Abstract: Headphones provide a voice of a person as binaural sound during a telephone call with a user wearing the headphones. The headphones store HRTFs of the user, and a digital signal processor (DSP) processes the voice of the person with the HRTFs so the voice of the person externally localizes as binaural sound at a sound localization point (SLP) that is in empty space at least one meter away from a head of the user. The headphones track head movements of the user and select the HRTFs based on the head movements during the telephone call.

Abstract: An apparatus for generating an audio output signal is provided. The audio output signal has two or more audio output channels and is generated from an audio input signal having two or more audio input channels. The apparatus includes a provider and a signal processor. The provider is adapted to provide first covariance properties of the audio input signal. The signal processor is adapted to generate the audio output signal by applying a mixing rule on at least two of the two or more audio input channels. The signal processor is configured to determine the mixing rule based on the first covariance properties of the audio input signal and based on second covariance properties of the audio output signal, the second covariance properties being different from the first covariance properties.

Abstract: A smartphone provides a voice of a person during a telephone call to a user of the smartphone in binaural sound. The smartphone stores HRTFs of the user, and a digital signal processor (DSP) processes the voice of the person with the HRTFs so the voice of the person externally localizes in the binaural sound at a sound localization point (SLP) that is in empty space one meter away from a head of the user.

Abstract: Methods and apparatus include headphones that measure an interaural time difference (ITD) and sound impulse responses to determine user-specific head-related transfer functions (HRTFs) for a listener. The headphones include head tracking, and sound is adjusted so that a location of a source of the sound continues to originate from a sound localization point (SLP) in empty space that is at least one meter away from the head of the listener while the head orientations of the listener change with respect to the SLP.

Abstract: A mobile entertainment and announcement system has a trailer base and a trailer body mounted on the trailer base. A trailer hitch projecting from a forward end of the trailer base allows the mobile entertainment and announcement system to be transported to a desired venue by connecting the trailer hitch to a vehicle. A telescoping boom is supported by at least one of the trailer base and the trailer body, and is adapted to move from a transport position to a set-up position and to a broadcast position. At least one broadcast speaker is mounted on the telescoping boom when the telescoping boom is in a set-up position, so that when the telescoping boom is moved from the transport position to the broadcast position, the broadcast speaker is moved to a higher elevation above the trailer base. An audio component housed in the trailer body is connected to the broadcast speaker for transmitting audio content to an audience over a larger area.

Abstract: A processing load at a receiving side is reduced in a case where a plurality of classes of audio data is transmitted. A predetermined number of audio streams including coded data of a plurality of groups is generated and a container of a predetermined format having this predetermined number of audio streams is transmitted. Command information for creating a command specifying a group to be decoded from among the plurality of groups is inserted into the container and/or the audio stream. For example, a command insertion area for the receiving side to insert a command for specifying a group to be decoded is provided in at least one audio stream among the predetermined number of audio streams.

Abstract: A local wave field synthesis apparatus, which includes a determination module for determining desired sound pressures and desired particle velocity vectors at a plurality of control points, a computation module for computing sound pressures and particle velocity vectors at the plurality of control points based on a set of filter parameters, an optimization module for computing an optimum set of filter parameters by jointly optimizing computed sound pressures towards the desired sound pressures and computed particle velocity vectors towards the desired particle velocity vectors, and a generator module for generating the drive signals based on the optimum set of filter parameters, wherein the plurality of control points are located on one or more contours around the one or more audio zones.

Abstract: A method and apparatus for controlling earphone noise reduction. The method for controlling earphone noise reduction can include collecting a noise signal of an environment where a terminal microphone is placed, processing the collected noise signal to generate a judgement result, and controlling connected earphones to enable a noise reduction function or disable the noise reduction function according to the judgement result. Embodiments of the present disclosure can be realized without a hardware switch for noise reduction adjustment on earphones. Embodiments of the present disclosure are capable of automatic noise reduction, thereby improving the integration degree of the earphones and also enhancing the noise reduction efficiency of the earphones.

Abstract: An earphone noise reduction method and apparatus, which are applicable to the technical field of wearable devices. The earphone noise reduction method can include collecting, using an earphone microphone, a noise signal of an environment where the earphone microphone is placed. The method can also include transmitting the noise signal to a connected terminal or transmitting a noise value corresponding to the noise signal to the connected terminal. The method can further include receiving a judgement result returned by the terminal, and enabling a noise reduction function or disabling the noise reduction function according to the judgement result. Embodiments of the present disclosure can be realized without keys or toggle switches for noise reduction adjustment on earphones. Embodiments of the present disclosure are capable of automatic noise reduction, thereby improving the integration degree of the earphones and also enhancing the battery life of the earphones.

Abstract: An optimal listening position for the multi-channel audio sound system is determined. A listening area and other audio playback controls may also be defined or modified by a listener. During subsequent audio playback, an imager captures images of at least a portion of the listener. When the listener has moved to a position that is different from a previous position, one or more audio parameters are adjusted for at least one channel in the multi-channel audio system in order to reposition the initial or previous optimal listening position to the current position of the listener. When the listener moves outside the listening area, an out of bounds response is initiated to disable or control the audio playback until the listener moves to a position within the listening area.

Abstract: The field of the invention relates to speaker devices, to speaker panels for speaker devices, to mechanical assemblies and mountings for speaker devices, to uses of speaker devices for example as products or as components for products, and to computer program products operable to control speaker devices. There is provided a speaker device, the speaker device including a body and a speaker panel assembly, the speaker panel assembly movable between an open configuration and a closed configuration, wherein in the open configuration the speaker panel assembly is displaced relative to the body, and in the closed configuration the speaker panel assembly is retracted into the body. The speaker device may be a smart device. The speaker device may be one configured to receive and to store a mobile device.

