Abstract: An apparatus includes a noise reduction headphone comprising one or more microphones and an acoustic transducer, the one or more microphones configured to generate an input signal; and a controller comprising one or more processing devices, the controller configured to: process the input signal through one or more noise reduction filters to generate a noise-reduction signal, compare the input signal to an estimate of ambient noise to determine if the energy of the input signal is greater than the estimate of ambient noise, wherein if the energy of the input signal is greater than the estimate of ambient noise by a predetermined amount, a change in the noise reduction signal is suppressed; and generate an output signal, the output signal comprising, at least in part, the noise-reduction signal, wherein the acoustic transducer is configured to produce an acoustic output in accordance with the output signal.

Abstract: In some embodiments, a low voltage system can include a capacitor between an output node of an amplifier and ground, with the output node connectable to a load, and the amplifier configured to operate with a series of pulses. The low voltage system can further include a monitoring circuit configured to monitor a voltage at the capacitor against a desired low voltage value, and a control system configured to generate the series of pulses for the amplifier, and to control charging and discharging of the capacitor based on an output of the monitoring circuit to regulate the voltage at the output node at approximately the desired low voltage value.

Abstract: An information processing method receives settings of a plurality of pieces of information on a plurality of physical spaces that respectively correspond to a plurality of pieces of information on a plurality of logical spaces, receives a first group of a plurality of pieces of first acoustic image localization information that indicates a position of an acoustic image to be localized in each of the plurality of logical spaces using first coordinates in the plurality of logical spaces, receives a change in one piece of first acoustic image localization information, among the first group of the plurality of pieces of first acoustic image localization information, changes other pieces of first acoustic image localization information, among the first group of the plurality of pieces of first acoustic image localization, in response to the received change in the one piece of first acoustic image localization information, and transforms the first group of the plurality of pieces of first acoustic image localizatio

Abstract: Examples of the disclosure relate to apparatus, methods and computer programs for repositioning spatial audio streams. The apparatus is configured to receive a plurality of spatial audio streams wherein the spatial audio streams include one or more audio signals and associated spatial metadata. The apparatus is also configured to receive obtaining repositioning information relating to at least one of the plurality of spatial audio streams and repositioning the at least one of the plurality of spatial audio streams based on the repositioning information.

Abstract: A parametric array loudspeaker 10 includes: a plurality of ultrasonic wave generating elements 23A, 23B arranged in a first direction R1; and a modulation circuit 32 configured to generate a modulation signal in which audible sound is modulated with a carrier wave. The plurality of ultrasonic wave generating elements include a first ultrasonic wave generating element 23A included in a first area being at least one side of both end areas in the first direction, and a second ultrasonic wave generating element 23B included in a second area present between the both end areas. The modulation circuit generates a first modulation signal SG1 to be provided to the first ultrasonic wave generating element and a second modulation signal SG2 to be provided to the second ultrasonic wave generating element. The first modulation signal is a signal in which the phase of at least one of the carrier wave and a sideband wave of the second modulation signal is shifted by a predetermined phase shift amount.

Abstract: An apparatus for synthesizing a spatially extended sound source includes: a spatial information interface for receiving a spatial range indication indicating a limited spatial range for the spatially extended sound source within a maximum spatial range; a cue information provider for providing one or more cue information items in response to the limited spatial range; and an audio processor for processing an audio signal representing the spatially extended sound source using the one or more cue information items.

Abstract: A control method of microphone and an electronic apparatus are provided. In the control method, a switching operation is received. In response to receiving the switching operation, a sound-receiving state of a target program is detected. The sound-receiving state of the microphone is switched through an audio effect processing corresponding to the target program according to the detection result of the sound-receiving state, to switch the sound-receiving state of the target program from one of an activated state and a disabled state to the other. The switching operation is used for switching the sound-receiving state of the microphone. The sound-receiving state represents whether to receive sound through the microphone. The sound-receiving state includes the activated state and the disabled state. The target program is a currently running application. The audio effect processing is implemented by an audio engine between an application layer and a driver layer.

Abstract: An audio bitstream is decoded into audio objects and audio metadata for the audio objects. The audio objects include a specific audio object. The audio metadata specifies frame-level gains that include a first gain and a second gain respectively for a first audio frame and a second audio frame. It is determined, based on the first and second gains, whether sub-frame gains are to be generated for the specific audio object. If so, a ramp length is determined for a ramp used to generate the sub-frame gains for the specific audio object. The ramp of the ramp length is used to generate the sub-frame gains for the specific audio object. A sound field represented by the audio objects with the sub-frame gains is rendered by audio speakers.

Abstract: A biological sound acquisition device (10) includes a housing (100), a first acceleration sensor (210), and a second acceleration sensor (220). The first acceleration sensor (210) is disposed in the housing (100). The first acceleration sensor (210) is mechanically connected to the housing (100) through a first vibration damping member (410). The second acceleration sensor (220) is disposed in the housing (100).

Abstract: An apparatus for immersive audio communication including circuitry configured to: receive at least a first audio data stream and a second audio data stream, wherein at least one of the first and second audio stream includes a spatial audio stream to enable immersive audio during a communication; determine a type of each of the first and second audio streams to identify which of the received first and second audio data streams the spatial audio stream; process the second audio data stream with at least one parameter dependent on the determined type; and render the first audio data stream and the processed second audio data stream.

