A method transmits an audio signal from a transmitter to a receiver and a hearing device, particularly a hearing aid, contains a communication facility which is provided and configured for transmitting and/or receiving an audio signal according to the method. A hearing device system has two hearing devices and is provided and configured to transmit audio signals between the two hearing devices by their communication facilities according to the method.

A method transmits an audio signal from a transmitter to a receiver and a hearing device, particularly a hearing aid, contains a communication facility which is provided and configured for transmitting and/or receiving an audio signal according to the method. A hearing device system has two hearing devices and is provided and configured to transmit audio signals between the two hearing devices by their communication facilities according to the method.

Noise filling of a spectrum of an audio signal is improved in quality with respect to the noise filled spectrum so that the reproduction of the noise filled audio signal is less annoying, by performing the noise filling in a manner dependent on a tonality of the audio signal.

Noise filling of a spectrum of an audio signal is improved in quality with respect to the noise filled spectrum so that the reproduction of the noise filled audio signal is less annoying, by performing the noise filling in a manner dependent on a tonality of the audio signal.

An audio encoder for encoding an audio signal includes: a first encoding processor for encoding a first audio signal portion in a frequency domain, wherein the first encoding processor includes: a time frequency converter for converting the first audio signal portion into a frequency domain representation having spectral lines up to a maximum frequency of the first audio signal portion; a spectral encoder for encoding the frequency domain representation; a second encoding processor for encoding a second different audio signal portion in the time domain; a cross-processor for calculating, from the encoded spectral representation of the first audio signal portion, initialization data of the second encoding processor, so that the second encoding processing is initialized to encode the second audio signal portion immediately following the first audio signal portion in time in the audio signal; a controller configured for analyzing the audio signal and for determining, which portion of the audio signal is the first audio signal portion encoded in the frequency domain and which portion of the audio signal is the second audio signal portion encoded in the time domain; and an encoded signal former for forming an encoded audio signal including a first encoded signal portion for the first audio signal portion and a second encoded signal portion for the second audio signal portion.

Systems and methods for spoken language understanding are described. Embodiments of the systems and methods receive audio data for a spoken language expression, encode the audio data using a multi-stage encoder comprising a basic encoder and a sequential encoder, wherein the basic encoder is trained to generate character features during a first training phase and the sequential encoder is trained to generate token features during a second training phase, and decode the token features to generate semantic information representing the spoken language expression.

Systems and methods for spoken language understanding are described. Embodiments of the systems and methods receive audio data for a spoken language expression, encode the audio data using a multi-stage encoder comprising a basic encoder and a sequential encoder, wherein the basic encoder is trained to generate character features during a first training phase and the sequential encoder is trained to generate token features during a second training phase, and decode the token features to generate semantic information representing the spoken language expression.

In accordance with one aspect of the invention, a selector supports the selection of a first encoding scheme or the second encoding scheme based upon the detection or absence of the triggering characteristic in the interval of the input speech signal. The first encoding scheme has a pitch pre-processing procedure for processing the input speech signal to form a revised speech signal biased toward an ideal voiced and stationary characteristic. The pre-processing procedure allows the encoder to fully capture the benefits of a bandwidth-efficient, long-term predictive procedure for a greater amount of speech components of an input speech signal than would otherwise be possible. In accordance with another aspect of the invention, the second encoding scheme entails a long-term prediction mode for encoding the pitch on a sub-frame by sub-frame basis. The long-term prediction mode is tailored to where the generally periodic component of the speech is generally not stationary or less than completely periodic and requires greater frequency of updates from the adaptive codebook to achieve a desired perceptual quality of the reproduced speech under a long-term predictive procedure.

A stereo signal encoding method includes determining a window length of an attenuation window based on an inter-channel time difference, determining a modified linear prediction analysis window based on the window length of the attenuation window, where values of at least some points from a point (L−sub_window_len) to a point (L−1) in the modified linear prediction analysis window are less than values of corresponding points from a point (L−sub_window_len) to a point (L−1) in an initial linear prediction analysis window, and the window length of the modified linear prediction analysis window is equal to a window length of the initial linear prediction analysis window, and performing linear prediction analysis on a to-be-processed sound channel signal based on the modified linear prediction analysis window.

A stereo signal encoding method includes determining a window length of an attenuation window based on an inter-channel time difference, determining a modified linear prediction analysis window based on the window length of the attenuation window, where values of at least some points from a point (L−sub_window_len) to a point (L−1) in the modified linear prediction analysis window are less than values of corresponding points from a point (L−sub_window_len) to a point (L−1) in an initial linear prediction analysis window, and the window length of the modified linear prediction analysis window is equal to a window length of the initial linear prediction analysis window, and performing linear prediction analysis on a to-be-processed sound channel signal based on the modified linear prediction analysis window.