Abstract: A sounding body is included in an audio reproduction device mounted on a mobile device. At least a part of the sounding body also serves as at least a part of one of an emblem, a horn section, and a mirror section of the mobile device.

Abstract: A loudspeaker device is presented which includes a zeolite material comprising zeolite particles having a silicon to aluminum mass ratio of at least 200. For an increased pore fraction of pores with a diameter in a range between 0.7 micrometer and 30 micrometer shows an increased shift of the resonance frequency down to lower frequencies has been observed.

Abstract: A vehicle headrest includes a main body portion having a front surface arranged to support a back of a head of a user. The main body portion includes first and second speakers which are disposed within the main body portion, and a cushion layer that is configured to surround the speakers and provide cushioning for a user's head. The cushion layer includes acoustic channels for controlling a radiation pattern of acoustic energy radiated from the speakers.

Abstract: A micro-electro-mechanical system (MEMS) microphone device is provided in the present disclosure. The MEMS microphone device includes a circuit board, an electromagnetic shielding cover mounted on the circuit board to define an accommodating space, electronic components received in the accommodating space and electrically connected to the circuit board, and a metal shielding member disposed between the electronic components and the electromagnetic shielding cover. The electromagnetic shielding cover has an inner surface and an outer surface; and at least one of the inner surface and the outer surface is provided with a first metal shielding layer. At least one of the electromagnetic shielding cover and the metal shielding member is electrically connected to the circuit board.

Abstract: A stereo microphone for mobile devices, comprising a coupling body provided with elements for electromechanical connection to a mobile device and a supporting body for sound wave transducer means, which are integral therewith, the supporting body being pivoted to the coupling body by way of pivoting elements that allow its rotation about a single rotation axis for transition between a first shooting configuration, which corresponds to the use of the mobile device in portrait mode, and a second shooting configuration, which corresponds to the use of the mobile device in landscape mode.

Abstract: The present embodiments relate to techniques (300) and apparatuses (100, 500) for implementing a flexible transducer for soft-tissue audio production. These techniques (300) and apparatuses (100, 500) enable an audio-production device (102) having a flexible transducer (116, 402) conformed to a person's pinna to create audio within the person's external ear canal.

Abstract: An ear pad to be mounted to a base body to which a speaker unit is fixed includes an elastic body in which a plurality of elastic members having different coefficients of restitution are laminated together in a sound emitting direction of the speaker unit. A skin material covers an exterior of the elastic body. The elastic body includes a first elastic member arranged at a side of the base body, and a second elastic member laminated on and fixed to the first elastic member. The first and second elastic members respectively have ring shapes having a substantially same external diameter. The second elastic member has an inner peripheral surface having a different dimension compared to an inner peripheral surface of the first elastic member, and has a portion having a larger inner diameter than an inner diameter of the first elastic member.

Abstract: A headset driver circuit is described which comprises a connector interface. The connector interface comprises a first terminal, a second terminal and a third terminal for establishing respective electrical connections to a first speaker, a microphone and a common ground node of a headphone, earphone or headset, respectively. A first power amplifier is coupled to the first terminal to supply a first audio output signal to the first speaker of the headset. A first switch arrangement comprises a first ground switch is configured for selectively connecting and disconnecting the second terminal and a ground node of the headset driver circuit. The headset driver circuit further comprises a second ground switch configured for selectively connecting and disconnecting the third terminal and the ground node. The headset driver circuit also comprises a differential preamplifier, e.g.

Abstract: A mobile terminal having a terminal body, an audio output module located in the terminal body, a power supply unit located in the terminal body, an earphone port formed on one side of the terminal body and a controller configured to, when a plug of earphone having a cable-type battery is inserted into the earphone port while performing an audio output function, to select an audio output path for at least one of the audio output module or the earphone inserted in the earphone port to output audio sound according to the audio output function and to charge the power supply unit by the cable-type battery is provided. Accordingly, an audio sound can be output through an audio output path selected according to the intention of the user who has inserted the earphone having the cable-type battery.

