Abstract: A multi-channel acoustic echo canceller arrangement comprises a microphone (111) providing a microphone signal having contributions from at least two audio sources (107, 109) to be cancelled. An echo canceling circuit (113, 115) performs echo cancellation of the two audio sources (107, 109) based on channel estimates for channels from each of the audio sources (107, 109) to the microphone (111). An estimation circuit (117) generates each of the channel estimates as a combination of a previous channel estimate and a channel estimate update where the combination includes applying a relative weight to the channel estimate update relative to the previous channel estimate. A weight processor 119 varies the relative weight in response to a time value. The arrangement may provide improved echo-cancellation for scenarios wherein the rendering of sound from the audio sources (107, 109) is time varying, such as when time varying decorrelation filters are used.

Abstract: Provided are methods and systems for noise suppression within multiple time-frequency points of spectral representations. A multi-feature cluster tracker is used to track signal and noise sources and to predict signal versus noise dominance at each time-frequency point. Multiple features, such as binaural and monaural features, may be used for these purposes. A Gaussian mixture model (GMM) is developed and, in some embodiments, dynamically updated for distinguishing signal from noise and performing mask-based noise reduction. Each frequency band may use a different GMM or share a GMM with other frequency bands. A GMM may be combined from two models, with one trained to model time-frequency points in which the target dominates and another trained to model time-frequency points in which the noise dominates. Dynamic updates of a GMM may be performed using an expectation-maximization algorithm in an unsupervised fashion.

Abstract: A processing apparatus estimates a noise amplitude spectrum of noise included in a sound signal. The processing apparatus includes an amplitude spectrum calculation part configured to calculate an amplitude spectrum of the sound signal for each one of frames obtained from dividing the sound signal into units of time; and a noise amplitude spectrum estimation part configured to estimate the noise amplitude spectrum of the noise detected from the frame. The noise amplitude spectrum estimation part includes a first estimation part configured to estimate the noise amplitude spectrum based on a difference between the amplitude spectrum calculated by the amplitude spectrum calculation part and the amplitude spectrum of the frame occurring before the noise is detected, and a second estimation part configured to estimate the noise amplitude spectrum based on an attenuation function obtained from noise amplitude spectra of the frames occurring after the noise is detected.

Abstract: A hearing assistance system includes an adaptive directionality controller to control a target direction for sound reception. The adaptive directionality controller includes a beamformer, a speech detector to detect off-axis speech being speech that is not from the target direction, and a steering module to steer the beamformer in response to a detection of the off-axis speech.

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: A noise processing apparatus measures a first potential difference signal, between a first electrode and a second electrode that is used as a reference electrode, and measures a second potential difference signal, between the second electrode and a third electrode that is arranged on the steering unit in the apparatus. The apparatus calculates the difference between the intensities of the first potential difference signal and the second potential difference signal calculated at the predetermined intervals. The apparatus corrects the first potential difference signal or the second potential difference signal by using the calculated difference such that the intensities of the first potential difference signal and the second potential difference signal are canceled out.

Abstract: A howling canceller applied to an acoustic system having a speaker and a microphone comprises: a filter insertion unit for inserting a notch filter at a frequency of an audio signal picked up by the microphone; a setting unit for setting the insertion time of the notch filter on the basis of the frequency at which the notch filter is inserted; and a releasing unit for, when the insertion time set by the setting unit has elapsed, releasing the notch filter, the insertion time of which has elapsed. The setting unit sets the insertion time of the notch filter to be shorter as the frequency at which the notch filter is inserted increases.

Abstract: An audio processing apparatus includes first and second audio pickup units. The second audio pickup unit includes an audio resistor provided to cover a sound receiving portion to suppress external wind introduction while passing an external audio. A first filter attenuates a signal having a frequency lower than a first cutoff frequency of the output signal of a first A/D converter. A second filter attenuates a signal having a frequency higher than a second cutoff frequency of the output signal of a second A/D converter. A third filter is provided between the first audio pickup unit and the first A/D converter to attenuate a signal having a frequency lower than a third cutoff frequency for suppressing the wind noise.

