EP1540987B1 - Verfahren zum Einstellen einer Beschallungsanlage - Google Patents

Verfahren zum Einstellen einer Beschallungsanlage Download PDF

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Publication number
EP1540987B1
EP1540987B1 EP03757113A EP03757113A EP1540987B1 EP 1540987 B1 EP1540987 B1 EP 1540987B1 EP 03757113 A EP03757113 A EP 03757113A EP 03757113 A EP03757113 A EP 03757113A EP 1540987 B1 EP1540987 B1 EP 1540987B1
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EP
European Patent Office
Prior art keywords
sfmoy
excursion
response
impulse response
acoustic
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EP03757113A
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English (en)
French (fr)
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EP1540987A1 (de
Inventor
Jacques Lewiner
Sylvain Javelot
Damien Lebrun
Stéphane DEBUSNE
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Cynove SARL
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Cynove SARL
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R27/00—Public address systems
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/04—Circuits for transducers, loudspeakers or microphones for correcting frequency response

Definitions

  • the present invention relates to sound reinforcement methods, including correction of acoustic speaker response.
  • amplitude can be corrected at an amplifier that powers one or more speakers, using a gain pattern of the amplifier as a function of frequency.
  • the amplification in said band is emphasized so that the emitted sound is substantially constant throughout the audible band.
  • US-A-4,458,362 it was proposed in the document US-A-4,458,362 , to develop the gain template in question from test signals emitted by the speaker.
  • the technique used in this document raises many problems of implementation in real situations and in particular in a reverberant environment. Above all, this technique does not retain the phase of electrical signals to transform into acoustic signals.
  • a second approach, widely used to correct the response of a speaker consists in.regrouping in a speaker several speakers each having good characteristics in a given spectral band and to interpose between the input of the speaker and the speakers, filters that will selectively send to each speaker the spectral components of the electrical signal best suited to the speaker.
  • This method which makes it possible to improve the overall amplitude response of the loudspeaker, has the serious drawback of introducing phase shifts at several levels in the system and thus of not allowing a faithful reproduction with regard to the phase of the signals. to reproduce.
  • Another known technique uses, from the initial impulse response of the acoustic enclosure, a series of operations based on the Fourier transform to firstly obtain the response of the speaker in the frequency domain, in amplitude and in phase and in a second step, the template of a correction filter, which, used to power the loudspeaker, is supposed to correct the phase defects while respecting in theory the amplitude of the signals.
  • the practical implementation of such a solution from signal processing processors has serious drawbacks.
  • the impulse response of loudspeakers in the frequency domain has considerable differences in the amplitude of the signals as a function of frequency: it is frequent for the amplitude response of an enclosure to exhibit peaks up and down which can reach 50 dB and whose frequency width is often low. Therefore, with the technique proposed in the document US-A-4,888,808 , the construction of the template of an effective correction filter to obtain a satisfactory correction involves considerable computing power, resulting in the use of expensive processors. Moreover even these expensive processors obviously do not have an infinite dynamic, which leads to insufficient improvements.
  • the object of the present invention is in particular to propose a method for correcting the response of an acoustic speaker which makes it possible to preserve the phase of the signals to be reproduced in a broad frequency band, while requiring a reduced computing power compatible with the dimensions. and the costs of sound reproduction devices for the general public.
  • the method according to the invention requires only a relatively low computing capacity, compatible with the moderate costs required for devices intended for the general public.
  • the inventors have found that the clipping of the signal S (f) does not affect the quality of listening, thanks to an effect called “mask effect", which makes that the human ear discerns with diminished sensitivity frequency sounds close to a given frequency where a signal is well audible.
  • the listening quality obtained with the present invention is excellent, for a moderate cost.
  • the method according to the invention makes it possible to sound a space 100 while ensuring optimal listening to a listener 102 in a target area 101 of the space 100.
  • the space to be sounded 100 may be for example a listening room equipped with at least one loudspeaker 2, comprising a number n of loudspeakers 22, 24, n being a natural integer at least equal to 1, for example equal to 2 or higher.
  • the loudspeakers 22, 24 of the enclosure 2 may for example be supplied by a common input 25 through passive filters, respectively 21, 23.
  • the input 25 receives an electrical signal P (t) from a computer 5 and amplified by an amplifier 6 (the amplifier 6 and the computer 5 can of course be included in the same housing).
  • the aforementioned filter 54 may simply be a software module loaded into the computer 5 and that the digital to analog converter could be removed using digital speakers.
  • the electrical signal X (t) is processed by the correction filter 54 of the computer 5 during the phases of sound, that is to say during normal operation of the sound system.
  • an acoustic calibration operation of the space 100 is carried out by determining the impulse response S (t) between the acoustic enclosure 2 and a calibration point 103 of the target zone 101.
  • the calibration point 103 may for example be between 50 cm and 1 m 50 above the ground.
  • the impulse response S (t) corresponds to the acoustic signal received at point 103 when the loudspeaker emits a short acoustic pulse.
  • This impulse response may preferably be measured at a time when the space 100 is not polluted by other acoustic signals than those emitted by the chamber 2, for example by causing the speaker 2 to emit a short acoustic pulse. and by measuring the acoustic signals received as a result of this pulse at the calibration point 103, by means of a microphone 11 previously arranged at the point 103.
  • the acoustic chamber 2 receives from the computer 5 the pulse signal to be transmitted.
  • the microphone 11 located at the calibration point 103 is connected to an amplifier 12 itself connected to an analog-digital converter 3, this converter can for example be connected to the computer 5, so that the signals picked up by the microphone 11 can be stored by the computer 5 for the calibration point 103.
  • the microphone 11 is disassembled with its amplifier 12 and its converter 3.
  • the computer 5 determines by a fast Fourier transform technique the frequency response S (f) of the impulse response S (t ).
  • the calculator (5) then calculates the inverse Fourier transform of I (f), namely I (t).
  • the filter template W (t) is then obtained by the computer 5 by making the convolution product S (-t) with I (t), which makes it possible to set up the filter software module 54 in the computer 5 and closes the learning step.

