WO2015085924A1 - Procédé d'égalisation automatique de haut-parleur - Google Patents

Procédé d'égalisation automatique de haut-parleur Download PDF

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Publication number
WO2015085924A1
WO2015085924A1 PCT/CN2014/093456 CN2014093456W WO2015085924A1 WO 2015085924 A1 WO2015085924 A1 WO 2015085924A1 CN 2014093456 W CN2014093456 W CN 2014093456W WO 2015085924 A1 WO2015085924 A1 WO 2015085924A1
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WO
WIPO (PCT)
Prior art keywords
signal
speaker
frequency
equalization
function
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PCT/CN2014/093456
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English (en)
Chinese (zh)
Inventor
叶超
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苏州上声电子有限公司
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Publication of WO2015085924A1 publication Critical patent/WO2015085924A1/fr

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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
    • H04R29/00—Monitoring arrangements; Testing arrangements
    • H04R29/001—Monitoring arrangements; Testing arrangements for loudspeakers
    • 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
    • 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
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
    • H04S7/30—Control circuits for electronic adaptation of the sound field
    • H04S7/301—Automatic calibration of stereophonic sound system, e.g. with test microphone
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
    • H04S7/30—Control circuits for electronic adaptation of the sound field
    • H04S7/307—Frequency adjustment, e.g. tone control

