Abstract: An integrated system for hearing protection and enhancement having earpieces including microphones and receivers to provide for user control of ambience level (mic gain), monitor input, control over limiting of excessive levels, and frequency equalization, all with very wide input dynamic range capability and controlled output dynamic range. The user control comprises a wired or wireless smartphone app. Also, the wide input dynamic range is achieved by placement of limiter circuitry capable of handling the entire input dynamic range prior to the most dynamic range limited circuit blocks: digital EQ block, output amplifier, earpiece receiver, and the users' ears. Switches allow combining left and right signals to be directed to a user's favored ear, if necessary.

Abstract: Disclosed herein are system, method, and computer program product embodiments for replicating a remote venue in a local venue. An embodiment operates by receiving original audio. Thereafter, the original stream of audio is modified to produce modified audio based on an audio profile unique to a remote venue smaller than the local venue. The audio profile comprises a virtual representation of the remote venue including (i) a virtual audio source corresponding to the local venue's audio source and configured to produce the original audio and (ii) a virtual reverberation point corresponding to a real-world location in the remote venue and the portion of the local venue. As such, the modified audio is determined based on receipt of the original audio from the reverberation point. Thus, after correlating the modified audio to the local venue's audio source, the modified audio is provided by the audio source to the portion of the local venue.

Abstract: In one implementation, a method for providing security on an externally connected controller includes launching, by the controller, a security layer that includes a whitelist of permitted processes on the controller, the whitelist including (i) signatures for processes that are authorized to be executed and (ii) context information identifying permitted controller contexts within which the processes are authorized to be executed; determining, by the security layer, whether the particular process is permitted to be run on the controller based on a comparison of the determined signature with a verified signature for the particular process from the whitelist; identifying, by the security layer, a current context for the controller; determining by the security layer, whether the particular process is permitted to be run on the controller based on a comparison of the current context with one or more permitted controller contexts for the particular process from the whitelist.

Abstract: An audio device including a housing having a side facing outwardly of the housing, the housing further having a first mounting position disposed on the side. The side has a first portion and a second portion, the first portion opposite the second portion relative to the first mounting position. The audio device also includes a first omnidirectional microphone disposed on the first portion. The audio device also includes a second microphone disposed on the first mounting position. The second microphone may be a second-order microphone. The second-order microphone has a first-order microphone disposed within a first platform mounted a distance from the first mounting position. The audio device also includes a first speaker disposed on the second portion. The first speaker is further disposed such that an audio output of the first speaker falls within a null of the second microphone.

Abstract: An audio system includes: a speaker; a microphone that generates a microphone signal based on sound output from the speaker; a mixer module configured to generate a mixed signal by mixing the microphone signal with an audio signal; a filter module configured to filter the mixed signal to produce a filtered signal and to apply the filtered signal to the speaker; and a detector module configured to determine a howling frequency in the microphone signal attributable to sound output from the speaker, where the filter module is configured to decrease a magnitude of the filtered signal at the howling frequency.

Abstract: A sound signal processing method includes: receiving a line-inputted sound signal; controlling a volume of the line-inputted sound signal; and generating an early reflected sound control signal using the line-inputted sound signal having the controlled volume.

Abstract: A method of suppressing wind noise of a microphone and/or an electronic device are disclosed. The method of suppressing wind noise of a microphone includes receiving an audio signal, obtaining a frequency spectrum of the audio signal and a power spectrum of the audio signal, determining a wind noise power spectrum of the audio signal based on the power spectrum, determining a wind noise suppression gain based on the wind noise power spectrum and the power spectrum, correcting the frequency spectrum according to the determined wind noise suppression gain, and converting the corrected frequency spectrum into a time domain to obtain a corrected audio signal.

Abstract: A playback device is configured to receive, via a network interface, a source stream of audio including first and second channel streams of audio, and to produce, via respective first and second speaker drivers, a first channel audio output and a second channel audio output. The playback device is also configured to receive, via one or more microphones, a captured stream of audio including first and second portions corresponding to the respective first and second channel audio outputs. The playback device is also configured to combine at least the first channel stream of audio and the second channel stream of audio into a compound audio signal and perform acoustic echo cancellation on the compound audio signal and thereby produce an acoustic echo cancellation output, then to apply the acoustic echo cancellation output to the captured stream of audio and thereby increase a signal-to noise ratio of the captured stream of audio.

