Abstract: The disclosure relates to an audio processing apparatus for localizing an audio source. The audio processing apparatus comprises a plurality of audio sensors, including a primary audio sensor and at least two secondary audio sensors, configured to detect an audio signal from a target audio source, wherein the primary audio sensor defines at least two pairs of audio sensors with the at least two secondary audio sensors; and processing circuitry configured to: determine for each pair of audio sensors a first set of likelihoods of spatial directions of the target audio source using a first localization scheme; determine a second set of likelihoods of spatial directions of the target audio source using a second localization scheme; and determine a third set of likelihoods of spatial directions of the target audio source on the basis of the first sets of likelihoods and the second set of likelihoods.

Abstract: An apparatus determines a spatial position of an audio source in multi moving audio sources scenarios. The apparatus receives audio signal versions as local sound waves. The apparatus determines first and second probabilities for a direction of arrival of the audio signal version based on the audio signal versions received within a first time interval; determines third and fourth probabilities for the direction of arrival of the audio signal version based on the audio signal versions received within a second time interval; determines a first probability difference between the first and third probabilities; determines a second probability difference between the second and fourth probabilities; combines the third probability and the first probability difference to obtain an updated third probability; combines the fourth probability with the second probability difference to obtain an updated fourth probability; and determines the spatial position based on the updated third and fourth probabilities.

Abstract: An apparatus determines a spatial position of an audio source in multi moving audio sources scenarios. The apparatus receives audio signal versions as local sound waves. The apparatus determines first and second probabilities for a direction of arrival of the audio signal version based on the audio signal versions received within a first time interval; determines third and fourth probabilities for the direction of arrival of the audio signal version based on the audio signal versions received within a second time interval; determines a first probability difference between the first and third probabilities; determines a second probability difference between the second and fourth probabilities; combines the third probability and the first probability difference to obtain an updated third probability; combines the fourth probability with the second probability difference to obtain an updated fourth probability; and determines the spatial position based on the updated third and fourth probabilities.

Abstract: The disclosure relates to an audio processing apparatus for localizing an audio source. The audio processing apparatus comprises a plurality of audio sensors, including a primary audio sensor and at least two secondary audio sensors, configured to detect an audio signal from a target audio source, wherein the primary audio sensor defines at least two pairs of audio sensors with the at least two secondary audio sensors; and processing circuitry configured to: determine for each pair of audio sensors a first set of likelihoods of spatial directions of the target audio source using a first localization scheme; determine a second set of likelihoods of spatial directions of the target audio source using a second localization scheme; and determine a third set of likelihoods of spatial directions of the target audio source on the basis of the first sets of likelihoods and the second set of likelihoods.

Abstract: A rendering system including a plurality of loudspeakers, at least one microphone and a signal processing unit. The signal processing unit is configured to determine at least some components of a loudspeaker-enclosure-microphone transfer function matrix estimate describing acoustic paths between the plurality of loudspeakers and the at least one microphone using a rendering filters transfer function matrix using which a number of virtual sources is reproduced with the plurality of loudspeakers.

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: A rendering system including a plurality of loudspeakers, at least one microphone and a signal processing unit. The signal processing unit is configured to determine at least some components of a loudspeaker-enclosure-microphone transfer function matrix estimate describing acoustic paths between the plurality of loudspeakers and the at least one microphone using a rendering filters transfer function matrix using which a number of virtual sources is reproduced with the plurality of loudspeakers.

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: A device for generating a multitude of loudspeaker signals based on a virtual source object which has a source signal and a meta information determining a position or type of the virtual source object. The device has a modifier configured to time-varyingly modify the meta information. In addition, the device has a renderer configured to transfer the virtual source object and the modified meta information to form a multitude of loudspeaker signals.

Abstract: A high-performance method evaluates a useful signal of an audio device, and in particular of an audio apparatus, for example for reducing interference. Accordingly, in the method at least two microphone signals are each obtained from a sound signal and a reference signal is obtained from the microphone signals, a portion of the microphone signals from a predetermined direction being blocked. The microphone signals are filtered by a filter such that an evaluation signal is obtained. To that end, a coherence value is determined from portions of the reference signal and a power density value is determined from the coherence value. The filter is parameterized on the basis of the power density value.

Abstract: A device for generating a multitude of loudspeaker signals based on a virtual source object which has a source signal and a meta information determining a position or type of the virtual source object. The device has a modifier configured to time-varyingly modify the meta information. In addition, the device has a renderer configured to transfer the virtual source object and the modified meta information to form a multitude of loudspeaker signals.

