Abstract: Various embodiments of the present invention relate to an earjack connection apparatus for an earphone antenna, and an electronic device comprising the same. To this end, the electronic device according to various embodiments of the present invention comprises: an earjack including a plurality of earjack electrodes; and a circuit board including a plurality of substrate electrodes and a plurality of substrate lines electrically connected to the plurality of substrate electrodes, wherein first and second substrate lines among the plurality of substrate lines can be configured to be electrically connected to a first substrate electrode among the plurality of substrate electrodes. Other embodiments are also possible.

Abstract: A computer-based method for reproducing a sound quality in a performance space is provided. The method includes accessing a multi-dimensional sound signature, stored in a computer, for an audience location in a first performance space. The method further includes receiving sound data from a sound input device in a second performance space. The method further includes modifying the sound data to match a sound characteristic of the multi-dimensional sound signature. The method further includes transmitting the modified sound data through a sound output device in the second performance space.

Abstract: Estimating the field strength from an ultrasonic phased array can be done by summing the contribution of each transducer to the point of interest. Since this contribution is already calculated when creating a converging spherical wave, it can be reused to add a virtual microphone to the system. By monitoring this microphone and moving it along with new focus points, a robust system of field estimates and regulation may be established.

Abstract: A system configured to perform microphone occlusion event detection. When a device detects a microphone occlusion event, the device will modify audio processing performed prior to speech processing, such as by disabling spatial processing and only processing audio data from a single microphone. The device detects the microphone occlusion event by determining inter-level difference (ILD) values between two microphone signals and using the ILD values as input features to a classifier. For example, when a far-end reference signal is inactive, the classifier may process a first ILD value within a high frequency band. However, when the far-end reference signal is active, the classifier may process the first ILD value and a second ILD value within a low frequency band.

Abstract: A wireless receiver includes a guided user interface to enable and assist users to optimally select groups and frequencies for wireless communication with wireless transmitters is provided. The wireless receiver may automatically detect the connectedness and parameters of wireless receivers in order to adaptively display and select setup options and messages. The guided user interface may improve user satisfaction and reduce the configuration time of the wireless receivers.

Abstract: An identification device, method and system for identifying a scene in an environment. The environment includes at least one sound capture device. The identification device is configured to identify the scene based on at least two sounds captured in the environment. Each of the at least two sounds are associated respectively with at least one sound class. The scene is identified by taking account of a chronological order in which the at least two sounds were captured.

Abstract: An electronic device is provided.

Abstract: A method performed by an in-ear headphone. Coupled to the in-ear headphone is a first ear tip that is inserted into an ear canal of a user. The method obtains an audio signal from an audio source device paired with the in-ear headphone and uses the signal to drive a speaker of the headphone to output a sound into the ear canal. The method obtains a microphone signal that is responsive to the outputted sound. The method notifies the user to replace the first ear tip with a second ear tip in response to a parameter associated with the microphone signal being less than a threshold.

Abstract: An integrated acoustic coupler for use in sound engineering testing of an IEM (in-ear monitor). The integrated acoustic coupler comprises an integrated coupler body having an input, a first chamber and an output port. The input defines an IEM seat defined by a foam member and into an which an IEM may be inserted to define an air-tight seal between the foam and the IEM. The first chamber is in fluid communication with the IEM seat and interconnected with at least one second chamber via a passageway configured for creating compliance and impedance to simulate a human ear. The output port is configured for electronically interconnecting to an XLR cable and outputting a signal from an IEM under test to a mixer or audio analyzer.

Abstract: A communication system, A power and data hub and a method for operating a communication device is disclosed, the power and data hub comprising: a hub housing; a first primary interface configured to communicate with a radio unit to communicate primary audio, primary power input, and primary data; a second primary interface configured to communicate with a communication device to communicate and/or provide secondary audio and primary power output; a first secondary interface configured to communicate and/or provide secondary data and/or secondary power output with a first peripheral device connectable to the first secondary interface; a second secondary interface configured to communicate and/or provide secondary data and/or secondary power output with a second peripheral device connectable to the second secondary interface.

