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: Various embodiments of the present invention are directed to methods and systems that reduce acoustic echoes in audio signals in accordance with changing conditions at first and second locations that are linked together in a communication system. In particular, one embodiment of the present invention relates to a method for determining an approximate impulse-response vector for canceling an acoustic echo resulting from an audio signal transmitted from the first location to the second location. This method includes forming a trust region within a search space based on computing a recursive specification vector defining the trust region. The method also includes computing a recursive shadow-impulse-response vector that lies substantially within the trust region, and computing the approximate impulse-response vector based on the recursive shadow-impulse-response vector and the recursive specification vector.

Abstract: A test system for evaluates acoustic feedback characteristics of a listening device by changing the transfer function of the signal path from an acoustic output to an acoustic input of the listening device. A variable filter comprises an acoustic input, an acoustic output, and a control unit for changing a transfer function of the filter. First and/or second acoustic propagation elements propagate acoustic sound, the first acoustic propagation element acoustically connecting the acoustic output of the listening device with the acoustic input of the variable filter, the second acoustic propagation element acoustically connecting the acoustic output of the variable filter with the acoustic input of the listening device; wherein the system is configured to allow a determination of acoustic feedback or a level of acoustic feedback of the listening device at different frequencies.

Abstract: An electronic device comprises a microphone, a transceiver circuit, a loudspeaker, a sense element and a processing circuit. The microphone is configured to receive a first audio signal. The transceiver circuit is configured to communicate the first audio signal to a remote device and to receive a second audio signal from the remote device. The loudspeaker is configured to provide an audible signal based on the second audio signal. The sense element is configured to sense the audible signal provided by the loudspeaker. The sense element may be positioned at a distance from the loudspeaker different than a distance between the microphone and the loudspeaker. The processing circuit is configured to process at least one of the first audio signal and the second audio signal based on a sensed signal from the sense element.

Abstract: An acoustic feedback suppression apparatus includes an amplitude measurement section for measuring the amplitude of a sound signal input from a microphone, an amplitude attenuation section for attenuating the amplitude of a sound signal output to a speaker, and an acoustic feedback determination section for determining whether or not acoustic feedback is occurring based on the sound signal input from the microphone. If it is determined that acoustic feedback is occurring, a sound-deadening control section enables sound-deadening control so that the amplitude of the sound signal output to the speaker is attenuated to a sound-deadening level. A sound-deadening release section releases the sound-deadening control if the amplitude of the sound signal input from the microphone is more than specified threshold amplitude and if it is determined that acoustic feedback is not occurring while the sound-deadening control is conducted.

Abstract: Improving the acoustical communication between interlocutors in at least two positions in a room includes generating electrical signals representative of acoustical signals present at the respective interlocutor positions, amplifying each of the electrical signals and converting the amplified electrical signals into acoustical signals. A time delay is applied to the electrical signals such that the acoustical signal arriving first at one of the interlocutor positions originates from the direction of the other interlocutor position.

Abstract: Electronics for altering the audio frequency response of a micro-speaker without modifying the micro-speaker itself are used to provide a selected frequency response of the micro-speaker. The micro-speaker has a resonant peak region, from which the response declines for both higher and lower frequencies. In one embodiment the electronics includes a first circuit for modifying the frequency response curve up to the resonant peak region, and a second circuit for modifying the frequency response curve for audio frequencies higher than this region. Each filter yields an integer multiple of 6 dB per octave slope. In the described embodiment, for approximately correcting presbyacusis (age related hearing loss), a set of high pass filters, low pass filters, and/or high order filters are connected to the micro-speaker to progressively attenuate the frequency response curve as the frequency decreases from 10000 Hz to 100 Hz.

