Abstract: In an embodiment, a circuit arrangement for active noise cancellation, comprises a first input (E1) for supplying a playback signal (Spb), a second input (E2) for supplying a sensor signal (Sanc), a first and a second terminal (A1, A2) of an output that is designed for being connected to a loudspeaker (Lsp) and a compensating device for respectively generating a first and a second noise signal (Sanc1, Sanc2) as a function of the sensor signal (Sanc),wherein the first and the second input (E1, E2) are coupled to the first and the second terminal of the output (A1, A2) by means of the compensating device (Komp) in such a way that a virtual playback signal (Ssp1) is provided at the first terminal (A1) of the output (A1, A2) and a superposition signal (Ssp2) is provided at the second terminal (A2) of the output (A1, A2) such that a differential signal between the virtual playback signal (Ssp1) and the superposition signal (Ssp2) can be fed to the loudspeaker.

Abstract: The invention provides a stem for an earplug and an earplug incorporating the stem where the stem includes an attachment portion configured to receive and retain a sound attenuating element, the attachment portion extending substantially along an attachment axis, and a handle portion extending from the attachment portion substantially along a handle axis, where at least part of the handle axis is non-collinear with respect to the attachment axis.

Abstract: A multiple error filtered-x least mean square (MEFxLMS) algorithm using a channel equalization virtual secondary path for an active noise control/cancellation (ANC) system for treating noise in a Multiple-Input Multiple-Output (MIMO) system. The channel equalization technique equalizes amplitude levels of the estimated response of all primary channels to overcome limitations caused by the frequency dependent property of standard filtered-x least mean square (FxLMS) algorithm, reduce the variation of convergence speed existed in the multiple channels, and improve the overall performance of the control system. The convergence property of the algorithm is analyzed in the frequency domain.

Abstract: A method in a communications device includes the following operations. During a call, a process automatically detects that the device has moved from an at-the-ear position to an away-from-the-ear position. Based on the detection, a noise suppressor that operates upon an uplink signal for the call is signaled to change its noise suppression performance. Other embodiments are also described and claimed.

Abstract: A noise reduction is provided for a hearing apparatus, with which both stationary and also non-stationary interference noises can be attenuated in an input signal. An output signal is in this way to convey a quite sound impression. A signal processing is provided, which effects a noise reduction on the basis of two different methods. Provision is made on the one hand for a noise reduction for stationary interference noises and on the other hand for a noise reduction for spatially oriented interference noises. A selection facility selects between the two noise reductions.

Abstract: Echo cancellation is handled using a pattern classification technique. During a training phase, the communications device is trained to learn patterns of signal inputs that correspond to certain communication modes. After numerous different patterns have been classified by mode, the classified patterns are used during real-time use of the communications device in order to determine, dynamically, the mode in which the communications device is currently operating. The communications device may then apply, to the microphone-produced signal, a suppression action.

Abstract: An active noise control apparatus includes a first active noise controller for generating a first canceling signal for a first noise type, a second active noise controller for generating a second canceling signal for a second noise type that is different from the first noise type, a mixer for mixing the first canceling signal and the second canceling signal into a mixed canceling signal, a canceling sound output unit for outputting a canceling sound based on the mixed canceling signal, and an amplitude suppressor for suppressing the amplitude of the second canceling signal depending on the amplitude of the first canceling signal.

Abstract: An apparatus and method for active cancellation of interference at a hearing assistive devices are presented. The apparatus includes an interference cancellation circuit for cancelling an interference component of a composite signal, and an activator circuit for enabling interference cancellation circuit. The interference cancellation circuit generates an estimated replica of the interference from an interference profile, inverts the replica to form a cancellation waveform, then adds the cancellation waveform to the composite signal to cancel the interference component. An interference profile can be provided by performing a training sequence on a composite signal to detect a repetitive signal and building a profile using its parameters, retrieving a profile stored in memory, or using an antenna to capture an RF signal.

