EP2168121A1 - Quantification after linear conversion combining audio signals of a sound scene, and related encoder - Google Patents

Quantification after linear conversion combining audio signals of a sound scene, and related encoder

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Publication number
EP2168121A1
EP2168121A1 EP08806144A EP08806144A EP2168121A1 EP 2168121 A1 EP2168121 A1 EP 2168121A1 EP 08806144 A EP08806144 A EP 08806144A EP 08806144 A EP08806144 A EP 08806144A EP 2168121 A1 EP2168121 A1 EP 2168121A1
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Prior art keywords
quantization
components
function
audio signals
module
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EP08806144A
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German (de)
French (fr)
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EP2168121B1 (en
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Adil Mouhssine
Abdellatif Benjelloun Touimi
Pierre Duhamel
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Orange SA
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France Telecom SA
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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/008—Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
    • G10L19/032—Quantisation or dequantisation of spectral components
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S3/00—Systems employing more than two channels, e.g. quadraphonic
    • H04S3/008—Systems employing more than two channels, e.g. quadraphonic in which the audio signals are in digital form, i.e. employing more than two discrete digital channels
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S3/00—Systems employing more than two channels, e.g. quadraphonic
    • H04S3/02—Systems employing more than two channels, e.g. quadraphonic of the matrix type, i.e. in which input signals are combined algebraically, e.g. after having been phase shifted with respect to each other
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S2420/00—Techniques used stereophonic systems covered by H04S but not provided for in its groups
    • H04S2420/01—Enhancing the perception of the sound image or of the spatial distribution using head related transfer functions [HRTF's] or equivalents thereof, e.g. interaural time difference [ITD] or interaural level difference [ILD]

Definitions

  • the present invention relates to audio signal coding devices, intended in particular to take place in applications for transmission or storage of digitized and compressed audio signals.
  • the invention relates more specifically to the quantization modules included in these audio coding devices.
  • a 3D sound scene also called spatialized sound, comprises a plurality of audio channels each corresponding to monophonic signals.
  • a signal coding technique for a sound stage used in the "MPEG Audio Surround” encoder includes the extraction and coding of spatial parameters from the set of monophonic audio signals on the different channels. These signals are then mixed to obtain a monophonic or stereophonic signal, which is then compressed by a conventional mono or stereo encoder (for example of the MPEG-4 AAC, HE-AAC type, etc.). At the level of the decoder, the synthesis of the rendered 3D sound scene is made from the spatial parameters and the decoded mono or stereo signal.
  • the coding of the multichannel signals in certain cases requires the introduction of a transformation (KLT, Ambiophonic, DCT, etc.) making it possible to better take into account the interactions that may exist between the different signals of the sound scene to be encoded.
  • KLT KLT, Ambiophonic, DCT, etc.
  • the invention proposes a method for quantifying components, at least some of these components being each determined according to a plurality of audio signals of a scene. sound and calculable by applying a linear transformation on said audio signals.
  • a quantization function is determined to be applied to said components in a given frequency band by testing a condition relating to at least one audio signal and depending at least on a comparison made between a psychoacoustic masking threshold relative to the audio signal. in the given frequency band, and a value determined according to the inverse linear transformation and quantization errors of the components by said function on the given frequency band.
  • Such a method therefore makes it possible to determine a quantization function which makes it possible to mask, in the playback listening field, the noise introduced with respect to the audio signal of the initial sound scene.
  • the sound scene restored after the coding and decoding operations therefore presents a better audio quality.
  • the introduction of a multichannel transform transforms the real signals into a new domain different from the listening domain.
  • the quantization of the components resulting from this transform according to the methods of the state of the art, based on a perceptual criterion (ie respecting the masking threshold on the latter), does not guarantee a minimum distortion on the real signals restored in the listening domain.
  • the calculation of the quantization function according to the invention makes it possible to guarantee that the quantization noises induced on the real signals by the quantization of the transformed components are minimal in the sense of a perceptual criterion. The condition of a maximum improvement of the perceptual quality of the signals in the listening domain is then verified.
  • the condition is relative to several audio signals and depends on several comparisons, each comparison being made between a psychoacoustic masking threshold relative to a respective audio signal in the given frequency band, and a value determined according to the inverse linear transformation and quantization errors of the components by said function. This arrangement further enhances the audio quality of the restored sound stage.
  • the determination of the quantization function is repeated when updating the values of the components to be quantized. This arrangement also makes it possible to increase the audio quality of the restored sound scene, by adapting the quantization over time according to the characteristics of the signals.
  • B ⁇ (s) represents a parameter the quantization function s in the band on the j th component
  • ⁇ ⁇ (s) is the expected value in the strip s of the square root of the j-th component.
  • a quantization function is determined to apply components in the given frequency band using an iterative process generating at each iteration a parameter of the candidate quantization function satisfying the condition and associated with a corresponding flow rate, the iteration being stopped when the flow rate is below a given threshold.
  • Such an arrangement thus makes it possible to simply determine a quantization function based on the determined parameters, allowing the noise to be masked in the playback listening domain while reducing the coding bit rate below a given threshold.
  • the linear transformation is an ambiophonic transformation.
  • the linear transformation is an ambiophonic transformation (called “ambisonic").
  • ambisonic ambiophonic transformation
  • This arrangement makes it possible on the one hand to reduce the number of data to be transmitted since, in general, the N signals can be very satisfactorily described by a reduced number of ambiophonic components (for example, a number equal to 3 or 5). , which is smaller than N.
  • This arrangement also allows coding adaptability to any type of sound rendering system, since it is sufficient at the decoder level to apply an inverse surround transform of size Q'x (2p '+ 1). , (where Q 'is equal to the number of loudspeakers of the sound rendering system used at the output of the decoder and 2p' + 1 the number of received surround components), to determine the signals to be supplied to the sound rendering system.
  • the invention can be implemented with any linear transformation, for example the DCT or the KLT (in English "Karhunen Loeve Transform") transform which corresponds to a decomposition on principal components in a space representing the statistics of the signals and allows to distinguish the most energetic components from the least energy components.
  • any linear transformation for example the DCT or the KLT (in English "Karhunen Loeve Transform") transform which corresponds to a decomposition on principal components in a space representing the statistics of the signals and allows to distinguish the most energetic components from the least energy components.
  • the invention proposes a quantization module adapted to quantify components, at least some of these components being each determined according to a plurality of audio signals of a sound scene and calculable by application of a transformation. linearly on said audio signals, said quantization module being adapted to implement the steps of a method according to the first aspect of the invention.
  • the invention provides an audio coder adapted to encode an audio scene comprising a plurality of respective signals into an output bit stream, comprising: a transform module adapted to calculate by applying a linear transformation on said audio signals, components at least some of which are determined each according to a plurality of audio signals of a sound scene; and a quantization module according to the second aspect of the invention adapted to determine at least one quantization function over at least a given frequency band and for quantizing the components on the given frequency band as a function of at least the determined quantization function; the audio coder being adapted to constitute a bit stream according to at least quantization data delivered by the quantization module.
  • the invention proposes a computer program to be installed in a quantization module, said program comprising instructions for implementing the steps of a method according to the first aspect of the invention during execution. of the program by means of processing said module.
  • the invention proposes coding data, determined following the implementation of a quantization method according to the first aspect of the invention.
  • FIG. 1 shows an encoder in an embodiment of the invention
  • FIG. 2 represents a decoder in one embodiment of the invention
  • Fig. 3 is a flowchart showing steps of a method in one embodiment of the invention.
  • Figure 1 shows an audio coder 1 in one embodiment of the invention. It relies on the technology of perceptual audio coders, for example MPEG-4 AAC type.
  • the encoder 1 comprises a time / frequency transformation module 2, a linear transformation module 3, a quantization module 4, a Huffman entropy coding module 5 and a masking curve calculation module 6, for transmission.
  • a bit stream ⁇ representing the signals supplied at the input of the encoder 1.
  • a 3D sound scene comprises N channels on each a respective audio signal S 1 , ..., S N is delivered.
  • Figure 2 shows an audio decoder 100 in one embodiment of the invention.
  • the decoder 100 comprises a bit sequence reading module 101, an inverse quantization module 102, an inverse linear transformation module 103, a frequency / time transformation module 104.
  • the decoder 100 is adapted to receive as input the bitstream ⁇ transmitted by the encoder 1 and to output Q 'signals S ⁇ , ..., S' Q. for supplying the respective loudspeakers H1, H2 ..., HQ 'of a sound rendering system 105.
  • the time / frequency conversion module 2 of the encoder 1 receives as input the N signals S 1 ,... S N of the 3D sound scene to be encoded, in the form of successive blocks.
  • Each block m received has N time frames each indicating different values taken over time by a respective signal.
  • the time / frequency transformation module 2 On each time frame of each of the signals, the time / frequency transformation module 2 performs a time / frequency transformation, in this case a modified discrete cosine transform (MDCT).
  • MDCT modified discrete cosine transform
  • the coding of multichannel signals comprises in the case considered a linear transformation, making it possible to take into account the interactions between the different audio signals to be coded, before the monophonic coding, by the quantization module 4, of the components resulting from the linear transformation.
  • the linear transformation module 3 is adapted to perform a linear transformation of the coefficients of the spectral representations (X t ⁇ ⁇ ⁇ N provided, in one embodiment it is adapted to perform a spatial transformation, and it determines the spatial components of the signals ⁇ x, ⁇ ⁇ ⁇ N in the frequency domain, resulting from the projection on a spatial referential depending on the order of the transformation
  • the order of a spatial transformation is related to the angular frequency according to which it "scrutinizes" The sound field.
  • the surround components are determined as follows:
  • R is the ambiophonic transformation matrix
  • Each of the ambiophonic components is therefore determined according to several signals (S 1 ) ⁇ N.
  • the masking curve calculation module 6 is adapted to determine the spectral masking curve of each frame of a signal Si considered individually in the block m, using its spectral representation Xi and a psychoacoustic model.
