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: 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: 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: 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: 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: 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: 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.

Abstract: A virtual surround sound production circuit includes: a subtraction unit configured to perform subtraction operation on first and second signals to output a third signal; a first addition unit configured to perform addition operation on the first and second signals to output a center channel signal; a surround sound adjustment unit configured to adjust the third signal or another third signal that has undergone frequency compensation to output a fourth signal; a center compensation unit configured to adjust the center channel signal to output a fifth signal; a second addition unit configured to perform addition operation on the first, fourth and fifth signals to output a first output signal; a phase inverter unit configured to perform phase inversion on the fourth signal to generate a phase-inverted signal; and a third addition unit configured to perform addition operation on the second, fifth and phase-inverted signals to output a second output signal.

Abstract: An audio apparatus and method employ a first renderer circuit, wherein the first renderer circuit is arranged to render audio elements by generating a first set of audio signals for a set of loudspeakers; and a second renderer circuit, wherein the second renderer circuit is arranged to render audio elements by generating a second set of audio signals for headphones. The audio apparatus and method select between the first renderer circuit and the second renderer circuit such that the rendering of at least a first part of a first audio element is in response to a first audio rendering property indicator which is received in an audio signal. The first audio rendering property indicator is indicative of an audio format of the first audio element.

Abstract: Membrane unit for a speaker device, the membrane unit comprising a membrane and at least two coils. The respective longitudinal axes of the at least two coils are arranged in parallel along a main axis, a main surface being formed perpendicular to the main axis. The at least two coils are positioned adjacent to each other and mechanically connected to the membrane, the membrane being substantially flat and having a major surface parallel to the main surface. Wiring to the at least two coils is provided at least partly on a flat part of the membrane.

Abstract: Embodiments relate to audio processing for opposite facing speaker configurations that results in multiple optimal listening regions around the speakers. A system includes a left speaker and a right speaker in an opposite facing speaker configuration, and a crosstalk cancellation processor connected with the left speaker and the right speaker. The crosstalk cancellation processor applies a crosstalk cancellation to an input audio signal to generate left and right output channels. The left output channel is provided to the left speaker and the right output channel is provided to the right speaker to generate sound including multiple crosstalk cancelled listening regions that are spaced apart.

Abstract: Methods, apparatus, systems and articles of manufacture are disclosed for dynamic volume adjustment via audio classification.

Abstract: Systems and methods for optimizing network microphone devices using noise classification are disclosed herein. In one example, individual microphones of a network microphone device (NMD) detect sound. The sound data is analyzed to detect a trigger event such as a wake word. Metadata associated with the sound data is captured in a lookback buffer of the NMD. After detecting the trigger event, the metadata is analyzed to classify noise in the sound data. Based on the classified noise, at least one performance parameter of the NMD is modified.

Abstract: An audio preamplifier (1) for a microphone, including: a chassis (2) with at least one audio input (17) and at least one audio output, a control panel (3) including a gain selector (10) being arranged on one of the sides of the chassis (2), the chassis (2) further including a receiving compartment inside which a first connector is arranged; and a cartridge (6) comprising a preamplification circuit and a gain control circuit, the cartridge (6) further having a second connector complementary to the first connector, the cartridge (6) being configured for being inserted in a removable manner into the receiving compartment of the chassis (2) so as to connect the first and second connectors. An audio preamplification kit includes an audio preamplifier (1) and a plurality of additional cartridges.

Abstract: Provided are a display apparatus and a computing apparatus including the same. A display apparatus includes: a display module including a display panel configured to display an image, a vibration plate on the display module, a first air gap between the vibration plate and the rear surface of the display module, a vibration module on the vibration plate, and a system rear cover on the vibration plate, a second air gap between the system rear cover and the vibration plate, wherein the vibration plate includes a communication part configured to allow air to flow between the first air gap and the second air gap.

Abstract: A frequency-dependent signal limiter using a paired combination of a bandpass filter (or “selection filter”) and an attenuating equalizing filter (or “cutting filter”) limit the amount of distortion generated by an audio output device in an acoustic rendering system. A bank of one or more of these pairs in series can be used to limit selected frequencies identified as primary causes of distortion in the audio output device to specified levels that avoid distortion.

Abstract: A method for directional signal processing of signals of a microphone array, including first and second microphones generating first and second input signals from an ambient sound, uses the first input signal to form a reference signal transformed into the frequency domain, thereby generating a frequency-space reference signal. The first and second input signals are transformed into the frequency domain, and a first frequency-space directional signal is formed in the frequency domain using the transformed first and second input signals. A frequency-resolved comparison of the frequency-space reference signal with the first frequency-space directional signal or signal derived therefrom in the frequency domain, is used to generate frequency-dependent first gain factors to generate a time filter in the time domain. The reference signal is filtered by the time filter and an output signal is generated from the reference signal filtered by the time filter.