Abstract: A howling canceller is adapted to an acoustic system having a speaker and first and second microphones. The speaker and the first microphone form a first acoustic feedback loop; the speaker and the second microphone form a second acoustic feedback loop. The howling canceller includes a howling suppressing unit for performing suppression processing in such a way that: frequency components at which howling is possibly occurring are detected in each of the sound signals picked up by the first and second microphones; the detected frequency components of the sound signals picked up by the first and second microphones are compared with each other on a per-frequency basis and a frequency component having larger power is detected; and based on the comparison results, the larger power frequency component of at least one of the sound signals picked up by the first and second microphones is suppressed.

Abstract: An apparatus and method for optimizing a parametric emitter system having a pot core inductive device coupled between an amplifier and emitter. The pot core inductive device allows for adjustments of the air gap formed between the two halves of the pot core structure to adjust its inductive value. This post-manufacture adjustability allows for corrections of differences caused by operations of other components in the audio system and to account for slight differences in the electrical circuit of different amplifier/emitter combinations. As efficiency of the system is dependent on the functional relationship between the amplifier, inductive device, and emitter, this allows for fine tuning of the signal to obtain high quality.

Abstract: Noise suppression systems and methods suppress far field noise in a microphone signal. A telephony system includes a main microphone and a reference microphone. In one example, the main microphone and the reference microphone can be located in the same device. In another example, the main microphone and the reference microphone can be located in two separate devices. A DSP can use the reference microphone signal to carry out suppression of far field noise in the main microphone signal. In one approach the DSP can determine an estimate of far field noise in the main microphone signal based on a noise estimate of the reference microphone signal and a reference and main microphone coupling estimate, and then subtract the far field noise estimate from the main microphone signal. Alternatively, the DSP can suppress the main microphone signal if it determines that a local talker is inactive.

Abstract: Techniques are described herein that use sensors (e.g., microphones) for noise reduction in a mobile communication device. For example, one technique enables a first sensor that is initially configured to be a speech sensor to be used as a noise reference sensor. This technique also enables a second sensor that is initially configured to be a noise reference sensor to be used as a speech sensor. Another technique enables a primary sensor and/or a secondary sensor in a handset of a mobile communication device to be used as a speech sensor while a sensor in a headset of the mobile communication device is used as a noise reference sensor, or vice versa. In yet another technique, a secondary sensor in a mobile communication device is configured to be a directional sensor.

Abstract: A computer-implemented method including receiving a first signal from an input device of a hearing aid. The first signal may include a noise signal. The computer-implemented method may include low-pass filtering first periodic samples of the first signal, and the first periodic samples may be approximately periodic with respect to a period of the noise signal. The computer-implemented method may further include low-pass filtering second periodic samples of the first signal, and the second periodic samples may be approximately periodic with respect to the period of the noise signal. The second periodic samples may also be phase shifted relative to the first periodic samples. Hearing aid systems and apparatuses are also disclosed.

Abstract: A microphone module disposed in an electronic device for reducing echo noise. The microphone module includes a casing, a first diaphragm disposed in the casing, a second diaphragm disposed in the casing and a substrate disposed between the first diaphragm and the second diaphragm and joined to the casing to define a first space and a second space which are isolated and separated from each other. The first diaphragm is disposed in the first space, the second diaphragm is disposed in the second space, and the substrate is electrically connected with the first diaphragm and the second diaphragm.

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: An MEMS microphone is provided which includes a reference voltage/current generator configured to generate a DC reference voltage and a reference current; a first noise filter configured to remove a noise of the DC reference voltage; a voltage booster configured to generate a sensor bias voltage using the DC reference voltage the noise of which is removed; a microphone sensor configured to receive the sensor bias voltage and to generate an output value based on a variation in a sound pressure; a bias circuit configured to receive the reference current to generate a bias voltage; and a signal amplification unit configured to receive the bias voltage and the output value of the microphone sensor to amplify the output value. The first noise filter comprises an impedance circuit; a capacitor circuit connected to a output node of the impedance circuit; and a switch connected to both ends of the impedance circuit.