Abstract: Described herein are systems and methods for reliably detecting of the presence or absence of accessories and other components connected to an external electrical device or connector. A low-cost detection circuit reliably distinguishes between a compromised connector and the presence of an actual device, thereby, preventing false detection scenarios. In various embodiments, the detection circuit accomplishes this by using a zero-crossing detector and/or a digital signal, which serves as a measure of impedance, to detect, within an identification period, the presence of the external device.

Abstract: A method and multimedia device for calibrating audio delays of an audio system with wireless loudspeakers are disclosed. The method allows to emulate surround or 3D sound systems with spatially distributed wireless loudspeakers. A first probe signal for audio playback, having a first audio pattern, is transmitted by a loudspeaker. A signal quality is determined by listening to an acoustic response of the loudspeaker to the first probe signal. A second audio pattern is determined by modifying the first audio pattern based on the signal quality. A second probe signal for audio playback having the second audio pattern is transmitted by the loudspeaker. An audio delay compensation is then determined for the loudspeaker based on a measure of an audio delay between the transmitted second probe signal and an acoustic response of the loudspeaker in response to the second probe signal.

Abstract: Implementations of the disclosed subject matter provide a method of transmitting, from a mobile robot device, sound and/or at least one image captured by a sensor to a remote user device. The mobile robot device may receive at least one first control operation for the mobile robot device to move within an area via a communications network from a remote user device. An audio signal may be transmitted based on sound received at a microphone of the mobile robot device in the area. The audio signal received from the remote user device may be output at a speaker of the mobile robot device. A volume of the audio signal output by the speaker may be adjusted based on a size of the area and on an average or a median of an amplitude of frequencies in the area based on the sound received by the microphone.

Abstract: The invention discloses a test circuit, a test method and an audio codec for testing a microphone module that includes a first microphone and a second microphone. The first microphone outputs the first data, and the second microphone outputs the second data. The test circuit includes a comparison circuit, a counter, and a decision circuit. The comparison circuit is configured to compare the first data with the second data and generate a comparison result. The counter is coupled to the comparison circuit and configured to generate a count value based on the comparison result. The decision circuit is coupled to the counter and configured to indicate, based on the count value and a threshold value, whether the microphone module has an error.

Abstract: Systems, devices, and methods for encoding the harmonic structure of a musical composition in a digital data structure are described. Tonal and rhythmic commonalities are identified across the musical bars that make up a musical composition. Individual bars of the musical composition are each analyzed to characterize their respective harmonic fingerprints in various forms, and the respective harmonic fingerprints are compared to sort the musical bars into harmonic equivalence categories. Isomorphic mappings between hierarchical data structures that encode the musical composition based on musicality and harmony, respectively, are also described. The systems, devices, and methods for encoding the harmonic structure of a musical composition in a digital data structure have broad applicability in computer-based composition and variation of music.

Abstract: Disclosed herein are computer-implemented method, system, and computer-readable storage-medium embodiments for implementing sound generation with adaptive directivity. An embodiment includes at least one computer processor that may retrieve and process an audio sample of a content instance, via an algorithm configured to generate a classification of the audio sample. Directivity may be determined, corresponding to a first audio signal to be output via an audio output device. The first audio signal may correspond to the audio sample of the content instance, and the audio output device may include at least one loudspeaker, for example. A second audio signal may be generated from the classification and the directivity. In some embodiments, another classification of the audio sample may be generated based at least in part on a second algorithm that may process information including a video image corresponding to the audio sample and/or additional audio characteristic(s) detected in the audio sample.

