Abstract: Techniques are disclosed that enable processing of audio data to generate one or more refined versions of audio data, where each of the refined versions of audio data isolate one or more utterances of a single respective human speaker. Various implementations generate a refined version of audio data that isolates utterance(s) of a single human speaker by processing a spectrogram representation of the audio data (generated by processing the audio data with a frequency transformation) using a mask generated by processing the spectrogram of the audio data and a speaker embedding for the single human speaker using a trained voice filter model. Output generated over the trained voice filter model is processed using an inverse of the frequency transformation to generate the refined audio data.

Abstract: An AI device is provided. The AI device includes a memory to store data, a voice acquisition interface to acquire a voice signal, and a processor to perform preprocessing for the voice signal based on a parameter, to provide the preprocessed voice signal to a voice recognition model, to acquire a voice recognition result, to store a characteristic of the preprocessed voice signal in the memory, and to change the parameter using a distribution of characteristics of voice signals accumulated in the memory.

Abstract: A spatial audio processing system operable to enable audio signals to be spatially extracted from, or transmitted to, discrete locations within an acoustic space. Embodiments of the present disclosure enable an array of transducers being installed in an acoustic space to combine their signals via inverting physical and environmental models that are measured, learned, tracked, calculated, or estimated. The models may be combined with a whitening filter to establish a cooperative or non-cooperative information-bearing channel between the array and one or more discrete, targeted physical locations in the acoustic space by applying the inverted models with whitening filter to the received or transmitted acoustical signals. The spatial audio processing system may utilize a model of the combination of direct and indirect reflections in the acoustic space to receive or transmit acoustic information, regardless of ambient noise levels, reverberation, and positioning of physical interferers.

Abstract: A method for processing sound that includes, generating one or more noise component estimates relating to an electrical representation of the sound and generating an associated confidence measure for the one or more noise component estimates. The method further comprises processing, based on the confidence measure, the sound.

Abstract: Microphone signals are received from microphones of a computer device. Each microphone signal of the microphone signals is acquired by a respective microphone of the microphones. A previously unselected microphone is selected from the microphones as a reference microphone, which generates a reference microphone signal. An adaptive filter is used to create, based on microphone signals of the microphones other than the reference microphone, predicted microphone signals for the reference microphone. Based on the predicted microphone signals for the reference microphone, an enhanced microphone signal is outputted for the reference microphone. The enhanced microphone signal may be used as microphone signal for the reference microphone in subsequent audio processing operations.

Abstract: A hearing device for enhancing speech intelligibility, the hearing device includes: an input transducer for providing an input signal comprising a speech signal and a noise signal; a processing unit; an acoustic output transducer coupled to the processing unit, the acoustic output transducer configured to provide an audio output signal based on an output signal form the processing unit; wherein the processing unit is configured to determine one or more parameters of the input signal based on a codebook based approach (CBA) processing; and wherein the processing unit is configured to perform a Kalman filtering of the input signal based on the determined one or more parameters so that the output signal has an enhanced speech intelligibility.

Abstract: A parallel processing starting unit 3 that partitions an inputted image into tiles each having a predetermined size, and distributes tiles obtained through the partitioning, and N tile encoding units 5-1 to 5-N each of that carries out a prediction difference encoding process on a tile distributed thereto by the parallel processing starting unit 3 to generate a local decoded image are disposed, and each of N tile loop filter units 7-1 to 7-N determines a filter per tile suitable for a filtering process on the local decoded image generated by the corresponding one of the tile encoding units 5-1 to 5-N, and carries out the filtering process on the local decoded image by using the filter.

Abstract: A method for encoding an input audio stream including the steps of obtaining a first playback stream presentation of the input audio stream intended for reproduction on a first audio reproduction system, obtaining a second playback stream presentation of the input audio stream intended for reproduction on a second audio reproduction system, determining a set of transform parameters suitable for transforming an intermediate playback stream presentation to an approximation of the second playback stream presentation, wherein the transform parameters are determined by minimization of a measure of a difference between the approximation of the second playback stream presentation and the second playback stream presentation, and encoding the first playback stream presentation and the set of transform parameters for transmission to a decoder.

Abstract: In a method for testing an audio communication system of an aircraft, it is detected whether a jack plug of a headset is plugged into a jack of the audio communication system. Subject to the condition that no jack plug is detected, an electrical connection is made between an audio output of the jack and a microphone input of the jack by means of a test bridge circuit provided for at the jack. Furthermore, a test input signal is applied to an audio input connected to the audio output of the jack, and a test output signal is tapped off at a microphone output connected to the microphone input of the jack. The test output signal is used to ascertain a functional state of the audio communication system. Furthermore, an aircraft having an audio communication system is described.