  • the masking curve calculation module 6 thus calculates a masking threshold M TM (s, i) relative to the frame of each signal (S t ) 1 ⁇ n ⁇ N in the block m, for each frequency band s considered during the quantification.
  • Each frequency band s is part of a set of frequency bands including for example the bands as normalized for the MPEG-4 AAC encoder.
  • the masking thresholds M TM (s, i) for each signal S 1 and each frequency band s are delivered to the quantization module 4.
  • the quantization module 4 is adapted to quantify the components ⁇ Y ⁇ ) ⁇ r that are input to it, so as to reduce the bit rate required for transmission. Respective quantization functions are determined by the quantization module 4 on each frequency band s.
  • the quantization module 4 quantizes each spectral coefficient (Y ] t ) 1]] ⁇ r such that the frequency F t is an element of the
  • k takes the values of the set +1) is equal to the number of spectral coefficients to be quantized in the s-band for all the surround components.
  • O ⁇ t ⁇ M-1 signals takes the following form, according to MPEG-4 AAC
  • Arr is a rounding function that delivers an integer value.
  • Arr (x) is for example the function providing the integer closest to the variable x, or the function "integer part" of the variable x, etc.
  • the quantization module 4 is adapted to determine a quantization function to be applied on a frequency band, verifying that the masking threshold M TM (s, i) of each signal S 1 in the listening domain, with 1 ⁇ i ⁇ N, is greater than the power of the error made, on an audio signal restored in the listening domain corresponding to the channel i (and not in the linear transformation domain), by the quantization errors made to the ambiophonic components.
  • the quantization module 4 is therefore adapted to determine, during the processing of a block m of signals, the quantization function defined using the scale parameters ⁇ Bf is)) ⁇ ⁇ r relative to each band s, such that, for all i, 1 ⁇ i ⁇ N, the error introduced on the signal S 1 in the band s by the quantization of the ambiophonic components is less than the mask threshold M TM (s, i) of the signal S 1 on the band s.
  • a problem to be solved by the quantization module 4 is therefore to determine, on each band s, the set of scaling coefficients ( ⁇ j (S)) ⁇ satisfying the following formula (1):
  • B ⁇ (s) represents a parameter characterizing the quantization function s in the band on the j-th component.
  • the choice of B ⁇ (s) determines in a bijective manner the quantization function used.
  • This arrangement has the effect that the noise brought into the listening domain by the quantization on the components resulting from the linear transformation remains masked by the signal in the listening domain, which contributes to a better quality of the signals restored in the listening domain.
  • e TM (k) are the quantization errors introduced on the (k max s - k ⁇ an + ls + l) spectral coefficients of ambiophonic components corresponding to frequencies in the band s.
  • the quantization errors e TM (k) are independent random variables equi-distributed according to the index k; the quantization errors e TM (k) are random variables according to the index i; the number of samples in a band s is large enough; the coder 1 works at high resolution.
  • the power P e m (s, i) of the quantization error, in a subband s and for a signal S 1 tends, when the number of coefficients in a band s increases, to a Gaussian whose mean m um / e ⁇ and the variance ⁇ um , e ⁇ are given by the following formulas:
  • e R the rounding error specific to the rounding function Arr. For example, if Arr (x) is the function providing the integer closest to the variable x, e R is equal to 0.5. If Arr (x) is the function "integer part" of the variable x, e R is equal to 1.
  • This last equation represents a sufficient condition for the noise corresponding to the channel i to be masked at the output in the listening domain.
  • the quantization module 4 is adapted to determine using the latter equation, for a block m of current frames, scale coefficients [BJ (s)) ⁇ r guaranteeing that the noise in the listening domain is hidden.
  • the quantization module 4 is adapted to determine, for a block m of current frames, scaling coefficients [BJ (s)) ⁇ ensuring that the noise in the d domain listening is masked and further to respect a flow constraint.
  • the conditions to be respected are the following:
  • D TM (s) is the bit rate assigned to the surround component Y 1 in the s band.
  • bit rate assigned to an ambiophonic component in a band s is a logarithmic function of the scale coefficient, ie:
  • the resolution of this constrained optimization problem is for example carried out using the Lagrangian method.
  • the Lagrangian function is written in the following form:
  • the iterative relative gradient method (see in particular the Derrien document) is used to solve this system.
  • the vector m is chosen equal to:
  • the quantization module 4 is adapted to implement the steps of the method described below with reference to FIG. 3 on each quantization band s during the quantization of a block m of signals ( S t ) 1 ⁇ N .
  • the method is based on an iterative algorithm comprising instructions for implementing the steps described below during the execution of the algorithm on calculation means of the quantization module 4.
  • the steps of the iterative loop for a (k + 1) th iteration, with k integer greater than or equal to 0, are as follows.
  • a step d / the value of the function F is calculated on the band s, representing the corresponding bit rate for the band s:
  • a step e / the calculated value F (s) is compared with the given threshold D.
  • a step g / the index k is incremented by one unit and the steps b /, c /, d / and e / are repeated.
  • the quantization function thus determined for the respective s-bands and respective surround components is then applied to the spectral coefficients of the surround components.
  • the quantization indices as well as definition elements of the quantization function are provided to the Huffman coding module.
  • the coding data delivered by the Huffman coding module 5 is then transmitted as a bit stream ⁇ to the decoder 100. Operations performed at the decoder:
  • the bit sequence reading module 101 is adapted to extract coding data present in the stream ⁇ received by the decoder and to deduce, in each band s, quantization indices i (k) and scale coefficients (B TM (s)) ⁇ ] ⁇ r .
  • the inverse quantization module 102 is adapted to determine the spectral coefficients, relative to the band s, of the corresponding ambiophonic components as a function of the quantization indices i (k) and the scale coefficients (B TM (s)) ⁇ ] ⁇ r in each band s.
  • Ambiophonic decoding is then applied to the decoded surround components, so as to determine the signals S'i, S ' 2 , ..., S'Q ⁇ for the Q' speakers H1, H2 ..., HQ .
  • the quantization noise at the output of the decoder 100 is a constant which depends only on the transform R used and the quantization module 4 because the psychoacoustic data used during the coding do not take into consideration the processing performed during the rendering by the processor. decoder. Indeed, the psychoacoustic model does not take into account the acoustic interactions between the different signals, but calculates the masking curve of a signal as if it were the only one listened to. The error calculated on this signal therefore remains constant and masked for any surround decoding matrix used. This surround decoding matrix will simply change the distribution of the error on the different speakers output.

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Abstract

The invention relates to a method for quantifying components ((yj)1=j=r ), wherein certain components are each determined based on a plurality of audio signals ( (sj)11=j=N) and can be calculated by the application of a linear conversion on the audio signals, said method comprising: determining a quantification function (Qm) to be applied to the components by testing a condition relative to an audio signal ( Si ) and depending on a comparison made between a psycho-acoustic masking threshold (Mm t(s,i) ) relative to the audio signal and a value determined based on the reverse linear conversion and quantification errors of the components by the function.

Description

QUANTIFICATION APRES TRANSFORMATION LINEAIRE COMBINANT LES SIGNAUX AUDIO D'UNE SCENE SONORE, CODEUR ASSOCIE QUANTIFICATION AFTER LINEAR TRANSFORMATION COMBINING THE AUDIO SIGNALS OF A SOUND SCENE, ENCODER
La présente invention concerne les dispositifs de codage de signaux audio, destinés notamment à prendre place dans des applications de transmission ou de stockage de signaux audio numérisés et compressés.The present invention relates to audio signal coding devices, intended in particular to take place in applications for transmission or storage of digitized and compressed audio signals.
L'invention est relative plus précisément aux modules de quantification compris dans ces dispositifs de codage audio.The invention relates more specifically to the quantization modules included in these audio coding devices.
L'invention concerne plus particulièrement le codage de scène sonore 3D. Une scène sonore 3D, encore appelée son spatialisé, comprend une pluralité de canaux audio correspondant chacun à des signaux monophoniques.The invention more particularly relates to 3D sound stage coding. A 3D sound scene, also called spatialized sound, comprises a plurality of audio channels each corresponding to monophonic signals.
Une technique de codage de signaux d'une scène sonore utilisée dans le codeur « MPEG Audio Surround » (cf. « Text of ISO/IEC FDIS 23003-1 , MPEG Surround », ISO/IEC JTC1 / SC29 / WG11 N8324, JuIy 2006, Klagenfurt, Austria), comprend l'extraction et le codage de paramètres spatiaux à partir de l'ensemble des signaux audio monophoniques sur les différents canaux. Ces signaux sont ensuite mélangés pour obtenir un signal monophonique ou stéréophonique, qui est alors comprimé par un codeur mono ou stéréo classique (par exemple de type MPEG-4 AAC, HE-AAC, etc). Au niveau du décodeur, la synthèse de la scène sonore 3D restituée se fait à partir des paramètres spatiaux et du signal mono ou stéréo décodé.A signal coding technique for a sound stage used in the "MPEG Audio Surround" encoder (see "Text of ISO / IEC FDIS 23003-1, MPEG Surround", ISO / IEC JTC1 / SC29 / WG11 N8324, July 2006 , Klagenfurt, Austria), includes the extraction and coding of spatial parameters from the set of monophonic audio signals on the different channels. These signals are then mixed to obtain a monophonic or stereophonic signal, which is then compressed by a conventional mono or stereo encoder (for example of the MPEG-4 AAC, HE-AAC type, etc.). At the level of the decoder, the synthesis of the rendered 3D sound scene is made from the spatial parameters and the decoded mono or stereo signal.
Le codage des signaux multicanaux nécessite dans certains cas l'introduction d'une transformation (KLT, Ambiophonique, DCT...) permettant de mieux prendre en compte les interactions qui peuvent exister entre les différents signaux de la scène sonore à coder.The coding of the multichannel signals in certain cases requires the introduction of a transformation (KLT, Ambiophonic, DCT, etc.) making it possible to better take into account the interactions that may exist between the different signals of the sound scene to be encoded.
