Abstract: A multi-channel acoustic cancellation system 40 with, for example, stereo speakers and a pair of microphones in the transmitting and receiving rooms (11 and 21) has time varying all-pass filters (45, 47) in the signal path between the microphones (13, 15) in the transmitting room (11) and the speakers (27, 29) in the receiving room (21) to provide decorrelation.

Abstract: Filter coefficients of a beamformer are computed based on a segment of input samples. The segment of input samples is divided into a plurality of blocks of input samples wherein the plurality of blocks of input samples are received by a shared memory at a first rate. The first block of the plurality of blocks is received in the shared memory at a first time. The plurality of blocks of input samples from the shared memory are read out at a second rate wherein the first block of the plurality of blocks is read from the shared memory at a second time. A plurality of partial covariance matrices for the plurality of blocks read from the shared memory are computed and added together to determine a covariance matrices used to compute the filter coefficients.

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: An adaptive resonance canceller system and method for attenuating narrowband noise signals within an input signal, where the narrowband noise signals may vary significantly in frequency. The sensor comprises a notch filter, an error reference and gradient generator, and a complex correlator circuit for attenuating the narrowband noise component of the input signal. The input signal is applied to the notch filter and to the error reference and gradient generator. The latter component generates an error reference signal and an error gradient signal. These two signals are applied to the complex correlator circuit which, in turn, generates a tuning parameter signal which is applied to the notch filter to tune the filter to the center frequency of the noise component of the input signal.

Abstract: A noise shaping system including an inner loop and outer noise shaping loops. The inner noise shaping loop includes an inner loop filter and a quantizer for quantizing an output of the inner loop filter. The outer noise shaping loop includes an outer loop filter having an input receiving feedback from the quantizer of the inner noise shaping loop and an output driving an input of the inner loop filter of the inner noise shaping loop.

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: A new subband feedback cancellation scheme is proposed, capable of providing additional stable gain without introducing audible artifacts. The subband feedback cancellation scheme employs a cascade of two narrow-band filters Ai(Z) and Bi(Z) along with a fixed delay, instead of a single filter Wi(Z) and a delay to represent the feedback path in each subband. The first filter, Ai(Z), is called the training filter, and models the static portion of the feedback path in ith subband, including microphone, receiver, ear canal resonance, and other relatively static parameters. The training filter can be implemented as a FIR filter or as an IIR filter. The second filter, BI(Z), is called a tracking filter and is typically implemented as a FIR filter with fewer taps than the training filter. This second filter tracks the variations of the feedback path in the ith subband caused by jaw movement or objects close to the ears of the user.

Abstract: In order to improve the feedback compensation in hearing d vices th xtent of compensation is controlled. An estimated signal is acquired with which the intensity of the feedback signal is estimated. The damping of the feedback signal is thus controlled using the estimated signal. With this, it is, among other things, possible to disconnect the feedback compensation given no present feedback such that artifacts can be prevented.

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: Harmonic interference caused by limit cycles occurs in the resultant signals of filters for noise conversion as a consequence of limit cycles. A feedback loop is connected downstream of the actual filter and is used to effectively suppress the limit cycles. A feedback signal yFB that is added to the output signal of the filter block is generated in the feedback loop. The invention can be implemented using both analog and digital technology.

Abstract: Filter system designed to remove loudspeaker-generated sound signals from a microphone signal, comprising a stereo sampling unit for sampling the interference-bearing microphone signal as well as the interfering loudspeaker signal in time (t) to generate samples x(t) and z(t) for provision to a computation unit which derives the transfer function H(f,T) effective between the loudspeaker and the microphone and applies that function to a filter unit which acts to restore the interference-free microphone signal y(t) from x(t), z(t) and H(f,T).

