Abstract: The present disclosure is directed to a device that includes a headphone speaker housing that includes a coil having a first terminal and a second terminal that is configured to operate in a sound generation mode and a battery charging mode. A class D amplifier circuit is configured to rectify in a battery charging mode and amplify in a sound generation mode, the class D amplifier is coupled to the first terminal and the second terminal of the coil. The class D amplifier including a first, second, third, and fourth switch, the first terminal coupled between the first and second switch, the second terminal coupled between the third and fourth switch. An audio generation circuit having a third terminal and a fourth terminal, the third terminal coupled between the first and third switch of the class D amplifier and the fourth terminal coupled between the second and fourth switch of the class D amplifier. A battery charging circuit coupled to the third terminal and the fourth terminal.

Abstract: Audio signals are obtained and filtered. First and second audio processing tools are selected, and priorities associated with the first and second audio processing tools are assigned. In response to a higher priority associated with the first audio processing tool, the audio signal is encoded using the first audio processing tool. A multi-channel audio format for decoding the encoded may be selected and the encoded audio signal is decoded to the selected multi-channel audio format. Further, metadata associated with audio objects may be obtained. A gain level for each audio object may be determined based on the metadata and the gain level may be adjusted using the second audio processing tool. Another multi-channel audio format may be selected, and the gain level adjusted audio signal may be encoded based on the other selected multi-channel audio format. The gain level adjusted audio signal is output to be rendered via playback devices.

Abstract: An identifying method of a sound watermark and a sound watermark identifying apparatus are provided. The method includes the following. A synthesized sound signal is received through a network. Noise interference transferred through the network in the synthesized sound signal is determined according to a reflection-cancelling sound signal. A coding threshold is determined according to the noise interference. A sound watermark signal in the synthesized sound signal is identified according to the coding threshold.

Abstract: The present disclosure discloses an acoustic output device. The acoustic output device may include a first acoustic driver including a first diaphragm; a second acoustic driver including a second diaphragm; a control circuit electrically connected with the first acoustic driver and the second acoustic driver respectively, the control circuit provides a first electrical signal for driving a vibration of the first diaphragm, and a second electrical signal for driving a vibration of the second diaphragm, and a phase of the first electrical signal and a phase of the second electrical signal are opposite; and a housing supporting the first acoustic driver and the second acoustic driver, wherein a sound generated by the vibration of the first diaphragm is radiated outward through a first sound guide hole on the housing, and a sound generated by the vibration of the second diaphragm is radiated outward through a second sound guide hole on the housing.

Abstract: Proposed is a space monitoring apparatus using a sound signal and a space monitoring method thereof and, more specifically, is a technology in which a space monitoring apparatus using a sound to determine a spatial condition sets a synthesized sound having a plurality of frequency components to utilize the performance of a speaker efficiently and emits the synthesized sound to monitor the spatial condition.

Abstract: A method for operating a hearing aid which has an auditory canal microphone for capturing sound in the auditory canal of the wearer and an own voice recognition unit for carrying out an adaptive own voice recognition. Sound in the auditory canal is captured by the auditory canal microphone and a signal that corresponds to the sound is output by the auditory canal microphone, and training of the adaptive own voice recognition is controlled only based on the signal output by the auditory canal microphone. A corresponding hearing aid configured to carry out the method.

Abstract: A triple-membrane MEMS device includes a first membrane, a second membrane and a third membrane spaced apart from one another, wherein the second membrane is between the first membrane and the third membrane, a sealed low pressure chamber between the first membrane and the third membrane, a first stator and a second stator in the sealed low pressure chamber, and a signal processing circuit configured to read-out output signals of the triple-membrane MEMS device.

Abstract: The present disclosure provides a sound generating unit, a sound generating module and electronic terminal, comprising a magnetic circuit system, a vibration system, and a circuit board. The voice coil is jointly wound by two voice coil wires, and the two end portions of each voice coil wire respectively form a wire-in end and a wire-out end; the lead-out positions of the two wire-in ends are located on one side of the voice coil, and the lead-out positions of the two wire-out ends are located on the other side of the voice coil; the two wire-in ends and the two wire-out ends are electrically connected to the corresponding pads of the circuit board respectively, and the internal circuit of the circuit board is configured to electrically connect the two wire-in ends together, and electrically connect the two wire-out ends together.

Abstract: Various embodiments of a system and methods for audio source separation via multi-scale feature learning are disclosed.

Abstract: A boundary of a highlight of audiovisual content depicting an event is identified. The audiovisual content may be a broadcast, such as a television broadcast of a sporting event. The highlight may be a segment of the audiovisual content deemed to be of particular interest. Audio data for the audiovisual content is stored, and the audio data is automatically analyzed to detect one or more audio events indicative of one or more occurrences to be included in the highlight. Each audio event may be a brief, high-energy audio burst such as the sound made by a tennis serve. A time index within the audiovisual content, before or after the audio event, may be designated as the boundary, which may be the beginning or end of the highlight.