Il est toujours besoin d'accroitre la qualité audio des scènes sonores restituées après une opération de codage et décodage.It is always necessary to increase the audio quality of the sound scenes restored after a coding and decoding operation.
Suivant un premier aspect, l'invention propose un procédé de quantification de composantes, certaines au moins de ces composantes étant déterminées chacune en fonction d'une pluralité de signaux audio d'une scène sonore et calculables par application d'une transformation linéaire sur lesdits signaux audio.According to a first aspect, the invention proposes a method for quantifying components, at least some of these components being each determined according to a plurality of audio signals of a scene. sound and calculable by applying a linear transformation on said audio signals.
Selon le procédé, on détermine une fonction de quantification à appliquer audites composantes dans une bande de fréquence donnée en testant une condition relative à au moins un signal audio et dépendant au moins d'une comparaison effectuée entre un seuil de masquage psychoacoustique relatif au signal audio dans la bande de fréquence donnée, et une valeur déterminée en fonction de la transformation linéaire inverse et d'erreurs de quantification des composantes par ladite fonction sur la bande de fréquence donnée.According to the method, a quantization function is determined to be applied to said components in a given frequency band by testing a condition relating to at least one audio signal and depending at least on a comparison made between a psychoacoustic masking threshold relative to the audio signal. in the given frequency band, and a value determined according to the inverse linear transformation and quantization errors of the components by said function on the given frequency band.
Un tel procédé permet donc de déterminer une fonction de quantification qui permette de masquer, dans le domaine d'écoute de restitution, le bruit introduit par rapport au signal audio de la scène sonore initiale. La scène sonore restituée après les opérations de codage et décodage présente donc une meilleure qualité audio.Such a method therefore makes it possible to determine a quantization function which makes it possible to mask, in the playback listening field, the noise introduced with respect to the audio signal of the initial sound scene. The sound scene restored after the coding and decoding operations therefore presents a better audio quality.
En effet, l'introduction d'une transformée multicanal (par exemple de type ambiophonique) transforme les signaux réels dans un nouveau domaine différent du domaine d'écoute. La quantification des composantes résultant de cette transformée selon les méthodes de l'état de l'art, basées sur un critère perceptuel (i.e. respectant le seuil de masquage sur ces derniers), ne garantit pas une distorsion minimale sur les signaux réels restitués dans le domaine d'écoute. En effet, le calcul de la fonction de quantification selon l'invention permet de garantir que les bruits de quantification induits sur les signaux réels par la quantification des composantes transformées sont minimaux au sens d'un critère perceptuel. La condition d'une amélioration maximale de la qualité perceptuelle des signaux dans le domaine d'écoute est alors vérifiée.Indeed, the introduction of a multichannel transform (for example of the ambiophonic type) transforms the real signals into a new domain different from the listening domain. The quantization of the components resulting from this transform according to the methods of the state of the art, based on a perceptual criterion (ie respecting the masking threshold on the latter), does not guarantee a minimum distortion on the real signals restored in the listening domain. Indeed, the calculation of the quantization function according to the invention makes it possible to guarantee that the quantization noises induced on the real signals by the quantization of the transformed components are minimal in the sense of a perceptual criterion. The condition of a maximum improvement of the perceptual quality of the signals in the listening domain is then verified.
Dans un mode de réalisation la condition est relative à plusieurs signaux audio et dépend de plusieurs comparaisons, chaque comparaison étant effectuée entre un seuil de masquage psychoacoustique relatif à un signal audio respectif dans la bande de fréquence donnée, et une valeur déterminée en fonction de la transformation linéaire inverse et d'erreurs de quantification des composantes par ladite fonction. Cette disposition accroît encore la qualité audio de la scène sonore restituée.In one embodiment, the condition is relative to several audio signals and depends on several comparisons, each comparison being made between a psychoacoustic masking threshold relative to a respective audio signal in the given frequency band, and a value determined according to the inverse linear transformation and quantization errors of the components by said function. This arrangement further enhances the audio quality of the restored sound stage.
Dans un mode de réalisation, la détermination de la fonction de quantification est réitérée lors de l'actualisation des valeurs des composantes à quantifier. Cette disposition permet également d'accroître la qualité audio de la scène sonore restituée, en adaptant la quantification dans le temps en fonction des caractéristiques des signaux.In one embodiment, the determination of the quantization function is repeated when updating the values of the components to be quantized. This arrangement also makes it possible to increase the audio quality of the restored sound scene, by adapting the quantization over time according to the characteristics of the signals.
Dans un mode de réalisation, on teste la condition relative à un signal audio au moins en comparant le seuil de masquage psychoacoustique relatif i au signal audio et un élément représentant la valeur ∑(h?JBJ (s)2μι (s)) , où sIn one embodiment, one tests the condition relating to an audio signal at least by comparing the masking threshold psychoacoustic on i the audio signal and an element representing the Σ value (h? J B J (s) 2 μ ι (s )), where s
est la bande de fréquence donnée, r est le nombre de composantes, \ } est le coefficient de la transformée linéaire inverse relatif au signal audio et à la jeme composante avec j=1 à r, B} (s) représente un paramètre de la fonction de quantification dans la bande s relative à la jeme composante et μι (s) est l'espérance mathématique dans la bande s de la racine carrée de la jeme composante.is the frequency band, r is the number of components, \} is the coefficient of the linear inverse transform on the audio signal and the j-th component with j = 1 to r, B} (s) represents a parameter the quantization function s in the band on the j th component and μ ι (s) is the expected value in the strip s of the square root of the j-th component.
Dans un mode de réalisation, on détermine une fonction de quantification à appliquer audites composantes dans la bande de fréquence donnée à l'aide d'un processus itératif générant à chaque itération un paramètre de la fonction de quantification candidat vérifiant la condition et associé à un débit correspondant, l'itération étant stoppée lorsque le débit est inférieur à un seuil donné.In one embodiment, a quantization function is determined to apply components in the given frequency band using an iterative process generating at each iteration a parameter of the candidate quantization function satisfying the condition and associated with a corresponding flow rate, the iteration being stopped when the flow rate is below a given threshold.
Une telle disposition permet ainsi de déterminer simplement une fonction de quantification à partir des paramètres déterminés, permettant le masquage du bruit dans le domaine d'écoute de restitution tout en réduisant le débit de codage en dessous d'un seuil donné.Such an arrangement thus makes it possible to simply determine a quantization function based on the determined parameters, allowing the noise to be masked in the playback listening domain while reducing the coding bit rate below a given threshold.
Dans un mode de réalisation, la transformation linéaire est une transformation ambiophonique.In one embodiment, the linear transformation is an ambiophonic transformation.
Dans un mode de réalisation particulier, la transformation linéaire est une transformation ambiophonique (appelée en anglais « ambisonic »). Cette disposition permet d'une part de réduire le nombre de données à transmettre puisque, en général, les N signaux peuvent être décrits d'une manière très satisfaisante par un nombre de composantes ambiophoniques réduit (par exemple, un nombre égal à 3 ou 5), inférieur à N. Cette disposition permet en outre une adaptabilité du codage à tout type de système de rendu sonore, puisqu'il suffit au niveau du décodeur, d'appliquer une transformée ambiophonique inverse de taille Q'x(2p'+1 ), (où Q' est égal au nombre de haut- parleurs du système de rendu sonore utilisé en sortie du décodeur et 2p'+1 le nombre de composantes ambiophoniques reçues), pour déterminer les signaux à fournir au système de rendu sonore.In a particular embodiment, the linear transformation is an ambiophonic transformation (called "ambisonic"). This This arrangement makes it possible on the one hand to reduce the number of data to be transmitted since, in general, the N signals can be very satisfactorily described by a reduced number of ambiophonic components (for example, a number equal to 3 or 5). , which is smaller than N. This arrangement also allows coding adaptability to any type of sound rendering system, since it is sufficient at the decoder level to apply an inverse surround transform of size Q'x (2p '+ 1). , (where Q 'is equal to the number of loudspeakers of the sound rendering system used at the output of the decoder and 2p' + 1 the number of received surround components), to determine the signals to be supplied to the sound rendering system.
L'invention peut être mise en œuvre avec toute transformation linéaire, par exemple la DCT ou encore la transformée KLT (en anglais « Karhunen Loeve Transform ») qui correspond à une décomposition sur des composantes principales dans un espace représentant les statistiques des signaux et permet de distinguer les composantes les plus énergétiques des composantes les moins énergétiques.The invention can be implemented with any linear transformation, for example the DCT or the KLT (in English "Karhunen Loeve Transform") transform which corresponds to a decomposition on principal components in a space representing the statistics of the signals and allows to distinguish the most energetic components from the least energy components.
Suivant un deuxième aspect, l'invention propose un module de quantification adapté pour quantifier des composantes, certaines au moins de ces composantes étant déterminées chacune en fonction d'une pluralité de signaux audio d'une scène sonore et calculables par application d'une transformation linéaire sur lesdits signaux audio, ledit module de quantification étant adapté pour mettre en œuvre les étapes d'un procédé suivant le premier aspect de l'invention.According to a second aspect, the invention proposes a quantization module adapted to quantify components, at least some of these components being each determined according to a plurality of audio signals of a sound scene and calculable by application of a transformation. linearly on said audio signals, said quantization module being adapted to implement the steps of a method according to the first aspect of the invention.