Abstract: The present invention discloses a wireless telephone system using microphone arrays together with additional signal processing to suppress the background noise in the surrounding environment. The signal processing resources of a wireless telephone and multi-channel transmission capabilities of the Bluetooth transmission are used to suppress the background noise. The wireless telephone system includes a Bluetooth transceiver communicating to a wireless telephone through a multi-channel Bluetooth transmission, and an array of microphones coupled to the Bluetooth transceiver. The array of microphones includes a first microphone producing a first audio signal output and a second microphone producing a second audio signal output. The first audio signal output and second audio signal output are transmitted to the wireless telephone through the first channel and second channel of multi-channel Bluetooth transmission respectively.

Abstract: The invention relates to a hearing instrument for processing an input sound to an output sound according to a user's needs. The invention further relates to a method of operating a hearing instrument and to use of a hearing instrument. The object of the present invention is to provide an alternative scheme for handling acoustic feedback in a hearing instrument.

Abstract: A method and device for transforming ambient audio are provided. Example embodiments may include monitoring ambient audio proximate to a sound processing device located in an environment. The device may analyze the ambient audio to derive one or more first characteristics. Memory may be accessed to obtain at least one transformation audio from a number of stored transformation audio. Each stored transformation audio may have one or more associated characteristics. A characteristic associated with the transformation audio may match one or more of the first characteristics. The device may generate output transformation audio based on the one or more transformation audio to provide modified output audio for propagation into the environment.

Abstract: A method and device for transforming ambient audio are provided. Example embodiments may include monitoring ambient audio proximate to a sound processing device located in an environment. The device may access memory to obtain transformation audio and generate output transformation audio based on the transformation audio and the ambient audio to provide modified output audio for propagation into the environment. The device may at least reduce feedback of the modified output audio received by the sound processing device from the environment.

Abstract: An audio processing apparatus is provided, comprising: a main microphone for receiving sounds from a source and noises from non-source sources and generating a main input; a reference microphone for receiving the sounds and the noises and generating a reference input; a short-time Fourier transformation (STFT) unit for applying short time Fourier transformation to convert the main input of a time domain signals into a main signal of a frequency domain and convert the reference input of the time domain signals into a reference signal of the frequency domain; a sensitivity calibrating unit for performing sensitivity calibration on the main signal and the reference signal and generating a main calibrated signal and a reference calibrated signal; and a voice active detector (VAD) for generating a voice active signal according to the main calibrated signal, the reference calibrated signal and a direction of arrival (DOA) signal.

Abstract: Provided is a dereverberation system or the like which copes with an arbitrary condition flexibly and is capable of recognizing a sound or a sound source signal. According to the dereverberation system, an inverse filter (h) is set by using a pseudo-inverse matrix (R+) of a non-square matrix (R) as a correlation matrix of input signals (x). On the basis of the inverse filter (h) and an estimated correlation matrix (R^) generated according to a window function (w), an error cost (J(h) between a correlation value of the input signals (x) and output signals (y) and a desired correlation value (d) is calculated. On the basis of the error cost (J(h)), the inverse filter (h) is adaptively updated according to a gradient method.

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 system and method for analyzing a signal to monitor the dynamics of its magnitude and frequency characteristics over time. An electronic circuit for identifying feedback in an audio signal, formed in accordance with embodiments of the invention may comprise a feedback control block operable to determine a candidate frequency having potential feedback such that the feedback control block is further operable to perform an iterative analysis of the magnitude of the audio signal at the candidate frequency to determine the growth characteristics of the signal. The electronic circuit may further include a test filter block operable to deploy a test filter at a candidate frequency and a permanent filter block operable to deploy a permanent filter at the candidate frequency if the feedback control block determines that the growth characteristics of the signal at the candidate frequency comprises feedback characteristics after the test filter has been deployed.

Abstract: A configuration and associated methods are used for detecting acoustic feedback in a hearing device. One embodiment contains a first feedback detection unit, which determines the probability of feedback, a second feedback detection unit, which determines a weighting factor, and an arithmetic unit, which multiplies the feedback probability by the weighting factor. As an alternative to determining the weighting factor, a threshold value may also be controlled. This offers the advantage of improved acoustic feedback detection by a combination of two different feedback detection methods.