Abstract: A user terminal device, and a method of controlling a user terminal device are discussed. The user terminal device includes an ear accessory of the user terminal device including a first electrocardiogram (ECG) electrode contacted with an ear of a user, a second ECG electrode contacted with a hand of the user, and a first speaker configured to output an audio signal; a display; and a controller configured to display, on the display, bio information based on a first ECG signal from the first ECG electrode when the ear of the user is contacted with the first ECG electrode, and control an audio playback in response to a second ECG signal from the second ECG electrode when the hand of the user is contacted with the second ECG electrode.

Abstract: A method includes comparing a voltage level detected by a detection circuit to a set of threshold voltages to determine a type of a headphone, determining the headphone to be a first type when a first signal travels through the first path to ground, and determining the headphone to be a second type when a second signal travels through the second path. The detection circuit includes a first path, a second path, and a bias capacitor. The headphone of the second type includes a plug that has a microphone connection.

Abstract: This application relates to earbuds configured with one or more biometric sensors. At least one of the biometric sensors is configured to be pressed up against a portion of the tragus for making biometric measurements. In some embodiments, the housing of the earbud can be symmetric so that the earbud can be worn interchangeably in either a left or a right ear of a user. In such an embodiment, the earbud can include a sensor and circuitry configured to determine and alter operation of the earbud in accordance to which ear the earbud is determined to be positioned within.

Abstract: A non-directional microphone includes a housing, a microphone unit attached to the housing, and including a diaphragm that receives a sound wave, and a connector portion to which a cable connector including a cable that transmits an audio signal from the microphone unit is connected, and a pressure equalization opening that allows a back-side space of the diaphragm in the housing and an outer space to communicate is provided in the connector portion. In the non-directional microphone, the communication between the back-side space of the diaphragm and the outer space through the pressure equalization opening is cut off when the cable connector is connected to the connector portion, and the back-side space of the diaphragm communicates into the outer space through the pressure equalization opening when the cable connector is disconnected from the connector portion.

Abstract: The present invention discloses a miniature loudspeaker module, a method for enhancing frequency response of a miniature loudspeaker module, and an electronic device. The method comprises the steps of: additionally providing a passive driver in a cavity where an active driver of a miniature loudspeaker module is located, the passive driver and the active driver radiating together, wherein, after the passive driver is additionally provided in the miniature loudspeaker module, the amplitude of a vibrating diaphragm of the active driver shows a local dip on frequency bands below a resonant frequency point F0, and the lowest point of the local dip is corresponding to a frequency point Fb; and, perform, according to amplitude characteristics of the vibrating diaphragm of the active driver of the miniature loudspeaker module additionally provided with the passive driver, matching enhancement to an input signal of the active driver.

Abstract: Accordingly, the present disclosure is directed to an audio capturing enhancement method and an audio capturing system using the same method. The audio capturing system includes at least but not limited to two microphones for recording an audio data, an amplifier coupled to the at least two microphones for adjusting the audio data by applying automatic gain control (AGC) in order to generate a gain adjusted data that is within a predefined level, and a processing circuit coupled to the amplifier for calculating a linear predictive coding (LPC) residue of the gain adjusted data, determining from the LPC residue a first source at a first direction relative to the at least two microphones based on time different of arrival (TDOA), and attenuating any source at a second direction that is outside of a predefined direction.

Abstract: A method includes generating a command at a first microphone and sending the command from the first microphone to a second microphone. The command is sent to the second microphone via a bus that is coupled to the first microphone and to the second microphone.

Abstract: An analog signal processing circuit of a microphone includes a bias circuit including a first sub-circuit which receives a signal from the microphone to output a first signal and a second sub-circuit which receives a reference voltage to output a second signal. A fully differential circuit receives the first signal and the second signal to output a fully differential signal. Each of the first sub-circuit and the second sub-circuit includes a bias sub-circuit to apply a bias voltage.