Definitions

  • the present invention relates to the field of audio signal processing technologies, and in particular, to a speaker automatic equalization system, which aims to correct the frequency characteristics of a listening position in a room, improve the sound reproduction capability of the speaker system, and improve the sound quality. More specifically, this equalization method includes a virtual bass enhancement technique that generates harmonic components of low frequency components by nonlinearity to improve the sensing ability of low frequency components.
  • the ideal sound reproduction system should have a relatively flat frequency response in the whole frequency band, but due to the limitation of the manufacturing process in the production process, the speaker system cannot have an ideal frequency response, and there is a certain distortion. On the other hand, due to the influence of the room modality and the interaction between the speaker system and the room, true sound reproduction cannot be achieved completely at the listening position. Therefore, the sound field correction technique is required to equalize the speaker system such that the frequency response at one or more position points approaches an ideal straight curve to ensure true playback of the original signal.
  • the existing equalization techniques have a graphic equalizer and a parametric equalizer, which are mainly a set of cascaded peak or slope filters, and the center frequency of each filter corresponds to an octave or a 1/3 octave.
  • the frequency band is controlled by adjusting the gain of each filter to achieve correction of the entire frequency band.
  • the frequency response of the speaker system at one or more points is first measured by a microphone, and then the equalizer design is performed according to the measured curve.
  • the equalizer is in the form of FIR (Finite Impulse Response) or IIR ( An infinite impulse response filter that filters the input signal such that an approximately flat frequency response is obtained at each location.
  • FIR Finite Impulse Response
  • IIR An infinite impulse response filter that filters the input signal such that an approximately flat frequency response is obtained at each location.
  • FIR Finite Impulse Response
  • IIR An infinite impulse response filter that filters the input signal such that an approximately flat frequency response is obtained at each location.
  • the frequency resolution of the low frequency band is low, in order to improve the low frequency resolution, it is necessary to increase the order of the filter, which increases the computational complexity.
  • the direct equalization method is used to increase the energy of the low-frequency signal, the playback signal may be distorted and the speaker system may be damaged.
  • the virtual bass enhancement technology based on the psychoacoustic fundamental frequency missing principle can solve this problem well.
  • the perception of low frequency sound can be improved subjectively, and the low frequency playback capability of the small aperture speaker can be improved. .
  • the present invention provides a speaker automatic equalization method, which in turn includes the following steps:
  • the equalization filter output signal drives the speaker unit through the power amplifier.
  • the method of measuring the speaker system transfer function mentioned in the above described scheme may employ a frequency sweep signal or a maximum length sequence (MLS), or other impulse response measurement method.
  • the selected measurement location point should be the preferred listening location in the room, or cover a preferred listening area, such as a home theater or various seat locations within the car.
  • the calculation process of the prototype function of determining a plurality of transfer functions is as follows:
  • a i is a weighting coefficient
  • different location points can be weighted according to actual conditions. For example, in a home theater, the sound quality of the position of the screen is emphasized, and other positions are considered. Another example is that in the interior of the car, the front or rear seats can be weighted differently according to actual needs.
  • the prototype function is the root mean square value of the M transfer functions; or
  • b i is a weighting coefficient, and different location points can be weighted according to actual conditions.
  • the prototype function is the arithmetic mean of the M transfer functions.
  • the prototype function describes the common features of multiple position point transfer functions. In the room sound field, the prototype function extracts the common characteristics of each position point from the aspects of direct sound, early reflection sound and reverberation sound, which can be achieved by equalizing the prototype function. Correction of the sound field for multiple locations.
  • the design method of the prototype function equalization filter in step 3) may adopt a time domain adaptive optimization algorithm, including a minimum mean square error method, a recursive least square method, and the like.
  • the adaptive algorithm adjusts its filter parameters by automatic iteration to meet the minimum criteria, thus achieving optimal filter coefficients.
  • the low frequency lower limit frequency of the speaker in step 4) is determined by its physical characteristics; the low pass filter may be in the form of FIR (finite impulse response) or IIR (infinite impulse response) filter.
  • the amplitude-frequency characteristic of the IIR filter is highly accurate.
  • the system function can be written in the form of a closed function. It is implemented in a recursive structure with low computational complexity, but the phase characteristics are not linear, and the stability of the system needs to be considered.
  • the amplitude and frequency characteristics of the FIR filter are lower than that of the IIR. Generally, there is no analytical expression and the computational complexity is high. The significant advantage is that the system is stable and has a linear phase.
  • the nonlinear algorithm for generating the higher harmonic signal in step 5) may be a polynomial function, an exponential function or a power function and other nonlinear functions to generate a higher harmonic component of the input low frequency signal.
  • the dynamic range control in the step 6) refers to dynamic control of the higher harmonic signal, and the control of the perceptible low frequency signal is realized by peak detection and gain control of the higher harmonic signal.
  • the power amplifier of step 7) can have both analog and digital implementations. If the analog implementation is adopted, the digital signal output by the equalization filter is digital-to-analog converted into an analog signal, and then the power amplifier performs signal power amplification; if digital implementation is used, the digital signal output from the equalization filter is directly fed to the digital power amplifier. Signal power is amplified.
  • the speaker unit described in step 7) may be a dynamic speaker of various sizes and specifications.
  • the present invention can combine the speaker system with the acoustic characteristics of the room by on-line measurement and real-time equalization of the speaker system in the use environment, thereby improving the performance of the speaker system in a specific use environment.
