Abstract: For online audio/video conferencing applications deployed in an open office environment, using shared conference devices, it can be advantageous to define an acoustic fence. A non-participant audio received from outside the acoustic fence can be considered noise and filtered out before transmission of an audio signal to a far end recipient. Three suppression stages are used to filter the non-participant audio. The first suppression stage uses beamformers for suppression. The second suppression stage is mask-based, and the third suppression stage is reference-based. The three suppression stages filter out non-participant audio signals, having a wide range of frequencies.

Abstract: A virtual reality (VR), augmented reality (AR) and/or mixed reality (MR) system in a physical environment with a plurality of loudspeakers includes a user-worn head mounted display (HMD), a VR/AR/MR processor, and a VR/AR/MR user tracking processor. The HMD includes a microphone and a user tracking device configured to track a user orientation and position. The VR/AR/MR processor delivers a digital video signal to the head-mounted display, and a digital control signal and a digital audio signal to a receiver/preamplifier. The VR/AR/MR user tracking processor receives user tracking data from the HMD user tracking device and provides a digital user tracking data signal to the receiver preamplifier. the receiver/preamplifier receives the digital user tracking data signal, the digital control signal, the digitized microphone signal, and the digital audio signal, and provides a processed audio signal to the amplifier. An amplifier receives the processed audio signal and provides amplified audio signals.

Abstract: A method for generating an active noise reduction filter includes: obtaining a physically noise-reduced signal, the physically noise-reduced signal being a signal received by a feedback microphone after a noise signal passes through an earphone, obtaining a mixed signal, the mixed signal being a signal received by the feedback microphone when the same noise signal is played and the earphone plays a through signal in a through state, calculating an input signal according to the mixed signal and the physically noise-reduced signal, performing adaptive filtering on the input signal and the physically noise-reduced signal according to an adaptive filtering algorithm to obtain a transfer function, and generating an active noise reduction filter according to the transfer function.

Abstract: An audio processor for reproducing multichannel audio has a normal layer processing and a lower layer processing. The normal layer processing is configured for processing one or more channels of the object-specific audio or the multichannel audio belonging to a normal layer. The lower layer processing is configured for processing at least one channel or more channels of the object-specific audio or the multichannel audio belonging to a lower layer. The lower layer processing is configured to feed at least one signal belonging to the one channel or the more channels of the lower layer to a subwoofer output.

Abstract: A microphone unit having only one PCB, a plurality of transducers of acoustic-electric type with digital output mounted on a back side of the PCB, through holes obtained in the PCB in correspondence with each transducer, an A2B audio interface mounted on the back side of the PCB and electrically connected to the transducers; and input/output electrical connectors mounted on the back side of the PCB and electrically connected to said A2B audio interface; the plurality of transducers include a central transducer and at least three peripheral transducers disposed at the same radial distance from the central transducer and equally angularly spaced. A microphone meta-array includes a plurality of microphone units connected by one or more A2b networks.

Abstract: Presented herein are techniques to enhance the audio portion of a video conference. In one embodiment, a method includes determining, using a multi-microphone array, a direction of arrival of sound signals from a user, detecting, using an image from a camera, a face of the user, determining a position of the face of the user with respect to a position of the camera, and forming a spatial beam for the multi-microphone array based on the direction of arrival of sound signals from the user and the position of the face of the user.

Abstract: The present disclosure provides systems and method for adjusting the audio output of a wearable device based on an audio gain profile of a user. The wearable device may receive an audiogram indicating one or more frequency ranges associated with hearing loss. The wearable device may determine the audio gain profile based on the audiogram. The wearable device may use one or more adjustment modules to determine a gain to apply to the audio output based on the audio gain profile. The adjustment modules may include an active noise control module, a hearing assistance module, and a transparency control module. The wearable device may determine the amount of gain to apply using a least mean square algorithm and/or a machine learning model.

Abstract: An electronic device having circuitry, which is configured to estimate a distraction level of an audio object stream, and to modify the audio object stream based on the estimated distraction Audio object level to obtain a modified audio object stream.

Abstract: A digital stethoscope includes a stethoscope housing defining a housing edge. The digital stethoscope also includes a surface region secured to the stethoscope housing at the housing edge, and a number of microphones. The digital stethoscope also includes a processing device disposed within the stethoscope housing and in communication with the microphones. The processing device receives the digital audio data from the microphones.

Abstract: A method for generating an impulse response for a listening point in a room includes: receiving a 3D model of the room, the position of at least one sound source in the 3D model, and acoustic properties of at least one boundary in the 3D model; using a wave based solver for determining a wave based impulse response of a wave based propagation of an impulse emitted at the at least one sound source in the 3D model and received at the listening point within a first acoustic frequency range; using a geometrical acoustics based solver for determining a geometrical impulse response of the ray based propagation of an impulse emitted at the at least one sound source in the 3D model and received at the listening point within a second acoustic frequency range; and generating the impulse response by merging the wave impulse response and the geometrical impulse response.

