EP3111667A1 - Verfahren und system zur automatischen akustischen entzerrung - Google Patents

Verfahren und system zur automatischen akustischen entzerrung

Info

Publication number
EP3111667A1
EP3111667A1 EP15703951.2A EP15703951A EP3111667A1 EP 3111667 A1 EP3111667 A1 EP 3111667A1 EP 15703951 A EP15703951 A EP 15703951A EP 3111667 A1 EP3111667 A1 EP 3111667A1
Authority
EP
European Patent Office
Prior art keywords
response
target
curve
averaged
difference
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.)
Granted
Application number
EP15703951.2A
Other languages
English (en)
French (fr)
Other versions
EP3111667B1 (de
Inventor
Frédéric AMADU
Delphine Devallez
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.)
Arkamys SA
Original Assignee
Arkamys SA
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Filing date
Publication date
Application filed by Arkamys SA filed Critical Arkamys SA
Publication of EP3111667A1 publication Critical patent/EP3111667A1/de
Application granted granted Critical
Publication of EP3111667B1 publication Critical patent/EP3111667B1/de
Active legal-status Critical Current
Anticipated expiration legal-status Critical

Links

Classifications

    • 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
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2499/00—Aspects covered by H04R or H04S not otherwise provided for in their subgroups
    • H04R2499/10—General applications
    • H04R2499/13—Acoustic transducers and sound field adaptation in vehicles