Abstract: A method for enabling a user to proactively reduce the likelihood of audio feedback in an application requiring audio input and output, comprising the steps of: generating a graphical user interface (GUI) display screen including a first area for displaying information about preventing audio feedback and a second area for user selections and controls; displaying a list of available audio outputs in the second area; prompting the user to select one of the audio outputs from the list; prompting the user to select one of a plurality of muting options for each selected one of the audio outputs; and, displaying in the GUI display screen an explanation for each one of the plurality of muting options, whereby muting selections for proactively reducing the likelihood of audio feedback can be made based on user experience and knowledge. Only one of the muting option explanations is displayed at a time, responsive to the user selection of one of the muting options.

Abstract: Herein disclosed is a microphone-speaker apparatus, comprising: audio signal dividing means for dividing the audio signal into a plurality of raw component signals each indicative of the raw wave components; coherent component signal extracting means for extracting a plurality of coherent component signals respectively indicative of the coherent wave components from the raw component signals divided by the audio signal dividing means; power value calculating means for calculating the raw power value of each of the coherent component signals extracted by the coherent component signal extracting means; power value adjusting means for adjusting the raw power value of each of the coherent component signals calculated by the power value calculating means to produce an adjusted power value of each of the coherent component signals; power value judging means for judging whether or not the adjusted power value of each of the coherent component signals of the current frame exceeds the adjusted power value of each of t

Abstract: An earset including a housing positionable with respect to an ear of a person, a microphone disposed with respect to the housing for insertion into the ear of a person, the microphone operable to detect a change in air pressure within the ear while the person speaks and to produce an electrical microphone signal corresponding to the internally detected change in air pressure and a speaker disposed with respect to the housing and operable to produce a sound corresponding to an electrical speaker signal. The earset also includes a circuit coupled to receive the microphone signal and the speaker signal and operable to produce a corrected microphone signal having a reduced feedback component of the microphone signal, the feedback component resulting from the detection by the microphone of the sound produced by the speaker to produce a corrected microphone signal.

Abstract: A digital voice enhancement, DVE, communication system includes an instability detector detecting an unstable acoustic feedback condition from a loudspeaker to a microphone by sensing a condition of the electrical signal transmitted from the microphone to the loudspeaker, and a corrective processor responsive to the instability detector to modify the electrical signal to reduce unstable acoustic feedback. The sensed condition may be magnitude, power, or, preferably, the sinusoidal characteristic of the electrical signal, namely the electrical signal becoming sinusoidal in nature.

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: There is provided a speaker control apparatus capable of efficiently improving frequency characteristic of a speaker apparatus of phase inversion type, which is originally small in size and high in efficiency. The speaker control apparatus includes a detection section for detecting sound pressure within a speaker apparatus of phase inversion type and generating a sound pressure signal Spo corresponding to the detected sound pressure, a signal processing section for generating a feedback signal Sfb for feedback on the basis of the detected sound pressure signal Spo, a feedback section for feeding back the feedback signal Sfb to an input signal corresponding to sound to be radiated and generating a feedback input signal Ssy, and an amplification section for driving a speaker unit included in the speaker apparatus on the basis of the feedback input signal Ssy and causing radiation of sound.

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, the adaptive filter modifying at least one of the adaptive coefficients based on a feedback output. The invention further includes a feedback component that provides the feedback output based, at least in part, upon a non-linear function of the acoustic reverberation reduced output.

Abstract: Amplitude is corrected by subtracting a component signal from a composite signal to produce a remainder signal, correlating the remainder signal with the component signal to produce a product signal, averaging the product signal, and adjusting the magnitude of the component signal in accordance with the averaged product signal to minimize the product signal. The amplitude of the component signal is adjusted in a programmable gain amplifier controlled by an up-down counter. The up-down counter is part of a digital control loop including a pseudo-multiplier for multiplying the remainder signal with the component signal. The output of the multiplier controls the direction of the count, which is generally continuous except that it cannot roll over or roll under. The remainder signal is the received signal with the echo removed.