Abstract: An apparatus for computing filter coefficients for an adaptive filter is disclosed. The adaptive filter is used for filtering a microphone signal so as to suppress an echo due to a loudspeaker signal. The apparatus has: an echo decay modeling means for modeling a decay behavior of an acoustic environment and for providing a corresponding echo decay parameter; and computing means for computing the filter coefficients of the adaptive filter on the basis of the echo decay parameter. A corresponding method has: providing echo decay parameters determined by means of an echo decay modeling means; and computing the filter coefficients of the adaptive filter on the basis of the echo decay parameters.

Abstract: The disclosure includes a system and method for sonically customizing an audio reproduction device. The system includes a processor and a memory storing instructions that when executed cause the system to determine an application environment associated with an audio reproduction device associated with a user; determine one or more sound profiles based on the application environment; provide the one or more sound profiles to the user; receive a selection of a first sound profile from the one or more sound profiles; and generate tuning data based on the first sound profile, the tuning data configured to sonically customize the audio reproduction device.

Abstract: The present technology substantially reduces undesirable effects of multi-level noise suppression processing by applying an adaptive signal equalization. A noise suppression system may apply different levels of noise suppression based on the (user-perceived) signal-to-noise-ratio (SNR) or based on an estimated echo return loss (ERL). The resulting high-frequency data attenuation may be counteracted by adapting the signal equalization. The present technology may be applied in both transmit and receive paths of communication devices. Intelligibility may particularly be improved under varying noise conditions, e.g., when a mobile device user is moving in and out of noisy environments.

Abstract: Systems and methods are provided for device playback calibration. An example implementation involves a network device detecting, at a first time, a first audio content rendered by at least one first playback device. The example implementation also involves the network device detecting, at a second time, a second audio content rendered by at least one second playback device. The example implementation further involves causing, via the network device, determination of an equalization adjustment for one or more of the at least one first playback device and the at least one second playback device based on an analysis of the detected first audio content and the detected second audio content.

Abstract: Embodiments disclosed herein enable detection and improvement of the quality of the audio signal using a mobile device by determining the loss in the audio signal and enhancing audio by streaming the remainder portion of audio. Embodiments disclosed herein enable an improvement in the sound quality rendered by rendering devices by emitting an test audio signal from the source device, measuring the test audio signal using microphones, detecting variation in the frequency response, loudness and timing characteristics using impulse responses and correcting for them. Embodiments disclosed herein also compensate for the noise in the acoustic space by determining the reverberation and ambient noise levels and their frequency characteristics and changing the digital filters and volumes of the source signal to compensate for the varying noise levels.

Abstract: Disclosed are an apparatus and method of processing an audio signal to optimize audio for a room environment. One example method of operation may include recording the audio signal generated within a particular room environment and processing the audio signal to create an original frequency response based on the audio signal. The method may also include identifying a target sub-region of the frequency response which has a predetermined area percentage of a total area under a curve generated by the frequency response, determining whether the target sub-region is a narrow energy region, creating a filter to adjust the frequency response, and applying the filter to the audio signal.

Abstract: A multi-channel audio system that can provide a variable sub-sample delay between two or more audio channels. In one embodiment, a variable timing clock generator generates multiple clock signals where each may have different phase, and the clock generator can vary the phase difference, in accordance with a sub-sample delay setting input. These clock signals are used by respective digital-to-analog converters (DACs) to convert the digital audio channels into analog form. In another embodiment, a variable delay block is added to an oversampling DAC, on a per channel basis. Other embodiments are also described and claimed.

Abstract: Described herein are systems, devices, and methods for generating a synchronized time value for a wireless speaker having a first timer and a second timer. In some embodiments, the systems, devices, and method may include reading a first timer value associated with the first timer, wherein the first timer is synchronized with a wireless network. Further, a first timer value associated with the second timer may be read. The first timer value associated with the second timer may be correlated to the first timer value associated with the first timer to generate the synchronized time value for the wireless speaker.