Abstract: Techniques described herein are directed to performing back-end single-channel suppression of one or more types of interfering sources (e.g., additive noise) in an uplink path of a communication device. The back-end single-channel suppression techniques may suppress types(s) of additive noise using one or more suppression branches (e.g., a non-spatial (or stationary noise) branch, a spatial (or non-stationary noise) branch, a residual echo suppression branch, etc.). The non-spatial branch may be configured to suppress stationary noise from the single-channel audio signal, the spatial branch may be configured to suppress non-stationary noise from the single-channel audio signal and the residual echo suppression branch may be configured to suppress residual echo from the signal-channel audio signal. The spatial branch may be disabled based on an operational mode (e.g.

Abstract: Described herein are multi-channel noise suppression systems and methods that are configured to detect and suppress wind and background noise using at least two spatially separated microphones: at least one primary speech microphone and at least one noise reference microphone. The multi-channel noise suppression systems and methods are configured, in at least one example, to first detect and suppress wind noise in the input speech signal picked up by the primary speech microphone and, potentially, the input speech signal picked up by the noise reference microphone. Following wind noise detection and suppression, the multi-channel noise suppression systems and methods are configured to perform further noise suppression in two stages: a first linear processing stage that includes a blocking matrix and an adaptive noise canceler, followed by a second non-linear processing stage.

Abstract: Methods and systems for enhancing signal quality are disclosed. A method includes receiving buffers of sound samples including a first microphone signal and a second microphone signal from a first and a second microphone; generating a first cardioid shape signal by subtracting a delayed second microphone signal from the first microphone signal; generating a second cardioid shape signal by subtracting the second microphone signal from a delayed first microphone signal; generating a first level output signal based on the first cardioid shape signal; detecting at least one speech and non-speech region of the first level output signal; generating a second level output signal based on the second cardioid shape signal, and at least one of the speech and non-speech regions of the first level output signal; and removing residuals of noise from the first level output signal based on adaptive weights output and generated second level output signal.

Abstract: A method of reducing noise in an environment where the noise source is in a fixed location relative to a pair of microphones, such as in a camera with a zoom motor, involves receiving signals x1(t), x2(t) from the respective microphones, and filtering each of the signals x1(t), x2(t) with respective first and second linear filters having filter coefficients obtained by computing eigenfilters corresponding to data samples from the respective microphones for noise only and signal only conditions.

Abstract: A sound mixture may be received that includes a plurality of sources. A model may be received for one of the source that includes a dictionary of spectral basis vectors corresponding to that one source. At least one feature of the one source in the sound mixture may be estimated based on the model. In some examples, the estimation may be constrained according to temporal data.

Abstract: Multi-channel noise suppression systems and methods are described that omit the traditional delay-and-sum fixed beamformer in devices that include a primary speech microphone and at least one noise reference microphone with the desired speech being in the near-field of the device. The multi-channel noise suppression systems and methods use a blocking matrix (BM) to remove desired speech in the input speech signal received by the noise reference microphone to get a “cleaner” background noise component. Then, an adaptive noise canceler (ANC) is used to remove the background noise in the input speech signal received by the primary speech microphone based on the “cleaner” background noise component to achieve noise suppression. The filters implemented by the BM and ANC are derived using closed-form solutions that require calculation of time-varying statistics of complex frequency domain signals in the noise suppression system.

Abstract: One or more microphones of a first devices receive a first audio. The first device generates a first audio signal from the received first audio. The first device analyzes the first audio signal to determine noise characteristics included in the first audio received by the first device. The first device generates noise compensation information based at least in part on the noise associated with the first audio. The first device transmits the noise compensation information to at least one of a second device or a third device that is intermediate to the first device and the second device. The first device receives a second audio signal generated by the second device, wherein the second audio signal is based at least in part on the noise compensation information transmitted by the first device. The first device then outputs the second audio signal to a speaker of the first device.

Abstract: Systems and methods to automatically equalize coloration in speech recordings is provided. In example embodiments, a reference spectral shape based on a reference signal is determined. An estimated spectral shape for an input signal is derived. Using the estimated spectral shape and the reference spectral shape a comparison is performed to determine gain settings. The gain settings comprise a gain value for each filter of a filter system. Using gain values associated with the gain setting, automatic equalization is performed on the input signal.

