Abstract: Various embodiments provide an electronic device and a method, the device comprising: an audio module; a display; a connection terminal connected to earphones; a communication interface; and a processor electrically connected to the audio module, the display, the connection terminal, or the communication interface, wherein the processor is set to display execution screens associated with a first application and a second application through a multi-window, respectively, and output an audio signal corresponding to the first application and an audio signal corresponding to the second application to a left terminal or a right terminal of the earphones separately based on a window position corresponding to each execution screen. In addition, other embodiments are also possible.

Abstract: Methods, systems, and computer readable media for a phase array directed speaker are disclosed. One system includes a phase array and a controller. The controller is configured to determine an area to send directed sound. The controller is also configured to generate input instructions for the phase array to send directed sound to the area. The input instructions indicate that at least one speaker in the phase array is to delay outputting an audio signal relative to at least one other speaker in the phase array. The phase array is configured to generate, using the input instructions and ultrasonic signals emitted by the speakers in the phase array, the directed sound.

Abstract: A device is provided for capturing vibrations produced by an object such as a musical instrument such as a cymbal of a drum kit. The device comprises a detectable element, such as a ferromagnetic element, such as a metal shim and a sensor spaced apart from and located relative to the musical instrument. The detectable element is located between the sensor and the musical instrument. When the musical instrument vibrates, the sensor remains stationary and the detectable element is vibrated relative to the sensor by the musical instrument.

Abstract: Array microphone systems and methods having adjustable lobe shapes are provided. The lobe shapes of pickup patterns in an array microphone may be adjusted by weighting the audio signals of subsets of the microphone elements that make up the array. The lobe shapes may be adjusted in a direction independent of a steering vector of the lobe. Users may have greater control of lobes which can result in more efficient and optimal coverage of audio sources in environments.

Abstract: A microphone can include an adapter housing. The adapter housing can include an opening and an outer acoustic port. The microphone can include an internal microphone assembly disposed at least partially within the adapter housing. The internal microphone assembly can include an internal housing having an internal acoustic port. The internal microphone assembly can include a plurality of contacts disposed on the internal housing. The contacts can be accessible through the opening of the adapter housing. An interior of the internal housing can be acoustically coupled to the outer acoustic port via the internal acoustic port.

Abstract: A microphone can include a cover having a series of slits and a nest. The nest can be configured to receive a first diaphragm, a second diaphragm, and a PCB in a stacked arrangement, such that the PCB is positioned between the first diaphragm and the second diaphragm. Also, the first diaphragm can define a first plane, the second diaphragm can define a second plane, and the PCB can define a third plane and the first plane, the second plane, and the third plane can extend parallel to one another. The cover can also include slits having a first length and a second length, and the first length can be greater than the second length. The slits can extend both radially and axially.

Abstract: A MEMS microphone with a hybrid packaging structure is provided. The microphone comprises: a first circuit board; a second circuit board, spaced apart from the first circuit board and parallel to the first circuit board; a packaging cover, covering the second circuit board, and forming an acoustic cavity with the second circuit board; wherein the first circuit board and the second circuit board form an accommodating space in which a pressure sensor is disposed. In the present invention, the chip and electronics of the MEMS microphone are installed on two different circuit boards, respectively, so that an interior space of the microphone is fully utilized, a good heat dissipation is achieved, a better sound transmission effect is achieved, and the microphone can be widely used.

Abstract: A microphone assembly is disclosed including a microelectromechanical system (MEMS) transducer and an electrical circuit disposed in a housing having an external-device interface. The electrical circuit is configured to determine whether a speech characteristic is present in an electrical signal produced by the transducer, attempt to authenticate the speech characteristic, and provide an interrupt signal to the external device interface only upon successful authentication of the speech characteristic.

Abstract: An arrangement (2) to calibrate a capacitive sensor interface (1) includes a capacitive sensor (10) having a capacitance (cmem, cmemsp, cmemsm) and a charge storing circuit (20) having a changeable capacitance (cdum, cdump, cdumm). A test circuit (30) applies a test signal (vtst) to the capacitive sensor (10) and the charge storing circuit (20). An amplifier circuit (40) has a first input connection (E40a) coupled to the capacitive sensor (10) and a second input connection (E40b) coupled to the charge storing circuit (20). The amplifier circuit (40) provides an output signal (Vout) in dependence on a first input signal (?Verr1) applied to the first input connection (E40a) and a second input signal (?Verr2) applied to the second input connection (E40b). A control circuit (60) is configured to trim the capacitance (cdum, cdump, cdumm) of the charge storing circuit (20) such that the level of the output signal (Vout) tends to the level of zero.