Abstract: There is presented a hearing aid that comprises an input transducer for transforming an acoustic input signal into an electrical input signal, a compressor for generating an electrical output signal from the electrical input signal, an output transducer for transforming the electrical output signal into an acoustic output signal, an autocorrelation estimator for calculating an autocorrelation estimate of the electrical input signal, and an acoustic loop gain estimator for determining a dynamic gain limit from the autocorrelation estimate and an instantaneous gain level of the signal processor. The invention further provides a method of adjusting signal path gain in a hearing aid, and a system for providing increased stability in a hearing aid.

Abstract: A noise reducing comprises a first microphone that picks up noise signal at a first location and that is electrically coupled to a first microphone output path; a loudspeaker that is electrically coupled to a loudspeaker input path and that radiates noise reducing sound at a second location; a second microphone that picks up residual noise from the noise and the noise reducing sound at a third location and that is electrically coupled to a second microphone output path; a first active noise reducing filter that is connected between the first microphone output path and the loudspeaker input path; and a second active noise reducing filter that is connected between the second microphone output path and the loudspeaker input path; in which the first active noise reduction filter is a shelving or equalization filter or comprises at least one shelving or equalization filter or both.

Abstract: The sound device includes an audio-information output unit, an analysis unit, an audio-division-spectrum output unit, a noise-division-spectrum output unit and a correction unit. The analysis unit receives audio information from the audio-information output unit, and then outputs sound spectrum information. The noise-division-spectrum output unit outputs sound-volume information for each critical band width of a noise, and the audio-division-spectrum output unit outputs the sound-volume information for each critical band width of the sound-spectrum information. The correction unit corrects the information from the audio-division-spectrum output unit based on the information from the noise-division-spectrum output unit. The audio-signal properties can be well corrected corresponding to the auditory-sense properties of the human, and thus the audio sound, in which an uncomfortable feeling to the auditory sense of the human has been adequately controlled, can be transmitted to a user.

Abstract: A method of removing a signal from among received signals, the method including: filtering the received signals; detecting a time band of the filtered received signals where an energy value of the filtered received signals exceeds a reference energy value; and applying a gain value to one or more received signals, from among the received signals, in the detected time band.

Abstract: Improvements in voice signals transmitted within communication systems are obtained by use of adaptive filters, front and rear microphones, noise cancelling systems and other means and methods. Disclosed embodiments include the use of directional microphones, primary inputs, secondary inputs, adaptive weight generators, canceller outputs to improve signal to noise ratios and other communication attributes.

Abstract: A personal audio device, such as a wireless telephone, generates an anti-noise signal from an error microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. The error microphone is also provided proximate the speaker to provide an error signal indicative of the effectiveness of the noise cancellation. A secondary path estimating adaptive filter is used to estimate the electro-acoustical path from the noise canceling circuit through the transducer so that source audio can be removed from the error signal. Noise bursts are injected intermittently and the adaptation of the secondary path estimating adaptive filter controlled, so that the secondary path estimate can be maintained irrespective of the presence and amplitude of the source audio.

Abstract: The suppression device for outdoor noises in indoor space solves the technical problems on realizing better sound insulation effect within the room space and creating a quiet atmosphere for people's living environment. The structure includes an anti-noise sound source synchronized to and invert with outdoor noises. The circuit structure of suppression device includes a directional reception and processing circuit for outdoor noises, a attenuation control circuit, a multi-channel digital amplifier circuit, and loudspeaker being installed at the inlet of indoor noises. The directional reception and processing circuit for outdoor noises processes the noise signals collected from the outdoor environment and generates the inverted-phase signals, and then transmits them to the input terminal of attenuation control circuit.

Abstract: A system for speech enhancement in a room, having: a microphone arrangement with at least two spaced apart microphones for capturing audio signals from a speaker's voice, an acoustic beamforming unit for processing the captured audio signals in a manner so as to impart one of a plurality of different directional patterns to the microphone arrangement; a feedback cancellation unit for applying a feedback cancellation algorithm to the processed audio signals and for providing a feedback status signal indicating how close an acoustic feedback loop of the system is to feedback; an amplifier for amplifying the processed audio signals; a loudspeaker arrangement to be located in the room for generating sound according to the amplified audio signals; and a control for selecting the directional pattern imparted to the microphone arrangement as a function of the feedback status signal.

