The purpose of the present invention is to reduce distortion a frequency band component encoded with a small number of bits in a time domain and improve quality. An audio decoding device (10) decodes an encoded audio signal and outputs the audio signal. A decoding unit (10a) decodes an encoded sequence containing an encoded audio signal and obtains a decoded signal. A selective temporal envelope shaping unit (10b) shapes a temporal envelope of a decoded signal in the frequency band on the basis of decoding related information concerning decoding of the encoded sequence.

A wireless audio system for encoding and decoding an audio signal using spectral bandwidth replication is provided. Bandwidth extension is performed in the time-domain, enabling low-latency audio coding.

An apparatus for generating an enhanced signal from an input signal, wherein the enhanced signal has spectral values for an enhancement spectral region, the spectral values for the enhancement spectral regions not being contained in the input signal, includes a mapper for mapping a source spectral region of the input signal to a target region in the enhancement spectral region, the source spectral region including a noise-filling region; and a noise filler configured for generating first noise values for the noise-filling region in the source spectral region of the input signal and for generating second noise values for a noise region in the target region, wherein the second noise values are decorrelated from the first noise values or for generating second noise values for a noise region in the target region, wherein the second noise values are decorrelated from first noise values in the source region.

A first encoding unit generates a first encoded signal by encoding a component within a first band in a voice signal. A frequency shifting unit shifts the frequency of a component within a second band in the voice signal, the second band having a frequency higher than that of the first band, to the frequency of a component within the first band. A second encoding unit generates a second encoded signal by encoding the component whose frequency has been shifted in the frequency shifting unit. An output unit outputs both the first encoded signal generated in the first encoding unit and the second encoded signal generated in the second encoding unit.

An apparatus and method are disclosed for processing an audio signal. The apparatus includes an input interface, a digital filterbank having an analysis part and a synthesis part, a first phase shifter, a spectral envelope adjuster, a second phase shifter, and an output interface. The first phase shifter and the second phase shifter reduce a complexity of the digital filterbank, which includes both analysis and synthesis filters that are complex-exponential modulated versions of a prototype filter.

An audio signal encoding and decoding method, an audio signal encoding and decoding apparatus, a transmitter, a receiver, and a communications system, which can improve encoding and/or decoding performance. The audio signal encoding method includes dividing a to-be-encoded time domain signal into a low band signal and a high band signal; encoding the low band signal to obtain a low frequency encoding parameter; calculating a voiced degree factor, and predicting a high band excitation signal; weighting the high band excitation signal and random noise using the voiced degree factor, so as to obtain a synthesized excitation signal; and obtaining a high frequency encoding parameter based on the synthesized excitation signal and the high band signal. Technical solutions in the embodiments of the present invention can improve an encoding or decoding effect.

An audio signal is transmitted from a transmitter to a receiver in a hearing device, particularly a hearing aid. A communication facility configured for transmitting and/or receiving the audio signal. A hearing device system has two hearing devices configured to transmit audio signals between the two hearing devices by way of their communication facilities in accordance with the method.

A gain adjustment apparatus for use in decoding of audio that has been encoded with separate gain and shape representations includes an accuracy meter configured to estimate an accuracy measure of the shape representation, and to determine a gain correction based on the estimated accuracy measure. An envelope adjuster further included in the apparatus is configured to adjust the gain representation based on the determined gain correction.

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.

For an efficient encoding of subband configuration data the first, penultimate and last subband groups are treated differently than the other subband groups. Further, subband group bandwidth difference values are used in the encoding. The number of subband groups N.sub.SB is coded using a fixed number of bits representing N.sub.SB−1. The bandwidth value B.sub.SB[1] of the first subband group is coded using a unary code representing B.sub.SB[1]−1. No bandwidth value B.sub.SB[g] is coded for the last subband g=N.sub.SB. For subband groups g=2, . . . , N.sub.SB−2 bandwidth difference values ΔB.sub.SB [g]=B.sub.SB [g]−B.sub.SB[g−1] are coded using a unary code, and the bandwidth difference value ΔB.sub.SB[N.sub.SB−1] for subband group g=N.sub.SB−1 is coded using a fixed number of bits.