Suivant un troisième aspect, l'invention propose un codeur audio adapté pour coder une scène audio comprenant plusieurs signaux respectifs en un flux binaire de sortie, comprenant : un module de transformation adapté pour calculer par application d'une transformation linéaire sur lesdits signaux audio, des composantes dont certaines au moins sont déterminées chacune en fonction d'une pluralité des signaux audio d'une scène sonore ; et un module de quantification suivant le deuxième aspect de l'invention adapté pour déterminer au moins une fonction de quantification sur au moins une bande de fréquence donnée et pour quantifier les composantes sur la bande de fréquence donnée en fonction d'au moins la fonction de quantification déterminée ; le codeur audio étant adapté pour constituer un flux binaire en fonction au moins de données de quantification délivrées par le module de quantification.According to a third aspect, the invention provides an audio coder adapted to encode an audio scene comprising a plurality of respective signals into an output bit stream, comprising: a transform module adapted to calculate by applying a linear transformation on said audio signals, components at least some of which are determined each according to a plurality of audio signals of a sound scene; and a quantization module according to the second aspect of the invention adapted to determine at least one quantization function over at least a given frequency band and for quantizing the components on the given frequency band as a function of at least the determined quantization function; the audio coder being adapted to constitute a bit stream according to at least quantization data delivered by the quantization module.
Suivant un quatrième aspect, l'invention propose un programme d'ordinateur à installer dans un module de quantification, ledit programme comprenant des instructions pour mettre en œuvre les étapes d'un procédé suivant le premier aspect de l'invention lors d'une exécution du programme par des moyens de traitement dudit module.According to a fourth aspect, the invention proposes a computer program to be installed in a quantization module, said program comprising instructions for implementing the steps of a method according to the first aspect of the invention during execution. of the program by means of processing said module.
Suivant un cinquième aspect, l'invention propose des données de codage, déterminées suite à la mise en œuvre d'un procédé de quantification suivant le premier aspect de l'invention.According to a fifth aspect, the invention proposes coding data, determined following the implementation of a quantization method according to the first aspect of the invention.
D'autres caractéristiques et avantages de l'invention apparaîtront encore à la lecture de la description qui va suivre. Celle-ci est purement illustrative et doit être lue en regard des dessins annexés sur lesquels : la figure 1 représente un codeur dans un mode de réalisation de l'invention ; la figure 2 représente un décodeur dans un mode de réalisation de l'invention ; la figure 3 est un organigramme représentant des étapes d'un procédé dans un mode de réalisation de l'invention. La figure 1 représente un codeur audio 1 dans un mode de réalisation de l'invention. Il s'appuie sur la technologie des codeurs audio perceptuels, par exemple de type MPEG-4 AAC.Other features and advantages of the invention will become apparent on reading the description which follows. This is purely illustrative and should be read with reference to the accompanying drawings in which: Figure 1 shows an encoder in an embodiment of the invention; FIG. 2 represents a decoder in one embodiment of the invention; Fig. 3 is a flowchart showing steps of a method in one embodiment of the invention. Figure 1 shows an audio coder 1 in one embodiment of the invention. It relies on the technology of perceptual audio coders, for example MPEG-4 AAC type.
Le codeur 1 comprend un module 2 de transformation temps/fréquence, un module 3 de transformation linéaire, un module 4 de quantification, un module 5 de codage entropique de Huffman et un module 6 de calcul de courbe de masquage, en vue de la transmission d'un flux binaire Φ représentant les signaux fournis en entrée du codeur 1. Une scène sonore 3D comprend N canaux sur chacun un signal audio respectif S1 , ..., SN est délivré.The encoder 1 comprises a time / frequency transformation module 2, a linear transformation module 3, a quantization module 4, a Huffman entropy coding module 5 and a masking curve calculation module 6, for transmission. a bit stream Φ representing the signals supplied at the input of the encoder 1. A 3D sound scene comprises N channels on each a respective audio signal S 1 , ..., S N is delivered.
La figure 2 représente un décodeur audio 100 dans un mode de réalisation de l'invention.Figure 2 shows an audio decoder 100 in one embodiment of the invention.
Le décodeur 100 comprend un module 101 de lecture de séquence binaire, un module 102 de quantification inverse, un module 103 de transformation linéaire inverse, un module 104 de transformation fréquence/temps.The decoder 100 comprises a bit sequence reading module 101, an inverse quantization module 102, an inverse linear transformation module 103, a frequency / time transformation module 104.
Le décodeur 100 est adapté pour recevoir en entrée le flux binaire Φ transmis par le codeur 1 et pour délivrer en sortie Q' signaux S\ , ..., S'Q. destinés à alimenter les Q' haut-parleurs H1 , H2 ..., HQ' respectifs d'un système de rendu sonore 105.The decoder 100 is adapted to receive as input the bitstream Φ transmitted by the encoder 1 and to output Q 'signals S \, ..., S' Q. for supplying the respective loudspeakers H1, H2 ..., HQ 'of a sound rendering system 105.
Opérations réalisées au niveau du codeur :Operations performed at the encoder level:
Le module 2 de transformation temps/fréquence du codeur 1 reçoit en entrée les N signaux S1 , ..., SN de la scène sonore 3D à coder, sous forme de blocs successifs.The time / frequency conversion module 2 of the encoder 1 receives as input the N signals S 1 ,... S N of the 3D sound scene to be encoded, in the form of successive blocks.
Chaque bloc m reçu comporte N trames temporelles indiquant chacune différentes valeurs prises au cours du temps par un signal respectif.Each block m received has N time frames each indicating different values taken over time by a respective signal.
Sur chaque trame temporelle de chacun des signaux, le module 2 de transformation temps/fréquence effectue une transformation temps/fréquence, dans le cas présent, une transformée en cosinus discrète modifiée (MDCT).On each time frame of each of the signals, the time / frequency transformation module 2 performs a time / frequency transformation, in this case a modified discrete cosine transform (MDCT).
Ainsi, suite à la réception d'un nouveau bloc comportant une nouvelle trame pour chacun des signaux S1 , il détermine, pour chacun des signaux S1 , i=1 à N, sa représentation spectrale Xj, caractérisée par M coefficients MDCT Xi t, avec t = 0 à M-1. Un coefficient MDCT Xi t représente ainsi le spectre du signal Si pour une fréquence Ft .Thus, following the reception of a new block comprising a new frame for each of the signals S 1 , it determines, for each of the signals S 1 , i = 1 to N, its spectral representation Xj, characterized by M coefficients MDCT Xi t with t = 0 to M-1. An MDCT coefficient X it thus represents the spectrum of the signal Si for a frequency F t .
Les représentations spectrales Xi des signaux S1 , i= 1 à N, sont fournies en entrée du module 3 de transformation linéaire.The spectral representations Xi of the signals S 1 , i = 1 to N, are provided at the input of the linear transformation module 3.
Les représentations spectrales Xi des signaux S1 , i= 1 à N, sont en outre fournies en entrée du module 6 de calcul des courbes de masquage. Le codage de signaux multicanaux comporte dans le cas considéré une transformation linéaire, permettant de prendre en compte les interactions entre les différents signaux audio à coder, avant le codage monophonique, par le module 4 de quantification, des composantes résultant de la transformation linéaire.The spectral representations Xi of the signals S 1 , i = 1 to N, are further provided at the input of the module 6 for calculating the masking curves. The coding of multichannel signals comprises in the case considered a linear transformation, making it possible to take into account the interactions between the different audio signals to be coded, before the monophonic coding, by the quantization module 4, of the components resulting from the linear transformation.
Le module 3 de transformation linéaire est adapté pour effectuer une transformation linéaire des coefficients des représentations spectrales (Xt \<ι<N fournis. Dans un mode de réalisation, il est adapté pour effectuer une transformation spatiale. Il détermine alors les composantes spatiales des signaux {x, \<ι<N dans le domaine fréquentiel, résultant de la projection sur un référentiel spatial dépendant de l'ordre de la transformation. L'ordre d'une transformation spatiale se rattache à la fréquence angulaire selon laquelle elle « scrute » le champ sonore.The linear transformation module 3 is adapted to perform a linear transformation of the coefficients of the spectral representations (X t \ <ι <N provided, in one embodiment it is adapted to perform a spatial transformation, and it determines the spatial components of the signals {x, \ <ι <N in the frequency domain, resulting from the projection on a spatial referential depending on the order of the transformation The order of a spatial transformation is related to the angular frequency according to which it "scrutinizes" The sound field.
Dans le mode de réalisation considéré, le module 3 de transformation linéaire effectue une transformation ambiophonique d'ordre p (par exemple p=1 ), qui donne une représentation spatiale compacte d'une scène sonore 3D, en réalisant des projections du champ sonore sur les fonctions harmoniques sphériques ou cylindriques associées.In the embodiment considered, the linear transformation module 3 performs an ambiophonic transformation of order p (for example p = 1), which gives a compact spatial representation of a 3D sound scene, by making projections of the sound field onto the spherical or cylindrical harmonic functions associated.
Pour plus d'information sur les transformations ambiophoniques, on pourra se référer aux documents suivants : « Représentation de champs acoustiques, application à la transmission et à la reproduction de scènes sonores complexes dans un contexte multimédia », Thèse de doctorat de l'université Paris 6, Jérôme DANIEL, 31 juillet 2001 , « A highly scalable spherical microphone array based on an orthonormal décomposition of the sound field », Jens Meyer - Gary Elko, Vol. Il - pp. 1781-1784 in Proc. ICASSP 2002.For more information on the ambiophonic transformations, one can refer to the following documents: "Representation of acoustic fields, application to the transmission and the reproduction of complex sound scenes in a multimedia context", Thesis of doctorate of the university Paris 6, Jerome DANIEL, July 31, 2001, "A highly scalable spherical array based microphone on an orthonormal decomposition of the sound field," Jens Meyer - Gary Elko, Vol. He - pp. 1781-1784 in Proc. ICASSP 2002.
