Abstract: Examples described herein involve configuring a playback device based on distortion, such as that caused by a barrier. One implementation may involve causing the playback device to play audio content according to an existing playback configuration, determining an existing frequency response of the playback device in a given system, and determining whether a difference between the existing frequency response of the playback device in the given system and a predetermined frequency response for the playback device is greater than a predetermined distortion threshold. If it is determined that the difference between the existing frequency response of the playback device and the predetermined frequency response for the playback device is greater than the predetermined distortion threshold, then the existing playback configuration of the playback device is changed to an updated playback configuration of the playback device and the playback device plays audio content according to the updated playback configuration.

Abstract: The invention relates to an evaluation device (6) for analysing a public address system (1), the public address system (1) comprising at least one loudspeaker (3a-3d) and at least one audio signal output unit (4), the loudspeaker (3a-3d) having an impedance plot (10), the evaluation device (6) comprising a neural network (11), the neural network (11) being trained to determine a loudspeaker type on the basis of an impedance plot (10), the evaluation device (4) being provided with the impedance plot (10) of the loudspeaker (3a-3d) and an additional parameter, and the evaluation device being designed to analyse the loudspeaker (3a-3d) and/or determine the loudspeaker type by means of the neural network (11) on the basis of the provided impedance plot (10) and the additional parameter.

Abstract: In some examples, a non-transitory computer-readable medium stores machine-readable instructions which, when executed by a processor, cause the processor to cause a playback device to produce a sound using an audio file; cause a recording device to record the sound; compare the audio file to the recorded sound; determine, based on the comparison, whether the recorded sound comprises a noise not present in the audio file; and cause, based on the determination, a second recording device to be selected, a multiplexer to select a second playback device, or a combination thereof.

Abstract: Connected speaker (1) comprising: a LAN interface (2); a WPAN interface (3); pairing means (5) designed to pair the connected speaker with one or more external speakers (6) each equipped with an external WPAN interface (7); processing means (8) designed to acquire an input audio signal from an audio source (15) via the LAN interface, to produce at least one distributed audio signal from the input audio signal, to allocate each distributed audio signal to one of the external speakers (6) paired with the connected speaker, and to transmit, to each external speaker, via the WPAN interface (3) of the connected speaker (1) and the external WPAN interface (7) of said external speaker (6), the distributed audio signal that is allocated to said external speaker (6), such that the connected speaker (1) and the external speakers (6) form a multichannel audio playback system.

Abstract: Techniques are provided for audio capture path evaluation of microphones incorporated into a device under test (DUT). A methodology implementing the techniques according to an embodiment includes estimating impulse responses (IRs) of the DUT microphones based on a comparison of a test audio signal received through the DUT microphones, at a selected measurement angle, to the test audio signal received through a reference microphone. The method also includes calculating group delays for the DUT microphones based on phase responses of the estimated IRs and calculating an average of the group delays. The method further includes calculating a distance, projected onto the measurement angle, between the DUT microphones and a geometric center of the DUT microphones. The distance is calculated as a product of the speed of sound and a difference between the average delay and the group delays for the DUT microphones. The process is repeated for additional measurement angles.

Abstract: Example techniques may involve calibration with multiple recording devices. An implementation may include a mobile device receiving data indicating that a calibration sequence for multiple playback devices has been initiated in a venue. The mobile device displays a prompt to include the first mobile device in the calibration sequence for the multiple playback devices and a particular selectable control that, when selected, includes the first mobile device in the calibration sequence. During the calibration sequence, the mobile device records calibration audio as played back by the multiple playback devices and transmits data representing the recorded calibration audio to a computing device.

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: Embodiments include an audio system comprising a speaker array comprising a plurality of drivers arranged in a first planar configuration; a microphone array comprising a plurality of transducers arranged in a second planar configuration; and a beamformer communicatively coupled to the speaker array and the microphone array, the beamformer configured to: generate an individual speaker output signal for each of the drivers in the speaker array based on one or more input audio signals received from an audio source, and generate a microphone output signal for the microphone array based on one or more microphone signals captured by one or more of the transducers. Embodiments also include a method performed by one or more processors coupled to a microphone array having a plurality of transducers and a speaker array having a plurality of drivers.

Abstract: Systems and methods for determining and adapting to changes in microphone performance of playback devices are disclosed herein. In one example, an audio input is received at an array of individual microphones of a network microphone device. Output microphone signals are generated from each of the individual microphones based on the audio input. The output microphone signals are analyzed to detect a trigger event. After detecting the trigger event, the output microphone signals are compared to detect aberrant behavior of one or more of the microphones. Optionally, corrective actions can be taken or suggested based on the detection of aberrant behavior of one or more microphones.

Abstract: An embodiment sets forth a technique for configuring one or more audio parameters. The technique includes presenting a first plurality of setting options associated with an audio parameter, wherein the first plurality of setting options comprises a first option having a same value as a first setting of the audio parameter and a second option having a first value different than the first setting; receiving a selection of one of the first option and the second option; based on the selection, presenting a second plurality of setting options associated with the audio parameter, wherein the second plurality of setting options comprises a third option having a same value as a second setting of the audio parameter and a fourth option having a second value different than the second setting; and setting the audio parameter based on a selection of at least one of the second plurality of setting options.

Abstract: A stereo signal processing 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: A method, system and product including obtaining a media sample of a media stream; retrieving from a database a media record that matches the media sample of the media stream; aligning the media record with the media stream, wherein aligning the media record comprises determining an offset from a start of the media record; comparing the media stream with the media record according to the offset; and upon identifying a semantic deviation between the media stream and the media record, determining that the media stream is semantically manipulated.

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: A test module for testing microphones comprises an outer chamber being airtight, and a sound chamber comprising an electrical test device for testing the microphones. The sound chamber is located within the outer chamber, and the sound chamber is coupled to the outer chamber with a connection suppressing structure-borne noise between the outer chamber and the sound chamber. A space between the outer chamber and the sound chamber has a gas pressure being lower than an ambient air pressure. A method of testing microphones comprises evacuating the space between the outer chamber and the sound chamber to having a lower gas pressure than an ambient air pressure, and testing the microphone.

Abstract: A data processing method includes: in the case where an operation state of an audio/video device in a target driving device is an out of operation state, generating an out of operation state notification message of the audio/video device; and sending the out of operation state notification message to a push client for stopping a data push between the push client and a push server using the out of operation state notification message.

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: Disclosed are a robot which localizes a sound source based on sound spot information of at least one sound spot, and a sound source localization method. The robot can communicate with other electronic devices and a server in a 5G communication environment.

Abstract: An internal failure detection method of an external failure detection system for industrial equipment, the external failure detection system including an array of transducers, the method including: (a) receiving a plurality of signals, each signal being measured by a corresponding transducer of the transducers array; (b) for each pair of transducers among a number of pairs of transducers, calculating at least one value of a correlation parameter between the pair of signals received at step (a) at the pair of transducers, by correlating at least part of the signals or of invertible transforms thereof; (c) for at least one transducer among the number of pairs of transducers, estimating from the values of the correlation parameters calculated at step (b) if the transducer is working properly.

Abstract: A signal source identification device includes: a feature value calculation unit configured to calculate feature values corresponding to paths where signals from generation sources of the signals transmit, based on signals received by a plurality of sensors; and an identification unit configured to identify whether or not a feature value calculated by the feature value calculation unit is a signal from a predetermined signal source by using an identification range that is a range within which feature values based on signals from the predetermined signal source fall and that is previously determined based on the feature values calculated by the feature value calculation unit.

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: 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: 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 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: 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, 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: 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: 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: 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: 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: 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: 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: 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: 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.