Huawei Technologies Co., Limited

Liyun Pang

Fons Adriaensen

Roman Schlieper

Christof Faller

Yue Lang

Peter Grosche

Alexis Favrot

Song Li

Anisse Taleb

David Virette

Karim Helwani

Pablo Hoffmann

An audio controller for an active noise reduction, ANR, reduces an ambient noise signal. The audio controller includes processing circuitry configured to provide a fixed ANR filter configured to generate a first noise reduction signal. Moreover, the processing circuitry is further configured to provide an adaptive ANR filter. The adaptive ANR filter includes one or more adjustable filter coefficients for adapting the adaptive ANR filter and the adaptive ANR filter is configured to generate a second noise reduction signal. The processing circuitry is further configured to generate a total noise reduction signal as an adjustable weighted linear combination of the first noise reduction signal and the second noise reduction signal.

Audio controller for a semi-adaptive active noise reduction device

The present disclosure provides a method for evaluating an electronic device. The method comprises determining, with an acoustic tube, a value of a first parameter, the value of the first parameter being indicative of the acoustic impedance of a reference termination. The method further comprises determining, with the acoustic tube, a value of a second parameter, the value of the second parameter being indicative of the acoustic impedance of the reference termination, when occluded by the electronic device. The method then comprises calculating a value of a third parameter, the value of the third parameter being indicative of the acoustic impedance of the electronic device, based on the value of the first parameter and the value of the second parameter.

System and method for evaluating an acoustic characteristic of an electronic device

The disclosure relates to a method for processing a stereo signal. The method can include obtaining a center channel signal by up-mixing the stereo signal. The method can also include generating a filtered center channel signal by applying one or more peak filters and one or more notch filters to the center channel signal. Furthermore, the method can include generating a binaural signal based on the filtered center channel signal.

Method and apparatus for processing a stereo signal

The invention relates to the technical fields of binaural audio rendering and, to this end, estimation of room acoustic parameters like reverberation time. In particular, the invention provides a device and method for estimating such acoustic parameters. The device is configured to record an acoustic signal, particularly a speech signal, to estimate a frequency-dependent reverberation time in a lower frequency range based on the recorded acoustic signal, and to extend the frequency-dependent reverberation time to a higher frequency range based on a predetermined model to obtain an extended frequency-dependent reverberation time. Virtual 3D audio can thus be adapted to a real room.

Device and method for adaptation of virtual 3D audio to a real room

A method for processing an audio signal, the method including: processing the audio signal according to a pair of mouth to ear transfer functions to obtain a processed audio signal; filtering the processed audio signal, using a pair of equalization filters, to obtain a filtered audio signal, where a parameter of the equalization filter is depends on an acoustic impedance of a headphone; and outputting the filtered audio signal to the headphone. Accordingly, this method counteracts the occlusion effect and to provides a natural perceived sound pressure.

Method and apparatus for processing an audio signal based on equalization filter

An audio signal processing apparatus, comprising a memory configured to store a set of pairs of predefined left ear and right ear transfer functions, which are predefined for a plurality of reference positions relative to the listener, a processor configured to determine a pair of left ear and right ear transfer functions on the basis of the set of predefined pairs of predefined left ear and right ear transfer functions for the azimuth angle and the elevation angle of the virtual target position and an adjustment filter configured to filter the input audio signal on the basis of the determined pair of left ear and right ear transfer functions and an adjustment function configured to adjust a delay between the left ear transfer function and the right ear transfer function to obtain a left ear output audio signal and a right ear output audio signal.

Audio signal processing apparatus and method

The disclosure relates to an audio signal processing apparatus for processing a plurality of audio signals defining a plurality of audio signal spectra, the audio signals to be transmitted to a listener in such a way that the listener perceives the audio signals to originate from virtual positions of a plurality of audio signal sources. The audio signal processing apparatus comprises a selector configured to select a spatial arrangement of the virtual positions of the audio signal sources relative to the listener from a plurality of possible spatial arrangements, and a filter configured to filter the plurality of audio signals on the basis of the selected spatial arrangement.

The disclosure relates to an apparatus for manipulating an input audio signal associated to a spatial audio source within a spatial audio scenario, wherein the spatial audio source has a certain distance to a listener within the spatial audio scenario. The apparatus comprises an exciter adapted to manipulate the input audio signal to obtain an output audio signal, and a controller adapted to control parameters of the exciter for manipulating the input audio signal based on the certain distance.

Apparatus and a method for manipulating an input audio 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.

Signal processing apparatus, method and computer program for dereverberating a number of input audio signals

The invention relates to a method for processing a multi-channel audio signal which carries a plurality of audio channel signals. The method comprises determining a time-scaling position using the plurality of audio channel signals and time-scaling each audio channel signal of the plurality of audio channel signals according to the time-scaling position to obtain a plurality of time scaled audio channel signals.

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Munich, Germany

Liyun Pang