Le module 3 de transformation spatiale délivre ainsi r (r= 2p+1 ) composantes ambiophoniques {Y} ) . Chaque composante ambiophoniqueThe spatial transformation module 3 thus delivers r (r = 2p + 1) ambiophonic components {Y } ). Each surround component
Y} considérées dans le domaine fréquentiel, comporte M paramètres spectraux Yj t pour t = 0 à M-1. Le paramètre spectral Y] t se rapporte à la fréquence Ft pour t = 0 à M-1. Les composantes ambiophoniques sont déterminés de la façon suivante :Y } considered in the frequency domain, has M spectral parameters Y jt for t = 0 to M-1. The spectral parameter Y ] t relates to the frequency F t for t = 0 to M-1. The surround components are determined as follows:
où R = est la matrice de transformation ambiophonique where R = is the ambiophonic transformation matrix
d'ordre p pour la scène sonore spatiale, avec R1 } = 1 Rl } = yf2 cos θ. SI Iof order p for the spatial sound stage, with R 1} = 1 R l} = yf2 cos θ. IF I
Ï - 1 pai irr et i?î ; = V2 ssiin θ. si i impair supérieur ou égale à 3, et θj est l'angle1 - 1 irr and i? î; = V2 ssiin θ. if i odd greater than or equal to 3, and θj is the angle
de propagation du signal S} dans l'espace de la scène 3D.of propagation of the signal S } in the space of the 3D scene.
Chacune des composantes ambiophoniques est donc déterminée en fonction de plusieurs signaux (S1 )^N .Each of the ambiophonic components is therefore determined according to several signals (S 1 ) ^ N.
Le module 6 de calcul de courbe de masquage est adapté pour déterminer la courbe de masquage spectral de chaque trame d'un signal Si considéré individuellement dans le bloc m, à l'aide de sa représentation spectrale Xi et d'un modèle psychoacoustique.The masking curve calculation module 6 is adapted to determine the spectral masking curve of each frame of a signal Si considered individually in the block m, using its spectral representation Xi and a psychoacoustic model.
Le module 6 de calcul de courbe de masquage calcule ainsi un seuil de masquage M™ (s,i) , relatif à la trame de chaque signal (St )1≤ι≤N dans le bloc m, pour chaque bande de fréquence s considérée lors de la quantification. Chaque bande de fréquence s est élément d'un ensemble de bandes de fréquence comprenant par exemple les bandes telles que normalisées pour le codeur MPEG-4 AAC.The masking curve calculation module 6 thus calculates a masking threshold M ™ (s, i) relative to the frame of each signal (S t ) 1 ι n ≤ N in the block m, for each frequency band s considered during the quantification. Each frequency band s is part of a set of frequency bands including for example the bands as normalized for the MPEG-4 AAC encoder.
Les seuils de masquage M™ (s,i) pour chaque signal S1 et chaque bande de fréquences s sont délivrés au module 4 de quantification.The masking thresholds M ™ (s, i) for each signal S 1 and each frequency band s are delivered to the quantization module 4.
Le module 4 de quantification est adapté pour quantifier les composantes {Y} ) <r qui lui sont fournies en entrée, de manière à réduire le débit nécessaire à la transmission. Des fonctions de quantification respectives sont déterminées par le module 4 de quantification sur chaque bande de fréquence s.The quantization module 4 is adapted to quantify the components {Y } ) <r that are input to it, so as to reduce the bit rate required for transmission. Respective quantization functions are determined by the quantization module 4 on each frequency band s.
Dans une bande s quelconque, le module 4 de quantification quantifie chaque coefficient spectral (Y] t)1≤]≤r tel que la fréquence Ft est élément de laIn any band s, the quantization module 4 quantizes each spectral coefficient (Y ] t ) 1]] ≤r such that the frequency F t is an element of the
O≤t≤M-l bande de fréquence s. Il détermine ainsi un indice de quantification i(k) pour chaque coefficient spectral (Y] t\≤]≤r tel que la fréquence Ft est élément de laO≤t≤Ml frequency band s. It thus determines a quantization index i (k) for each spectral coefficient (Y ] t \ ≤] ≤r such that the frequency F t is an element of the
O≤t≤M-l bande de fréquence s.O≤t≤M-l frequency band s.
Pour une bande s considérée, k prend les valeurs de l'ensemble +1) est égal au nombre de coefficients spectraux à quantifier dans la bande s pour l'ensemble des composantes ambiophoniques.For a considered band, k takes the values of the set +1) is equal to the number of spectral coefficients to be quantized in the s-band for all the surround components.
La fonction de quantification Qm appliquée par le module 4 de quantification pour les coefficients (Y] t)1≤]≤r calculés pour un bloc m deThe quantization function Q m applied by the quantization module 4 for the coefficients (Y ] t ) 1]] ≤ r calculated for a block m of
O≤t≤M-l signaux prend la forme suivante, conformément à la norme MPEG-4 AACO≤t≤M-1 signals takes the following form, according to MPEG-4 AAC
CT[Y, t ) = Arr avec la fréquence F élément de la bande de fréquence s, et il existe k élément de 1^,^^,-A^J tel que Q m{Y] t ) = i(k).CT [Y, t ) = Arr with frequency F element of the band of frequency s, and there exists the element of 1 ^, ^^, - A ^ J such that Q m {Y ] t ) = i (k).
Bf (S) , coefficient d'échelle relatif à la composante ambiophonique Yj, prend des valeurs discrètes. Il dépend du paramètre d'échelle entierBf (S), scale coefficient relative to the ambiophonic component Y j , takes discrete values. It depends on the whole scale setting
1 relatif φj (s) : B" (s) = :1 relative φj (s): B "(s) =:
Arr est une fonction d'arrondi délivrant une valeur entière. Arr(x) est par exemple la fonction fournissant l'entier le plus proche de la variable x, ou encore la fonction « partie entière » de la variable x, etc.Arr is a rounding function that delivers an integer value. Arr (x) is for example the function providing the integer closest to the variable x, or the function "integer part" of the variable x, etc.
Le module 4 de quantification est adapté pour déterminer une fonction de quantification à appliquer sur une bande de fréquence s vérifiant que le seuil de masquage M™ (s,i) de chaque signal S1 dans le domaine d'écoute, avec 1 ≤ i ≤ N, est supérieur à la puissance de l'erreur apportée, sur un signal audio restitué dans le domaine d'écoute correspondant au canal i (et non pas dans le domaine de transformation linéaire), par les erreurs de quantification apportée aux composantes ambiophoniques.The quantization module 4 is adapted to determine a quantization function to be applied on a frequency band, verifying that the masking threshold M ™ (s, i) of each signal S 1 in the listening domain, with 1 ≤ i ≤ N, is greater than the power of the error made, on an audio signal restored in the listening domain corresponding to the channel i (and not in the linear transformation domain), by the quantization errors made to the ambiophonic components.
Le module 4 de quantification est donc adapté pour déterminer, lors du traitement d'un bloc m de signaux, la fonction de quantification définie à l'aide des paramètres d'échelle {Bf is))^ <r relatifs à chaque bande s, telle que, pour tout i, 1 ≤ i ≤ N, l'erreur introduite sur le signal S1 dans la bande s par la quantification des composantes ambiophoniques est inférieure au seuil de masquage M™ (s,i) du signal S1 sur la bande s.The quantization module 4 is therefore adapted to determine, during the processing of a block m of signals, the quantization function defined using the scale parameters {Bf is)) ^ <r relative to each band s, such that, for all i, 1 ≤ i ≤ N, the error introduced on the signal S 1 in the band s by the quantization of the ambiophonic components is less than the mask threshold M ™ (s, i) of the signal S 1 on the band s.
Un problème à résoudre par le module 4 de quantification est donc de déterminer, sur chaque bande s, l'ensemble des coefficients d'échelle (βj (S)) < < vérifiant la formule (1 ) suivante :A problem to be solved by the quantization module 4 is therefore to determine, on each band s, the set of scaling coefficients (βj (S)) << satisfying the following formula (1):
{B; /Pe m (s,i) ≤ M? (s,i),l ≤ i ≤ N }ι≤]≤r où Pe m(s,i) est la puissance d'erreur introduite sur le signal S1 suite aux erreurs de quantification introduites par la quantification, définie par les coefficients d'échelle [BJ (S))^ <r , des composantes ambiophoniques.{B; / P e m (s, i) ≤ M? (s, i), l ≤ i ≤ N} ι≤] ≤r where P e m (s, i) is the error power introduced on the signal S 1 following the quantization errors introduced by the quantization, defined by the scaling coefficients [BJ (S)) ^ <r , ambiophonic components.
Ainsi, B}(s) représente un paramètre caractérisant la fonction de quantification dans la bande s relative à la jeme composante. Le choix de B}(s) détermine de manière bijective la fonction de quantification utilisée.Thus, B} (s) represents a parameter characterizing the quantization function s in the band on the j-th component. The choice of B } (s) determines in a bijective manner the quantization function used.
Cette disposition a pour effet que le bruit apporté dans le domaine d'écoute par la quantification sur les composantes issues de la transformation linéaire reste masqué par le signal dans le domaine d'écoute, ce qui contribue à une meilleure qualité des signaux restitués dans le domaine d'écoute.This arrangement has the effect that the noise brought into the listening domain by the quantization on the components resulting from the linear transformation remains masked by the signal in the listening domain, which contributes to a better quality of the signals restored in the listening domain.
Dans un mode de réalisation, le problème indiqué ci-dessus par la formule (1 ) est traduit sous la forme de la formule (2) suivante :In one embodiment, the problem indicated above by the formula (1) is translated as the following formula (2):
{β; m / Probabilité (if (s,i) ≤ M?(s,i)) ≥ a,l ≤ i ≤ N }1<;<r , où a est un taux fixé de respect du seuil de masquage. La probabilité est calculée pour la trame relative au signal S1 du bloc m considéré et sur l'ensemble des bandes de fréquence s.{β ; m / Probability (if (s, i) ≤ M (s, i)) ≥ a, l ≤ i ≤ N} 1 <;<r , where a is a fixed rate of compliance with the masking threshold. The probability is calculated for the frame relating to the signal S 1 of the block m considered and on all the frequency bands s.
