Abstract: A device and a method are provided for producing a sound field are disclosed. The device comprises a plurality of loudspeakers arranged at a plurality of locations within a plane and a processing circuitry configured to drive the plurality of loudspeakers. A first subset of the plurality of loudspeakers defines a first rhombus within the plane and a second subset of the plurality of loudspeakers defines a second rhombus within the plane. The first rhombus is oriented substantially perpendicular to the second rhombus. Further subsets of the plurality of loudspeakers may define further rhombic sub-arrays. The audio device may be implemented as a soundbar or a sound panel, which provides a richer sound experience.

Abstract: A device for estimating Direction of Arrival (DOA) of sound from Q?1 sound sources is provided. The device is configured to obtain a phase difference matrix, which includes measured phase difference values, each of the measured phase difference values being a measured value of a phase difference between two microphone units for a frequency bin in a range of frequencies of the sound. The device is further configured to generate a replicated phase difference matrix by replicating the measured phase difference values to other potential sinusoidal periods, calculate a DOA value for each phase difference value in the replicated phase difference matrix, and determine, as Q DOA results, the Q most prominent peak values in a histogram generated based on the calculated DOA values.

Abstract: The present disclosure relates to an audio device (900) for providing an improved three-dimensional sound experience by means of the generated soundfield. To achieve this, the audio device (900) comprises a housing (901), which has an elliptical torus shape and a plurality of loudspeakers (903a-903h), and a processing circuitry (1310). The processing circuitry is configured to process a plurality of input signals (L, R, UL, UR) in a manner, which enables the plurality of loudspeakers (903a-903h) to form at least a first (DH1, DH3) and second (DH2) horizontal dipoles for crosstalk cancellation within at least two different frequency ranges (HF, MF), and to form at least a first vertical dipole (DV1, DV3) for sound elevation (1204a, 1204b) of the soundfield. Hereby, the desired frequency ranges (HF, MF) may be adjusted using an appropriated distance of the plurality of loudspeakers (903a-903h).

Abstract: An apparatus configured to obtain a plurality of phase differences is provided. Each phase difference represents a difference in phase between a spectral component in a first sound signal and a corresponding spectral component in a second sound signal, to estimate an aliasing frequency. Phase differences in frequency bands above the aliasing frequency are expected to be wrapped, and to unwrap the plurality of phase differences in dependence on the estimated aliasing frequency. A phase difference obtained from spectral components that are comprised in one frequency band is unwrapped independently of any phase difference obtained from spectral components that are comprised in another frequency band.

Abstract: A method for acoustic scene playback is described, which comprises: providing recording data comprising microphone signals of microphone setups positioned within an acoustic scene and microphone metadata of the microphone setups, each of the microphone setups has a recording spot which is a center position of the respective microphone setup; specifying a virtual listening position within the acoustic scene; assigning each microphone setup Virtual Loudspeaker Objects, VLOs, wherein each VLO is an abstract sound output object within a virtual free field; generating an encoded data stream based on the recording data, the virtual listening position and VLO parameters of the VLOs assigned to the microphone setups; and decoding the encoded data stream based on a playback setup, thereby generating a decoded data stream; and feeding the decoded data stream to a rendering device, thereby driving the rendering device to reproduce sound of the acoustic scene at the virtual listening position.

Abstract: A signal generator has a filter bank that provides weighted versions of audio signals to speakers. The weights were derived by identifying a first constraint that limits a weight that can be applied to an audio signal to be provided to a first speaker. A characteristic of a second speaker that affects how a user will perceive audio signals output by that speaker relative to audio signals output by the first speaker was also determined. A second constraint was determined based on the determined characteristic and the first constraint. The weights were then determined so as to minimize a difference between an actual balance of each signal that is expected to be heard by a user and a target balance. The signal generator can achieve sweet spot correction and sound stage widening simultaneously. It also achieves a balanced sound stage, particularly when the speakers are asymmetric.