Abstract: Provided is a pop-noise removing method. The pop-noise removing method analyzes an acoustic noise, occurring when receiving digital broadcasting under a weak electric field environment, and reduces a pop-noise on the basis of the analyzed result, for transferring digital broadcasting with a minimized pop-noise to consumers.

Abstract: A Voice Activity Detection (VAD) algorithm provides a simple binary signal indicating the presence or absence of speech in a microphone signal. The VAD algorithm includes a first step of noise suppression which both estimates and removes (i.e., filters) ambient noise from the microphone signal to create a filtered signal. The magnitude of the filtered signal is then compared to a threshold in order to produce a VAD output signal. The threshold is dynamic and may be derived either from the filtered signal itself, or from a noise spectrum estimate calculated by the noise suppression step.

Abstract: An encoding apparatus comprises a frame processor (105) which receives a multi channel audio signal comprising at least a first audio signal from a first microphone (101) and a second audio signal from a second microphone (103). An ITD processor 107 then determines an inter time difference between the first audio signal and the second audio signal and a set of delays (109, 111) generates a compensated multi channel audio signal from the multi channel audio signal by delaying at least one of the first and second audio signals in response to the inter time difference signal. A combiner (113) then generates a mono signal by combining channels of the compensated multi channel audio signal and a mono signal encoder (115) encodes the mono signal. The inter time difference may specifically be determined by an algorithm based on determining cross correlations between the first and second audio signals.

Abstract: A headphone accessory for use with a portable audio device and headphones. The headphone accessory includes an audio source input for receiving audio signals from an audio device. An audio sound transducer receives external sounds and converts the external sound to external sound signals. A signal mixer for continuously varies the balance of a source audio volume to an external sound volume. The signal mixer mixes the volume-adjusted source audio signals with volume-adjusted external sound signals. An audio output outputs the mixed source audio and external sound signals.

Abstract: The disclosure is directed to techniques for noise handling in audio or video recording. Sounds that may be generated during the audio recording, such as by activation of switches, may appear as undesired artifacts in the recording. Techniques and apparatus are described whereby, when an undesired sound is generated, an inverse of the sound is applied to the sound, thereby canceling the unwanted sound from the recording. Further, the techniques can be adapted to many kinds of portable electronic devices having many kinds of switches and many kinds of undesirable sounds.

Abstract: An active noise cancellation earphone (1) has an acoustic path including a cavity (36) and a pipe (20) leading to the auditory canal (40) which are arranged to form an oscillator in use which has the effect of recovering the open loop system phase characteristics at a selected frequency or frequency range. The earphone (1) also has two parts (5,18) which can be adjusted relative to each other to allow the earphone (1) to be comfortably and correctly positioned in use.

Abstract: A sound reproducing device has a loudspeaker arranged to produce sound from an audio signal provided by an audio signal source. A microphone is positioned to pick up ambient noise and generate a microphone signal which comprises the noise. An ambient noise cancellation (ANC) system receives the microphone signal from the microphone and generates anti-noise corresponding to the ambient noise in the microphone signal. An automatic polarity adaptation (AAP) system monitors the ANC system and, when a decision criterion is fulfilled, causes a switch in polarity for the generated anti-noise.

Abstract: A method for determining leakage factors or adaptation rates, or both, for adaptive filters in an active noise reduction system. The leakage factor or adaptation rate, or both, may vary depending on a parameter of an input reference signal. The parameter may include one or more of reference signal input frequency, rate of change of reference input signal frequency, if a predetermined triggering condition exits, or if a predetermined event has occurred.

Abstract: A system and method for enhancing the sound signal produced by an audio system in a listening environment by compensating for ambient sound in the listening environment, comprises producing an audio sound in the time domain from an electrical sound signal in the time domain. The electrical sound signal in the time domain is transformed into an electrical sound signal in the frequency domain and the electrical sound signal in the frequency domain is retransformed into an audio sound in the time domain. The total sound level in the environment is measured and a signal representative thereof is generated. The audio sound signal and the total sound signal are processed to extract a signal representing the ambient sound level within the environment, and equalization is performed in the frequency domain to adjust the output from the audio sound signal to compensate for the ambient noise level.

