Abstract: A method and apparatus for mobile emulator determination using sound fingerprinting is disclosed. The method includes a verification computer system receiving a transaction request from a computing device purporting to be a mobile device. Responsive to receiving the request, the verification computer system transmits a request for verification information to the computing device. The verification system includes information regarding a tone to be generated by a speaker of the computing device. Thereafter, verification information is received from the computing device. The verification information includes information tone information generated by the computing device, wherein the tone is, after generation, detected by a microphone. The verification system then verifies, based on the receive verification information, whether the information indicates that the computing device is a mobile device.

Abstract: The present disclosure relates to an apparatus and method for monitoring a space using a three-dimensional acoustic web, and to a method of emitting a plurality of acoustic signals, forming a three-dimensional acoustic web in a monitoring target space based on interference between acoustic waves, and recognizing a situation of the monitoring target space based on a change in measured acoustic signals.

Abstract: A system for sound localization can include a first electronic device having a microphone to detect a sound, and a second electronic device. A processor can be in communication with the first electronic device and the second electronic device. The processor can receive a first signal from the first electronic device corresponding to the detected sound, determine a location of origin of the detected sound based at least in part on the first signal, and provide a second signal to the second electronic device based at least in part on the location of origin.

Abstract: An acoustic system includes an in-vehicle device including a vehicle-side communicator; and an information processing device configured to be worn at an ear of a user and to communicate with the vehicle-side communicator. The information processing device includes: a sound pickup configured to pick up external sound; a sound emitter configured to emit sound to the ear of the user; a communicator configured to communicate with the in-vehicle device; and a computer. The computer is programmed to: determine, based on a result of communication with the in-vehicle device by the communicator, whether or not a predetermined condition is satisfied in a state where the information processing device is in a vehicle; and when determining that the predetermined condition is satisfied, cause the sound emitter to emit the external sound picked up by the sound pickup.

Abstract: A signal representing a sound can be received. A machine learning model can be run to identify that the sound triggers a reaction in a user hearing the sound. A preventive action can be automatically activated to mitigate the reaction. The user's reactions can be monitored. Responsive to determining that the user's reaction has been mitigated or suppressed, the preventive action can be deactivated. The machine learning model can be retrained using at least the signal as new training data.

Abstract: This application relates an activity detector (100) for detecting signal activity in an input audio signal (SIN), such as may be used for always-on speech detection. The activity detector has a first time-encoding modulator (TEM) 101 including a first hysteretic comparator (201) for generating a PWM (pulse-width modulation) signal based on the input audio signal. A second TEM (103) having a second hysteretic comparator (401) is arranged to receive a reference voltage (VMID) and generate a clock signal (SCLK). A time-decoding converter (102) receives the clock signal and generates count values of a number of cycles of the clock signal in periods defined by the PWM signal. An activity monitor (104) is responsive to a count signal (SCT) from the TDC 102 to determine whether the input audio signal comprises signal activity above a defined threshold.

Abstract: Techniques are described in which sensor data is used to determine one or more of background noise or occupancy associated with a passenger cabin of a vehicle. The sensor data, in turn, is used to determine an operating state for one or more components of the vehicle (e.g., pumps, compressors, fans, blowers, etc.) such that an amount of background noise within the passenger cabin is reduced (e.g., when a passenger/occupant is present). In various examples, the operating state of the component may operate in a different, though louder, state (e.g., higher efficiency, greater power, etc.) when an occupant is not present or proximate the component.

Abstract: Provided is an electronic device. The electronic device includes a motor, a memory configured to store at least one of first frequency information associated with a sound that is generated when the motor is driven or second frequency information associated with a surrounding environment of the electronic device, and a processor configured to control the motor to be driven by avoiding a driving speed corresponding to the frequency information stored in the memory.

Abstract: Embodiments include a microphone array with a plurality of microphone elements comprising a first set of elements arranged along a first axis, comprising at least two microphone elements spaced apart by a first distance; a second set of elements arranged along the first axis, comprising at least two microphone elements spaced apart by a second, greater distance, such that the first set is nested within the second set; a third set of elements arranged along a second axis orthogonal to the first axis, comprising at least two microphone elements spaced apart by the second distance; and a fourth set of elements nested within the third set along the second axis, comprising at least two microphone elements spaced apart by the first distance, wherein each set includes a first cluster of microphone elements and a second cluster of microphone elements spaced apart by the specified distance.