Abstract: An earphone and connection cable can be connected to the housing of the earphone with a robust and reliable plug connection. The plug connection includes a coaxial plug having rotationally symmetrical plug contacts and is surrounded with a plastic encapsulation. When the plug connection is inserted the plastic encapsulation forms a first bearing location for receiving mechanical forces acting laterally on the plug while at least one of the plug contacts of the coaxial plug forms a second bearing location for receiving such forces. The distance from the first to the second bearing location in that case is greater than the diameter of the plug contacts of the coaxial plug.

Abstract: A method, an apparatus, and logic to post-process raw gains determined by input processing to generate post-processed gains, comprising using one or both of delta gain smoothing and decision-directed gain smoothing. The delta gain smoothing comprises applying a smoothing filter to the raw gain with a smoothing factor that depends on the gain delta: the absolute value of the difference between the raw gain for the current frame and the post-processed gain for a previous frame. The decision-directed gain smoothing comprises converting the raw gain to a signal-to-noise ratio, applying a smoothing filter with a smoothing factor to the signal-to-noise ratio to calculate a smoothed signal-to-noise ratio, and converting the smoothed signal-to-noise ratio to determine the second smoothed gain, with smoothing factor possibly dependent on the gain delta.

Abstract: A method performed by a wearable audio output device worn by a user is provided for controlling the reproduction of external noise and/or noise masking. A first a first lower-power detector determines if the user is in transport or a change in the user's transport. In response to the determination, a second higher-power detector exits a power saving state and determines a mode of the transport based. The wearable audio output device adjust a level of attenuation applied by the wearable audio output device to the external noise or adjusting a level of noise masking output by the wearable audio output device based on the determined mode of the transport.

Abstract: A method, system, and computer program product, include obtaining window status decision information based on vehicle interior background noise and determining switch status of the vehicle window(s) based on the obtained window status decision information.

Abstract: The disclosure provides a method and an apparatus for detecting a voice activity in an input audio signal composed of frames. A noise characteristic of the input signal is determined based on a received frame of the input audio signal. A voice activity detection (VAD) parameter is derived based on the noise characteristic of the input audio signal using an adaptive function. The derived VAD parameter is compared with a threshold value to provide a voice activity detection decision. The input audio signal is processed according to the voice activity detection decision.

Abstract: An ear-plug assembly presents audio content to an ear canal of a user. The audio content may be based in part on sound in a local area surrounding the user. The ear-plug assembly detects, via one or more acoustic sensors, sound in the area around the user. The sound waves travel through an aperture in a body of the ear-plug assembly and are propagated to a waveguide to the one or more acoustic sensors. The ear-plug assembly processes the detected sound data in a controller, which instructs a speaker assembly to present audio content based in part on the detected sound data. The detected sounds may be amplified, attenuated, filtered, and/or augmented when presented by the speaker assembly.

Abstract: A speech recognition method and an apparatus which recognize speech, based on speaker recognition, and output a result of the speech recognition are provided. The speech recognition method includes activating a session for receiving an input of an audio signal, performing speech recognition on a speech signal detected from the input audio signal while the session is maintained, determining whether a speaker of the speech signal is a registered speaker based on speaker information generated from the speech signal, determining whether to maintain the session based on a result of the determination, and outputting a result of performing the speech recognition.

Abstract: A sound signal processing device includes: a microphone terminal to which a sound signal derived from sound received by a microphone is input; a loudspeaker terminal from which a sound signal directed to a loudspeaker is output; a first input terminal to which a sound signal from another proximal-end device is input; a first output terminal from which a sound signal directed to the other device is output; a distal-end input terminal to which a distal-end sound signal is input via a network; a distal-end output terminal from which a sound signal directed to the network is output; and at least one processor configured to execute stored instructions to establish at least one signal path from at least one of the microphone terminal, the first input terminal, or the distal-end input terminal, to at least one of the loudspeaker terminal, the first output terminal, or the distal-end output terminal.