Abstract: Provided are an audio calibration method and device. The method includes: transmitting an audio calibration signal to a first loudspeaker and a second loudspeaker such that the first loudspeaker and the second loudspeaker generate sound signals; determining first loudspeaker sound information through a sound signal generated by the first loudspeaker and acquired by a first microphone and a sound signal generated by the first loudspeaker and acquired by the second microphone; determining second loudspeaker sound information through a sound signal generated by the second loudspeaker and acquired by the first microphone and a sound signal generated by the second loudspeaker and acquired by the second microphone; and determining a calibration parameter of the first loudspeaker or a calibration parameter of the second loudspeaker according to the first loudspeaker sound information and the second loudspeaker sound information.

Abstract: Systems, methods, and non-transitory computer-readable storage media for performing a role swapping operation between a pair of non-tethered wireless ear buds after detecting a triggering event. Further, state information can be coordinated between devices, including in connection with performing a role swap between buds in a pair of wireless, untethered ear buds, where one wireless ear bud is in a primary role and is responsible for a connection with a companion device, and another wireless ear bud in the pair is in a secondary role.

Abstract: An inspection system includes two or more microphone units and an inspection controller. The two or more microphone units are disposed side by side in a vehicle. The inspection controller is configured to cause an inspection-target speaker to output an inspection sound, and determine, on the basis of a sound picked up by a main microphone unit and a sound picked up by one or more satellite microphone units, whether the inspection-target speaker passes or fails. The inspection-target speaker is any of one or more speakers mounted on the vehicle. The main microphone unit is one, of the microphone units, that is disposed at a position corresponding to a position of the inspection-target speaker. The one or more satellite microphone units are each one of the microphone units excluding the main microphone unit.

Abstract: A remote speaker microphone (RSM) unit enables a user to use either a headset or an in-ear speaker with the RSM unit. The RSM unit includes a first RSM connection for communicating and connectable with the first connection of the headset; and a second RSM connection for communicating and connectable with the second connection of the in-ear speaker. Connecting either the headset or the in-ear speaker to the RSM unit disables the speaker and microphone of the RSM unit. At least one of the first and second connections are wireless, e.g., Bluetooth or NFC, and at least one of the first and second RSM connections are wireless connections communicable with the first and second connections, respectively. In addition or the alternative, at least one of said first connection and said second connection are electrical plugs and at least one of said first and second RSM connections are corresponding electrical jacks.

Abstract: Systems and methods are provided for presenting subtitles. The systems and methods include accessing, by a user device, a video discovery graphical user interface that includes a plurality of videos; receiving a user input that gradually reduces volume of the user device; determining that the volume of the user device has gradually been reduced by the user input until a mute state has been reached in which audio output of the user device is disabled; and in response to determining that the volume of the user device has gradually been reduced until the mute state has been reached, automatically causing subtitles of a first video of the plurality of videos to be displayed during playback of the first video.

Abstract: Disclosed in the embodiments of the present application are a microphone hole blockage detection method and a related product. The method may include: a microphone of a first wireless earphone, acquiring audio of a preset frequency band, the preset frequency band being an audible sound frequency band; determining a first volume parameter of the audio acquired by the first wireless earphone; and, on the basis of the first volume parameter, determining that the microphone of the first wireless earphone is in a blocked hole state.

Abstract: An electronic device is in communication with one or more wearable audio output devices. The electronic device detects occurrence of an event and outputs, via the one or more wearable audio output devices, one or more audio notifications corresponding to the event. After beginning to output the one or more audio notifications, the electronic device detects an input directed to the one or more wearable audio output devices. In response, if the input is detected within a predefined time period with respect to the one or more audio notifications corresponding to the event, the electronic device performs a first operation associated with the one or more audio notifications corresponding to the event; and, if the input is detected after the predefined time period has elapsed, the electronic device performs a second operation not associated with the one or more audio notifications corresponding to the first event.

Abstract: A test system comprises a device under test and a testing device. The device under test comprises a first initiation unit, wherein the first initiation unit is configured to generate a first wireless initiation signal. The initiation signal consists of at least one of electromagnetic waves and sound waves, wherein the initiation signal comprises a first test command. The testing device comprises a first sensor unit, wherein the first sensor unit is configured to receive the initiation signal via the first sensor unit. The testing device is configured to generate a first electromagnetic test signal based on the first test command.