Abstract: Provided are a method and apparatus for reproducing sound. A sound signal may be generated by receiving sound around the apparatus. An output signal may be generated by performing a process according to functions of the apparatus on a residual signal obtained by canceling a feedback signal from the sound signal.

Abstract: An oscillation-echo preventing circuit has a microphone/speaker unit and a voltage-canceling circuit for canceling voltages of audio receive signals. The microphone/speaker unit has a main body, at least two microphones, and a speaker or an earphone. The microphone seals a first inside space from an outside space. The microphone seals the first inside space from a second inside space. The speaker or the earphone seals the first inside space from the outside space. The voltage-canceling circuit 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: A system and method for analyzing a signal to identify one or more candidate frequencies that may have a potential feedback problem. An electronic circuit for identifying feedback in an audio signal, formed in accordance with embodiments of the invention, may comprise a feedback control block operable to: 1) determine a magnitude of a first dominant spectral component in a wide-band span of frequencies of a signal, 2) determine a magnitude of a second dominant spectral component in a mid-band span of frequencies of the signal, and 3) determine a magnitude of a third dominant spectral component in a narrow-band span of frequencies of the signal. The feedback control block may then further analyze the magnitude and frequency of the first, second, and third dominant spectral components to determine the presence of feedback characteristics in the signal.

Abstract: Audio processing systems, such as full duplex digital telephone systems, are provided in which analog automatic level control (ALC) circuitry is employed to automatically control and stabilize the microphone output audio signal levels to a constant level for optimal digital signal processing. For example, an audio communication device (10) includes a microphone (11) to generate an audio signal, an analog automatic level control (ALC) circuit (13), coupled to an output of the microphone (11), to control a level of the audio signal output from the microphone (11) and output a level-controlled audio signal, and a DSP (digital signal processing) circuit (14) to process the level-controlled audio signal output from the ALC circuit (13). The ALC (13) allows audio signals to be captured by the microphone (11) with wide dynamic range, while stabilizing the microphone output level within a reduced dynamic range compatible with the limited dynamic range of the DSP circuitry (14) for enhanced performance.

Abstract: A method, medium, and apparatus for extracting a target sound from mixed sound. The method includes receiving a mixed signal through a microphone array, generating a first signal whose directivity is emphasized toward a target sound source and a second signal whose directivity toward the target sound source is suppressed based on the mixed signal, and extracting a target sound signal from the first signal by masking an interference sound signal, which is contained in the first signal, based on a ratio of the first signal to the second signal. Therefore, a target sound signal can be clearly separated from a mixed sound signal which contains a plurality of sound signals and is input to a microphone array.

Abstract: An anti-feedback device includes an anti-feedback filter provided in a closed loop. The anti-feedback device down-samples a signal of specific band selected from an output signal of an adaptive target signal transfer system and a signal of the same band selected from an input signal of the transfer system, and a filtering coefficient of the adaptive filter is updated by use of the down-sampled signals. The filter controller controls a filtering characteristic of the anti-feedback filter so that a peak gain of a frequency of an amplitude characteristic within a specific band of a closed loop determined from the filtering coefficient of the adaptive filter is suppressed. Moreover, the filter controller estimates a gain of the closed loop outside the specific band from the amplitude characteristic in the specific band and controls the amount of suppression of the anti-feedback filter outside the band in accordance with a result of estimation.

Abstract: Harmonic distortion residual echo suppression (HDRES) technique embodiments are presented which act to suppress the residual echo remaining after a near-end microphone signal has undergone AEC, including harmonic distortion in the signal that was caused by the speaker audio signal playback. In general, an AEC module is employed which suppresses some parts of the speaker audio signal found in a near-end microphone signal and generates an AEC output signal. A HDRES module then inputs the AEC output signal and the speaker audio signal, and suppresses at least a portion of a residual part of the speaker audio signal that was left unsuppressed by the AEC module. This includes at least a portion of the harmonic distortion exhibited in the AEC output signal.

