An audio signal encoding method includes obtaining a current frame of an audio signal, where the current frame includes a high frequency band signal and a low frequency band signal; obtaining a parameter of bandwidth extension of the current frame based on the high frequency band signal, the low frequency band signal, and configuration information of the bandwidth extension; obtaining tile information, where the tile information indicates a first frequency range in which tonal component detection needs to be performed on the high frequency band signal; performing tonal component detection in the first frequency range to obtain information about a tonal component of the high frequency band signal; and performing bitstream multiplexing on the parameter of the bandwidth extension and the information of the tonal component to obtain a payload bitstream.

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.

Audio objects are associated with positional metadata. A received downmix signal comprises downmix channels that are linear combinations of one or more audio objects and are associated with respective positional locators. In a first aspect, the downmix signal, the positional metadata and frequency-dependent object gains are received. An audio object is reconstructed by applying the object gain to an upmix of the downmix signal in accordance with coefficients based on the positional metadata and the positional locators. In a second aspect, audio objects have been encoded together with at least one bed channel positioned at a positional locator of a corresponding downmix channel. The decoding system receives the downmix signal and the positional metadata of the audio objects. A bed channel is reconstructed by suppressing the content representing audio objects from the corresponding downmix channel on the basis of the positional locator of the corresponding downmix channel.

A signal processing method and device includes obtaining spectral coefficients of a current frame of an audio signal, in which N sub-bands of the current frame comprises at least one of the spectral coefficients. A total energy of M successive sub-bands of the N sub-bands, a total energy of K successive sub-bands of the N sub-bands, and an energy of a first sub-band are obtained to determine whether to modify original envelope values of the M sub-bands. When the original envelope values of the M sub-bands are modified, encoding bits are allocated to each of the N sub-bands according to the modified envelope values of the M sub-bands.

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 bit allocating method is provided that includes determining the allocated number of bits in decimal point units based on each frequency band so that a Signal-to-Noise Ratio (SNR) of a spectrum existing in a predetermined frequency band is maximized within a range of the allowable number of bits for a given frame; and adjusting the allocated number of bits based on each frequency band.

Encoding and decoding systems are described for the provision of high quality digital representations of audio signals with particular attention to the correct perceptual rendering of fast transients at modest sample rates. This is achieved by optimising downsampling and upsampling filters to minimise the length of the impulse response while adequately attenuating alias products that have been found perceptually harmful.

The present document relates to audio source coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), as well as to digital effect processors, e.g. exciters, where generation of harmonic distortion add brightness to the processed signal, and to time stretchers where a signal duration is prolonged with maintained spectral content. A system and method configured to generate a time stretched and/or frequency transposed signal from an input signal is described. The system comprises an analysis filterbank configured to provide an analysis subband signal from the input signal; wherein the analysis subband signal comprises a plurality of complex valued analysis samples, each having a phase and a magnitude. Furthermore, the system comprises a subband processing unit configured to determine a synthesis subband signal from the analysis subband signal using a subband transposition factor Q and a subband stretch factor S. The subband processing unit performs a block based nonlinear processing wherein the magnitude of samples of the synthesis subband signal are determined from the magnitude of corresponding samples of the analysis subband signal and a predetermined sample of the analysis subband signal. In addition, the system comprises a synthesis filterbank configured to generate the time stretched and/or frequency transposed signal from the synthesis subband signal.

A signal processing method and device includes obtaining spectral coefficients of a current frame of an audio signal, in which N sub-bands of the current frame comprises at least one of the spectral coefficients. A total energy of M successive sub-bands of the N sub-bands, a total energy of K successive sub-bands of the N sub-bands, and an energy of a first sub-band are obtained to determine whether to modify original envelope values of the M sub-bands. When the original envelope values of the M sub-bands are modified, encoding bits are allocated to each of the N sub-bands according to the modified envelope values of the M sub-bands.

This application relates to a speech coding method performed by a computer device. The method includes: obtaining initial frequency bandwidth feature information corresponding to a speech signal; performing feature compression on initial feature information corresponding to a second band in the initial frequency bandwidth feature information to obtain target feature information corresponding to a compressed band, a frequency interval of the second band being greater than a frequency interval of the compressed band; obtaining, based on the target feature information corresponding to the compressed band, a compressed speech signal corresponding to the speech signal; and coding the compressed speech signal to obtain coded speech data corresponding to the speech signal, a target sampling rate corresponding to the compressed speech signal being less than a sampling rate corresponding to the speech signal.