[go: up one dir, main page]

EP0195441B1 - Verfahren zur Sprachcodierung mit niedriger Bitrate unter Verwendung eines Mehrimpulsanregungssignals - Google Patents

Verfahren zur Sprachcodierung mit niedriger Bitrate unter Verwendung eines Mehrimpulsanregungssignals Download PDF

Info

Publication number
EP0195441B1
EP0195441B1 EP86103770A EP86103770A EP0195441B1 EP 0195441 B1 EP0195441 B1 EP 0195441B1 EP 86103770 A EP86103770 A EP 86103770A EP 86103770 A EP86103770 A EP 86103770A EP 0195441 B1 EP0195441 B1 EP 0195441B1
Authority
EP
European Patent Office
Prior art keywords
signal
pulse
filter
excitation
pulses
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Expired - Lifetime
Application number
EP86103770A
Other languages
English (en)
French (fr)
Other versions
EP0195441A1 (de
Inventor
Jean-Paul Lefevre
Olivier Passien
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Alcatel CIT SA
Original Assignee
Alcatel CIT SA
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Alcatel CIT SA filed Critical Alcatel CIT SA
Publication of EP0195441A1 publication Critical patent/EP0195441A1/de
Application granted granted Critical
Publication of EP0195441B1 publication Critical patent/EP0195441B1/de
Anticipated expiration legal-status Critical
Expired - Lifetime legal-status Critical Current

Links

Images

Classifications

    • 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
    • G10L19/04—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 predictive techniques
    • G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • G10L19/10—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a multipulse excitation

