René Boite

A new procedure for the design of high order minimum phase FIR digital or CCD filters

Abstract The approximation problem for high-order minimum phase FIR filter is solved without requiring any polynomial factorization. A modified Parks-McClellan program is used to compute the amplitude function; the minimum phase function is then derived by a method using the FFT algorithm. The procedure is illustrated by the design of various high order filters; short computation time with no numerical troubles is achieved.

Voiced speech coding at very low bit rates based on forward-backward waveform prediction (FBWP)

The authors introduce a novel algorithm, called forward-backward waveform prediction (FBWP), for voiced speech coding at very low bit rates, that aims to produce high-quality speech with low complexity. This algorithm encodes and transmits partial representative waveforms (RWs) with which the complete voiced speech waveforms are reconstructed in the time domain. The RW can be encoded at 20-30 ms intervals, considering the special initial conditions of short- and long-terms. The basic idea of the FBWP method is essentially consistent with that of the PWI algorithm, which is capable of reproducing high-quality voiced speech. The FBWP algorithm does not require two synchronous prototype waveforms decomposed into sinusoidal components; thus it is very fast while the high-quality speech can be obtained at a bit rate of about 3 kbit/s. Like the PWI method, the proposed algorithm is easily combined with other linear-prediction-based speech codes using a noise-like excitation to reproduce unvoiced speech.<<ETX>>

Voiced speech coding at very low bit rates based on forward-backward waveform prediction

Techniques for coding voiced speech at very low bit rates are investigated and a new algorithm, designed to produce high quality speech with low complexity, is proposed. This algorithm encodes and transmits partial representative waveforms (RWs) from which the complete speech waveforms are reconstructed by using a method called forward-backward waveform prediction (FBWP). The RW is encoded at 20-30 ms intervals with a low complexity approach, taking into account the special initial conditions of short- and long-term filters. The basic idea of FBWP is essentially consistent with that of the prototype waveform interpolation (PWI) algorithm, which was reported to be capable of producing high-quality voiced speech at a bit rate of between 3.0 and 4.0 kb/s. By implementing the FBWP in the time domain, fast computation is thereby made possible while high-quality speech can be obtained at bit rate of about 3 kb/s. As in the PWI method, the proposed algorithm may be combined with an LP-based speech coder which uses a noise-like excitation to reproduce unvoiced speech. >

Multiband code-excited linear prediction (MBCELP) for speech coding

Abstract This paper presents a new speech coding model targeted at the bit-rate above 4 kbit/s, referred to as multiband code-excited linear prediction (MBCELP). The analysis and synthesis of speech are accomplished in the time domain by comparing the original to the synthetic speech while a perceptual criterion is used. A usual short-term linear predictive filter is employed as the synthesis filter; the excitation signal is modelled as a linear combination of a long-term predictive excitation, periodic multiband excitations and a noise-like excitation; no voiced/unvoiced decision is required. The periodic multiband excitation is produced by convoluting a periodic impulse sequence with a sinc function corresponding to a frequency band; the noise-like excitation is represented by a codebook. We estimate a pitch which is appropriate not only to the long-term predictive filter but also to the periodic multiband excitations and to the ‘pitch’ prefilter in the decoder. Several CELP vocoders are developed as a reference to test the property of the MBCELP vocoder. Listening tests clearly indicate that this vocoder reconstructed very high quality speech without ‘buzziness’ or ‘hoarseness’ for both clean and noisy speech. A 4.8 kbit/s MBCELP vocoder is shown as an example. Its perceptual quality is virtually identical to the original 8 kbit/s CELP vocoder and the improved 7.2 kbit/s CELP vocoder. Since less subframes are used for the MBCELP vocoders, their complexity is not greater than that of usual CELP vocoders with the same type of codebook. Alot of techniques used to simplify CELP coding can be also adopted for the MBCELP coding.

