Abstract: Systems and methods to provide acoustic monitoring are disclosed, comprising receiving a first acoustic signal and providing a first analog signal output representative of the received first acoustic signal using a first sensor, determining at least one of a continuous noise level of the first analog signal output of the first sensor or a peak level of impulse noise of the first analog signal output of the first sensor using analog signal processing circuitry, and sampling, using digital signal processing circuitry, at least one of the determined continuous noise level of the first analog signal output or the peak level of impulse noise of the first analog signal output from the analog signal processing circuitry to provide data privacy of the first analog signal output.

Abstract: A receiver control method and device, and an electronic device are provided. The electronic device includes at least one receiver, wherein the at least one receiver is electrically connected to a baseband circuit, and a feedback circuit is arranged between at least one receiver and the baseband circuit. The receiver control method includes: detecting the at least one receiver; and in a case of detecting that a first receiver of the at least one receiver is blocked, adjusting an electrical signal gain of the feedback circuit corresponding to the first receiver to reduce an electrical signal input to the first receiver.

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 earbud with temperature sensing is provided, and the earbud includes a housing, a speaker and a first temperature sensor. The housing defines a first opening and a second opening. The speaker is disposed within the housing and oriented so that audio emitted by the speaker exits the housing through the first opening defined by the housing. The first temperature sensor is positioned along an interior surface of the housing and aligned with the second opening in the housing for non-contact temperature sensing. In response to the earbud being positioned for use within an ear of a user, the second opening aligns with an inner side of the tragus close to the ear canal, the cavum conchae, an inner edge of the antitragus, or the cheek close to the ear, such that the first temperature sensor detects a body temperature of the user through the second opening.

Abstract: The present disclosure relates to an ear protection system (200) for a medical imaging device. It comprises an ear protection device (210), adapted to be fitted around or in the ears of a patient (P) to be imaged, and at least comprising a first communication interface (211) and at least one sensor device (212) adapted to determine a measurement of noise passing through the ear protection device (210) towards the ears of the patient. The system (200) further comprises a controllable signal emitter (230), adapted to output a proxy signal representing an expected imaging device noise and to be measured by the at least one sensor device (212), and a patient assistance device (220), adapted to assist the patient to fit the ear protection device (210), and at least comprising a second communication interface.

Abstract: A method for eliminating sound leakage includes: controlling, upon detecting call voice being outputted by a receiver, a collection device to determine a first frequency response curve of first sound wave of the call voice at a first position outside a terminal device; controlling a speaker to generate a second sound wave; controlling the collection device to determine a second frequency response curve of the second sound wave at the first position; and regulating, according to the first frequency response curve, the second frequency response curve to a third frequency response curve. Frequency responses of the first and third frequency response curves at a corresponding frequency are superimposed on and cancel each other. The speaker actively generates second sound wave that is modulated according to a parameter of first sound wave, so as to ensure that second sound wave effectively eliminates leakage of call content caused by first sound wave.

Abstract: This document relates to training and employing of quality estimation models to estimate the quality of different signal characteristics. One example includes a method or technique that can be performed on a computing device. The method or technique can include obtaining training signals exhibiting diverse impairments introduced when the training signals are captured or diverse artifacts introduced by different processing characteristics of a plurality of data enhancement models. The method or technique can also include obtaining quality labels for different signal characteristics of the training signals. The method or technique can also include training at least two different quality estimation models to estimate quality of at least two different signal characteristics based at least on the training signals and the quality labels.

Abstract: A processing device according to this embodiment includes: a measurement signal output unit configured to output a frequency sweep signal whose frequency is swept as a measurement signal; a sound pickup signal acquisition unit configured to acquire L-ch and R-ch sound pickup signals obtained by picking up the measurement signal by a left microphone and a right microphone; an evaluation signal acquisition unit configured to calculate an evaluation signal in a time domain in accordance with the L-ch and R-ch sound pickup signals; an extraction unit configured to extract a partial section of the evaluation signal as an extraction section; a comparison unit configured to compare the L-ch sound pickup signal with the R-ch sound pickup signal using the evaluation signal in the extraction section; and a determination unit configured to determine whether the fit of the right and left microphones or the output unit is good or not based on the results of the comparison made in the comparison unit.

