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: The present disclosure provides methods, devices and computer program products which provide less complex and more flexible control of the introduced decorrelation in an audio coding system. According to the disclosure, this is achieved by calculating and using two weighting factors, one for an approximated audio object and one for a decorrelated audio object, for introduction of decorrelation of audio objects in the audio coding system.

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: 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: In one embodiment, a pickup system includes a wind detector and a wind suppressor. The wind detector has a plurality of analyzers each configured to analyze first and second input signals, and a combiner configured to combine outputs of the plurality of analyzers and issue, based on the combined outputs, a wind level indication signal indicative of wind activity. The analyzers can be selected from a group of analyzers including a spectral slope analyzer, a ratio analyzer, a coherence analyzer, a phase variance analyzer and the like. The wind suppressor has a ratio calculator configured to generate a ratio of the first and second input signals, and a mixer configured to select one of the first or second input signals and to apply to the selected input signal one of first or second panning coefficients based on the wind level indication signal and on the ratio.

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: 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: A spatial adaptation system for multiple-microphone sound capture systems and methods thereof are described. A spatial adaptation system includes an inference and weight module configured to receive a inputs. The inputs based on two or more input signals captured by at least two microphones. The inference and weight module to determine one or more weight values base on at least one of the inputs. The spatial adaptation system also including a noise magnitude ratio update module coupled with the inference and weight module. The noise magnitude ratio update module to determine an updated noise target based on the one or more weight values from the inference and weight module.

Abstract: A spatial adaptation system for multiple-microphone sound capture systems and methods thereof are described. A spatial adaptation system includes an inference and weight module configured to receive a inputs. The inputs based on two or more input signals captured by at least two microphones. The inference and weight module to determine one or more weight values base on at least one of the inputs. The spatial adaptation system also including a noise magnitude ratio update module coupled with the inference and weight module. The noise magnitude ratio update module to determine an updated noise target based on the one or more weight values from the inference and weight module.

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 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: The invention provides a layered audio coding format with a monophonic layer and at least one sound field layer. A plurality of audio signals is decomposed, in accordance with decomposition parameters controlling the quantitative properties of an orthogonal energy-compacting transform, into rotated audio signals. Further, a time-variable gain profile specifying constructively how the rotated audio signals may be processed to attenuate undesired audio content is derived. The monophonic layer may comprise one of the rotated signals and the gain profile. The sound field layer may comprise the rotated signals and the decomposition parameters. In one embodiment, the gain profile comprises a cleaning gain profile with the main purpose of eliminating non-speech components and/or noise. The gain profile may also comprise mutually independent broadband gains.

Abstract: The invention provides a layered audio coding format with a monophonic layer and at least one sound field layer. A plurality of audio signals is decomposed, in accordance with decomposition parameters controlling the quantitative properties of an orthogonal energy-compacting transform, into rotated audio signals. Further, a time-variable gain profile specifying constructively how the rotated audio signals may be processed to attenuate undesired audio content is derived. The monophonic layer may comprise one of the rotated signals and the gain profile. The sound field layer may comprise the rotated signals and the decomposition parameters. In one embodiment, the gain profile comprises a cleaning gain profile with the main purpose of eliminating non-speech components and/or noise. The gain profile may also comprise mutually independent broadband gains.

Abstract: An audio decoding system (100) for processing a two-channel input signal (X) comprises a parametric mixing stage (110). The parametric mixing stage receives the two-channel input signal and a set of mixing parameters (P1), and outputs a two-channel output signal (Y1). The parametric mixing stage comprises a decorrelation stage (111) outputting a decorrelated signal (D1) based on the input signal. The parametric mixing stage further comprises a mixing matrix (112) receiving the input signal and the de-correlated signal, and forming a two-channel linear combination of channels from the input signal and the decorrelated signal. The mixing matrix outputs the linear combination as the two-channel output signal. Coefficients of the linear combination are controllable by the set of mixing parameters, and at least four mixing parameters of the set are independently assignable.

Abstract: In one embodiment, a pickup system includes a wind detector and a wind suppressor. The wind detector has a plurality of analyzers each configured to analyze first and second input signals, and a combiner configured to combine outputs of the plurality of analyzers and issue, based on the combined outputs, a wind level indication signal indicative of wind activity. The analyzers can be selected from a group of analyzers including a spectral slope analyzer, a ratio analyzer, a coherence analyzer, a phase variance analyzer and the like. The wind suppressor has a ratio calculator configured to generate a ratio of the first and second input signals, and a mixer configured to select one of the first or second input signals and to apply to the selected input signal one of first or second panning coefficients based on the wind level indication signal and on the ratio.

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: Exemplary embodiments provide encoding and decoding methods, and associated encoders and decoders, for encoding and decoding of an audio scene which at least comprises one or more audio objects (106a). The encoder (108, 110) generates a bit stream (116) which comprises downmix signals (112) and side information which includes individual matrix elements (114) of a reconstruction matrix which enables reconstruction of the one or more audio objects (106a) in the decoder (120).

Abstract: There is provided encoding and decoding methods for encoding and decoding of object based audio. An exemplary encoding method includes inter alia calculating M downmix signals by forming combinations of N audio objects, wherein M?N, and calculating parameters which allow reconstruction of a set of audio objects formed on basis of the N audio objects from the M downmix signals. The calculation of the M downmix signals is made according to a criterion which is independent of any loudspeaker configuration.

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.