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:
The invention proposes the synthesis of a signal consisting of consecutive blocks. It proposes more particularly, on receipt of such a signal, to replace, by synthesis, lost or erroneous blocks of this signal. To this end, it proposes an attenuation of the overvoicing during the generation of a signal synthesis. More particularly, a voiced excitation is generated on the basis of the pitch period (T) estimated or transmitted at the previous block, by optionally applying a correction of plus or minus a sample of the duration of this period (counted in terms of number of samples), by constituting groups (A′,B′,C′,D′) of at least two samples and inverting positions of samples in the groups, randomly (B′,C′) or in a forced manner. An over-harmonicity in the excitation generated is thus broken and the effect of overvoicing in the synthesis of the generated signal is thereby attenuated.
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 proposes the synthesis of a signal consisting of consecutive blocks. It proposes more particularly, on receipt of such a signal, to replace, by synthesis, lost or erroneous blocks of this signal. To this end, it proposes an attenuation of the overvoicing during the generation of a signal synthesis. More particularly, a voiced excitation is generated on the basis of the pitch period (T) estimated or transmitted at the previous block, by optionally applying a correction of plus or minus a sample of the duration of this period (counted in terms of number of samples), by constituting groups (A′,B′,C′,D′) of at least two samples and inverting positions of samples in the groups, randomly (B′,C′) or in a forced manner. An over-harmonicity in the excitation generated is thus broken and the effect of overvoicing in the synthesis of the generated signal is thereby attenuated.
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:
Decoder for an audio signal coded by a coder including a long-term prediction filter wherein the decoder comprises: a block (211) for detecting transmission frame losses; a module (222) for calculating values of an error indication function representative of the cumulative adaptive excitation error during decoding following said transmission frame loss, an arbitrary value being assigned to said adaptive excitation gain for the lost frame; a module (213) for calculating an error indication parameter from said values of the error indication function; a comparator (214) for comparing said error indication parameter to at least one given threshold; and a discriminator (215) adapted to determine as a function of the results supplied by the comparator (214) a value of at least one adaptive excitation gain to be used by the decoder.
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.