Abstract: An audio signal processing device includes a first input terminal which receives an audio signal from a record player, a memory which stores a correction parameter for audio signal processing and characteristic information indicating the characteristic of an audio signal for measurement, a signal processing circuit which performs the audio signal processing on the audio signal received based on the correction parameter, and a microcomputer. The microcomputer measures the sound characteristic of the audio signal received when the audio signal for measurement recorded on a phonograph record for measurement are reproduced, generates the correction parameter based on a result of comparison between the result of the measurement and the characteristic information, and stores, in the memory, setting information and the correction parameter generated, in association with each other, the setting information indicating a setting on the audio signal processing when the measurement is performed.

Abstract: A selection method is implemented in electronic equipment and includes steps consisting of: acquiring and memorizing, for each microphone of the equipment, a first reference audio signal produced by said microphone from a reception by the set of microphones of a first reference sound signal; analyzing the first reference audio signals to produce at least one directional parameter representative of a direction of arrival of the first reference sound signal; selecting, according to the directional parameters from among the set of microphones, the first microphones which make it possible to maximize an effectiveness of a first voice recognition method.

Abstract: An audio device including: an audio signal input section that receives an audio signal; a signal processor that performs signal processing on the audio signal; an amplifier that amplifies the audio signal; an audio signal output section that outputs the amplified audio signal; a feeder that feeds electric power to an external device; a feed detector that detects a state of feeding of the electric power by the feeder; a controller that causes the signal processor to perform level limiting processing to limit a level of the audio signal in a case where the feed detector detects that the feeder is feeding the electric power to the external device.

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 acoustic device is described and includes an acoustic sensor element configured to sense acoustic energy and produce an output signal and a threshold detector circuit including a switch having an input coupled to the output of the acoustic sensor element to receive the output signal, a control port that receives a control signal, and first and second output ports, a first channel including an analog-to-digital converter that operates at a first power level a second analog-to-digital converter that operates at a second higher power level, relative to the first power level and a threshold level detector that receives an output from the first analog-to-digital converter to produce the control signal having a first state that causes the switch feed the output signal from the acoustic sensor element to the second analog-to-digital converter when the first digitized output signal meets a threshold criteria.

Abstract: A computing system includes an interface configured to interface and communicate with an external media resource, a memory that stores operational instructions and a processing module operably coupled to the interface and to the memory, such that the processing module, when operable within the computing system based on the operational instructions, is configured to is configured to perform various operations. The computing system receives an audio fingerprint from the external medial resource, divides the audio file segment into a plurality of portions and compares the audio file segment with a plurality of stored audio segments. The computing system then determines whether a threshold number of the plurality of portions of a stored audio segment compare favorably with a threshold number of the plurality of portions of the audio file segment and based thereon determines whether the audio file segment and the stored audio segment correspond to alternate versions of a same song.

Abstract: An audio processing device includes: a sound level measuring circuit arranged to operably generate multiple sound level values, wherein the multiple sound level values respectively correspond to the sound levels generated by an audio playback device at multiple time points or the sound levels received by a microphone at multiple time points; an audio dose calculating circuit coupled with the sound level measuring circuit and arranged to operably generate an audio dose value corresponding to a measuring period based on the multiple sound level values and contents of a weighting table; a control circuit coupled with the audio dose calculating circuit and arranged to operably compare the audio dose value with a dose threshold to determine whether to generate a control signal or not; and an indication signal generating circuit coupled with the control circuit and arranged to operably generate a corresponding indication signal according to the control signal.

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: 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: 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: 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: 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 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: 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: 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: 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: 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.