The present technology relates to an encoding device and an encoding method, a decoding device and a decoding method, and a program, configured to obtain a high quality audio with less encoding amount. A number-of-sections determining feature amount calculating circuit calculates a number-of-sections determining feature amount for determining the number of divisions to divide a process target section into continuous frame sections each including a frame for which the same estimation coefficient is selected, based on sub-band signals of a plurality of sub-bands constituting an input signal. A quasi-high frequency sub-band power difference calculating circuit determines the number of continuous frame sections in the process target section based on the number-of-sections determining feature amount, selects an estimation coefficient for obtaining a high frequency component of the input signal by estimation for each continuous frame section, and generates data including a coefficient index for obtaining the estimation coefficient. A high frequency encoding circuit encodes the obtained data, and generates high frequency encoded data. The present technology can be applied to an encoding device.

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.

The present document relates to audio encoding and decoding. In particular, the present document relates to audio coding schemes which make use of high frequency reconstruction (HFR) methods. A system configured to determine a master scale factor band table of a highband signal (105) of an audio signal is described. The highband signal (105) is to be generated from a lowband signal (101) of the audio signal using a high frequency reconstruction (HFR) scheme. The master scale factor band table is indicative of a frequency resolution of a spectral envelope of the highband signal (105).

An audio encoder for encoding an audio signal having a lower frequency band and an upper frequency band includes: a detector for detecting a peak spectral region in the upper frequency band of the audio signal; a shaper for shaping the lower frequency band using shaping information for the lower band and for shaping the upper frequency band using at least a portion of the shaping information for the lower band, wherein the shaper is configured to additionally attenuate spectral values in the detected peak spectral region in the upper frequency band; and a quantizer and coder stage for quantizing a shaped lower frequency band and a shaped upper frequency band and for entropy coding quantized spectral values from the shaped lower frequency band and the shaped upper frequency band.

Described herein are methods, apparatus and computer products for decoding an encoded MPEG-D USAC bitstream. Described herein are such methods, apparatus and computer products that reduce a computational complexity.

In accordance with an example embodiment of the present invention, disclosed is a method and an apparatus thereof for controlling a concealment method for a lost audio frame of a received audio signal. A method for a decoder of concealing a lost audio frame comprises detecting in a property of the previously received and reconstructed audio signal, or in a statistical property of observed frame losses, a condition for which the substitution of a lost frame provides relatively reduced quality. In case such a condition is detected, the concealment method is modified by selectively adjusting a phase or a spectrum magnitude of a substitution frame spectrum.

A device for signal processing includes a receiver and a high-band excitation signal generator. The receiver is configured to receive a parameter associated with a bandwidth-extended audio stream. The high-band excitation signal generator is configured to determine a value of the parameter. The high-band excitation signal generator is also configured to select, based on the value of the parameter, one of target gain information associated with the bandwidth-extended audio stream or filter information associated with the bandwidth-extended audio stream. The high-band excitation signal generator is further configured to generate a high-band excitation signal based on the one of the target gain information or the filter information.

An apparatus for processing an audio signal includes a configurable first audio signal processor for processing the audio signal in accordance with different configuration settings to obtain a processed audio signal, wherein the apparatus is adapted so that different configuration settings result in different sampling rates of the processed audio signal. The apparatus furthermore includes n analysis filter bank having a first number of analysis filter bank channels, a synthesis filter bank having a second number of synthesis filter bank channels, a second audio processor being adapted to receive and process an audio signal having a predetermined sampling rate, and a controller for controlling the first number of analysis filter bank channels or the second number of synthesis filter bank channels in accordance with a configuration setting.

An apparatus comprising means for: receiving values for sub-bands of a frame of an audio signal, the values comprising at least one azimuth value, at least one elevation value and at least one energy ratio value for each sub-band; determining an allocation of first number of bits to encode the values of the frame, wherein the first number of bits are fixed; encoding the at least one energy ratio value for a frame based on a defined allocation of a second number of bits from the first number of bits; encoding the at least one azimuth value and/or at least one elevation value of the frame based on a defined allocation of a third number of bits from the first number of bits, wherein the third number of bits is variably distributed on a sub-band-by-sub-band basis.

Methods, systems, and apparatus, including computer programs encoded on computer storage media, for coding speech using neural networks. One of the methods includes obtaining a bitstream of parametric coder parameters characterizing spoken speech; generating, from the parametric coder parameters, a conditioning sequence; generating a reconstruction of the spoken speech that includes a respective speech sample at each of a plurality of decoder time steps, comprising, at each decoder time step: processing a current reconstruction sequence using an auto-regressive generative neural network, wherein the auto-regressive generative neural network is configured to process the current reconstruction to compute a score distribution over possible speech sample values, and wherein the processing comprises conditioning the auto-regressive generative neural network on at least a portion of the conditioning sequence; and sampling a speech sample from the possible speech sample values.