Abstract: A noise reduction device including a noise detection microphone including a high-frequency noise detection microphone and a low-frequency noise detection microphone for respectively detecting a high-frequency noise and a low-frequency noise generated from a noise source; a noise control unit for generating a control sound signal for cancelling a noise detected by the noise detection microphone in a control center of control space; and a loudspeaker for outputting a control sound based on the control sound signal from the noise controlling unit. The high-frequency noise microphone is disposed in a vicinity of a head portion of a user in a state in which directivity in an opposite direction with respect to the control center is added, and the low-frequency noise detection microphone is disposed outside of a sound-insulating wall.

Abstract: A portable audio device, which includes active noise cancellation circuitry, a hearing aid compliant magnetic radiator, and a speaker/earpiece, is surrounded by ambient acoustic noise. The active noise cancellation circuitry provides an anti-noise signal at an input of the speaker to control/reduce the ambient acoustic noise outside of the device. In addition, the active noise cancellation circuitry provides an inverse anti-noise signal to an input of the magnetic radiator. The magnetic fields produced by the speaker driven by the anti-noise signal and the magnetic radiator driven by the inverse anti-noise signal cancel each other out through phase cancellation such that a hearing aid using a telecoil coupled to the audio device does not produce significant audio waves based on either of these signals. Other embodiments are also described.

Abstract: A sound emitting and collecting apparatus that can accomplish stable echo removal if the relative positions of a loudspeaker and a microphone change is provided. A control section 52 inputs information of an arm rotation angle from a sensor 54. The control section 52 reads the filter coefficient corresponding to the input rotation angle from memory 53 and sets the coefficient in an adaptive filter 412A. When the arm rotates, the installing position of a microphone is changed and an acoustic transmission system changes, but a preset (or in the installing environment, updated) filter coefficient is set, whereby stable echo removal can be accomplished.

Abstract: A noise reduction circuit 200 for a headphone 100 is disclosed herein. In a described embodiment, the headphone 100 includes a speaker driver 110 and the circuit 200 comprises a microphone 112 configured to convert ambient sound into a corresponding electrical ambient signal and which is disposed adjacent to the speaker driver's diaphragm. The circuit 200 further includes an active noise reduction path configured to provide active noise reduction of the ambient sound based on the corresponding electrical ambient signal and a vocal signal compensation path configured to restore attenuated signals within the vocal range of the corresponding electrical ambient signal to increase audibility of vocal signals of the ambient sound. The circuit 200 also includes a switching device 204,210 arranged to selectively deliver the corresponding electrical ambient signal to the active noise reduction path or to the vocal signal compensation path.

Abstract: A method of increasing the intelligibility of an audio broadcast in an at least partially enclosed space from at least one amplified audio source. An input microphone receives an incident audio wavefront at a first position in the at least partially enclosed space. An active noise control system is employed to generate a cancelling audio wavefront having a magnitude substantially equal to the magnitude of incident audio wavefront and a phase substantially opposite to the phase of the incident audio wavefront. The cancelling audio wavefront is broadcast at a second position in the at least partially enclosed space adjacent to a reflective surface of the at least partially enclosed space so as to attenuate the incident audio wavefront substantially at or near the reflective surface in order to reduce reverberations of the incident audio wavefront. In this manner, reverberations which could reduce the intelligibility of the audio broadcast to an audience is reduced.

Abstract: An acoustical noise reduction system which comprises a primary acoustic to electric transducer, digital signal processor (DSP), an electric to acoustic transducer, and a secondary acoustic to electric transducer is disclosed in the present invention. The active noise reduction system is located close to the noise source with the sound being sensed by a primary transducer before the noise enters the active noise reduction area. The system functions to generate an anti-noise cancellation sound wave with an acoustic propagation speed of approximately 330 m/sec and an electromagnetic propagation speed of approximately 3×10?8 m/sec.

