EP1383109A1 - Method and device for wide band speech coding - Google Patents

Abstract

The product of the word extracted by long term excitation with the associated long term gain (Ga) is added (SM) to the product of the word extracted by short term excitation with the associated short term gain (Gc). The dataword is filtered using a low-pass filter having a cut-off frequency which is greater than 25 % of the sampling frequency, but less than 50 % of the latter. The filtered code is then combined with the adaptive coded index. Speech is sampled in order to obtain a succession of speech frames, each one comprising a pre-determined number of samples. The parameters for a code excitation-based linear prediction model are determined for each speech frame and these parameters comprise a long term excitation dataword (vi) extracted from an associated coded index (DLT) and an associated short term gain, as well as a short term excitation dataword (cj) extracted from an algebraic coded index (DCT) and an associated short term gain, providing an adaptive coded index obtained from the long term excitation extracted word and the short term excitation extracted word. Independent claims are also included for the following: (1) Speech coding device (CD) for carrying out this method; and (2) Wireless communication terminal provided with the coding device.

Description

The invention relates to speech encoding / decoding extended band, in particular but not limited to telephony mobile.

In wideband, the bandwidth of the speech signal is between 50 and 7000 Hz.

Successive speech sequences sampled at one predetermined sampling frequency, for example 16 kHz, are processed in a coding device using a prediction linear excitation by coded sequences (ACELP: “algebraic-code-excited linear-prediction ”), well known to those skilled in the art, and described in particular in recommendation ITU-TG 729, version 3/96, titled “speech coding at 8 kbit / s by prediction linear with excitation by coded sequences with algebraic structure conjugate ”.

We will now briefly recall, referring to the Figure 1, the main features and functionality of such coder, the skilled person can refer for all practical purposes, for more details, see the above-mentioned recommendation G 729.

The prediction coder CD, of the ACELP type, is based on the linear predictive coding model with code excitation. The coder operates on vocal superframes equivalent for example to 20 ms of signal and each comprising 320 samples. The extraction of the linear prediction parameters, i.e. the coefficients of the linear prediction filter also called short-term synthesis filter 1 / A (z), is carried out for each speech superframe. On the other hand, each superframe is subdivided into 5 ms frames comprising 80 samples. All the frames, the voice signal is analyzed to extract the parameters of the CELP prediction model (that is to say, in particular, a long-term digital excitation word V _i extracted from an adaptive coded directory DLT, also called "adaptive long-term dictionary", an associated long-term gain Ga, a short-term excitation word C _j , extracted from an algebraic coded directory DCT, also known as "fixed coded directory" or "short dictionary algebraic term ", and an associated short-term gain Gc).

These parameters are then coded and transmitted.

On reception, these parameters are used, in a decoder, to retrieve the excitation and predictive filter parameters. We then reconstitutes speech by filtering this excitation flow in a short-term synthesis filter.

While the Adaptive DLT Dictionary Contains Words digital representative of representative tonal delays of past excitations, the short-term dictionary DCT is founded on an algebraic structure using a permutation model intertwined with Dirac pulses. In this coded directory, which contains innovative excitations also called excitations algebraic or short-term, each vector contains a certain number of non-zero pulses, for example four, each of which can have amplitude +1 or -1 with predetermined positions.

The CD encoder processing means include functionally of the first MEXT1 extraction means intended to extract the word long-term excitement, and second MEXT2 extraction means intended to extract the word short-term excitement. Functionally, these means are made for example in software within a processor.

These extraction means include a predictive filter FP having a transfer function equal to 1 / A (z), as well as a filter FPP perceptual weighting with a transfer function W (z). The perceptual weighting filter is applied to the signal to model the perception of the ear.

Furthermore, the extraction means include means MECM intended to perform a minimization of a square error average.

The linear prediction FP synthesis filter models the spectral envelope of the signal. Linear predictive analysis is performed all superframes, so as to determine the linear predictive filter coefficients. These are converted to spectral line pairs (LSP: “Line Spectrum Pairs”) and digitized by predictive vector quantization in two stages.