Abstract: [Object] To provide a sound source separation technology capable of improving the separation performance. [Solution] An information processing apparatus including: an acquisition section configured to acquire an observation signal obtained by observing a sound; and a sound source separation section configured to separate the observation signal acquired by the acquisition section into a plurality of separated signals corresponding to a plurality of assumed sound sources by applying a non-linear function to a matrix product of an input vector and a coefficient vector corresponding to each of the plurality of sound sources.

Abstract: Provided is a delta-sigma AD converter including a delta-sigma modulating section that outputs a digital signal obtained by performing delta-sigma modulation with an oversampling ratio on an input analog signal; a digital filtering section that filters the digital signal with the oversampling ratio; a control terminal into which an external control signal is input; an output control section that performs control to output an output signal based on the filtered digital signal, according to the external control signal; and a setting section that sets the oversampling ratio based on interval information of the external control signal.

Abstract: Techniques are provided for generating sound using a speaker mounted to an enclosure (e.g., speaker cabinet) wherein a gas pressure level (e.g., air pressure level) inside the enclosure is lower than an ambient air pressure level outside the enclosure. The reduced gas pressure level within the enclosure provides an environment with a reduced pressure level at a back side of a speaker cone of the speaker, which enhances a low frequency response for a given speaker size, while also minimizing resonant frequencies and phase cancellation issues which could otherwise occur with conventional speaker systems in which acoustic sound waves are generated at the back side of the speaker cone. A pressure compensation system is utilized counteract a force applied to the front side of the speaker cone as a result of the gas pressure level inside the enclosure being lower than the ambient air pressure level outside the enclosure.

Abstract: An electret condenser microphone is provided. The electret condenser microphone comprises a diaphragm, a backplate with a metal layer on the side facing the diaphragm and an amplifier on the other side, the input of the amplifier electrically connecting the metal layer, a spacer separating the diaphragm and the backplate PWB; and a metal sleeve accommodating the diaphragm, the backplate and the spacer.

Abstract: Disclosed is a sound recording circuit capable of adjusting microphone sensitivity and preventing sound cracks caused by overly loud sound. The sound recording circuit includes: a microphone bias circuit configured to provide a bias voltage for a microphone circuit; an AC coupling capacitor configured to output an analog input signal according to a microphone signal of the microphone circuit; an analog amplifier circuit configured to output an analog output signal according to the analog input signal; an analog-to-digital converter configured to output a digital input signal according to the analog output signal; a digital amplifier circuit configured to output a digital output signal according to the digital input signal; and a signal detector configured to control an analog gain of the analog amplifier circuit, a digital gain of the digital amplifier circuit, and the bias voltage of the microphone bias circuit.

Abstract: Examples described herein involve a playback device performing one or more playback device actions based on locations of one or more physical contacts on an external surface of the playback device. A processor of the playback device may receive from an array of proximity sensors underlying an external surface of the playback device, location data indicating a physical contact at a location on the external surface. Based on at least the location, the processor may identify a playback device action from a plurality of playback device actions, and cause at least the playback device to perform the identified playback device action. The playback device may further include at least one orientation sensor from which the processor may also receive orientation data indicating an orientation of the playback device. The processor may identify the playback device action also based on the orientation of the playback device.

Abstract: Provided are, among other things, systems, methods and techniques for reducing echo in an audio signal. One representative embodiment involves obtaining an input signal, an estimate of a system-characterizing function, and a reference signal, each at a corresponding sample rate and each divided into a plurality of sub-bands; separately processing such sub-bands, where for a given sub-band the estimate of the system-characterizing function and the reference signal are processed to generate an echo-estimation signal and then the echo-estimation signal is subtracted from the input signal to provide an echo-corrected signal for such given sub-band; and combining the echo-corrected signal from each of different ones of the plurality of the sub-bands to provide a final output signal, with the echo-estimation signal generated using a processing sample rate that is lower than the sample rate for the input signal.

