EP0597040A1 - Verfahren zur Ermittlung von Ubergangssignalen insbesondere für akustische Signale - Google Patents

Verfahren zur Ermittlung von Ubergangssignalen insbesondere für akustische Signale

Info

Publication number
EP0597040A1
EP0597040A1 EP92918452A EP92918452A EP0597040A1 EP 0597040 A1 EP0597040 A1 EP 0597040A1 EP 92918452 A EP92918452 A EP 92918452A EP 92918452 A EP92918452 A EP 92918452A EP 0597040 A1 EP0597040 A1 EP 0597040A1
Authority
EP
European Patent Office
Prior art keywords
signal
filter
parameters
signals
regressive
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.)
Withdrawn
Application number
EP92918452A
Other languages
English (en)
French (fr)
Inventor
Alain Lemer
Jean-Marie Nicolas
Jean-Pierre Pignon
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.)
Thales SA
Original Assignee
Thomson CSF 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 Thomson CSF SA filed Critical Thomson CSF SA
Publication of EP0597040A1 publication Critical patent/EP0597040A1/de
Withdrawn legal-status Critical Current

Links

Classifications

    • GPHYSICS
    • G01MEASURING; TESTING
    • G01RMEASURING ELECTRIC VARIABLES; MEASURING MAGNETIC VARIABLES
    • G01R29/00Arrangements for measuring or indicating electric quantities not covered by groups G01R19/00 - G01R27/00
    • G01R29/02Measuring characteristics of individual pulses, e.g. deviation from pulse flatness, rise time or duration
    • G01R29/027Indicating that a pulse characteristic is either above or below a predetermined value or within or beyond a predetermined range of values
    • GPHYSICS
    • G01MEASURING; TESTING
    • G01RMEASURING ELECTRIC VARIABLES; MEASURING MAGNETIC VARIABLES
    • G01R23/00Arrangements for measuring frequencies; Arrangements for analysing frequency spectra
    • G01R23/16Spectrum analysis; Fourier analysis
    • G01R23/165Spectrum analysis; Fourier analysis using filters
    • G01R23/167Spectrum analysis; Fourier analysis using filters with digital filters
    • GPHYSICS
    • G01MEASURING; TESTING
    • G01SRADIO DIRECTION-FINDING; RADIO NAVIGATION; DETERMINING DISTANCE OR VELOCITY BY USE OF RADIO WAVES; LOCATING OR PRESENCE-DETECTING BY USE OF THE REFLECTION OR RERADIATION OF RADIO WAVES; ANALOGOUS ARRANGEMENTS USING OTHER WAVES
    • G01S7/00Details of systems according to groups G01S13/00, G01S15/00, G01S17/00
    • G01S7/52Details of systems according to groups G01S13/00, G01S15/00, G01S17/00 of systems according to group G01S15/00
    • G01S7/523Details of pulse systems
    • G01S7/526Receivers
    • G01S7/527Extracting wanted echo signals
    • G01S7/5273Extracting wanted echo signals using digital techniques

