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

Technische Universitaet Ilmenau

Technische Universitat Ilmenau

Christoph Sladeczek

Andreas Franck

Thomas Sporer

Karlheinz Brandenburg

Annika Neidhardt

Stephan Werner

Michael Rath

Daniel Beer

Hanna Lukashevich

Jakob Abesser

Georg Fischer

Martina Böhme

Albert Zhykhar

Florian Klein

Ulrike Sloma

Christian Schneiderwind

Claudia Stirnat

Johannes Nowak

A system and a corresponding method for assisting selective hearing are provided. The system includes a detector for detecting an audio source signal portion of one or more audio sources by using at least two received microphone signals of a hearing environment. In addition, the system includes a position determiner for allocating position information to each of the one or more audio sources. In addition, the system includes an audio type classifier for assigning an audio source signal type to the audio source signal portion of each of the one or more audio sources. In addition, the system includes a signal portion modifier for varying the audio source signal portion of at least one audio source of the one or more audio sources depending on the audio signal type of the audio source signal portion of the at least one audio source so as to obtain a modified audio signal portion of the at least one audio source. In addition, the system includes a signal generator.

System and method for assisting selective hearing

The effectiveness of calculating FIR filter coefficients for beam-forming filters for transducer arrays such as arrays of microphones or loudspeakers, for example, is increased in that the calculation is performed in two stages; namely, on the one hand, by calculating frequency domain filter weights of the beam-forming filters, i.e., coefficients describing the transfer functions of the beam-forming filters within the dimension of the frequency so as to obtain target frequency responses for the beam-forming filters, so that applying the beam-forming filters to the array approximates a desired directional selectivity, and followed by calculating the FIR filter coefficients for the beam-forming filters, i.e., of coefficients describing the impulse response of the beam-forming filters within the time domain, such that the frequency responses of the FIR beam-forming filters approximate the target frequency responses in an optimum manner in accordance with defined criteria.

FIR filter coefficient calculation for beam-forming filters

A device for calculating loudspeaker signals for a plurality of loudspeakers while using a plurality of audio sources, an audio source including an audio signal, includes a forward transform stage for transforming each audio signal, block-by-block, to a spectral domain so as to obtain for each audio signal a plurality of temporally consecutive short-term spectra, a memory for storing a plurality of temporally consecutive short-term spectra for each audio signal, a memory access controller for accessing a specific short-term spectrum among the plurality of short-term spectra for a combination consisting of a loudspeaker and an audio signal on the basis of a delay value, a filter stage for filtering the specific short-term spectrum for the combination of the audio signal and the loudspeaker by using a filter provided for the combination of the audio signal and the loudspeaker, so that a filtered shot-term spectrum is obtained for each combination of an audio signal and a loudspeaker, a summing stage for summing up the filtered short-term spectra for a loudspeaker so as to obtain summed-up short-term spectra for each loudspeaker, and a backtransform stage for backtransforming, block-by-block, summed-up short-term spectra for the loudspeakers to a time domain so as to obtain the loudspeaker signals.

Device and method for calculating loudspeaker signals for a plurality of loudspeakers while using a delay in the frequency domain

A mixing console () for processing at least a first and a second source signal and for providing a mixed audio signal comprises an audio signal generator () for providing an audio signal () for a virtual listening position () within a space (), in which an acoustic scene is recorded by at least a first microphone () at a first known position within the space () as the first source signal () and by at least a second microphone () at a second known position within the space () as the second source signal (). The audio signal generator () comprises an input interface () configured to receive the first source signal () recorded by the first microphone () and the second source signal () recorded by the second microphone (), and a geometry processor () configured to determine a first piece of geometry information () based on the first position and the virtual listening position () and a second piece of geometry information () based on the second position and the virtual listening position (). A signal generator () for providing the audio signal () is configured to combine at least the first source signal () and the second source signal () according to a combination rule using the first piece of geometry information () and the second piece of geometry information ().

Mixing desk, sound signal generator, method and computer program for providing a sound signal

A device for determining room-optimized transfer functions for a listening room serving for room-optimized post-processing of audio signals in spatial production, is configured to analyze room acoustics of the listening room and to determine, based on the analysis of the room acoustics, the room-optimized transfer functions for the listening room where the spatial reproduction by means of a binaural close-range sound transducer is to take place. The spatial reproduction of the audio signals by means of the binaural close-range sound transducer may then be emulated using known head-related transfer functions und using the room-optimized transfer functions, wherein a room to be synthesized may be emulated based on the head-related transfer functions, and wherein the listening room may be emulated based on the room-optimized transfer functions.

Determining and using room-optimized transfer functions

A loudspeaker system for a vehicle includes a loudspeaker array including a plurality of electroacoustic sound transducers that can be controlled individually, such that a user-specific audio signal can be reproduced for different users at different listening positions in a vehicle interior of the vehicle via the plurality of electroacoustic sound transducers. Here, the loudspeaker array or a sound outlet of the loudspeaker array is arranged in particular between at least two of the listening positions in the vehicle interior, i.e. for example between the driver and the passenger seat.

Loudspeaker system

A more clear-cut separation of a first audio signal within a first region of a sonication area of a plurality of loudspeakers is achieved in that a calculator calculates that version of the audio signals which results from the spatially selective reproduction of the audio signals at this first region, in that a masking threshold is calculated as a function of the version of that audio signal which is to be separated from the one or more other audio signals at this region, and in that the emission of the audio signals for spatially selective reproduction to the outputs of the plurality of loudspeakers is influenced as a function of a comparison of the masking threshold with the version of the one or more other, i.e. spurious, audio signals.

Device and method for spatially selective audio reproduction

A device for calculating loudspeaker signals using a plurality of audio sources, an audio source including an audio signal, includes a forward transform stage for transforming each audio signal to a spectral domain to obtain a plurality of temporally consecutive short-term spectra, a memory for storing a plurality of temporally consecutive short-term spectra for each audio signal, a memory access controller for accessing a specific short-term spectrum for a combination consisting of a loudspeaker and an audio signal based on a delay value, a filter stage for filtering the specific short-term spectrum by using a filter, so that a filtered short-term spectrum is obtained for each audio signal and loudspeaker combination, a summing stage for summing up the filtered short-term spectra for a loudspeaker to obtain summed-up short-term spectra, and a backtransform stage for backtransforming summed-up short-term spectra for the loudspeakers to a time domain to obtain the loudspeaker signals.

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Ilmenau, Germany

Christoph Sladeczek