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: 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: 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: 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.

Abstract: A vehicle may include a speaker; a display; a plurality of microphones configured to receive sound waves outside the vehicle; and a controller connected to the speaker, the display, and the plurality of microphones, and configured to determine sound wave characteristics of a terrain around a road on which the vehicle travels, based on map information, to determine a direct wave and a reflected wave of the received sound waves based on the terrain, the determined sound wave characteristics of the terrain, and the received sound waves, to determine a position and a velocity of an object that has generated the received sound waves based on the direct wave and the reflected wave, and to control at least one of the speaker and the display to output information on the position and the velocity of the object.

Abstract: Some disclosed systems may include a microphone system having two or more microphones, an interface system and a control system. In some examples, the control system may be capable of receiving, via the interface system, audio data from two or more microphones of the microphone system, of determining a gesture location based, at least in part, on the audio data and of controlling one or more settings of the system based on the gesture location.

Abstract: This disclosure describes, in part, techniques to process audio signals to lessen the impact that wind and/or other environmental noise has upon the resulting quality of these audio signals. For example, the techniques may determine a level of wind and/or other noise in an environment and may determine how best to process the signals to lessen the impact of the noise, such that one or more users that hear audio based on output of the signals hear higher-quality audio.

Abstract: Confirmation can be made what sound has been made under a restriction in which transmittable traffic is small. An abnormal sound detection system including an artificial sound creating function is configured, the abnormal sound detection system including a statistic calculation unit configured to calculate a statistic set expressing sizes of a direct current component, an alternating current component, and a noise component in an amplitude time series at each of frequencies of a sound inputted at a terminal, a statistic transmitting unit configured to transmit the statistic set from the terminal to a server, a statistic receiving unit configured to receive the statistic set in the server, and an artificial sound reproducing unit configured to reproduce a cyclostationary artificial sound based on the statistic set received in the server.

Abstract: Described herein is a speakerphone system (system) comprising: at least one loudspeaker, the loudspeaker adapted to generate mechanical vibrations on an enclosure of the system; at least one microphone (mic) adapted to convert an input sound acoustic signal into an input sound electrical signal and adapted to convert the mechanical vibrations into a mechanical vibrations electrical signal, and to output both of the input sound electrical signal and the mechanical vibrations electrical signal as a mic output signal; at least one mechanical vibration sensor (MVS) adapted to convert the mechanical vibrations to a mechanical vibration error signal to output the mechanical vibration error signal as an MVS output signal; and circuitry adapted to subtract the MVS output signal from the mic output signal and output the resultant signal as a speakerphone output signal.

Abstract: An information processing device having a to-be-worn portion capable of being worn at one ear of a user, the information processing device comprising: a first sound pickup section configured to pick up external sound; a sound emission section configured to emit sound to the one ear of the user; and a control section configured to determine 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 emission section to emit the external sound picked up by the first sound pickup section.

Abstract: A method or apparatus for delivering audio programming such as music to listeners may include identifying, capturing and applying a listener's audiometric profile to transform audio content so that the listener hears the content similarly to how the content was originally heard by a creative producer of the content. An audio testing tool may be implemented as software application to identify and capture the listener's audiometric profile. A signal processor may operate an algorithm used for processing source audio content, obtaining an identity and an audiometric reference profile of the creative producer from metadata associated with the content. The signal processor may then provide audio output based on a difference between the listener's and creative producer's audiometric profiles.

Abstract: A system for removing noise from an audio signal is described. For example, noise caused by content playing in the background during a voice command or phone call may be removed from the audio signal representing the voice command or phone call. By removing noise, the signal to noise ratio of the audio signal may be improved.

Abstract: This disclosure falls into the field of audio coding, in particular it is related to the field of providing a framework for providing loudness consistency among differing audio output signals. In particular, the disclosure relates to methods, computer program products and apparatus for encoding and decoding of audio data bitstreams in order to attain a desired loudness level of an output audio signal.

Abstract: Embodiments may relate to a wearable audio device. The wearable audio device may be configured to generate a data related to a sound environment in which the wearable audio device is located, and transmit an indication of that data to a computing device that is located remote from the wearable audio device. The wearable audio device may further be able to receive an indication of an audio signal parameter that is based on the data, and generate an audio signal based on the audio parameter. Other embodiments may be described or claimed.