Definitions

  • the present invention relates to methods which make it possible to detect, and possibly identify, the transient acoustic signals characteristic of various phenomena which it is desired to study. It can be applied as well to aerial acoustic signals as to underwater acoustic signals, for applications as diverse as the identification of breakdowns in an automobile and the identification of ships.
  • the invention proposes a method for assisting in the detection of transient signals, in particular in acoustic signals, mainly characterized in that in a first step the parameters of a filter are determined making it possible to modify the received signal to obtain a signal formed by a first substantially continuous and low-level part and a second impulse part having a level significantly higher than this low level and representing the transient signals to be detected, and that in a second step the signal is filtered reception by a filter having the parameters thus determined.
  • the electrical signal representative of the audio signal received is applied to an anti-aliasing filter 101 adapted to the maximum frequency received, to the maximum frequency audible by the operator (approximately 20 KHz) and at the sampling frequency of a converter analog-digital (102) located after this filter 101.
  • This analog-digital converter delivers a digital signal Y (T) which is applied to several processing chains whose outputs lead to a switch 103.
  • One of the inputs of this switch is directly connected to converter 102 in order to allow the operator to listen directly to the received signal.
  • the output of the switch 103 is connected to the input of a digital-analog converter 104, followed by a low-pitch filter 105 and an amplifier 106 which supplies headphones 107.
  • a digital-analog converter 104 which supplies headphones 107.
  • the operator provided with the headset 107 hears the same signal as that received by the hydrophone, the passage by the digital channel not causing significant distortion of this signal.
  • the operator directs the output signal of the converter 102, using a switch 108, to a learning module 109
  • This training module makes it possible to model the signal listened to according to a stationary broadband component synthesized by a predictive model, and a prediction residue corresponding to the difference between the signal predicted by the predictive model and the true signal.
  • This modeling will preferably be carried out by the known method of autoregressive modeling of the signal with a P order.
  • the order P of such an autoregressive model corresponds to the P temporal previous samples of the output of the autoregressive filter
  • these P predictor coefficients can be obtained in a known manner by solving the Yule-Walker equations using the Levinson algorithm. This algorithm converges quickly and we can get the P coefficients in the space of a few seconds during which the operator closes the switch 108.
  • this selection of the learning sequences can be carried out automatically, either repeatedly (for example 1 second every 10 seconds), or by detection of an abnormal phenomenon identified in any way.
  • This gives a vector formed of P coefficients a. which can be used in different ways, for example by reverse filtering or by so-called "caricatural” filtering. In the case of the use of reverse filtering, the coefficients a.
  • the signal e (kT) obtained at the output of this adapted average filter MA is applied via the switch 103 to the converter 104, and the processed signal finally arrives on the headphones 107.
  • AR recursive filters of the self-regressive type which receive the coefficients a. and apply a processing defined by the formula to the input signal:
  • the signal at the output of converter 102 is applied to a first AR filter
  • the inventors have proposed to call signal c. (kT) thus obtained, order 1 caricature of the signal s (t) because it has a power spectral density substantially equal to the square of the power spectral density of this signal s (t), which amounts to saying that the permanent noise contained in the signal s (t) has been "flattened” by bringing out the impulse noise contained in this same signal.
  • This impulse noise is thus exaggerated by reducing the permanent noise, which corresponds pictorially to a caricature of the signal s (t). In this way the operator who listens to the signal resulting from this processing in the helmet 107 more easily perceives this impulse signal, which emerges better from the permanent signal.
  • the invention further proposes to further increase the influence of this processing by applying the signal at the output of the first AR filter in a second AR filter 111.2 then possibly in a succession of AR filters put in series up to an n th AR filter 111. n.
  • the signal cj thus obtained at the output of the AR filter 111. j will quite naturally be called caricature of order j of the input signal, and its power spectral density is substantially equal to the power n of the power spectral density of the signal s .
  • the treatment thus carried out is defined by the formula:
  • the outputs of the AR filters are respectively connected to inputs of the switch 103 and the operator can thus select, in addition to the direct signal and the signal filtered by reverse filtering using the MA filter, a caricature of order between 1 and n of the input signal.
  • the operator after having learned will successively select the different outputs and then come back to the one which seems to him to have particularly interesting characteristics. Indeed the passage in such a system, whether by reverse filtering or by j-order caricature, deeply distorts the signal, since in the best of cases there is a continuous signal of very low level barely audible from which come out more or less regular pulses which are clearly perceptible to the operator but which are distorted with respect to the initial signal received.
  • this signal is very difficult to analyze, even by a trained operator, to obtain precise information on the source of the disturbing signal, because the deformations which are made to this disturbing signal, if they make it more detectable, distort it too much compared to the original signal to allow effective identification.
  • these modifications are essentially variable according to the modalities brought, within the framework of the invention, to the processing chains, for example the order P of the auto-regression or the order j of the caricature.
  • the method according to the invention therefore essentially serves as a detection aid for an operator, who, after having identified the presence of a disturbing signal, may return for example to direct listening without filtering to concentrate his attention on the identification of the source of this disturbing signal. Possibly he can pass the baton to a more trained operator who will more easily carry out this identification.
  • the invention has been described in the case where the selection of the learning range is made by the operator, and the possibility of using automatic criteria based on scales has been mentioned. of time.
  • LEVINSON for example those of BURG, ITAKURA, GUEGUEN-LEROUX, MORF and FALCONER, or MORF and LEE. . .
  • the invention extends beyond self-regressive analysis to any technique for modeling the initial signal which allows a decomposition into a set of parameters characteristic of the chosen model and an "innovation" ( difference between the prediction provided by the model and the true signal).
  • ARMA model the KALMAN filter
  • neural networks . . .
  • a device applying the method according to the invention in order to obtain an automatic function making it possible to improve the performance of a device for detecting and automatically classifying acoustic signals.
  • the adapted medium filter to produce a pre-processing module making it possible to whiten and minimize the energy of a stationary signal, in order to favor the automatic detection of transient signals contained in an acoustic signal, by controlling for example the learning phases by a clock validated by a transient non-detection signal.

Landscapes

  • Physics & Mathematics (AREA)
  • General Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Radar, Positioning & Navigation (AREA)
  • Remote Sensing (AREA)
  • Mathematical Physics (AREA)
  • Measurement Of Mechanical Vibrations Or Ultrasonic Waves (AREA)
  • Filters That Use Time-Delay Elements (AREA)
  • Measurement Of Velocity Or Position Using Acoustic Or Ultrasonic Waves (AREA)
EP92918452A 1991-08-02 1992-07-31 Verfahren zur Ermittlung von Ubergangssignalen insbesondere für akustische Signale Withdrawn EP0597040A1 (de)

Applications Claiming Priority (3)

Application Number Priority Date Filing Date Title
FR9109874 1991-08-02
FR9109874A FR2680006B1 (fr) 1991-08-02 1991-08-02 Procede d'aide a la detection de signaux transitoires notamment dans les signaux acoustiques.
PCT/FR1992/000760 WO1993003395A1 (fr) 1991-08-02 1992-07-31 Procede d'aide a la detection de signaux transitoires, notamment dans les signaux acoustiques

Publications (1)