Abstract: A method for detecting a prominent tone of an input audio includes establishing a first analysis audio signal based on the input audio signal, establishing a second analysis audio signal based on the input audio signal, wherein an analysis audio signal of the first analysis audio signal and the second analysis audio signal is established by applying an analysis audio filter to the input audio signal, comparing the first analysis audio signal and the second analysis audio signal to obtain an energy level contrast, and determining a representation of the prominent tone by converting the energy level contrast by a contrast-to-frequency mapping function.

Abstract: An audio processing method may involve receiving media input audio data corresponding to a media stream and headphone microphone input audio data, determining a media audio gain for at least one of a plurality of frequency bands of the media input audio data and determining a headphone microphone audio gain for at least one of a plurality of frequency bands of the headphone microphone input audio data. Determining the headphone microphone audio gain may involve determining a feedback risk control value, for at least one of the plurality of frequency bands, corresponding to a risk of headphone feedback between at least one external microphone of a headphone microphone system and at least one headphone speaker and determining a headphone microphone audio gain that will mitigate actual or potential headphone feedback in at least one of the plurality of frequency bands, based at least partly upon the feedback risk control value.

Abstract: Technology described in this document can be embodied in an earpiece of an active noise reduction (ANR) device. The earpiece includes a plurality of microphones, wherein each of the plurality of microphones is usable for capturing ambient audio to generate input signals for both an ANR mode of operation and a hear-through mode of operation of the ANR device. The earpiece further includes a controller configured to: process a first subset of microphones from the plurality of microphones to generate input signals for the ANR mode of operation, process a second subset of microphones from the plurality of microphones to generate input signals for the hear-through mode of operation, detect that a particular microphone of the second subset is acoustically coupled to an acoustic transducer of the ANR device in the hear-through mode of operation, and in response to the detection, process the input signals from the second subset of microphones without using input signals from the particular microphone.

Abstract: An electrical device for reducing noise, comprises a first microphone configured to receive soundwave from a sound source and convert the soundwave to a first electrical signal including a noise component, a second microphone configured to receive ambient noise from an ambient environment and convert the ambient noise to a second electrical signal. The second electrical signal is reversed in polarity to the first electrical signal. The electrical device further comprises a circuit connecting the first microphone and the second microphone. The circuit is configured to combine the first electrical signal and the second electrical signal in order to reduce the noise component in the first electrical signal with the second electrical signal that is reversed in polarity.

Abstract: Methods and systems for assisting tonally-challenged singers. A microphone can be integrated with a sound reinforcement system used in a live performance. The microphone, which can transduce the performer's voice, can serve multiple purposes such as, for example, to feed input to the natural ear and to the sound reinforcement system. The processed sound of the performer's voice (with fundamental frequencies emphasized) can be mixed into the signal fed to a stage “monitor” speaker facing the performer or a headset worn by the performer.

Abstract: An apparatus includes a housing having a printed circuit board having a top surface and a bottom surface. The top surface of the printed circuit board forms an outer surface of the housing. The apparatus further includes a bottom-port microphone sensor mounted on the bottom surface of the printed circuit board. The printed circuit board has a port opening formed therein to provide an acoustic path from outside of the housing to the microphone sensor. A method of detecting ultrasonic signals includes receiving ultrasonic signals within a port opening of a printed circuit board forming part of a surface of a housing, and directing ultrasonic signals to a microphone sensor secured to a printed circuit board through the port of the printed circuit board.

Abstract: An audio system and method for eliminating audio feedback including a first audio input arranged to receive a first audio signal, a first audio output arranged within a first predefined zone, the first audio output arranged to receive the first audio signal, a second audio output arranged within the first predefined zone, the second audio output arranged to receive the first audio signal, and one or more processors connected to a virtual matrix including a plurality of virtual channels connecting the first audio input, the first audio output, and the second audio output, and the one or more processors arranged to receive the first audio signal and attenuate or eliminate the first audio signal to the first audio output or the second audio output if an audio feedback is detected.