A very simple and efficient weighting filter with application to a CELP coder for high quality speech at 4800 bits/s

Abstract In this paper we propose a very simple and efficient weighting filter with which the computational complexity of CELP coders can be considerably reduced. Other algorithms using a weighting filter could also benefit from the advantages of this simplified weighting filter. The estimation of the long term prediction with the close-loop method is described. A binary codebook is used for the excitation vectors. It is shown how the excitation sequence can be obtained by a non-exhaustive method in two steps with a simplified algorithm and the simple weighting filter. Several coders have been implemented showing that the perceptual quality of the simplified algorithm is equivalent to that of the original CELP.

Fast measurements of the attenuation and delay characteristics of data channels

A procedure for the fast measurement of the attenuation and delay of data channels in the frequency band of interest (100 to 3400 Hz) is described. The method is based upon the analysis of the steady-state response to a special periodic input sequence; that sequence is pseudo-random and such that the Discrete FOURIER Transform of one period has strict constant amplitude; moreover its length can be chosen equal to a power of 2, in order to facilitate the computation of the D.F.T. by a F.F.T. algorithm. The complete procedure is described; it has been simulated on a computer and the influence of various parameters is reported.

Contribution à une meilleure approche dans la détection et l’estimation de la fréquence laryngienne

RésuméL’utilisation combinée des informations sur la fréquence laryngienne, portées simultanément par le module de la transformée de Fourier court-terme et la fonction de distribution des fréquences instantanées, conduit à une meilleure précision dans l’estimation de cette fréquence. Deux paramètres sont définis : le coefficient de dissymétrie spectrale et la fonction d’intercorrélation. Ils introduisent une bonne discrimination dans la nature des tranches analysées assurant ainsi la détection exacte du voisement et du non-voisement. L’exploitation de la sensibilité de ces paramètres par rapport à la nature des tranches permet l’élaboration d’une décision voisé/mixte/non voisé et la spécification de la source mixte.AbstractCombined use of different laryngeal frequency information furnished simultaneously by short-time Fourier transform and by instantaneous frequency distribution function leads to a higher precision of the frequency estimation. Two parameters are defined : the spectral dissymmetry coefficient and the “inter-correlation function. These parameters give a clear insight into the nature of the analysed speech windows, leading to an exact detection of the voicing feature. The exploitation of the sensibility with respect to the nature of the speech windows enables the extraction of voiced/mixed/unvoiced decision and also the specification of the mixed source.

Sensibilité des filtres LC en échelle avec terminaisons résistives

AnalyseL’auteur établit l’expression de la sensibilité au premier ordre et aux ordres supérieurs pour l’affaiblissement effectif d’un filtre LC avec terminaisons résistives. Il justifie ensuite d’une façon algébrique l’existence de zéros de sensibilité au premier ordre communs à tous les composants réactifs du filtre; l’existence de zéros de sensibilité propres à certains composants est aussi expliquée.AbstractExpressions are derived for the first and higher order sensitivities of the effective loss introduced by a (LC) filter with resistive terminations. An algebraic proof is given for the first-order sensitivity zeros which are shared by each non dissipative component; the existence of « private » zeros for certain component is also explained.

Synthèse de filtres numériques simulant Des filtres a inductances et capacités

RésuméDes structures de filtres numériques récursifs moins sensibles que les structures classiques peuvent être obtenues par simulation de filtres analogues non dissipatifs. Les principes généraux de la transposition des filtres (LC) en échelle sont établis et la synthèse selon Fettweis des filtres d’onde est décrite. Des résultats numériques pour un filtre passebas de Cauer sont présentés et des conclusions générales terminent cette étude.AbstractRecursive digital filters structures much less sensitive than ordinary cascade or parallel structures are obtained when imitating analogue non dissipative filters. General principles for the transposition of (L, C) ladders filters are established and the synthesis of the socalled Fettweis wave filtersis described. Numerical results for a lowpass Cauer filter and final conclusions are also given.