Abstract: A noise canceling audio in/out device includes an audible in/out transducer operable to convert an audible noise signal to a noise signal and convert an audio transmit (TX) signal to an audible output signal. A transducer signal of the audible in/out transducer generated by the audio TX signal is affected by the noise signal. The noise canceling audio in/out device further includes a noise canceling circuit operable to convert a digital TX signal to a TX reference signal, compare the transducer signal with the TX reference signal to produce an analog comparison signal, where the analog comparison signal includes a representation of the audio TX signal and the noise signal, and regulate the transducer signal to substantially match the TX reference signal to remove the effect of the noise signal on the transducer signal.

Abstract: The present technology relates to an information processing device and method, a reproduction device and method, and a program that are capable of performing gain correction more easily. An information processing device includes a gain correction value decision unit that decides a correction value of a gain value for performing gain correction on an audio signal of an audio object in accordance with a direction of the audio object viewed from a listener. The present technology can be applied to a gain decision device and a reproduction device.

Abstract: In one aspect, there is a method for rendering a spatially-heterogeneous audio element. In some embodiments, the method includes obtaining two or more audio signals representing the spatially-heterogeneous audio element, wherein a combination of the audio signals provides a spatial image of the spatially-heterogeneous audio element. The method also includes obtaining metadata associated with the spatially-heterogeneous audio element, the metadata comprising spatial extent information indicating a spatial extent of the audio element. The method further includes rendering the audio element using: i) the spatial extent information and ii) location information indicating a position (e.g. virtual position) and/or an orientation of the user relative to the audio element.

Abstract: A current detection circuit and method for protecting a speaker, and a chip are provided. The current detection circuit includes a current detection module and an analog-to-digital converter. The current detection module samples a current flowing through the speaker based on current mirror technology, and converts the current into a voltage to output a first voltage signal and a second voltage signal. The analog-to-digital converter converts the first voltage signal and the second voltage signal into a PDM code, to calculate a current parameter of the speaker.

Abstract: Methods and systems are provided for an automatic noise control system. Automatic noise control includes evaluating an amplitude of an acceleration acting on an acceleration sensor and generating a reference signal representative of the amplitude of the acceleration, the acceleration being representative of unwanted noise sound generated by a noise source, filtering the reference signal with a noise control transfer function to generate an anti-noise signal, and converting with a loudspeaker the anti-noise signal into anti-noise sound.

Abstract: Disclosed is a headphone including a headband and two ear cups connected to opposite ends of the headband. Each of the ear cups includes a shell, a front cover, and a light sensor. The shell has an open end. The front cover covers the open end of the shell. The front cover is in a basin form and has a peripheral sidewall and a bottom wall. The light sensor is located on the peripheral sidewall of the front cover. The light sensor includes a light-emitting unit and a light detection unit. When a user wears the headphone, light emitted by the light-emitting unit irradiates a flat portion of a back surface of an ear of the user.

Abstract: A computer-implemented method for generating spatial audio with uniform reverberation in a real-time communication session is performed by a real-time communication software application running on an electronic communication device. The method includes removing the reverberation of recorded speech signals from far-end participants by the dereverberation approach, rendering the direct sound parts by filtering the output signals by head-related transfer functions of desired directions, generating reverberant sound parts by convolving the output signals from with uniform room impulse responses or an artificial reverberator, combining direct and reverberant sound components to generate spatialized speech signals. When speakers and listeners are located in two virtual conference rooms, the reverberation of the two rooms are coupled. The reverberant sound parts are then generated by convolving the output signals and coupled RIRs from the two rooms.

Abstract: A wearable device includes a body having fasteners and a frame coupled between two fasteners. The frame includes first and second sections. A first portion of the body includes the first section of the frame and one fastener and a second portion of the body includes the second section of the frame and the other fastener. A speaker and a microphone are connected to the first portion and another speaker and another microphone are connected to the second portion. The body also includes a processor, memory accessible to the processor, and programming in the memory for configuring the processor to selectively activate the speakers and microphones such that a first speaker emits an output sound signal while a first microphone and a second speaker are deactivated and a second microphone captures an input sound signal during the emission of the output sound signal by the first speaker.

Abstract: Embodiments are disclosed for volume control of ear devices. In an embodiment, a method comprises: determining a reference attitude of the ear device based on sensor data collected by motion sensors of the ear device; storing the reference attitude; receiving, with at least one processor, a first user input indicating a user request for a volume control mode of the ear device; receiving a rotation input; determining, with the at least one processor, an amount of volume level increase/decrease based on attitude change relative to the reference attitude due to the rotation input; adjusting a volume level of the ear device in accordance with the volume level increase/decrease; and receiving second user input indicating a second user request to release the volume control mode.

Abstract: The present application provides a sounding device. The sounding device includes a front surface with a sound outlet hole; a back surface; a sounding component and an outer casing. The outer casing is covered outside the sounding component. A cavity is formed between the outer casing and the end of the sounding component close to the back side. A sound guiding channel is arranged in the outer casing. One end of the sound guiding channel is communicated with the sound outlet hole, and the other end of the sound guiding channel is communicated with the cavity. The sounding device of the present application provides a smooth transmission channel for the sound from the back side by arranging the sound guiding channel in the outer casing, thereby strengthening the frontal sound of the sounding device and greatly improving the loudness of the sounding device.

