Huawei Technologies Co., Limited

Friedrich-Alexander-Universitat Erlangen-Nürnberg

Huawei Technologies Duesseldorf Gmbh

University of Cambridge

Karim Helwani

Peter Grosche

Yue Lang

Panji Setiawan

Lukasz Kondrad

Liyun Pang

Simone Fontana

Walter Kellermann

Simon J Godsill

Robert Höldrich

Jürgen Geiger

Milos Markovic

Kainan Chen

Juergen Geiger

Herbert Buchner

Nicola Piotto

Michael Buerger

Heinrich Löllmann

Michael Bürger

Franz Zotter

Christian Schörkhuber

Matthias Frank

Christian SCHÖRKHUBER

The proposed method for localizing a target sound source from a plurality of sound sources, wherein a multi-channel recording signal of the plurality of sound sources comprises a plurality of microphone channel signals, comprises converting each microphone channel signal into a respective channel spectrogram in a time-frequency domain, blindly separating the channel spectrograms to obtain a plurality of separated source signals, identifying, among the plurality of separated source signals, the separated source signal that best matches a target source model, estimating, based on the identified separated source signal, a binary mask reflecting where the target sound source is active in the channel spectrograms in terms of time and frequency, applying the binary mask on the channel spectrograms to obtain masked channel spectrograms, and localizing the target sound source from the plurality of sound sources based on the masked channel spectrograms.

Localization algorithm for sound sources with known statistics

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.

Signal processing apparatus and method

The invention relates to audio signal processing apparatuses and methods, such as an audio signal downmixing apparatus () for processing an input audio signal comprising a plurality of input channels () into an output audio signal comprising a plurality of primary output channels () and at least one auxiliary output channel () using a downmix matrix D, wherein the downmix matrix D comprises a primary downmix matrix Dproviding the plurality of primary output channels () and an auxiliary downmix matrix Dproviding the at least one auxiliary output channel (). The audio signal downmixing apparatus () comprises an auxiliary downmix matrix determiner () configured to determine the auxiliary downmix matrix D, and a processor () configured to process the input audio signal into the output audio signal using the downmix matrix D.

Audio signal processing apparatuses and methods

A sound signal processing apparatus including a plurality of microphones, where each microphone is configured to receive the sound signal from the target source, a processor configured to estimate a first power measure on the basis of the sound signal from the target source received by a first microphone of the microphones and a second power measure on the basis of the sound signal from the target source received by at least a second microphone of the microphones, which is located more distant from the target source than the first microphone, and the processor is further configured to determine a gain factor on the basis of a ratio between the second power measure and the first power measure, and an amplifier configured to apply the gain factor to the sound signal from the target source received by the first microphone.

Sound signal processing apparatus and method for enhancing a sound signal

An image processing device performing color balancing of a first image and at least a second image is provided. The image processing device comprises a color balancing determination unit and a color balancing calculation unit. The color balancing determination unit determines a global gain vector (t) comprising at least two gain factors (&#xe2;, &#xe2;) of the first and second images by minimizing a cost function based upon reference pixel values of the first and second images. The first and second image reference pixels depict a shared color scene of the two images. The color balancing calculation unit performs color balancing of the first image based upon the gain factor (&#xe2;) of the first image and to perform a color balancing of the second image based upon the gain factor (&#xe2;) of the second image.

Image processing device and method for color balancing

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.

Apparatus and method for driving an array of loudspeakers

Audio signal processing apparatuses and methods are provided, such as an audio signal downmixing apparatus for processing an input audio signal into an output audio signal, wherein the input audio signal comprises a plurality of input channels recorded at a plurality of spatial positions and the output audio signal comprises a plurality of primary output channels. The audio signal downmixing apparatus comprises a downmix matrix determiner configured to determine for each frequency bin j of a plurality of frequency bins a downmix matrix Dwith j being an integer in the range from 1 to N, and a processor configured to process the input audio signal using the downmix matrix Dinto the output audio signal.

An audio signal processing device for generating a plurality of output signals for a plurality of loudspeakers from an input audio signal comprises a driving function determining unit adapted to determined driving functions of a plurality of loudspeakers for generating a virtual left binaural signal source and a virtual right binaural signal source based upon a position and a directivity of the virtual left binaural signal source, a position and a directivity of the virtual right binaural signal source and positions of the plurality of loudspeakers. Moreover, it comprises a filtering unit adapted to filter a left binaural signal and a right binaural signal using the driving functions of the plurality of loudspeakers resulting in the plurality of output signals. The left binaural signal and the right binaural signal constitute the input audio signal or are derived there from.

Audio signal processing device and method for reproducing a binaural signal

A signal processing apparatus for dereverberating a number of input audio signals, where the signal processing apparatus includes a processor configured to transform the number of input audio signals into a transformed domain to obtain input transformed coefficients, the input transformed coefficients being arranged to form an input transformed coefficient matrix, determine filter coefficients upon the basis of eigenvalues of a signal space, the filter coefficients being arranged to form a filter coefficient matrix, convolve input transformed coefficients of the input transformed coefficient matrix by filter coefficients of the filter coefficient matrix to obtain output transformed coefficients, and the output transformed coefficients being arranged to form an output transformed coefficient matrix.

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Munich, Germany

Karim Helwani