Abstract: An operation method of a terminal may include: executing a voice call service between the terminal and at least one terminal; determining at least one short-range terminal existing within a preset range from among the at least one terminal, based on location information of the terminal; detecting a howling frequency band in which howling occurs between the terminal and the at least one short-range terminal from among a plurality of frequency bands in which the voice call service is performed, based on information of the plurality of frequency bands; and removing the howling by adjusting a gain of the howling frequency band.

Abstract: A digital signal processor for audio includes: an interface circuit communicating with an microcomputer; a multi-band equalizer including digital filters corresponding respectively to bands, wherein the frequency characteristic of each digital filter is changed according to coefficient group; a memory; and a coefficient setting circuit setting the coefficient group of each digital filter, wherein the frequency characteristic of the multi-band equalizer is selected from presets, wherein, when the digital signal processor start, the interface circuit receives setting data corresponding to the presets from the microcomputer and stores the received setting data in the memory, and wherein, at the time of audio reproduction, the interface circuit receives selection data to designate one of the plurality of presets and the coefficient setting circuit sets the coefficient group of each digital filter according to one of the setting data stored in the memory, the one setting data corresponding to the selection data.

Abstract: A method and system for a noise cancellation comprises an amplifier in communication with the three or more speakers disposed in an area. A system controller produces a driver signal for each of the speakers in response to a signal from at least one microphone detecting sound in the area and communicates the driver signals to the amplifier. The amplifier drives each speaker with the driver signal produced for that speaker. In response to the driver signals, the speakers emit sound that combined produces a substantially uniform sound pressure field for a particular zone within the area. The substantially uniform sound pressure field produced by the speakers has a magnitude and phase adapted to attenuate a noise field in the area corresponding to the sound detected by the at least one microphone.

Abstract: In embodiments of this invention howling sounds are removed without disturbing speech and music. Microphone signals are converted to the frequency domain where howling frequencies are detected and estimated using howling to noise ratios. A notch filter adaptively designed for the howling frequency is applied to remove the howling.

Abstract: Commands for modifying audio playback, such as to mute and unmute or pause and play audio, may be input to a mobile device by a user through interacting with the speaker in the device. The user input may be facilitated by monitoring a characteristic of the speaker and identifying signatures in the changing characteristics of the speaker that correspond with predetermined user activities. For example, a resonance frequency of the speaker may be monitored for a change resulting from a user placing a hand to cover the speaker output. When the resonance frequency change is detected, the audio playback may be muted. The speaker may continue to be monitored for a change indicating removal of the user's hand, and then audio playback may be unmuted.

Abstract: An acoustic feedback detection method and device. According to the method, whether acoustic feedback occurs is determined based on a frequency characteristic of an acoustic feedback signal. Specifically, a judgment value is determined using a power peak value and an average peak value, and it is determined whether acoustic feedback occurs in a signal based on a magnitude of the judgment value and a duration of the power peak value. In this case, whether acoustic feedback occurs can be determined based on the frequency characteristic of the signal.

Abstract: The invention includes a method, apparatus, and computer program to selectively suppress wind noise while preserving narrow-band signals in acoustic data. Sound from one or several microphones is digitized into binary data. A time-frequency transform is applied to the data to produce a series of spectra. The spectra are analyzed to detect the presence of wind noise and narrow band signals. Wind noise is selectively suppressed while preserving the narrow band signals. The narrow band signal is interpolated through the times and frequencies when it is masked by the wind noise. A time series is then synthesized from the signal spectral estimate that can be listened to. This invention overcomes prior art limitations that require more than one microphone and an independent measurement of wind speed. Its application results in good-quality speech from data severely degraded by wind noise.

Abstract: A method for detecting whistling in an audio system includes determining an average frequency of an input signal of the audio system, sampling the input signal in consecutive blocks of at least one sample, wherein the average frequency is determined blockwise, and determining whether feedback related whistling is present in the input signal of the audio system by evaluating a stability of the average frequency, wherein the evaluation of the stability of the average frequency comprises: determining a difference of two values of the determined average frequency for two blocks, and comparing the determined difference to a first threshold value.