Abstract: A method for forming a complete ring network of a plurality of Bluetooth® speakers, the method including populating a configurable speaker register of each of the plurality of Bluetooth® speakers with an address of an upstream Bluetooth® speaker that is in the plurality of Bluetooth® speakers, populating the configurable speaker register of each of the plurality of Bluetooth® speakers with an address of a downstream Bluetooth® speaker that is in the plurality of Bluetooth® speakers, and coupling an audio source to one Bluetooth® speaker of the plurality of Bluetooth® speakers.

Abstract: A method of operating an audio system having first and second sources includes outputting a first audio signal from the first source on first and second speakers. In response to a listener input, the first audio signal and a second audio signal from the second source are output simultaneously on the first and second speakers. The first signal is output from each of the first and second speakers such that the first signal is perceived by a listener to originate from a first direction toward the listener. The second signal is output from each of the first and second speakers such that the second signal is perceived by the listener to originate from a second direction toward the listener. The second direction is offset at least ninety degrees from the first direction.

Abstract: An audio output control method and apparatus using a wired/wireless communication system is provided. An audio output control method of an audio output device connected to at least one other audio output device according to the present disclosure includes converting specific audio data to at least one modified audio signal designated to the respective audio output devices, playing at least one of the modified audio signals through a channel designated to the specific audio data, and transmitting a remaining of the modified audio signals to the at least one other audio output device, wherein the modified audio signals include at least one of sound sources and channel characteristics of the specific audio data.

Abstract: The invention relates to a method for rendering a stereo audio signal over a first loudspeaker and a second loudspeaker with respect to a desired direction, the stereo audio signal comprising a first audio signal component (L) and a second audio signal component (R), the method comprising: providing a first rendering signal based on a combination of L and a first difference signal obtained based on a difference between L and R to the first loudspeaker, and providing a second rendering signal based on a combination of R and a second difference signal obtained based on the difference between L and R to the second loudspeaker, such that both difference signals are different with respect to sign and one difference signal is delayed by a delay compared to the other difference signal to define a dipole signal, wherein the delay is adapted according to the desired direction.

Abstract: An audio adaptor and method of communication is provided. The audio adaptor includes a first connector for connecting to a source of audio, a second connector for connecting to a telephone, a third connector for connecting to an audio output device, and an attenuator connected to the first connector and the second connector. The method involves receiving a first audio signal at a line level from the source of audio, receiving a second audio signal at the line level from a telephone, transmitting the first audio signal and the second audio signal at the line level to an audio output device, attenuating the first audio signal from the line level to a microphone level and transmitting the attenuated first audio signal.

Abstract: Several embodiments of the instant invention incorporate a loudspeaker design utilizing hard magnetic poles and yokes. The magnetic field of the loudspeaker is created and guided with a series of critically formed hard magnetic structures guiding the magnetic field from the primary magnet to the gap in a loop of hard magnetic material, replacing the soft magnetic materials normally used. Magnetization of the components of the hard magnetic loudspeaker driver with radial pole patterns—a yoke radial magnet, a washer pole magnet and a yoke upper ring magnet—is performed with these components in their final assembly relationships.

Abstract: The present invention provides an electromagnetic signal converter for an osteophone, comprising a soft magnetic yoke, an electrical coil arranged concentrically to the longitudinal axis of the yoke, an elastically suspended, soft-magnetic armature which, viewed in the direction of the longitudinal axis of the yoke, is separated from the yoke by a working air gap and can move along the longitudinal axis of the yoke, and a permanent magnet which is magnetized in the direction of the longitudinal axis of the yoke in order to generate a magnetic biasing voltage of the yoke and of the armature, characterized in that the permanent magnet and the coil do not overlap one another in the direction of the longitudinal axis of the yoke and that means is provided for dividing the magnetic flux that can be produced by the coil onto at least two flux paths, wherein one flux path runs outside of the permanent magnet, and characterized in that the permanent magnet, yoke and coil are surrounded in the signal converter of the

Abstract: A wearable communication device includes an annular shell, a processing module, a first piezoelectric unit, and a driving unit. A cavity is formed at a part of the annular shell. The processing module processes communication data. The first piezoelectric unit is disposed in the cavity. The driving unit is used to receive the communication data and drive the first piezoelectric unit according to the communication data to make the first piezoelectric unit vibrate and trigger a corresponding audio signal. The wearable communication device can be wirelessly connected to an electronic device without physical wires to improve the convenience greatly.