Abstract: The technology described in this document can be embodied in a computer-implemented method that includes receiving identification information associated with a geographic location. The identification information includes one or more features that affect an acoustic environment of the geographic location at a particular time. The method also includes determining one or more parameters representing at least a subset of the one or more features, and estimating at least one acoustic parameter that represents the acoustic environment of the geographic location at the particular time. The at least one parameter can be estimated using a mapping function that generates the estimate of the at least one acoustic parameter as a weighted combination of the one or more parameters. The method further includes presenting, using a user-interface displayed on a computing device, information representing the at least one acoustic parameter estimated for the geographic location for the particular time.

Abstract: A compensation system for an implantable actuator is disclosed where the implantable actuator includes a sealed housing containing a driving arrangement for the actuator. The compensation system includes an external pressure sensor for measuring an external pressure outside of the sealed housing and a compensation module for determining a compensation factor for the implantable actuator based on the external pressure. In one embodiment, the compensation is directed to a direct acoustic cochlear stimulation (DACS) implantable actuator.

Abstract: Systems and methods for digital signal processing using a sliding windowed infinite Fourier transform (“SWIFT”) algorithm are described. A discrete-time Fourier transform (“DTFT”) of an input signal is computed over an infinite-length temporal window that is slid from one sample in the input signal to the next. The DTFT with the temporal window at a given sample point is effectively calculated by phase shifting and decaying the DTFT calculated when the temporal window was positioned at the previous sample point and adding the current sample to the result.

Abstract: A computing device comprising a processor, the processor configured to: receive, from an image capture system, an image captured in an environment and image metadata associated with the image, the image metadata comprising an image capture time; receive a sound recognition message from a sound recognition module, the sound recognition message comprising (i) a sound recognition identifier indicating a target sound or scene that has been recognised based on captured audio data captured in the environment, and (ii) time information associated with the sound recognition identifier; detect that the target sound or scene occurred at a time that the image was captured based on the image metadata and the time information in the sound recognition message; and output a camera control command to said image capture system based on said detection.

Abstract: A system and method for matching audio content to virtual reality visual content. The method includes analyzing received visual content and received metadata to determine an optimal audio source associated with the received visual content; configuring the optimal audio source to capture audio content; synthesizing the captured audio content with the received visual content; and providing the synthesized captured audio content and received visual content to a virtual reality (VR) device.

Abstract: A system automatically controls an electronic device's audio by detecting an active sound source presence within an auditory detection space. The system transitions the electronic device to selectively output a desired sound when the active sound source presence is detected and detects sound in the auditory detection space. The system enhances sound and transforms it into electrical signals. The system converts the electrical signals into a digital signal and identifies active sound segments in the digital signals. The system attenuates noise components in the digital signals and locates the physical location of the active sound source. It adjusts an output automatically by muting a second sound source in a second detection space.

Abstract: The present invention relates to a content producing device and method for matching and storing music information when an electronic device captures an image and, particularly, to a content producing device and method for storing, together with a captured image, information on music played by an electronic device or around the electronic device when the image is captured. According to one embodiment of the present disclosure, a control method for an electronic device comprises the steps of: capturing an image when a photographing instruction is inputted by a user; acquiring, during capturing of the image, sound source information on music played in a space in which the electronic device is located; and matching the sound source information on music to the captured image and storing the same.

Abstract: A sheet conveying device includes conveyance passage forming bodies to define a sheet conveyance passage, a nip forming body to form a conveying nip region in the sheet conveyance passage, and a flexible plate body having a support portion to be attached to another body and having a downstream-side end portion as a free end in a sheet conveying direction. The support portion of the flexible plate body is located on an upstream side in the sheet conveying direction with respect to the conveying nip region. The free end being located on a downstream side in the sheet conveying direction with respect to the conveying nip region.