Each 20 ms speech superframe is divided into four frames of 5 ms each containing 80 samples. The settings Quantized LSPs are transmitted to the decoder once per superframe while long term and short term parameters are passed at each frame.

The coefficients of the linear prediction filter, quantified and not quantified, are used for the most recent frame of a super-frame, while the other three frames of the same super-frame use an interpolation of these coefficients. Tonal delay open loop is estimated every two frames based on the perceptually weighted voice signal. Then, the following operations are repeated at each frame:

The long-term target signal X _LT is calculated by filtering the sampled speech signal s (n) by the perceptual weighting filter FPP.

We then subtract the response to null input of the weighted synthesis filter FP, FPP, so that get a new long-term target signal.

The impulse response of the weighted synthesis filter is calculated.

A closed loop tonal analysis using a minimization of the mean square error is then carried out in order to determine the long-term excitation word v _i and the associated gain Ga, by means of the target signal and the impulse response, by searches around the value of the tone delay in open loop.

The long-term target signal is then updated by subtracting the filtered contribution y from the adaptive coded directory DLT and this new short-term target signal X _ST is used when exploring the fixed coded directory DCT in order to determine the password. short term excitation c _j and the associated gain G _c . Again, this closed loop search is performed by minimizing the mean square error.

Finally, the long-term adaptive DLT dictionary as well that the memories of the filters FP and FPP, are updated by means of the long term and short term excitation words so determined.

The quality of a CELP algorithm strongly depends on the richness of the DCT algebraic excitation dictionary. If the efficiency of such an algorithm is indisputable for band signals narrow bandwidth (300-3400 Hz), problems arise for wideband signals.

1) une qualité globale de parole reconstruite totalement insuffisante (la parole reconstruite manque de présence, le niveau d'énergie est très variable, le timbre de la voix est méconnaissable, ...),

2) un signal reconstruit corrompu par trois sortes de bruits :

• un bruit harmonique en haute fréquence (comb-like noise),
• un fort bruit à haute fréquence, tel un bruit de quantification,
• un bruit en basse fréquence (rumbling noise), tel qu'un balai de paille frappé à intervalles réguliers sur le sol.

The inventors have indeed observed that even with a very rich algebraic dictionary, the speech encoding algorithm produces two types of problems:

1) an overall quality of reconstructed speech totally insufficient (the reconstructed speech lacks presence, the energy level is very variable, the timbre of the voice is unrecognizable, ...),

2) a reconstructed signal corrupted by three kinds of noise:

• harmonic noise at high frequency (comb-like noise),
• strong noise at high frequency, such as quantization noise,
• a low frequency noise (rumbling noise), such as a straw broom struck at regular intervals on the ground.

It was then observed that an improvement in the quality overall speech could be achieved by elimination partial or total of these noises.

An object of the invention is to reduce the harmonic noise and the high frequency noise.

The invention also aims to suppress noise from "whistling" type tainting the voiced speech frames.

Another object of the invention is to independently control short-term and long-term distortions.

The invention therefore provides a speech encoding method with wide band, in which the speech is sampled so as to obtain successive voice frames each comprising a predetermined number of samples, and for each voice frame, we determines parameters of a linear prediction model at excitation by code, these parameters comprising a numeric word of long-term excitement extracted from an adaptive coded repertoire and a associated long-term gain, as well as a word of short-term excitement extract from an algebraic coded repertoire and short-term gain associated, and we update the adaptive coded directory from the word excerpt long term excitement and short term excitement word extract.

According to a general characteristic of the invention, we sum the product of the word extracted from long-term excitation by long-term gain associated term with the product of the excerpt short word term by the associated short-term gain, we filter the digital word summed in a low pass filter having a cutoff frequency greater than a quarter of the sampling frequency and less than half of it, and we update the adaptive coded directory with the filtered word.