Abstract: A method of automatic announcer voice removal from a televised sporting event. A sound processing circuit divides an audio input signal of a televised sporting event into multiple audio segments. The audio input signal includes crowd noise and announcer commentary. If an audio segment does not exceed a pre-defined amplitude threshold, a voice removal utility adds the audio segment to a recent crowd noise library and stores the segment in an output buffer. If the amplitude of a segment exceeds the threshold, the utility adds the segment to a recent announcer voice library. The sound processing circuit generates an attenuated version of the segment and blends the attenuated version with one or more mixed segments from the recent crowd noise library. The voice removal utility stores the attenuated and blended segment in the output buffer and outputs one or more audio segments from the buffer in a chronological order.

Abstract: A microphone array apparatus includes: an acquisition unit configured to acquire samples from a sound signal inputted from each of a plurality of microphones, at predetermined time intervals; an operation unit configured to calculate a value based on volumes of the sound signal possessed by a plurality of the samples for each of the sound signals inputted from the plurality of microphones; a correlation coefficient calculator configured to calculate a coefficient of correlation between the sound signals, on the basis of the values calculated for the respective sound signals; and a gain calculator configured to calculate reduction gain for the sound signals inputted from the plurality of microphones, on the basis of the coefficient of correlation.

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 estimate an electro-acoustical path from the noise canceling circuit through the transducer. A processing circuit uses the reference and/or error microphone, optionally along with a microphone provided for capturing near-end speech, to determine whether one of the reference or error microphones is obstructed by comparing their received signal content and takes action to avoid generation of erroneous anti-noise.

Abstract: Null processing noise subtraction is performed per sub-band and time frame for acoustic signals received from multiple microphones. The acoustic signals may include a primary acoustic signal and one or more additional acoustic signals. A noise component signal may be determined for each additional acoustic signal in each sub-band of signals received by N microphones by subtracting a desired signal component within every other acoustic signal weighted by a complex-valued coefficient ? from the secondary acoustic signal. The noise component signals, each weighted by a corresponding complex-valued coefficient ?, may then be subtracted from the primary acoustic signal resulting in an estimate of a target signal (i.e., a noise subtracted signal).

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: 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: Methods and systems for power conservation in mobile devices are presented. In one example, a mobile communication device includes a first microphone and a second microphone. The mobile communication monitors an ambient noise level and responsive to the ambient noise level operates the mobile communication device in a normal operation mode or a power conservation mode. In power conservation mode, use of the second microphone is discretionary.

Abstract: Systems and methods for controlling adaptivity of noise cancellation are presented. One or more audio signals are received by one or more corresponding microphones. The one or more signals may be decomposed into frequency sub-bands. Noise cancellation consistent with identified adaptation constraints is performed on the one or more audio signals. The one or more audio signals may then be reconstructed from the frequency sub-bands and outputted via an output device.

Abstract: The headset comprises two earpieces each having a transducer for playing back the sound of an audio signal and received in an acoustic cavity defined by a shell having an ear-surrounding cushion. The active noise control comprises, in parallel, a feedforward bandpass filter receiving the signal from an external microphone, a feedback bandpass filter receiving as input an error signal delivered by an internal microphone, and a stabilizer bandpass filter locally increasing the phase of the transfer function of the feedback filter in an instability zone, in particular a waterbed effect zone around 1 kHz. A summing circuit delivers a weighting linear combination of the signal delivered by these filters together with the audio signal to be played back. Control is non-adaptive, with the parameters of the filters being static.

Abstract: The present invention is directed to a wireless telephone having a first microphone and a second microphone and a method for processing audio signal in a wireless telephone having a first microphone and a second microphone. The wireless telephone includes a first microphone, a second microphone, and a signal processor. The first microphone outputs a first audio signal, the first audio signal comprising a voice component and a background noise component. The second microphone outputs a second audio signal. The signal processor increases a ratio of the voice component to the noise component of the first audio signal based on the content of at least one of the first audio signal and the second audio signal to produce a third audio signal.