La justification de cette traduction est réalisée dans le document « Optimisation de la quantification par modèles statistiques dans le codeur MPEG Advanced Audio coder (AAC) - Application à la spatialisation d'un signal comprimé en environnement MPEG-4 », Thèse de doctorat de Olivier Derrien - ENST Paris, 22 novembre 2002, nommé ci-après « document Derrien ». Selon ce document, on cherche à modifier la quantification de manière à diminuer la distorsion perçue par l'oreille d'un signal résultant d'un filtrage de spatialisation HRTF (en anglais « Head Related Transfer Function » encore appelé filtre de tête modélisant l'effet de chemin de propagation entre la position de la source sonore et l'oreille humaine et prenant en compte l'effet dû à la tête et au torse d'un auditeur, appliqué après le décodage.The justification of this translation is realized in the document "Optimization of the quantization by statistical models in the MPEG coder Advanced Audio coder (AAC) - Application to the spatialization of a compressed signal in MPEG-4 environment", PhD Thesis of Olivier Derrien - ENST Paris, 22 November 2002, hereinafter referred to as the Derrien document. According to this document, it is sought to modify the quantization so as to reduce the distortion perceived by the ear of a signal resulting from a spatialization filtering HRTF (in English "Head Related Transfer Function" also called head filter modeling the propagation path effect between the position of the sound source and the human ear and taking into account the effect due to the head and the torso of a listener, applied after the decoding.
Par ailleurs, Pe m (s,i) = ∑<" (fc)2 , où {e™ sont les erreurs introduites sur les Ks = (kmΛX S -kimn+l s +1) coefficients spectraux du signal S1 correspondant à des fréquences dans la bande s.On the other hand, P e m (s, i) = Σ <"(fc) 2 , where {e ™ are the errors introduced on the K s = (k mΛX S -k imn + ls +1) spectral coefficients of the signal S 1 corresponding to frequencies in the band s.
Soit H = (fy ji≤î≤îv la matrice inverse de la matrice de transformationLet H = (fy ji≤i≤iv be the inverse matrix of the transformation matrix
ambiophonique R, alors e™ (k) sont les erreurs de quantification introduites sur les (kmax s - kπan+l s + l) coefficients spectraux de composantes ambiophoniques correspondant à des fréquences dans la bande s.ambiophonic R, then e ™ (k) are the quantization errors introduced on the (k max s - k πan + ls + l) spectral coefficients of ambiophonic components corresponding to frequencies in the band s.
Ainsi So
On effectue les hypothèses suivantes : les erreurs de quantification e™(k) sont des variables aléatoires indépendantes équi-distribuées selon l'indice k ; les erreurs de quantification e™(k) sont des variables aléatoires selon l'indice i ; le nombre d'échantillons dans une bande s est suffisamment grand ; le codeur 1 travaille à haute résolution.The following assumptions are made: the quantization errors e ™ (k) are independent random variables equi-distributed according to the index k; the quantization errors e ™ (k) are random variables according to the index i; the number of samples in a band s is large enough; the coder 1 works at high resolution.
Sous ces hypothèses et par application du théorème de la limite centrale, la puissance Pe m(s,i) de l'erreur de quantification, dans une sous- bande s et pour un signal S1 , tend, lorsque le nombre de coefficients dans une bande s augmente, vers une gaussienne dont la moyenne mum/ e λ et la variance σum, e λ sont données par les formules suivantes :Under these assumptions and by applying the central limit theorem, the power P e m (s, i) of the quantization error, in a subband s and for a signal S 1 , tends, when the number of coefficients in a band s increases, to a Gaussian whose mean m um / e λ and the variance σ um , e λ are given by the following formulas:
σrM = Σ E[er(*)4]-E[er(*)2]2 fc=fcm σr M = Σ E [e r ( * ) 4] -E [e r ( * ) 2] 2 fc = fc m
où la fonction E[x] délivre la moyenne de la variable x.where the function E [x] delivers the average of the variable x.
La contrainte « Prόbabïïité(Pe m(s,i) ≤ M™(s,i)) ≥ a » indiquée dans la formule 2 ci-dessus s'écrit alors à l'aide de la formule (3) suivante : mpr<s [) +β(a)σpr<s [) < M-(s,i)The constraint "Prefability (P e m (s, i) ≤ M ™ (s, i)) ≥ a" indicated in formula 2 above is then written using the following formula (3): m pr <s [) + β (a) σ pr <s [) <M- (s, i)
Avec : et la fonction Erf~ι (x) est l'inverse de la fonction d'erreur d'Euler.With: and the function Erf ~ ι (x) is the inverse of the error function of Euler.
Les variables e™(k) étant indépendantes selon l'indice i , on en déduit :The variables e ™ (k) being independent according to the index i, we deduce:
E[er(*)2] = ∑X E[Vf1C*)2]E [er (*) 2 ] = ΣXE [Vf 1 C *) 2 ]
Par conséquent, on obtient :Therefore, we get:
m^ = ∑ Σ% E[vrw2] = ∑% ∑ E[v;( kf] k=kmm s j=l J=I k=kmm s Les variables aléatoires e™ (k) étant indépendantes et équi-distribuées selon l'indice k , les variables aléatoires v™(k) sont également indépendantes et équi-distribuées selon l'indice k . Par conséquent : mP^rκ>£h>Mv> (Φ avec :m ^ = Σ Σ% E [vrw 2 ] = Σ% Σ E [v ; ( k f] k = k mm s j = l J = I k = k mm s The random variables e ™ (k) being independent and equi-distributed according to the index k, the random variables v ™ (k) are also independent and equi-distributed according to the index k. Therefore: P ^ r m κ> £ h> M v> (Φ with:
J\. — /l /C "T" J-J \. - / l / C "T" J-
On suppose que les puissances Pe m(s,i) d'erreur de quantification tendent vers des gaussiennes, alors :We assume that the powers P e m (s, i) of quantization error tend to Gaussian, then:
E[e:(k)4] = 3E[e: (k)2]2 D'où : Ainsi on peut écrire : E [e: (k) 4 ] = 3E [e: (k) 2 ] 2 Where: So we can write:
A partir de cette dernière équation, et en appliquant l'inégalité de Cauchy-Schwartz : From this last equation, and applying the Cauchy-Schwartz inequality:
Ce qui implique que : Which implies :
Par ailleurs, en haute résolution : In addition, in high resolution:
avec μγ représentant l'espérance mathématique de Y: 2 dans lawith μ γ representing the expected expectation of Y: 2 in the
2J sous bande s traitée et eR l'erreur d'arrondi propre à la fonction d'arrondi Arr. Si Arr(x) est par exemple la fonction fournissant l'entier le plus proche de la variable x, eR est égale à 0,5. Si Arr(x) est la fonction « partie entière » de la variable x, eR est égale à 1.2 J under treated band and e R the rounding error specific to the rounding function Arr. For example, if Arr (x) is the function providing the integer closest to the variable x, e R is equal to 0.5. If Arr (x) is the function "integer part" of the variable x, e R is equal to 1.
Ainsi la contrainte donnée par la formule (3) relative au signal S1 , i= 1 à N, sur une bande s, s'écrit sous la forme suivante : Thus the constraint given by the formula (3) relating to the signal S 1 , i = 1 to N, on a band s, is written in the following form:
II est ainsi possible, à partir de cette dernière équation, de déterminer si des coefficients d'échelle <r calculés par le module 4 de quantification pour coder les composantes de la transformée, permettent ou non de respecter le seuil de masquage tel que considéré dans le domaine du signal.It is thus possible, from this last equation, to determine whether scaling coefficients <r calculated by the quantization module 4 to code the components of the transform, allow or not to respect the masking threshold as considered in the signal domain.
Cette dernière équation représente une condition suffisante pour que le bruit correspondant au canal i soit masqué en sortie dans le domaine d'écoute.This last equation represents a sufficient condition for the noise corresponding to the channel i to be masked at the output in the listening domain.
Dans un mode de réalisation de l'invention, le module 4 de quantification est adapté pour déterminer à l'aide de cette dernière équation, pour un bloc m de trames courant, des coefficients d'échelle [BJ (s)) <r garantissant que le bruit dans le domaine d'écoute est masqué.In one embodiment of the invention, the quantization module 4 is adapted to determine using the latter equation, for a block m of current frames, scale coefficients [BJ (s)) <r guaranteeing that the noise in the listening domain is hidden.
Dans un mode de réalisation particulier de l'invention, le module 4 de quantification est adapté pour déterminer, pour un bloc m de trames courant, des coefficients d'échelle [BJ (s)) < < garantissant que le bruit dans le domaine d'écoute est masqué et en outre permettant de respecter une contrainte de débit.In a particular embodiment of the invention, the quantization module 4 is adapted to determine, for a block m of current frames, scaling coefficients [BJ (s)) << ensuring that the noise in the d domain listening is masked and further to respect a flow constraint.
Dans un mode de réalisation, les conditions à respecter sont les suivantes : rIn one embodiment, the conditions to be respected are the following:
- Minimiser le débit global Dm = ∑DJ- Minimize the overall flow D m = ΣDJ
- Sous la contrainte : ≤ M?(s,i) pour toute bande s, avec D; m le débit global attribué à la composante ambiophonique Y1.- Under duress : ≤ M? (S, i) for any band s, with D ; m the overall bit rate assigned to the surround component Y 1 .
On peut écrire que :We can write that:
où D™{s) est le débit attribué à la composante ambiophonique Y1 dans la bande s.where D ™ (s) is the bit rate assigned to the surround component Y 1 in the s band.