The invention here uses a "total correction" filter which combines a harmonic noise correction filter and a high frequency correction.

The invention thus improves the quality during voiced speech frames. Furthermore, the complexity of the encoder is reduced by merging the filter of harmonic correction and the high frequency correction filter.

The invention differs in particular from a solution described in an article by Kroon and Atal, entitled "Strategies for Improving the Performance of CELP Coders at Low Bit Rates ”, Proc., IEEE, Int. Conf. Acoustics, Speech, and Signal Processing, ICASSP'88, New York, USA, 1988, Pages 151-154, which offers filtering of adaptive dictionary made at the output of this dictionary and not not at the entrance according to the invention.

Thus, the prefiltering of the adaptive dictionary according to the invention presents in relation to the post-filtering of the article of Kroon and Atal, the advantage that filtering is taken into account when error minimization performed to choose excitation adaptive to the next frame. This is not the case for the solution of Kroon and Atal, since the proposed filtering takes place on the excitation chosen next. Also, to take into account the filtering in the minimization of the error, it would then be necessary to increase considerably complexity and filter out any excitement to be tested.

According to a preferred embodiment, the summed word with a finite impulse response digital filter at linear phase having an order at least equal to 10. For example, when the sampling frequency is 16 kHz, you can choose a filter of order 20 having a cut-off frequency of around 6 kHz.

Although the quality of the speech is thus improved, the voiced speech frames still seem to be corrupted by “hissing” noise. This high frequency noise comes from short term excitation which introduces unwanted artifacts. Two types of solutions to solve this problem have already been proposed in the literature. A first type of solution, described for example
in the article by Gerson and Jasiuk, entitled "Techniques for Improving the Performance of CELP-Type Speech Coders", IEEE, Journal on Selected Areas In Communications, Vol. 10, N ° 5, June 1992, pages 858-865, or else
in the article by Miki et al., entitled "A Pitch Synchronous Innovation CELP (PSI-CELP) Coder for 2-4 kbit / s", Proc. IEEE Int. Conf. Acoustics, Speech, and Signal Processing, ICASSP'94, Adelaide, South Australia, 1994, Vol. II, pages 113-116,
proposes to make the short-term contribution periodic.

Another type of solution, described for example
in the article by Taniguchi Johnson and Ohta, entitled “Pitch Sharpening for Perceptually Improved CELP, and the Sparse-Delta Codebook for Reduced Computation”, Proc. IEEE Int. Conf. Acoustics, Speech, and Signal Processing, ICASSP'91, Toronto, Canada, 1991, pages 241-244, or
in Shoham's article, "Constrained-Stochastic Excitation Coding of Speech At 4.8 kb / s", Advances in Speech Coding, BS Atal, V. Cuperman, and A. Gersho, Eds., Dordrecht, The Netherlands, Kluwer, 1991, pages 339-348,
offers adaptive short-term gain control.

The invention also provides a control type solution gain, but totally different from that described in particular in the articles of Taniguchi and others and of Shoham.

More specifically, according to an embodiment of the invention, the extraction of the short term excitation word comprises digital linear prediction filtering and the method includes an update of the state of the linear prediction filter with the word short-term excitation filtered by a filter whose coefficients depend on the value of the long-term gain, so that weaken the contribution of short-term excitement when the gain long-term excitement is above a predetermined threshold, for example equal to 1.

In other words, the solution according to the invention consists here to weaken the contribution of short-term excitement if the gain of long-term excitement is important. However, this is the contribution of undiminished short-term excitement which is stored in the adaptive dictionary for updating. So the reduction occurs only on exit. Preserving the magnitude of the short-term contribution to be stored is important, since the richness of the adaptive dictionary is thus preserved for the lowest frequencies.

Of course, the gain correction must also be applied when reconstructing the signal at the decoder.