Abstract: The sound field reproducing apparatus includes: a signal obtaining unit which obtains input audio signals of plural channels; an effect adding unit which gives an effect on an output to a sound field; and an effect amount control unit which controls an amount of the effect based on a characteristic of the input audio signal. The sound field reproducing apparatus receives input audio signals of plural channels from a medium such as a DVD, and gives a sound effect to them to output. The amount of the effect given to the input audio signal is controlled based on the characteristic of the input audio signal, especially the cross correlation of the reverberation component between the channels.

Abstract: A transceiver includes an analog front end (AFE) device and first, second, third, and fourth transmitters. A digital signal processor (DSP) receives digital receive signal and digital transmit signals and generates a shared error control signal and first, second, third, and fourth individual error control signals. An echo canceller system generates an estimated echo signal. First, second, and third NEXT canceller systems generate first, second, and third estimated NEXT signals, respectively, each based on the second analog receive signal and respective ones of the analog transmit signals from the second, third, and fourth transmitters during the first mode and each based on the shared error control signal, respective ones of the second, third, and fourth individual error control signals, and the respective ones of the analog transmit signals from the second, third, and fourth transmitters during the second mode.

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: A system of signal processing an input signal in a hearing assistance device to avoid entrainment wherein the hearing assistance device including a receiver and a microphone, the method comprising using an adaptive filter to estimate an acoustic feedback path from the receiver to the microphone, generating one or more estimated future pole positions of a transfer function of the adaptive filter, analyzing stability of the one or more estimated pole positions for an indication of entrainment and adjusting the adaptation of the adaptive filter based on the stability.

Abstract: Disclosed herein are conferencing products implementing an acoustic echo cancellation system that utilizes converging coefficients and a detector of turn-off and/or turn-on events of push-to-talk microphones, and, further, that mitigates against divergence and/or drift of coefficients and other variables of an echo canceller. A push-to-talk detector may be used that includes a high-pass filter, a transient detector, or an adjustable high-pass filter. An echo canceller may be disabled as to a push-to-talk microphone that has been turned off.

Abstract: A conference system for independently adjusting audio parameters is provided, including a microphone module, a storage module, a digital signal processing module and an amplifier. When any microphone of the microphone module generates an audio signal transmitted to the digital signal processing module, the priority controller determines the priority order of the audio signals entering the digital signal processing module, the digital signal processing module will read the corresponding adjustment parameters stored in the storage module, and the volume controller, equalizer, feedback suppressor perform adjustment on the audio signal. Finally, the mixer mixes the audio signals and outputs to the amplifier. The amplifier amplifies the mixed audio signal and outputs to speakers. In this manner, a single digital signal processing module can adjust different audio signals from different microphones based on different adjustment parameters, without affecting the audio signal from other microphones.

Abstract: An audio device includes a first audio path with a loudspeaker for reproducing an audio signal, and a second audio path. The second audio path includes in series a band-pass filter for filtering an audio signal, a detector for detecting the amplitude of the band-pass filtered audio signal, a multiplier for multiplying a periodic signal by the amplitude of the band-pass filtered audio signal, and a vibration device for reproducing the multiplied periodic signal. The frequency of the periodic signal is substantially equal to the resonance frequency of the vibration device.

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. The invention further provides a noise statistics component that stores noise statistics associated with a noise portion of an input signal and a signal+noise statistics component that stores signal+noise statistics associated with a signal and noise portion of the input signal.