Abstract: Representative embodiments disclose mechanisms to separate and recognize multiple audio sources (e.g., picking out individual speakers) in an environment where they overlap and interfere with each other. The architecture uses a microphone array to spatially separate out the audio signals. The spatially filtered signals are then input into a plurality of separators, so each signal is input into a corresponding signal. The separators use neural networks to separate out audio sources. The separators typically produce multiple output signals for the single input signals. A post selection processor then assesses the separator outputs to pick the signals with the highest quality output. These signals can be used in a variety of systems such as speech recognition, meeting transcription and enhancement, hearing aids, music information retrieval, speech enhancement and so forth.

Abstract: An analog microphone and a control method thereof are disclosed. The analog microphone includes a sensor configured to sense an audio signal and convert the audio signal into an electrical signal; a charge pump configured to provide a bias voltage for the sensor to drive the sensor; a source follower configured to receive the electrical signal and convert the electrical signal into a source follower signal; a gain adjustable amplifier configured to receive the source follower signal, multiply the source follower signal by an amplifying factor, and output an amplified signal; and a detecting module configured to adaptively control the bias voltage of the charge pump and the amplified signal of the gain adjustable amplifier in response to the source follower signal of the source follower.

Abstract: A hearing aid device comprising a microphone, a processing unit and a receiver for compensating for a hearing loss in the auditory system of a human is disclosed. The hearing aid device is configured with a microphone inlet assembly which is optimized with regards to maintenance of the hearing aid and improved sound performance. That is, the microphone inlet assembly comprises an inlet element attached to a PCB and further configured to connect with a sound inlet structure.

Abstract: Provided is a delta-sigma AD converter including a delta-sigma modulating section that outputs a digital signal obtained by performing delta-sigma modulation with an oversampling ratio on an input analog signal; a digital filtering section that filters the digital signal with the oversampling ratio; a control terminal into which an external control signal is input; an output control section that performs control to output an output signal based on the filtered digital signal, according to the external control signal; and a setting section that sets the oversampling ratio based on interval information of the external control signal.

Abstract: A signal processing apparatus that generates a reproducing signal from an input audio signal includes an information acquisition unit that acquires information about an arrangement of a plurality of speakers used for reproduction of a sound that is based on the reproducing signal, a specifying unit that specifies a target range for localization of a sound corresponding to the input audio signal, a setting unit that sets a plurality of virtual sound sources used for localization of a sound based on the specified target range based on the acquired information about the arrangement of the plurality of speakers, and a generation unit that generates the reproducing signal by processing the input audio signal based on setting of the plurality of virtual sound sources.

Abstract: A speaker unit for a vehicle includes a subwoofer unit for outputting sound in a base sound region, an amplifier (AMP) circuit unit mounted on a side of the subwoofer unit for amplifying an acoustic current signal and supplying the amplified acoustic current signal to the subwoofer unit, and an amplifier protection cover unit coupled to the subwoofer unit at one side of the subwoofer unit and configured to cover the amplifier circuit unit.

Abstract: An electroacoustic transducer includes: a diaphragm; a frame that is disposed around the diaphragm so as to surround the diaphragm; an edge that closes a space between the diaphragm and the frame in a manner of forming a membrane so as to connect the diaphragm and the frame; a base body that supports the frame and has a through sound output hole in front of the diaphragm; a magnet that is attached to the frame; and a voice coil body that is attached to the diaphragm. The frame has an annular first fitting portion that is disposed around the diaphragm so as to surround the diaphragm. The base body has a second fitting portion that fits with the first fitting portion. The electroacoustic transducer further includes a sealing member that is made of a thermoplastic substance and seals a space between the first fitting portion and the second fitting portion fitted with each other.

Abstract: A speaker device according to an embodiment includes a vibration element, a driving unit, and a panel. The driving unit applies, to the vibration element, a driving signal where a carrier wave in an ultrasonic wave band is modulated by a sound signal in an audible wave band. The panel is provided with the vibration element and arranged in such a manner that a driving signal is applied to such a vibration element by the driving unit to form a vibration region and first and second ultrasonic waves that are generated from such a vibration region and travel in mutually different directions travel toward respectively different users.