Abstract: An apparatus for upmixing a downmix audio signal describing one or more downmix audio channels into an upmixed audio signal describing a plurality of upmixed audio channels includes an upmixer and a parameter determinator. The upmixer is configured to apply temporally variable upmix parameters to upmix the downmix audio signal in order to obtain the upmixed audio signal, wherein the temporally variable upmix parameters include temporally variable smoothened phase values. The parameter determinator is configured to obtain one or more temporally smoothened upmix parameters for usage by the upmixer on the basis of a quantized upmix parameter input information. The parameter determinator is configured to combine a scaled version of a previous smoothened phase value with a scaled version of an input phase information using a phase change limitation algorithm, to determine a current smoothened phase value on the basis of the previous smoothened phase value and the phase input information.

Abstract: A wireless microphone system with a first wireless microphone comprising a first microphone element arranged to generate a first audio signal, wherein the first microphone is arranged to generate a first output signal that comprises the first audio signal and information identifying the first wireless microphone; a second wireless microphone with a second microphone element arranged to generate a second audio signal, wherein the second microphone is arranged to generate a second output signal that comprises the second audio signal and information identifying the second wireless microphone; and a monitoring device with a display, wherein the monitoring device is arranged to receive the first and second output signals and to use the display to generate a visualisation of at least one aspect of the first and second audio signals, and to associate each visualisation with a representation of the information identifying the corresponding first or second wireless microphone.

Abstract: Methods and systems for controlling the receipt and transmission of audio transmissions are provided. In one embodiment, a method is provided that includes selecting a first audio channel and transmitting a first audio transmission using the first audio channel. A second audio transmission may then be received that contains an acknowledgment of the first audio transmission. In certain instances, the second audio transmission may be received on the first audio channel. If a second audio transmission is received on the first audio channel, it may be determined that the first audio transmission was successfully received by another computing device.

Abstract: An audio signal playing method and apparatus, and an electronic device are provided. The method comprises: separating, from a first audio signal, a recorded audio signal corresponding to each of at least one sound source; on the basis of the first audio signal, determining a real-time orientation of each of the at least one sound source relative to the head of a user; for each sound source, according to the real-time orientation of the sound source and the recorded audio signal corresponding to the sound source, generating a target direct audio signal corresponding to the sound source, and generating a target reverberated audio signal corresponding to the sound source; and playing a second audio signal that is generated by means of fusing the target direct audio signal and the target reverberated audio signal corresponding to each sound source.

Abstract: A method for designing a loudspeaker excursion estimator comprises measuring an excursion-related parameter for a loudspeaker, for each of a plurality of loudspeaker input signal levels and each of a plurality of loudspeaker input signal frequencies. The method further comprises, for each of the loudspeaker input signal frequencies and based on the measured excursion-related parameters, identifying a respective loudspeaker input signal level corresponding to a target maximum excursion-related parameter value. The method further comprises determining a filter response, based on the identified loudspeaker input signal levels and their respective loudspeaker input signal frequencies, and implementing a filter, based on the calculated filter response, for generating an excursion estimation based on loudspeaker input signal levels.

Abstract: An apparatus configured to: receive spatial audio data comprising audio captured from a space extending around a capture device and directional information; receive video imagery captured by a camera, wherein the extent of the space from which the spatial audio data is captured is greater than the field of view of the camera; associate each audio sources determined from the directional information, with, for audio sources within the field of view, a region of the video imagery that corresponds to the direction towards the audio source, and, for audio sources outside the field of view, a part of an out-of-view graphic, the out-of-view graphic indicative of the spatial extent of the space outside the field of view; display the video imagery and the out-of-view graphic; receive user input; and provide for control of at least one audio capture property of a selected one of said one or more audio sources.

Abstract: A sound effect adjustment method is provided. In the method, a video frame and an audio signal of a corresponding time unit of a target video are obtained. A sound source orientation and a sound source distance of a sound source object in the video frame are determined. Scene information corresponding to the video frame is determined. The audio signal is filtered based on the sound source orientation and the sound source distance. An echo coefficient is determined according to the scene information. Further, an adjusted audio signal with an adjusted sound effect is generated based on the filtered audio signal and the echo coefficient.

Abstract: This disclosure describes an apparatus and method of an embodiment of an invention that is improves Direction of Arrival (DOA) determinations. This embodiment of the apparatus includes a plurality of microphones coupled together as a microphone array used for beamforming, the plurality of microphones are positioned at predetermined locations and produce audio signals to be used to form a directional pickup pattern; a processor, memory, storage, and a power supply operably coupled to the microphone array, the processor configured to execute the following steps: processing an algorithm for a Direction of Arrival (DOA) determination; supplemental processing that improves the DOA processing.

Abstract: The present disclosure provides an acoustic device including a microphone array, a processor, and at least one speaker. The microphone array may be configured to acquire an environmental noise. The processor may be configured to estimate a sound field at a target spatial position using the microphone array. The target spatial position may be closer to an ear canal of a user than each microphone in the microphone array. The processor may be configured to generate a noise reduction signal based on the environmental noise and the sound field estimation of the target spatial position. The at least one speaker may be configured to output a target signal based on the noise reduction signal. The target signal may be used to reduce the environmental noise. The microphone array may be arranged in a target area to minimize an interference signal from the at least one speaker to the microphone array.

Abstract: A system and method implementing an acoustic mapping suite that equips intelligent vehicles with the ability to identify and respond to primary sounds of interest in their environment. The invention performs sound separation, localization and classification techniques, while also generating a sound-based map of the vicinity as well as making inferences and qualitative assessments about sound sources with information over time.