This filter can be of order 0 or of higher or equal order to 1. In the latter case, the filter of order greater than or equal to 1 can be finite impulse response.

According to an embodiment of the invention, in which filter is of order 1 and its transfer function equal to B0 + B1 z ^-1 , the first coefficient B0 of the filter is equal to 1 / (1 + β. min (Ga, 1)), and the second coefficient B1 of the filter is equal to β.min (Ga, 1) / (1 + β.min (Ga, 1)), where β is a real number with a lower absolute value at 1, Ga is the long-term gain and min (Ga, 1) designates the minimum value between Ga and 1.

According to another variant of the invention, which can be taken into account combination or independently of the previous variant, we performs long term excitation word extraction using a first perceptual weighting filter comprising a first formantic weighting filter, and we extract the word short-term excitation using the first filter perceptual weighting cascaded to a second filter of perceptual weighting including a second filter formantic weighting. The denominator of the transfer function of the first formantic weighting filter is equal to the numerator of the second formantic weighting filter.

Thus, according to this variant, the use of two filters of weighting different formant allows to control regardless of short-term and long-term distortions. The short-term weighting filter is cascaded to the filter of long-term weighting. Also, tying the denominator of the long-term weighting filter in the numerator of the short-term weighting allows these two to be controlled separately filters and also allows a clear simplification when these two filters are cascaded.

Of course, when this variant is used in combination with the gain control variant there is a update the status of the two perceptual weighting filters with the short term excitation word filtered by the order filter greater than or equal to 1.

• des moyens d'échantillonnage aptes à échantillonner la parole de façon à obtenir des trames vocales successives comportant chacune un nombre prédéterminé d'échantillons,
• des moyens de traitement aptes à chaque trame vocale, à déterminer des paramètres d'un modèle de prédiction linéaire à excitation par code, ces moyens de traitement comportant des premiers moyens d'extraction aptes à extraire un mot numérique d'excitation à long terme d'un répertoire codé adaptatif et à calculer un gain à long terme associé, et des deuxièmes moyens d'extraction aptes à extraire un mot d'excitation à court terme d'un répertoire codé algébrique et à calculer un gain à court terme associé, et
• des premiers moyens de mise à jour aptes à mettre à jour le répertoire codé adaptatif à partir du mot d'excitation à long terme extrait et du mot d'excitation à court terme extrait. Selon une caractéristique générale de l'invention, les premiers moyens de mise à jour comportent
• des premiers moyens de calcul aptes à sommer le produit du mot extrait d'excitation à long terme par le gain à long terme associé, avec le produit du mot extrait d'excitation à court terme par le gain à court terme associé, de façon à délivrer un mot numérique sommé, et
• un filtre passe-bas ayant une fréquence de coupure supérieure au quart de la fréquence d'échantillonnage et inférieure à la moitié de celle-ci, et connecté entre la sortie des premiers moyens de calcul et le répertoire codé adaptatif de façon à mettre à jour ce répertoire adaptatif avec le mot filtré.

The subject of the invention is also a device for encoding speech with a wide band, comprising

• sampling means able to sample the speech so as to obtain successive speech frames each comprising a predetermined number of samples,
• processing means suitable for each voice frame, in determining parameters of a linear prediction model with code excitation, these processing means comprising first extraction means capable of extracting a digital word of long-term excitation d '' an adaptive coded repertoire and calculating an associated long-term gain, and second extraction means capable of extracting a short-term excitation word from an algebraic coded repertoire and calculating an associated short-term gain, and
• first updating means capable of updating the adaptive coded directory on the basis of the extracted long-term excitation word and the extracted short-term excitation word. According to a general characteristic of the invention, the first updating means comprise
• first calculation means capable of summing the product of the extracted word of long-term excitation by the associated long-term gain, with the product of the extracted word of short-term excitation by the associated short-term gain, so as to issue a summed digital word, and
• a low-pass filter having a cutoff frequency greater than a quarter of the sampling frequency and less than half of it, and connected between the output of the first calculation means and the adaptive coded directory so as to update this adaptive directory with the filtered word.