Publication Number Publication Date
EP0597040A1 true EP0597040A1 (de) 1994-05-18

Family

ID=9415882

Family Applications (1)

Application Number Title Priority Date Filing Date
EP92918452A Withdrawn EP0597040A1 (de) 1991-08-02 1992-07-31 Verfahren zur Ermittlung von Ubergangssignalen insbesondere für akustische Signale

Country Status (5)

Country Link
EP (1) EP0597040A1 (de)
AU (1) AU657792B2 (de)
CA (1) CA2114632A1 (de)
FR (1) FR2680006B1 (de)
WO (1) WO1993003395A1 (de)

Family Cites Families (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US188667A (en) * 1877-03-20 Improvement in carbureters
JPS52125251A (en) * 1976-02-23 1977-10-20 Bio Communication Res Electric filter and method of designing same
FR2593608B1 (fr) * 1986-01-28 1988-07-15 Thomson Csf Procede et dispositif de reconnaissance automatique de cibles a partir d'echos " doppler "
US4982150A (en) * 1989-10-30 1991-01-01 General Electric Company Spectral estimation utilizing an autocorrelation-based minimum free energy method

Non-Patent Citations (1)

* Cited by examiner, † Cited by third party
Title
See references of WO9303395A1 *

Also Published As

Publication number Publication date
FR2680006B1 (fr) 1993-10-29
WO1993003395A1 (fr) 1993-02-18
FR2680006A1 (fr) 1993-02-05
CA2114632A1 (fr) 1993-02-18
AU2470892A (en) 1993-03-02
AU657792B2 (en) 1995-03-23

Similar Documents

Publication Publication Date Title
EP2309499B1 (de) Verfahren zur optimierten Filterung nicht stationärer Geräusche, die von einem Audiogerät mit mehreren Mikrophonen eingefangen werden, insbesondere eine Freisprechtelefonanlage für Kraftfahrzeuge
EP2293594B1 (de) Verfahren zur Filterung von seitlichem nichtstationärem Rauschen für ein Multimikrofon-Audiogerät
EP1356461B1 (de) Rauschverminderungsverfahren und -einrichtung
EP2415047B1 (de) Klassifizieren von in einem Tonsignal enthaltenem Hintergrundrauschen
EP2772916B1 (de) Verfahren zur Geräuschdämpfung eines Audiosignals mit Hilfe eines Algorithmus mit variabler Spektralverstärkung mit dynamisch modulierbarer Härte
EP2538409B1 (de) Verfahren zur Geräuschdämpfung für Audio-Gerät mit mehreren Mikrofonen, insbesondere für eine telefonische Freisprechanlage
EP0752181B1 (de) Echokompensation mit adaptivem filter im frequenzbereich
FR2864860A1 (fr) Systeme d'annulation de bruit, systeme de reconnaissance vocale et systeme de navigation d'automobile
FR3012928A1 (fr) Modificateurs reposant sur un snr estime exterieurement pour des calculs internes de mmse
EP2898707A1 (de) Optimierte kalibrierung eines klangwiedergabesystems mit mehreren lautsprechern
EP0998166A1 (de) Anordnung zur Verarbeitung von Audiosignalen, Empfänger und Verfahren zum Filtern und Wiedergabe eines Nutzsignals in Gegenwart von Umgebungsgeräusche
FR3012929A1 (fr) Modificateur de la presence de probabilite de la parole perfectionnant les performances de suppression du bruit reposant sur le log-mmse
FR2568014A1 (fr) Dispositif de detection de parasites en forme d'impulsion et dispositif de suppression de parasites en forme d'impulsion muni d'un dispositif de detection de parasites en forme d'impulsion
EP1473709A1 (de) Verfahren zur Identifizierung von spezifischen Klängen
FR3012927A1 (fr) Estimation precise du rapport signal a bruit par progression reposant sur une probabilite de la presence de la parole mmse
FR2724029A1 (fr) Procede de detection de signaux acoustiques provenant de torpilles
EP0597040A1 (de) Verfahren zur Ermittlung von Ubergangssignalen insbesondere für akustische Signale
EP3627510A1 (de) Filterung eines tonsignals, das durch ein stimmerkennungssystem erfasst wurde
FR2799321A1 (fr) Procede de controle en continu de la qualite des sons numeriques en distribution
EP2515300B1 (de) Verfahren und System für die Geräuschunterdrückung
EP1359788B1 (de) Panoramische Audiovorrichtung für Passiv- Sonar
EP2452293A1 (de) Quelllokation
FR3106691A1 (fr) Conversion de la parole par apprentissage statistique avec modélisation complexe des modifications temporelles
FR2466901A1 (fr) Dispositif permettant l'ajustage automatique du niveau des sources sonores utiles dans les milieux a bruyance variable
EP1239291A1 (de) Anwesenheitsnachweisverfahren einer reinen Frequenz oder einer linear modulierten Frequenz

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

17P Request for examination filed

Effective date: 19931223

AK Designated contracting states

Kind code of ref document: A1

Designated state(s): DE GB

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

Free format text: STATUS: THE APPLICATION IS DEEMED TO BE WITHDRAWN

18D Application deemed to be withdrawn

Effective date: 19960201