Abstract: The present document relates to a method of layered encoding of a frame of a compressed higher-order Ambisonics, HOA, representation of a sound or sound field. The compressed HOA representation comprises a plurality of transport signals. The method comprises assigning the plurality of transport signals to a plurality of hierarchical layers, the plurality of layers including a base layer and one or more hierarchical enhancement layers, generating, for each layer, a respective HOA extension payload including side information for parametrically enhancing a reconstructed HOA representation obtainable from the transport signals assigned to the respective layer and any layers lower than the respective layer, assigning the generated HOA extension payloads to their respective layers, and signaling the generated HOA extension payloads in an output bitstream.

Abstract: An encoding system (400) encodes an N-channel audio signal (X), wherein N?3, as a single-channel downmix signal (Y) together with dry and wet upmix parameters ({tilde over (C)}, {tilde over (P)}). In a decoding system (200), a decorrelating section (101) outputs, based on the downmix signal, an (N?1)-channel decorrelated signal (Z); a dry upmix section (102) maps the downmix signal linearly in accordance with dry upmix coefficients (C) determined based on the dry upmix parameters; a wet upmix section (103) populates an intermediate matrix based on the wet upmix parameters and knowing that the intermediate matrix belongs to a predefined matrix class, obtains wet upmix coefficients (P) by multiplying the intermediate matrix by a predefined matrix, and maps the decorrelated signal linearly in accordance with the wet upmix coefficients; and a combining section (104) combines outputs from the upmix sections to obtain a reconstructed signal ({circumflex over (X)}) corresponding to the signal to be reconstructed.

Abstract: The present document relates to the determination of a 200 bitrate related to an encoded bitstream, and describes a method for determining an estimate of a bitrate of a bitstream comprising a sequence of frames comprising a varying number of bitsand corresponding to excerpts of an audio and/or video signal. At least two frames of the sequence of frames comprise a parameter indicative of a processing delay for the corresponding frame. The method comprises determining: a total number of bits for a subsequence of frames from the bitstream; a corrected number 201 of frames based on a number of frames comprised within the subsequence and the parameters of at least two frames of the subsequence; and a lower bitrate bound and an upper bitrate bound of the bitrate based on the total number of bits, the corrected number of frames and a frame rate of the bitstream.

Abstract: On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (?) from the bitstream, where 1?m<n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

Abstract: Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content.

Abstract: An audio processing system (100) accepts an audio bitstream having one of a plurality of predefined audio frame rates. The system comprises a front-end component (110), which receives a variable number of quantized spectral components, corresponding to one audio frame in any of the predefined audio frame rates, and performs an inverse quantization according to predetermined, frequency-dependent quantization levels. The front-end component may be agnostic of the audio frame rate. The audio processing system further comprises a frequency-domain processing stage (120) and a sample rate converter (130), which provide a reconstructed audio signal sampled at a target sampling frequency independent of the audio frame rate. By its frame-rate adaptability, the system can be configured to operate frame-synchronously in parallel with a video processing system that accepts plural video frame rates.

Abstract: 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.

Abstract: The present disclosure provides methods, devices and computer program products for non-uniform quantization of parameters relating to parametric spatial coding of audio signals. The disclosure further relates to a method and apparatus for reconstructing an audio object in an audio decoding system taking the non-uniformly quantized parameters into account. According to the disclosure, such an approach renders it possible to reduce bit consumption without substantially reducing the quality of the reconstructed audio object.

Abstract: Methods and systems for improving coding decoding efficiency of video by providing a syntax modeler, a buffer, and a decoder. The syntax modeler may associate a first sequence of symbols with syntax elements. The buffer may store tables, each represented by a symbol in the first sequence, and each used to associate a respective symbol in a second sequence of symbols with encoded data. The decoder decodes the data into a bitstream using the second sequence retrieved from a table.

Abstract: A method for representing a second presentation of audio channels or objects as a data stream, the method comprising the steps of: (a) providing a set of base signals, the base signals representing a first presentation of the audio channels or objects; (b) providing a set of transformation parameters, the transformation parameters intended to transform the first presentation into the second presentation; the transformation parameters further being specified for at least two frequency bands and including a set of multi-tap convolution matrix parameters for at least one of the frequency bands.

Abstract: Higher Order Ambisonics (HOA) signals can be compressed by decomposition into a predominant sound component and a residual ambient component. The compressed representation comprises pre-dominant sound signals, coefficient sequences of the ambient component and side information. For efficiently combining HOA decompression and HOA rendering to obtain loudspeaker signals, combined rendering and decoding of the compressed HOA signal comprises perceptually decoding the perceptually coded portion and decoding the side information, without reconstructing HOA coefficient sequences. For reconstructing components of a first type, fading of coefficient sequences is not required, while for components of a second type fading is required. For each second type component, different linear operations are determined: one for coefficient sequences that in a current frame require no fading, one for those that require fading-in, and one for those that require fading-out.