According to one embodiment of the invention, the first extraction means include a digital prediction filter linear, and by the fact that the device comprises second updating means capable of updating the state of the linear prediction filter with short term excitation word filtered by a filter whose coefficient (s) depend on the value long-term gain, so as to weaken the contribution of short-term excitement when gaining long-term excitement is above a predetermined threshold.

According to another embodiment of the invention, the first extraction means include a first filter perceptual weighting including a first weighting filter formantic, by the fact that the second means of extraction include the first perceptual weighting filter cascaded to a second perceptual weighting filter comprising a second formantic weighting filter, and by the fact that the denominator of the transfer function of the first filter of formantic weighting is equal to the numerator of the second filter formantic weighting.

The invention also relates to a terminal of a system wireless communication, such as a mobile phone cell, incorporating a device as defined above.

• la figure 1, déjà décrite, illustre schématiquement un dispositif d'encodage de la parole, selon l'art antérieur ;
• la figure 2 illustre schématiquement un premier mode de réalisation d'un dispositif d'encodage, selon l'invention ;
• la figure 3 illustre schématiquement un deuxième mode de réalisation d'un dispositif d'encodage, selon l'invention, et la figure 3a illustre schématiquement un mode de réalisation d'un décodeur correspondant ;
• la figure 4 illustre schématiquement un troisième mode de réalisation d'un dispositif d'encodage, selon l'invention ;
• la figure 5 illustre schématiquement un quatrième mode de réalisation d'un dispositif d'encodage, selon l'invention ; et
• la figure 6 illustre schématiquement l'architecture interne d'un téléphone mobile cellulaire incorporant un dispositif de codage, selon l'invention.

Other advantages and characteristics of the invention will appear on examining the detailed description of embodiments and implementation, in no way limiting, and the appended drawings, in which:

• Figure 1, already described, schematically illustrates a speech encoding device, according to the prior art;
• Figure 2 schematically illustrates a first embodiment of an encoding device according to the invention;
• Figure 3 schematically illustrates a second embodiment of an encoding device according to the invention, and Figure 3a schematically illustrates an embodiment of a corresponding decoder;
• Figure 4 schematically illustrates a third embodiment of an encoding device according to the invention;
• FIG. 5 schematically illustrates a fourth embodiment of an encoding device according to the invention; and
• FIG. 6 schematically illustrates the internal architecture of a cellular mobile telephone incorporating a coding device according to the invention.

The encoding device, or encoder, CD, according to the invention, as illustrated in Figure 2, differs from that of the prior art as illustrated in FIG. 1 by the fact that the MAJ means of DLT adaptive long-term dictionary update feature a total correction filter FLCT connected between the output of a SM summer and DLT dictionary entry.

The two inputs of the summator SM respectively receive the product of the extracted word of long-term excitation v; by the associated long-term gain Ga, and the product of the extracted short-term excitation word c _j by the associated gain Gc.

This FLCT total correction filter is a low pass filter generally having a cutoff frequency greater than quarter of the sampling frequency and less than half of it.

This filter is in the example described a digital filter with linear phase finite impulse response with order at less than 10.

More specifically, when the sampling frequency is of 16 kHz, a cutoff frequency of preferably will be used of the order of 6 kHz and a filter of order 20, which achieves a good trade-off between memory complexity and signal quality reconstituted vocal.

Harmonic noise is introduced by the contribution of long-term excitement and by repeating samples for values of the fundamental period (pitch) less than the length of a speech frame, here of 5 ms. This noise is also present for values of the fundamental period greater than the size of a frame. It is also linked to adaptive gain, extracts a single times per speech frame.

Using long-pass low-pass filtering term is a solution to reduce harmonic noise.

In addition, high frequency noise is introduced by old high frequency short dictionary contributions term, present in the adaptive dictionary.