Abstract: Disclosed is a portable piezoelectric speaker including a body configured to be coupled to and separated from a rear surface of an electronic device, and a piezoelectric speaker module coupled to a predetermined area of the body, wherein the piezoelectric speaker module includes a piezoelectric device, and a vibration transfer member provided to be in contact with at least one area of the piezoelectric device and to be spaced apart from at least one surface of the piezoelectric device.

Abstract: A measurement system includes an ear model including an artificial auricle (51) and an artificial external ear canal (53); and an air-conducted sound gauge (200) that measures air-conducted sound in the artificial external ear canal (53), and while an acoustic device (1) that includes a vibrating body and transmits sound to a user by contacting the vibrating body to a human auricle is placed in contact with the ear model (50) of the measurement system, the measurement system executes control for measurement, with the air-conducted sound gauge, of harmonics in an air-conducted component generated by a pure tone emitted by the acoustic device (1) and control to display the result of the measurement on a display.

Abstract: A voice signal processing apparatus and a voice signal processing method are provided. A last sampling point of an mth original frequency-lowered signal frame is determined according to a phase reference sampling point number of the mth original frequency-lowered signal frame. Here, the phase reference sampling point number corresponds to a middle sampling point of an mth renovating frequency-lowered signal frame, and the last sampling point is phase-matched with a sampling point corresponding to the phase reference sampling point number in the mth original frequency-lowered signal frame. P consecutive sampling points starting from the last sampling point are applied as sampling points of an (m+1)th renovating frequency-lowered signal frame.

Abstract: A hearing aid (200) configured to provide enriched sound, by forming successive chords defined by a common musical scale. The invention further provides a method of providing enriched sound using a hearing aid system.

Abstract: An audio system includes a transient suppression circuit that rectifies an input signal representing target sounds and transient sounds and produces a threshold envelope of the rectified input signal. The transient suppression circuit accumulates the difference between the rectified input signal and the threshold envelope, and normalizes the accumulated difference using the rectified input signal. A gain is calculated using this normalized accumulated difference and applied to the input signal to suppress the transient sounds without substantially affecting the target sounds.

Abstract: An example of an apparatus configured to be worn by a person who has an ear and an ear canal includes a first microphone adapted to be worn about the ear of the person, and a second microphone adapted to be worn at a different location than the first microphone. The apparatus includes a sound processor adapted to process signals from the first microphone to produce a processed sound signal, a receiver adapted to convert the processed sound signal into an audible signal to the wearer of the hearing assistance device, and a voice detector to detect the voice of the wearer. The voice detector includes an adaptive filter to receive signals from the first microphone and the second microphone.

Abstract: The present disclosure relates in a first aspect to a method of superimposing spatial auditory cues to an externally picked-up sound signal in a hearing instrument. The method comprises steps of a generating an external microphone signal by an external microphone arrangement and transmitting the external microphone signal to a wireless receiver of a first hearing instrument via a first wireless communication link. Further steps of the methodology comprise determining response characteristics of a first spatial synthesis filter by correlating the external microphone signal and a first hearing aid microphone signal of the first hearing instrument and filtering the external microphone signal by the first spatial synthesis filter to produce a first synthesized microphone signal comprising first spatial auditory cues.