Abstract: In one embodiment, a directional microphone array having (at least) two microphones generates forward and backward cardioid signals from two (e.g., omnidirectional) microphone signals. An adaptation factor is applied to the backward cardioid signal, and the resulting adjusted backward cardioid signal is subtracted from the forward cardioid signal to generate a (first-order) output audio signal corresponding to a beampattern having no nulls for negative values of the adaptation factor. After low-pass filtering, spatial noise suppression can be applied to the output audio signal. Microphone arrays having one (or more) additional microphones can be designed to generate second- (or higher-) order output audio signals.

Abstract: An echo prevention circuit comprises an input terminal to which a first input signal is input; a first FIR filter into which the first input signal is input through the input terminal; a second FIR filter into which the first input signal is input at the same time as into the first FIR filter; an input/output terminal to which an output signal of the first FIR filter is output or a second input signal is input; a subtracter that subtracts an output signal of the second FIR filter from a combined signal of the output signal of the first FIR filter and the second input signal; and an output terminal to which an output signal of the subtracter is output. The first and the second FIR filters have such filter coefficients that the output signal through the output terminal has only the output signal from the first FIR filter removed.

Abstract: The subject disclosure is directed towards a noise adaptive beamformer that dynamically selects between microphone array channels, based upon noise energy floor levels that are measured when no actual signal (e.g., no speech) is present. When speech (or a similar desired signal) is detected, the beamformer selects which microphone signal to use in signal processing, e.g., corresponding to the lowest noise channel. Multiple channels may be selected, with their signals combined. The beamformer transitions back to the noise measurement phase when the actual signal is no longer detected, so that the beamformer dynamically adapts as noise levels change, including on a per-microphone basis, to account for microphone hardware differences, changing noise sources, and individual microphone deterioration.

Abstract: The invention automatically enables and disables noise reduction based on a noise threshold. This threshold can be pre-defined by a user for a particular machine or can be defined “on the fly” before/during a telephonic conversation. With this flexibility, the users can “by-pass” the noise reduction and preserve the voice quality which are usually altered/modified by noise reduction algorithms. The present invention provides a novel system and method for monitoring the audio signals, analyze selected audio signal components, compare the results of analysis with a threshold value, and enable or disable noise reduction capability of a communication device.

Abstract: Unlike sound based pressure waves that go everywhere, air turbulence caused by wind is usually a fairly local event. Therefore, in a system that utilizes two or more spatially separated microphones to pick up sound signals (e.g., speech), wind noise picked up by one of the microphones often will not be picked up (or at least not to the same extent) by the other microphone(s). Embodiments of methods and apparatuses that utilize this fact and others to effectively detect and suppress wind noise using multiple microphones that are spatially separated are described.

Abstract: A mechanical noise suppression apparatus includes: a framing section adapted to divide an input signal into frames of a predetermined time length; a Fourier transform section adapted to transform framed signals obtained by the framing section into a frequency spectrum of a frequency domain; a mechanical noise reduction section adapted to correct the frequency spectrum of the input signal obtained by the Fourier transform section based on frequency spectrum information of mechanical noise to suppress the mechanical noise; an inverse Fourier transform section adapted to return the frequency spectrum corrected by the mechanical noise reduction section into framed signals of a time domain; and a frame synthesis section adapted to carry out frame synthesis of the framed signals of frames obtained by the inverse Fourier transform section to obtain an output signal in which the mechanical noise is suppressed.

Abstract: Disclosed herein are systems, methods, and non-transitory computer-readable storage media for suppressing spatial noise based on phase information. The method transforms audio signals to frequency-domain data and identifies time-frequency points that have a parameter (e.g., signal-to-noise ratio) above a threshold. Based on these points, unwanted signals can be attenuated the desired audio source can be isolated. The method can work on a microphone array that includes two microphones or more.