Abstract: A method, system and computer-usable medium for utilizing personalized audio selection to facilitate achieving a target state comprising: identifying a target state for a user; identifying a set of audio selections for the user to achieve the target state; monitoring reactions by the user to the set of audio selections to determine an assessment; and responsive to determining when the assessment is below a desired level, adjusting the set of audio selections.

Abstract: A method includes wirelessly receiving audio content via a first transceiver included in a housing of a first device and producing a first audible signal from the audio content via a speaker included in the housing of the first device. The audio content is wirelessly transmitted via the first transceiver to a second transceiver integrated within a first remote speaker. A second audible signal is produced from the audio content via the first remote speaker. The method also includes wirelessly relaying the audio content via the second transceiver to a third transceiver integrated within a second remote speaker and producing a third audible signal from the audio content via the second remote speaker. The second remote speaker is located outside of the transmission range of the first transceiver.

Abstract: Disclosed are a sound source focus method and device in which the sound source focus device, in a 5G communication environment by amplifying and outputting a sound source signal of a user's object of interest extracted from an acoustic signal included in video content by executing a loaded artificial intelligence (AI) algorithm and/or machine learning algorithm. The sound source focus method includes playing video content including a video signal including at least one moving object and the acoustic signal in which sound sources output by the object are mixed, determining the user's object of interest from the video signal, acquiring unique sound source information about the user's object of interest, extracting an actual sound source for the user's object of interest corresponding to the unique sound source information from the acoustic signal, and outputting the actual sound source extracted for the user's object of interest.

Abstract: An audio system generates customized head-related transfer functions (HRTFs) for a user. The audio system receives an initial set of estimated HRTFs. The initial set of HRTFs may have been estimated using a trained machine learning and computer vision system and pictures of the user's ears. The audio system generates a set of test locations using the initial set of HRTFs. The audio system presents test sounds at each of the initial set of test locations using the initial set of HRTFs. The audio system monitors user responses to the test sounds. The audio system uses the monitored responses to generate a new set of estimated HRTFs and a new set of test locations. The process repeats until a threshold accuracy is achieved or until a set period of time expires. The audio system presents audio content to the user using the customized HRTFs.

Abstract: A system, method and computer product for estimating a component of a provided audio signal. The method comprises converting the provided audio signal to an image, processing the image with a neural network trained to estimate one of vocal content and instrumental content, and storing a spectral mask output from the neural network as a result of the image being processed by the neural network. The neural network is a U-Net. The method also comprises providing the spectral mask to a client media playback device, which applies the spectral mask to a spectrogram of the provided audio signal, to provide a masked spectrogram. The media playback device also transforms the masked spectrogram to an audio signal, and plays back that audio signal via an output user interface.

Abstract: Various implementations include audio devices and related computer-implemented methods for controlling playback of augmented reality (AR) audio. In some cases, a method includes: receiving a first audio notification from one of a set of applications; determining an importance value for the first audio notification; detecting an activity state of the audio device, the activity state assigned to a notification importance threshold; and either: delaying release of the first audio notification at the audio device in response to the importance value failing to meet the notification importance threshold, or releasing the first audio notification at the audio device in response to the importance value meeting or exceeding the notification importance threshold.

Abstract: A gunshot detection/security system for detecting dangerous events in a school or other building includes one or several pods placed throughout the building premises. Each pod includes a camera, a thermal camera, and an acoustic sensor for detecting video, images, heat signatures, and sound within a detection area for the respective pod. The sensor data is then analyzed to identify a dangerous event in the building, and provide alerts regarding the dangerous event via the pods or client computing devices of students/occupants in the building, administrators, parents, and emergency responders. A server computing device generates digital maps of the interior and exterior of the building having location information regarding the occupants of the building and a danger zone indicating the epicenter for the dangerous event.

Abstract: Methods, systems, and devices for auditory enhancement are described. A device may receive a respective auditory signal at each of a set of microphones, where each auditory signal includes a respective representation of a target auditory component and one or more noise artifacts. The device may identify a directionality associated with a source of the target auditory component (e.g., based on an arrangement of the multiple microphones). The device may determine a distribution function for the target auditory component based at least in part on the directionality associated with the source and on the received plurality of auditory signals. The device may generate an estimate of the target auditory component based at least in part on the distribution function and output the estimate of the target auditory component.