Abstract: A hearing aid device comprising a receiver unit arranged in a compartment in a shell, wherein the shell comprises a venting passage between first and second opposite faces of the shell to provide air passage from one side of the shell to another, and wherein the shell comprises an opening in the first face into the compartment and a closing member positioned to close off the compartment. The opening is shaped to allow the receiver unit to be inserted into the compartment through the opening, and the compartment is acoustically connected to the venting passage to allow the sound to emit through the venting passage. The receiver unit may be releasably mounted in the compartment such that the receiver unit can be retracted from the shell through the opening.

Abstract: A system for hearing assistance devices to assist hearing aid fitting applied to individual differences in hearing impairment. The system is also usable for assisting fitting and use of hearing assistance devices for listeners of music. The method uses a subjective space approach to reduce the dimensionality of the fitting problem and a non-linear regression technology to interpolate among hearing aid parameter settings. This listener-driven method provides not only a technique for preferred aid fitting, but also information on individual differences and the effects of gain compensation on different musical styles.

Abstract: A wireless interface device for at least one of wireless transmission from an electric analog audio device or wireless reception at an electric analog audio device of an audio signal, comprises an audio connector jack plug or jack socket in communication with a system that is at least one of a wireless internet system or a WLAN-enabled system and connectable to at least one of an audio connector jack plug of the electric analog audio device, or an audio connector jack socket of the electric analog audio device.

Abstract: In a multi-speaker audio system for, e.g., a home entertainment system or other entertainment system, each networked-speaker (wired or wireless) is activated or deactivated as appropriate such that audio play follows a user as the user moves around the home.

Abstract: The volume of a sound generated from a source is monitored. Contextual data is collected from sensors. Based on the collected contextual data listeners of the sound are identified. The identified listeners are grouped into listeners that are bothered by the sound and listeners that are unbothered by the sound. An acceptable range for the volume of the sound is determined based on the bothered listeners and the unbothered listeners.

Abstract: An apparatus comprising: a detector configured to determine at least one microphone is impaired by analyzing at least one audio signal from the at least one microphone; and an controller configured to determine an indicator based on the determination of the impairment of the at least one microphone; and configured to apply the indicator based on the determination of the impairment of the at least one microphone, such that the at least one audio signal is processed based on the indicator.

Abstract: A self calibrating dipole microphone formed from two omni-directional acoustic sensors. The microphone includes a sound source acoustically coupled to the acoustic sensors and a processor. The sound source is excited with a test signal, exposing the acoustic sensors to acoustic calibration signals. The responses of the acoustic sensors to the calibration signals are compared by the processor, and one or more correction factors determined. Digital filter coefficients are calculated based on the one or more correction factors, and applied to the output signals of the acoustic sensors to compensate for differences in the sensitivities of the acoustic sensors. The filtered signals provide acoustic sensor outputs having matching responses, which are subtractively combined to form the dipole microphone output.

Abstract: A method improves computer performance to provide binaural sound in a telephone call. The method includes providing the telephone call with a voice of a calling party and a coordinate location that describes where a voice of the calling party should localize as binaural sound to the called party. The voice of the calling party is convolved to localize at the coordinate location.

Abstract: An apparatus for generating one or more audio output channels is provided. The apparatus includes a parameter processor for calculating output channel mixing information and a downmix processor for generating the one or more audio output channels. The downmix processor is configured to receive an audio transport signal including one or more audio transport channels, wherein two or more audio object signals are mixed within the audio transport signal, and wherein the number of the one or more audio transport channels is smaller than the number of the two or more audio object signals. The audio transport signal depends on a first mixing rule and on a second mixing rule. The first mixing rule indicates how to mix the two or more audio object signals to obtain a plurality of premixed channels. Moreover, the second mixing rule indicates how to mix the plurality of premixed channels.

Abstract: A multi-channel decoder for generating a binaural signal from a downmix signal using upmix rule information on an energy-error introducing upmix rule for calculating a gain factor based on the upmix rule information and characteristics of head related transfer function based filters corresponding to upmix channels. The one or more gain factors are used by a filter processor for filtering the downmix signal so that an energy corrected binaural signal having a left binaural channel and a right binaural channel is obtained.