Abstract: Personal audio systems and methods are disclosed. A personal audio system includes a class table storing processing parameters respectively associated with a plurality of annoyance noise classes, a controller, and a processor. The controller identifies an annoyance noise class of an annoyance noise included in an ambient audio stream and retrieves, from the class table, one or more processing parameters associated with the identified annoyance noise class. The processor to processes the ambient audio stream according to the one or more retrieved processing parameters class to provide a personal audio stream. The processor includes a pitch tracker to identify a fundamental frequency of the annoyance noise and a filter bank including a band reject filter tuned to the fundamental frequency.

Abstract: A communication system includes a carrier generator configured to generate a first carrier signal and a demodulator configured to demodulate a modulated signal responsive to the first carrier signal. The demodulator includes a filter and a bandwidth adjusting circuit. The filter is configured to filter a first signal. The first signal is a product of the first carrier signal and the modulated signal. The filter has a first cutoff frequency and a bandwidth. The bandwidth of the filter is controlled by a set of control signals. The bandwidth adjusting circuit is configured to adjust the bandwidth of the filter based on a frequency of the filtered first signal and a frequency of the first signal, or a phase of the filtered first signal and a phase of the first signal. The bandwidth adjusting circuit is configured to generate the set of control signals.

Abstract: An audio signal attenuation system and method for detecting an audio emergency warning signal (or alarm) in a vehicle in which an audio signal is being played. Embodiments of the system and method make it easier for a police, fire, or other emergency alarm or siren to be heard in a loud or noisy listening environment when audio signal is being reproduced. This is achieved using selective frequency attenuation, which identifies a frequency of the alarm and then selectively attenuates the alarm frequency in the audio signal. Moreover, direction data that includes information about from which direction the alarm is coming can be used to selectively attenuate the alarm frequency in certain channels (or speakers) of the audio signal. In some embodiments, audio cues are used to alert the listener to the alarm signal and are adjusted based on alarm distance from the vehicle, speed, and the type of alarm.

Abstract: The invention includes a method, apparatus, and computer program to selectively suppress wind noise while preserving narrow-band signals in acoustic data. Sound from one or several microphones is digitized into binary data. A time-frequency transform is applied to the data to produce a series of spectra. The spectra are analyzed to detect the presence of wind noise and narrow band signals. Wind noise is selectively suppressed while preserving the narrow band signals. The narrow band signal is interpolated through the times and frequencies when it is masked by the wind noise. A time series is then synthesized from the signal spectral estimate that can be listened to. This invention overcomes prior art limitations that require more than one microphone and an independent measurement of wind speed. Its application results in good-quality speech from data severely degraded by wind noise.

Abstract: A listening device includes an input transducer, an output transducer, a signal processor, and a feedback cancellation system for estimating acoustic feedback. The feedback cancellation system includes an adaptive FBC filter arranged in parallel to the forward path, an adaptive whitening filter, and a howl detection unit that performs howl detection based on an output of the adaptive whitening filter. The adaptive FBC filter includes a variable FBC filter part and an FBC update algorithm part for updating the variable FBC filter part, the FBC update algorithm part receiving first and second FBC algorithm input signals influenced by the electrical input and the electrical output signals. Further, the feedback cancellation system receives an electrical update signal essentially consisting of the direct part of the electrical input signal and coefficients of the adaptive whitening filter are based on the electrical update signal.

Abstract: An audio conference system has a plurality of specially designed microphone housings into each of which a directional microphone is positioned. Each microphone housing is positioned entirely within the body of the audio conferencing system, the microphone housings are strategically positioned at each one of four corners of the audio conference system body in order to provide maximum exposure of a microphone to its operating environment, and the interior structure of the microphone housing is designing to reflect unwanted energy harmlessly away from the microphone. Each directional microphone is positioned in the microphone housing such that the most sensitive node in its polar response pattern is oriented normal (at right angles) to a line radiating outward from the center of the system.

Abstract: A hearing aid comprises a microphone configured to receive an acoustic input signal and convert the signal into an electric input signal, an audiological signal processing unit coupled to the microphone and configured to process the electric input signal to a processed signal, an amplifier coupled to the audiological signal processing unit and configured to amplify the processed signal to an amplified signal, and a receiver coupled to the amplifier and configured to transform the amplified signal into an acoustic output signal. The hearing aid further comprises a howl detector coupled to the signal processing unit and configured to detect the presence of a feedback howl originating from the hearing aid an analyzer coupled to the howl detector and configured to determine a value indicative of whether the feedback howl detected is potentially audible to a person neighboring the user of the hearing aid.