Definitions

  • the invention relates to low bit rate digital codings which are used for speech in vocoders and which do not restore the original form of the speech signal but parameters allowing the excitation signal and the parameters to be defined over successive time windows. characteristics of a filter generating a synthetic speech signal resembling listening to the original speech signal. It relates more particularly to a form of generation of the excitation signal of the filter known under the name multi-pulse.
  • the filter models the vocal tract considered to be invariant over short periods of time on the order of 20 ms. It restores the spectrum of short-term frequencies of the speech signal, especially its maxima or formants which are more perceived by the ear than its minima. It can be performed in different analog or digital ways: channel synthesis, formant synthesis or linear prediction synthesis.
  • This mode of elaboration poses the problem of an effective distinction between voiced and unvoiced sounds. It results in an excitation signal having only a distant relationship with the vocal excitation signal and producing via the vocal tract modeling filter a synthetic speech signal which is not very faithful and which is sometimes difficult to understand.
  • This minimization is done according to the criterion of minimization of the quadratic error on the window considered with a so-called perceptual weighting of the error taking into account the property of the human ear to be less sensitive to distortions in the regions forming speech frequency spectrum where energy is relatively concentrated.
  • the successive approximation is stopped after a certain number of iterations determined according to the available computing capacities and the coding bit rate.
  • the present invention aims to combat the degradation of the signal to noise ratio of a synthesized speech signal due to the method by successive approximations used for the determination of the pulses of the excitation signal of the filter producing the synthesized speech signal, without significantly increasing the number of calculations to be undertaken.
  • FIG. 1 there is a transmitter equipment 1 connected by a digital link 2 at low speed to a receiver equipment 3.
  • the receiving equipment 3 comprises a demultiplexer 31 and two decoders 32, 33 placed at the input, which are adapted to the multiplexer 16 and to the coders 14, 15 of the transmitting equipment 1 and which extract from the signal received from the digital transmission link 2 the sets of prediction coefficients a (k) and the multi-pulse excitation signal v (k), and a filter 34 for modeling the vocal tract, the characteristics of which are adjusted from the sets of prediction coefficients a (k) and which generates from the excitation multi-pulse signal v (k) samples S (k) of a synthesized speech signal reproducing the original speech signal.
  • the analysis circuit 11 of the transmitting equipment 1 is a digital processing circuit which is not detailed because it is well known to those skilled in the art and is not within the scope of the invention.
  • a (k) For the way in which it proceeds to extract the sets of prediction coefficients a (k) from the samples of the speech signal to be coded, reference may be made to the book by Markel J., Gray A. entitled “Linear prediction of speech” published by Springer. Verlag, New York, 1976.
  • the predicted signal S (n) is defined from the elapsed values of the speech signal to be coded S (n) by means of the prediction coefficients a (k) by the relation:
  • the prediction is considered to be optimal when the quadratic error between the predicted values and the real values defined by: is minimal.
  • the voice path modeling filter 34 of the reception equipment has the transfer function H (z) which is expressed from the prediction coefficients a (k) by: Its synthesis is outside the scope of the present invention. It can be done from the prediction coefficients a (k) by applying the previous relation but is preferably carried out by the Itakura-Saito method in the form of a lattice defined from so-called transmitted reflection coefficients instead of the prediction coefficients a (k) to which they correspond by well-known equivalence relations.
  • the circuit 12 for generating the excitation multi-pulse signal generates, for each time window of analysis of the signal, to code a sequence of pulses in minimum number with positions and amplitudes chosen so as to obtain from the filter modeling the conduit. vocal a synthesized speech signal reproducing as faithfully as possible for a listener the original speech signal.
  • the criterion adopted to estimate the fidelity of reproduction of a speech signal by a synthetic signal is that of minimization of the quadratic error, over a time window of analysis, between the original speech signal and the speech signal synthesized with error weighting taking into account the perceptual properties of a listener that make it less sensitive to distortions occurring in the forming regions of the frequency spectrum of the speech signal where energy is relatively concentrated.
  • One known way of achieving this weighting in particular by American patent No. 4,133,976, consists in subjecting the error signal formed by the difference between the original speech signal and the synthesized speech signal to filtering, the function of which is transfer W (z) is expressed as a function of that H (z) of the vocal tract modeling filter by the relation:
  • This filtering can be obtained by passing the error signal or its components through a predictive filter whose transfer function is H - '(z) then through a so-called perceptual filter with transfer function H (yz) which can be determined as a function of the prediction coefficients by the definition relation:
  • predictive filtering is done on the components of the error signal, explicitly on the speech signal to be coded and implicitly on the synthesized speech signal, while perceptual filtering is done on the error signal itself, the components of which have been combined after predictive filtering.
  • the processing circuit 12 comprises a delay circuit 120 which receives the packets of N successive samples S (k) of the speech signal to be coded corresponding to the successive time windows on which the analysis circuit 11 and which store them the time necessary for the latter to establish each set of prediction coefficients a (k), and a predictive filter 121 which receives its set of coefficients a (k) from the analysis circuit 11 and the successive sample packets S (k) of the delay circuit 120 and which delivers a prediction residue signal r (k).
  • the predictive filtering of the synthesized speech signal is obtained implicitly by replacing this signal by the multi-pulse excitation signal v (k) from which it follows by a filtering in H (z) carried out by the modeling filter of the vocal tract.
  • a subtractor 122 forms the error signal by subtracting the multi-pulse signal v (k) from the prediction residue signal r (k) and applies it to a perceptual filter 123 receiving its coefficients from a processing circuit 124 the developing from the set of prediction coefficients a (k) by implementing the last relation mentioned.
  • the pulse sequences forming the excitation multi-pulse signal for each of the time windows on which the analysis circuit 11 operates are generated in the processing circuit 12 by a pulse synthesizer circuit 125 which receives the signal d weighted error from the perceptual filter 123.
  • This pulse synthesizer circuit 125 generates for each sequence of the excitation multi-pulse signal a number of pulses compatible with the transmission capacity of the digital link 2 which connects the equipment d transmission 1 to the reception equipment 3 while giving them positions in the time window considered and amplitudes minimizing the energy of the weighted error.
  • a (i) be the amplitudes of these pulses assumed at most in number Q and m (i) their respective positions in the time window chosen from the discrete positions 1, ..., N of samples staggered along the window.
  • the pulse sequence V (k) is expressed by: where d (k, m (i)) is a function taking the value one for k equal m (i) and zero everywhere else.
  • h '(k) the samples of the impulse response of the perceptual filter 123 having the transfer function H (yz)
  • the weighted error e (k) is expressed by: where B (j) and b (j) define the pulses relating to the preceding windows.
  • the weighted error in step (1 + 1) is expressed according to relation (1) by: or: which makes it possible to define the energy E (I + 1) of the weighted error in step (1 + 1) relative to the energy of the weighted error E (I) in step (I) by: or by noting with function and by C (i, j) the samples of the autocorrelation function of the impulse response of the perceptual filter 123
  • This expression reaches its minimum when its derivative with respect to the amplitude A (I + 1) of the (1 + 1) th pulse is canceled, that is to say for the value: and then takes as value:
  • the amplitude A (i) of each of them is corrected using the corrective term A '(i) deduced from relation (4): corrective term which can still be expressed taking into account relations (2) and (6) in the form: and which is defined as a ratio of two terms with the numerator the partial derivative, compared to the amplitude A (i), of the weighted quadratic error between the speech signal to be coded and the synthesized speech signal and the denominator the zero value of the autocorrelation function of the impulse response of the perceptual filter delayed by a delay corresponding to the position of the pulse considered relative to the start of the window.
  • T (j) can be expressed by:
  • This system of equations (10) can be rewritten or again, in terms of correction A "(i)
  • a comparison of this system of equations with relations (2) and (9) shows that the definition of the corrective term A '(i) is deduced from that of the corrective term A "(i) given by the optimal solution by admitting that the values C (i, j) of the correlation between two impulse responses from the perceptual filter are zero when they are not simultaneous.
  • the corrective term A '(i) has the advantage of having a definition relationship of the same kind as that (4) of the amplitude A (I + 1) of the pulse placed during each step of the method by approximations and consequently of being able to be worked out with a very limited number of additional operations, without common measure with the number of operations necessary for the resolution of the system of equations (12).
  • the step of developing the set of Q corrective terms A '(i) takes place after the Q th step of the approximation method during which the Q th pulse was determined by means of the study of the function It looks like, as we will see below, an additional step of the approximation method in which the calculation of the function is not carried out but replaced by the systematic calculation of the pulse amplitude for all the pulse positions already determined.
  • FIG. 2 illustrates an embodiment of the analysis circuits 11 and of preparation 12 of the transmitting equipment.
  • This consists of a microprocessor 40 connected by address 41, data 42 and control 43 buses to a random access memory 44 making it possible to temporarily store the samples of the speech signal to be coded S (k) as well that calculation variables, to a read only memory 45 containing programs for packetizing the samples S (k) of the speech signal to be coded, for calculating the set of prediction coefficients a (k) corresponding to each packet and samples h '(k) of the impulse response of the perceptual filter as well as of determination of the positions and amplitudes of the pulses of the sequence of the excitation multi-pulse signal, and at an input-output interface 46 allowing the introduction of the digital samples S (k) of speech to be coded and the delivery to the coders of the sets of prediction coefficients a (k) and of the positions and amplitudes of the pulses of the sequences of the excitation multi-pulse signal.
  • the microprocessor 40 carries out several simultaneous operations under the control of the programs recorded in the read-only memory 45.
  • the microprocessor then stores in memory the values of this function then calculates the function by the formula: determines the value of k for which this function is maximum and takes it as the value of the index m (t + 1) locating the position of the (1 + 1) th pulse of which it determines the amplitude A (I + 1) by calculating the relation:
  • the function is calculated from its definition using samples r (k) of the prediction residue signal taking into account the fact that the sequence of the multi-pulse signal on the current window is then a zero signal:
  • stage of drawing up corrective terms which does not require operations which are very different from those carried out during a stage of the method by successive approximations is easily integrated into the framework of the latter without appreciably increasing the duration of implementation which is fundamental in the context of vocoders where the development of each sequence of the excitation multi-pulse signal must be done over the limited duration of a time window of analysis.