Abstract: A warning system for a hearing-impaired person of a vehicle includes: a sound sensing unit disposed at a vehicle to sense sound outside of the vehicle; a database storing characteristic information of a plurality of specific sounds and visual information corresponding to the characteristic information of the plurality of specific sounds; an output unit outputting visual information corresponding to the plurality of specific sounds; and a controller extracting characteristic information of the sound sensed by the sound sensing units. In particular, the controller determines a type of the sensed sound by comparing the extracted characteristic information of the sensed sound with the characteristic information of the plurality of specific sounds stored in the database, and outputs visual information corresponding to the determined type of the sensed sound through the output unit.

Abstract: A hearing device, a hearing system and a method for improving a hearing impaired person's ability to perceptually separate a target sound from competing sounds, the target sound and the competing sounds forming a composite sound signal having a given frequency range, where the method comprises the steps of: (i) subdividing the frequency range of the composite sound signal into a plurality of frequency sub-bands; (ii) grouping frequency sub-bands based on comparable characteristics of the plurality of frequency sub-bands; (iii) for each of the groups calculating a group envelope; and (iv) multiplying the signal in the frequency sub-bands of each individual group by a function or functions that enhance(s) peaks of the group envelope and/or attenuates energy in troughs in the group envelope. The comparable characteristics may be the correlation between the envelope of each of the bands in the specific group of frequency sub-bands and the corresponding group envelope.

Abstract: Provided are a method of recognizing a fingerprint and a device including the same. The method of recognizing a fingerprint includes obtaining a fingerprint image from a fingerprint sensor, determining whether a pixel in the fingerprint image is defective based on a dispersion value of the pixel with respect to time, in response to determining that the pixel is defective, generating an interpolated pixel value of the pixel by performing interpolation on a first pixel value of the pixel based on a second pixel value of another pixel different from the pixel, and correcting the fingerprint image based on the interpolated pixel value.

Abstract: In environments (such as acoustic and bioelectrical environments) characterized by multiple simultaneous sources, effective blind source separation from sensor response mixtures becomes difficult as the number of sources increases—especially when the true number of sources is both unknown and changing over time. However, in some environments, non-sensor information can provide useful hypotheses for some sources. Embodiments of the present invention provide an adaptive filtering architecture for validating such source hypotheses, extracting an estimated representation of source signals corresponding to valid hypotheses, and improving the separation of the remaining “hidden” source signals from the sensor response mixtures.

Abstract: A processing unit can acquire datasets from respective data sources, each having a respective unique data domain. The processing unit can determine values of a plurality of features based on the plurality of datasets. The processing unit can modify input-specific parameters or history parameters of a computational model based on the values of the features. In some examples, the processing unit can determine an estimated value of a target feature based at least in part on the modified computational model and values of one or more reference features. In some examples, the computational model can include neural networks for several input sets. An output layer of at least one of the neural networks can be connected to the respective hidden layer(s) of one or more other(s) of the neural networks. In some examples, the neural networks can be operated to provide transformed feature value(s) for respective times.

Abstract: An audio processing apparatus includes an audio input unit configured to input a plurality of audio signals, a reduction unit configured to reduce the amount of wind noise in the plurality of input audio signals, and a controller configured to control the reduction amount of the wind noise by the reduction unit based on a difference between an in-phase component and an antiphase component of the plurality of audio signals and control the speed of change of the reduction amount of the wind noise by the reduction unit based on a magnitude of the in-phase component of the plurality of audio signals. The audio processing apparatus reduces the adverse effects of the wind noise reduction process on the audio quality of the audio signals.

Abstract: A method includes: sensing a portion of a human body to generate a series of image frames by an optical sensing element; determining a complexion region from each of the image frames; then calculating a central coordinate of the complexion region; determining a sample block by expanding a surrounding range from the central coordinate, for getting a complexion fluctuation waveform; comparing the central coordinate with a central coordinate of a later image frame, to calculate a displacement in a unit time for determining whether the portion of the human body is in a static state. In the static state, an operating unit performs noise filter and peak detecting for complexion fluctuation waveform, and calculates a physiological signal. The method can reduce the amount of data required to be processed in physiological signal detection, and improve the accuracy of the detection.

Abstract: A method, an apparatus, and logic to post-process raw gains determined by input processing to generate post-processed gains, comprising using one or both of delta gain smoothing and decision-directed gain smoothing. The delta gain smoothing comprises applying a smoothing filter to the raw gain with a smoothing factor that depends on the gain delta: the absolute value of the difference between the raw gain for the current frame and the post-processed gain for a previous frame. The decision-directed gain smoothing comprises converting the raw gain to a signal-to-noise ratio, applying a smoothing filter with a smoothing factor to the signal-to-noise ratio to calculate a smoothed signal-to-noise ratio, and converting the smoothed signal-to-noise ratio to determine the second smoothed gain, with smoothing factor possibly dependent on the gain delta.