Definitions

  • the present invention relates to the field of sound signal processing.
  • the present invention relates more particularly to a method and an automated acoustic equalization system.
  • a case of use of the present invention is as follows: in the automotive field: a plurality of microphones are placed in a vehicle and pre-calibrated sound sequences are broadcast in the vehicle speakers. A system compares the sound signals emitted and the sound signals received and recorded. We deduce the "acoustic signature" of the passenger compartment of the vehicle. The user then defines a target acoustic signature curve, which is different from the vehicle's native acoustic signature. A second algorithm calculates digital filter coefficients so that, when these filters are applied before sound signals are broadcast in the vehicle loudspeakers, the acoustic signature of the vehicle becomes the target sound signature curve, and not the native acoustic signature of the vehicle.
  • IIR or Infinity Impulse Response filters are used in English terminology.
  • the method and the system according to the present invention concern the equalization of the amplitude of the frequency response of the passenger compartment.
  • digital audio filters are iteratively generated, and the digital audio filters are applied to the second digital data to generate third digital data for a compensated response curve. .
  • the frequency, gain and bandwidth of the digital audio filters are automatically optimized until the compensated response curve is within the tolerance range or a predetermined limit of the number of digital audio filters has been reached. of the two taking place first.
  • scientific publication “Filter Design Method for Loudspeaker Equalization Based on IIR Parametric Filters” by German Ramos and Jose J. Lopez. Presentation of the invention
  • the present invention provides a method for providing equalization of a signal by determining filter parameters to reduce the difference between the amplitude of a frequency response representing the acoustic signature of a set of speakers in their environment and a target sound signature curve.
  • the present invention relates, in its most general sense, to an automated acoustic equalization method, characterized in that it comprises the following steps:
  • said method further comprising a step of optimizing the parameters of the filters to improve the performance of the system.
  • the method according to the present invention makes it possible to obtain automated acoustic equalization thanks to a precise and optimized calculation of filter parameters.
  • the interpolation step of the target sound signature curve is performed using the Hermite method.
  • said method further comprises a step of automatically optimizing the offset of the target response C ec , repeated at each iteration.
  • said method further comprises a step of smoothing the N frequency responses.
  • said method implements filters corresponding to the following types: "peak”, “notch” “high-shelf” and “low-shelf” depending on the shape of local maxima and local minima.
  • said method furthermore implements a global optimization algorithm to minimize the error.
  • the present invention also relates to an automated acoustic equalization system, characterized in that it comprises means for:
  • said method further comprising means for optimizing filter parameters to improve system performance.
  • FIG. 2 represents the target sound signature curve C ec within the meaning of the present invention, the frequency responses derived from the N impulse response measurements, as well as the average M;
  • FIG. 1 illustrates the different steps of the method according to the present invention.
  • the automated acoustic equalization method according to the present invention comprises the following steps:
  • RI N after issuing a precalibrated sound signal received by N microphones.
  • a logarithmic frequency scale translation of said average M of the N frequency responses is performed.
  • a target sound signature curve C ec is interpolated and then translated into a logarithmic frequency scale.
  • the averaged response M and the target sound signature response C ec are compared by calculating the difference between the averaged response M and the target response C ec .
  • Curve C of itt resulting from the difference between said averaged response M and C ec said target response is analyzed.
  • filter parameters are determined for reducing the difference between said averaged response M and said target response C ec by firstly processing the local maxima in descending order according to their gain, then the local minima, and performing successive iterations.
  • the method according to the present invention further comprises a step of optimizing the parameters of the filters in order to improve the performance of the system.
  • Frequency responses can be averaged "standard” (that is, with identical weights), or with different weights.
  • FIG. 2 represents the target sound signature curve C ec within the meaning of the present invention, the frequency responses derived from the N impulse response measurements, as well as the average M.
  • a comparison of the averaged response M and the target response C ec by calculating the difference between the averaged response M and the target response C ec .
  • Figure 3 illustrates the detection and ranking of local maxima ("peaks") and local minima ("troughs").
  • the local maxima (peaks) are first processed in descending order according to their gain, then the local minima (troughs) are processed in ascending order. This makes it possible to determine filter parameters for reducing the difference between the averaged response M and the target response C ec . Successive iterations are performed. It has been shown in scientific studies that it is preferable to first equalize the peaks and then the troughs. Indeed, the human ear is more sensitive to peaks than hollows.
  • optimizing the offset of the target curve is performed as follows:
  • the target curve and the average frequency response are recalculated on a log scale to approximate the nonuniform resolution of the auditory system. This is achieved by a smoothing function that resamples the frequency response on a logarithmic scale with for example a frequency resolution of 1/48 octave.
  • the "FreqRange” optimization frequency band is applied as a FreqWeight weight vector that is 0 outside the frequency band and 1 within the frequency band.
  • the initial offset value "Offset" (in dB) is calculated as the average value of the average frequency response in the equalization frequency band:
  • n are respectively the first and the last of the frequency points of the logarithmic equalization frequency band.
  • Parameters and number of filters are optimized using an algorithm.
  • the parameters f, Q and G (respectively central frequency, quality factor and gain of the biquads) are optimized from intervals of values that can be predefined by a user, and the ranges of values of Q and G may depend on frequency. Thus, for example in high frequencies, the low gain filters are more easily eliminated because they are not perceptible.
  • the goal is to find the optimal parameters (fc op t, G t op, Qopt) of a filter and the optimum offset of the target curve Offset op t.
  • the limits of the parameters are determined like this: J
  • TargetGain- 100 ⁇ TargetGain opt ⁇ TargetGain + 100
  • f c and G are respectively the center frequency and the gain of a biquad filter modeling the nth peak and QRange is the range of admissible values of the quality factor Q.
  • a post-optimization process is performed. This postoptimization process consists of reclassifying the filters by increasing frequency and reoptimizing the coefficients. If a filter is canceled during this process, a new peak / dip is searched in order to output the maximum number of filters. The optimization process is implemented until the maximum number of filters is reached.
  • the interpolation step of the target curve is performed using the Hermite method.
  • the method according to the present invention further comprises a step of automatically optimizing the offset of the target response C ec , repeated at each iteration.
  • the method according to the present invention further comprises a step of smoothing the N frequency responses.
  • filters corresponding to the following types are used: "peak”, “notch” shelf “and” low-shelf “depending on the shape of local maxima (peaks) and local minima (troughs).
  • the filter is made according to whether or not a certain threshold is exceeded by the quality factor.
  • the method according to the present invention further implements a global optimization algorithm to minimize the error.
  • the present invention also relates to an automated acoustic equalization system, comprising means for:
  • said system further comprising means for optimizing filter parameters to improve system performance.

