Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V.

U.S. Philips Corporation

Siemens Audiologische Technik Gmbh

Sivantos Pte. Ltd.

Other

Siemens Aktiengesellschaft

Nuance Communications, Inc.

Siemens Medical Instruments Pte. Ltd.

Friedrich-Alexander-Universitat Erlangen-Nürnberg

Kellermann, Walter

Buchner, Herbert

Spors, Sascha

Herbordt, Wolfgang

Walter Kellermann

Martin Schneider

Klaus Reindl

Yuanhang Zheng

Herbert Buchner

Eghart Fischer

Henning Puder

Christian Hofmann

Homayoun Kamkar Parsi

Marc Aubreville

Andreas Franck

Rainer Martin

Kees Janse

Stefan Meier

Charles Knibbler

Wolfgang Herbordt

Parijat Ashok Oak

Robert Aichner

Sascha Spors

Wolfgang Herbodt

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.

Rendering system

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.

Apparatus and method for providing a loudspeaker-enclosure-microphone system description

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.

Device and method for decorrelating loudspeaker signals

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.

Method for evaluating a useful signal and audio device

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. At least one of the subfilters has a number of the received loudspeaker signals that is smaller than a total number of the plurality of transformed loudspeaker signals.

Apparatus and method for listening room equalization using a scalable filtering structure in the wave domain

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.

Method for adjusting the useful signal in binaural hearing aid systems and hearing aid system

A method determines a bias reduced noise and interference estimation in a binaural microphone configuration with a right and a left microphone signal at a time-frame with a target speaker active. The method includes a determination of the auto power spectral density estimate of the common noise formed of noise and interference components of the right and left microphone signals and a modification of the auto power spectral density estimate of the common noise by using an estimate of the magnitude squared coherence of the noise and interference components contained in the right and left microphone signals determined at a time frame without a target speaker active. An acoustic signal processing system and a hearing aid implement the method for determining the bias reduced noise and interference estimation. The noise reduction performance of speech enhancement algorithms is improved by the invention. Further, distortions of the target speech signal and residual noise and interference components are reduced.

Method and acoustic signal processing system for interference and noise suppression in binaural microphone configurations

A method and an acoustic signal processing system for noise reduction of a binaural microphone signal (x, x) with one target point source and M interfering point sources (n, n, . . . , n) as input sources to a left and a right microphone of a binaural microphone system, include: where His the Wiener filter, &#x3a6;is the auto power spectral density of the sum of all of the M interfering point sources components (x1,n, x2,n) contained in the left and right microphone signals and &#x3a6;is the auto power spectral density of the sum of the left and right microphone signals. Due to the linear-phase property of the calculated Wiener filter, original binaural cues are perfectly preserved not only for the target source but also for the residual interfering sources.

Blind source separation method and acoustic signal processing system for improving interference estimation in binaural wiener filtering

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-field components representing a processed input signal are obtained on the output side.

Apparatus and method for processing continuous wave fields propagated in a room

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.

Signal separator, method for determining output signals on the basis of microphone signals, and computer program

It is intended to improve and automate the calculation of calibration filters connected downstream from the microphones of an RGSC beamformer. To this end it is proposed that an adaptive calibration filter calculation unit be used, by means of which calibration filters are calculated from the output signals of adaptive blocking filters such that the power of an output signal of a blocking filter subtracted from a reference signal and filtered by means of a calibration filter respectively is minimized. The calibration filters connected downstream from the microphones are then replaced by the calibration filters thus determined.

Microphone calibration with an RGSC beamformer

A speech signal processing system combines acoustic noise reduction and echo cancellation to enhance acoustic performance. The speech signal processing system may be used in vehicles or other environments where noise-suppressed communication is desirable. The system includes an adaptive beamforming signal processing unit, an adaptive echo compensating unit to reduce acoustic echoes, and an adaptation unit to combine noise reduction and adaptive echo compensating.

Speech signal processing with combined noise reduction and echo compensation

Mobile radiotelephone with handsfree device

A speech processing arrangement has at least two microphones for supplying microphone signals formed by speech components and noise components to microphone signal branches that are coupled to an adder device used for forming a sum signal. The microphone signals are delayed and weighted by weight factors in the microphone signal branches. The arrangement includes an evaluation circuit that a) receives the microphone signals, b) estimates the noise components, c) estimates the speech components by forming the difference between one of the microphone signals and the estimated noise component for this microphone signal, d) selects one of the microphone signals as a reference signal which contains a reference noise component and a reference speech component, e) forms speech signal ratios by dividing the estimated speech components by the estimated reference speech component, f) forms noise signal ratios by dividing the powers of the estimated noise components by the power of the estimated reference noise component, and g) determines the weight factors by dividing each speech signal ratio by the associated noise signal ratio. The signal-to-noise ratio corresponds to the ratio of the power of the speech component to the power of the noise component of the sum signal. Because the speech signals are correlated and noise signals are uncorrelated, the sum signal available on the output of the adder device has a reduced noise component yielding improved speech audibility. Real-time computation of the weight factors eliminates any annoying delay during a conversation held using the speech processing arrangement.

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Eckental, Germany

Walter Kellermann