  • the invention processes the low frequency and the high frequency of the frequency response of the speaker system separately, performs virtual bass enhancement for the low frequency signal below the lower limit frequency, adaptively equalizes the signal above the lower limit frequency, and improves the speaker system in the full frequency band. Sound playback capability.
  • the invention can equalize multiple points in the room, realize multi-point balance by extracting prototype functions of multiple position point transfer functions, and avoid the sound quality damage that may be caused to other points after equalization of one point in the room.
  • the present invention adopts a time domain adaptive algorithm to calculate the equalization filter, which can effectively improve the accuracy of the equalization, and adopts a time domain equalization algorithm, which can avoid the frequency domain algorithm to simultaneously consider the amplitude and phase balance, and reduce the computational complexity. .
  • 1 is a flow chart of signal processing of the speaker automatic equalization system of the present invention
  • FIG. 2 is a flow chart of the sound field equalization process in FIG. 1;
  • FIG. 3 is a schematic diagram of calculating an equalization filter using an adaptive algorithm in FIG. 2;
  • FIG. 4 is a flow chart of signal processing for realizing virtual bass boosting in FIG. 1;
  • Figure 5 is a flow chart of signal processing for implementing dynamic range control in Figure 1;
  • FIG. 6 is a time-domain graph of an equalization filter according to an embodiment of the present invention.
  • FIG. 7A is a time-domain impulse response graph of a speaker system according to an embodiment of the present invention.
  • 7B is a time-domain impulse response graph of a speaker system after equalization according to an embodiment of the present invention.
  • 8A is a graph showing a frequency response curve of a speaker system according to an embodiment of the present invention.
  • 8B is a graph showing a frequency response of a speaker system after equalization according to an embodiment of the present invention.
  • the invention firstly measures the impulse response of the speaker system at a plurality of position points in the room through a microphone, determines a prototype function from the plurality of impulse responses, calculates an equalization filter of the prototype function by using an adaptive optimization algorithm, and determines the speaker system according to the prototype function.
  • the low frequency lower limit frequency is used to perform virtual bass boost on the input signal below this frequency to achieve automatic equalization of the speaker system in the full frequency band.
  • the speaker automatic equalization system according to the present invention as shown in FIG. 1 is mainly composed of a sound source 101, a virtual bass enhancement module 102, a dynamic range control 103, a delay unit 104, an equalization filter 105, and a power amplifier 106. It is composed of a speaker unit 107 and the like.
  • the sound source 101 is connected to the input end of the virtual bass enhancement 102 for enhancing the bass below the lower limit frequency of the speaker system; the output of the virtual bass enhancement module 102 is connected to the input of the dynamic range control 103, The signal of the virtual bass processing is dynamically controlled to remove noise; the output of the dynamic range control 103 is added to the output of the delay unit 104, and then connected to the input end of the equalization filter 105, and the input signal is equalized and then sent to the power.
  • the amplifier 106 amplifies the equalized signal and drives the speaker unit 107 to sound.
  • the calculation process of the equalization filter 105 in FIG. 1 is shown in FIG. 2.
  • the specific implementation step is to first measure the impulse response of a plurality of position points in the room by using a microphone, and obtain the prototype function by using the above formula (1) or (2).
  • 202 since the prototype function 202 is generally a non-minimum phase system, it can be divided into a minimum phase system 203 and an all-pass system 204, respectively obtaining amplitude information 205 and phase information 206; and then frequency transforming it using equation (3) 207. Transform the prototype function from a linear frequency to a bending frequency to improve low frequency resolution.
  • z -1 is a frequency domain delay unit
  • ⁇ is a bending factor
  • the value ranges from -1 ⁇ 1.
  • the equalization filter 209 is obtained using the adaptive optimization algorithm 208.
  • the schematic diagram of the adaptive optimization algorithm is shown in Figure 3, where x(t) is the input signal, H(z) is the filter, P(z) is the system function, and d(t) and y(t) are the expected
  • the signal and output signals, e(t) d(t) - y(t), are error signals.
  • the error signal e(t) is minimized by updating the coefficients of the filter H(z) by an optimization algorithm.
  • the calculation process of the adaptive optimization algorithm is specifically described by taking the minimum mean square error method as an example.
  • the filter coefficient is calculated as
  • ⁇ is the step factor.
  • the virtual bass boost module is shown in Figure 4.
  • the input signal 401 first passes through the low pass filter 402, generates the higher harmonics 403 by nonlinearity, and then performs the band pass filtering 404 and the original input signal through the delay unit 405.
  • the number 401 is superimposed to obtain an output signal 406 that is enhanced by the virtual bass.
  • the manner in which the harmonics 403 are nonlinearly generated may be a polynomial function, an exponential function or a power function, and other nonlinear functions.
  • the manner in which the harmonics are generated nonlinearly may be in the form of a polynomial of equation (6).
  • the dynamic range control processing flow is shown in Figure 5.
  • the virtual bass enhanced output signal 406 is used as the dynamic range controlled input signal 501, multiplied by the peak detection 502 and the gain control 503 and the original input signal passed through the delay unit 504 to achieve dynamic range of the original input signal 501. control.
  • the speaker unit is 3.5 inches in size, and the impulse response and frequency response of the speaker system are first measured using a microphone as shown in Figs. 7A and 8A, respectively.
  • the adaptive optimization method is used to obtain the equalization filter, which adopts the 300-order FIR form, and the time domain waveform is shown in Fig. 6.
  • the pulse response and frequency response after equalization of the speaker system are shown in Figures 7B and 8B, respectively. It can be seen from the figure that after the equalization process, the impulse response of the speaker system is sharper, the frequency response is flatter, and the frequency characteristics are better. And use the virtual bass enhancement algorithm for bass compensation. Through the actual audition, the performance of the low frequency band is obviously enhanced, the music of the middle and high frequency is brighter and the sound is more natural.