Abstract: A level detection unit detects a level of an audio signal. A level control unit controls the level of the audio signal according to a gain. A gain control unit performs a limit operation and a recovery operation according to the level of the audio signal. A noise generation unit outputs a noise signal. A combination unit combines the audio signal from the level control unit with the noise signal. In a second recovery mode, the gain control unit increases the gain in a longer time than in a first recovery mode. The gain control unit controls the noise generation unit to output a noise signal with a predetermined level in the first recovery mode, and to output a noise signal with a level according to the gain of the level control unit in the second recovery mode.

Abstract: Embodiments of the present disclosure provide a headphone and an interaction system. The headphone includes: a controller electrically connected to the Type-C interface, a biological features detection module connected to the controller, a microphone electrically connected to the controller, and a loudspeaker electrically connected to the controller. The biological features detection module is configured to detect biological features of a user wearing the headphone; and the controller is configured to control paring between the Type-C interface and a terminal and communication between the terminal and the biological features detection module, the microphone and the loudspeaker when the headphone is in a digital mode, to control detection of biological features and processing of audio data.

Abstract: A method and related processing system and input device are disclosed for power consumption optimization using interference measurements. The method comprises applying, within a predefined first low-power operational mode, a first set of values for at least one predefined sensing parameter and corresponding to a first power consumption level; acquiring, within the first low-power operational mode, a first interference measurement using the plurality of sensor electrodes; transitioning, upon determining the first interference measurement exceeds a first interference threshold value, into a predefined high-power operational mode; and applying, within the high-power operational mode, a second set of values for the at least one predefined sensing parameter and corresponding to a second power consumption level greater than the first power consumption level.

Abstract: A beamformer system that can isolate a desired portion of an audio signal resulting from a microphone array. A fixed beamformer is used to dampen diffuse noise while an adaptive beamformer is used to cancel directional coherent noise. A gain is calculated using a signal quality value such as signal-to-noise ratio, signal-to-null ratio or other value. The adaptive beamformer output is adjusted by the gain prior to combining the fixed beamformer output and the adaptive beamformer output to determine the output audio data.

Abstract: A noise cancellation device includes an anti-noise filter circuit, an output circuit, and a detection circuit. The anti-noise filter circuit provides a corresponding one of transfer functions to process a digital signal, in order to generate a noise cancellation signal, in which the transfer functions are different from each other. The output circuit mixes the noise cancellation signal, a reference signal, and an input signal to generate a mixed signal, and generates a sound output signal based on the mixed signal, in which the digital signal is associated with the sound output signal. The detection circuit controls the anti-noise filter circuit to provide the corresponding one of the transfer functions according to a comparison result of a first ratio and a first threshold value, in which the first ratio is a ratio of a first power of the mixed signal to a second power of the digital signal.

Abstract: A test apparatus for binaurally-coupled acoustic devices is disclosed. The apparatus includes a base, a lid, a primary speaker, and a binaural test fixture. The lid is coupled to the base and movable between a closed position in which the lid and base cooperate to form a closed sound chamber that is symmetric about a vertical mirror plane, and an open position. The primary speaker is coupled to one of the base and the lid, and faces a direction that lies on the mirror plane. The binaural test fixture is positioned inside the sound chamber, and includes first and second acoustic coupler mounts. The vertical mirror plane extends symmetrically between the first and second acoustic coupler mounts. A binaural test fixture, an acoustic coupler assembly, and a method of testing binaurally-coupled acoustic devices are also disclosed.

Abstract: An output stage circuit including an amplifier, a switching circuit and a selection circuit is provided. The amplifier converts an input signal into an output signal. The switching circuit is coupled between an output terminal of the amplifier and ground. The selection circuit includes a first-mode-selection-unit and a second-mode-selection unit. The first control signal is selected as a switching signal by the first-mode-selection unit when a power-detection signal is in a high level, so that the switching circuit is selectively turned on. The switching circuit is turned on according to a voltage stored in a storage capacitor of the second-mode-selection unit when the power-detection signal is in a low level, so that the output terminal of the amplifier is coupled to the ground.

Abstract: A method of operating a hearing aid system with virtually zero delay and phase distortion. The invention also provides a hearing aid system (100) adapted for carrying out such a method.

