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

Abstract: [Problem] To provide an earphone device with which a user can satisfactorily listen to audio data regardless of the wearing state of an earphone.

Abstract: Example techniques relate to audio generation in a media playback system. Based on one or more first functions and first characteristics of an area, the system may generate first audio that includes a first audio signal and a second audio signal. The system provides the first audio signal to at least one first audio driver and the second audio signal to at least one second audio driver, thereby causing a first playback device and a second playback device to play back the first audio synchronously. The system receives second characteristics of the area and based on one or more second functions and the second characteristics, generates second audio comprising a third audio signal and a fourth audio signal. The system provides the third audio signal to the at least one first audio driver and the fourth audio signal to the at least one second audio driver.

Abstract: There are included a first speaker processing step in which a first speaker 2 produces sound based on a first filtered signal, a gain multiplication step in which a gain multiplication unit 4 generates a gain multiplied signal by multiplying a second filtered signal by a gain, a second speaker processing step in which a second speaker 5 produces sound based on the gain multiplied signal, and a gain control step in which a gain control unit 7 controls the gain such that a root mean square of a collected sound signal collected by a microphone 6 installed at a mute position, which is a position at which the sound produced by the first speaker and the sound produced by the second speaker are to be muted, is relatively small.

Abstract: An electronic device is provided. The electronic device includes a housing, a support member disposed inside the housing and forming a connection passage, a printed circuit board disposed on the support member and including a first area and a second area spaced apart from the first area, and a conductive pattern disposed on the support member and including a first portion and a second portion, wherein the first portion may be electrically connected to a first electrode disposed in the first area, the second portion may be electrically connected to a second electrode disposed in the second area, and the first area and the second area may be electrically connected to each other via a shunt member.

Abstract: Encoding/decoding techniques where multiple transform parameter sets are encoded together with a rendered playback presentation of an input audio content. The multiple transform parameters are used on the decoder side to transform the playback presentation to provide a personalized binaural playback presentation optimized for an individual listener with respect to their hearing profile. This may be achieved by selection or combination of the data present in the metadata streams.

Abstract: A display device includes: a control unit that specifies a sound source position from an image displayed on a display unit, and performs different types of signal processing on a voice signal synchronized with the image according to the sound source position, the voice signal being output to a plurality of sets of speaker units, the plurality of sets of speaker units including at least one set of speaker units provided on an upper portion of the display unit.

Abstract: A method for detecting anomalies has the following steps: Obtaining a long-term recording having a plurality of first audio segments associated to respective first time windows; analyzing the plurality of the first audio segments to obtain, for each of the plurality of the first audio segments, a first characteristic vector describing the respective first audio segment; obtaining a further recording having one or more second audio segments associated to respective second time windows; analyzing the one or more second audio segments to obtain one or more characteristic vectors describing the one or more second audio segments ABCD; matching the one or more second characteristic vectors with the plurality of the first characteristic vectors to recognize at least one anomaly, like a temporal, sound or spatial anomaly.

Abstract: Disclosed embodiments include systems and methods for position-based playback of multichannel audio. Some embodiments include (i) receiving, via a network interface of a first playback device, an indication of a position of a second playback device, (ii) determining, based on the received indication of the position of the second playback device, an audio characteristic of the first playback device, and (iii) playing back, via one or more transducers associated with the first playback device, a first portion of media content according to the determined audio characteristic in substantial synchrony with playback, via the second playback device, of a second portion of the media content.

Abstract: Disclosed embodiments include systems and methods for position-based playback of multichannel audio. Some embodiments include (i) determining first and second positions of first and second playback devices, respectively, (ii) receiving media content comprising multichannel content, (iii) generating first audio according to a first sound characteristic based on the received multichannel audio content and the first position of the first playback device, (iv) generating first audio according to a first sound characteristic based on the received multichannel audio content and the first position of the first playback device, and (v) causing synchronous playback of the generated first audio via the first playback device and the generated second audio via the second playback device.

Abstract: The present invention discloses an audio processing circuit, wherein when the audio processing circuit determines that a signal being processed is a small signal, an output stage uses a regulated supply voltage provided by a voltage regulator, and the output stage uses an open-loop structure to reduce noise of an output audio signal; and when the audio processing circuit determines that the signal being processed is a large signal, the output stage directly uses the supply voltage without using the regulated supply voltage, and the output stage uses a closed-loop structure to reduce the total harmonic distortion of the output audio signal. By using the present invention, the audio processing circuit can have a good performance indicator with a small chip area design.