Abstract: A footsteps tracking method, including the steps of: receiving a first sound signal of a user's first footstep; calculating a first position of the first footstep according to relative position relationship of at least three microphones in the microphone array and time differences of sound arrival of the first sound signal received by the three microphones respectively; receiving a second sound signal of a second footstep of the user, wherein an audio frequency of the second sound signal is the same as an audio frequency of the first sound signal; and calculating a second position of the second footstep according to the first position, a time difference between receiving the first sound signal and the second sound signal, receiving angles between the first sound signal and a pair of the three microphones, and receiving angles between the second sound signal and the pair of the three microphones.

Abstract: In some examples, a content consumer device configured to play one or more of a plurality of audio streams includes a memory configured to store the plurality of audio streams and audio location information associated with the plurality of audio streams and representative of audio stream coordinates in an acoustical space where an audio stream was captured or synthesized or both. Each of the audio streams is representative of a soundfield. The content consumer device also includes one or more processors coupled to the memory, and configured to determine device location information representative of device coordinates of the content consumer device in the acoustical space. The one or more processors are configured to select, based on the device location information and the audio location information, a subset of the plurality of audio streams, and output, based on the subset of the plurality of audio streams, one or more speaker feeds.

Abstract: A method for detecting blocking of a microphone and related products are provided. The method includes the following. Voice data is collected through a microphone of the first wireless earphone. Whether the voice data has a missing voice segment is detected. The microphone (e.g., microphone-hole) of the first wireless earphone is determined to be blocked in response to detecting that the voice data has the missing voice segment.

Abstract: A device that includes an adaptive acoustic detection circuit and an acoustic sensor device such as a microphone is described. The device includes in addition to the sensor a circuit configured to detect when an input stimulus to the sensor satisfies an adaptive threshold, and further configured to produce a signal upon detection that causes adjustment of performance of the device, wherein the adaptive threshold is a threshold value that varies over time in accordance with detected changes to sound of an environment in which the device is located.

Abstract: A driving assistance system includes a plurality of vehicles on which a plurality of microphones is mounted respectively and a server having an acquisition unit configured to acquire sound signals recorded by the microphones and position information of the vehicles. The server further has an estimation unit configured to estimate a position of one or more sound sources based on the sound signals and the position information and a providing unit configured to provide the estimated position of the one or more sound sources to the vehicles.

Abstract: A method is for the classification of temporally sequential digital audio data that describe acoustic signals that identify hazardous situations. The method includes calculating a large number of frequency spectrograms for stepwise progressive time intervals of the temporally sequential audio data, and forming a specific number of frequency segments for each octave of each individual frequency spectrogram. The frequency segments include a subset of the individual frequency spectrograms. The method further includes adding corresponding frequency segments of the octaves of each individual frequency spectrogram, calculating frequency components through the formation of mean values for the individual, added frequency segments in each individual frequency spectrogram, and generating a classification vector using a classifier and the number of frequency components of the large number of frequency spectrograms.

Abstract: A leakage inspection device according to an aspect of the present disclosure includes: at least one memory storing a set of instructions; and at least one processor configured to execute the set of instructions to: determine a measurement time; measure vibration waveforms for the measurement time by using at least two sensors set to a pipe; calculate a cross-correlation function of the measured vibration waveforms; detect peaks of the cross-correlation function at least twice in the measurement time; and determine that leakage occurs in a case where the peaks are repeatedly detected in the measurement time.

Abstract: An audio device for receiving radio communication. The audio device is configured to receive radio communication as a received radio signal. The audio device includes a hear-through element configured to provide a hear-through signal to a user in response to a received ambient sound signal, and an adaptive auto-gain element configured to perform an auto-gain function of the received radio signal according to an adaptive gain value resulting in a modified radio signal, and to set the adaptive target level for the auto-gain function in response to the hear-through signal.

Abstract: A sound inspection system that can reduce power consumption is provided. A sound inspection system 1 that determines a state based on a sound of an inspection target object 2 includes a sound sensor device 10 that collects the sound of the inspection target object 2, analyzes the collected sound, and transmits an analysis result, and a sound data determination device 30 that determines a state of the inspection target object based on the analysis result from the sound sensor device. The sound sensor device transmits, as the analysis result, intensities of the collected sound for each predetermined frequency set in advance.