Landscapes

  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Soundproofing, Sound Blocking, And Sound Damping (AREA)
  • Circuit For Audible Band Transducer (AREA)

Claims (9)

  1. Beschallungsverfahren eines Raums (100) zur Übertragung von Informationen in diesen Raum in Form akustischer Wellen, die für ein Signal (X)t stehen, mittels einer Akustikbox (2), die einen Eingang (25) zur Steuerung einer Anzahl n von Lautsprechern (22,24) umfasst, wobei n ein natürliches Ganzes zumindest gleich 1 ist und dieses Verfahren zumindest eine Beschallungsetappe einschließt, in deren Verlauf am Eingang der Akustikbox (2) ein elektrisches Signal P(t)=W(t)⊗X(t) angewandt wird, wobei:
    - ⊗ der mathematische Operator als Produkt der Konvolution ist und
    - W(t) eine im Vorfeld bestimmte und gespeicherte Filterschablone darstellt,
    und das Verfahren eine Lernetappe umfasst, in deren Verlauf die Filterschablone W(t) wie folgt bestimmt wird: W t = S - t ⊗ I t ,
    Figure imgb0008
     wobei:
    - S (-t) die zeitliche Umkehr der Impulsantwort S(t) zwischen der Box und einem Zielbereich (101) ist, der dem zu beschallenden Raum (100) angehört, t entspricht hierbei der Zeit,
    - und I(t) die zeitliche Antwort des Produkts e-2iπft0.Sc (f) ist, wobei F die Frequenz, t0 eine zeitliche Verschiebungskennziffer und Sc(f)=1/(S1(f))α darstellt, α hierbei eine positive Ziffer ungleich 0 ist und S1 (f) eine reale Funktion darstellt, die durch den Ausgleich des Moduls |S(f)| der Frequenzantwort S(f) der Impulsantwort S(t) ermittelt wird.
  2. Verfahren nach Anspruch 1, bei dem die Funktion Sc(f) im Verlauf des Lernverfahrens wie folgt bestimmt wird:
    - für Sfmoy.R2<|S(f)|< Sfmoy.R1 Sc(f)=1/S|(f)|α, wobei R1 und R2 zwei positive Zahlen sind, R1 größer als R2 und Sfmoy dem Durchschnittswert von |S(f)| entspricht,
    - für |S(f)| ≤ Sfmoy.R2, Sc(f)=1/(Sfmoy.R2)α.
    - für |S(f)| ≥ Sfmoy.R1, Sc(f) = 1/(Sfmoy.R1)α,
  3. Verfahren nach einem der vorstehenden Ansprüche, bei dem die zeitliche Verschiebungskennziffer t0 zwischen 0 und Tmax liegt und Tmax der Aufzeichnungsdauer der Antwort S(t) gleicht.
  4. Verfahren nach einem der vorstehenden Ansprüche, bei dem I(t) durch die Verwendung des realen Teil der Fourier-Transformation des Produkts e-2iπft0.Sc(f) ermittelt wird.
  5. Verfahren nach einem der vorstehenden Ansprüche, bei dem die Impulsantwort S (t) auf einer Anzahl 2K von Abtastwerten gespeichert und S(f) ab S(t) berechnet wird, indem eine schnelle Fourier-Transformationstechnik von S(t) angewandt wird.
  6. Verfahren nach einem der vorstehenden Ansprüche, bei dem die Impulsantwort S(t) auf einer Anzahl 2K von Abtastwerten gespeichert und I(t) auf der Grundlage des Produkts e-2insft0.Sc(f) ermittelt wird, indem eine schnelle Fourier-Transformationstechnik von S(t) angewandt wird.
  7. Verfahren nach einem der vorstehenden Ansprüche, bei dem α gleich 1 ist.
  8. Verfahren nach einem der vorstehenden Ansprüche, bei dem die Kennziffern R1 und R2 derart ausgewählt werden, dass ein Schwingungsweitenhub erreicht wird, der unter einem Schwingungsweitenhub von etwa 12dB, einem Schwingungsweitenhub von 24 dB, einem Schwingungsweitenhub von 36 dB und einem Schwingungsweitenhub von 48 dB ausgewählt wird.
  9. Verfahren nach einem der vorstehenden Ansprüche, bei dem der Wert Sfmoy auf einem Frequenzband fb ermittelt wird, das einzig einen Teil der hörbaren Frequenzen darstellt.
EP03757113A 2002-06-10 2003-06-06 Verfahren zum Einstellen einer Beschallungsanlage Expired - Lifetime EP1540987B1 (de)

