Abstract: An encoding system 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, a decorrelating section outputs, based on the downmix signal, an (N?1)-channel decorrelated signal (Z); a dry upmix section maps the downmix signal linearly in accordance with dry upmix coefficients (C) determined based on the dry upmix parameters; a wet upmix section 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 combines outputs from the upmix sections to obtain a reconstructed signal ({circumflex over (X)}) corresponding to the signal to be reconstructed.

Abstract: The present disclosure relates to the field audio coding, an in particular to an audio decoder having at least two decoding modes, and associated decoding methods and decoding software for such audio decoder. In one of the decoding modes, at least one dynamic audio object is mapped to a set of static audio objects, the set of static audio objects corresponding to a predefined speaker configuration. The present disclosure further relates to a corresponding audio encoder, and associated encoding methods and encoding software for such audio encoder.

Abstract: Described herein is a method of low-bitrate coding of audio data and generating enhancement metadata for controlling audio enhancement of the low-bitrate coded audio data at a decoder side, including the steps of: (a) core encoding original audio data at a low bitrate to obtain encoded audio data; (b) generating enhancement metadata to be used for controlling a type and/or amount of audio enhancement at the decoder side after core decoding the encoded audio data; and (c) outputting the encoded audio data and the enhancement metadata. Described is further an encoder configured to perform said method. Described is moreover a method for generating enhanced audio data from low-bitrate coded audio data based on enhancement metadata and a decoder configured to perform said method.

Abstract: Methods for generating an object based audio program, renderable in a personalizable manner, and including a bed of speaker channels renderable in the absence of selection of other program content (e.g., to provide a default full range audio experience). Other embodiments include steps of delivering, decoding, and/or rendering such a program. Rendering of content of the bed, or of a selected mix of other content of the program, may provide an immersive experience. The program may include multiple object channels (e.g., object channels indicative of user-selectable and user-configurable objects), the bed of speaker channels, and other speaker channels. Another aspect is an audio processing unit (e.g., encoder or decoder) configured to perform, or which includes a buffer memory which stores at least one frame (or other segment) of an object based audio program (or bitstream thereof) generated in accordance with, any embodiment of the method.

Abstract: A reference picture information decoding unit (13) omits decoding of a reference list sorting presence or absence flag and/or a reference list sorting order based on the number of current picture referable pictures.

Abstract: The present document relates to audio source coding systems. In particular, the present document relates to audio source coding systems which make use of linear prediction in combination with a filterbank. A method for estimating a first sample (615) of a first subband signal in a first subband of an audio signal is described. The first subband signal of the audio signal is determined using an analysis filterbank (612) comprising a plurality of analysis filters which provide a plurality of subband signals in a plurality of subbands from the audio signal, respectively.

Abstract: Encoding and decoding devices for encoding the channels of an audio system having at least four channels are disclosed. The decoding device has a first stereo decoding component which subjects a first pair of input channels to a first stereo decoding, and a second stereo decoding component which subjects a second pair of input channels to a second stereo decoding. The results of the first and second stereo decoding components are crosswise coupled to a third and a fourth stereo decoding component which each performs stereo decoding on one channel resulting from the first stereo decoding component, and one channel resulting from the second stereo decoding component.

Abstract: A method for decoding an encoded audio bitstream is disclosed. The method includes receiving the encoded audio bitstream and decoding the audio data to generate a decoded lowband audio signal. The method further includes extracting high frequency reconstruction metadata and filtering the decoded lowband audio signal with an analysis filterbank to generate a filtered lowband audio signal. The method also includes extracting a flag indicating whether either spectral translation or harmonic transposition is to be performed on the audio data and regenerating a highband portion of the audio signal using the filtered lowband audio signal and the high frequency reconstruction metadata in accordance with the flag. The high frequency regeneration is performed as a post-processing operation with a delay of 3010 samples per audio channel.

Abstract: Described herein is a method for setting up a decoder for generating processed audio data from an audio bitstream, the decoder comprising a Generator of a Generative Adversarial Network, GAN, for processing of the audio data, wherein the method includes the steps of (a) pre-configuring the Generator for processing of audio data with a set of parameters for the Generator, the parameters being determined by training, at training time, the Generator using the full concatenated distribution; and (b) pre-configuring the decoder to determine, at decoding time, a truncation mode for modifying the concatenated distribution and to apply the determined truncation mode to the concatenated distribution. Described are further a method of generating processed audio data from an audio bitstream using a Generator of a Generative Adversarial Network, GAN, for processing of the audio data and a respective apparatus. Moreover, described are also respective systems and computer program products.