To eliminate this high frequency noise, we can eliminate high frequency residual components of the adaptive dictionary, using a correction filter, and this before updating the dictionary.

The total correction filter according to the invention therefore achieves the double harmonic correction and high correction function frequency. This allows an improvement in quality during voiced speech frames.

In addition, the location of this filter, i.e. at the input of the adaptive dictionary, allows filtering to be taken into account minimization of the error made to choose the excitation adaptive of the following speech frame.

In the embodiment illustrated in FIG. 3, the coder CD further comprises second updating means MAJ2 able to update the state of the linear prediction filter FP and the state of the filter perceptual weighting FPP with the short-term excitation word C _j filtered by a filter which is represented here schematically by a gain Gc '. This filter can be of order 0 and its gain Gc 'is less than the gain Gc. As a variant, this filter can be of finite impulse response and of order greater than or equal to 1, with for example a filter of finite impulse response of order 1.

The coefficients of this first order filter depend on the value long-term gain Ga, so as to weaken the contribution of short-term excitement when gaining long-term excitement Ga is greater than a predetermined threshold, for example equal to 1.

The transfer function of this filter is equal to B0 + B1 z ^-1 . By way of example, the first coefficient of the filter B0 can be determined by the formula (I) below. 1 / (1 + 0.98 min (Ga, 1)) while the second coefficient of filter B1 can be determined by formula (II) below. 0.98 min (Ga, 1) / (1 + 0.98 min (Ga, 1))

However, it is the short-term contribution not weakened (gain Gc) which is stored in the adaptive DLT dictionary for its update.

Thus, the attenuation occurs only on the signal preserving the magnitude of the short contribution term to store keeps the richness of the dictionary adaptive for the lowest frequencies.

Naturally, the correction of the Gc gain must also be applied for updating the memory status of the filters in the DCD decoder, as shown diagrammatically in FIG. 3a.

The variant embodiment illustrated in FIG. 3 allows, in addition to the benefits of the total correction filter, to eliminate hissing noise on speech frames voiced.

The FPP perceptual weighting filter uses the masking properties of the human ear compared to the spectral envelope of the speech signal, whose shape is a function resonances of the vocal tract. This filter allows you to assign more importance of the error appearing in the spectral valleys by compared to formic peaks.

In the variants illustrated in FIGS. 2 and 3, the same FPP perceptual weighting filter is used for short-term research and for long-term research. The transfer function W (z) of this FPP filter is given by the formula (III) below. W ( z ) = AT ( z / γ 1 ) AT ( z / γ 2 ) in which 1 / A (z) is the transfer function of the predictive filter FP and γ1 and γ2 are the perceptual weighting coefficients, the two coefficients being positive or zero and less than or equal to 1 with the coefficient γ2 less than or equal to the coefficient γ1.

In general, the perceptual weighting filter consists of a formantic weighting filter and a weighting of the slope of the spectral envelope of the signal (tilt).

In this case, assume that the weighting filter perceptual is only formed by the weighting filter formant whose transfer function is given by the formula (III) above.

The spectral nature of the long-term contribution is different from the short-term contribution. Therefore, it is advantageous to use two formantic weighting filters different, allowing independent control of distortions at short term and long term.

Such an embodiment is illustrated in FIG. 4, in which, compared to Figure 3, the unique FPP filter was replaced by a first formantic weighting filter FPP1 for long-term research, cascaded with a second filter of FPP2 formant weighting for short-term research.

Since the short-term weighting filter FPP2 is cascaded to the long-term weighting filter, the filters appearing in the long-term research loop should also appear in the short-term research loop.