Applications Claiming Priority (3)

Application Number Priority Date Filing Date Title
FR0207110A FR2840759B1 (fr) 2002-06-10 2002-06-10 Procede de sonorisation
FR0207110 2002-06-10
PCT/FR2003/001694 WO2003105525A1 (fr) 2002-06-10 2003-06-06 Procede de sonorisation

Publications (2)

Publication Number Publication Date
EP1540987A1 EP1540987A1 (de) 2005-06-15
EP1540987B1 true EP1540987B1 (de) 2012-05-30

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EP03757113A Expired - Lifetime EP1540987B1 (de) 2002-06-10 2003-06-06 Verfahren zum Einstellen einer Beschallungsanlage

Country Status (6)

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US (1) US7171005B2 (de)
EP (1) EP1540987B1 (de)
AU (1) AU2003258785A1 (de)
DK (1) DK1540987T3 (de)
FR (1) FR2840759B1 (de)
WO (1) WO2003105525A1 (de)

Families Citing this family (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US8249260B2 (en) * 2007-04-13 2012-08-21 Qualcomm Incorporated Method and apparatus for audio path filter tuning
EP3508836B1 (de) * 2018-01-05 2020-07-29 Infineon Technologies AG Photoakustisches system und verfahren zur abschätzung einer gaskonzentration
GB2600538B (en) * 2020-09-09 2023-04-05 Tymphany Worldwide Enterprises Ltd Method of providing audio in a vehicle, and an audio apparatus for a vehicle

Family Cites Families (12)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US4458362A (en) 1982-05-13 1984-07-03 Teledyne Industries, Inc. Automatic time domain equalization of audio signals
US4683590A (en) * 1985-03-18 1987-07-28 Nippon Telegraph And Telphone Corporation Inverse control system
US4888808A (en) 1987-03-23 1989-12-19 Matsushita Electric Industrial Co., Ltd. Digital equalizer apparatus enabling separate phase and amplitude characteristic modification
JP2646210B2 (ja) * 1987-05-27 1997-08-27 ヤマハ株式会社 電気音響的残響支援装置
JP2569872B2 (ja) * 1990-03-02 1997-01-08 ヤマハ株式会社 音場制御装置
JPH042928A (ja) * 1990-04-19 1992-01-07 Matsushita Electric Ind Co Ltd 音場制御装置
GB9026906D0 (en) * 1990-12-11 1991-01-30 B & W Loudspeakers Compensating filters
US5384856A (en) * 1991-01-21 1995-01-24 Mitsubishi Denki Kabushiki Kaisha Acoustic system
US5572443A (en) * 1993-05-11 1996-11-05 Yamaha Corporation Acoustic characteristic correction device
DE4332804C2 (de) * 1993-09-27 1997-06-05 Klippel Wolfgang Adaptive Korrekturschaltung für elektroakustische Schallsender
JP3537674B2 (ja) * 1998-09-30 2004-06-14 パイオニア株式会社 オーディオシステム
JP4663085B2 (ja) * 2000-09-19 2011-03-30 パナソニック株式会社 音響再生装置

Also Published As

Publication number Publication date
AU2003258785A1 (en) 2003-12-22
EP1540987A1 (de) 2005-06-15
FR2840759B1 (fr) 2004-07-23
US7171005B2 (en) 2007-01-30
FR2840759A1 (fr) 2003-12-12
US20050226440A1 (en) 2005-10-13
DK1540987T3 (da) 2012-08-20
WO2003105525A1 (fr) 2003-12-18

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