Abstract: An electronic device displays a user interface for controlling an audio mode for a wearable audio output device. While displaying the user interface, the device electronic outputs first audio using a first audio mode while displaying in the user interface a first indication that the first audio mode is used for the first audio. While displaying the user interface and outputting the first audio using the audio mode, the electronic device receives a request to switch to using a second audio mode. In response, the device outputs second audio using the second audio mode while displaying a second indication that the second audio mode is used for the second audio. While outputting the second audio using the second audio mode and displaying the second indication, the electronic device detects an input, and, in response, selects one of the first audio mode or the second audio mode the active audio mode.

Abstract: Example techniques involve calibration of one or more playback devices. An example implementation involves a playback device playing back audio content using a first calibration via the one or more audio transducers and the one or more amplifiers. The first calibration is based on a response of a listening environment to audio content playback by the playback device. The playback device records, via one or more microphones, at least a portion of the played back audio content. Based on the recorded audio content, the playback device detects a change in the response of the listening environment to audio content playback by the playback device. Responsive to detecting the change in the response, the playback device causes output of a prompt to initiate a calibration procedure for the playback device. The calibration procedure involves a mobile device recording playback by the playback device.

Abstract: In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

Abstract: Apparatus and methods for bone conduction detection are disclosed herein. An example apparatus includes a first sensor to output a first vibration signal associated with a first bone structure of a user, a second sensor to output a second vibration signal associated with a second bone structure of the user, memory, and at least one processor to execute instructions to determine a sound associated with at least one of the first vibration signal or the second vibration signal originated external to the user; determine a direction from which the sound originated relative to the user; and generate a signal to cause an output device to present an alert to the user, the alert indicative of the direction of the sound.

Abstract: A method is provided. Intermediate audio features are generated from respective segments of an input acoustic time series for a same scene. Using a nearest neighbor search, respective segments of the input acoustic time series are classified based on the intermediate audio features to generate a final intermediate feature as a classification for the input acoustic time series. Each respective segment corresponds to a respective different acoustic window. The generating step includes learning the intermediate audio features from Multi-Frequency Cepstral Component (MFCC) features extracted from the input acoustic time series, dividing the same scene into the different windows having varying MFCC features, and feeding the MFCC features of each window into respective LSTM units such that a hidden state of each respective LSTM unit is passed through an attention layer to identify feature correlations between hidden states at different time steps corresponding to different ones of the different windows.

Abstract: In one aspect, a network microphone device includes a plurality of microphones and is configured to detect sound via the one or more microphones. The network microphone device may capture sound data based on the detected sound in a first buffer, and capture metadata associated with the detected sound in a second buffer. The network microphone device may classify one or more noises in the detected sound and cause the network microphone device to perform an action based on the classification of the respective one or more noises.

Abstract: An image capture device includes a sensor, a microphone array, and a processor. The microphone array may include a first microphone, a second microphone, a third microphone, or any combination thereof. The first microphone may be configured to face a first direction. The second microphone may be configured to face a second direction. The second direction may be diametrically opposed to the first direction. The third microphone may be configured to face a third direction. The third direction may be substantially perpendicular to the first direction, the second direction, or both. The processor may be configured to determine a microphone capture pattern and detect wind noise. The microphone capture pattern may be determined based on data obtained from the sensor. The sensor data may include image data, audio data, image capture device orientation data, location data, accelerometer data, or any combination thereof.

Abstract: A method includes receiving a volume modifier value; when the volume modifier value is in a soft sound emphasizer range, modifying a volume control curve by increasing a value of the volume control curve in a range below a reference value, the volume control curve defining how much a sound signal with a specific volume is amplified to a desired volume; when the volume modifier value is in a loud sound limiter range, modifying the volume control curve by increasing a value of the volume control curve in a range above a reference value; and applying the modified volume control curve to a sound processor of a hearing device, such that a sound signal processed by the hearing device and output by the hearing device to the user is amplified according to the modified volume control curve.