Abstract: Herein disclosed is a microphone-speaker apparatus, comprising: audio signal dividing means for dividing the audio signal into a plurality of raw component signals each indicative of the raw wave components; coherent component signal extracting means for extracting a plurality of coherent component signals respectively indicative of the coherent wave components from the raw component signals divided by the audio signal dividing means; power value calculating means for calculating the raw power value of each of the coherent component signals extracted by the coherent component signal extracting means; power value adjusting means for adjusting the raw power value of each of the coherent component signals calculated by the power value calculating means to produce an adjusted power value of each of the coherent component signals; power value judging means for judging whether or not the adjusted power value of each of the coherent component signals of the current frame exceeds the adjusted power value of each of t

Abstract: A circuit for adaptive suppression of acoustic feedback forms part of a digital hearing aid, comprising a microphone (1), subtracter (3), hearing correcting means (4), receiver (6), delay element (9), filter (10), updating unit (11), lattice decorrelators (12, 13) and control unit (14). The transmission path is modeled with the feedback characteristic (7) and an adder (8). First decorrelator (12) decorrelates the echo-compensated input signal (en) and second decorrelator (13) decorrelates the delayed output signal (xn) by using coefficients (kn) from first decorrelator (12). The coefficients (kn) of the two filters (12, 13) are calculated by adaptive decorrelation of the echo-compensated input signal (en). This permit maximum convergence rates for minimum distortions. Updating of the filter coefficients mainly takes place where the greatest amplifications occur in the hearing correcting means (4). The fed-back signal components are continuously removed from the input signal.

Abstract: Filter system designed to remove loudspeaker-generated sound signals from a microphone signal, comprising a stereo sampling unit for sampling the interference-bearing microphone signal as well as the interfering loudspeaker signal in time (t) to generate samples x(t) and z(t) for provision to a computation unit which derives the transfer friction H(f,T) effective between the loudspeaker and the microphone and applies that function to a filter unit which acts to restore the interference-free microphone signal y(t) from x(t), z(t) and H(f,T).

Abstract: A complementary comb filter includes a plurality of band pass filters and a plurality of notch filters, wherein the notch filters have the same center frequencies as the band pass filters or the same center frequencies as the dead bands defined by the band pass filters. Phase distortion by the notch filters occurs primarily in the notch, thereby eliminating distortion of the signal.

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 digital signal processing algorithm to cancel howling in a telephone circuit, the telephone circuit being characterized by a receive path and a transmit path in an effective closed loop configuration. The algorithm monitors input samples taken from the receive path and, if a howling signal is detected, controlled amounts of attenuation are introduced to the loop until howling is cancelled. In a preferred embodiment a prediction logic is included in the algorithm to verify the howling signal.

Abstract: A gain control method for reducing or eliminating the undesirable effects of acoustic echo coupling between a speaker and a microphone while providing a full-duplex connection. Weighted normalized far-end and near-end powers are used to first calculate a suppression value which is used to determine the attenuation factor between an upper and a lower limit, thus improving reliability in noisy environments. To further improve the quality of the full-duplex connection, a smoothing factor is applied to the attenuation calculation to provide a low power and constant sounding echo without annoying transient-like sounds.

Abstract: An anti-oscillation loudspeaker device where a reproduced sound is compensated based on a acoustic signal detected by a microphone and a reproduced band is enlarged is provided. In this loudspeaker device, a microphone is placed near a position where sound pressure of resonance occurring in a closed space for at least one of height, width, and depth of the inside of a cabinet is minimum. An influence of the resonance is restrained, a feedback circuit becomes stable, and a feedback amount increases.

