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).

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: There is provided encoding and decoding methods for encoding and decoding of object based audio. An exemplary decoding method described is for reconstructing audio objects based on a data stream, wherein the data stream corresponds to a plurality of time frames, wherein the data stream comprises a plurality of side information instances, wherein the data stream further comprises, for each side information instance, transition data including two independently assignable portions which in combination define a point in time to begin a transition from a current reconstruction setting to a desired reconstruction setting specified by the side information instance, and a point in time to complete the transition.

Abstract: There is provided encoding and decoding methods for encoding and decoding of object based audio. An exemplary decoding method described is for reconstructing audio objects based on a data stream, wherein the data stream corresponds to a plurality of time frames, wherein the data stream comprises a plurality of side information instances, wherein the data stream further comprises, for each side information instance, transition data including two independently assignable portions which in combination define a point in time to begin a transition from a current reconstruction setting to a desired reconstruction setting specified by the side information instance, and a point in time to complete the transition.

Abstract: An encoding system (100) encodes a first (E1) and further (E2, E3) audio signals as a layered bitstream (B), wherein a quantizer for each frequency band of each signal is selected using a rate allocation rule based on signal-specific rate allocation data, a spectral envelope of the signal and a reference level (EnvE1Max), which is determined based on the spectral envelope of the first signal and is not necessarily included in the bitstream. Further disclosed is a decoding system for reconstructing the audio signals based on the bitstream. In embodiments, the bitstream has a basic layer (BE1), which contains data that enable decoding of the first audio signal, and a spatial layer (Bspatial) facilitating decoding of the further audio signal(s). In embodiments, the encoding system prepares the bitstream subject to a basic-layer bitrate constraint and a total bitrate constraint.

Abstract: A method and apparatus for performing an adaptive down-mixing of a multichannel audio signal comprising a number of input channels, wherein a signal adaptive transformation of said input channels is performed by multiplying the input channels with a downmix block matrix comprising a fixed block for providing a set of backward compatible primary channels and a signal adaptive block for providing a set of secondary channels.

Abstract: A method for encoding sound field signals includes allocating coding rate by application of a uniform criterion to all subbands of all signals in a joint process. An allocation criterion may be based on a comparison, in a given subband, between a spectral envelope of the signals to be encoded and a coding noise profile, wherein the noise profile may be a sum of a noise shape and a noise offset, which noise offset is computed on the basis of the coding bit budget. The rate allocation process may be combined with an energy-compacting orthogonal transform, for which there is proposed a parameterization susceptible of efficient coding and having adjustable directivity. In a further aspect, the invention provides a corresponding decoding method.

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 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).

Abstract: An encoding system (100) encodes a first (E1) and further (E2, E3) audio signals as a layered bitstream (B), wherein a quantizer for each frequency band of each signal is selected using a rate allocation rule based on signal-specific rate allocation data, a spectral envelope of the signal and a reference level (EnvE1Max), which is determined based on the spectral envelope of the first signal and is not necessarily included in the bitstream. Further disclosed is a decoding system for reconstructing the audio signals based on the bitstream. In embodiments, the bitstream has a basic layer (BE1), which contains data that enable decoding of the first audio signal, and a spatial layer (Bspatial) facilitating decoding of the further audio signal(s). In embodiments, the encoding system prepares the bitstream subject to a basic-layer bitrate constraint and a total bitrate constraint.

Abstract: In a method for reconstructing a source signal, which is encoded by a set of at least two descriptions, the method comprises: receiving a subset of the set of descriptions; reconstructing a reconstructed signal at an operating bitrate of a set of operating bitrates upon the basis of the subset of descriptions, the reconstructed signal having a second probability density, wherein the second probability density comprises a first statistical moment and a second statistical moment; and manipulating the reconstructed signal, wherein the reconstructed signal is manipulated such that, irrespective of the operating bitrate, a predetermined minimum similarity between a first statistical moment of a third probability density and a first statistical moment of a first probability density and between a second statistical moment of the third probability density and a second statistical moment of the first probability density is maintained.

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 (112) configured to quantize a first coefficient of a block (141) of coefficients is described. The block (141) of coefficients comprises a plurality of coefficients for a plurality of corresponding frequency bins (301). The quantization unit (112) is configured to provide a set (326, 327) of quantizers. The set (326, 327) of quantizers comprises a plurality of different quantizers (321, 322, 323) associated with a plurality of different signal-to-noise ratios, referred to as SNR, respectively. The plurality of different quantizers (321, 322, 323) includes a noise-filling quantizer (321); one or more dithered quantizers (322); and one or more un-dithered quantizers (323).

Abstract: In a method for reconstructing a source signal, which is encoded by a set of at least two descriptions, the method comprises: receiving a subset of the set of descriptions; reconstructing a reconstructed signal at an operating bitrate of a set of operating bitrates upon the basis of the subset of descriptions, the reconstructed signal having a second probability density, wherein the second probability density comprises a first statistical moment and a second statistical moment; and manipulating the reconstructed signal, wherein the reconstructed signal is manipulated such that, irrespective of the operating bitrate, a predetermined minimum similarity between a first statistical moment of a third probability density and a first statistical moment of a first probability density and between a second statistical moment of the third probability density and a second statistical moment of the first probability density is maintained.

Abstract: A method for encoding sound field signals includes allocating coding rate by application of a uniform criterion to all subbands of all signals in a joint process. An allocation criterion may be based on a comparison, in a given subband, between a spectral envelope of the signals to be encoded and a coding noise profile, wherein the noise profile may be a sum of a noise shape and a noise offset, which noise offset is computed on the basis of the coding bit budget. The rate allocation process may be combined with an energy-compacting orthogonal transform, for which there is proposed a parameterization susceptible of efficient coding and having adjustable directivity. In a further aspect, the invention provides a corresponding decoding method.

Abstract: In a method for reconstructing a source signal, which is encoded by a set of at least two descriptions, the method comprises: receiving a subset of the set of descriptions; reconstructing a reconstructed signal at an operating bitrate of a set of operating bitrates upon the basis of the subset of descriptions, the reconstructed signal having a second probability density, wherein the second probability density comprises a first statistical moment and a second statistical moment; and manipulating the reconstructed signal, wherein the reconstructed signal is manipulated such that, irrespective of the operating bitrate, a predetermined minimum similarity between the first statistical moment of the third probability density and the first statistical moment of the first probability density and between the second statistical moment of the third probability density and the second statistical moment of the first probability density is maintained.

Abstract: The invention relates to an estimator for estimating a probability distribution of a quantization index generated from a source coder encoding a source signal, into a sequence of quantization indices, the source signal being described by a signal model, the source coder providing a current quantization index and current side information, the estimator being configured to obtain auxiliary parameters based on a configuration of the source coder and the current available side information and the signal model, the estimator being further configured to adaptively update the probability distribution of a quantization index upon the basis of a probability density function relating to a previous state of the estimator, the auxiliary parameters, the current quantization index and the current side information.