Abstract: The present disclosure discloses a method and an apparatus for testing a speaker, an electronic device and a storage medium. A specific implementation includes: obtaining first audio data recorded by a microphone integrated with the speaker in ambient white noise; analyzing the first audio data to derive a first analysis result; and determining whether there is a defect in the microphone according to the first analysis result. Hence, these allow for testing a completed set on an assembled speaker to ensure the consistency of a microphone test and improve the accuracy of the test result.

Abstract: A stereo signal processing method and apparatus, where the method includes performing delay estimation on a stereo signal of a current frame to determine an inter-channel time difference of the current frame, identifying a sign of the inter-channel time difference of the current frame is different from a sign of an inter-channel time difference of a previous frame of the current frame, performing delay alignment processing on the first-channel signal of the current frame based on the inter-channel time difference of the current frame, and performing delay alignment processing on the second-channel signal of the current frame based on the inter-channel time difference of the previous frame.

Abstract: Media content typically includes closed captioning information such as subtitled in domestic and foreign languages. Techniques and mechanisms provide that closed captioning information may be toggled on/off using menu options and preferences as well as automatically managed by intelligently monitoring the environment surrounding a device. Device sensors such as microphones and vibration monitors determine the noise level of an environment as well as the spectral characteristics of the noise to determine whether the noise profile would interfere with the video playback experience. A particular environmental noise profile could automatically trigger the display of closed captioning information or present an easy access, otherwise unavailable toggle to display closed captioning information associated with a video stream.

Abstract: An acoustic testing apparatus may include connections for multiple devices under test (DUTs) to support simultaneous testing of two or more miniature electroacoustic devices. The acoustic testing apparatus may allow the testing of multiple DUTs with a ratio of less than one reference microphone per DUT. Thus, the speed of testing DUTs may be increased without adding significant cost through additional reference microphones. For example, a single reference microphone may be used to test two or four DUTs coupled together through an acoustic test cavity.

Abstract: This disclosure describes techniques that include aligning processing of audio samples collected by multiple audio sensors or microphones. In one example, this disclosure describes a method comprising enabling a first microphone; processing, by an audio processor and using a first processing pipeline, audio data samples collected by the first microphone; enabling a second microphone a period of time after enabling the first microphone; processing, by the audio processor and using a second processing pipeline, a sample of audio data collected by the second microphone by synchronizing starting times for the first and second processing pipelines.

Abstract: Example techniques may involve multiple calibrations for a playback device. In an example implementation, a playback device plays back audio content according to a first calibration via the one or more audio amplifiers. The playback device detects that an arrangement of one or more listeners relative to the first playback device has changed from a first arrangement to a second arrangement. Based on detection that the arrangement of one or more listeners relative to the first playback device has changed from the first arrangement to the second arrangement, the playback device applies a second calibration to playback by the first playback device.

Abstract: A method for controlling a display apparatus includes: based on a communication connection with an audio device being established through a first interface, transmitting, to the audio device, audio setup information of the display apparatus; based on receiving a first audio signal comprising a plurality of channels from an external server, transmitting a second audio signal comprising the plurality of channels to the audio device; receiving, from the audio device, a third audio signal comprising a height channel obtained from the second audio signal based on the audio setup information of the display apparatus; and outputting the received third audio signal.

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

Abstract: A media playlist is generated in which media items have a tempo matching a target tempo. The content server receives a seed value and a tempo signal from a client device. The tempo signal indicates a target tempo specified by the user. The content server calculates a target tempo range associated with the target tempo and generates a filtered playlist of media items based on the seed value and the target tempo range. The media items in the in the filtered playlist have tempos within the target tempo range. The content server provides the media items in the filtered playlist to the client device.