The transfer function W ₁ (z) of the formantic weighting filter FPP1 is given by formula (IV) below. W 1 ( z ) = AT ( z / γ 11 ) AT ( z / γ 12 ) while the transfer function W ₂ (z) of the formantic weighting filter FPP2 is given by the formula (V) below. W 2 ( z ) = AT ( z / γ 21 ) AT ( z / γ 22 )

Furthermore, the coefficient γ ₁₂ is equal to the coefficient γ ₂₁ . This allows a clear simplification when cascading these two filters. Thus, the filter equivalent to the cascade of these two filters has a transfer function given by the formula (VI) below. AT ( z / γ 11 ) AT ( z / γ 12 )

Furthermore, if the value 1 is used for the coefficient γ ₁₁ , then the synthesis filter FP (having the transfer function 1 / A (z)) followed by the long-term weighting filter FPP1 and the weighting filter FPP2 is then equivalent to the filter whose transfer function is given by formula (VII) below. 1 AT ( z / γ 22 )

This further reduces the complexity of the excitation extraction algorithm.

As an indication, one can for example use for the coefficients γ ₁₁ , γ ₂₁ = γ ₁₂ and γ ₂₂ , the respective values 1; 0.1 and 0.9.

Of course, the variant providing for the use of two different form filters can be used independently of that providing for the weakening of the gain Gc.

Such an embodiment is illustrated in FIG. 5, where one see that the use of the two form filters is taken into account combination with the use of the total correction filter.

The invention advantageously applies to telephony mobile, and in particular to all remote terminals belonging to a wireless communication system.

Such a terminal, for example a TP mobile telephone, such as that illustrated in FIG. 6, conventionally comprises a antenna connected via a DUP duplexer to a chain reception CHR and a CHT transmission chain. A baseband processor BB is connected to the chain respectively of reception CHR and to the chain of transmission CHT by via analog digital ADCs and analog digital DACs.

Conventionally, the processor BB performs processing in baseband, including DCN channel decoding, followed by DCS source decoding.

For transmission, the processor performs source coding CCS followed by CCN channel coding.

When the mobile phone incorporates an encoder according to the invention, it is incorporated within the coding means of CCS source, while the decoder is incorporated within the means DCS source decoding.

Claims (18)

