Abstract: A sound processing node is provided for an arrangement of sound processing nodes configured to receive a plurality of sound signals. The sound processing node comprises a processor configured to generate an output signal based on the plurality of sound signals weighted by a plurality of beamforming weights. The processor is configured to adaptively determine the plurality of beamforming weights on the basis of an adaptive linearly constrained minimum variance beamformer using a transformed version of a least mean squares formulation of a constrained gradient descent approach. The transformed version of the least mean squares formulation of the constrained gradient descent approach is based on a transformation of the least mean squares formulation of the constrained gradient descent approach to the dual domain.

Abstract: The invention relates to a sound processing node for an arrangement of sound processing nodes, the sound processing nodes being configured to receive a plurality of sound signals, wherein the sound processing node comprises a processor configured to generate an output signal on the basis of the plurality of sound signals weighted by a plurality of beamforming weights, wherein the processor is configured to adaptively determine the plurality of beamforming weights on the basis of an adaptive linearly constrained minimum variance beamformer using a transformed version of a least mean squares formulation of a constrained gradient descent approach, wherein the transformed version of the least mean squares formulation of the constrained gradient descent approach is based on a transformation of the least mean squares formulation of the constrained gradient descent approach to the dual domain.

Abstract: The application relates to a hearing device adapted for being located at or in a first ear of a user, or to be fully or partially implanted in the head at a first ear of a user, the hearing device comprising a first input transducer for converting a first input sound signal from a sound field around the user at a first location, the first location being a location of the first input transducer, to a first electric input signal, the sound field comprising a mixture of a target sound from a target sound source and possible acoustic noise; a transceiver unit configured to receive a first quantized electric input signal via a communication link, the first quantized electric input signal being representative of the sound field around the user at a second location, the first quantized electric input signal comprising quantization noise due to a specific quantization scheme; a beamformer filtering unit adapted to receive said first electric input signal and said quantized electric input signal and to determine be

Abstract: A sound processing node for an arrangement of sound processing nodes is disclosed. The sound processing nodes being configured to receive a plurality of sound signals, wherein the sound processing node comprises a processor configured to determine a beamforming signal on the basis of the plurality of sound signals weighted by a plurality of weights, wherein the processor is configured to determine the plurality of weights using a transformed version of a linearly constrained minimum variance approach, the transformed version of the linearly constrained minimum variance approach being obtained by applying a convex relaxation to the linearly constrained minimum variance approach.

Abstract: A sound processing node for an arrangement of sound processing nodes is disclosed. The sound processing nodes being configured to receive a plurality of sound signals, wherein the sound processing node comprises a processor configured to determine a beamforming signal on the basis of the plurality of sound signals weighted by a plurality of weights, wherein the processor is configured to determine the plurality of weights using a transformed version of a linearly constrained minimum variance approach, the transformed version of the linearly constrained minimum variance approach being obtained by applying a convex relaxation to the linearly constrained minimum variance approach.

Abstract: The application relates to a hearing device adapted for being located at or in a first ear of a user, or to be fully or partially implanted in the head at a first ear of a user, the hearing device comprising a first input transducer for converting a first input sound signal from a sound field around the user at a first location, the first location being a location of the first input transducer, to a first electric input signal, the sound field comprising a mixture of a target sound from a target sound source and possible acoustic noise; a transceiver unit configured to receive a first quantized electric input signal via a communication link, the first quantized electric input signal being representative of the sound field around the user at a second location, the first quantized electric input signal comprising quantization noise due to a specific quantization scheme; a beamformer filtering unit adapted to receive said first electric input signal and said quantized electric input signal and to determine be

Abstract: Public address systems or other systems for emitting audio signals, like music, speech or announcements, in different locations like supermarkets, schools, universities, and auditoriums are widely known. In one embodiment, invention proposes a system for emitting an audio signal in an environment. The system includes an audio source for providing the audio signal and at least one loudspeaker for emitting the audio signal. The system also includes at least one microphone for receiving an acoustic signal from the environment. The acoustic signal is based on the audio signal and may comprise disturbing components. The system also includes an analyzing module for analyzing the acoustic signal and for providing an intelligibility measure from an objective intelligibility measure method. The intelligibility measure is used as a feedback signal.