Abstract: A wearable audio device that includes an electro-acoustic transducer for creating audio output and a magnetometer system comprising a magnetic field sensor with an output, a temperature sensor that is configured to determine an internal device temperature, a processor, and memory. The magnetometer system is configured to derive from the magnetic field sensor output a directional heading of the Earth's magnetic field. The magnetometer system is further configured to compensate the magnetic field sensor output, wherein the compensation is temperature dependent. The memory is configured to store temperature-dependent compensation information. The magnetometer system is further configured to use the temperature sensor output to retrieve compensation information from the memory in order to compensate the magnetic field sensor output at the current temperature.

Abstract: A test system comprises a device under test and a testing device. The device under test comprises a first microphone, wherein the first microphone is configured to receive sound waves. The testing device comprises a first speaker, wherein the first speaker is configured to generate sound waves. The testing device is configured to generate a first acoustic command signal via the first speaker, wherein the first acoustic command comprises a first test command. The device under test is configured to receive the first acoustic command signal via the first microphone. The device under test is configured to generate a first electromagnetic test signal based on the first test command.

Abstract: A device may receive a request to initiate an audio test and may generate, based on the request, one or more instructions to control a bone transducer device and a microphone device. The device may send the one or more instructions to a control device and may receive, after sending the one or more instructions, audio data from the control device. The device may process the audio data to determine whether the audio data includes an otoacoustic emission (OAE) and may cause at least one action to be performed based on determining whether the OAE is included in the audio data.

Abstract: A device or system is provided which is configured to detect one or more sound events and/or scenes associated with a predetermined context, and to provide an assistive output on fulfilment of that context.

Abstract: A system and method for associating audio feeds to corresponding video feeds, including determining a subject of interest within a video feed based on the video feed and metadata associated with the video feed; analyzing the metadata to determine an optimal audio source for the subject of interest; configuring the optimal audio source to capture an audio feed; and associating the captured audio feed with the video feed.

Abstract: Disclosed are a low-power communication method and a low-power communication apparatus in a communication system. A low-power station may comprise a processor; a memory storing at least one instruction executable by the processor; a receiver for receiving a WUR PPDU according to the at least one instruction; and a transceiver for transmitting and receiving a legacy PPDU according to the at least one instruction, and the at least one instruction may be configured to cause the receiver to receive a WUR wake-up frame from the access point in an on-duration within a WUR duty cycle period; when the WUR wake-up frame is received, cause the processor to transmit a first signal requesting wake-up to the transceiver; and when the first signal is received, cause the transceiver to transition from a sleep state to a wake-up state at a TWT configured between the access point and the low-power station.

Abstract: A signal processing device according to an embodiment includes a measurement signal generation unit that generates a first measurement signal and outputs the signal to headphones, a sound pickup signal acquisition unit that acquires a first sound pickup signal obtained by picking up the first measurement signal by a microphone, a signal analysis unit that calculates a frequency response of the first sound pickup signal, a fit determination unit that determine a fit of the headphones by comparing a response value of the frequency response in a specified frequency band with a reference value, and a display unit that outputs gradual results of the determination in accordance with a percentage of the response value with respect to the reference value.

Abstract: A computing device includes: a memory storing an audio sequence and a reference attribute associated with playback of the audio sequence; a speaker; a microphone; an audio enclosure supporting the speaker and the microphone; a processor configured to: simultaneously (i) control the speaker to play the audio sequence, and (ii) control the microphone to capture a test recording corresponding to playback of the audio sequence; based on a comparison of the reference attribute to a test attribute associated with the test recording, detect occlusion of the audio enclosure; and in response to detecting occlusion of the audio enclosure, generate a notification message.

Abstract: A method of overlaying an optical and acoustic reproduction of a real scene with virtual image and audio data comprises reading image data which represent an optical and/or acoustic reproduction of the real scene captured by at least one environment sensor of a mobile device, determining marker data from the image data, wherein the marker data represent a reproduction and a positioning of a marker arranged in the real scene, reading virtual image and audio data, wherein the virtual image and audio data comprise a representation instruction for representing the virtual image, a positioning instruction for positioning the virtual image and a positioning instruction for replay of the audio data, determining object data from the image and audio data, and ascertaining a positioning rule for positioning the virtual image with reference to the reproduction of the object portion using the object data and the virtual image and audio data.

