Abstract: The invention concerns data processing by passage between different subband domains, of a first number L to a second number M of subband components. After determining a third number K, least common multiple between the first number L and the second number M: a) if K is different from L, it consists in arranging in blocs, by a serial/parallel conversion, an input vector X(z) to, obtain p2 polyphase component vectors (p2=KL); b) applying a square matrix filtering T(z) of dimensions K×K, to the p2 polyphase component vectors to obtain p1 polyphase component vectors for forming an output vector Y(z), with p1=K/M, and if the third number K is different from the second number M, providing a block arrangement by a parallel/serial conversion to obtain the output vector Y(z).

Abstract: The invention relates to a method for quantifying components, wherein certain components are each determined based on a plurality of audio signals and can be calculated by the application of a linear conversion on the audio signals, said method comprising: determining a quantification function to be applied to the components by testing a condition relative to an audio signal and depending on a comparison made between a psycho-acoustic masking threshold relative to the audio signal and a value determined based on the reverse linear conversion and quantification errors of the components by the function.

Abstract: The invention relates to a method for sequencing spectral components of elements to be encoded (A1, . . . , AQ) originating from an audio scene comprising N signals (Sii=1 to N), in which N>1, an element to be encoded comprising spectral components associated with respective spectral bands, characterised in that it comprises the following steps: calculation of the respective influence of at least some spectral components which can be calculated as a function of the spectral parameters originating from at least some of the N signals on the mask-to-noise ratios determined over the spectral bands as a function of the encoding of said spectral components; and allocation of an order of priority to at least one spectral component as a function of the influence calculated for said spectral component compared to the other influences calculated.

Abstract: The invention relates to a method for ordering spectral parameters of ambisonic components to be encoded (A1, . . . , AQ) originating from an audio scene comprising N signals (Sii=1 to N), in which N>1, comprising the following steps: calculation of the respective influence of at least some spectral parameters, taken from a set of spectral parameters to be ordered, on an angle vector defined as a function of energy and velocity vectors associated with Gerzon's criteria and calculated as a function of a reverse ambisonic transformation in relation to said quantified ambisonic components; and allocation of a precedence order to at least one spectral parameter as a function of the influence calculated for said spectral parameter compared to the other calculated influences.

Abstract: The invention relates to a method for sequencing spectral components of elements to be encoded (A1, . . . , AQ) originating from an audio scene comprising N signals (Sii=1 to N), in which N>1, an element to be encoded comprising spectral components associated with respective spectral bands, characterised in that it comprises the following steps: calculation of the respective influence of at least some spectral components which can be calculated as a function of the spectral parameters originating from at least some of the N signals on the mask-to-noise ratios determined over the spectral bands as a function of the encoding of said spectral components; and allocation of an order of priority to at least one spectral component as a function of the influence calculated for said spectral component compared to the other influences calculated.

Abstract: The invention concerns a method for reproducing digital content including the following steps: receiving (E222) an identifier (Id?) of the digital content from a secure electronic entity; extracting (E226) a digital watermark of the digital content; controlling (E232) the reproduction of the content based on a comparison between the extracted watermark and the identifier. The invention concerns a reproducing device, an electronic entity and related systems.

Abstract: The invention relates to a method for encoding the components (Xi,k) of an audio scene including N signals (Si, . . . , SN) with N>1, that comprises the step of quantifying at least some of said components, wherein the quantification is defined based on at least an energy vector and/or one velocity vector associated with Gerzon criteria and based on said components.

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: The invention relates to a method for quantifying components, wherein certain components are each determined based on a plurality of audio signals and can be calculated by the application of a linear conversion on the audio signals, said method comprising: determining a quantification function to be applied to the components by testing a condition relative to an audio signal and depending on a comparison made between a psycho-acoustic masking threshold relative to the audio signal and a value determined based on the reverse linear conversion and quantification errors of the components by the function.

Abstract: The invention relates to a method for ordering spectral parameters of ambisonic components to be encoded (A1, . . . , AQ) originating from an audio scene comprising N signals (Sii=1 to N), in which N>1, comprising the following steps: calculation of the respective influence of at least some spectral parameters, taken from a set of spectral parameters to be ordered, on an angle vector defined as a function of energy and velocity vectors associated with Gerzon's criteria and calculated as a function of a reverse ambisonic transformation in relation to said quantified ambisonic components; and allocation of a precedence order to at least one spectral parameter as a function of the influence calculated for said spectral parameter compared to the other calculated influences.

