An apparatus for processing an audio signal having associated therewith a pitch lag information and a gain information, includes a domain converter for converting a first domain representation of the audio signal into a second domain representation of the audio signal; and a harmonic post-filter for filtering the second domain representation of the audio signal, wherein the post-filter is based on a transfer function including a numerator and a denominator, wherein the numerator includes a gain value indicated by the gain information, and wherein the denominator includes an integer part of a pitch lag indicated by the pitch lag information and a multi-tap filter depending on a fractional part of the pitch lag.

In methods and systems for filtering an information input signal, a system may have: a first filter unit filtering an input signal at an initial subinterval in a current update interval according to parameters associated to the preceding update interval, the parameters being scaled by a first scaling factor changing towards 0; and a second filter unit filtering a second filter input signal, based on the output of the first filter unit, at the initial subinterval, according to parameters associated to the current update interval, the parameters being scaled by a second scaling factor changing from 0, or a value close to 0, toward a value more distant from 0.

In methods and systems for filtering an information input signal, a system may have: a first filter unit filtering an input signal at an initial subinterval in a current update interval according to parameters associated to the preceding update interval, the parameters being scaled by a first scaling factor changing towards 0; and a second filter unit filtering a second filter input signal, based on the output of the first filter unit, at the initial subinterval, according to parameters associated to the current update interval, the parameters being scaled by a second scaling factor changing from 0, or a value close to 0, toward a value more distant from 0.

Disclosed is an LPC residual signal encoding/decoding apparatus of an MDCT based unified voice and audio encoding device. The LPC residual signal encoding apparatus analyzes a property of an input signal, selects an encoding method of an LPC filtered signal, and encode the LPC residual signal based on one of a real filterbank, a complex filterbank, and an algebraic code excited linear prediction (ACELP).

Disclosed is an LPC residual signal encoding/decoding apparatus of an MDCT based unified voice and audio encoding device. The LPC residual signal encoding apparatus analyzes a property of an input signal, selects an encoding method of an LPC filtered signal, and encode the LPC residual signal based on one of a real filterbank, a complex filterbank, and an algebraic code excited linear prediction (ACELP).

A hearing device comprises a) at least one input transducer configured to pick up sound from an acoustic environment around the user when the user is wearing the hearing device, the at least one input transducer providing at least one electric input signal representative of said sound, b) at least one analysis filter bank configured to provide said at least one electric input signal as a multitude of frequency sub-band signals, the at least one analysis filter bank comprising b1) a plurality of M first filters h.sub.m(n), whose impulse responses are modulated from a first prototype filter h(n), where m=0, 1, . . . , M−1 is a frequency band index, and n is a time index, c) a processor for processing said at least one electric input signal provided by said at least one analysis filter bank, or a signal originating therefrom, and providing a processed signal, d) an output transducer configured to provide stimuli perceivable as sound to the user in dependence of said processed signal, and e) a controller for controlling said analysis filter bank by applying a different first prototype filter to said at least one filter bank in dependence of said current acoustic environment. A method of operating a hearing device is further disclosed.

A hearing device comprises a) at least one input transducer configured to pick up sound from an acoustic environment around the user when the user is wearing the hearing device, the at least one input transducer providing at least one electric input signal representative of said sound, b) at least one analysis filter bank configured to provide said at least one electric input signal as a multitude of frequency sub-band signals, the at least one analysis filter bank comprising b1) a plurality of M first filters h.sub.m(n), whose impulse responses are modulated from a first prototype filter h(n), where m=0, 1, . . . , M−1 is a frequency band index, and n is a time index, c) a processor for processing said at least one electric input signal provided by said at least one analysis filter bank, or a signal originating therefrom, and providing a processed signal, d) an output transducer configured to provide stimuli perceivable as sound to the user in dependence of said processed signal, and e) a controller for controlling said analysis filter bank by applying a different first prototype filter to said at least one filter bank in dependence of said current acoustic environment. A method of operating a hearing device is further disclosed.

Methods and systems for encoding and decoding data in an audio channel are provided. At least one notch attribute for each of a set of notches to be applied to a source audio channel corresponding to data to be encoded is determined. The data is encoded by applying notch-filtering to the source audio channel to create a modified audio channel having the set of notches having the at least one notch attribute. The notches in the modified audio channel are then analyzed to determine at least one characteristic of each of the notches, and data is then decoded from the at least one characteristic of each of the notches.

The present disclosure relates to an apparatus for decoding an encoded Unified Audio and Speech stream. The apparatus comprises a core decoder for decoding the encoded Unified Audio and Speech stream. The core decoder includes a fast Fourier transform, FFT, module implementation based on a Cooley-Tuckey algorithm. The FFT module is configured to determine a discrete Fourier transform, DFT. Determining the DFT involves recursively breaking down the DFT into small FFTs based on the Cooley-Tucker algorithm and using radix-4 if a number of points of the FFT is a power of 4 and using mixed radix if the number is not a power of 4. Performing the small FFTs involves applying twiddle factors. Applying the twiddle factors involves referring to pre-computed values for the twiddle factors. The present disclosure further relates to an apparatus for decoding an encoded Unified Audio and Speech stream, in which the core decoder is configured to decode an LPC filter that has been quantized using a line spectral frequency, LSF, representation from the Unified Audio and Speech stream. Decoding the LPC filter from the Unified Audio and Speech stream comprises computing a first-stage approximation of a LSF vector, reconstructing a residual LSF vector, if an absolute quantization mode has been used for quantizing the LPC filter, determining inverse LSF weights for inverse weighting of the residual LSF vector by referring to pre-computed values for the inverse LSF weights or their respective corresponding LSF weights, inverse weighting the residual LSF vector by the determined inverse LSF weights, and calculating the LPC filter based on the inversely-weighted residual LSF vector and the first-stage approximation of the LSF vector. The present disclosure further relates to corresponding methods and storage media.

The present disclosure relates to an apparatus for decoding an encoded Unified Audio and Speech stream. The apparatus comprises a core decoder for decoding the encoded Unified Audio and Speech stream. The core decoder includes a fast Fourier transform, FFT, module implementation based on a Cooley-Tuckey algorithm. The FFT module is configured to determine a discrete Fourier transform, DFT. Determining the DFT involves recursively breaking down the DFT into small FFTs based on the Cooley-Tucker algorithm and using radix-4 if a number of points of the FFT is a power of 4 and using mixed radix if the number is not a power of 4. Performing the small FFTs involves applying twiddle factors. Applying the twiddle factors involves referring to pre-computed values for the twiddle factors. The present disclosure further relates to an apparatus for decoding an encoded Unified Audio and Speech stream, in which the core decoder is configured to decode an LPC filter that has been quantized using a line spectral frequency, LSF, representation from the Unified Audio and Speech stream. Decoding the LPC filter from the Unified Audio and Speech stream comprises computing a first-stage approximation of a LSF vector, reconstructing a residual LSF vector, if an absolute quantization mode has been used for quantizing the LPC filter, determining inverse LSF weights for inverse weighting of the residual LSF vector by referring to pre-computed values for the inverse LSF weights or their respective corresponding LSF weights, inverse weighting the residual LSF vector by the determined inverse LSF weights, and calculating the LPC filter based on the inversely-weighted residual LSF vector and the first-stage approximation of the LSF vector. The present disclosure further relates to corresponding methods and storage media.