Abstract: A noise eliminating apparatus includes a first microphone, a second microphone and a signal processing unit, the signal processing unit includes a linear prediction filter and a noise resynthesis filter, the linear prediction filter receives an output signal of the first microphone, predicts the output signal of the first microphone by linear prediction and generates a prediction signal, and the noise resynthesis filter is an adaptive filter which receives, as a main input signal, a first difference signal obtained by subtracting one of the output signal of the first microphone and the prediction signal from the other, receives, as an error signal, a second difference signal obtained by subtracting one of an output signal of the second microphone and an output signal of the noise resynthesis filter itself from the other, and updates a filter coefficient so that the error signal is minimized.

Abstract: A feedback reduction system adaptively processes a microphone signal. An adaptive feedback suppression filter circuit processes the microphone signal, and an adaptive feedback compensation filter circuit further processes the microphone signal. The adaptive feedback suppression filter circuit processes the signal based upon output from the adaptive filter compensation filter circuit.

Abstract: A microphone assembly for desktop communication systems utilizes a directional microphones in a desktop conferencing system without exposing the microphone to unfavorable mechanical or acoustic influence. The microphones is built into the front portion of the base of the system, in a mechanically controlled and robust way. The microphone assembly maximizes microphone sensitivity in the direction of a near end user while simultaneously minimizing microphone sensitivity in the direction of the loudspeaker.

Abstract: Methods and systems for masking audio noise are disclosed. One apparatus includes a silence detector configured to detect a period of substantial silence in an audio signal; a masking noise source operably coupled to the silence detector, the masking noise source configured to generate a noise signal in response to the silence detector detecting the period of substantial silence; and at least one combining device operably coupled to the masking noise source, the at least one combining device configured to contribute to combining the audio signal and the noise signal. A method includes detecting a period of substantial silence in an audio signal; and combining masking noise with the audio signal during the period of substantial silence.

Abstract: An acoustic echo cancellation device for canceling an echo in a microphone signal in response to first and second input signals includes a first combination unit for combining the first and second input signals into an aggregate input signal. The device further includes an adaptive filter unit for filtering the aggregate input signal so as to produce an aggregate echo cancellation signal. A second combination unit combines the aggregate echo cancellation signal with the microphone signal so as to produce a residual signal, and an additional filter unit filters either the first or second input signal so as to produce a first or a second partial echo cancellation signal. A post-processor unit suppresses remaining echo components in the residual signal. The post-processor unit uses at least one partial echo cancellation signal to suppress echo components corresponding with the first input signal to a greater extent than echo components corresponding with the second input signal.

Abstract: It is an object of the present invention to provide a howling suppressing apparatus and an acoustic apparatus which can suppress the howling component effectively without suppressing a periodic component other than the howling component. The howling control apparatus comprises a signal separating unit 23 operable to separate an input signal into two components including a periodic component and a nonperiodic component, a howling detecting unit 24 operable to detect a howling component from the input signal on the basis of signals selected from among the input signal and periodic and nonperiodic component signals from the signal separating unit 23, a howling suppressing unit 25 operable to suppress the howling component of the input signal on the basis of detection information from the howling detecting unit 24, and a signal synthesizing unit 26 operable to synthesize a signal from signals from the howling suppressing unit 25.

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: An audio signal is supplied to a loudspeaker array to perform wavefront synthesis. A virtual sound source is produced at an infinite distance using wavefront synthesis. A propagation direction of a sound wave emitted from the virtual sound source is changeable.

Abstract: An echo compensation system may remove undesirable audio signals. The echo compensation system may utilize adaptive filters to remove echoes and undesirable signals received by a microphone. The echo compensation system may inhibit adaptation of an adaptive filter when left and right channel audio signals are highly correlated. In a two-channel system, inhibiting adaptation of one of two adaptive filters may reduce computational power requirements, while still removing undesirable signals. In a four-channel system, inhibiting adaptation of all but one of the four adaptive filters may reduce computational power requirements by a greater percentage.