Abstract: Disclosed is a wireless earset, including an antenna, a plurality of pattern matching circuits, a radio frequency switch circuit, and a wireless chip. The radio frequency switch circuit is electrically connected to one pattern matching circuit based on a control signal, and grounds the connected pattern matching circuit. After the wireless earset is activated, the wireless chip outputs different control signals to control the radio frequency switch circuit to be electrically connected to different pattern matching circuits to detect a received signal strength indication value for each channel within a preset channel range through the antenna electrically connected with different pattern matching circuits. When the wireless chip determines that a detection result generated by the antenna electrically connected to a currently connected pattern matching circuit meets a default condition, the radio frequency switch circuit maintains electrical connection with the currently connected pattern matching circuit.

Abstract: Methods and systems for performing automatic speed-based audio control. One method includes receiving, with an electronic control unit included in a vehicle, a speed of the vehicle and receiving, with the electronic control unit, an audio signal. The method also includes accessing, with the electronic control unit, a plurality of equalization curves based on the speed of the vehicle, each of the plurality of equalization curves associated with the speed of the vehicle and each of the plurality of equalization curves defining a gain adjustment for one of a plurality of frequencies, and, for each curve of the plurality of equalization curves, applying the gain adjustment defined by the curve to one of the plurality of frequencies of the audio signal.

Abstract: An electronic device may include control circuitry that allows a user to listen to audio playing from an external speaker in the environment from the user's personal speaker device such as a pair of headphones, earbuds, or a personal loudspeaker. In response to receiving user input, the control circuitry may gather audio input from the environment with a microphone. The control circuitry may process the detected audio input to identify what audio track is currently playing from the external speaker and to determine which part of the audio track is currently playing. The control circuitry may then start playing the identified audio track from the user's personal speaker device in sync with the audio track playing from the external speaker. The electronic device may search music streaming platforms, public radio station information, and other public music content sources to anticipate and buffer upcoming songs.

Abstract: Presented herein are contaminant-proof microphone assemblies for use with devices/apparatuses, such as auditory prostheses, that include one or more microphones disposed within a housing. A contaminant-proof microphone assembly in accordance with certain embodiments presented herein includes a microphone, a microphone plug, and a contaminant-proof membrane. The microphone plug has a first end coupled to the microphone and a second end that is configured to be positioned adjacent the contaminant-proof membrane. As such, the microphone plug is disposed between a sound inlet of the microphone and the contaminant-proof membrane. The microphone plug may be configured to mate with the housing or a gasket attached to the housing.

Abstract: In accordance with an embodiment, a method for canceling noise in a vehicle includes receiving noise information from an earphone worn by a user inside the vehicle; generating a compensating signal based on the noise information; and outputting the compensating signal through the earphone and an internal speaker of the vehicle.

Abstract: A method and system for determining sound equalisation filters of an audio system having at least one loudspeaker adapted to emit audio signals and a computing device adapted to implement at least one equalisation filter. The system includes modules configured to transmit a measurement signal, and for a first predetermined spatial location, obtain a spectral amplitude curve of a received audio signal, calculate a target spectral amplitude curve, and then make a first determination of a first set of parameterizable equalisation filters which, when applied to the audio signal, reduce a distance between the spectral amplitude curve of the audio signal and the target spectral amplitude curve, and a second determination, from the first set of parameterizable equalisation filters, of a second set of parameterizable equalisation filters, by applying an iterative learning-based method. A first set of optimised equalisation filters is obtained from the second set of parameterizable equalisation filters.

Abstract: A method (900) for rendering audio in a virtual reality rendering environment (180) is described. The method (900) comprises rendering (901) an origin audio signal of an origin audio source (113) of an origin audio scene (111) from an origin source position on a sphere (114) around a listening position (201) of a listener (181). Furthermore, the method (900) comprises determining (902) that the listener (181) moves from the listening position (201) within the origin audio scene (111) to a listening position (202) within a different destination audio scene (112). In addition, the method (900) comprises applying (903) a fade-out gain to the origin audio signal to determine a modified origin audio signal, and rendering (903) the modified origin audio signal of the origin audio source (113) from the origin source position on the sphere (114) around the listening position (201, 202).

Abstract: An audio device is disclosed, the audio device comprising an interface, memory, and a processor, wherein the processor is configured according to any of the following: obtain a first microphone input signal and a second microphone input signal; process the first microphone input signal and the second microphone input signal for provision of an output audio signal; and output the output audio signal. Processing of the input signals includes determine a first set of covariance parameters and a second set of covariance parameters; determine a first beamforming; apply the first beamforming to the first microphone input signal and the second microphone input signal; determine a second beamforming; apply the second beamforming to the first microphone input signal and the second microphone input signal; and provide the output audio signal based on first beamforming output signal and/or second beamforming output signal.

Abstract: In one implementation, a method of transforming a sound into a virtual sound for a synthesized reality (SR) setting is performed by a head-mounted device (HMD) including one or more processors, non-transitory memory, a microphone, a speaker, and a display. The method includes displaying, on the display, an image representation of a synthesized reality (SR) setting including a plurality of surfaces associated with an acoustic reverberation property of the SR setting. The method includes recording, via the microphone, a real sound produced in a physical setting. The method further includes generating, using the one or more processors, a virtual sound by transforming the real sound based on the acoustic reverberation property of the SR setting. The method further includes playing, via the speaker, the virtual sound.

Abstract: An acoustic reproduction method includes: localizing a first sound image at a first position in a target space in which a user is present; and localizing, at a second position in the target space, a second sound image that represents an anchor sound for indicating a reference position.