Abstract: A system (100) for controlling noise comprises an array (2) of concelling tranducers (loudspeakers) (2a). Located some distance away from cancelling array (2) is a detection system (3) comprising a series of microphones (3a), the system casting an acoustic “shadow” or quiet region (4). Located on or adjacent the primary source (1) emitting the noise to be controlled is a synchronising sensor (5) which may be a microphone, vibration transducer or electrical transducer, the output of sensor (5) being fed, along with the output from detection array (3), into an adaptive control system (102) the output of which is fed back to the cancelling units (2). The adaptive control system (102) comprises a low pass filter (15) producing a dc component V from a mathematical convolver (multiplier) (14), and a digital oscillator (16) which generates the cancelling frequency which is controlled by V. Also included arc frequency multipliers (17, 18 and 19) which multiply (2, 3) and n times respectively.

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: The present invention teaches a system for improving the clarity of a voice spoken within an enclosed space. The system comprises a first microphone, positioned at a first location, for receiving the audible communication and for converting the audible communication at the first location into a first audio signal. The system also comprises a loudspeaker for receiving the first audio signal, and for converting the first audio signal into a first reproduced audible communication, the reproduced audible communication also being fed back and received by the first microphone and converted with the audible communication into the first audio signal.

Abstract: A wave-discriminating inductive electronic device includes a special integrated circuit that is inducible by a proximity feeble electromagnetic signal generated by a general mobile phone during receiving and transmitting signals, so that the induced integrated circuit is able to control and drive another integrated circuit that includes voice and motor driving circuits. Voices and vibrations produced by the integrated circuit enable the wave-discriminating inductive electronic device to provide a highly interesting entertaining effect.

Abstract: A microphone for a full duplex speaker phone comprises a loudspeaker, a uni-directional microphone, a omni-directional microphone, and a background noise suppression circuit. The uni-directional microphone is a directional microphone such that the intensity of an acoustic source detected at the uni-directional microphone is dependent on the angular direction of the acoustic source relative to a reference direction. The omni-directional microphone is a non-directional microphone such that the intensity of the acoustic source detected at the omni-directional microphone is substantially independent of the angular direction of the acoustic source relative to the reference direction. The background noise suppression circuit determines the direction of the acoustic source by calculating a quotient between a signal strength of the acoustic source as detected by the two microphones and provides for suppressing an acoustic source that is not within a threshold angle of the reference angle.

Abstract: The present invention is embodied in a method and apparatus for eliminating audio feedback which includes an active microphone coupled to an amplifier which transmits signals received at the active microphone to a parametric speaker to generate a feedback controlled signal for broadcasting. The apparatus may also be adapted for use with an audio system wherein the invention comprises (i) at least one transducer for detecting at least one sonic frequency and generating an electrical signal representative of the at least one sonic frequency, (ii) a processor for receiving the electrical signal and generating a first ultrasonic frequency which has been modulated with the at least one sonic frequency, (iii) a parametric demodulator for recovering the at least one sonic frequency from the first ultrasonic frequency, and (iv) a speaker for directly emitting the recovered sonic frequency as a feedback controlled audio emission.

Abstract: Herein disclosed is a base station and a base station comprising the base station, which does not limit the time period required for the processing performed by the signal processing section of the base station, but can prevent the talk back sound from being delayed and misheard for reverberations, thereby making it easy for an order taker to accurately take the order.

Abstract: The present invention is embodied in a method and apparatus for eliminating audio feedback which includes an active microphone coupled to an amplifier which transmits signals received at the active microphone to a parametric speaker to generate a feedback controlled signal for broadcasting. The apparatus may also be adapted for use with an audio system wherein the invention comprises (i) at least one transducer for detecting at least one sonic frequency and generating an electrical signal representative of the at least one sonic frequency, (ii) a processor for receiving the electrical signal and generating a first ultrasonic frequency which has been modulated with the at least one sonic frequency, (iii) a parametric demodulator for recovering the at least one sonic frequency from the first ultrasonic frequency, and (iv) a speaker for directly emitting the recovered sonic frequency as a feedback controlled audio emission.