Abstract: An out-of-head localization processing apparatus according to an embodiment includes headphones, left and right microphones, a measurement unit configured to measure left and right headphone transfer characteristics, respectively, an inverse-filter calculation unit configured to calculate inverse filters of the headphone transfer characteristics, a correction unit configured to calculate correction filters by correcting the inverse filters, and an input unit configured to receive a user input. The correction unit corrects the inverse filters by using a predefined correction function in a first frequency band. The correction unit corrects the inverse filters according to a correction pattern selected based on the user input in a second frequency band.

Abstract: A method of manufacturing an Active Noise Reduction (ANR) device includes providing at a stage during manufacture a pre-completion ANR device in a non-final configuration, the pre-completion ANR device including a plurality of inputs, a plurality of signal processing resources, an output for driving an earphone driver, and a programmable switch arrangement capable of assigning any of the plurality of inputs to any of the plurality of signal processing resources, selecting from the plurality of signal processing resources a subset of signal processing resources to contribute to the output, whereby the remaining signal processing resources of the plurality do not contribute to the output in any mode of operation of the ANR device, and in a configuration step during manufacture, programming the programmable switch arrangement to assign each of at least a subset of the plurality of inputs to a different one of the selected subset of signal processing resources.

Abstract: An earpiece adapts to the acoustics characteristics and needs of its user to provide a perceptually transparent hearing protection, apart from uniform loudness reduction. An occlusion effect (OE) active noise control (ANC) system reduces the augmented perception of one's own voice while occluded. This occlusion effect active control adapts to the specific acoustic characteristics of the user to provided better control of the details occlusion effect reduction, and enhanced performances relative to fixed or one-size-fits-all solutions. An isolation effect (IE) filtering algorithm adapts itself to the user's acoustic characteristics to provide a uniform attenuation either in dB or in phons. Additionally, the device may be used as an in-ear monitor that also adapts to its user characteristics to provide in-ear quality sound.

Abstract: An audio playback control method and a terminal device are described. The method includes starting an application, and playing a background audio of the application, acquiring a foreground audio, and determining duration and volume of the foreground audio. If the duration of the foreground audio is greater than a first threshold and the volume of the foreground audio is greater than a second threshold, the method can reduce volume of the background audio.

Abstract: A schematic block diagram of a decoder for decoding an encoded audio signal is shown. The decoder includes an adaptive spectrum-time converter and an overlap-add-processor. The adaptive spectrum-time converter converts successive blocks of spectral values into successive blocks of time values, e.g. via a frequency-to-time transform. Furthermore, the adaptive spectrum-time converter receives a control information and switches, in response to the control information, between transform kernels of a first group of transform kernels including one or more transform kernels having different symmetries at sides of a kernel, and a second group of transform kernels including one or more transform kernels having the same symmetries at sides of a transform kernel. Moreover, the overlap-add-processor overlaps and adds the successive blocks of time values to obtain decoded audio values, which may be a decoded audio signal.

Abstract: In a binaural hearing aid system audio signals are transmitted between an antenna facility of a left ITE hearing aid and an antenna facility of a right ITE hearing aid. Binaural beam forming is based on a natural directivity of the pinna and/or based on a head shadowing effect. Each antenna facility has an antenna arrangement with a coil core made of magnetically permeable material, and extending along a longitudinal axis, a further electric hearing aid component, which emits electromagnetic interference radiation, and an at least partially planar shield made of magnetically permeable material. The shield is arranged between the antenna arrangement and the further hearing aid component transversely to the longitudinal axis of the coil core and the shield is arranged at a distance of 50 to 150 micrometers from the coil core, preferably 75 to 100 micrometers.

Abstract: The techniques disclosed herein provide application programming interfaces (APIs) for enabling a system to select a spatialization technology. The APIs also enable a system to balance resources by allocating audio objects to a number of applications executing on a computer system. The system coordinates the audio objects between applications and each application can control the number of objects they individually generate. In some configurations, the system can also fold audio objects across different applications. Different spatialization technologies can be selected based on an analysis of contextual data and policy data. For instance, when a new headphone system is plugged in, the system may switch from Dolby Atmos to the Microsoft HoloLens HRTF spatialization technology. The system can dynamically control a number of generated audio objects and dynamically change a utilized spatialization technology based on changes to a computing environment.

Abstract: Audio signal processing apparatuses and methods are provided, such as an audio signal downmixing apparatus for processing an input audio signal into an output audio signal, wherein the input audio signal comprises a plurality of input channels recorded at a plurality of spatial positions and the output audio signal comprises a plurality of primary output channels. The audio signal downmixing apparatus comprises a downmix matrix determiner configured to determine for each frequency bin j of a plurality of frequency bins a downmix matrix DU with j being an integer in the range from 1 to N, and a processor configured to process the input audio signal using the downmix matrix DU into the output audio signal.

Abstract: The invention relates to a calibration unit (20) for an audio system (2) having a generator (4) for generating a pilot tone (6), and a detector (16a-c) for the pilot tone (6), wherein at least one parameter (P1, 2) determining the pilot tone (6) is changeable, which calibration unit comprises an input unit (22) for optionally setting the parameter (P1, 2), wherein for at least one value (W1, 2) of the parameter (P1, 2), all detectors indicate a detection of the pilot tone (6). An audio system (2) comprises said calibration unit (20). In a method for operating the calibration unit (20), the parameters (p1, 2) is regulated in such a way that all detectors (16a-c) indicate a detection of the pilot tone (6). A computer program carries out all steps of the method, when the program is performed on a computer and/or the calibration unit (20) or the audio system (2).