, teleconferencing, voice controlled video/audio apparatuses, etc. To improve the existing filters, an adaptive filter is used along with a signal processing device which obtain the coefficient updates in the transformed domain, reducing the required calculation complexity. Further, the filter includes circuitry for reducing the correlation between the input signals on the coefficient updates.

Abstract: A sound reinforcement system includes several microphones, a microphone beamformer coupled to the microphones, adaptive echo compensator (EC) coupled to the microphone beamformer for generating an echo compensated microphone signal, and several loudspeakers coupled to the adaptive EC. An adaptive loudspeaker beamformer is coupled between the adaptive EC and the loudspeakers for shaping the directional pattern of the loudspeakers. The adaptive loudspeaker beamformer creates a beam pattern which is capable of creating a “null” in the direction of speaker(s) such that howling is effectively prevented.

Abstract: This invention includes a circuit for the prevention of acoustic feedback between an electronic device and an audio accessory. In a preferred embodiment, the circuit prevents audio feedback between a cellular telephone and a speakerphone accessory. The circuit includes a current limiting device coupled serially in the receive (Rx) line. The current limiting device is actuated via a delay circuit coupled between the current limiting device and the transmit (Tx) line. When a bias current is presented to the Tx line, the bias propagates through the delay circuit, thereby actuating the current limiting device a predetermined time after the presentation of the bias. The circuit keeps the Rx line open long enough for the phone to deactivate its internal microphone.

Abstract: In a hearing aid having an oscillation detector and a method for establishing the presence of oscillations in a hearing aid, sinusoidal input signals of the microphone can be detected, and so oscillations that are present can also be detected. To this end, the number of digitized sample values in consecutive periods of the input signal is determined, and a long-term average value NL and a short-term average value NK are formed from these numbers. When NL and NK are essentially identical, the presence of oscillations is detected.

Abstract: For reducing feedback-conditioned oscillations in a hearing aid device, microphone signals of a first microphone and of a distanced, second microphone are compared to one another. When oscillations are detected at the same frequency in both microphone signals, these oscillations are determined to be useful (non-feedback) tonal signals. Oscillations that are only present in one of the microphone signals, in contrast, are feedback-conditioned and are suppressed using suitable measures.

Abstract: A device for operating voice-controlled systems, such as communication and/or intercommunication systems in motor vehicles, includes a plurality of microphones and at least one loudspeaker. Voice signals received by the microphones are transmitted to the at least one loudspeaker. The voice signals are subjected to a low-value frequency shift before being transmitted to the loudspeaker(s) or to the input of a voice-controlled device to thereby suppress feedback.

Abstract: An RF active deflector is described for use with orthogonal frequency division multiplex signals. An input antenna receives an RF input signal and an output antenna transmits an RF signal on substantially the same frequency as the input signal. An amplification path between the input antenna and the output antenna provides substantially linear processing and includes a delay. Means responsive to the signal in the amplification path produce a plurality of control coefficients. A transversal filter receives the signal in the amplification path and, under control by the control coefficient, provides a modified signal. A combiner is provided for combining the modified signal with the signal in said amplification path so as to reduce the effect of the feedback.

Abstract: For an especially low loss and echo free communication between several participants in a room (1) an inventive process for the compensation of losses of a signal(S) along a transmission path between at least one transmission point (6) and one receiving point (4) in a room (1) determining the transmission path of signal (S) and via the transmission paths at least one parameter of an associated transmission function will be determined, whereby via the determined parameters the signal level for a given position (P1 through P4) on the transmission path is controlled. Additionally an especially suitable device is provided for the implementation of the process.

Abstract: A feedback compensator for a hearing aid device has a filter arrangement that splits a signal path, to implement an adaptive feedback compensation with only one buffer memory, two splitting nodes, and two addition nodes. The feedback compensation ensues only in the feedback-susceptible frequency range of the input signal. In addition to the filtering of the input signal, it is advantageous to feed to the adaptive feedback compensation filter a bandwidth-limited signal that is taken from the amplified output signal.

Abstract: An audio conferencing system in which each participant in the conference generates a monaural signal for transmission on an output channel to the other participants, and receives a plurality of input channels from the other participants. The input channels are individually converted to stereophonic signal pairs by respective filter banks having selected predetermined transfer functions. The stereo channels are combined linearly to provide a stereophonic audio output. The system has an echo cancellation system in which an adaptive filter process is applied to the input channels to generate a combined echo cancellation signal for applying to the output channel.