Abstract: Conversion between sub-band field representations for time-dependent filter banks. The invention relates to a transcoding processing operation between different sub-band fields, aiming to compact the application of a first vector representing the signal in a first sub-band field to a synthesis filter bank, and then to an analysis filter bank, in order to obtain a second vector representing the signal in a second sub-band field. In particular, the synthesis bank and/or the analysis bank are time-dependent. Within the scope of the invention, matrix filtering of the first vector is anticipated in order to directly obtain the second vector, this matrix filtering being represented by a global conversion matrix comprising pre-calculated sub-blocks of matrices (Ai0, . . . , Aip2?1) taking into consideration possible time-dependent variations in the filter banks, then stored into memory.

Abstract: The invention concerns data processing by passage between different subband domains, of a first number L to a second number M of subband components. After determining a third number K, least common multiple between the first number L and the second number M: a) if K is different from L, it consists in arranging in blocs, by a serial/parallel conversion, an input vector X(z) to, obtain p2 polyphase component vectors (p2=KL); b) applying a square matrix filtering T(z) of dimensions K×K, to the p2 polyphase component vectors to obtain p1 polyphase component vectors for forming an output vector Y(z), with p1=K/M, and if the third number K is different from the second number M, providing a block arrangement by a parallel/serial conversion to obtain the output vector Y(z).

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. According to the invention, a) the functional blocks (BF10, BFnN) 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. (BF1CC, BFnCC).

Abstract: A method of setting up a communication between a plurality of terminals (T1, . . . , Tn) in a network comprises the activation of at least one chosen node (N1, . . . , Nm) of the network. A node is selected beforehand as a function of predetermined criteria, then the presence of possible means of management of communication on this node (N1, . . . , Nm) is thereafter detected. In case of absence of the means of management on the node, means of management of communication are dispatched to the node. This method applies in particular to computer networks that can support teleconferences.

Abstract: The invention relates to the treatment of sound data for spatialized restitution of acoustic signals. At least one first and one second series of weighting terms are obtained for each acoustic signal, said terms representing a direction of perception of said acoustic signal by a listener. The acoustic signals are then applied to at least two sets of filtering units, which are disposed in parallel, in order to provide at least one first and one second output signal (L,R), corresponding to a linear combination of signals provided by said filtering units, which are respectively weighted by the weighting terms of the first and second series. According to the invention, each acoustic signal to be treated is at least partially compression coded and is expressed in the form of a vector of sub-signals associated with respective frequency sub-bands. Matrix filtering applied to each vector is carried out by each filtering unit in the space of the frequential sub-bands.

Abstract: A system for accessing data (10) associated with user rights (12) for the data (10), expressed in a specific language for expressing rights, on at least one consultation station (4) including elements for recovering (14) user rights adapted to identify user rights expressed in one or more different languages for access to the data (10). The system is characterized in that it includes a unit (16) for adapting user rights associated with elements (18) for determining the specific language wherein the user rights (12) are expressed associated with the data (10), to elements (20; 78) for determining the language(s) expressing user rights identified by the elements (14) for recovery from the consultation station (4), and to elements (22) converting user rights (12) for converting the user rights (12) associated with the data (10).

Abstract: At least one coded binary audio flux organized into frames is created from digital audio signals which were coded by transforming them from the time domain into the frequency domain. Transform coefficients of the signals in the frequency domain are quantized and coded according to a set of quantizers. The set is determined from a set of values extracted from the signals. The values make up selection parameters of the set of quantizers. The parameters are also present in the frames. A partial decoding state decodes then dequantizes transform coefficients produced by the coding based on a set of quantizers determined from the selection parameters contained in the frames of the coded binary audio flux or of each coded binary audio flux. The partially decoded frames are subjected to processing in the frequency domain. The thus-processed frames are then made available for use in a later utilization step.

Abstract: At least one coded binary audio flux organized into frames is created from digital audio signals which were coded by transforming them from the time domain into the frequency domain. Transform coefficients of the signals in the frequency domain are quantized and coded according to a set of quantizers. The set is determined from a set of values extracted from the signals. The values make up selection parameters of the set of quantizers. The parameters are also present in the frames. A partial decoding state decodes then dequantizes transform coefficients produced by the coding based on a set of quantizers determined from the selection parameters contained in the frames of the coded binary audio flux or of each coded binary audio flux. The partially decoded frames are subjected to processing in the frequency domain. The thus-processed frames are then made available for use in a later utilization step.