Abstract: Provided are methods and systems for enhancing speech when corrupted by transient noise (e.g., keyboard typing noise). The methods and systems utilize a reference microphone input signal for the transient noise in a signal restoration process used for the voice part of the signal. A robust Bayesian statistical model is used to regress the voice microphone on the reference microphone, which allows for direct inference about the desired voice signal while marginalizing the unwanted power spectral values of the voice and transient noise. Also provided is a straightforward and efficient Expectation-maximization (EM) procedure for fast enhancement of the corrupted signal. The methods and systems are designed to operate easily in real-time on standard hardware, and have very low latency so that there is no irritating delay in speaker response.

Abstract: A device provides a first data signal that indicates an activity of at least one muscle of a patient that is relevant for an inspiratory breathing effort and a second data signal that indicates an activity of at least one muscle of the patient that is relevant for an expiratory breathing effort. The data signals are generated from electromyography (EMG) signals detected by surface electromyography sensors. A computer is configured to determine breathing phase information on the basis of a breathing signal and to check at least one of the electromyography signals or at least one of the separated signals for detectability of a heart signal component and further to assign the signals to an inspiratory breathing activity as well as to an expiratory breathing activity of the patient as a function of the breathing phase information.

Abstract: The invention refers to a system for handling digital content including an input interface, a calculator, and an output interface. The input interface receives digital content and includes a plurality of input channels. At least one input channel receives digital content from a sensor or a group of sensors belonging to a recording session. The calculator provides output digital content by adapting received digital content to a reproduction session in which the output digital content is to be reproduced. The output interface outputs the output digital content and includes a plurality of output channels, wherein at least one output channel outputs the output digital content to an actuator or a group of actuators belonging to the reproduction session. Further, the input interface, the calculator, and the output interface are connected with each other via a network.

Abstract: Provided are methods and systems for enhancing speech when corrupted by transient noise (e.g., keyboard typing noise). The methods and systems utilize a reference microphone input signal for the transient noise in a signal restoration process used for the voice part of the signal. A robust Bayesian statistical model is used to regress the voice microphone on the reference microphone, which allows for direct inference about the desired voice signal while marginalizing the unwanted power spectral values of the voice and transient noise. Also provided is a straightforward and efficient Expectation-maximization (EM) procedure far fast enhancement of the corrupted signal. The methods and systems are designed to operate easily in real-time on standard hardware, and have very low latency so that there is no irritating delay in speaker response.

Abstract: Provided are methods and systems for enhancing speech when corrupted by transient noise (e.g., keyboard typing noise). The methods and systems utilize a reference microphone input signal for the transient noise in a signal restoration process used for the voice part of the signal. A robust Bayesian statistical model is used to regress the voice microphone on the reference microphone, which allows for direct inference about the desired voice signal while marginalizing the unwanted power spectral values of the voice and transient noise. Also provided is a straightforward and efficient Expectation-maximization (EM) procedure for fast enhancement of the corrupted signal. The methods and systems are designed to operate easily in real-time on standard hardware, and have very low latency so that there is no irritating delay in speaker response.

Abstract: A signal processing apparatus and method are provided for separating a plurality of mixture signals from a MIMO system to iteratively obtain a plurality of output signals. The plurality of mixture signals are a response of the MIMO system to a plurality of source signals. The signal processing apparatus comprises a plurality of blind source separators including a first blind source separator based on a first blind source separation technique or algorithm and a second blind source separator based on a second blind source separation technique or algorithm, wherein the first blind source separator is configured to compute a first plurality of preliminary output signals on the basis of a first set of coefficients describing the MIMO system and wherein the second blind source separator is configured to compute a second plurality of preliminary output signals on the basis of a second set of coefficients describing the MIMO system.

