Abstract: A method for providing media content within a media distribution network. The method comprises transforming source media content into an interim format, thereby providing transformed content. Furthermore, the method comprises storing the transformed content on at least one core storage unit. In addition, the method comprises receiving a request for the source media content from a client. The method further comprises encoding the transformed content or intermediate coded content derived therefrom into encoded content suitable for transmission over a core network and/or an edge network, as well as sending the encoded content via the core network and/or the edge network to the client.

Abstract: Described herein is a method of waveform decoding, the method including the steps of: (a) receiving, by a waveform decoder, a bitstream including a finite bitrate representation of a source signal; (b) waveform decoding the finite bitrate representation of the source signal to obtain a waveform approximation of the source signal; (c) providing the waveform approximation of the source signal to a generative model that implements a probability density function, to obtain a probability distribution for a reconstructed signal of the source signal; and (d) generating the reconstructed signal of the source signal based on the probability distribution. Described are further a method and system for waveform coding and a method of training a generative model.

Abstract: A method for providing media content within a media distribution network. The method comprises transforming source media content into an interim format, thereby providing transformed content. Furthermore, the method comprises storing the transformed content on at least one core storage unit. In addition, the method comprises receiving a request for the source media content from a client. The method further comprises encoding the transformed content or intermediate coded content derived therefrom into encoded content suitable for transmission over a core network and/or an edge network, as well as sending the encoded content via the core network and/or the edge network to the client.

Abstract: Described herein is a method of decoding an audio or speech signal, the method including the steps of: (a) receiving, by a decoder, a coded bitstream including the audio or speech signal and conditioning information; (b) providing, by a bitstream decoder, decoded conditioning information in a format associated with a first bitrate; (c) converting, by a converter, the decoded conditioning information from the format associated with the first bitrate to a format associated with a second bitrate; and (d) providing, by a generative neural network, a reconstruction of the audio or speech signal according to a probabilistic model conditioned by the conditioning information in the format associated with the second bitrate. Described are further an apparatus for decoding an audio or speech signal, a respective encoder, a system of the encoder and the apparatus for decoding an audio or speech signal as well as a respective computer program product.

Abstract: The present document describes a method (900) for establishing control information for a control policy of a client (102) for streaming data (103) from at least one server (101, 701). The method (900) comprises performing (901) a message passing process between a server agent of the server (101, 701) and a client agent of the client (102), in order to iteratively establish control information. Furthermore, the method (900) comprises generating (902) a convergence event for the message passing process to indicate that the control information has been established.

Abstract: The present document describes a method (900) for establishing control information for a control policy of a client (102) for streaming data (103) from at least one server (101, 701). The method (900) comprises performing (901) a message passing process between a server agent of the server (101, 701) and a client agent of the client (102), in order to iteratively establish control information. Furthermore, the method (900) comprises generating (902) a convergence event for the message passing process to indicate that the control information has been established.

Abstract: A method for providing media content within a media distribution network. The method comprises transforming source media content into an interim format, thereby providing transformed content. Furthermore, the method comprises storing the transformed content on at least one core storage unit. In addition, the method comprises receiving a request for the source media content from a client. The method further comprises encoding the transformed content or intermediate coded content derived therefrom into encoded content suitable for transmission over a core network and/or an edge network, as well as sending the encoded content via the core network and/or the edge network to the client.

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 audio codec system which is particularly well suited for voice encoding/decoding. A transform-based speech encoder is configured to encode a speech signal into a bitstream is described. A speech decoder configured to decode audio signals from a bitstream is further described.

Abstract: A method for performing inter-channel encoding of a multi-channel audio signal comprising channel signals for N channels, with N being an integer, with N>1, is described. The method comprises determining a basic graph comprising the N channels as nodes and comprising directed edges between at least some of the N channels. Furthermore, the method comprises determining an inter-channel coding graph from the basic graph, such that the inter-channel coding graph is a directed acyclic graph, and such that a cumulated a cumulated cost of the signals of the nodes of the inter-channel coding graph is reduced.

Abstract: Systems, methods, and storage media for audio processing adapted to voice encoding/decoding are disclosed. In some implementations, a system is adapted to determine prediction coefficients based on coefficient data included within a bitstream to determine quantized prediction coefficients, the coefficient data including one or more model parameters indicating at least one attribute of a signal model. The system inversely quantizes the quantized prediction coefficients to determine dequantized prediction coefficients. The system then determines multiple spectral energy values for multiple corresponding frequency bands based on the dequantized prediction coefficients.

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 performing inter-channel encoding of a multi-channel audio signal comprising channel signals for N channels, with N being an integer, with N>1, is described. The method comprises determining a basic graph comprising the N channels as nodes and comprising directed edges between at least some of the N channels. Furthermore, the method comprises determining an inter-channel coding graph from the basic graph, such that the inter-channel coding graph is a directed acyclic graph, and such that a cumulated a cumulated cost of the signals of the nodes of the inter-channel coding graph is reduced.

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 audio codec system which is particularly well suited for voice encoding/decoding. A transform-based speech encoder is configured to encode a speech signal into a bitstream is described. A speech decoder configured to decode audio signals from a bitstream is further described.

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 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: 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 control section (1009) receives signaling (S) indicating one of at least two coding formats (F1, F2, F3) of an M-channel audio signal (L, LS, LB, TFL, TBL), the coding formats corresponding to different partitions of the channels of the audio signal into respective first and second groups (601, 602), wherein, in the indicated coding format, first and second channels (L1, L2) of a downmix signal correspond to linear combinations of the first and second groups, respectively; and a decoding section (900) reconstructs the audio signal based on the downmix signal and associated upmix parameters (?L). In the decoding section: a decorrelation input signal (D1, D2, D3) is determined based on the downmix signal and the indicated coding format; and wet and dry upmix coefficients, controlling linear mappings of the downmix signal and a decorrelated signal, generated based on the decorrelation input signal, are determined based on the upmix parameters and the indicated coding format.

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: Disclosed is a system and computer program product of encoding audio content and corresponding method. The method includes determining a characteristic of the audio content, the characteristic of the audio content including at least one of a type or a property of the audio content. Also the method includes classifying the audio content based on the characteristic of the audio content and determining probabilities for multiple predefined audio coding symbols associated with the audio content by calculating a probability for each of the audio coding symbols based on the result of the classification, the probability for an audio coding symbol indicating a frequency at which the audio coding symbol occurs in the audio content. Further, the method encoded the audio content based on the audio coding symbols and the corresponding probabilities to obtain a code value, the code value representing a compression coding format of the audio content.

Abstract: The present document relates to multichannel audio coding and more precisely to techniques for discrete multichannel audio encoding and decoding. In particular, the present document relates to systems and method for coding soundfields. An audio encoder (200) configured to encode a frame of a soundfield signal (110) comprising a plurality of audio signals is described. The audio encoder (200) comprises a transform determination unit (203, 204) configured to determine an energy-compacting orthogonal transform (V) based on the frame of the soundfield signal (110). Furthermore, the encoder (200) comprises a transform unit (202) configured to apply the energy-compacting orthogonal transform (V) to the frame of the soundfield signal (110), and configured to provide a frame of a rotated soundfield signal (112) comprising a plurality of rotated audio signals (E1, E2, E3).