Abstract: A system comprising converter logic that introduces noise to a signal. The system also comprises amplifier logic that reduces a sound pressure associated with the noise by amplifying the signal prior to providing the signal to the converter logic and de-amplifying the signal after providing the signal to the converter logic.

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: There is described a method for operation of a hearing device system with two microphones arranged spatially separated from one another and with sound-generating output units assigned to these microphones, in which, by comparison of the microphone signals or of signals derived therefrom, feedback is detected, and on detection of the feedback measures are initiated for reducing the feedback, and with the comparison of the microphone signals or of signals derived therefrom comprising a frequency-selective power comparison. There is also described a hearing device system suitable for this method.

Abstract: A system for reducing an echo and/or a residual echo in a microphone output signal includes an echo compensation filter configured to receive audio input signals and generate an estimated echo signal. A speech activity detector is configured to detect speech of a local speaker, and a combining circuit is configured to generate an echo compensated signal by subtracting the estimated echo signal from the microphone output signal. A residual echo reduction circuit is configured to suppress the residual echo in the echo compensated signal based on the detected speech activity, and output an echo suppressed output signal. The echo reduction system may reduce local background noise and residual echoes in a microphone output signal so that local speech may be transmitted free from undesirable signals.

Abstract: A system and method to detect and remediate unacceptable levels of speech intelligibility evaluates received test audio transmitted across and received in a space or region of interest. Intelligibility is improved by altering the rate, pitch, amplitude and frequency bands energy during presentation of the speech signal.

Abstract: Disclosed herein are detectors of audio ringing feedback, that is decaying feedback with a gain of less than one, those detectors utilizing a repeated gain measurement that applied to a range of gain values characteristic of ringing-type feedback. Those gain measurements, while in the range, increase a probability measurement of feedback. When the probability of feedback reaches a threshold, a detection of feedback is made and feedback countermeasures, such as the application of a notch filter, may be applied. Optionally, the audio gain around likely frequencies of feedback may be enhanced for a time to increase the resolution of identification of a feedback frequency, which may be identified through an interpolative method. Repeated gain measurements may also identify building-type feedback. A ringing detector may include more than one range of detection, for example for building, strong-ringing and weak-ringing feedback.

Abstract: A system and method to detect and measure remediated speech intelligibility by evaluating received test audio transmitted across and received in a space or region of interest. Remediation of the test audio may include altering the rate, pitch, amplitude and frequency bands energy during presentation of the speech signal.

Abstract: A sound field controlling apparatus for a public-address system comprises a microphone that picks up a sound of a speaker, a loudspeaker that sound a sound signal based on the sound picked up by the microphone, a sound source position detector that detects a position of a sound source, and a signal processor that controls a level, delay time and equalizing property of the sound signal output to the loudspeaker in accordance with the sound source position detected by the sound source position detector.

Abstract: An equalization system enhances the quality of communications between a remote party and a local party. The equalization system includes an equalization filter that equalizes an acoustic signal received from the remote party. The equalized acoustic signal is transmitted to a speaker based on the equalized acoustic signal. A device converts sound into electrical signals. The electrical signals are transmitted to an echo compensation filter that compensates for reflected sound. Filter characteristics of the equalization filter are based on filter characteristics of the echo compensation filter.

Abstract: Disclosed herein are detectors of audio ringing feedback, that is decaying feedback with a gain of less than one, those detectors utilizing a repeated gain measurement that applied to a range of gain values characteristic of ringing-type feedback. Those gain measurements, while in the range, increase a probability measurement of feedback. When the probability of feedback reaches a threshold, a detection of feedback is made and feedback countermeasures, such as the application of a notch filter, may be applied. Optionally, the audio gain around likely frequencies of feedback may be enhanced for a time to increase the resolution of identification of a feedback frequency, which may be identified through an interpolative method. Repeated gain measurements may also identify building-type feedback. A ringing detector may include more than one range of detection, for example for building, strong-ringing and weak-ringing feedback.