Abstract:
This application provides an upgrade method including: An electronic device acquires a first verification voice entered by a user; processes the first verification voice by using a first model stored in the electronic device, to obtain a first voiceprint feature; verifies an identity of the user based on the first voiceprint feature and a first user feature template stored in the electronic device; after the identity of the user is verified, if the electronic device has received a second model, processes the first verification voice by using the second model, to obtain a second voiceprint feature; and updates the first user feature template based on the second voiceprint feature, and updates the first model by using the second model.
Abstract:
The invention relates to a method for determining an encoding parameter for an audio channel signal of a multi-channel audio signal, the method comprising: determining a frequency transform of the audio channel signal; determining a frequency transform of a reference audio signal; determining inter channel differences for at least each frequency sub-band of a subset of frequency sub-bands, each inter channel difference indicating a phase difference or time difference between a band-limited signal portion of the audio channel signal and a band-limited signal portion of the reference audio signal in the respective frequency sub-band the inter-channel difference is associated to; determining a first average based on positive values of the inter-channel differences and determining a second average based on negative values of the inter-channel differences; and determining the encoding parameter based on the first average and on the second average.
Abstract:
A wave field synthesis apparatus for driving an array of loudspeakers with drive signals, the apparatus includes a sound field synthesizer for generating sound field drive signals for causing the array of loudspeakers to generate one or more sound fields at one or more audio zones, a binaural renderer for generating binaural drive signals for causing the array of loud-speakers to generate specified sound pressures at at least two positions, wherein the at least two positions are determined based on a detected position and/or orientation of a listener, and a decision unit for deciding whether to generate the drive signals using the sound field synthesizer or using the binaural renderer.
Abstract:
A method and apparatus for decoding a multichannel audio signal comprising the steps of receiving a downmix audio signal and an interchannel cross correlation parameter; deriving an interchannel phase difference parameter from the received interchannel cross correlation parameter; and calculating a decoded multichannel audio signal for the received downmix audio signal depending on the derived interchannel phase difference parameter.
Abstract:
This application provides a voice control method and apparatus, a wearable device, and a terminal. The method includes: obtaining voice information of a user; obtaining identity information of the user based on a first voiceprint recognition result of a first voice component of the voice information, a second voiceprint recognition result of a second voice component of the voice information, and a third voiceprint recognition result of a third voice component of the voice information, where the first voice component is captured by an in-ear voice sensor of a wearable device, the second voice component is captured by an out-of-ear voice sensor of the wearable device, and the third voice component is captured by a bone vibration sensor of the wearable device; and executing an operation instruction when the identity information of the user matches the preset identity information.
Abstract:
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.
Abstract:
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.
Abstract:
A stereo decoding method and apparatus are disclosed. The method includes: restoring a monophonic signal from a received code stream through decoding; restoring an interchannel level difference, a group delay, and a group phase from the received code stream through decoding; and processing the monophonic signal according to the interchannel level difference, group delay, and group phase to obtain a first channel signal and a second channel signal. According to the stereo decoding method and apparatus provided in embodiments of the present invention, the first and second channel signals are obtained according to the monophonic signal, ILD, group delay, and group phase by referring to not only the ILD but also the group delay and group phase, thereby yielding favorable stereo sound field effect for the obtained first and second channel signals.
Abstract:
The invention relates to a sound processing node for an arrangement of sound processing nodes, the sound processing nodes being configured to receive a plurality of sound signals, wherein the sound processing node comprises a processor configured to generate an output signal on the basis of the plurality of sound signals weighted by a plurality of beamforming weights, wherein the processor is configured to adaptively determine the plurality of beamforming weights on the basis of an adaptive linearly constrained minimum variance beamformer using a transformed version of a least mean squares formulation of a constrained gradient descent approach, wherein the transformed version of the least mean squares formulation of the constrained gradient descent approach is based on a transformation of the least mean squares formulation of the constrained gradient descent approach to the dual domain.
Abstract:
The invention relates to a method for determining an encoding parameter for an audio channel signal of a multi-channel audio signal, the method comprising: determining for the audio channel signal a set of functions from the audio channel signal and a reference audio signal; determining a first set of encoding parameters based on a smoothing of the set of functions with respect to a frame sequence of the multi-channel audio signal, the smoothing being based on a first smoothing coefficient; determining a second set of encoding parameters based on a smoothing of the set of functions with respect to the frame sequence of the multi-channel audio signal, the smoothing being based on a second smoothing coefficient; and determining the encoding parameter based on a quality criterion with respect to the first set of encoding parameters and/or the second set of encoding parameters.