Abstract: A microphone assembly for a vehicle headliner includes a housing arranged to be received within a substrate layer of the headliner and having an upper portion and a lower portion. A circuit board is mounted in the upper portion and has a microphone element coupled thereto. An insert bracket includes a base and a shaft member extending upwardly therefrom, the base having a plurality of apertures aligned with the shaft member, wherein the shaft member engages the lower portion to connect the insert bracket to the housing. A sealing gasket having at least one channel defining an air path extending therethrough is arranged to be received within the shaft member and extend between the base and the upper portion, providing acoustic sealing between the insert bracket and the housing such that the air path directs sound from a cabin of the vehicle through the apertures to the microphone element.

Abstract: A method of inspecting a sound input/output device is disclosed. A method of inspecting a sound input/output device according to an embodiment of the present disclosure can diagnose an error state of either a speaker or a microphone based on a cross-correlation of input/output signals by receiving a sound signal from an AI device through the microphone. The method of inspecting of the present disclosure may be associated with an artificial intelligence module, a drone ((Unmanned Aerial Vehicle, UAV), a robot, an AR (Augmented Reality) device, a VR (Virtual Reality) device, a device associated with 5G services, etc.

Abstract: A method for spatialized sound reproduction using an array of loudspeakers, to generate a chosen sound field selectively audible in a first predetermined sub-area of an area covered by the loudspeakers, the area further including a second sub-area in which the chosen sound field is inaudible. The loudspeakers are supplied with respective control signals so that each one continuously emits an audio signal. The method includes, iteratively and continuously: estimating a sound pressure in the second sub-area as a function of the respective control signals of the loudspeakers, and of a respective initial weight of the control signals; calculating an error between the estimated sound pressure and a target sound pressure in the second sub-area; and, calculating and applying respective weights to the control signals, based on the error. The sound pressure in the second sub-area is calculated again based on the respective weighted control signals of the loudspeakers.

Abstract: A method for detecting wind using a microphone and a speaker of an electronic device. The method obtains a microphone signal produced by the microphone. The method obtains a speaker input signal produced by the speaker that is emulating a microphone capturing ambient sound in an environment through the speaker. The method determines a coherence between the microphone signal and the speaker input signal and determines whether the coherence is below a coherence intensity threshold. In response to determining that the coherence is below the coherence intensity threshold, the method determines a presence of wind in the environment.

Abstract: In an embodiment, a method comprises: capturing, by one or more microphone arrays of a vehicle, sound signals in an environment; extracting frequency spectrum features from the sound signals; predicting, using an acoustic scene classifier and the frequency spectrum features, one or more siren signal classifications; converting the one or more siren signal classifications into one or more siren signal event detections; computing time delay of arrival estimates for the one or more detected siren signals; estimating one or more bearing angles to one or more sources of the one or more detected siren signals using the time delay of arrival estimates and a known geometry of the microphone array; and tracking, using a Bayesian filter, the one or more bearing angles. If a siren is detected, actions are performed by the vehicle depending on the location of the emergency vehicle and whether the emergency vehicle is active or inactive.

Abstract: An adaptive noise filtering system and method detect sounds using a sensor onboard a vehicle system. A value of a signal associated with operation of the vehicle system is determined. One or more sounds detected by the sensor are filtered out based on the value of the signal that is determined. Operation of the vehicle system may be controlled using the remaining sound(s).

Abstract: Disclosed are video playback and data providing methods in a virtual scene, and a client and server implementing the same. The video playback method comprises: receiving current video segment data sent from a server, wherein the current video segment data represents a video segment, and the current video segment data comprises at least one specified viewing angle and a data identifier representing video segment data to which the specified viewing angle is directed; playing the video segment represented by the current video segment data, and acquiring a current viewing angle of a user during playback; and determining a target specified viewing angle matching the current viewing angle from the at least one specified viewing angle, wherein a video segment represented by video segment data to which the target specified viewing angle is directed is played at the end of the playback of the video segment represented by the current video segment data.

