A hearing system configured to be located at or in the head of a user, comprises a) at least two microphones providing at least two electric input signals, b) an own voice detector, c) access to a database (O.sub.l, H.sub.l) comprising c1) relative or absolute own voice transfer function(s), and corresponding c2) absolute or relative acoustic transfer functions for a multitude of test-persons, d) a processor connectable to the at least two microphones, to the own voice detector, and to the database. The processor is configured A) to estimate an own voice relative transfer function for sound from the user's mouth to at least one of the at least two microphones, and B) to estimate personalized relative or absolute head related acoustic transfer functions from at least one spatial location other than the user's mouth to at least one of the microphones of the hearing system in dependence of the estimated own voice relative transfer function(s) and the database (O.sub.l, H.sub.l). The hearing system further comprises e) a beamformer configured to receive the at least two electric input signals, or processed versions thereof, and to determine personalized beamformer weights based on the personalized relative or absolute head related acoustic transfer functions or impulse responses. A method of determining personalized beamformer coefficients (w.sub.k) is further disclosed.

Provided is a signal processing device including: a reverberation sound signal generation unit that generates a reverberation sound signal according to a sound source position of a virtual sound source and a distance to a reference point; and a drive signal generation unit that generates a drive signal for a speaker array by a wavefront synthesis filter, in which the drive signal generation unit generates the drive signal on the basis of a signal obtained by performing wavefront synthesis filtering processing on a signal obtained by convolving the reverberation sound signal with a signal of the virtual sound source and/or a signal obtained by performing wavefront synthesis filtering processing on the reverberation sound signal to make the reverberation sound signal into a virtual sound source.

A stochastic channel state information transmission/reception method and apparatus is provided for use in a multiuser radio communication system. The signal method of transmitting and receiving signals in a terminal in a mobile communication system according to the present disclosure includes receiving a reference signal transmitted by a base station, estimating channel information based on the reference signal, predicting a channel estimation error based on the channel information, generating feedback information based on the channel estimation error, and transmitting feedback information to the base station.

A communication system provides audio content to one or more client devices capable of playing spatialized audio content. For example, the communication system receives audio content from a client device and transmits the audio content to other client devices to be played for users. The communication system dynamically modifies audio content transmitted to different client devices based on a payload including audio parameters (e.g., local area acoustic properties, an audiogram for a user, a head related transfer function for a user, etc.) received from a client device.

A method is disclosed, the method comprising obtaining at least one first information indicative of audio data gathered by at least one first microphone, and at least one second information indicative of audio data gathered by at least one second microphone; determining a differential information indicative of one or more differences between at least two pieces of information, wherein the differential information is determined based, at least in part, on the at least one first information and the at least one second information; and compensating of an impact onto the audio data, wherein audio data of the first information and/or the second information is compensated based, at least in part, on the determined differential information. Further, an apparatus, and a system are disclosed.

Embodiments include a microphone assembly comprising an array microphone and a housing configured to support the array microphone and sized and shaped to be mountable in a drop ceiling in place of at least one of a plurality of ceiling tiles included in the drop ceiling. A front face of the housing includes a sound-permeable screen having a size and shape that is substantially similar to the at least one of the plurality of ceiling tiles. Embodiments also include an array microphone system comprising a plurality of microphones arranged, on a substrate, in a number of concentric, nested rings of varying sizes around a central point of the substrate. Each ring comprises a subset of the plurality of microphones positioned at predetermined intervals along a circumference of the ring.

Embodiments of the disclosure provide methods and apparatuses processing audio data. The method can include: acquiring audio data by an audio capturing device, determining feature information of an enclosure in which the audio capturing device is located, and reverberating the feature information into the audio data.

A method of generating a signal for driving a first linear array of sound sources. The first linear array of sound sources comprises a primary sound source and one or more secondary sound sources. The method comprises the steps of receiving an audio signal for a first channel of an audio system, deriving, from the audio signal, a first signal and a second signal, applying a low-pass filter to the second signal to generate a second drive signal for driving the one or more secondary sound sources, and applying a corresponding high-frequency shelving filter to the first signal to generate a first drive signal for driving the primary sound source. A computer program product and an audio system for generating a levelled sound field is also provided.

The description relates to rendering directional sound. One implementation includes receiving directional impulse responses corresponding to a scene. The directional impulse responses can correspond to multiple sound source locations and a listener location in the scene. The implementation can also include encoding the directional impulse responses to obtain encoded departure direction parameters for individual sound source locations. The implementation can also include outputting the encoded departure direction parameters, the encoded departure direction parameters providing sound departure directions from the individual sound source locations for rendering of sound.

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.