Abstract: The invention refers to a system for handling digital content including an input interface, a calculator, and an output interface. The input interface receives digital content and includes a plurality of input channels. At least one input channel receives digital content from a sensor or a group of sensors belonging to a recording session. The calculator provides output digital content by adapting received digital content to a reproduction session in which the output digital content is to be reproduced. The output interface outputs the output digital content and includes a plurality of output channels, wherein at least one output channel outputs the output digital content to an actuator or a group of actuators belonging to the reproduction session. Further, the input interface, the calculator, and the output interface are connected with each other via a network.

Abstract: A device provides a first data signal that indicates an activity of at least one muscle of a patient that is relevant for an inspiratory breathing effort and a second data signal that indicates an activity of at least one muscle of the patient that is relevant for an expiratory breathing effort. The data signals are generated from electromyography (EMG) signals detected by surface electromyography sensors. A computer is configured to determine breathing phase information on the basis of a breathing signal and to check at least one of the electromyography signals or at least one of the separated signals for detectability of a heart signal component and further to assign the signals to an inspiratory breathing activity as well as to an expiratory breathing activity of the patient as a function of the breathing phase information.

Abstract: A signal processing apparatus and method are provided for separating a plurality of mixture signals from a MIMO system to iteratively obtain a plurality of output signals. The plurality of mixture signals are a response of the MIMO system to a plurality of source signals. The signal processing apparatus comprises a plurality of blind source separators including a first blind source separator based on a first blind source separation technique or algorithm and a second blind source separator based on a second blind source separation technique or algorithm, wherein the first blind source separator is configured to compute a first plurality of preliminary output signals on the basis of a first set of coefficients describing the MIMO system and wherein the second blind source separator is configured to compute a second plurality of preliminary output signals on the basis of a second set of coefficients describing the MIMO system.

Abstract: Provided are methods and systems for acoustic keystroke transient cancellation/suppression for user communication devices using a semi-blind adaptive filter model. The methods and systems are designed to overcome existing problems in transient noise suppression by taking into account some less-defective signal as side information on the transients and also accounting for acoustic signal propagation, including the reverberation effects, using dynamic models. The methods and systems take advantage of a synchronous reference microphone embedded in the keyboard of the user device, and utilize an adaptive filtering approach exploiting the knowledge of this keybed microphone signal.

Abstract: Provided are methods and systems for acoustic keystroke transient cancellation/suppression for user communication devices using a semi-blind adaptive filter model. The methods and systems are designed to overcome existing problems in transient noise suppression by taking into account some less-defective signal as side information on the transients and also accounting for acoustic signal propagation, including the reverberation effects, using dynamic models. The methods and systems take advantage of a synchronous reference microphone embedded in the keyboard of the user device, and utilize an adaptive filtering approach exploiting the knowledge of this keybed microphone signal.

Abstract: Provided are methods and systems for enhancing speech when corrupted by transient noise (e.g., keyboard typing noise). The methods and systems utilize a reference microphone input signal for the transient noise in a signal restoration process used for the voice part of the signal. A robust Bayesian statistical model is used to regress the voice microphone on the reference microphone, which allows for direct inference about the desired voice signal while marginalizing the unwanted power spectral values of the voice and transient noise. Also provided is a straightforward and efficient Expectation-maximization (EM) procedure for fast enhancement of the corrupted signal. The methods and systems are designed to operate easily in real-time on standard hardware, and have very low latency so that there is no irritating delay in speaker response.

Abstract: An audio signal decorrelator for deriving an output audio signal from an input audio signal has a frequency analyzer for extracting from the input audio signal a first partial signal descriptive of an audio content in a first audio frequency range and a second partial signal descriptive of an audio content in a second audio frequency range having higher frequencies compared to the second audio frequency range. A partial signal modifier modifies the first and second partial signals, to obtain first and second processed partial signals, so that a modulation amplitude of a time variant phase shift or time variant delay applied to the first partial signal is higher than that applied to the second partial signal, or for modifying only the first partial signal. A signal combiner combines the first and second processed partial signals, or combines the first processed partial signal and the second partial signal, to obtain an output audio signal.