Abstract: The present disclosure provides a magnetic bowl, including: a main body made of a first magnetic material and a reinforcing portion made of a second magnetic material, wherein the reinforcing portion is superposed and fixed on the main body, and the second magnetic material has a higher magnetoconductivity than the first magnetic material. The present disclosure further provides a sounding device using the magnetic bowl. Compared with the related art, the magnetic bowl of the present disclosure has a simple forming process and low production cost, and the sounding device using the magnetic bowl has excellent acoustic performance.

Abstract: A method and circuit for testing an acoustic sensor are disclosed. In a first aspect, the method comprises using electro-mechanical features of the acoustic sensor to measure characteristic of the acoustic sensor. In a second aspect, the method comprises utilizing an actuation signal to evaluate mechanical characteristics of the acoustic sensor. In a third aspect, the method comprises using a feedthrough cancellation system to measure a capacitance of the acoustic sensor. In the fourth aspect, the circuit comprises a mechanism for driving an electrical signal into a signal path of the acoustic sensor to cancel an electrical feedthrough signal provided to the signal path, wherein any of the electrical signal and the electrical feedthrough signal are within or above an audio range.

Abstract: A capacitive MEMS device, a capacitive MEMS sound transducer, a method for forming a capacitive MEMS device and a method for operating a capacitive MEMS device are disclosed. In an embodiment the capacitive MEMS device includes a first electrode structure comprising a first conductive layer and a second electrode structure comprising a second conductive layer, wherein the second conductive layer at least partially opposes the first conductive layer, and wherein the second conductive layer includes a multiple segmentation which provides an electrical isolation between at least three portions of the second conductive layer.

Abstract: A MEMS assembly includes a package, wherein the package includes a substrate and a cover element, wherein a through opening is provided in the cover element, a MEMS component within the package on the cover element, an integrated circuit arrangement within the package on the substrate, and a support component within the package on the substrate, wherein the support component on the substrate is electrically coupled, by first electrical connection lines, to the MEMS component on the cover element and is electrically coupled, by second electrical connection lines, to the circuit arrangement on the substrate in order to produce an electrical connection between the MEMS component and the integrated circuit arrangement.

Abstract: A treble loudspeaker with an improved mounting structure for a phase plug, having a T-shaped iron, a magnet, a washer, a phase plug and a diaphragm. The magnet is located between the T-shaped iron and the washer. The T-shaped iron has a central hole. The phase plug is inserted into the central hole and fixedly connected to the T-shaped iron. A soft gasket is provided at a plane junction between the phase plug and the T-shaped iron. The diaphragm is located above the phase plug to cover the phase plug, and a sound guiding gap is provided between the diaphragm and the phase plug.

Abstract: A speaker (10) comprises an enclosure (8) defining an internal volume (11); and at least one sound generator (7), the sound generator (7) comprising one or more surfaces defining an air-path conduit (15) through which air operably passes in and out of the internal volume. The sound generator (7) further comprises a plurality of electrodes comprising at least one air-exposed electrode (1, 4) and at least one insulated electrode (2, 3). A voltage source (6) is configured to generate an electrical field between the at least one air-exposed electrode (1, 4) and the at least one insulated electrode (2, 3, 70) to operatively generate a plasma proximal (100) to the plurality of electrodes and within the air-path conduit (15).

Abstract: One embodiment provides a method, including: detecting, using a touch surface, facial contact of a user with an electronic device; determining, based on the facial contact of the user, a position of the electronic device relative to user's face; and toggling, based on the identified position, a characteristic of the electronic device. Other aspects are described and claimed.

Abstract: A thermal model system for estimating a voice coil temperature of a loudspeaker that has frequency dependent parameters to model thermal behavior of the loudspeaker may include a loudspeaker having a voice coil and a magnet, and a thermal model configured to have multiple frequency dependent thermal circuits including the voice coil and the magnet that determine a voice coil temperature which is used to limit input to the loudspeaker to prevent thermal overload of the loudspeaker.