Abstract: A sound signal processing apparatus including a plurality of microphones, where each microphone is configured to receive the sound signal from the target source, a processor configured to estimate a first power measure on the basis of the sound signal from the target source received by a first microphone of the microphones and a second power measure on the basis of the sound signal from the target source received by at least a second microphone of the microphones, which is located more distant from the target source than the first microphone, and the processor is further configured to determine a gain factor on the basis of a ratio between the second power measure and the first power measure, and an amplifier configured to apply the gain factor to the sound signal from the target source received by the first microphone.

Abstract: A device for estimating Direction of Arrival (DOA) of sound from Q?1 sound sources is provided. The device is configured to obtain a phase difference matrix, which includes measured phase difference values, each of the measured phase difference values being a measured value of a phase difference between two microphone units for a frequency bin in a range of frequencies of the sound. The device is further configured to generate a replicated phase difference matrix by replicating the measured phase difference values to other potential sinusoidal periods, calculate a DOA value for each phase difference value in the replicated phase difference matrix, and determine, as Q DOA results, the Q most prominent peak values in a histogram generated based on the calculated DOA values.

Abstract: An audio signal processing apparatus, comprising a memory configured to store a set of pairs of predefined left ear and right ear transfer functions, which are predefined for a plurality of reference positions relative to the listener, a processor configured to determine a pair of left ear and right ear transfer functions on the basis of the set of predefined pairs of predefined left ear and right ear transfer functions for the azimuth angle and the elevation angle of the virtual target position and an adjustment filter configured to filter the input audio signal on the basis of the determined pair of left ear and right ear transfer functions and an adjustment function configured to adjust a delay between the left ear transfer function and the right ear transfer function to obtain a left ear output audio signal and a right ear output audio signal.

Abstract: Example audio signal processing apparatuses and methods for processing an input audio signal into an output audio signal are disclosed. An example audio signal processing apparatus comprises a decomposer configured to decompose the input audio signal into a direct audio signal and a diffuse audio signal, a modifier configured to modify the direct audio signal in order to obtain a modified direct audio signal, wherein the modifier comprises a bandwidth extender configured to extend an upper cutoff frequency of a frequency range of the direct audio signal, and a combiner configured to combine the modified direct audio signal with the diffuse audio signal to obtain the output audio signal.

Abstract: An audio signal processing apparatus, comprising a memory configured to store a set of pairs of predefined left ear and right ear transfer functions, which are predefined for a plurality of reference positions relative to the listener, a processor configured to determine a pair of left ear and right ear transfer functions on the basis of the set of predefined pairs of predefined left ear and right ear transfer functions for the azimuth angle and the elevation angle of the virtual target position and an adjustment filter configured to filter the input audio signal on the basis of the determined pair of left ear and right ear transfer functions and an adjustment function configured to adjust a delay between the left ear transfer function and the right ear transfer function to obtain a left ear output audio signal and a right ear output audio signal.

Abstract: A signal generator has a filter bank that provides weighted versions of audio signals to speakers. The weights were derived by identifying a first constraint that limits a weight that can be applied to an audio signal to be provided to a first speaker. A characteristic of a second speaker that affects how a user will perceive audio signals output by that speaker relative to audio signals output by the first speaker was also determined. A second constraint was determined based on the determined characteristic and the first constraint. The weights were then determined so as to minimize a difference between an actual balance of each signal that is expected to be heard by a user and a target balance. The signal generator can achieve sweet spot correction and sound stage widening simultaneously. It also achieves a balanced sound stage, particularly when the speakers are asymmetric.

Abstract: An input audio signal is separated into input audio signal components (X(k,b)). A set of two or more band branches provides output audio signal components (Y(k,b)). The set of band branches comprises one or more compressor branches Each compressor branch compresses a respective input audio signal component (X(k,b)) into a respective output audio signal component (Y(k,b)). A summed audio signal (y(t)) is generated by summing the output audio signal components (Y(k,b)). A residual audio signal (v(t)) is a difference between the input audio signal and the summed audio signal(y(t)) A virtual bass signal (w(t)) comprises one or more harmonics of the residual audio signal (v(t)). An output audio signal is generated by summing the summed audio signal (y(t)) and the virtual bass signal (w(t)).