Landscapes

  • Engineering & Computer Science (AREA)
  • Computational Linguistics (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)

Claims (1)

  1. Verfahren zur Sprachverschlüsselung mit niedriger Bitrate und Mehrimpulsanregungssignal, wobei das zu verschlüsselnde Sprachsignal durch Parameter ersetzt wird, welche über aufeinanderfolgende Zeitfenster hinweg die Kennwerte eines Filters definieren, das den Lautverlauf sowie Impulspositionen und -amplituden modelliert, die ein Mehrimpulssignal für die Anregung des Filters bilden und durch aufeinanderfolgende Annäherungen nach dem Kriterium der Minimisierung des quadratischen Fehlers zwischen dem zu verschlüsselnden Sprachsignal und dem durch das Filter wiederhergestellten synthetischen Sprachsignal bestimmt werden, wobei das Verfahren darin besteht, daß nach der Bestimmung der Positionen und Amplituden der Impulse des Mehrimpulsanregungssignals durch aufeinanderfolgende Annäherung der Amplitude jedes Impulses ein Korrekturterm zugefügt wird, der vom Wert der partiellen Ableitung des quadratischen Fehlers bezüglich der als unabhängige Variable betrachteten Impulsamplitude abhängt, wobei weiter der quadratische Fehler durch eine Filterung in einem Wahrnehmungsfilter gewichtet wird, dessen Impulsantwort relativ zu derjenigen des Modellierungsfilters des Lautverlaufs definiert ist und der Korrekturterm proportional zur partiellen Ableitung des gewichteten quadratischen Fehlers bezüglich der als unabhängige Variable betrachteten Impulsamplitude ist, geteilt durch den Nullwert der Autokorrelationsfunktion der Impulsantwort des Wahrnehmungsfilters und verzögert um eine der Position des betrachteten Impulses in Bezug auf den Anfang des Fensters entsprechenden Verzugszeit.
EP86103770A 1985-03-22 1986-03-20 Verfahren zur Sprachcodierung mit niedriger Bitrate unter Verwendung eines Mehrimpulsanregungssignals Expired - Lifetime EP0195441B1 (de)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
FR8504304A FR2579356B1 (fr) 1985-03-22 1985-03-22 Procede de codage a faible debit de la parole a signal multi-impulsionnel d'excitation
FR8504304 1985-03-22