Abstract: The present disclosure relates generally to digital microphone and other sensor assemblies including a transducer and a delta-sigma analog-to-digital converter (ADC) with digital-to-analog converter (DAC) element mismatch shaping and more particularly to sensor assemblies and electrical circuits therefor including a dynamic element matching (DELM) entity configured to select DAC elements based on data weighted averaging (DWA) and a randomized non-negative shift.

Abstract: Various embodiments are directed to a modular headset assembly comprising a headband, a detachable audio module in wired communication with a computing device via a cord coupled to the detachable audio module, the detachable audio module being detachably secured relative to the headband and comprising: a module housing; and a cord retention element disposed within an interior of the module housing and comprising an at least partially dynamic configuration relative to the module housing defined at least in part by a range of relative rotational motion between the cord retention element and the module housing, wherein the cord retention element may rotate within the module housing at least partially independently of an angular configuration of the module housing; wherein the cord is engaged with the cord retention element such that the cord retention element may rotates along the range of relative rotational motion based on forces acting on the cord.

Abstract: Example systems and methods are audio identification based on data structure are disclosed. An example apparatus includes memory, and one or more processors to execute instructions to execute a constant Q transform on query time slices of query audio, binarize the constant Q transformed query time slices, execute a two-dimensional Fourier transform on query time windows within the binarized and constant Q transformed query time slices to generate two-dimensional Fourier transforms of the query time windows, sequentially order the two-dimensional Fourier transforms in a query data structure, and identify the query audio as a cover rendition of reference audio based on a comparison between the query data structure and a reference data structure associated with the reference audio.

Abstract: A dipole loudspeaker for producing sound at bass frequencies. The dipole loudspeaker includes: a diaphragm having a first radiating surface and a second radiating surface, wherein the first radiating surface and the second radiating surface are located on opposite faces of the diaphragm; a drive unit configured to move the diaphragm at bass frequencies such that the first and second radiating surfaces produce sound at bass frequencies, wherein the sound produced by the first radiating surface is in antiphase with sound produced by the second radiating surface; a frame, wherein the diaphragm is suspended from the frame via one or more suspension elements, wherein the frame is configured to allow sound produced by the first radiating surface to propagate out from a first side of the dipole loudspeaker and to allow sound produced by the second radiating surface to propagate out from a second side of the dipole loudspeaker.

Abstract: A vibration device can include a vibration portion; a first electrode portion disposed on a first surface of the vibration portion; and a second electrode portion disposed on a second surface of the vibration portion, in which each of the first and second electrode portions include at least one of a conductive metal particle and a carbon particle.

Abstract: An electronic device is provided.

Abstract: A method of managing an audio stream that is read by audio playback equipment in a manner that is synchronized on a reference clock is provided. The method includes the following steps implemented in said piece of audio playback equipment: receiving the audio stream, the audio stream being in the form of packets, each packet comprising a succession of audio samples associated with identification information characteristic of a timestamp of said packet; playing the audio samples contained in each packet and storing a time given by the reference clock that is characteristic of the moment at which a predetermined sample of at least one of the packets is played; and sending a message including correspondence information concerning the correspondence between the identification information characteristic of a timestamp of said packet and the time stored in the preceding step.

Abstract: A compression driver is provided. In one embodiment, the compression driver comprises an annular diaphragm, a phasing plug, and a housing, wherein the housing has a rectangular exit proximate to a blade of the phasing plug.

Abstract: The present invention provides an audio control circuit, wherein the audio control circuit includes a USB interface circuit, a specific interface circuit and a processing circuit. In the operation of the audio control circuit, the processing circuit receives a plurality of parameters from a host device through the specific interface circuit; and during an enumeration between the audio control circuit and the host device, the processing circuit uses the plurality of parameters to perform the enumeration.

Abstract: A method and apparatus for audio signal processing in an audio chain to correct a non-linearity of the electroacoustic transducers in the audio chain by adding non-linearities in the audio chain in front of at least one electroacoustic transducer in the audio chain using an approximation of the quadratic function. The method accommodates the psychoacoustical characteristics of the human ear by adding non-linearities in the audio chain in front of at least one electroacoustic transducer in the audio chain approximating by a non-linear fifth degree polynomial function for a pressure change by the human ear up to p_?. The method and apparatus reduce non-linearities of the entire audio chain with the human ear, by adding non-linearities in the audio chain so that an audio chain characteristic reduces the non-linearity of the human ear polynomial approximation to the pressure change p_?=±1 Pa.