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  • Health & Medical Sciences (AREA)
  • General Health & Medical Sciences (AREA)
  • Otolaryngology (AREA)
  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Tone Control, Compression And Expansion, Limiting Amplitude (AREA)
  • Circuit For Audible Band Transducer (AREA)

Abstract

La présente invention concerne un procédé d'égalisation automatique de haut-parleur qui améliore la performance de lecture sonore d'un système de haut-parleur dans toute une bande de fréquence. Le procédé consiste à : mesurer des réponses d'impulsions au niveau d'une ou de plusieurs positions ponctuelles d'une pièce au moyen d'un haut-parleur, obtenir une réponse de fréquence au niveau de chaque position ponctuelle et une limite basse de basse fréquence du système de haut-parleur au moyen d'un algorithme d'optimisation auto-adaptative pour obtenir un filtre d'égalisation et compenser le système de haut-parleur. En ce qui concerne un signal basse fréquence de fréquence inférieure à une fréquence limite basse du système de haut-parleur, on utilise le principe de la fondamentale manquante basé sur la psychoacoustique, et une composante d'harmonique de rang élevé d'un signal de fréquence fondamentale est générée et est superposée sur un signal audio d'origine retardé après une commande de gain et, ainsi, on améliore la capacité de lecture sonore du système de haut-parleur dans toute la bande de fréquence.
PCT/CN2014/093456 2013-12-11 2014-12-10 Procédé d'égalisation automatique de haut-parleur WO2015085924A1 (fr)

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US9794688B2 (en) 2015-10-30 2017-10-17 Guoguang Electric Company Limited Addition of virtual bass in the frequency domain
US9794689B2 (en) 2015-10-30 2017-10-17 Guoguang Electric Company Limited Addition of virtual bass in the time domain
US10405094B2 (en) 2015-10-30 2019-09-03 Guoguang Electric Company Limited Addition of virtual bass
US10893363B2 (en) 2018-09-28 2021-01-12 Apple Inc. Self-equalizing loudspeaker system
US10893362B2 (en) 2015-10-30 2021-01-12 Guoguang Electric Company Limited Addition of virtual bass
WO2021008684A1 (fr) * 2019-07-16 2021-01-21 Ask Industries Gmbh Procédé de reproduction d'un signal audio dans un habitacle de voiture par l'intermédiaire d'un système audio de voiture
CN113852903A (zh) * 2021-10-21 2021-12-28 杭州爱华智能科技有限公司 电容型测试传声器的声场特性转换方法与电容型测试传声器系统
CN116367076A (zh) * 2023-03-30 2023-06-30 潍坊歌尔丹拿电子科技有限公司 车辆内音频处理方法、设备及存储介质
CN117676418A (zh) * 2023-12-06 2024-03-08 广州番禺职业技术学院 一种用于混合相位系统中的声场均衡方法及系统