Abstract: One example discloses an apparatus for sound signal detection, comprising: a first wireless device including a first pressure sensor having a first acoustical profile and configured to capture a first set of acoustic energy within a time window; wherein the first wireless device includes a wireless signal input; wherein the first wireless device includes a processing element configured to: receive, through the wireless signal input, a second set of acoustic energy captured by a second pressure sensor, having a second acoustical profile, within a second wireless device and within the time window; apply a signal enhancement technique to the first and second sets of acoustic energy based on the first and second acoustical profiles; search for a predefined sound signal within the enhanced sets of acoustic energy; and initiate a subsequent set of sound signal detection actions if the search finds the sound signal.

Abstract: Provided is an electronic device including a communicator including communication circuitry configured to perform wireless communication with a peripheral device and to receive information indicating at least one audio connection protocol for connecting the peripheral device to the electronic device; a processor; a memory; and one or more programs comprising instructions, stored in the memory, which, when executed by the processor, cause the processor to perform operations corresponding to the instructions, the one or more programs including instructions for selecting one of the at least one audio connection protocol based on pre-set priorities of the audio connection protocols; and instructions for outputting an audio signal to the peripheral device based on the selected audio connection protocol.

Abstract: A system and method of blind bandwidth extension. The system selects a prediction model from a number of stored prediction models that were generated using an unsupervised clustering method (e.g., a k-means method) and a supervised regression process (e.g., a support vector machine), and extends the bandwidth of an input musical audio signal.

Abstract: Systems and techniques for removing non-stationary and/or colored noise can include one or more of the three following innovative aspects: (1) detection of an unwanted target signal, or component thereof, within an observed signal; (2) removal of the target (component) from the observed signal; and (3) filling of a gap in the observed signal generated by removal of the unwanted target (component). Removal regions, frequency bands, and/or regions of the observed signal used to train the gap filler can be adapted in correspondence with local characteristics of the observed signal and/or the target signal (component). Related aspects also are described. For example, disclosed noise detection and/or removal methods can include converting an incoming acoustic signal to a corresponding machine-readable form. And, a corrected signal in machine-readable form can be converted to a human-perceivable form, and/or to a modulated signal form conveyed over a communication connection.

Abstract: Embodiment apparatus and associated methods relate to automatically improving the quality of a segmented audio stream containing a desired signal, based on filtering an audio segment with an audio filter configured to remove a predetermined noise, and adapting the audio filter degree of noise removal determined as a function of a characteristic of the desired signal in the audio segment. In an illustrative example, the desired signal may be human voice. The predetermined noise may be, for example, wireless ring, hum, or tick resulting from a wireless microphone. In various implementations, the desired signal characteristic may be voice activity detected in the audio segment, and the audio filter degree of noise removal may be adapted as a function of the voice activity. Various examples may advantageously provide faster and more accurate vocal dialog editing in sound production procedures such as ADR (known as Alternative Dialog Replacement, or, Additional Dialog Recording).

Abstract: Techniques for online information search and retrieval for a query including a digital audio waveform. In an aspect, an audio waveform is received and digitized by at least one of a plurality of audio input devices. The digitized audio waveforms are transmitted to a central processing unit, which formulates and submits a query to an online engine. The formulated query may include the at least one digital audio waveform. The online engine retrieves one or more online results relevant to the formulated query. The online results may include one or more relevant visual results, and/or one or more relevant audio results. The retrieved results are served in real-time back to a user, via a device having audio output capability, and/or a device having visual data output capability.

Abstract: At least some embodiments are directed to a system comprising an amplifier containing a first bias current source and configured to provide an output voltage at a node, a gain stage coupled to the node and comprising a second bias current source, and a buffer stage coupled to the node and comprising third and fourth bias current sources and an additional set of bias current sources, the third and fourth bias current sources are able to activate output transistors that are configured to increase current provided to a load. The system also comprises a controller configured to activate the first bias current source, to activate the second bias current source after the first bias current source is activated, to activate the bias current sources in the set after the first bias current source is activated, and to activate the third and fourth bias current sources after the first and second bias current sources are activated and after the bias current sources in the set are activated.

Abstract: Systems and methods are presented for power supply noise cancellation in a microphone package. A first node on a MEMS microphone package is coupled to an inverted voltage signal of a power supply output. A summing node is connected to both the first node and a second node. The second node is coupled to a non-inverted voltage signal of the power supply output. Because the parasitic capacitance of the MEMS microphone package is substantially equivalent relative to both the first node and the second node, the summing node outputs a noise-reduced summed voltage signal. An adjustable trim module is coupled to the first node and configured to adjust a reduction of a power supply noise in the summed voltage signal from the summing node.