Abstract: An earset including a housing positionable with respect to an ear of a person, a microphone disposed with respect to the housing for insertion into the ear of a person, the microphone operable to detect a change in air pressure within the ear while the person speaks and to produce an electrical microphone signal corresponding to the internally detected change in air pressure and a speaker disposed with respect to the housing and operable to produce a sound corresponding to an electrical speaker signal. The earset also includes a circuit coupled to receive the microphone signal and the speaker signal and operable to produce a corrected microphone signal having a reduced feedback component of the microphone signal, the feedback component resulting from the detection by the microphone of the sound produced by the speaker to produce a corrected microphone signal.

Abstract: A digital voice enhancement, DVE, communication system includes an instability detector detecting an unstable acoustic feedback condition from a loudspeaker to a microphone by sensing a condition of the electrical signal transmitted from the microphone to the loudspeaker, and a corrective processor responsive to the instability detector to modify the electrical signal to reduce unstable acoustic feedback. The sensed condition may be magnitude, power, or, preferably, the sinusoidal characteristic of the electrical signal, namely the electrical signal becoming sinusoidal in nature.

Abstract: An echo sound signal suppressing apparatus for suppressing an echo sound signal to ensure speech communication between a far-end speaker and a near-end speaker, comprising: first power value calculating means for calculating the power value of the second sound signal; second power value calculating means for calculating the power value of the echo replica sound signal produced by the echo replica sound signal producing means; echo sound suppressing means for suppressing the echo sound difference signal component outputted from the difference signal producing means to a minimum level with the power value of the second sound signal calculated by the first power value calculating means and the power value of the echo replica sound signal calculated by second power value calculating means with the near-end speaker sound signal component being allowed to pass therethrough, the echo sound suppressing means having an output terminal to the communication line to output the near-end speaker sound signal component unde

Abstract: A howling canceling apparatus is provided in a sound system containing a microphone, a loudspeaker and an amplifier for canceling howling which may occur by feedback of sound from the loudspeaker to the microphone. In the howling canceling apparatus, a measuring section measures an impulse response of the sound system to determine a time length of a decay portion of the impulse response. A detecting section detects an occurrence of the howling when the determined time length is longer than a predetermined reference time length, and further analyzes a frequency spectrum of the decay portion of the impulse response to determine a frequency point at which the howling occurs. An attenuating section attenuates a frequency component of the sound around the determined frequency point so as to cancel the howling.

Abstract: In order to reduce acoustic echo and restore stereophonic sound in a digital transmission system of sound signals on multiple transmission channels, between a local site and one or several remote sites, comprising several microphones and loudspeakers, the method consists in the following steps: computing a weighting factor to be applied to each microphone signal before emitting, based on two attenuation factors computed from the microphone and loudspeaker signals; transmitting on each transmission channel the linear combination of weighted microphone signals; computing a weighting factor to be applied to each loudspeaker signal; and emitting on the loudspeaker the linear combination of transmitted, weighted signals.

Abstract: A sound collector comprises switch means (55) for selecting one of the microphones (50a, 50b) directed to sound collection objects (60, 65) the angle between which is &thgr;. The microphone is an optical microphone comprising a diaphragm (2), a light source (3) for irradiating the diaphragm (2), a photodetector (5) outputting a vibration signal of the diaphragm (2), a light source driving circuit (13), and a negative feedback circuit (100) for supplying the output from the photodetector (5) to the light source driving circuit (13). The difference of the sensitivity between the sound collecting objects is eliminated by selecting one of the microphones through the switch means (55) and thereby changing the amount of negative feedback.

Abstract: In a digital voice enhancement communication system, a plurality of individually customized equalization filters are provided, one for each electrical path between a respective microphone and loudspeaker and custom tailored to the respective electroacoustic transfer function therebetween. Increased gain is enabled in a digital voice enhancement communication system by an equalization filter filtering a respective electrical signal from a microphone to a loudspeaker in inverse relationship to the electroacoustic transfer function between the microphone and loudspeaker.