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FR3052951B1 (fr) * 2016-06-20 2020-02-28 Arkamys Procede et systeme pour l'optimisation du rendu sonore de basses frequences d'un signal audio
CN108668193A (zh) * 2017-03-30 2018-10-16 展讯通信（上海）有限公司 一种用于播放设备的低音增强方法、装置以及播放设备
CN107222808B (zh) * 2017-05-03 2019-11-19 上海大学 一种高保真扬声器重放系统设计方法
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WO2021236076A1 (fr) * 2020-05-20 2021-11-25 Harman International Industries, Incorporated Système, appareil et procédé pour ensembles de réseaux de microphone-haut-parleur adaptatif multidimensionnel pour correction et égalisation de pièce
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CN115226001B (zh) * 2021-11-24 2024-05-03 广州汽车集团股份有限公司 声能量补偿方法、装置及计算机设备

Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN1630427A (zh) * 2004-06-30 2005-06-22 深圳兰光电子集团有限公司 一种低音增效处理的方法
CN101053152A (zh) * 2005-07-29 2007-10-10 哈曼国际工业有限公司 音频调谐系统
US20090016543A1 (en) * 2007-07-12 2009-01-15 Oki Electric Industry Co., Ltd. Acoustic signal processing apparatus and acoustic signal processing method
CN103634726A (zh) * 2013-08-30 2014-03-12 苏州上声电子有限公司 一种扬声器自动均衡方法

Family Cites Families (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
ATE470323T1 (de) * 2006-01-03 2010-06-15 Sl Audio As Verfahren und system zum entzerren eines lautsprechers in einem raum
CN102447446A (zh) * 2011-12-09 2012-05-09 苏州上声电子有限公司 基于振动元件运动状态反馈的扬声器频响均衡方法和装置

Patent Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN1630427A (zh) * 2004-06-30 2005-06-22 深圳兰光电子集团有限公司 一种低音增效处理的方法
CN101053152A (zh) * 2005-07-29 2007-10-10 哈曼国际工业有限公司 音频调谐系统
US20090016543A1 (en) * 2007-07-12 2009-01-15 Oki Electric Industry Co., Ltd. Acoustic signal processing apparatus and acoustic signal processing method
CN103634726A (zh) * 2013-08-30 2014-03-12 苏州上声电子有限公司 一种扬声器自动均衡方法

Cited By (12)

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Publication number Priority date Publication date Assignee Title
US9794688B2 (en) 2015-10-30 2017-10-17 Guoguang Electric Company Limited Addition of virtual bass in the frequency domain
US9794689B2 (en) 2015-10-30 2017-10-17 Guoguang Electric Company Limited Addition of virtual bass in the time domain
US10405094B2 (en) 2015-10-30 2019-09-03 Guoguang Electric Company Limited Addition of virtual bass
US10893362B2 (en) 2015-10-30 2021-01-12 Guoguang Electric Company Limited Addition of virtual bass
US10893363B2 (en) 2018-09-28 2021-01-12 Apple Inc. Self-equalizing loudspeaker system
WO2021008684A1 (fr) * 2019-07-16 2021-01-21 Ask Industries Gmbh Procédé de reproduction d'un signal audio dans un habitacle de voiture par l'intermédiaire d'un système audio de voiture
US11800311B2 (en) 2019-07-16 2023-10-24 Ask Industries Gmbh Method of reproducing an audio signal in a car cabin via a car audio system
CN113852903A (zh) * 2021-10-21 2021-12-28 杭州爱华智能科技有限公司 电容型测试传声器的声场特性转换方法与电容型测试传声器系统
CN113852903B (zh) * 2021-10-21 2022-05-31 杭州爱华智能科技有限公司 电容型测试传声器的声场特性转换方法与电容型测试传声器系统
CN116367076A (zh) * 2023-03-30 2023-06-30 潍坊歌尔丹拿电子科技有限公司 车辆内音频处理方法、设备及存储介质
CN117676418A (zh) * 2023-12-06 2024-03-08 广州番禺职业技术学院 一种用于混合相位系统中的声场均衡方法及系统
CN117676418B (zh) * 2023-12-06 2024-05-24 广州番禺职业技术学院 一种用于混合相位系统中的声场均衡方法及系统

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