Abstract: An audio signal control apparatus includes a sound input circuit, a sound output circuit, and a control circuit. The sound input circuit is configured to receive an ambient sound signal from a microphone that captures ambient sound. The sound output circuit is configured to generate an output sound signal and transmit the output sound signal to a speaker to generate sound based on the output sound signal. The control circuit is configured to determine characteristics of the ambient sound based on the ambient sound signal received by the sound input circuit, determine a frequency band of the sound to be generated by the speaker based on the characteristics of the ambient sound, and control the sound output circuit to generate an output sound signal corresponding to the sound of the determined frequency band.

Abstract: A headphone volume control method calculates a level of a sound signal, which is captured by a microphone, during a first period, determines a target value of a sound volume based on the above-mentioned level, and gradually changes a reproduction volume of content to be reproduced to the above-mentioned target value over a second period.

Abstract: A processor, that may include at least one neural network that comprises at least one leaky spiking neuron; wherein the at least one leaky spiking neuron is configured to directly receive an input pulse density modulation (PDM) signal from a sensor; wherein the input PDM signal represents a detected signal that was detected by the sensor; and wherein the at least one neural network is configured to process the input PDM signal to provide an indication about the detected input signal.

Abstract: A scalable, distributed load control system for home automation based on a network of microphones may include control devices (e.g., load control devices) that may include microphones for monitoring the system and communicating audio data to a cloud server for processing. The control devices of the load control system may receive a single voice command and may be configured to choose one of the load control devices to transmit the voice command to the cloud server. The load control devices may be configured to receive a voice command, control a connected load according to the voice command if the voice command is a validated command, and transmit the voice command to a voice service in the cloud if the voice command is not a validated command. The voice service to which the load control devices transmit audio data to may be selectable.

Abstract: Disclosed is a microphone-mounted earphone including a housing, a speaker unit accommodated in the housing, and a microphone. Here, the housing includes a sound emission path formed in front of the speaker unit to emit a sound generated by the speaker unit toward the outside. The housing includes a microphone installation groove recessed from an outer surface of a front of the speaker unit toward the sound emission path. The microphone is installed in the microphone installation groove. Also, the microphone-mounted earphone further includes a cover configured to cover the microphone installation groove not to allow the microphone to be exposed outward.

Abstract: A sound collection device includes a sensor, a database, a microphone, and an electronic controller. The sensor detects a state of at least one of the sound collection device or a device equipped with the sound collection device, or both. The database is a database of noise sounds. The electronic controller includes a signal processing unit configured to read at least one noise sound from the database based on a detection value of the sensor and carry out a noise reduction process to reduce noise from a sound signal acquired by the microphone based on the at least one noise sound read from the database.

Abstract: An audio playback device comprises a microphone, a speaker, and a processor. The processor is arranged to output by the speaker first audio content and receive by the microphone an indication of the first audio content. A first acoustic response of a room in which the audio playback device is located is determined based on the received indication of first audio content. A mapping is applied to the first acoustic response to determine a second acoustic response. The second acoustic response is indicative of an approximated acoustic response of the room at a spatial location different from a spatial location of the microphone. The second audio content output by the speaker is adjusted based on the second response.

Abstract: The present invention provides a signal processing device that allows a user, who adjusts a signal level to match a reference level, to adjust the signal level while checking a degree of difference between the signal level and the reference level. A signal processing device according to the present invention comprises: an input part 20 to which an input signal is input; a storage 40 that stores therein one or more setting values of a parameter to be used for amplifying the input signal and a reference value of the setting value; an amplification part 30 that generates an output signal by amplifying a level of the input signal based on the setting value; an output part 70 that outputs the output signal; an adjustment part 50 that adjusts the setting value; and a display 60 that displays a display result determined based on the setting value adjusted by the adjustment part and the reference value.

Abstract: Method and apparatus for generating a model, and a method and apparatus for generating information are disclosed, an embodiment includes: acquiring a training sample set for an audio frame set, an audio frame in the audio frame set being in one-to-one correspondence with a training sample in the training sample set, the training sample in the training sample set including feature data and identification information of the audio frame in the audio frame set, and the identification information being used to identify a non-voice audio, a consonant audio and a vowel audio in the audio frame; and training to obtain a voice recognition model using a machine learning algorithm, by taking the feature data included in the training sample in the training sample set as an input, and the identification information corresponding to the input feature data as a desired output.