Abstract: A protected extended playback mode protects the integrity of audio and side information of a spatial audio signal and sound object and position information of audio objects in an immersive audio capture and rendering environment. Integrity verification data for audio-related data determined. An integrity verification value is computable dependent on the transmitted audio-related data. The integrity verification value can be compared with the integrity verification data for verifying the audio-related data transmitted in the audio stream for generating a playback signal having a mode dependent on the verification of the audio-related data A transmitting device transmits that integrity verification data and the audio-related data in an audio stream for reception by a receiving device. The audio stream, including the audio-related data and integrity verification data are received by the receiving device.

Abstract: According to one embodiment, a sound processing apparatus extracts a feature of first speech uttered outside an objective area from first speech obtained at positions different from each other in a space of the objective area and a place outside the objective area. The apparatus creates, by learning, a determination model configured to determine whether an utterance position of second speech in the space is outside the objective area based at least in part on the feature uttered outside the objective area. The apparatus eliminates a portion of the second speech uttered outside the objective area from the second speech obtained by a second microphone based at least in part on the feature and the model. The apparatus detects and outputs remaining speech from the second speech.

Abstract: Disclosed herein is a method of constructing and utilizing a sound engineering evaluation and comparison process to allow for improved finished results. Such a method entails the utilization of a high-pass filter for listening evaluation of recorded music or sounds including consistency with low-frequency mixing to allow for a tool to implement changes in relation to the filtered results in order to accommodate sensitivities of the human ear (with the optional inclusion of a comparison method to provide possible further enhanced results and the avoidance of biases). In such a manner, a facilitating method for sound engineering mixing adjustments that provide such accommodations are provided for improved sound recordings for distribution within on-line or recording product frameworks.

Abstract: A method for processing signals, a terminal device, and a non-transitory readable storage medium are provided. The method includes the following operations. A sound signal of external environment is collected via an electroacoustic transducer of a headphone when a user talks through the headphone. Feature audio of the sound signal is identified and whether the feature audio matches a preset sound model is determined. The user is reminded in a preset reminding manner corresponding to the feature audio based on a determination that the feature audio matches the preset sound model.

Abstract: The invention relates to a method for monitoring at least one loudspeaker line (120), wherein a main module (210) arranged at a first point in the at least one loudspeaker line (120) and at least one monitoring module (220) arranged at a second, remote point in each of the at least one loudspeaker lines (120) communicate, wherein a message is transmitted from the main module (210) to the at least one monitoring module (220), and/or the at least one monitoring module (220) is supplied with power, by using the loudspeaker line (120), wherein a message is transmitted from the at least one monitoring module (220) to the main module (210) by using a wireless communication connection, and a device for monitoring at least one loudspeaker line (120).

Abstract: The present disclosure discloses a sound control device for controlling load based on continuous sound control signal, which comprises a sound control component, a sound control processing component, a CPU processing device, an execution port component and a load, wherein the sound control component is connected to the sound control processing component; the sound control component collects a sound control signal and transmits the sound control signal to the sound control processing component; the sound control processing component is connected to the CPU processing device; and the CPU processing device is respectively connected with the execution port component and the load.

Abstract: A method, apparatus, and system provides the ability to navigate within a building structure. A building information model (BIM) for a building structure is obtained, validated, and stored in a database in the cloud. A request for the validated BIM is received from a mobile device. The validated BIM is provided from the database to the mobile device. A destination within the building structure, is accepted from the mobile device via a navigation application on the mobile device. A route from a location of the mobile device to the destination is determined and provided to the mobile device. The navigation application utilizes the route and the validated BIM to direct the mobile application to the destination.

Abstract: Systems and methods for beamforming audio signals received from a microphone array. One example embodiment provides an electronic device. The electronic device includes a microphone array and an electronic processor communicatively coupled to the microphone array. The electronic processor is configured to estimate an ambient noise level. The electronic processor is configured to compare the ambient noise level to a first threshold and a second threshold, the second threshold being lower than the first threshold. The electronic processor is configured to determine a beam pattern for the microphone array based on the comparison of the ambient noise level to the first threshold and the second threshold. The electronic processor is configured to apply the beam pattern to an audio signal received by the microphone array.