Minimiser le débit global Dm revient donc à minimiser le débit rMinimizing the overall flow D m is therefore to minimize the flow rate r
Dm (s) = ∑D™ (s) dans chaque bande s. Dans une première approximation, onD m (s) = ΣD ™ (s) in each band s. In a first approximation, we
peut écrire que le débit attribué à une composante ambiophonique dans une bande s est une fonction logarithmique du coefficient d'échelle, soit : can write that the bit rate assigned to an ambiophonic component in a band s is a logarithmic function of the scale coefficient, ie:
La nouvelle fonction à minimiser s'écrit donc sous la forme suivante : The new function to be minimized is written in the following form:
Pour résoudre le problème de quantification par bande en minimisant le débit global sous la contrainte (3), il faut donc minimiser la fonction F sous la contrainte (3).To solve the band quantization problem by minimizing the overall rate under the constraint (3), we must therefore minimize the function F under the constraint (3).
La résolution de ce problème d'optimisation sous contrainte est par exemple effectuée à l'aide de la méthode des Lagrangiens. La fonction Lagrangienne s'écrit sous la forme suivante :The resolution of this constrained optimization problem is for example carried out using the Lagrangian method. The Lagrangian function is written in the following form:
1616
L(B,λ) = -∑ln(β; (*)) + ∑4 K —E[eR 2](l + y[2β(a))∑(hï]B;'t(Syμ1 (s)) -M?(s,i)L (B, λ) = -Σln (β; (*)) + Σ4 K -E [e R 2 ] (1 + y [2β (a)) Σ (h 1 ) B; t ( S y 1 (s)) -M? (s, i)
/=1 I=I J-I 2'1 / = 1 I = I JI 2 ' 1
L(B,λ) = -∑in(β; (,))+Δ7 (λ)β; (,)f -∑ΛM™ (,,0 L (B, λ) = -Σin (β; (,)) + Δ7 (λ) β; (,) f -ΣΛM ™ (,, 0
;=1 1=1= 1 1 = 1
Avec : 16With: 16
Δ7 (λ) = //1 (S)K, -±E[eî](l + j2fl(.ay)∑%AΔ7 (λ) = / / 1 ( S ) K, - ± E [ei] (l + j2fl ( . Ay) Σ% A
2J ι=l et les valeurs X3 , l ≤ j ≤ N , sont les coordonnées du vecteur de2 J ι = 1 and the values X 3 , l ≤ j ≤ N, are the coordinates of the vector of
Lagrange λ .Lagrange λ.
La mise en œuvre de la méthode des Lagrangiens permet d'écrire tout d'abord que, pour 1 ≤ j < r :The implementation of the method of Lagrangians allows to write first that for 1 ≤ j <r:
B; (S) = ^—B; ( S ) = ^ -
2 Δ; (λ)2 Δ; (Λ)
On remplace par ces termes les coefficients d'échelle dans l'équation de Lagrange. Et on cherche alors à déterminer la valeur du vecteur de Lagrange λ qui maximise la fonction ω{λ) = L{{B™ {s),...,B™{s)),λ) , par exemple à l'aide de la méthode du gradient de la fonction ω .These terms are used to replace the scale coefficients in the Lagrange equation. And we then try to determine the value of the Lagrange vector λ which maximizes the function ω {λ) = L {{B ™ {s), ..., B ™ (s)), λ), for example at using the gradient method of the function ω.
D'après la méthode du gradient d'Uzawa Ww(X) , oùAccording to the Uzawa Ww (X) gradient method, where
les dérivées partielles ne sont autres que les contraintes calculées pour les B^ (S) = --1 the partial derivatives are none other than the computations calculated for B ^ (S) = - 1
I tT J (X)I tT J ( X)
On utilise la méthode itérative de gradient relatif (cf. notamment le document Derrien) pour résoudre ce système.The iterative relative gradient method (see in particular the Derrien document) is used to solve this system.
L'équation générale (formule (4)) de mise à jour du vecteur de Lagrange lors d'une (k+1 )ιeme itération de la méthode s'écrit alors sous la forme suivante : λi+1 - λ* ® (l + pm ® Vω(λk )) avec le vecteur de Lagrange λ avec un exposant (k+1 ) indiquant le vecteur actualisé et le vecteur de Lagrange λ avec un exposant k indiquant le vecteur calculé précédemment lors de la kιeme itération, ® désignant le produit terme à terme entre deux vecteurs de même taille, p désignant le pas de l'algorithme itératif et m étant un vecteur de pondération.The general equation (formula (4)) for updating the Lagrange vector during a (k + 1) th iteration of the method is then written in the following form: λ i + 1 - λ * ® ( l + pm ® Vω (λ k )) with the Lagrange vector λ with an exponent (k + 1) indicating the updated vector and the Lagrange vector λ with an exponent k indicating the vector calculated previously during the k th iteration, ® designating the product term term between two vectors of the same size, p designating the pitch of the iterative algorithm and m being a weighting vector.
Dans un mode de réalisation, de manière à assurer la convergence de la méthode itérative, on choisit le vecteur m égal à :In one embodiment, in order to ensure the convergence of the iterative method, the vector m is chosen equal to:
Dans le mode de réalisation considéré, le module 4 de quantification est adapté pour mettre en œuvre les étapes du procédé décrit ci-dessous en référence à la figure 3 sur chaque bande de quantification s lors de la quantification d'un bloc m de signaux (St )1≤ι≤N .In the embodiment considered, the quantization module 4 is adapted to implement the steps of the method described below with reference to FIG. 3 on each quantization band s during the quantization of a block m of signals ( S t ) 1≤ι≤N .
Le procédé est basé sur un algorithme itératif comprenant des instructions pour mettre en œuvre les étapes décrites ci-dessous lors de l'exécution de l'algorithme sur des moyens de calcul du module 4 de quantification.The method is based on an iterative algorithm comprising instructions for implementing the steps described below during the execution of the algorithm on calculation means of the quantization module 4.
Dans une étape a/ d'initialisation (k=0) : on définit la valeur du pas d'itération p , une valeur D représentant un seuil de débit et la valeur des coordonnées (/L1 ...λN) du vecteur de Lagrange initial avec λ} = λ° , 1 ≤ j ≤ N .In a step a / of initialization (k = 0): the value of the iteration step p is defined, a value D representing a rate threshold and the value of the coordinates (/ L 1 ... λ N ) of the vector initial Lagrange with λ } = λ °, 1 ≤ j ≤ N.
Les étapes de la boucle itérative pour une (k+1 )eme itération, avec k entier supérieur ou égal à 0, sont les suivantes.The steps of the iterative loop for a (k + 1) th iteration, with k integer greater than or equal to 0, are as follows.
Dans une étape b/, les valeurs des coordonnées λ} , l ≤ j ≤ N du vecteur de Lagrange considérées étant celles calculées précédemment lors de la kιeme itération, on calcule pour l ≤ j ≤ N : In a step b /, the values of the coordinates λ } , l ≤ j ≤ N of the Lagrange vector considered being those calculated previously during the k th iteration, we calculate for l ≤ j ≤ N:
Puis dans une étape c/, on calcule les coefficients d'échelle, pour l ≤ j ≤ r : Then in a step c /, the scaling coefficients are calculated, for l ≤ j ≤ r:
Dans une étape d/, on calcule la valeur de la fonction F sur la bande s, représentant le débit correspondant pour la bande s :In a step d /, the value of the function F is calculated on the band s, representing the corresponding bit rate for the band s:
FW =-ΣM*;«) F W = -ΣM *; ")
Dans une étape e/, on compare la valeur F (s) calculée avec le seuil donné D.In a step e /, the calculated value F (s) is compared with the given threshold D.
Si la valeur FO) calculée est supérieure au seuil donné D, on calcule, dans une étape il, la valeur du vecteur de Lagrange λ pour la (k+1 )eme itération à l'aide de l'équation (4) indiquée ci-dessus et du vecteur de Lagrange calculé lors de la keme itération.If the value FO) calculated is greater than the given threshold D is calculated in a step there, the value of the Lagrange λ vector for the (k + 1) th iteration using Equation (4) shown below -Dessus and Lagrange vector calculated at the kth iteration.
Puis, dans une étape g/, on incrémente l'indice k d'une unité et on réitère les étapes b/, c/, d/ et e/.Then, in a step g /, the index k is incremented by one unit and the steps b /, c /, d / and e / are repeated.
Si la valeur FO) calculée à l'étape e/, est inférieure au seuil donné D, on stoppe les itérations. On a alors déterminé des coefficients d'échelle [BJ (s))^ <r pour la bande de quantification s permettant de masquer, dans le domaine d'écoute, le bruit dû à la quantification dans la bande s, des composantes ambiophoniques (^ )1 , tout en garantissant que le débit nécessaire pour cette quantification dans la bande s est inférieur à une valeur déterminée, fonction de D.If the value FO) calculated in step e / is less than the given threshold D, the iterations are stopped. Scale coefficients [BJ (s)) ^ <r were then determined for the quantization band s to mask, in the listening domain, the noise due to the s-band quantization of the surround components ( ^) 1 , while guaranteeing that the bit rate necessary for this quantification in the band s is less than a determined value, a function of D.
On applique ensuite la fonction de quantification ainsi déterminée pour les bandes s respectives et composantes ambiophoniques respectives aux coefficients spectraux des composantes ambiophoniques. Les indices de quantification ainsi que des éléments de définition de la fonction de quantification sont fournis au module 5 de codage de Huffman.The quantization function thus determined for the respective s-bands and respective surround components is then applied to the spectral coefficients of the surround components. The quantization indices as well as definition elements of the quantization function are provided to the Huffman coding module.
Les données de codage délivrées par le module 5 de codage de Huffman sont ensuite transmises sous forme de flux binaire Φ au décodeur 100. Opérations réalisées au niveau du décodeur :The coding data delivered by the Huffman coding module 5 is then transmitted as a bit stream Φ to the decoder 100. Operations performed at the decoder:
Le module 101 de lecture de séquence binaire est adapté pour extraire des données de codage présentes dans le flux Φ reçu par le décodeur et en déduire, dans chaque bande s, des indices de quantification i(k) et des coefficients d'échelle (B™ (s))ι≤]≤r .The bit sequence reading module 101 is adapted to extract coding data present in the stream Φ received by the decoder and to deduce, in each band s, quantization indices i (k) and scale coefficients (B ™ (s)) ι≤] ≤r .