Abstract: The present technology provides adaptive noise reduction of an acoustic signal using a sophisticated level of control to balance the tradeoff between speech loss distortion and noise reduction. The energy level of a noise component in a sub-band signal of the acoustic signal is reduced based on an estimated signal-to-noise ratio of the sub-band signal, and further on an estimated threshold level of speech distortion in the sub-band signal. In various embodiments, the energy level of the noise component in the sub-band signal may be reduced to no less than a residual noise target level. Such a target level may be defined as a level at which the noise component ceases to be perceptible.

Abstract: The invention relates to a noise canceling audio transmitting/receiving device; a stereo headset with an integrated array of microphones utilizing an adaptive beam forming algorithm. The invention also relates to a method of using an adaptive beam forming algorithm that may be incorporated into a stereo headset. The sensor array used herein has adaptive filtering capabilities.

Abstract: A noise cancellation system generates a noise cancellation signal from a signal representing ambient noise by signal processing. The signal processing applies a controllable gain value, and includes a high pass filter with a controllable cut-off frequency. A control block detects a wind amplitude. The cut-off frequency of the high pass filter is controlled based on the detected wind amplitude. A low pass function is applied to the detected wind amplitude, and the controllable gain is adjusted based on the output of the low pass function.

Abstract: Provided is an active noise interference system. The active noise interference system may comprise a first microphone, a controller, a speaker, and an electrical power unit. The first microphone may be mounted within a first cavity defined by a tire-wheel system. The speaker may be mounted within the first cavity. The electrical power unit may be engaged with a component set comprising the first microphone, or the controller, or the speaker, or a combination thereof. The electrical power unit may be adapted to provide electrical power to the component set.

Abstract: Audio processing devices and methods. An ambient microphone picks up ambient sound and a voice microphone picks up a speaker's voice. The voice microphone is located farther from the speaker's mouth when a device is held to one side of the speaker's face than when held to the other side. A signal indicates whether an expected position of the handheld body is a position on a left side or right side of a speaker's face. Ambient sound in the voice signal is reduced by applying a first algorithm configuration selected to process voice signals with lower signal-to-noise ratio when the expected position in on one side of the speaker's face, and by applying a second sound cancellation algorithm configuration selected to process higher signal-to-noise ratios when the expected position is on the other side of the speaker's face.

Abstract: Disclosed are an apparatus and a method for voice processing in a mobile communication terminal. A plurality of microphones are used to remove environmental noise at the time of voice communication, so that it is possible to perform high-quality voice communication and video telephony. Moreover, it is possible to perform voice recording even when a user does not open a mobile communication terminal. Furthermore, when voice is recorded or sound is recorded during moving image photographing, a plurality of microphones are effectively utilized to achieve good-quality recording and to perform recording conveniently even when the folder or the slider of the mobile communication terminal is closed. Therefore, it is possible to provide improved convenience in using the mobile communication terminal.

Abstract: A system and method for localized noise cancellation. An audio signal is received from an environment in close proximity to a primary area. The audio signal is processed to generate an inverse signal of the audio signal. The inverse signal is broadcast within the primary area to destructively interfere with the audio signal. The inverse signal is configured to prevent the audio signal from being broadcast through a telephone conversation ongoing in the primary area.

Abstract: An A/D conversion device includes a first A/D converter that receives an analog vibration signal corresponding to a vibration detected by a vibration detecting sensor and converts the analog vibration signal to a first digital value, an analog amplifier that amplifies the analog vibration signal and outputs an amplified signal, a second A/D converter that converts the amplified signal to a second digital value, a determiner that determines whether an amplitude of the amplified signal input to the second A/D converter exceeds a range of amplitude that is convertible by the second A/D converter, and a selector that receives the first digital value and the second digital value and output one of the first and second digital values based on the determination of the determiner.

Abstract: Apparatus and method of controlling provision of ANR, possibly of a personal ANR device, in which amplitudes of a piece of audio employed in the provision of ANR are monitored, and the compression of one or both of feedback and feedforward ANR reference sounds is made dependent on frequency such that a first sound of one frequency need only reach a lower amplitude to trigger compression, while a second sound of another frequency must reach a higher amplitude to trigger compression.