Abstract: This disclosure falls into the field of audio coding, in particular it is related to the field of spatial audio coding, where the audio information is represented by multiple signals, where the signals may comprise audio channels or/and audio objects. In particular the disclosure provides a method and apparatus for reconstructing audio objects in an audio decoding system. Furthermore, this disclosure provides a method and apparatus for encoding such audio objects.

Abstract: The present invention relates to transposing signals in time and/or frequency and in particular to coding of audio signals. More particular, the present invention relates to high frequency reconstruction (HFR) methods including a frequency domain harmonic transposer. A method and system for generating a transposed output signal from an input signal using a transposition factor T is described. The system comprises an analysis window of length La, extracting a frame of the input signal, and an analysis transformation unit of order M transforming the samples into M complex coefficients. M is a function of the transposition factor T. The system further comprises a nonlinear processing unit altering the phase of the complex coefficients by using the transposition factor T, a synthesis transformation unit of order M transforming the altered coefficients into M altered samples, and a synthesis window of length Ls, generating a frame of the output signal.

Abstract: The present document relates an audio encoding and decoding system (referred to as an audio codec system). In particular, the present document relates to a transform-based audio codec system which is particularly well suited for voice encoding/decoding. A transform-based speech encoder (100, 170) configured to encode a speech signal into a bitstream is described. The encoder (100, 170) comprises a framing unit (101) configured to receive a set (132, 332) of blocks; wherein the set (132, 332) of blocks comprises a plurality of sequential blocks (131) of transform coefficients; wherein the plurality of blocks (131) is indicative of samples of the speech signal; wherein a block (131) of transform coefficients comprises a plurality of transform coefficients for a corresponding plurality of frequency bins (301).

Abstract: From an HOA signal representation (c(t)) of a sound field having an order of N and a number 0=(N+1)2 of coefficient sequences a mezzanine HOA signal representation (wMEZZ(t)) is generated that consists of an arbitrary number I<0 of virtual loudspeaker signals WMEZZ1(t), WMEZZ,2(t), . . . , WMEZZ,I(t). 0 directions are computed which are nearly uniformly distributed on the unit sphere. The mode vectors with respect to these directions are linearly weighted for constructing a matrix, of which the pseudo-inverse is used for multiplying the HOA signal representation (c(t)) in order to form (11) the mezzanine HOA signal representation (WMEZZ(t)).

Abstract: Techniques for low bit rate parametric encoding of haptic-tactile signals. The techniques encompass a parametric encoding method. The parametric encoding method includes the steps of: for at least one frame of a plurality of frames of a source haptic-tactile signal, representing the source haptic-tactile signal in the frame as a set of parameters and according to a functional representation; and including the set of parameters in a bit stream that encodes the source haptic-tactile signal. The functional representation is based on one of a set of orthogonal functionals, or polynomial approximation. For example, the functional representation can be based on one of Chebyshev functionals of the first kind through order n, Chebyshev functionals of the second kind through order n, or k-th order polynomial approximation.

Abstract: A method for performing audio inpainting, wherein missing portions in an input audio signal are recovered and a recovered audio signal is obtained, comprises computing a Short-Time Fourier Transform (STFT) on portions of the input audio signal, computing conditional expectations of the source power spectra of the input audio signal, wherein estimated source power spectra P(f, n, j) are obtained and wherein the variance tensor V and complex Short-Time Fourier Transform (STFT) coefficients of the input audio signals are used, iteratively re-calculating the variance tensor V from the estimated power spectra P(f, n, j) and re-calculating updated estimated power spectra P(f, n, j), computing an array of STFT coefficients ? from the resulting variance tensor V according to ?(f, n, j)=E{S(f, n, j)|x, Is, IL, V}, and converting the array of STFT coefficients ? to the time domain, wherein coefficients {tilde over (s)}1, {tilde over (s)}2, . . . , {tilde over (s)}j of the recovered audio signal are obtained.

Abstract: The present document relates an audio encoding and decoding system (referred to as an audio codec system). In particular, the present document relates to a transform-based audio codec system which is particularly well suited for voice encoding/decoding. A quantization unit configured to quantize a first coefficient of a block of coefficients is described. The block of coefficients comprises a plurality of coefficients for a plurality of corresponding frequency bins. The quantization unit is configured to provide a set of quantizers. The set of quantizers comprises a plurality of different quantizers associated with a plurality of different signal-to-noise ratios, referred to as SNR, respectively. The plurality of different quantizers includes a noise-filling quantizer; one or more dithered quantizers; and one or more un-dithered quantizers.

Abstract: A method for coding includes; segmenting an image into blocks; grouping blocks into a number of subsets; coding, using an entropy coding module, each subset, by associating digital information with symbols of each block of a subset, including, for the first block of the image, initializing state variables of the coding module; and generating a data sub-stream representative of at least one of the coded subsets of blocks. Where a current block is the first block to be coded of a subset, symbol occurrence probabilities for the first current block are determined based on those for a coded and decoded predetermined block of at least one other subset. Where the current block is the last coded block of the subset: writing, in the sub-stream representative of the subset, the entire the digital information associated with the symbols during coding of the blocks of the subset, and implementing the initializing sub-step.