Abstract: An apparatus for processing an input signal, wherein the input signal comprises a plurality of subsignals associated to discrete transmitters or receivers, wherein the discrete transmitters or receivers are disposed at predetermined geometrical positions with regard to a room, comprises a means for providing a plurality of wave-field components, wherein a superposition of the plurality of wave-field components results in a composite wave field, wherein the composite wave field can be propagated in the room, wherein the plurality of wave-field components are derived from the input signal by wave field decomposition based on orthogonal wave field base functions and the predetermined geometrical positions, a plurality of single filters, wherein a wave-field component of the plurality of wave-field components is associated to a single filter, wherein the single filter is formed to influence the associated wave-field component such that with regard to the plurality of single filters a plurality of filtered wave-fi

Abstract: A signal separator, a method and computer product for determining a first output signal describing an audio content of a useful-signal source in a first microphone signal, and for determining a second output signal describing an audio content of the useful-signal source in a second microphone signal.

Abstract: A signal separator, a method and computer product for determining a first output signal describing an audio content of a useful-signal source in a first microphone signal, and for determining a second output signal describing an audio content of the useful-signal source in a second microphone signal.

Abstract: An audio signal decorrelator for deriving an output audio signal from an input audio signal has a frequency analyzer for extracting from the input audio signal a first partial signal descriptive of an audio content in a first audio frequency range and a second partial signal descriptive of an audio content in a second audio frequency range having higher frequencies compared to the second audio frequency range. A partial signal modifier for modifies the first and second partial signals, to obtain first and second processed partial signals, so that a modulation amplitude of a time variant phase shift or time variant delay applied to the first partial signal is higher than that applied to the second partial signal, or for modifying only the first partial signal. A signal combiner combines the first and second processed partial signals, or combines the first processed partial signal and the second partial signal, to obtain an output audio signal.

Abstract: An apparatus for processing an input signal, wherein the input signal comprises a plurality of subsignals associated to discrete transmitters or receivers, wherein the discrete transmitters or receivers are disposed at predetermined geometrical positions with regard to a room, comprises a means for providing a plurality of wave-field components, wherein a superposition of the plurality of wave-field components results in a composite wave field, wherein the composite wave field can be propagated in the room, wherein the plurality of wave-field components are derived from the input signal by wave field decomposition based on orthogonal wave field base functions and the predetermined geometrical positions, a plurality of single filters, wherein a wave-field component of the plurality of wave-field components is associated to a single filter, wherein the single filter is formed to influence the associated wave-field component such that with regard to the plurality of single filters a plurality of filtered wave-fi

Abstract: A multichannel full-duplex audio signal transmission system, comprising an adaptive filter (2) provided for multichannel acoustic echo cancellation. A channel combining device (5) is provided between the preprocessing units (V1, . . . VD) and loudspeakers (L1, . . . ,LD), in which several (C>D) loudspeakers can be connected to one and the same preprocessing unit and by means of which the remaining D-C preprocessing units can be separated from the loudspeakers (L1, . . , LD). The channel combining device permits an optimization of the convergence ratio of the adaptive adjustment of filter coefficients in said adaptive filter (2).

Abstract: The present invention relates to a device and method for carrying out acoustic echo cancellation (2) when playing back C-channel audio signals on a D-channel audio signal transmission system with C<D. This invention can be used, for example, in videoconferencing, in which a variable number of active speakers are spatially played back according to the seating positions thereof. The aim of the invention is to solve for the poor convergence of the adaptive echo cancellation when the loudspeaker signals are strongly correlated. The invention provides that in addition to the known method of subjecting audio signals to preprocessing (V1, . . . , VD), which preferably induces a decorrelation, C output signals of the preprocessing units are selected by a channel combining device (5) and distributed to the loudspeakers (L1, . . . , LD), several signals being played back on a number of loudspeakers.