Abstract: Methods and systems are provided for implementing a distributed algorithm for beam-forming (e.g., MVDR beam-forming) using a message-passing algorithm. The message-passing algorithm provides for computations to be performed in a distributed manner across a network, rather than in a centralized processing center or “fusion center”. The message-passing algorithm may also function for any network topology, and may continue operations when various changes are made in the network (e.g., nodes appearing, nodes disappearing, etc.). Additionally, the message-passing algorithm may minimize the transmission power per iteration and, depending on the particular network, also may minimize the transmission power required for communication between network nodes.

Abstract: Provided are methods and systems for calibrating a distributed sensor (e.g., microphone) array using time-of-flight (TOF) measurements for a plurality of spatially distributed acoustic events at the sensors. The calibration includes localization and gain equalization of the sensors. Accurate measurements of TOFs are obtained from spatially distributed acoustic events using a controlled signal emitted at known intervals by a moving acoustic source. A portable user device capable of playing out audio is used to produce a plurality of acoustic events (e.g., sound clicks) at known intervals of time and at different, but arbitrary locations based on the device being moved around in space by a user while producing the acoustic events. As such, the times of the acoustic event generation are known, and are spatially diverse. The calibration signals emitted by the acoustic source are designed to provide robustness to noise and reverberation.

Abstract: Provided are methods and systems for finding the location of sensors (e.g., microphones) with unknown internal delays based on a set of events (e.g., acoustic events) with unknown event time. A localization algorithm may iteratively run to compute the acoustic event times, the observation delays, and the relative locations of the events and the sensors.

Abstract: Methods and systems are provided for implementing a distributed algorithm for beam-forming (e.g., MVDR beam-forming) using a message-passing algorithm. The message-passing algorithm provides for computations to be performed in a distributed manner across a network, rather than in a centralized processing center or “fusion center”. The message-passing algorithm may also function for any network topology, and may continue operations when various changes are made in the network (e.g., nodes appearing, nodes disappearing, etc.). Additionally, the message-passing algorithm may minimize the transmission power per iteration and, depending on the particular network, also may minimize the transmission power required for communication between network nodes.

Abstract: Provided are methods and systems for calibrating a distributed sensor (e.g., microphone) array using time-of-flight (TOF) measurements for a plurality of spatially distributed acoustic events at the sensors. The calibration includes localization and gain equalization of the sensors. Accurate measurements of TOFs are obtained from spatially distributed acoustic events using a controlled signal emitted at known intervals by a moving acoustic source. A portable user device capable of playing out audio is used to produce a plurality of acoustic events (e.g., sound clicks) at known intervals of time and at different, but arbitrary locations based on the device being moved around in space by a user while producing the acoustic events. As such, the times of the acoustic event generation are known, and are spatially diverse. The calibration signals emitted by the acoustic source are designed to provide robustness to noise and reverberation.

Abstract: The application relates to a method of providing a speech intelligibility predictor value for estimating an average listener's ability to understand of a target speech signal when said target speech signal is subject to a processing algorithm and/or is received in a noisy environment. The application further relates to a method of improving a listener's understanding of a target speech signal in a noisy environment and to corresponding device units. The object of the present application is to provide an alternative objective intelligibility measure, e.g. a measure that is suitable for use in a time-frequency environment. The invention may e.g. be used in audio processing systems, e.g. listening systems, e.g. hearing aid systems.

Abstract: Provided are methods and systems for finding the location of sensors (e.g., microphones) with unknown internal delays based on a set of events (e.g., acoustic events) with unknown event time. A localization algorithm may iteratively run to compute the acoustic event times, the observation delays, and the relative locations of the events and the sensors.