Abstract: A method, device and computer program product for processing audio signals, the method including determining beamformer filter coefficients to be applied to the audio signals; applying the beamformer filter coefficients to the audio signals; outputting the filtered audio signals from an audio output comprising a plurality of speakers coupled with the device; and receiving at a microphone coupled with the device, the filtered audio signals output from the audio output. The filter coefficients are determined such that the filtered audio signals are suppressed when received at the microphone.

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

Abstract: A personal audio device, such as a wireless telephone, includes noise canceling circuit that adaptively generates an anti-noise signal from a reference microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone may also be provided proximate the speaker to measure the output of the transducer in order to control the adaptation of the anti-noise signal and to estimate an electro-acoustical path from the noise canceling circuit through the transducer. A processing circuit that performs the adaptive noise canceling (ANC) function also either adjusts the frequency response of the anti-noise signal with respect to the reference microphone signal, and/or by adjusting the response of the adaptive filter independent of the adaptation provided by the reference microphone signal.

Abstract: A pseudo noise superimposing unit superimposes a pseudo noise (M-sequence) to an audio signal picked up by a microphone and outputs the superimposed signal to an amplifying system. An calculating unit calculates a correlation value between the audio signal picked up by the microphone and the pseudo noise. The calculating unit estimates a gain of a closed loop based on the correlation value. A gain control unit suppresses a gain of the audio signal based on the estimated gain of the closed loop.

Abstract: An audio signal measurement method for a speaker and an electronic apparatus having the speaker are provided. The electronic apparatus further has a processing circuit and a power amplifier. The processing circuit is coupled to the speaker and configured to execute a time domain to frequency domain transform according to a voltage value of an audio signal and a current value of current feedback from the speaker so as to obtain a frequency response curve. The power amplifier is coupled to the speaker and configured to drive the speaker according the voltage value of the audio signal. The processing circuit is capable of determining whether the frequency response curve is located within a predetermined area such that the processing circuit generates a signal when the frequency response curve is located out of the predetermined area. Thereby, the electronic apparatus may measure its transducer distortion and acoustic box leakage.

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

Abstract: An acoustic echo suppression unit according to an embodiment of the present invention includes and input interface for extracting a downmix signal from an input signal, the input signal including the downmix signal and parametric side information, wherein the downmix and the parametric side information together represent a multichannel signal, a calculator for calculating filter coefficients for an adaptive filter, wherein the calculator is adapted to determine the filter coefficients based on the downmix signal and a microphone signal or a signal derived from the microphone signal, and an adaptive filter adapted to filter the microphone signal or the signal derived from the microphone signal based on the filter coefficients to suppress an echo caused by the multichannel signal in the microphone signal.

Abstract: The adjustment of a hearing device with regard to a current feedback path is to be designed in a more user-friendly manner. For this purpose, provision is made for hearing aid equipment, in particular a hearing device or a non-programmable device such as a headset, including a measuring device for measuring a feedback path and an operating facility for operating the hearing aid equipment by the wearer. The operating facility allows a measurement cycle to be activated to determine at least one characteristic of the feedback path. The hearing aid wearer can thus make adjustments themselves in terms of the feedback path and therefore does not have to rely on an acoustician.

Abstract: An echo suppression system and method, and a computer-readable storage medium that is configured with instructions that when executed carry out echo suppression. Each of the system and the method includes the elements of a linear echo suppressor having a reference signal path, with a nonlinearity introduced in the reference signal path to introduce energy in spectral bands. Unlike an echo canceller, the echo suppression system and method are relatively robust to errors in the introduced nonlinearity.

Abstract: A method and an audio processing system determine a system parameter, e.g. step size, in an adaptive algorithm, e.g. an adaptive feedback cancellation algorithm so as to provide an alternative scheme for feedback estimation in a multi-microphone audio processing system. A feedback part of the system's open loop transfer function is estimated and separated in a transient part and a steady state part, which can be used to control the adaptation rate of the adaptive feedback cancellation algorithm by adjusting the system parameter, e.g. step size parameter, of the algorithm when desired system properties, such as a steady state value or a convergence rate of the feedback, are given/desired. The method can be used for different adaptation algorithms such as LMS, NLMS, RLS, etc. in hearing aids, headsets, handsfree telephone systems, teleconferencing systems, public address systems, etc.