Abstract: A bias circuit for a capacitive sensor may include a variable impedance element coupled to a capacitor of the capacitive sensor wherein an impedance of the variable impedance element is varied in accordance with a temperature associated with the bias circuit and an active feedback circuit coupled between the variable impedance element and an output of a processing circuit for processing a signal generated by the capacitive sensor and configured to drive the variable impedance element to force a direct-current (DC) voltage level of an output of the capacitive sensor to a desired voltage.

Abstract: A microelectromechanical loudspeaker may include: a plurality of elementary loudspeakers each including a drive unit and a diaphragm deflectable by the drive unit, and a controller configured to respectively supply control signals to the drive units. The drive units may be respectively configured to deflect the corresponding diaphragms according to the respective control signals supplied by the controller to generate acoustic waves. The control signal supplied to at least one control unit may have at least one local extremum and a global extremum of a curvature of the control signal with a highest absolute value of the curvature may be located at a position of the control signal preceding a position of the at least one local extremum of the control signal.

Abstract: A digital frequency measuring apparatus includes a frequency divider dividing an input frequency signal and providing a divided frequency signal; a period counter counting clock cycles in a period of the divided frequency signal using a clock signal and providing a period count value for each period; and a digital filter amplifying the period count value using an accumulated gain, converting an amplified period count value into a frequency, and providing a first digital output value. The digital filter determines the accumulated gain using a predetermined stage number and a predetermined decimator factor.

Abstract: The present disclosure relates to a protection system for protecting a MEMS transducer of a MEMS device from electrostatic capture, wherein the MEMS transducer is operable in a normal-sensitivity, mode and in a reduced-sensitivity mode.

Abstract: Facilitating an enhanced linearity and acoustic overload point of a sensor is presented herein. A system can comprise a first conductive component that is biased at a first direct current (DC) voltage; a second conductive component that is biased at a second DC voltage that is opposite in polarity to the first DC voltage; a third conductive component that is capacitively coupled to the first conductive component and the second conductive component; and a feedback component that generates a non-inverted output signal, comprising a sum of buffered signals generated via capacitive coupling between the third conductive component and the first and second conductive components, generates an inverted output signal comprising an amplified inversion of the non-inverted output signal, and applies the inverted output signal to the third conductive component.

Abstract: In the present invention, a digital electroacoustic transducer apparatus with a noise canceling system includes: a signal processing circuit that generates a digital processing signal based on a digital signal from a sound source; a first voice coil that receives the digital processing signal; a microphone that picks up noise and generates a noise signal; a noise canceling circuit that generates a cancel signal based on the noise signal; and a second voice coil that receives the cancel signal, the first voice coil and the second voice coil driving a diaphragm, thereby avoiding a difference between the phase of the cancel signal and the phase opposite to that of noise.

Abstract: An apparatus includes a printed wiring board (PWB) that defines an aperture. A microphone is mounted on the PWB such that the aperture provides an acoustic path to the microphone. An acoustic interface member defines a cavity that is acoustically coupled to the microphone via the aperture. A first gasket between the printed wiring board and the acoustic interface member forms an acoustic seal. A housing is included, and a second gasket is disposed between the acoustic interface member and the housing to form an acoustic seal. An acoustic chamber is defined by a sealed volume that extends from a first (bottom/inner) surface of the housing down to a junction between the microphone and the PWB. The housing defines apertures which provide an acoustic path between a region external to the housing and the acoustic chamber. The acoustic chamber and the apertures in the housing form a Helmholtz resonator.

Abstract: The present disclosure relates to a microphone driving device and a digital microphone including the same. A microphone driving device according to an embodiment of the inventive concept includes a voltage-to-current converter, a current-to-voltage converter, an analog-to-digital converter, a digital amplification unit, and a gain controller. The voltage-to-current converter converts an acoustic signal to an output current signal based on a gain control signal. The current-to-voltage converter converts the output current signal to an amplified voltage signal. The analog-to-digital converter converts the amplified voltage signal to a digital signal. The digital amplification unit amplifies the digital signal to an amplified digital signal based on the gain control signal. The gain controller generates a gain control signal. The microphone driving device and the digital microphone including the same according to the inventive concept may have a wide dynamic range and reduce the influence of noise.