Publications (2)

Publication Number Publication Date
EP0195441A1 EP0195441A1 (de) 1986-09-24
EP0195441B1 true EP0195441B1 (de) 1990-04-25

Family

ID=9317484

Family Applications (1)

Application Number Title Priority Date Filing Date
EP86103770A Expired - Lifetime EP0195441B1 (de) 1985-03-22 1986-03-20 Verfahren zur Sprachcodierung mit niedriger Bitrate unter Verwendung eines Mehrimpulsanregungssignals

Country Status (6)

Country Link
US (1) US4847905A (de)
EP (1) EP0195441B1 (de)
CA (1) CA1241117A (de)
DE (1) DE3670712D1 (de)
DK (1) DK126986A (de)
FR (1) FR2579356B1 (de)

Families Citing this family (15)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
USRE35057E (en) * 1987-08-28 1995-10-10 British Telecommunications Public Limited Company Speech coding using sparse vector codebook and cyclic shift techniques
CA1337217C (en) * 1987-08-28 1995-10-03 Daniel Kenneth Freeman Speech coding
AU629637B2 (en) * 1989-05-11 1992-10-08 Telefonaktiebolaget Lm Ericsson (Publ) Excitation pulse positioning method in a linear predictive speech coder
JP2940005B2 (ja) * 1989-07-20 1999-08-25 日本電気株式会社 音声符号化装置
US5673364A (en) * 1993-12-01 1997-09-30 The Dsp Group Ltd. System and method for compression and decompression of audio signals
AU696092B2 (en) * 1995-01-12 1998-09-03 Digital Voice Systems, Inc. Estimation of excitation parameters
US6012025A (en) * 1998-01-28 2000-01-04 Nokia Mobile Phones Limited Audio coding method and apparatus using backward adaptive prediction
US5963897A (en) * 1998-02-27 1999-10-05 Lernout & Hauspie Speech Products N.V. Apparatus and method for hybrid excited linear prediction speech encoding
JP4460165B2 (ja) * 1998-09-11 2010-05-12 モトローラ・インコーポレイテッド 情報信号を符号化する方法および装置
EP2009623A1 (de) * 2007-06-27 2008-12-31 Nokia Siemens Networks Oy Sprachkodierung
US8036886B2 (en) * 2006-12-22 2011-10-11 Digital Voice Systems, Inc. Estimation of pulsed speech model parameters
US11270714B2 (en) 2020-01-08 2022-03-08 Digital Voice Systems, Inc. Speech coding using time-varying interpolation
GB2596821A (en) 2020-07-07 2022-01-12 Validsoft Ltd Computer-generated speech detection
US12254895B2 (en) 2021-07-02 2025-03-18 Digital Voice Systems, Inc. Detecting and compensating for the presence of a speaker mask in a speech signal
US11990144B2 (en) 2021-07-28 2024-05-21 Digital Voice Systems, Inc. Reducing perceived effects of non-voice data in digital speech

Family Cites Families (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US4133976A (en) * 1978-04-07 1979-01-09 Bell Telephone Laboratories, Incorporated Predictive speech signal coding with reduced noise effects
US4472832A (en) * 1981-12-01 1984-09-18 At&T Bell Laboratories Digital speech coder
US4720861A (en) * 1985-12-24 1988-01-19 Itt Defense Communications A Division Of Itt Corporation Digital speech coding circuit

Also Published As

Publication number Publication date
DK126986D0 (da) 1986-03-19
EP0195441A1 (de) 1986-09-24
CA1241117A (fr) 1988-08-23
FR2579356A1 (fr) 1986-09-26
DK126986A (da) 1986-09-23
FR2579356B1 (fr) 1987-05-07
DE3670712D1 (de) 1990-05-31
US4847905A (en) 1989-07-11

