An apparatus comprising means configured to obtain direction parameter values (108) associated with at least two time-frequency parts (202) of at least one audio signal (102); and encode the obtained direction parameter values based on a codebook (206), wherein the codebook comprises two or more quantization levels arranged such that a first quantization level comprises a first set of quantization values, and a second or succeeding quantization level comprises a second or further set of quantization values and preceding quantization level quantization values.

Provided are an audio reconstruction method and device for providing improved sound quality by reconstructing a decoding parameter or an audio signal obtained from a bitstream, by using machine learning. The audio reconstruction method includes obtaining a plurality of decoding parameters of a current frame by decoding a bitstream, determining characteristics of a second parameter included in the plurality of decoding parameters and associated with a first parameter, based on the first parameter included in the plurality of decoding parameters, obtaining a reconstructed second parameter by applying a machine learning model to at least one of the plurality of decoding parameters, the second parameter, and the characteristics of the second parameter, and decoding an audio signal, based on the reconstructed second parameter.

The present disclosure provides methods, devices and computer program products for encoding and decoding of a vector of parameters in an audio coding system. The disclosure further relates to a method and apparatus for reconstructing an audio object in an audio decoding system. According to the disclosure, a modulo differential approach for coding and encoding a vector of a non-periodic quantity may improve the coding efficiency and provide encoders and decoders with less memory requirements. Moreover, an efficient method for encoding and decoding a sparse matrix is provided.

According to an aspect of the present invention, a method for reconstructing an audio signal having a baseband portion and a highband portion is disclosed. The method includes obtaining a decoded baseband audio signal by decoding an encoded audio signal and obtaining a plurality of subband signals by filtering the decoded baseband audio signal. The method further includes generating a high-frequency reconstructed signal by copying a number of consecutive subband signals of the plurality of subband signals and obtaining an envelope adjusted high-frequency signal. The method further includes generating a noise component based on a noise parameter. Finally, the method includes adjusting a phase of the high-frequency reconstructed signal and obtaining a time-domain reconstructed audio signal by combining the decoded baseband audio signal and the combined high-frequency signal to obtain a time-domain reconstructed audio signal.

A very coarse quantization exceeding the measure determined by the masking threshold without or only very little quality losses is enabled by quantizing not immediately the prefiltered signal, but a prediction error obtained by forward-adaptive prediction of the prefiltered signal. Due to the forward adaptivity, the quantizing error has no negative effect on the prediction on the decoder side.

An encoding device is provided with: a quantizing circuit which generates quantization parameters including first information on a vector quantization codebook, and second information on code vectors included in the codebook; and a control circuit which employs the second number of bits based on the difference between the first number of bits available for encoding of a sub-vector in the vector quantization, and the number of bits for the sub-vector quantization parameters, to control encoding of the first information with respect to the sub-vector.

The present technology relates to a signal processing apparatus and method, and a program that make it possible to reduce an arithmetic operation amount.

The signal processing apparatus performs, on the basis of audio object mute information indicative of whether or not a signal of an audio object is a mute signal, at least either one of a decoding process or a rendering process of an object signal of the audio object. The present technology can be applied to a signal processing apparatus.

The disclosure provides a packet loss recovery method for an audio data packet an electronic device and a storage medium. The method includes: receiving an audio data packet sent by a vehicle-mounted terminal, and identifying a discarded first sampling point set in response to detecting packet loss; obtaining a second sampling point set and a third sampling point set each adjacent to the first sampling point set, in which the second sampling point set is prior to the first sampling point set, the third sampling point set is behind the first sampling point set; and generating target audio data of the first sampling points based on first audio data sampled at the second sampling points and second audio data sampled at the third sampling points, and inserting the target audio data at sampling positions of the first sampling points.

The application relates to HFR (High Frequency Reconstruction/Regeneration) of audio signals. In particular, the application relates to a method and system for performing HFR of audio signals having large variations in energy level across the low frequency range which is used to reconstruct the high frequencies of the audio signal. A system configured to generate a plurality of high frequency subband signals covering a high frequency interval from a plurality of low frequency subband signals is described. The system comprises means for receiving the plurality of low frequency subband signals; means for receiving a set of target energies, each target energy covering a different target interval within the high frequency interval and being indicative of the desired energy of one or more high frequency subband signals lying within the target interval; means for generating the plurality of high frequency subband signals from the plurality of low frequency subband signals and from a plurality of spectral gain coefficients associated with the plurality of low frequency subband signals, respectively; and means for adjusting the energy of the plurality of high frequency subband signals using the set of target energies.

An illustrative frequency-domain encoder system transforms time-domain data representative of a content instance into frequency-domain data representative of the content instance. The frequency-domain data includes a plurality of complex coefficients each representing different frequency components of a plurality of frequency components incorporated by the content instance. The frequency-domain encoder system generates a frequency-domain data container that includes the complex coefficients of the frequency-domain data and metadata descriptive of the frequency-domain data. Additionally, within the frequency-domain data container, the frequency-domain encoder system integrates the complex coefficients of the frequency-domain data with timing data representative of a time-dependent feature of the content instance. Corresponding systems and methods are also disclosed.