Abstract: A method of reducing power consumed by an audio playback system includes: receiving a signal from a digital audio interface; comparing a sound pressure level (SPL) response of a first frequency band of the signal to a total SPL response of the signal, the first frequency band including frequencies lower than a selected frequency; and attenuating signal components in the first frequency band of the signal based upon the comparison of the SPL response of the first frequency band of the signal to the total SPL response of the signal.

Abstract: Embodiments of the disclosure generally include a method and apparatus for receiving and separating unwanted external noise from an audible input received from an audible source using an audible signal processing system that contains a plurality of audible signal sensing devices that are arranged and configured to detect an audible signal that is received from any position or angle within three dimensional space. The audible signal processing system is configured to analyze the received audible signals using a first signal processing technique that is able to separate unwanted low frequency range noise from the received audible signal and a second signal processing technique that is able to separate unwanted higher frequency range noise from the received audible signal. The audible signal processing system can then combine the signals processed by the first and second signal processing techniques to form a desired audible signal that has a high signal-to-noise ratio throughout the full speech range.

Abstract: A method for calibrating a microphone that includes a transducer element and an ASIC. The method includes calibrating the frequency characteristic of the ASIC such that the sensitivity (Smic(fLLF)) of the microphone at a predetermined cutoff frequency (fLLF) shows a predefined reduction (?) compared to the sensitivity (Smic(fstandard)) of the microphone at a standard frequency (fstandard). Another aspect of the present invention concerns a microphone.

Abstract: A first lifetime estimate is determined for a first acoustic component of a first audio playback device configured with first signal processing settings. A second audio playback device is configured with second signal processing settings. A second lifetime estimate is determined for a second acoustic component of the second audio playback device configured with the second signal processing settings, where the second lifetime estimate is determined based on first lifetime estimate and where the first acoustic component is substantially similar to the second acoustic component.

Abstract: Embodiments of the present application disclose a method and an apparatus for obtaining vibration information and user equipment.

Abstract: A microphone apparatus including a MEMS transducer, an acoustic activity detector, a local oscillator, and an external-device interface standardized for compatibility with devices from different manufacturers is disclosed. The microphone apparatus has a first mode of operation during which the apparatus is clocked by the internal clock signal when the acoustic activity detector processes digital data for acoustic activity, and a second mode of operation during which the microphone apparatus is clocked by an external clock signal received at the external-device interface after voice activity is detected by the acoustic activity detector.

Abstract: A pin-covering apparatus applied to a plurality of pins of an emitting end of a bi-directional optical module comprises a plurality of mediums and a plurality of metal sleeves. The mediums respectively cover the pins. The metal sleeves cover the mediums by taking the pins as centers, respectively. The metal sleeves are disposed at the emitting end of the bi-directional optical module. A bi-directional optical device using the pin-covering apparatus is also disclosed.

Abstract: A hearing device adapted for use by a wearer comprises an audio streaming circuit configured to receive electromagnetic audio streaming via a first communication link. A configuration circuit is configured to receive configuration parameters via a second communication link different from the first communication link for configuring the hearing device to receive the electromagnetic audio streaming. Control circuitry of the hearing device configures the hearing device to enable reception of the electromagnetic audio streaming in accordance with the received configuration parameters.

Abstract: In order to adjust an electric device, it is proposed to integrate a programmable memory unit into the device and to address said programmable memory unit without enlarging the footprint, via contact areas that are obtained by dividing previous contact areas. In this case, an adjustment value in particular for compensating for a fault tolerance is fed into the memory unit, an operating parameter being readjusted with the aid of said adjustment value. In each case two divided contact areas are short-circuited via a common soldering location during the mounting of the device.

Abstract: A method is provided for encoding multiple microphone signals into a composite source-separable audio (SSA) signal, conducive for transmission over a voice network. The embodiments enable the processing of source separation of the target voice signal from its ambient sound to be performed at any point in the voice communication network, including the internet cloud. A multiplicity of processing is possible over the SSA signal, based on the intended voice application. The level of processing is adapted with the availability of the processing power at the chosen processing node in the network in one embodiment. An apparatus for separating out the target source voice from its ambient sound is also provided. The apparatus includes a directed source separation (DSS) unit, which processes the two virtual microphone signals in the SSA representation, to generate a new SSA signal including the enhanced target voice and the enhanced ambient noise.