Abstract: Methods, systems, and apparatus for determining a maintenance issue are described. An audio signal is obtained and analyzed to generate an audio signature. A characteristic of a component is identified based on the audio signature and an action is determined based on the characteristic of the component.

Abstract: An audio system includes a first audio device and a second audio device. The first audio device receives an operation of increasing or reducing a parameter value, controls a first parameter using control amount according to operation amount, calculates a ratio of change in the control amount relative to a whole parameter adjustment range, and transmits the ratio of change to the second audio device. The second audio device receives the ratio of change, converts the ratio of change into a control amount of whole parameter adjustment range, controls a second parameter using a value of an integer part of the control amount, integrates the value of the fractional part of the control amount in a storage unit, and controls the second parameter using the value of the integer part when integer part is generated in the integrated value of the storage unit.

Abstract: A method of automatic switching between omnidirectional (OMNI) and directional (DIR) microphone modes in a binaural hearing aid comprising a first microphone system, a second microphone system, where the first microphone system is adapted to be placed in or at a first ear of a user, the second microphone system is adapted to be placed in or at a second ear of said user, the method includes a measurement step, where the spectral and temporal modulations of first and second input signals are monitored, an evaluation step, where the spectral and temporal modulations of the first and second input signal are evaluated by the calculation of an evaluation index of speech intelligibility for each of said signals, and an operational step, where the microphone mode of the first and the second microphone systems of the binaural hearing aid are selected in dependence of the calculated evaluation indexes.

Abstract: An apparatus for encoding a multi-channel signal having at least three channels includes an iteration processor, a channel encoder and an output interface. The iteration processor is configured to calculate inter-channel correlation values between each pair of the at least three channels, for selecting a pair including a highest value or including a value above a threshold, and for processing the selected pair using a multi-channel processing operation to derive first multi-channel parameters for the selected pair and to derive first processed channels. The iteration processor is configured to perform the calculating, the selecting and the processing using at least one of the processed channels to derive second multi-channel parameters and second processed channels. The channel encoder is configured to encode channels resulting from an iteration processing to obtain encoded channels.

Abstract: A device for volume control for at least two audio sources includes a loudness analyzer and a volume regulator. The loudness analyzer is configured to analyze an audio signal of the first one and of the second one of the two audio sources over a time period, to determine a first and a second loudness value, respectively, as a function thereof, and to store same while associating them with the first and the second audio source. The volume regulator is configured to adapt the audio signal of the currently selected first and/or second one of the two audio sources in accordance with the corresponding first and/or second loudness value.

Abstract: A smartphone provides a voice of a person during a telephone call to a user of the smartphone in binaural sound. The smartphone stores HRTFs of the user, and a digital signal processor (DSP) processes the voice of the person with the HRTFs so the voice of the person externally localizes in the binaural sound at a sound localization point (SLP) that is in empty space at least one meter away from a head of the user. The smartphone displays an AR image at the SLP during the telephone call.

Abstract: An audio system includes a first audio apparatus that receives a first audio signal with a first number of channels and a second audio apparatus connected to the first audio apparatus via a network. The first audio apparatus includes a signal processing unit configured to generate, on the basis of the first audio signal received from the outside of the first audio apparatus, a second audio signal with a second number of channels larger than the first number of channels, and a communication unit configured to transmit a third audio signal corresponding to a part of the channels in the second audio signal generated by the signal processing unit to the second audio apparatus.

Abstract: The disclosure provides a sound collection apparatus in which a user can easily find a sound collection direction, and furthermore, can readily control the sound collection direction. A sound collection apparatus comprises: an operation unit capable of visually indicating one direction and three-dimensionally and sequentially varying the direction; a sound collection unit that provides directivity for sound collection; and a control unit. The control unit includes a direction detection unit for detecting a direction indicated by the operation unit. The control unit controls the collection directivity of the sound collection unit according to the direction detected by the direction detection unit. The sound collection unit includes a plurality of microphones. The control unit further includes a beam forming production unit for controlling a beam by a microphone array including the plurality of microphones according to the direction detected by the direction detection unit to vary collection sound directivity.

Abstract: Provided is a condenser microphone circuit that can support variation in power supply voltage of a phantom power supply using a wiring system of two lines. A condenser microphone circuit includes a microphone unit, an FET, a constant current diode, a collector grounding first transistor that generates an operation power supply of the FET, a first resistor that sets base potential of the first transistor, a collector grounding second transistor that amplifies an output signal from the FET, a second resistor that sets base potential of the second transistor, and an output circuit. A base of the first transistor is connected to a source of the FET, an emitter of the first transistor is connected to a drain of the FET, a base of the second transistor is connected to the drain of the FET, an emitter of the second transistor is connected to the output circuit, and the second resistor divides voltage on a cathode side of the constant current diode.

Abstract: An earphone device (1) to be worn in the ear, which earphone device (1) comprising a main body (18) to be arranged in the concha (19) of the outer ear (14) of a user and an optical sensor (35), which optical sensor (35) comprises a light emitter (16) and a light detector (17). The earphone device (1) comprises window means (8, 9) through which light emitted by the light emitter (16) can be transmitted. The window means (8, 9) comprises an end face (31) adapted to abut a conchal wall (37) at a sensing position (36). The main body (18) comprises a speaker protrusion (26) to be inserted into to ear canal (15). The distance between the speaker protrusion (26) and the end face (31) is adjustable.