Abstract: Provided are a silicon-based microphone device and an electronic device. The silicon-based microphone device comprises a circuit board, a shielding housing and at least two differential silicon-based microphone chips, wherein at least two sound inlet holes are provided on the circuit board, the shielding housing covers one side of the circuit board and forms a sound cavity with the circuit board, the silicon-based microphone chips are all located inside the sound cavity, the differential silicon-based microphone chips are respectively disposed at the sound inlet holes, and a back cavity of each differential silicon-based microphone chip is communicated with the sound inlet hole at the corresponding position, each of the differential silicon-based microphone chips comprises a first microphone structure and a second microphone structure, all of the first microphone structures are electrically connected, and all of the second microphone structures are electrically connected.

Abstract: Certain embodiments relate generally to modifying audio playing on a first device based on detection of that audio by a second device. Other embodiments relate to transferring audio between a first device and a second device. More particularly, audio playing from a first device may be muted, stopped, or adjusted in volume based on detection of that audio by, or interaction with, a second device. Likewise, audio may be transferred from a first device to a second device based on communications between the first and second devices, proximity of the first and second devices relative to one another, proximity of a user to either the first or second device, and so on.

Abstract: A display device, a sound producing control method, a method and device for determining parameters. The display device includes a plurality of exciters arranged on the display panel, the display panel producing sound on a screen through the plurality of exciters; wherein, the plurality of the exciters are formed as at least one sound producing unit, each sound producing unit includes at least two exciters, and each exciter in the same sound producing unit is connected to a same audio signal input end, in one sound producing unit, a signal filter is provided on an audio transmission line from the audio signal input end to at least one exciter, and used for low frequency compensation of a sound produced by the entire sound producing unit.

Abstract: Various techniques pertaining to non-coherent noise reduction for audio enhancement on a multi-microphone mobile device are proposed. A processor receives a plurality of signals from a plurality of audio sensors corresponding to a plurality of channels responsive to sensing by the plurality of audio sensors. The processor then performs a non-coherent noise reduction on one or more signals of the plurality of signals to suppress one or more non-coherent noises in each of the one or more signals based on a respective signal-to-noise ratio (SNR) associated with each of the one or more signals. The processor further combines the plurality of signals subsequent the noise reduction to generate an output signal.

Abstract: A vibration device enabling a large area and a display apparatus including the vibration device are provided. The vibration device may be configured to include a vibration array including a plurality of vibration modules which are spaced apart from one another by a first interval of less than 3 cm in a first direction.

Abstract: The invention relates to a hearing aid a cochlear implant comprising a) at least one input transducer for capturing incoming sound and for generating electric audio signals which represent frequency bands of the incoming sound, b) a sound processor which is configured to analyze and to process the electric audio signals, c) a transmitter that sends the processed electric audio signals, d) a receiver/stimulator, which receives the processed electric audio signals from the transmitter and converts the processed electric audio signals into electric pulses, e) an electrode array embedded in the cochlear comprising a number of electrodes for stimulating the cochlear nerve with said electric pulses, and f) a control unit configured to control the distribution of said electric pulses to the number of said electrodes.

Abstract: One embodiment provides a computer-implemented method that includes acquiring, via at least one microphone, sound pressure data from a loudspeaker in a room. The sound pressure data is input into an artificial intelligence (AI) model. The AI model automatically estimates, without user interaction, at least one of energy average (EA) in a listening area or total sound power (TSP) produced by the loudspeaker. The AI model is trained prior to automatically estimating the at least one of the EA in the listening area or the TSP produced by the loudspeaker.

Abstract: A wireless system includes a first device in operative communication with an master device over a first link, and a second device to eavesdrop transmissions from the master device to the first device. The wireless system, based on a reference clock signal, adjusts at least one of a first clock signal for the first link and a second clock signal for a second link between the second device and the first device.

Abstract: The implementation of modal processors, which involve the parallel combination resonant filters, may be costly for applications such as artificial reverberation that can require thousands of modes. In one embodiment, the input signal is decomposed into a plurality of subbands, the outputs of which are downsampled. In each downsampled band, resonant filters are applied at the downsampled sampling rate, and their output is upsampled and filtered to form the band output. In these and other embodiments, a feature of responses of the mode filters have been optimized to minimize an aspect of a residual error after a point in time.