Abstract: A method of audio processing includes performing spatial analysis on a binaural signal to estimate level differences and phase differences characteristic of a binaural filter of the binaural signal, performing object extraction on the binaural audio signal using the estimated level and phase differences to generate a left/right main component signal and a left/right residual component signal. The system may process the left/right main and left/right residual components differently using different object processing parameters for e.g. repositioning, equalization, compression, upmixing, channel remapping or storage to generate a processed binaural signal that provides an improved listening experience. Repositioning may be based on head tracking sensor data.

Abstract: An apparatus and method are disclosed for processing an audio signal. The apparatus includes an input interface, a digital filterbank having an analysis part and a synthesis part, a first phase shifter, a spectral envelope adjuster, a second phase shifter, and an output interface. The first phase shifter and the second phase shifter reduce a complexity of the digital filterbank, which includes both analysis and synthesis filters that are complex-exponential modulated versions of a prototype filter.

Abstract: The present disclosure relates to a method of processing audio content including directivity information for at least one sound source, the directivity information comprising a first set of first directivity unit vectors representing directivity directions and associated first directivity gains. The disclosure further relates to corresponding methods of encoding and decoding audio content including directivity information for at least one sound source.

Abstract: The invention provides an efficient implementation of cross-product enhanced high-frequency reconstruction (HFR), wherein a new component at frequency Q?+r?0 is generated on the basis of existing components at ? and ?+?0. The invention provides a block-based harmonic transposition, wherein a time block of complex subband samples is processed with a common phase modification. Superposition of several modified samples has the net effect of limiting undesirable intermodulation products, thereby enabling a coarser frequency resolution and/or lower degree of oversampling to be used. In one embodiment, the invention further includes a window function suitable for use with block-based cross-product enhanced HFR. A hardware embodiment of the invention may include an analysis filter bank, a subband processing unit configurable by control data and a synthesis filter bank.

Abstract: 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.

Abstract: An audio communication endpoint receives a bitstream containing spectral components representing spectral content of an audio signal, wherein the spectral components relate to a first range extending up to a first break frequency, above which any spectral components are unassigned. The endpoint adapts the received bitstream in accordance with a second range extending up to a second break frequency by removing spectral components or adding neutral-valued spectral components relating to a range between the first and second break frequencies. The endpoint then attenuates spectral content in a neighbourhood of the least of the first and second break frequencies for thereby achieving a gradual spectral decay. After this, reconstructing the audio signal is reconstructed by an inverse transform operating on spectral components relating to said second range in the adapted and attenuated received bitstream. At small computational expense, the endpoint may to adapt to different sample rates in received bitstreams.

Abstract: 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.

Abstract: Some disclosed methods involve encoding or decoding directional audio data. Some encoding methods may involve receiving a mono audio signal corresponding to an audio object and a representation of a radiation pattern corresponding to the audio object. The radiation pattern may include sound levels corresponding to plurality of sample times, a plurality of frequency bands and a plurality of directions. The methods may involve encoding the mono audio signal and encoding the source radiation pattern to determine radiation pattern metadata. Encoding the radiation pattern may involve determining a spherical harmonic transform of the representation of the radiation pattern and compressing the spherical harmonic transform to obtain encoded radiation pattern metadata.

Abstract: A method is provided for coding at least one image split up into partitions, a current partition to be coded containing data, at least one data item of which is allotted a sign. The coding method includes, for the current partition, the following steps: calculating the value of a function representative of the data of the current partition with the exclusion of the sign; comparing the calculated value with a predetermined value of the sign; as a function of the result of the comparison, modifying or not modifying at least one of the data items of the current partition, in the case of modification, coding the at least one modified data item.

Abstract: The positions of a plurality of speakers at a media consumption site are determined. Audio information in an object-based format is received. Gain adjustment value for a sound content portion in the object-based format may be determined based on the position of the sound content portion and the positions of the plurality of speakers. Audio information in a ring-based channel format is received. Gain adjustment value for each ring-based channel in a set of ring-based channels may be determined based on the ring to which the ring-based channel belongs and the positions of the speakers at a media consumption site.

Abstract: A method (900) for rendering audio in a virtual reality rendering environment (180) is described. The method (900) comprises rendering (901) an origin audio signal of an origin audio source (113) of an origin audio scene (111) from an origin source position on a sphere (114) around a listening position (201) of a listener (181). Furthermore, the method (900) comprises determining (902) that the listener (181) moves from the listening position (201) within the origin audio scene (111) to a listening position (202) within a different destination audio scene (112). In addition, the method (900) comprises applying (903) a fade-out gain to the origin audio signal to determine a modified origin audio signal, and rendering (903) the modified origin audio signal of the origin audio source (113) from the origin source position on the sphere (114) around the listening position (201, 202).