Abstract: This application describes methods and apparatus for loudspeaker protection. A loudspeaker protection system (100) is described having a first frequency band-splitter (102) for splitting an input audio signal (Vin) into a plurality of audio signals (v1, v2 . . . , vn) in different respective frequency bands (?1, ?2 . . . , ?n). A first gain block (103) is configured to apply a respective frequency band gain (g1, g2 . . . , g3) to each of the audio signals in the different respective frequency bands and a gain controller (107, 108, 109) is provided for controlling the respective band gains. A displacement modeller (104, 105) determines a plurality of displacement signals (x1, x2 . . . , xn) based on the input audio signal (Vin) and a displacement model (104a) where each displacement signal corresponds to a modelled cone displacement for the loudspeaker for one of said different respective frequency bands.

Abstract: This application describes methods and apparatus for loudspeaker protection. A loudspeaker protection system (1100) is described having a first frequency band-splitter (102) for splitting an input audio signal (Vin) into a plurality of audio signals (v1, v2 . . . ,vn) in different respective frequency bands (?1, ?2 . . . ??). A first gain block (103) is configured to apply a respective frequency band gain (gt1, gt2 . . . ,gt3) to each of the audio signals in the different respective frequency bands and a gain controller (109; 1101) is provided for controlling the respective band gains. A thermal controller (1101) determines, for each of a plurality of the different respective frequency bands, a power dissipation for the loudspeaker in that frequency band and also determines a respective thermal gain setting based on the determined power dissipation for that frequency band. The gain controller is configured to control the respective frequency band gains based on the thermal gain settings.

Abstract: A novel microphone incorporates a phantom-powered JFET circuit for audio application. In one embodiment of the invention, a novel phantom-powered JFET preamplifier gain circuit can minimize undesirable sound distortions and reduce the cost of producing a conventional preamplifier gain circuit. Moreover, in one embodiment of the invention, one or more novel rounded-edge magnets may be placed close to a ribbon of a ribbon microphone, wherein the one or more novel rounded-edge magnets reduce or minimize reflected sound wave interferences with the vibration of the ribbon during an operation of the ribbon microphone. Furthermore, in one embodiment of the invention, a novel backwave chamber operatively connected to a backside of the ribbon can minimize acoustic pressure, anomalies in frequency responses, and undesirable phase cancellation and doubling effects.

Abstract: An apparatus for controlling a dynamic compressor of a hearing aid includes a combination signal analyzer for determining the binaural similarity of a left and right audio signal and an amplification adjuster for providing an amplification value for a band of the left or right audio signal in dependence on the binaural similarity and a level of the left or right audio signal in the band.

Abstract: [Object] To allow a listener to listen to ambient sounds of the external environment in an appropriate manner while wearing a head mounted acoustic device.

Abstract: Headphones provide a voice of a person as binaural sound during a telephone call with a user wearing the headphones. The headphones store HRTFs of the user, and a digital signal processor (DSP) processes the voice of the person with the HRTFs so the voice of the person externally localizes as binaural sound at a sound localization point (SLP) that is in empty space at least one meter away from a head of the user. The headphones track head movements of the user and select the HRTFs based on the head movements during the telephone call.

Abstract: An apparatus for generating an audio output signal is provided. The audio output signal has two or more audio output channels and is generated from an audio input signal having two or more audio input channels. The apparatus includes a provider and a signal processor. The provider is adapted to provide first covariance properties of the audio input signal. The signal processor is adapted to generate the audio output signal by applying a mixing rule on at least two of the two or more audio input channels. The signal processor is configured to determine the mixing rule based on the first covariance properties of the audio input signal and based on second covariance properties of the audio output signal, the second covariance properties being different from the first covariance properties.

Abstract: A smartphone provides a voice of a person during a telephone call to a user of the smartphone in binaural sound. The smartphone stores HRTFs of the user, and a digital signal processor (DSP) processes the voice of the person with the HRTFs so the voice of the person externally localizes in the binaural sound at a sound localization point (SLP) that is in empty space one meter away from a head of the user.

Abstract: Methods and apparatus include headphones that measure an interaural time difference (ITD) and sound impulse responses to determine user-specific head-related transfer functions (HRTFs) for a listener. The headphones include head tracking, and sound is adjusted so that a location of a source of the sound continues to originate from a sound localization point (SLP) in empty space that is at least one meter away from the head of the listener while the head orientations of the listener change with respect to the SLP.

Abstract: A mobile entertainment and announcement system has a trailer base and a trailer body mounted on the trailer base. A trailer hitch projecting from a forward end of the trailer base allows the mobile entertainment and announcement system to be transported to a desired venue by connecting the trailer hitch to a vehicle. A telescoping boom is supported by at least one of the trailer base and the trailer body, and is adapted to move from a transport position to a set-up position and to a broadcast position. At least one broadcast speaker is mounted on the telescoping boom when the telescoping boom is in a set-up position, so that when the telescoping boom is moved from the transport position to the broadcast position, the broadcast speaker is moved to a higher elevation above the trailer base. An audio component housed in the trailer body is connected to the broadcast speaker for transmitting audio content to an audience over a larger area.