Le module de quantification inverse 102 est adapté pour déterminer les coefficients spectraux, relatifs à la bande s, des composantes ambiophoniques correspondants en fonction des indices de quantification i(k) et des coefficients d'échelles (B™ (s))ι≤]≤r dans chaque bande s.The inverse quantization module 102 is adapted to determine the spectral coefficients, relative to the band s, of the corresponding ambiophonic components as a function of the quantization indices i (k) and the scale coefficients (B ™ (s)) ι] ≤r in each band s.
Ainsi un coefficient spectral YJ t relatif à la fréquence Ft élément de la bande s de la composante ambiophonique Y1 et représenté par l'indice de quantification i(k) est restitué par le module 102 de quantification inverse àThus a spectral coefficient Y j t relating to the frequency F t of the band s component of the ambiophonic component Y 1 and represented by the quantization index i (k) is restored by the inverse quantization module 102.
4 l'aide de la formule suivante : YJ t = AJ (s) i(k)3 4 using the following formula: Y J t = AJ (s) i (k) 3
Un décodage ambiophonique est ensuite appliqué aux r composantes ambiophoniques décodées, de manière à déterminer Q' signaux S'i, S'2, ..., S'Q< destinés aux Q' haut-parleurs H1 , H2 ..., HQ'.Ambiophonic decoding is then applied to the decoded surround components, so as to determine the signals S'i, S ' 2 , ..., S'Q < for the Q' speakers H1, H2 ..., HQ .
Le bruit de quantification à la sortie du décodeur 100 est une constante qui ne dépend que de la transformée R utilisée et du module 4 de quantification car les données psychoacoustiques utilisées lors du codage ne prennent pas en considération les traitements effectués lors de la restitution par le décodeur. En effet, le modèle psychoacoustique ne prend pas en compte les interactions acoustiques entre les différents signaux, mais calcule la courbe de masquage d'un signal comme s'il était le seul écouté. L'erreur calculée sur ce signal reste donc constante et masquée pour toute matrice de décodage ambiophonique utilisée. Cette matrice de décodage ambiophonique va simplement modifier la distribution de l'erreur sur les différents haut-parleurs en sortie. The quantization noise at the output of the decoder 100 is a constant which depends only on the transform R used and the quantization module 4 because the psychoacoustic data used during the coding do not take into consideration the processing performed during the rendering by the processor. decoder. Indeed, the psychoacoustic model does not take into account the acoustic interactions between the different signals, but calculates the masking curve of a signal as if it were the only one listened to. The error calculated on this signal therefore remains constant and masked for any surround decoding matrix used. This surround decoding matrix will simply change the distribution of the error on the different speakers output.

Claims

REVENDICATIONS
1. Procédé de quantification de composantes, certaines au moins desdites composantes ((^ )1< <r ) étant déterminées chacune en fonction d'une pluralité de signaux audio ) d'une scène sonore et calculables par application d'une transformation linéaire multicanal sur lesdits signaux audio, selon lequel on détermine une fonction de quantification (Qm) à appliquer audites composantes dans une bande de fréquence donnée (s) en testant une condition relative à au moins un signal audio ( S1 ) et dépendant au moins d'une comparaison effectuée entre :A method of quantizing components, at least some of said components (( 1 ) 1 <<r ) being each determined according to a plurality of audio signals ) of a sound scene and calculable by applying a multichannel linear transformation on said audio signals, according to which a quantization function (Q m ) is determined to apply audits components in a given frequency band (s) by testing a condition relating to at least one audio signal (S 1 ) and depending at least on a comparison between:
- un seuil de masquage psychoacoustique (M™ (s,ï) ) relatif au signal audio dans la bande de fréquence donnée, eta psychoacoustic masking threshold (M ™ (s, ï)) relating to the audio signal in the given frequency band, and
- une valeur déterminée en fonction de la transformation linéaire multicanal inverse et d'erreurs de quantification des composantes par ladite fonction sur la bande de fréquence donnée.a value determined as a function of the inverse multi-channel linear transformation and quantization errors of the components by said function on the given frequency band.
2. Procédé selon la revendication 1 , selon lequel la condition est relative à plusieurs signaux audio et dépend de plusieurs comparaisons, chaque comparaison étant effectuée entre un seuil de masquage psychoacoustique relatif à un signal audio respectif dans la bande de fréquence donnée, et une valeur déterminée en fonction de la transformation linéaire multicanal inverse et d'erreurs de quantification des composantes par ladite fonction.2. Method according to claim 1, wherein the condition is relative to several audio signals and depends on several comparisons, each comparison being made between a psychoacoustic masking threshold relative to a respective audio signal in the given frequency band, and a value determined according to the inverse multi-channel linear transformation and quantization errors of the components by said function.
3. Procédé selon la revendication 1 ou la revendication 2, selon laquelle la détermination de la fonction de quantification (Qm) est réitérée lors de l'actualisation des valeurs des composantes à quantifier.3. Method according to claim 1 or claim 2, wherein the determination of the quantization function (Q m ) is reiterated when updating the values of the components to be quantized.
4. Procédé selon l'une quelconque des revendications précédentes, selon lequel on teste la condition relative à un signal audio au moins en comparant le seuil de masquage psychoacoustique relatif au signal audio et un élément représentant la valeur mathématique4. Method according to any one of the preceding claims, according to which one tests the condition relating to an audio signal at least in comparing the psychoacoustic masking threshold relative to the audio signal and an element representing the mathematical value
11
∑(h?]B] (s)2μι (s)) , où s est la bande de fréquences donnée, r est leΣ (h? ] B ] (s) 2 μ ι (s)), where s is the given frequency band, r is the
nombre de composantes, hl } est le coefficient de la transformée linéaire multicanal inverse relatif au signal audio (Si) et à la jιeme composante avec j=1 à r, B}(s) représente un paramètre caractérisant la fonction de quantification (Qm) dans la bande s relative à la jιeme composante et μι (s) est l'espérance mathématique dans la bande s de la racine carrée de la jιeme composante.number of components, h l} is the coefficient of the inverse multichannel linear transform relative to the audio signal (Si) and to the j th component with j = 1 to r, B } (s) represents a parameter characterizing the quantization function ( Q m) in the band on the s j and μ ι ιeme component (s) is the expected value in the strip s of the square root of the j ιeme component.
5. Procédé selon l'une quelconque des revendications précédentes, selon lequel on détermine une fonction de quantification à appliquer audites composantes dans la bande de fréquence donnée à l'aide d'un processus itératif générant à chaque itération un paramètre de la fonction de quantification candidat vérifiant la condition et associé à un débit correspondant, l'itération étant stoppée lorsque le débit est inférieur à un seuil donné.A method according to any one of the preceding claims, wherein a quantization function is determined to be applied to said components in the given frequency band using an iterative process generating at each iteration a parameter of the quantization function. candidate checking the condition and associated with a corresponding rate, the iteration being stopped when the flow is below a given threshold.
6. Procédé selon l'une quelconque des revendications précédentes, selon lequel la transformation linéaire multicanal est une transformation ambiophonique.The method of any preceding claim, wherein the multichannel linear transformation is an ambiophonic transformation.
7. Module (4) de quantification adapté pour quantifier au moins des composantes ((F7 ) ) déterminées chacune en fonction d'une pluralité de signaux audio ) d'une scène sonore et calculables par application d'une transformation linéaire multicanal sur lesdits signaux audio, ledit module de quantification étant adapté pour mettre en œuvre les étapes d'un procédé selon l'une quelconque des revendications 1 à 6. 7. Quantization module (4) adapted to quantify at least components ((F 7 )) each determined according to a plurality of audio signals ) of a sound scene and calculable by applying a multichannel linear transformation on said audio signals, said quantization module being adapted to implement the steps of a method according to any one of claims 1 to 6.
8. Codeur audio (1) adapté pour coder une scène audio comprenant plusieurs signaux audio respectifs ((^; ) < < ) en un flux binaire de sortie (Φ), comprenant : un module (3) de transformation adapté pour calculer par application d'une transformation linéaire multicanal sur lesdits signaux audio, des composantes ((F; )1< <r ) dont au moins certaines sont déterminées chacune en fonction d'une pluralité des signaux audio ; et un module (4) de quantification selon la revendication 7 adapté pour déterminer au moins une fonction de quantification (Qm) sur au moins une bande de fréquence donnée (s) et pour quantifier les composantes sur la bande de fréquence donnée en fonction d'au moins la fonction de quantification déterminée ; ledit codeur étant adapté pour constituer un flux binaire en fonction au moins de données de quantification délivrées par le module de quantification.An audio encoder (1) adapted to encode an audio scene comprising a plurality of respective audio signals ((^ ; ) << ) into an output bit stream (Φ), comprising: a transform module (3) adapted to compute per application of a multichannel linear transformation on said audio signals, components ((F ; ) 1 <<r ), at least some of which are each determined according to a plurality of the audio signals; and a quantization module (4) according to claim 7 adapted to determine at least one quantization function (Q m ) over at least one given frequency band (s) and to quantize the components on the given frequency band as a function of at least the determined quantization function; said encoder being adapted to constitute a bit stream according to at least quantization data delivered by the quantization module.
9. Programme d'ordinateur à installer dans un module (4) de quantification, ledit programme comprenant des instructions pour mettre en œuvre les étapes d'un procédé selon l'une quelconque des revendications 1 à 6 lors d'une exécution du programme par des moyens de traitement dudit module.9. Computer program to be installed in a quantization module (4), said program comprising instructions for implementing the steps of a method according to any one of claims 1 to 6 during execution of the program by means for processing said module.
10. Données de codage (Φ), déterminées suite à la mise en œuvre d'un procédé de quantification selon l'une quelconque des revendications 1 à 6. 10. Coding data (Φ), determined following the implementation of a quantization method according to any one of claims 1 to 6.
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