Landscapes

  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Circuit For Audible Band Transducer (AREA)
  • Stereophonic System (AREA)
EP15703951.2A 2014-02-25 2015-02-03 Verfahren und system zur automatischen akustischen entzerrung Active EP3111667B1 (de)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
FR1451504A FR3018015B1 (fr) 2014-02-25 2014-02-25 Procede et systeme d'egalisation acoustique automatise
PCT/EP2015/052199 WO2015128160A1 (fr) 2014-02-25 2015-02-03 Procede et systeme d'egalisation acoustique automatise

Publications (2)

Publication Number Publication Date
EP3111667A1 true EP3111667A1 (de) 2017-01-04
EP3111667B1 EP3111667B1 (de) 2018-04-11

Family

ID=50933309

Family Applications (1)

Application Number Title Priority Date Filing Date
EP15703951.2A Active EP3111667B1 (de) 2014-02-25 2015-02-03 Verfahren und system zur automatischen akustischen entzerrung

Country Status (5)

Country Link
EP (1) EP3111667B1 (de)
CN (1) CN106063293B (de)
ES (1) ES2676577T3 (de)
FR (1) FR3018015B1 (de)
WO (1) WO2015128160A1 (de)

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP4040810A1 (de) 2021-02-09 2022-08-10 Arkamys Verfahren zur automatisierten einstellung von parametern für die digitale verarbeitung, die auf signale vor der ausstrahlung über lautsprecher angewendet werden, und vorrichtung zur durchführung eines solchen verfahrens

Families Citing this family (11)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
FR3050601B1 (fr) 2016-04-26 2018-06-22 Arkamys Procede et systeme de diffusion d'un signal audio a 360°
CN106877820B (zh) * 2017-01-12 2020-08-11 广州市迪声音响有限公司 一种动态改变均衡增益的均衡系统和方法
CN109889955B (zh) * 2019-01-28 2021-01-12 中科上声（苏州）电子有限公司 一种车内声场的鲁棒性自动均衡方法和系统
CN114287137B (zh) * 2019-09-20 2024-07-23 哈曼国际工业有限公司 基于高斯分布和k最近邻算法的房间校准
CN112584274B (zh) * 2019-09-27 2022-05-03 宏碁股份有限公司 等化处理的调整系统及其调整方法
FR3107982B1 (fr) * 2020-03-05 2023-05-19 Faurecia Clarion Electronics Europe Procédé et système de détermination de filtres d’égalisation sonore d’un système audio
CN112769410B (zh) * 2020-12-25 2024-06-11 西安讯飞超脑信息科技有限公司 滤波器构建方法、音频处理方法及电子设备、存储装置
CN113949968B (zh) * 2021-09-07 2024-10-01 万魔声学股份有限公司 一种频响校正方法、电子设备及信号处理方法
CN114157965B (zh) * 2021-11-26 2024-03-29 国光电器股份有限公司 一种音效补偿方法、装置、耳机和存储介质
CN115604628B (zh) * 2022-12-12 2023-04-07 杭州兆华电子股份有限公司 一种基于耳机喇叭频响的滤波器校准方法及装置
GB2628409A (en) * 2023-03-24 2024-09-25 Tymphany Worldwide Enterprises Ltd Calibration of a loudspeaker system

Family Cites Families (5)

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Publication number Priority date Publication date Assignee Title
US7583806B2 (en) * 2003-06-09 2009-09-01 Bose Corporation Convertible automobile sound system equalizing
CA2568916C (en) * 2005-07-29 2010-02-09 Harman International Industries, Incorporated Audio tuning system
JP4888163B2 (ja) * 2007-03-09 2012-02-29 ヤマハ株式会社 カラオケ装置
EP2326108B1 (de) * 2009-11-02 2015-06-03 Harman Becker Automotive Systems GmbH Phasenentzerrung für Audiosystem
FR2967861B1 (fr) * 2010-11-18 2013-11-22 Ct Scient Tech Batiment Cstb Systeme electroacoustique pour une salle de spectacle

Cited By (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP4040810A1 (de) 2021-02-09 2022-08-10 Arkamys Verfahren zur automatisierten einstellung von parametern für die digitale verarbeitung, die auf signale vor der ausstrahlung über lautsprecher angewendet werden, und vorrichtung zur durchführung eines solchen verfahrens
FR3119723A1 (fr) 2021-02-09 2022-08-12 Arkamys Procédé pour le réglage automatisé de paramètres de traitements numériques appliqués à des signaux avant diffusion par des haut-parleurs et dispositif pour la mise en œuvre d’un tel procédé

Also Published As

Publication number Publication date
WO2015128160A1 (fr) 2015-09-03
CN106063293B (zh) 2019-06-07
FR3018015A1 (fr) 2015-08-28
CN106063293A (zh) 2016-10-26
FR3018015B1 (fr) 2016-04-29
ES2676577T3 (es) 2018-07-23
EP3111667B1 (de) 2018-04-11

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