A stereo signal encoding method includes performing spectrum broadening on a quantized line spectral frequency (LSF) parameter of a primary channel signal in a current frame in a stereo signal to obtain a spectrum-broadened LSF parameter of the primary channel signal, determining a prediction residual of an LSF parameter of a secondary channel signal in the current frame based on an original LSF parameter of the secondary channel signal and the spectrum-broadened LSF parameter of the primary channel signal, and performing a quantization on the prediction residual of the LSF parameter of the secondary channel signal.

Various embodiments provide a method and an apparatus for determining a weighting factor during stereo signal encoding. In those embodiments, a parameter value corresponding to the encoding mode of the to-be-encoded signal is determining based on an encoding mode of a to-be-encoded signal in a stereo signal and a correspondence between an encoding mode and a parameter value. Based on the determined parameter value and an energy spectrum of a linear prediction filter corresponding to an original line spectral frequency parameter of the to-be-encoded signal is a weighting factor for calculating a distance between the original line spectral frequency parameter and a target original line spectral frequency parameter is calculated.

An encoding method includes determining a target adaptive broadening factor based on a quantized line spectral frequency (LSF) parameter of a primary channel signal in a current frame and an LSF parameter of a secondary channel signal in the current frame, and writing the quantized LSF parameter of the primary channel signal in the current frame and the target adaptive broadening factor into a bitstream.

The disclosure provides a speech processing method and a device thereof. The method includes: acquiring a speech sampling signal frame in a mixed-excitation linear prediction (MELP) speech coding system and estimating signal quality of the speech sampling signal frame; determining, based on the signal quality, a specific linear prediction coding (LPC) order used by an LPC circuit; controlling the LPC circuit to convert the speech sampling signal frame into a line spectrum pair parameter based on the specific LPC order; replacing a speech signal spectrum of the speech sampling signal frame with the line spectrum pair parameter to generate a predicted speech signal; and performing a speech coding operation and a signal synthesizing operation of the MELP speech coding system based on the predicted speech signal.

A User Equipment (UE) is operative to generate CN (Comfort Noise) control parameters, e.g., as part of audio-decoding processing by the UE. A buffer of a predetermined size implemented in the UE is configured to store CN parameters for SID (Silence Insertion Descriptor) frames and active hangover frames. Processing circuitry of the UE is configured to determine a CN parameter subset relevant for SID frames based on the age of the stored CN parameters and on residual energies, and use the determined CN parameter subset to determine CN control parameters for a first SID frame following an active signal frame.

A User Equipment (UE) is operative to generate CN (Comfort Noise) control parameters, e.g., as part of audio-decoding processing by the UE. A buffer of a predetermined size implemented in the UE is configured to store CN parameters for SID (Silence Insertion Descriptor) frames and active hangover frames. Processing circuitry of the UE is configured to determine a CN parameter subset relevant for SID frames based on the age of the stored CN parameters and on residual energies, and use the determined CN parameter subset to determine CN control parameters for a first SID frame following an active signal frame.

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.

An apparatus for decoding an audio signal includes a receiving interface, wherein the receiving interface is configured to receive a first frame and a second frame. Moreover, the apparatus includes a noise level tracing unit for determining noise level information being represented in a tracing domain. Furthermore, the apparatus includes a first reconstruction unit for reconstructing a third audio signal portion of the audio signal depending on the noise level information and a second reconstruction unit for reconstructing a fourth audio signal portion depending on noise level information being represented in the second reconstruction domain.

An apparatus for decoding an audio signal includes a receiving interface, wherein the receiving interface is configured to receive a first frame and a second frame. Moreover, the apparatus includes a noise level tracing unit for determining noise level information being represented in a tracing domain. Furthermore, the apparatus includes a first reconstruction unit for reconstructing a third audio signal portion of the audio signal depending on the noise level information and a second reconstruction unit for reconstructing a fourth audio signal portion depending on noise level information being represented in the second reconstruction domain.

A technology of accurately coding and decoding coefficients which are convertible into linear prediction coefficients even for a frame in which the spectrum variation is great while suppressing an increase in the code amount as a whole is provided. A coding device includes: a first coding unit that obtains a first code by coding coefficients which are convertible into linear prediction coefficients of more than one order; and a second coding unit that obtains a second code by coding at least quantization errors of the first coding unit if (A1) an index Q commensurate with how high the peak-to-valley height of a spectral envelope is, the spectral envelope corresponding to the coefficients which are convertible into the linear prediction coefficients of more than one order, is larger than or equal to a predetermined threshold value Th1 and/or (B1) an index Q commensurate with how short the peak-to-valley height of the spectral envelope is, is smaller than or equal to a predetermined threshold value Th1.