Abstract: An electronic device is disclosed. The electronic device comprises a communication unit, a speaker, and a processor for: outputting a test sound through the speaker when a preset signal is received from an external terminal device through the communication unit; acquiring, on the basis of sound data, reverberation time information for each frequency of the test sound and size information on a space in which the electronic device is positioned when sound data acquired by recording a test sound in a terminal device is received through the communication unit; acquiring, on the basis of the reverberation time information for each frequency and the size information on a space, a sound absorption rate of an object arranged in the space; and identifying information on the object on the basis of the sound absorption rate.

Abstract: A terminal may include: a sensor unit including a microphone configured to acquire a surrounding sound and a position sensor configured to detect a position of the terminal; a processor configured to identify characteristics of a voice of a specific person designated by a user of the terminal through learning, and determine a setting value determining operating characteristics of a hearing aid based on the characteristics of the voice of the specific person; and a communicator configured to transmit the setting value to the hearing aid.

Abstract: A system for enhancing audio including a plurality of sensors, an output device, and a processor in communication with the plurality of sensors and the output device. The processor is configured to process data captured by the plurality of sensors, and based on that, modify an output of the output device. The processor also is configured to determine whether there are a plurality of users associated with a video conferencing session, determine which user of the plurality of users is speaking, and enhance the audio or video output of the speaking user on the output device.

Abstract: An audio or haptic power output system manages and delivers power to multiple output transducers based on corresponding power or current limits that are adjusted dynamically according to measures of power or current that are required to generate power output signals from corresponding digital input values or signals. A power management subsystem controls electrical power or current consumed by power output stages that supply power to the transducers by comparing the measures of power or current required to generate the power output signals to the corresponding power or current limits, and limiting the power or current consumed by the individual power output stages measures of power or current exceeds the corresponding limits.

Abstract: Systems and methods for microphone degradation detection and compensation are disclosed. For example, microphones of an electronic device may capture audio and generate corresponding audio data, such as during a period of time where only ambient noise is present. Sound intensity level value differences between audio data from the various microphones may be determined and when one or more of the sound intensity level value differences satisfies a threshold amount, the microphone associated with the variant sound intensity level value may be determined to be degraded. The sound intensity level value difference may be compensated for, such as by utilizing sound boosting techniques and/or modifying parameters of a beamforming component.

Abstract: A self-calibration system may include a housing including a first calibration circuit configured to coordinate with a second calibration circuit to execute a calibration sequence for an active noise cancelling (ANC) earphone. The housing further includes a cavity configured to accommodate the ANC earphone, wherein the cavity is contoured to simulate the ANC earphone in a user's ear. The housing further includes a calibration microphone coupled with the first calibration circuit and configured to measure calibration sound waves from the ANC earphone, and a calibration speaker configured to emit calibration sound waves to the ANC earphone.

Abstract: Various implementations include audio devices and related computer-implemented methods for controlling playback of augmented reality (AR) audio. Certain implementations include approaches for managing audio notifications at an audio device according to a set of audio notification rules, for example, by delaying release of notifications and/or de-prioritizing notifications based upon the rules.

Abstract: The disclosed technology generally relates to a hearing device configured adjust the tap detection sensitivity on based on context. The disclosed technology can determine a context for a hearing device based on sound received at the hearing device (e.g., determine loud environment) or a wireless communication signal from an external device received at the hearing device (e.g., receive a message that phone call is incoming); adjust a tapping sensitivity threshold of the hearing device based on the context; detect a tap of the hearing device based on the adjusted sensitivity threshold; and modify a setting of the hearing device (e.g., reduce volume based on a tap) or transmitting instructions to the external device based on detecting the tap. The hearing device can be a hearing aid.