Abstract: An audio signal processing system and an echo signal removing method thereof are provided. The audio signal processing system includes a speaker that is configured to output an audio signal; a microphone that is configured to receive the audio signal output by the speaker including an echo signal generated by the audio signal; an echo signal delay unit that is configured to delay the echo signal for a bulk delay time, and output the echo signal that is delayed; and an echo signal removing unit that is configured to remove the echo signal that is delayed and output by the echo signal delay unit from the audio signal received by the microphone, wherein the echo signal delay unit includes a bulk delay measuring unit that is configured to measure a bulk delay by analyzing impulse response characteristics of an echo path.

Abstract: An apparatus for listening room equalization is provided. A system identification adaptation unit is configured to adapt a first loudspeaker-enclosure-microphone system identification to obtain a second loudspeaker-enclosure-microphone system identification. A filter adaptation unit is configured to adapt a filter based on the second loudspeaker-enclosure-microphone system identification a predetermined loudspeaker-enclosure-microphone system identification. A filter includes a plurality of subfilters each of which receive one or more of the transformed loudspeaker signals. Each of the subfilters is adapted to generate one of a plurality of filtered loudspeaker signals based on the one or more received loudspeaker signals. At least one of the subfilters is arranged to couple the at least two received loudspeaker signals to generate one of the plurality of the filtered loudspeaker signals.

Abstract: An apparatus for providing a current loudspeaker-enclosure-microphone system description of a loudspeaker-enclosure-microphone system is provided. The apparatus has a first transformation unit for generating a plurality of wave-domain loudspeaker audio signals. Moreover, the apparatus has a second transformation unit for generating a plurality of wave-domain microphone audio signals. Furthermore, the apparatus has a system description generator for generating the current loudspeaker-enclosure-microphone system description based on the plurality of wave-domain loudspeaker audio signals, based on the plurality of wave-domain microphone audio signals, and based on a plurality of coupling values, wherein the system description generator is configured to determine each coupling value assigned to a wave-domain pair of a plurality of wave-domain pairs by determining a relation indicator indicating a relation between a loudspeaker-signal-transformation value and a microphone-signal-transformation value.

Abstract: A method operates a hearing aid system and the hearing aid system has at least two hearing aid devices, between which a signal path is provided, and with at least one signal processing unit, which is provided to process audio signals. In the method the signal processing apparatus filters first audio signals with a filter predetermined for a defined spatial direction, from which a useful signal arrives, so that second audio signals are generated, in which the components of the useful signal in the second audio signals are equalized to a greater degree than in the first audio signals. The second audio signals are then filtered with an adaptive filter, so that third audio signals are generated, in which the components of the useful signal are equalized to an even greater degree than in the second audio signals.

Abstract: A high-performance method evaluates a useful signal of an audio device, and in particular of an audio apparatus, for example for reducing interference. Accordingly, in the method at least two microphone signals are each obtained from a sound signal and a reference signal is obtained from the microphone signals, a portion of the microphone signals from a predetermined direction being blocked. The microphone signals are filtered by a filter such that an evaluation signal is obtained. To that end, a coherence value is determined from portions of the reference signal and a power density value is determined from the coherence value. The filter is parameterized on the basis of the power density value.

Abstract: An apparatus for providing a current loudspeaker-enclosure-microphone system description of a loudspeaker-enclosure-microphone system is provided. The apparatus has a first transformation unit for generating a plurality of wave-domain loudspeaker audio signals. Moreover, the apparatus has a second transformation unit for generating a plurality of wave-domain microphone audio signals. Furthermore, the apparatus has a system description generator for generating the current loudspeaker-enclosure-microphone system description based on the plurality of wave-domain loudspeaker audio signals, based on the plurality of wave-domain microphone audio signals, and based on a plurality of coupling values, wherein the system description generator is configured to determine each coupling value assigned to a wave-domain pair of a plurality of wave-domain pairs by determining a relation indicator indicating a relation between a loudspeaker-signal-transformation value and a microphone-signal-transformation value.

Abstract: An apparatus for providing a current loudspeaker-enclosure-microphone system description of a loudspeaker-enclosure-microphone system is provided. The apparatus has a first transformation unit for generating a plurality of wave-domain loudspeaker audio signals. Moreover, the apparatus has a second transformation unit for generating a plurality of wave-domain microphone audio signals. Furthermore, the apparatus has a system description generator for generating the current loudspeaker-enclosure-microphone system description based on the plurality of wave-domain loudspeaker audio signals, based on the plurality of wave-domain microphone audio signals, and based on a plurality of coupling values, wherein the system description generator is configured to determine each coupling value assigned to a wave-domain pair of a plurality of wave-domain pairs by determining a relation indicator indicating a relation between a loudspeaker-signal-transformation value and a microphone-signal-transformation value.