The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals, which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

An apparatus for processing an input audio signal relies on a cascade of filterbanks, the cascade having a synthesis filterbank for synthesizing an audio intermediate signal from the input audio signal, the input audio signal being represented by a plurality of first subband signals generated by an analysis filterbank, wherein a number of filterbank channels of the synthesis filterbank is smaller than a number of channels of the analysis filterbank. The apparatus furthermore has a further analysis filterbank for generating a plurality of second subband signals from the audio intermediate signal, wherein the further analysis filterbank has a number of channels being different from the number of channels of the synthesis filterbank, so that a sampling rate of a subband signal of the plurality of second subband signals is different from a sampling rate of a first subband signal of the plurality of first subband signals.

This application provides example psychoacoustics-based audio encoding methods and apparatuses. One example method includes receiving audio data. The audio data can be decoded. Auditory feature information of a user can be obtained, where the auditory feature information includes at least one of the following: personal information, listening test result information, or frequency response curve information. A psychoacoustics model parameter of the user can be calculated based on the auditory feature information of the user, where the psychoacoustics model parameter includes at least one of the following: an intra-band masking parameter, a slope of a low-frequency inter-band masking line, a slope of a high-frequency inter-band masking line, or a human ear quiet threshold curve. The decoded audio data can be encoded based on the psychoacoustics model parameter of the user.

An apparatus comprising means configured to: generate spatial audio signal directional metadata parameters for a block of time-frequencies; generate encoded spatial audio signal directional metadata parameters (108) for a block of time-frequencies based on a first quantization resolution (203); compare a number of bits used for the encoded spatial audio signal directional parameters (108) for the block of time-frequencies based on the first quantization resolution against a determined number of bits; output or store the encoded spatial audio signal directional metadata parameters for a block of time-frequencies (108) based on a first quantization resolution when the number of bits used for the encoded spatial audio signal directional parameters for the block of time-frequencies (108) based on the first quantization resolution is less than a determined number of bits (217); generate encoded spatial audio signal directional metadata parameters (108) for the block of time-frequencies based on a second quantization resolution when the number of bits used for the encoded spatial audio signal directional parameters for the block of time-frequencies (108) based on the first quantization resolution is more than the determined number of bits and a difference between the determined number of bits and the number of bits used for the encoded spatial audio signal directional parameters (108) for the block of time-frequencies based on the first quantization resolution is less than a determined number of bits is within a determined threshold (217); generate encoded spatial audio signal directional metadata parameters (108) for the block of time-frequencies based on a third quantization resolution when the number of bits used for the encoded spatial audio signal directional parameters (108) for the block of time-frequencies based on the first quantization resolution is more than the determined number of bits and the difference between the determined number of bits and the number of bits used for the encoded spatial audio signal directional parameters (108) for the block of time-frequencies based on the first quantization resolution is greater than the determined threshold, wherein the third quantization resolution is determined such that a number of bits used for the encoded spatial audio signal directional parameters for the block of time-frequencies based on the third quantization resolution is always equal to or less than the determined number of bits (217).

This signal processing device comprises: an acquisition unit for acquiring an acoustic signal; a measurement unit for measuring an acoustic level of the acoustic signal for every one of first frequency bands, which are a plurality of frequency bands of a preset first bandwidth; a calculation unit that, on the basis of the plurality of acoustic levels of the first frequency bands, identifies an acoustic feature quantity indicating the separation degree from normal acoustic levels of second frequency bands, which are a plurality of frequency bands of a second bandwidth that is wider than the first bandwidth; a first determination unit for determining whether the acoustic levels measured for every one of the first frequency bands are a first threshold value or greater; and a second determination unit for determining whether the acoustic feature quantity is a second threshold value or greater.

An audio signal decoding device includes a non-transitory memory storage stores audio data in a form of a bitstream; and an audio decoder, by which a first spectral coefficient of a first sub-band of a current frame of an audio signal by decoding the bitstream is obtained; a first average quantity of allocated bits per spectral coefficient of the first sub-band is obtained; a first noise filling gain for the first sub-band is obtained when the first average quantity is less than a threshold; a second spectral coefficient is reconstructed according to the first noise filling gain; a frequency domain audio signal is obtained according to the first spectral coefficient and the second spectral coefficient; and a time domain audio signal is generated according to the frequency domain signal.

A method for mixing, processing and enhancing signals using signal decomposition is presented. A method for improving sorting of decomposed signal parts using cross-component similarity is also provided.

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 exemplary hearing device is configured to receive, from a source, a data packet, the data packet including a plurality of frames including a first frame and a second frame. The hearing device determines that the data packet has an invalid checksum. The hearing device accesses, in response to the determining that the data packet has the invalid checksum, a first frame checksum for the first frame and a second frame checksum for the second frame. The hearing device determines that the first frame checksum is invalid and that the second frame checksum is valid. The hearing device discards, based on the first frame checksum being invalid, the first frame and stores, based on the second frame checksum being valid, the second frame.

The present disclosure provides methods, devices and computer program products for encoding and decoding a stereo audio signal based on an input signal. According to the disclosure, a hybrid approach of using both parametric stereo coding and a discrete representation of the stereo audio signal is used which may improve the quality of the encoded and decoded audio for certain bitrates.