Abstract: Public address systems or other systems for emitting audio signals, like music, speech or announcements, in different locations like supermarkets, schools, universities, auditoriums are widely known. These systems usually comprise an audio source, for example a microphone or a recorder, and a plurality of loudspeakers, which are locally distributed in the locations, for emitting the audio signal from the audio source.

Abstract: A method estimates noise power spectral density (PSD) in an input sound signal to generate an output for noise reduction of the input sound signal. The method includes storing frames of a digitized version of the input signal, each frame having a predefined number N2 of samples corresponding to a frame length in time of L2=N2/sampling frequency. It further includes performing a time to frequency transformation, deriving a periodogram comprising an energy content |Y|2 from the corresponding spectrum Y, applying a gain function G(k,m)=f(?s2(km),?w2l (k,m?1), |Y(k,m)|2), to estimate a noise energy level |?|2 in each frequency sample, where ?s2 is the speech PSD and ?w2 the noise PSD. It further includes dividing spectra into a number of sub-bands, and providing a first estimate |{circumflex over (N)}|2 of the noise PSD level in a sub-band and a second, improved estimate |{circumflex over (N)}|2 of the noise PSD level in a subband by applying a bias compensation factor B to the first estimate.

Abstract: The application relates to a method of providing a speech intelligibility predictor value for estimating an average listener's ability to understand of a target speech signal when said target speech signal is subject to a processing algorithm and/or is received in a noisy environment. The application further relates to a method of improving a listener's understanding of a target speech signal in a noisy environment and to corresponding device units. The object of the present application is to provide an alternative objective intelligibility measure, e.g. a measure that is suitable for use in a time-frequency environment. The invention may e.g. be used in audio processing systems, e.g. listening systems, e.g. hearing aid systems.

Abstract: An audio encoder in which two or more preferably different encoders cooperate to generate a joint encoded audio signal. Encoding parameters of the two or more encoders are optimized in response to a measure of distortion of the joint encoded audio signal in accordance with a predetermined criterion. The distortion. measure is preferably a perceptual distortion measure. In one encoder embodiment comprising a sinusoidal and a waveform encoder, a constant total bit rate for each audio frame is distributed between the two encoders so as to minimize perceptual distortion for both the first and the second encoder. Other embodiments consider a set of encoding parameters that is larger than only those that minimize the perceptual distortion of the first encoder. In some embodiments, perceptual distortion may be minimized by optimizing encoding via optimizing entire encoding templates, i.e. a complex set of encoding parameters, for the separate encoders.

Abstract: The invention relates to a method of estimating noise power spectral density PSD in an input sound signal comprising a noise signal part and a target signal part. The invention further relates to a system to its use. The object of the present invention is to provide a scheme for estimating the noise PSD in an acoustic signal consisting of a target signal contaminated by acoustic noise. The problem is solved by a method comprising the steps of d) providing a digitized electrical input signal to a control path and performing; d1) storing a number of time frames of the input signal each comprising a predefined number N2 of digital time samples xn (n=1, 2, . . .

Abstract: An audio encoder in which two or more preferably different encoders cooperate to generate a joint encoded audio signal. Encoding parameters of the two or more encoders are optimized in response to a measure of distortion of the joint encoded audio signal in accordance with a predetermined criterion. The distortion. measure is preferably a perceptual distortion measure. In one encoder embodiment comprising a sinusoidal and a waveform encoder, a constant total bit rate for each audio frame is distributed between the two encoders so as to minimize perceptual distortion for both the first and the second encoder. Other embodiments consider a set of encoding parameters that is larger than only those that minimize the perceptual distortion of the first encoder. In some embodiments, perceptual distortion may be minimized by optimizing encoding via optimizing entire encoding templates, i.e. a complex set of encoding parameters, for the separate encoders.