Abstract: In a method for feedback recognition in a hearing aid and a hearing aid operating according to the method, a frequency band is defined, a first signal level in the frequency band is determined, the signal on a signal transmission path of the hearing aid is attenuated, and a second signal level of the attenuated signal in the frequency band is determined, and feedback is recognized on the basis of the identified first and second signal levels.

Abstract: The feedback caused between the output (39) and the input (30) of an amplification path (54, 42, 56, 60, 62, 64) is reduced by providing a delay (60) in the amplification path, passing through the amplification path a signal having an auto-correlation function which is substantially a delta function, correlating (72) the said signal before being delayed in the delay with the signal after being delayed in the delay to produce a plurality of correlation coefficients, modifying the signal in the amplification path to provide a modified signal, the modification being provided by a transversal filter (76) controlled by the said plurality of correlation coefficients, and combining (42) the modified signal with the signal in the amplification path so as to reduce the effect of the feedback. The signal having an auto-correlation function which is substantially a delta function may be an added noise signal (70) or may be constituted by the signal being processed itself.

Abstract: There is disclosed an audio terminal (10) and a method for operating in uncontrolled audio environments, the audio terminal (10) having an echo suppression unit (20) for reducing acoustic feedback (18). The echo suppression unit (20) includes a learner (22) for learning an audio environment of the audio terminal (10) and a control unit (21) for controlling the acoustic feedback (18) in accordance with the audio environment of the audio terminal (10).

Abstract: A microphone array apparatus includes a microphone array including microphones, one of the microphones being a reference microphone, filters receiving output signals of the microphones, and a filter coefficient calculator which receives the output signals of the microphones, a noise and a residual signal obtained by subtracting filtered output signals of the microphones other than the reference microphone from a filtered output signal of the reference microphone and which obtain filter coefficients of the filters in accordance with an evaluation function based on the residual signal.

Abstract: There is provided a method of controlling feedback in an acoustic system, for example a digital hearing aid, in which there is a potential feedback path between the output and the input. The method comprises making a spectral estimate of the input signal spectrum, and then subjecting the spectral estimate to a psycho-acoustic model to generate a control signal. A noise source is passed through a shaping filter, which is controlled with the control signal, to generate frequency-shaped noise, which is inaudible to someone hearing the output. The frequency-shaped noise is then added to the input signal to form a combined signal, which is processed in a forward path, to generate a first output signal. The first output signal and the frequency-shaped noise signal are analyzed, to determine the presence of feedback at difference frequencies, and the characteristics of the forward path are modified to reduce the gain thereof at frequencies where feedback is detected.

Abstract: A low pass limited, low frequency pseudo-random noise signal is added to the input signal for tagging the input signal in order to facilitate correlation with an echo.

Abstract: The feedback caused between the output and the input of an amplification path is reduced by providing a delay in the amplification path, passing through the amplification path a signal having an auto-correlation function which is substantially a delta function, correlating the said signal before being delayed in the delay with the signal after being delayed in the delay to produce a plurality of correlation coefficients, modifying the signal in the amplification path to provide a modified signal, the modification being provided by a transversal filter controlled by the said plurality of correlation coefficients, and combining the modified signal with the signal in the amplification path so as to reduce the effect of the feedback.

Abstract: Sound is converted into an electrical signal by a microphone and is converted into an inaudible, digitally modulated signal that is combined with the electrical signal from the microphone, amplified, and converted into sound waves by a speaker. Any sound traveling from the speaker back to the microphone includes the inaudible component representing the original sound. The inaudible component is separated from the audible components, and the original sound is reconstructed in a digital demodulator. The reconstructed original sound is subtracted from the signal from the microphone, thereby reducing any echo and canceling feedback. Digital modulation includes any form of shift keying, including coherent and noncoherent techniques for modulating frequency, phase, or amplitude.