Abstract: The application relates to a sound quality output method. The method includes: obtaining the current volume level input by the user; determine the audio signal gain difference between the current volume level and the baseline volume level according to the current volume level and the preset baseline volume level, determining an equal loudness curve difference between the current volume level and the baseline volume level according to the audio signal gain difference, and the current volume level and the baseline volume level, determining the audio quality response curve corresponding to the current volume level according to the equal loudness curve difference and the pre-stored audio quality response curve corresponding to the baseline volume level, determining the output signal parameters of multiple audio output channels according to the audio quality response curve corresponding to the current volume level. The application optimizes the sound quality effect and enhances acoustic experience.

Abstract: A method for adjusting the clarity of an audio output in a changing environment, including: receiving a content signal; applying a customized gain to the content signal; and outputting the content signal with the customized gain to at least one speaker for transduction to an acoustic signal, wherein the customized gain is applied on a per frequency bin basis such that frequencies of a lesser magnitude are enhanced with respect to frequencies of a greater magnitude and an intelligibility of the acoustic signal is set approximately at a desired level, wherein the customized gain is determined according to at least one of a gain applied to the content signal, a bandwidth of the content signal, and a content type encoded by the content signal.

Abstract: A method performed by a programmed processor of an audio system, the method includes receiving a sound track that has a track length, producing a binaural audio version of a sound track, the binaural audio version having an extended track length performing a fading operation upon the binaural audio version to gradually reduce a signal level of the binaural audio version to below a signal threshold level at a time along the extended track length that corresponds to an end time of the track length of the sound track; and storing the binaural audio version having the track length of the sound track in memory for later transmission to an audio playback device for driving one or more speakers.

Abstract: A system for monitoring sound comprising a primary circuit portion and a secondary circuit portion. The primary circuit portion is adapted to alternate between a first state of operation and a second state of operation. In the first state, the primary circuit portion is adapted to monitor sound. The primary circuit portion is adapted to periodically enter the first state with a predetermined frequency. The secondary circuit portion is adapted to monitor sound while the primary circuit portion is in the second state and provide an indication of the sound monitored to the primary circuit portion. The system is adapted to dynamically adjust the frequency with which the primary circuit portion enters the first state based on the indication.

Abstract: The control device includes: a detection unit that detects a predetermined speech/motion of a user who is present in an output destination space of a directional speaker; an identification unit that identifies, in accordance with the predetermined speech/motion, an audio source of interest among audio sources and a directional speaker as an object to be controlled; and an output control unit that causes the identified directional speakers to output audio supplied from the identified audio source.

Abstract: Disclosed are a sound tracing apparatus and a sound tracing method, and the sound tracing apparatus includes a first acceleration structure generation unit configured to generate a first acceleration structure for a static scene in a sound space, an intersection test execution unit configured to perform an intersection test on each of a plurality of dynamic objects constituting a dynamic scene in the sound space to detect whether or not the dynamic object affects a sound propagation path, a second acceleration structure generation unit configured to select the dynamic objects that affect the sound propagation path as a result of the intersection test and then generate the second acceleration structure for the dynamic scene, and a sound generation unit configured to generate a 3D sound by performing sound tracing based on the first and second acceleration structures.

Abstract: A method and system of improving frequency responses at lower frequencies from a loudspeaker includes simultaneously tuning a transmission line and a tuned port in the same system. The transmission line and the tuned ports are tuned to innate characteristics of the physical and electrical components within the loudspeaker. In one embodiment, an improved frequency response of a loudspeaker is achieved by tuning the transmission line to the frequency of the maximum excursion point of the loudspeakers. In another embodiment the transmission line is tuned to the resonance frequency of the loudspeaker having the tuned port.

Abstract: An audio system that is configured to convert a plurality of audio input channels to object-based audio, and a related computer program product. Correlation between input channels and energy balance between the input channels are determined. The determined correlation and energy balance are mapped to output three-dimensional spatial locations.

Abstract: A sound muffling chamber covers a nose and mouth in an airtight manner. The sound muffling chamber includes a microphone that detects vocal sound waves in the sound muffling chamber and generates a voice signal. A digital signal processing system analyzes the voice signal and generates a voice cancellation signal. A sound silencing chamber includes a speaker that generates out of phase sound waves in response to the voice cancellation signal that superimposes on and cancels the vocal sound waves. A sound decelerator is positioned between the sound muffling chamber and the sound silencing chamber and configured to increase a traveling time of the vocal sound waves such that the vocal sound waves' arrival at the sound silencing chamber may be synchronized with the arrival of a voice cancellation signal. The sound muffling chamber may include inflatable cells separated by slats such that the sound muffling chamber is foldable.

Abstract: Disclosed herein is a system for facilitating stress adaption in a workstation. Accordingly, the system may include microphones disposed on the workstation. Further, the one or more microphones may be configured for generating first sound signals of first sounds associated with an environment of the workstation. Further, the system may include a processing device communicatively coupled with the microphones. Further, the processing device may be configured for analyzing the first sound signals, determining first sound characteristics of the first sounds, determining second sound characteristics of second sounds, and generating second sound signals for the one or more second sounds. Further, the system may include acoustic devices disposed on the workstation. Further, the acoustic devices may be communicatively coupled with the processing device. Further, the acoustic devices may be configured for emitting the second sounds based on the second sound signals.