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

Abstract: An input signal is processed through noise suppression (NS) and echo control (EC) via a multipath model that reduces noise pumping effects while maintaining EC performance. A copy of a “noisy” input signal is sent to an EC component before the noisy signal is sent to a NS component, which processes the signal first, when there is a consistent noise level for estimation. The copy of the pre-processing noisy signal is sent to the EC component along with a “clean” or “noise-suppressed” signal output from the NS component. The EC component analyzes the noisy signal as if the EC was the first component in the signal chain to determine what actions to take. The EC component then applies these actions to the clean signal received from the NS component.

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

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

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

Abstract: Where information is to be conveyed as acoustic waves, embodiments of the present invention provide a transmitter capable of causing acoustic waves that convey information to be not easily perceived by the human ear. A transmitter, being an apparatus that converts various types of encoded information into acoustic wave(s) in the audible spectrum and carries out transmission thereof, comprises microphone(s) that cause ambient sound(s) from location(s) at which acoustic wave(s) is/are transmitted to be input as ambient sound signal(s); peak frequency detector(s) that detect, within ambient sound signal(s), peak frequency or frequencies of major constituent(s) of ambient sound(s); carrier wave generator(s) that generate carrier waves at a plurality of frequencies that are natural number multiples of peak frequency or frequencies and that can be used to mask ambient sound(s); and modulator(s) that modulate a plurality of carrier waves with baseband signal(s).

Abstract: It is described a method for controlling an adaptation of a behavior of an audio device (100) to a current acoustic environmental condition. The method comprises (a) monitoring an audio output signal (x(t), x?(t)) being provided to an acoustic output device (110) of the audio device (100) for outputting an acoustic output signal, (b) measuring an audio input signal (z(t)) being provided by an acoustic input device (120) of the audio device (100), wherein the audio input signal (z(t)) is indicative for a feedback portion of the acoustic output signal and for the current acoustic environmental condition, (c) determining a relation between the audio output signal (x?(t)) and the audio input signal (z(t)) and (d) adapting the behavior of the audio device (100) based on the determined relation.

Abstract: A method of enhancing an audio signal includes the steps of: a) receiving a primary audio input signal, b) receiving a detected audio signal which comprises: A) an echo component derived from play-out of the primary audio input signal and B) a noise component, and c) estimating from the primary audio input signal and the detected audio signal: 1) a set of frequency-specific lower bound gains, such that each frequency-specific lower bound gain, when applied to a respective frequency of the primary audio input signal, would cause the noise component to just mask the echo component at that respective frequency and 2) a set of frequency-specific upper bound gains, such that each frequency-specific upper bound gain, when applied to a respective frequency of the primary audio input signal, would cause the echo component to just mask the noise component at that respective frequency; d) estimating a set of frequency-specific gains in such a way that each frequency-specific gain falls between the respective frequency-

Abstract: The invention concerns in a first aspect an audio system comprising a microphone, audio signal processing means, an output transducer and means for detecting a possible feedback tone and the corresponding frequency of the feedback tone in the audio system between the output transducer and the microphone. According to the invention means for counteracting feedback are provided. Further, means are provided for changing the phase of the audio signal at a given frequency. In a further aspect, a binaural hearing aid system comprising first and second hearing instruments according to the first aspect, each hearing instrument comprising transceiver circuitry allowing an exchange of signals between the two hearing instruments, and wherein the binaural hearing aid system is adapted to provide that a phase change introduced in the first audio signal by a controller of the first hearing instrument is or can be introduced in the second audio signal of the second hearing instrument via said transceiver circuitry.

Abstract: A method for detecting whistling in an audio system includes determining an average frequency of an input signal of the audio system, sampling the input signal in consecutive blocks of at least one sample, wherein the average frequency is determined blockwise, and determining whether feedback related whistling is present in the input signal of the audio system by evaluating a stability of the average frequency, wherein the evaluation of the stability of the average frequency comprises: determining a difference of two values of the determined average frequency for two blocks, and comparing the determined difference to a first threshold value.