Abstract: In a method and system for fitting the gain of a hearing aid (300) for a hearing impaired person, a loop gain of the hearing aid in the ear canal (350) of the hearing impaired person is measured for at least one frequency band. An effective vent parameter such as a corresponding vent diameter for the hearing aid by determining a vent parameter that generates the best fit between a modelled and the measured loop gain is estimated, a vent effect value based on the estimated effective vent parameter is determined, and a corrected hearing aid gain is provided by means of the determined vent effect value. The invention provides a method, a computer program, a system for fitting a hearing aid, a hearing aid and a computer system.

Abstract: A system for damping audible noise from a drive train. One embodiment is a system with an audible noise signal from a drive train wherein a damping circuit produces a damping torque that is a phase shifted and amplified version of the noise signal that is applied to the air-gap torque of a generator or motor coupled to the drive train and effectively reduces the audible noise.

Abstract: A non-acoustic sensor is used to measure a user's speech and then broadcasts an obscuring acoustic signal diminishing the user's vocal acoustic output intensity and/or distorting the voice sounds making them unintelligible to persons nearby. The non-acoustic sensor is positioned proximate or contacting a user's neck or head skin tissue for sensing speech production information.

Abstract: A noise reduction device of the present invention comprises a control filter unit for generating a control sound signal to cancel out a noise, a control speaker for outputting a control sound according to the control sound signal from the control filter unit, an error microphone for detecting a residual sound by superimposing the noise upon the control sound output from the control speaker, and an obstacle detector for detecting an obstacle around the error microphone, wherein the control filter unit generates the control sound signal according to data from the error microphone and the obstacle detector.

Abstract: Systems and methods described herein provide for low latency active noise cancellation, which alleviates the problems associated with analog filter circuitry. The present technology utilizes low latency digital signal processing techniques that overcome the high latency conventionally associated with conversion between the analog and digital domains. As a result, low latency active noise cancellation is performed utilizing digital filter circuitry which is not subject to the inaccuracies and drift of analog filter components. In doing so, the present technology provides robust, high quality active noise cancellation.

Abstract: The present invention compensates output variability caused by the difference in the performance of and the individual difference between converter devices when processing the signals inputted by means of a converter device, and performs highly-accurate signal processing. The present invention is provided with an input means which inputs an input signal through a converter device, a memory means which stores a minimum value of a reference signal inputted through a reference converter device, a comparison means which compares a minimum value of the input signal and the minimum value of the reference signal, and a modification means which modifies the input signal in accordance with the comparison result of the comparison means.

Abstract: A noise injection system adds comfort noise to an audio signal. The system includes a background noise estimator that determines a spectral content of a background noise associated with the audio signal. A comfort noise generator generates a comfort noise signal having a random phase. A gain circuit adjusts the comfort noise signal based on the spectral content of the background noise. A combining circuit combines a gain-adjusted comfort noise signal and the audio signal to generate an output signal.

Abstract: A correction spectrum calculation unit 6 obtains a correction spectrum by smoothing an estimated noise spectrum in accordance with the degree of its variations, and a suppression quantity limiting coefficient calculation unit 7 decides a suppression quantity limiting coefficient from the correction spectrum. A suppression quantity calculation unit 9 obtains a suppression coefficient based on the suppression quantity limiting coefficient, and the spectrum suppression unit 10 carries out amplitude suppression of spectral components of an input signal.

Abstract: A method of processing an audio signal is disclosed. The present invention includes receiving, by an audio processing apparatus, an input signal; estimating indicator function using a signal power of the input signal; obtaining an adapted filter using the indicator function and an equalization filter; and, generating an output signal by applying the adapted filter to the input signal.

Abstract: An active noise control arrangement has a signal input (SI), a microphone input (MI), a signal output (SO) and a digital interface (DI). A signal processing block (SP) coupled to the microphone input (MI) by means of an amplifier (MA) has a digitally adjustable gain and comprises combining means (CM) and a filter (TP). The signal processing block (SP) is configured to generate an output signal at the signal output (SO) as a function of an input signal at the signal input (SI) and an amplified microphone signal. A control block (CB) is coupled to the digital interface (DI) and configured to adjust the gain of the amplifier (MA).