Wideband speech encoding method, in which speech is sampled so as to obtain successive speech frames each comprising a predetermined number of samples, and for each speech frame parameters of a linear prediction model are determined with code excitation, these parameters comprising a long term digital excitation word (v;) extracted from an adaptive coded directory (DLT) and an associated long term gain (Ga), as well as an excitation word short-term (cj) extracted from an algebraic coded repertoire (DCT) and an associated short-term gain (Gc), and the adaptive coded repertoire is updated from the extracted long-term excitation word and the word of short-term excitation extracted, characterized by the fact that we sum (SM) the product of the word extracted from long-term excitation by the associated long-term gain, with the product of the word extracted from short-term excitation by short-term gain assoc ié, we filter the summed digital word in a low-pass filter (FLCT) having a cut-off frequency greater than a quarter of the sampling frequency and less than half of it, and we update the adaptive coded directory with the filtered word. Method according to claim 1, characterized in that the summed word is filtered with a digital filter (FLCT) with finite impulse response in linear phase having an order at least equal to 10. Method according to Claim 2, characterized in that the sampling frequency is 16 kHz, and in that the filter (FLCT) is a filter of order 20 having a cut-off frequency of the order of 6 kHz . Method according to one of the preceding claims, characterized in that the extraction of the short-term excitation word comprises a digital linear prediction (FP) filtering, and in that the method comprises an update of the state of the linear prediction filter with the short term excitation word filtered by a filter (Gc ') whose coefficients or coefficients depend on the value of the long term gain, so as to weaken the contribution of the excitation to short term when the gain of the long term excitation is higher than a predetermined threshold. Method according to claim 4, characterized in that the predetermined threshold is equal to 1. Method according to claim 5, characterized in that the filter is of order 1 and has a transfer function equal to B0 + B1 z ^-1 , and in that the first coefficient of the filter B0 is equal to 1 / ( 1 + β.min (Ga, 1)), and the second coefficient of the filter B1 is equal to β.min (Ga, 1) / (1 + β.min (Ga, 1)), where β is a real number with an absolute value less than 1, Ga is the long-term gain and min (Ga, 1) designates the minimum value between Ga and 1. Method according to one of the preceding claims, characterized in that the extraction of the long-term excitation word is carried out using a first perceptual weighting filter (FPP1) comprising a first formant weighting filter, by the fact that the short term excitation word is extracted using the first perceptual weighting filter (FPP1) cascaded to a second perceptual weighting filter (FPP2) comprising a second formantic weighting filter, and by the fact that the denominator of the transfer function of the first formant weighting filter is equal to the numerator of the second formant weighting filter. Method according to claim 7 taken in combination with one of claims 4 to 6, characterized in that it includes an update of the state of the two perceptual weighting filters with the filtered short-term excitation word by said order filter 1. Wideband speech encoding device comprising sampling means able to sample the speech so as to obtain successive speech frames each comprising a predetermined number of samples, processing means suitable for each voice frame, in determining parameters of a linear prediction model with code excitation, these processing means comprising first extraction means (MEXT1) capable of extracting a digital excitation word with long term of an adaptive coded repertoire and calculating an associated long term gain, and second extraction means (MEXT2) able to extract a short term excitation word from an algebraic coded repertoire and calculating a gain associated short-term, and first updating means (MAJ) capable of updating the adaptive coded directory on the basis of the extracted long-term excitation word and the extracted short-term excitation word, characterized in that the first means of update feature first calculation means (SM) capable of summing the product of the word extracted from long-term excitation by the associated long-term gain, with the product of the word extracted from short-term excitation by the associated short-term gain, so as to deliver a summed digital word, and a low-pass filter (FLCT) having a cut-off frequency greater than a quarter of the sampling frequency and less than half of it, and connected between the output of the first calculation means and the adaptive coded repertoire so as to update this adaptive directory with the filtered word. Device according to Claim 9, characterized in that the low-pass filter (FLCT) is a digital finite impulse response filter with linear phase having an order at least equal to 10. Device according to Claim 10, characterized in that the sampling frequency is 16 kHz, and in that the filter is a filter of order 20 having a cut-off frequency of the order of 6 kHz. Device according to one of Claims 9 to 11, characterized in that the first extraction means comprise a digital linear prediction filter (FP), and in that the device comprises second updating means (MAJ2 ) capable of updating the state of the linear prediction filter with the short-term excitation word filtered (Gc ') by a filter whose coefficients depend on the value of the long-term gain, so as to weaken the contribution of short-term excitation when the gain of long-term excitation is above a predetermined threshold. Device according to claim 12, characterized in that the predetermined threshold is equal to 1. Device according to claim 13, characterized in that the filter is a filter of order 1 and has a transfer function equal to B0 + B1 z ^-1 , and in that the first coefficient B0 of the filter is equal to 1 /(1+β.min(Ga,1)), and the second coefficient B1 of the filter is equal to β.min (Ga, 1) / (1 + β.min (Ga, 1)), where β is a real number of absolute value less than 1, Ga is the long-term gain and min (Ga, 1) designates the minimum value between Ga and 1. Device according to one of Claims 9 to 14, characterized in that the first extraction means comprise a first perceptual weighting filter (FPP1) comprising a first formantic weighting filter, in that the second extraction means include the first perceptual weighting filter cascaded to a second perceptual weighting filter (FPP2) having a second formant weighting filter, and in that the denominator of the transfer function of the first formant weighting filter is equal to the numerator of the second formantic weighting filter. Device according to Claim 15 taken in combination with one of Claims 12 to 14, characterized in that the second updating means are capable of updating the state of the two perceptual weighting filters with the short term excitation word filtered by said filter of order 1. Terminal of a wireless communication system, characterized in that it incorporates a device according to one of claims 9 to 16. Terminal according to claim 17, characterized in that it forms a cellular mobile telephone.

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