Abstract:
A method for encoding and decoding a digital audio signal is provided, said method comprising the steps of: encoding a first sequence of samples of the digital signal according to a transform encoding; encoding a second sequence of samples of the digital signal according to a predictive encoding; wherein the second sequence starts before the end of the first sequence, a subsequence common to the first and second sequences being thus encoded both by predictive encoding and by transform encoding.
Abstract:
A system for coding a hierarchical audio signal, comprising, at least, a core layer using parametric coding by analysis by synthesis in a first frequency band, a band extension layer for widening said first frequency band into a second frequency band, or wideband. The system also comprises a wideband audio coding quality enhancement layer based on transform coding using a spectral parameter obtained from said band extension layer. Application to transmitting speech and/or audio signals over packet networks.
Abstract:
The present invention provides coding/decoding a digital signal, in particular using a transform with overlap employing weighting windows. In the invention, two consecutive and equal-size blocks of samples of the signal may be weighted by respective different successive windows. These two windows may be chosen independently of each other according to a criterion specific to the characteristics of the signal (entropy, data rate/distortion, etc.) that are determined for each of the two blocks.
Abstract:
The invention relates to the compression coding of digital signals such as multimedia signals (audio or video), and more particularly a method for multiple coding, wherein several encoders each comprising a series of functional blocks receive an input signal in parallel. Accordingly, a method is provided in which, a) the functional blocks forming each encoder are identified, along with one or several functions carried out of each block, b) functions which are common to various encoders are itemized and c) said common functions are carried out definitively for a part of at least all of the encoders within at least one same calculation module.
Abstract:
A system and a method for the scalable coding of a multi-channel audio signal comprising a principal component analysis (PCA) transformation of at least two channels (L, R) of the audio signal into a principal component (CP) and at least one residual sub-component (r) by rotation defined by a transformation parameter (θ), comprising the following steps: formation of a frequency subband-based residual structure (Sfr) on the basis of the at least one residual sub-component (r), and definition of a coded audio signal (SC) comprising the principal component (CP), at least one residual structure (Sfr) of a frequency subband and the transformation parameter (θ).
Abstract:
A system and a method for coding by principal component analysis (PCA) of a multi-channel audio signal comprising the following steps: decomposing at least two channels (L, R) of said audio signal into a plurality of frequency sub-bands (1(b1), . . . , 1(bN), r(b1), . . . , r(bN)), calculating at least one transformation parameter (θ(b1), . . . , θ(bN)) as a function of at least some of said plurality of frequency sub-bands, transforming at least some of said plurality of frequency sub-bands into a plurality of frequency sub-components as a function of said at least one transformation parameter (θ(b1), . . . , θ(bN)), said plurality of frequency sub-components comprising principal frequency sub-components (CP(b1), . . . , CP(bN)), combining at least some of said principal frequency sub-components (CP(b1), . . . , CP(bN)) in order to form a principal component (CP), and defining a coded audio signal (SC) representing said multi-channel audio signal (C1, . . . ,CM), said coded audio signal (SC) comprising said principal component (CP) and said at least one transformation parameter (θ(b1), . . . , θ(bN)).
Abstract:
The invention relates to the synthesis and the joint spatialization of sounds emitted by virtual sources. According to the invention, a step (ETA) is provided that consists of determining parameters including at least one gain (gi) for defining, at the same time, a loudness characterizing the nature of the virtual source and the position of the source relative to a predetermined origin.
Abstract:
A method for processing sound data is provided for the reconstruction of multi-channel audio data on the basis at least of data on a reduced number of channels and of spatialization data. A test is carried out to determine whether the spatialization data received are valid. If the test is positive, a spatialization value is predicted according to a per respective model of a plurality of models. A prediction model is chosen on the basis of the spatialization values thus predicted and on the basis of the spatialization data received, to permit, in case of subsequent reception of defective spatialization data, a prediction according to this chosen model of a spatialization value and to use this predicted spatialization value for the reconstruction of the multi-channel audio data.
Abstract:
A method is provided for coding a multi-channel audio signal representing a sound scene comprising a plurality of sound sources. The method comprises decomposing the multi-channel signal into frequency bands and the following performed per frequency band: obtaining data representative of the direction of the sound sources of the sound scene, selecting a set of sound sources constituting principal sources, adapting the data representative of the direction of the selected principal sources, as a function of restitution characteristics of the multi-channel signal, determining a matrix for mixing the principal sources as a function of the adapted data, matrixing the principal sources by the matrix determined so as to obtain a sum signal with a reduced number of channels and coding the data representative of the direction of the sound sources and forming a binary stream comprising the coded data, the binary stream being transmittable in parallel with the sum signal.
Abstract:
A method for processing a signal in the form of consecutive sample blocks, the method comprising filtering in a transformed domain of sub-bands, and particularly equalization processing, applied to a current block in the transformed domain, and filtering-adjustment processing that is applied in the transformed domain to at least one block adjacent to the current block.