Similar Documents

Publication Publication Date Title
EP0195441B1 (de) Verfahren zur Sprachcodierung mit niedriger Bitrate unter Verwendung eines Mehrimpulsanregungssignals
EP1316087B1 (de) Übertragungsfehler-verdeckung in einem audiosignal
US6691082B1 (en) Method and system for sub-band hybrid coding
FR2517452A1 (fr) Circuit de traitement numerique de la parole
EP1944755B1 (de) Veränderung eines Wortsignals
EP0865028A1 (de) Sprachdekodierung mittels Wellenforminterpolation unter Verwendung von Spline-Funktionen
EP0782128A1 (de) Verfahren zur Analyse eines Audiofrequenzsignals durch lineare Prädiktion, und Anwendung auf ein Verfahren zur Kodierung und Dekodierung eines Audiofrequenzsignals
EP0801790A1 (de) Verfahren zur sprachkodierung mittels analyse durch synthese
EP0428445B1 (de) Verfahren und Einrichtung zur Codierung von Prädiktionsfiltern in Vocodern mit sehr niedriger Datenrate
JP4180677B2 (ja) 音声符号化並びに復号化方法及びその装置
EP0865029A1 (de) Wellenforminterpolation mittels Zerlegung in Rauschen und periodische Signalanteile
EP2080194B1 (de) Dämpfung von stimmüberlagerung, im besonderen zur erregungserzeugung bei einem decoder in abwesenheit von informationen
EP0685833B1 (de) Verfahren zur Sprachkodierung mittels linearer Prädiktion
Srivastava Fundamentals of linear prediction
JP2829978B2 (ja) 音声符号化復号化方法及び音声符号化装置並びに音声復号化装置
EP2589045B1 (de) Adaptive lineare prädiktive codierung/decodierung
EP1192619B1 (de) Audio-kodierung, dekodierung zur interpolation
JP3055901B2 (ja) 音声信号符号化復号化方法及び音声信号符号化装置
JP2715437B2 (ja) マルチパルス符号化装置
EP1192618B1 (de) Audiokodierung mit adaptiver lifterung
EP1192621B1 (de) Audiokodierung mit harmonischen komponenten
Yuan The weighted sum of the line spectrum pair for noisy speech
EP1194923B1 (de) Verfahren und system für audio analyse und synthese
WO2002029786A1 (fr) Procede et dispositif de codage segmental d'un signal audio
EP0454552A2 (de) Verfahren und Einrichtung zur Sprachcodierung mit niedriger Bitrate

Legal Events

Date Code Title Description
PUAI Public reference made under article 153(3) epc to a published international application that has entered the european phase

Free format text: ORIGINAL CODE: 0009012

AK Designated contracting states

Kind code of ref document: A1

Designated state(s): BE DE FR GB IT NL SE

17P Request for examination filed

Effective date: 19870324

RAP1 Party data changed (applicant data changed or rights of an application transferred)

Owner name: ALCATEL CIT

17Q First examination report despatched

Effective date: 19881124

GRAA (expected) grant

Free format text: ORIGINAL CODE: 0009210

RAP1 Party data changed (applicant data changed or rights of an application transferred)

Owner name: ALCATEL CIT

AK Designated contracting states

Kind code of ref document: B1

Designated state(s): BE DE FR GB IT NL SE

REF Corresponds to:

Ref document number: 3670712

Country of ref document: DE

Date of ref document: 19900531

GBT Gb: translation of ep patent filed (gb section 77(6)(a)/1977)
ITF It: translation for a ep patent filed
PLBE No opposition filed within time limit

Free format text: ORIGINAL CODE: 0009261

STAA Information on the status of an ep patent application or granted ep patent

Free format text: STATUS: NO OPPOSITION FILED WITHIN TIME LIMIT

26N No opposition filed
ITTA It: last paid annual fee
PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: FR

Payment date: 19931230

Year of fee payment: 9

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: SE

Payment date: 19940127

Year of fee payment: 9

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: BE

Payment date: 19940210

Year of fee payment: 9

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: GB

Payment date: 19940221

Year of fee payment: 9

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: NL

Payment date: 19940331

Year of fee payment: 9

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: DE

Payment date: 19940416

Year of fee payment: 9

EAL Se: european patent in force in sweden

Ref document number: 86103770.3

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: GB

Effective date: 19950320

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: SE

Effective date: 19950321

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: BE

Effective date: 19950331

BERE Be: lapsed

Owner name: ALCATEL CIT

Effective date: 19950331

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: NL

Effective date: 19951001

GBPC Gb: european patent ceased through non-payment of renewal fee

Effective date: 19950320

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: FR

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 19951130

NLV4 Nl: lapsed or anulled due to non-payment of the annual fee

Effective date: 19951001

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: DE

Effective date: 19951201

EUG Se: european patent has lapsed

Ref document number: 86103770.3

REG Reference to a national code

Ref country code: FR

Ref legal event code: ST

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: IT

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20050320