Abstract: Techniques of content unification are disclosed. In some example embodiments, a computer-implemented method comprises: determining clusters based a comparison of a plurality of audio content using a first matching criteria, each cluster of the plurality of clusters comprising at least two audio content from the plurality of audio content; for each cluster of the plurality of clusters, determining a representative audio content for the cluster from the at least two audio content of the cluster; loading the corresponding representative audio content of each cluster into an index; matching the query audio content to one of the representative audio contents using a first matching criteria; determining the corresponding cluster of the matched representative audio content; and identifying a match between the query audio content and at least one of the audio content of the cluster of the matched representative audio content based on a comparison using a second matching criteria.

Abstract: There is disclosed active acoustic filter systems and methods. A processor is disposed within a housing configured to interface with a user's ear. A memory stores data defining one or more locations and a respective set of location-based processing parameters associated with each of the one or more locations. A personal computing device external to the housing is coupled to the processor via a first wireless communications link. The personal computing device determines a current location of the active acoustic filter system. The processor generates digitized processed sound by processing digitized ambient sound in accordance with a set of location-based processing parameters retrieved from the memory, the retrieved set of location-based processing parameters associated with the current location of the active acoustic filter system as determined by the personal computing device.

Abstract: A system that handles direction of arrival (DOA) estimation for acoustic signals using sub-array selection, identifies a plurality of microphone sub-arrays from the plurality of microphones in the microphone-array, selects a set of microphone sub-arrays from the plurality of microphone sub-arrays, and computes a relative time-delay for arrival of the acoustic signals between each pair of microphones of the selected set of microphone sub-arrays. The selection is based on a maximum distance between each pair of microphones of the identified plurality of microphone sub-arrays of the microphone-array. A first microphone sub-array is determined from the selected set of microphone sub-arrays and the DOA of the acoustic signals is estimated with reference to the determined first microphone sub-array. The estimation of the direction of arrival of the acoustic signals is based on the computed relative time-delay for the determined first microphone sub-array of the microphone-array.

Abstract: An audio processing device includes: a sound source localizing unit configured to determine a localized sound source direction, which is a direction of a sound source, on the basis of audio signals of a plurality of channels acquired from M (here, M is an integer equal to or greater than “3”) sound receiving units of which positions are different from each other; and a sound source position estimating unit configured to, for each set of two sound receiving units, estimate a midpoint of a segment perpendicular to both of half lines directed in estimated sound source directions, which are directions from the sound receiving units to an estimated sound source position of the sound source, as the estimated sound source position.

Abstract: The application aids users by providing supplemental audio content. For example, the application determines a subject of content provided to user equipment during a first time period of the content. The application retrieves a profile associated with the user equipment and retrieves supplemental audio related to the profile and related to the subject of the content. The application detects a supplementation point in the content corresponding with audio content similar to a supplementation signature and transmits the supplemental audio to the user equipment for output at the supplementation point.

Abstract: Implementations generally relate to automated equalizer adjustments based on audio characteristics. In some implementations, a method includes detecting music that is being currently played on an audio device. The method further includes adjusting one or more equalizer settings of an equalizer device based at least in part on a music genre associated with the music. The method further includes outputting the music based at least in part on the adjusting of the one or more equalizer settings of the equalizer device.

Abstract: Automatically recommending sound effects based on visual scenes enables sound engineers during video production of computer simulations, such as movies and video games. This recommendation engine may be accomplished by classifying SFX and using a machine learning engine to output a first of the classified SFX for a first computer simulation based on learned correlations between video attributes of the first computer simulation and the classified SFX.

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: Disclosed is a method for determining respective transmission delays between a node and a plurality of radiators of an infrared audio transmission system comprising a signal generator and said plurality of radiators connected to said signal generator by a network, the method comprising, at a node of said network, transmitting at least one test signal to said plurality of radiators over said network, detecting an event triggered by said at least one test signal, and determining respective transmission delays between said node and said radiators on the basis of said event. Also disclosed are non-transitory computer program product comprising code means configured to cause a processor to carry out the method, a configuration node for carrying out the method, and a system comprising the configuration node and the plurality of radiators.