Abstract: An apparatus is configured to obtain a plurality of phase differences, wherein each phase difference represents a difference in phase between a spectral component in a first sound signal and a corresponding spectral component in a second sound signal, to estimate an aliasing frequency, wherein phase differences in frequency bands above the aliasing frequency are expected to be wrapped, and to unwrap the plurality of phase differences in dependence on the estimated aliasing frequency, wherein a phase difference obtained from spectral components that are comprised in one frequency band is unwrapped independently of any phase difference obtained from spectral components that are comprised in another frequency band.

Abstract: The invention relates to an audio signal processing apparatus for processing an input earpiece audio signal upon the basis of a microphone audio signal, the audio signal processing apparatus comprising a voice activity detector being configured to determine a voice activity indicator signal upon the basis of the input earpiece audio signal, a noise magnitude determiner being configured to determine a microphone noise magnitude indicator signal upon the basis of the microphone audio signal, a gain factor determiner being configured to determine a gain factor signal upon the basis of the voice activity indicator signal and the microphone noise magnitude indicator signal, and a weighter being configured to weight the input earpiece audio signal by the gain factor signal to obtain an output earpiece audio signal.

Abstract: A method for acoustic scene playback is described, which comprises: providing recording data comprising microphone signals of microphone setups positioned within an acoustic scene and microphone metadata of the microphone setups, each of the microphone setups has a recording spot which is a center position of the respective microphone setup; specifying a virtual listening position within the acoustic scene; assigning each microphone setup Virtual Loudspeaker Objects, VLOs, wherein each VLO is an abstract sound output object within a virtual free field; generating an encoded data stream based on the recording data, the virtual listening position and VLO parameters of the VLOs assigned to the microphone setups; and decoding the encoded data stream based on a playback setup, thereby generating a decoded data stream; and feeding the decoded data stream to a rendering device, thereby driving the rendering device to reproduce sound of the acoustic scene at the virtual listening position.

Abstract: A wave field synthesis apparatus for driving an array of loudspeakers with drive signals, the apparatus includes a sound field synthesizer for generating sound field drive signals for causing the array of loudspeakers to generate one or more sound fields at one or more audio zones, a binaural renderer for generating binaural drive signals for causing the array of loud-speakers to generate specified sound pressures at at least two positions, wherein the at least two positions are determined based on a detected position and/or orientation of a listener, and a decision unit for deciding whether to generate the drive signals using the sound field synthesizer or using the binaural renderer.

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: An input audio signal (101) is separated into input audio signal components (X(k,b)). A set of two or more band branches (105e) provides output audio signal components (Y(k,b)). The set of band branches (105e) comprises one or more compressor branches Each compressor branch compresses a respective input audio signal component (X(k,b)) into a respective output audio signal component (Y(k,b)). A summed audio signal (y(t)) is generated by summing the output audio signal components (Y(k,b)). A residual audio signal (v(t)) is a difference between the input audio signal (101) and the summed audio signal (y(t)) A virtual bass signal (w(t)) comprises one or more harmonics of the residual audio signal (v(t)). An output audio signal (103) is generated by summing the summed audio signal (y(t)) and the virtual bass signal (w(t)).

Abstract: A signal processing apparatus for enhancing a voice component within a multi-channel audio signal comprising a left channel audio signal, a center channel audio signal, and a right channel audio signal, the signal processing apparatus comprising a filter and a combiner; wherein the filter is configured to determine an overall magnitude of the multi-channel audio signal over frequency based on the multi-channel audio signal, to obtain a gain function based on a ratio between a magnitude of the center channel audio signal and the overall magnitude of the multi-channel audio signal, and to weight the left channel audio signal, the center channel audio signal, and the right channel audio signal by the gain function; and wherein the combiner is configured to combine individually the left channel audio signal, the center channel audio signal, and the right channel audio signal with the weighted right channel audio signal.