Abstract: The present invention discloses a microphone array structure able to reduce noise and improve speech quality and a method thereof. The method of the present invention comprises steps: using at least two microphone to receive at least two microphone signals each containing a noise signal and a speech signal; using FFT modules to transform the microphone signals into frequency-domain signals; calculating an included angle between a speech signal and a noise signal of the microphone signal, and selecting a phase difference estimation algorithm, a noise reduction algorithm or both to reduce noise according to the included angle; if the phase difference estimation algorithm is used, calculating phase difference of the microphone signals to obtain a time-space domain mask signal; and multiplying the mask signal and the average of the microphone signals to obtain the speech signals of the microphone signals. Thereby is eliminated noise and improve speech quality.

Abstract: An apparatus for providing an audio signal to drive a speaker system includes first and second audio channels. The first audio channel has a first class-D amplifier for receiving an input signal, and a first reconstruction filter for receiving an output from the first class-D amplifier and reconstructing therefrom an output audio signal for driving the speaker system. The second audio channel has a second class-D amplifier for receiving an input audio signal, and a second reconstruction filter for receiving an output from the second class-D amplifier and reconstructing therefrom an output audio signal for driving the speaker system. The first and second reconstruction filters have corresponding first and second planar inductors, with the second planer inductor being magnetically coupled to the first planar inductor.

Abstract: There is provided a noise cancellation system, comprising: an input for a digital signal, the digital signal having a first sample rate; a digital filter, connected to the input to receive the digital signal; a decimator, connected to the input to receive the digital signal and to generate a decimated signal at a second sample rate lower than the first sample rate; and a processor. The processor comprises: an emulation of the digital filter, connected to receive the decimated signal and to generate an emulated filter output; and a control circuit, for generating a control signal on the basis of the emulated filter output. The control signal is applied to the digital filter to control a filter characteristic thereof.

Abstract: Systems and methods improve audio signals and include means and methods of reducing stochastic noise in wideband audio signals. Multiple microphones may acquire near and far end audio signals, the audio signals may undergo transformations via a general or specialized digital signal processor.

Abstract: Embodiments of the present invention include methods and apparatuses for adjusting audio content when more multiple audio objects are directed toward a single audio output device. The amplitude, white noise content, and frequencies can be adjusted to enhance overall sound quality or make content of certain audio objects more intelligible. Audio objects are classified by a class category, by which they are can be assigned class specific processing. Audio objects classes can also have a rank. The rank of an audio objects class is used to give priority to or apply specific processing to audio objects in the presence of other audio objects of different classes.

Abstract: A system, method, and computer program product are provided for cleaning an audio segment. For a given audio segment, an offset amount is calculated where the audio segment is maximally correlated to the audio segment as offset by the offset amount. The audio segment and the audio segment as offset by the offset amount are averaged to produce a cleaned audio segment, which has had noise features reduced while having signal features (such as voiced audio) enhanced.

Abstract: Signals are received, over a range of angles, at an input of a device. The signals include a primary signal with a principal direction of arrival and an interfering signal with a respective interfering direction of arrival at the input. Measurements are determined for the received signals over the range of angles. Each measurement relates to a particular angle and indicating the energy of the received signals which are received from the particular angle. For each angle over the range of angles, a value is removed from the measurement for that angle, the value being based on the minimum of: (i) the energy of the measurement for that angle, and (ii) the energy of a corresponding measurement for a corresponding angle mirrored around the principal direction of arrival, whereby the remaining values of the plurality of measurements are indicative of said at least one interfering direction of arrival.

Abstract: There is provided an in-ear earphone comprising a housing having at least one electroacoustic transducer, a control unit which outputs a test signal to the electroacoustic reproduction transducer for reproduction, and a tightness measuring unit for measuring a parameter representative of the tightness of a fit of the earphone in an ear canal. The control unit is adapted to output a second audio signal as confirmation of a tight fit of the earphone to the electroacoustic reproduction transducer for reproduction.

Abstract: A radio receiver comprising: an antenna for receiving a radio frequency signal amplitude modulated with an audio frequency signal; a digitizer for periodically sampling the radio frequency signal and generating a digital reception signal representative of the amplitude of the radio frequency signal; and a demodulator for demodulating the digital reception signal to generate a representation of the audio frequency signal.