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: 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: 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: 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: 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 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: 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: Recorded broadcast programs, including advertisements, are selected for playback responsive to user input. The user input may include pressing a button multiple times to cause recorded programs to be scanned through for a particular preset channel from newest recorded to oldest recorded. Pressing another button multiple times causes the recorded programs to be scanned through for the particular present channel from oldest recorded to newest recorded. Each time at least a portion of a recorded program is played back, the amount of time of the playback is kept track of. An advertisement credit is awarded for playback of the recorded program based on the amount of time of playback of the recorded program.

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.

Abstract: A deep learning based computer vision system to detect and analyses automobile crash videos comprises a crash detection module and a crash analysis module. The crash detection module uses dashcam video and or telemetry to detect the same. The usage of computer vision algorithm alongside the IMU sensor data make the system robust. The system uses a deep learning model which looks for an anomalous pattern in the video and/or the IMU signal to detect a crash. The system comprises different machine learning and deep learning based computer vision algorithm to analyze the crash detected videos. The system automates the labor-intensive task of reviewing crash videos and generates a crash report comprising accident type, road surface, weather condition and accident scenario among other things. The accident detection and analysis modules can reside on the vehicle as well as on the cloud based on the compute capabilities of the edge device.

Abstract: A distance-based score approximation having improved computational efficiency is provided. Responsive to receiving a score request, an observation point is identified based the score request. A geographic database is queried for map information corresponding to geometry elements located within a predetermined maximum distance of the observation point. A plurality of sections of the geometry elements are defined. Each section is associated with one of a plurality of groups of sections based on map information corresponding to a geometry element corresponding to the section. For each of the plurality of sections, an impact curtain area is determined. For each group of sections, the impact curtain area for each section associated with the group is aggregated to define a contribution for the group. The contribution for each group of sections is aggregated to determine the distance-based score. The distance-based score is provided for display via a display of a user computing entity.

Abstract: A virus detection method, a terminal, and a server are provided. The method includes performing preprocessing on an obtained to-be-processed file according to a preset policy, to obtain a part that is in the to-be-processed file and whose stability is greater than a first threshold as effective information. The effective information is calculated to obtain a first characteristic parameter value. The first characteristic parameter value is transmitted to a server for performing detection by means of virus comparison, and a detection result of the virus comparison is received. Virus scanning is performed on a local file according to the detection result.

Abstract: A method, apparatus and computer program product are provided to facilitate calibration of an audio playback system with a video presented upon a display. In the context of a method, information about a display from a first location is received. The method also includes determining an angle of the display based upon the received information about the display and the first location. The method further includes causing an audio playback system associated with the display to be calibrated based upon a modified azimuth of a source of audio signals based upon the determined angle. A corresponding apparatus and computer program product are also provided.

Abstract: A sound source localization position indicating a position where a sound source is localized is set, and a sound signal for localizing the sound source at the set sound source localization position is generated and output. Then, a vibration signal for, in conjunction with the sound source localization position, vibrating a vibrator for imparting a vibration to a user is generated and output.

Abstract: A flying object detection system includes a sensor including a microphone array configured to collect a sound in a first detection area of a monitoring area, a radar configured to measure a distance to a flying object by detecting the flying object in flight in a second detection area of the monitoring area, and a control device configured to detect presence or absence of the flying object based on sound data on the sound in the first detection area collected by the microphone array, and configured to receive the distance to the flying object. The control device is configured to display information on a position of the flying object viewed from arrangement places of the sensor and the radar based on the distance to the flying object and a detection output of the flying object by the radar on a first monitor.

Abstract: Audio information of audio content being listened to by a user is received. An aspect of a listening environment of the user is identified. A volume preset, based on the audio information and the aspect of the listening environment, is determined to be available. A first volume of the audio content being listened to by the user is determined to be different from the volume preset. The first volume of the audio content is adjusted to a second volume.

Abstract: A device includes a signal input to receive a data input as part of a bit stream. The device also includes a reference input to receive a reference signal. The device further includes push circuitry to receive a first weight value, receive a first correction value, and generate a push signal based on the first weight value and the first correction value to selectively modify the data input as well as pull circuitry to receive a second weight value, receive a second correction value, and generate a pull signal based on the second weight value and the second correction value to selectively modify the data input.

Abstract: A method of suppressing an acoustic reverberation in an audio signal includes providing an audio signal, carrying out a first level measurement of the audio signal and carrying out a second level measurement of the audio signal at least temporarily during the first level measurement. The first level measurement is carried out with a decay time and/or adjustment time that differs from a decay time or adjustment time of the second level measurement. An acoustic reverberation of the sound event in the audio signal is suppressed by attenuating the audio signal in response to a sound event in the audio signal. The attenuation is controlled as a function of a difference between the first level measurement and the second level measurement. A hearing device is also provided.

Abstract: This sound system includes a plurality of sound devices connected via a network, each sound device including: an audio acquirer; a sound source localizer that performs sound source localization; a sound source information exchanger that acquires sound source information relating to the audio acquired by another device; a sound source determiner that determines, when a sound source in the sound source information coincides with the sound source specified by the sound source localizer, whether or not a second sound pressure level of the sound source in the sound source information is higher than a first sound pressure level of the sound source specified by the sound source localizer; and a filter processor that performs filter processing of the audio of the sound source specified by the sound source localizer when the second sound pressure level is higher than the first sound pressure level.

Abstract: Embodiments relate generally to devices, systems, and methods for monitoring sounds. A method may comprise monitoring environmental sounds with a hearing protection device; transmitting acoustic signatures from the hearing protection device to a portable electronic device; transmitting the acoustic signatures from the portable electronic device to a server; correlating the acoustic signatures with information in a database; transmitting a message from the server to the portable electronic device indicating the correlation; creating a suggestion with the portable electronic device based on the correlation; and transmitting the suggestion from the portable electronic device to the hearing protection device based on the correlation.

Abstract: A computer implemented method includes receiving audio signals representative of speech via multiple audio channels transmitted from corresponding multiple distributed devices, designating one of the audio channels as a reference channel, and for each of the remaining audio channels, determine a difference in time from the reference channel, and correcting each remaining audio channel by compensating for the corresponding difference in time from the reference channel.

Abstract: Embodiments described herein provide methods and apparatus for predicting the excursion of a transducer. The method comprises determining whether an input signal to the transducer is a periodic signal; and responsive to determining that the input signal is a periodic signal, calculating the predicted excursion based on a direct current (“DC”) offset associated with the transducer.

Abstract: A cable apparatus, and systems and methods, are provided for mixing ambient sound signals generated by the cable apparatus with a monitor mix signal received by the cable apparatus for transmission to first and second in-ear monitoring devices. The cable apparatus may be connected between an audio signal transceiver and the first and second in-ear monitoring devices of an audio monitoring system. The cable apparatus comprises an electrical communications channel that includes first and second connectors for coupling with the first and second in-ear monitoring devices and a third connector for coupling with the audio signal transceiver. The cable apparatus is configured to generate first and second ambient sound signals associated with first and second microphones coupled along the electrical communications channel of the cable apparatus proximate to the first and second connectors. The cable apparatus actively mixes the first and second ambient sound signals with the monitor mix signal.