Abstract: A method and system for filtering a multi-channel audio signal having a speech channel and at least one non-speech channel, to improve intelligibility of speech determined by the signal. In typical embodiments, the method includes steps of determining at least one attenuation control value indicative of a measure of similarity between speech-related content determined by the speech channel and speech-related content determined by the non-speech channel, and attenuating the non-speech channel in response to the at least one attenuation control value. Typically, the attenuating step includes scaling of a raw attenuation control signal (e.g., a ducking gain control signal) for the non-speech channel in response to the at least one attenuation control value. Some embodiments are a general or special purpose processor programmed with software or firmware and/or otherwise configured to perform filtering in accordance the invention.

Abstract: The present invention relates to a method and an apparatus for processing an audio signal, and more particularly, to a method and an apparatus for processing an audio signal, which synthesizes an object signal and a channel signal and effectively binaural-render the synthesized signal.

Abstract: A system for noise removal is coupled to a signal unit that provides a digital signal. The noise removal system includes a transformation module to transform the digital signal into an f-digital signal, a threshold filter to generate a noiseless signal from the f-digital signal based on a threshold profile, and a signal synthesizer to provide a gain to the noiseless signal and to transform the noiseless signal into an output signal.

Abstract: A test apparatus for binaurally-coupled acoustic devices is disclosed. The apparatus includes a base, a lid, a primary speaker, and a binaural test fixture. The lid is coupled to the base and movable between a closed position in which the lid and base cooperate to form a closed sound chamber that is symmetric about a vertical mirror plane, and an open position. The primary speaker is coupled to one of the base and the lid, and faces a direction that lies on the mirror plane. The binaural test fixture is positioned inside the sound chamber, and includes first and second acoustic coupler mounts. The vertical mirror plane extends symmetrically between the first and second acoustic coupler mounts. A binaural test fixture, an acoustic coupler assembly, and a method of testing binaurally-coupled acoustic devices are also disclosed.

Abstract: Circuits, methods, and apparatus for grounding contacts in an audio jack. One example may provide a driver, such as a charge pump, driving a first transistor or switch coupled between a first contact in an audio jack and ground, and a second transistor or switch coupled between a second contact in the audio jack and ground. The first transistor or switch and second transistor or switch may be p-channel transistors or n-channel transistors depletion or enhancement-mode transistors, floating-gate transistors, MEMs, relays, or other switching devices. The first and second transistors or switches may be on and conducting when power is removed from the driver.

Abstract: Technologies are generally described for a noise cancellation scheme for voice communications. In some examples, a system configured to cancel background noise in voice communication may include a user device configured to capture a background sound; and a server configured to receive the background sound captured by the user device, identify a source of the background sound from among a plurality of predetermined sound sources, and generate a reference background sound based at least in part on the identified source.

Abstract: A first correlation matrix, which corresponds to observed signals, is calculated, a non-scan range is specified, a second correlation matrix, which corresponds to an acoustic signal from a sound source within the non-scan range, is estimated, a third correlation matrix, which corresponds to a target sound source within a scan range, is calculated by removing the second correlation matrix from the first correlation matrix, and a first spatial spectrum, which is a localization result, is calculated from the third correlation matrix. In the estimation, the second correlation matrix is estimated from direction vectors calculated from a direction range of the non-scan range and a second spatial spectrum, which is a localization result calculated immediately before the first spatial spectrum is calculated.

Abstract: A method performed by a processing system. The method includes generating silence signatures of audio signals from a plurality of device based on energy levels of the audio signals, providing the silence signatures to an interaction service, and out-putting interaction information corresponding to the devices.

Abstract: In one embodiment, a pickup system includes a wind detector and a wind suppressor. The wind detector has a plurality of analyzers each configured to analyze first and second input signals, and a combiner configured to combine outputs of the plurality of analyzers and issue, based on the combined outputs, a wind level indication signal indicative of wind activity. The analyzers can be selected from a group of analyzers including a spectral slope analyzer, a ratio analyzer, a coherence analyzer, a phase variance analyzer and the like. The wind suppressor has a ratio calculator configured to generate a ratio of the first and second input signals, and a mixer configured to select one of the first or second input signals and to apply to the selected input signal one of first or second panning coefficients based on the wind level indication signal and on the ratio.