KR101307430B1 - Method and device for real-time performance evaluation and improvement of speaker system considering power response of listening room - Google Patents

Abstract

PURPOSE: A real time performance assessment and improvement method and a device thereof are provided to efficiently improve a speaker system. CONSTITUTION: A real time performance assessment and improvement device of a speaker system comprises a microphone unit (21), a signal measuring unit (22), a signal analyzing unit (23) and a screen configuring unit (24). The microphone unit is installed at a listening expectation position in a listening space and measures an output signal of the speaker system. The signal measuring unit receives an input signal from the speaker system and receives the output signal from the microphone unit. The signal analyzing unit analyzes and corrects the time delay of the output signal from relation between the output signal and the input signal and performs frequency domain analysis about the output signal in which the time delay is corrected and the input signal. The screen configuring unit configures a screen and indicates the result of the frequency domain analysis. [Reference numerals] (12) Signal playing unit; (14) Signal amplifying unit; (22) Signal measuring unit; (23) Signal analyzing unit; (24) Screen configuring unit; (25) Filter planning and revising unit; (26) Verification unit for the validity of sound source selection

Description

TECHNICAL FIELD [0001] The present invention relates to a real-time performance evaluation and improvement method and apparatus for a speaker system considering a power response in a listening space,

The present invention relates to a method and apparatus for real-time performance evaluation and improvement of a speaker system in consideration of a power response in a listening room, which is an actual living space.

Frequency characteristics in speaker systems are important factors for performance evaluation. If the transfer function is obtained by using the speaker playback signal and the microphone measurement signal, the frequency characteristics of the speaker can be known. In Korean Patent Laid-Open Publication No. 2005-0091192 (titled "Method and Apparatus for Measuring the Characteristics of a Speaker"), a sound produced by a speaker is detected through a microphone, a frequency component is detected, Is disclosed.

In general, the HIFI Speaker System can predict most of the performance by measuring the On axis and the directivity (mainly horizontal) in free space conditions. In addition, the necessary tuning can be achieved by taking into account the gain due to the bottom when tuning the subwoofer attached to the floor such as 3 way (3 way), 2.1 channel, 5.1 channel, etc. in the room . This is because most of the speaker's playback energy is concentrated on the axis (on axis) and the listening window.

However, in recent years, digital TVs have been accelerated to be slimmer and adopt 2 or 2.1 channel type speakers, which are mainly composed of bottom emission or back emission (mainly woofer). In the case of lower radiation, horizontal non-directional characteristics are exhibited. In the listening space, interaction with the rear wall is a major variable, especially in digital TV installation.

In the case of tuning a speaker employed in such a digital TV, general measurement is carried out under free space conditions. The result of the tuning in this condition is that a real speaker system is installed, and a listening space as a living space Room), the distortion of the frequency and gain of each frequency is caused by the condition of the wall surface, and the sound quality is felt in a direction different from the result of tuning in the anechoic room. Therefore, in the actual listening space, there is a difficulty in efficiency and precise tuning since it is re-tuned depending on the feeling that the accuracy and reproducibility are relatively low.

In the case of a system to which transfer function calculation is applied in the field of noise / vibration, time delay due to sound propagation to be considered in the audio field is not considered. In the audio field, tuning is performed through the analysis after re-tuning or measuring the speaker tuned in the anechoic room. However, in order to ensure accuracy and reproducibility in the actual listening space, it is necessary to analyze the transfer function in real time There is a need. In addition, a real-time analysis system is required because the magnitude and frequency characteristics of sound reproduced in real time when a sound source is reproduced through a speaker vary.

The present invention calculates a real-time transfer function, an impulse response function, and a contribution function of a speaker system in consideration of a power response in a room acquired and analyzed through multi-point measurement in an actual listening room, (For example, LabVIEW), and to provide an apparatus and method for real-time performance evaluation and improvement of a speaker system that considers power response in a listening space capable of designing and correcting a filter according to a target frequency response characteristic.

In addition, the present invention provides a speaker system considering a power response in a listening space, which can verify the validity of a sound source selection for performance evaluation of a real-time speaker system using a contribution function between an input (sound source) signal and an output Time performance evaluation and improvement apparatus and method.

Other objects of the present invention will become readily apparent from the following description.

According to an aspect of the present invention, there is provided a method for evaluating and improving performance of a speaker system in real time in a listening space provided with a speaker system in which a predetermined sound source is output as an input signal, Measuring an output signal of the speaker system at a microphone; Receiving the input signal from the speaker system at a signal measurement unit; Analyzing and correcting a time delay of the output signal from a relationship between the input signal and the output signal in a signal analysis unit; Performing a frequency domain analysis on the output signal and the input signal in which the time delay is corrected by the signal analyzing unit; Displaying a frequency domain analysis result on a screen constitution unit; And designing and correcting a filter to be applied to the speaker system according to the frequency domain analysis result in the filter designing and correcting unit.

The step of analyzing and correcting the time delay may calculate a cross-correlation function between the input signal and the output signal, and determine a portion where a maximum peak value appears by applying a window function (Optimal Maximum Likelihood) as a delay time.

The performing the frequency domain analysis may include performing a transfer function analysis, performing an impulse response function analysis, and performing a contribution function analysis.

The step of performing the impulse response function analysis may include: a complex number conversion step of converting the transfer function into a complex number type divided into a real part and an imaginary part; a data mirroring step of generating a symmetric signal to increase the number of signals by two times; And performing a Fourier transform.

The step of performing the contribution function analysis may calculate a contribution function indicating the similarity between the input signal and the output signal, and the analysis result may be expressed as the degree of contribution of the input signal to the output signal.

는 주파수 영역에서의 상기 입력 신호와 상기 출력 신호의 기여도 함수이며,

는 주파수 영역에서의 상기 입력 신호의 자기상관함수이고,

는 주파수 영역에서의 상기 출력 신호의 자기상관함수이며,

는 주파수 영역에서의 상기 입력 신호와 상기 출력 신호의 상호상관함수이다.here,

Is a function of the contribution of the input signal and the output signal in the frequency domain,

Is an autocorrelation function of the input signal in the frequency domain,

Is an autocorrelation function of the output signal in the frequency domain,

Is a cross-correlation function of the input signal and the output signal in the frequency domain.

The step of designing and calibrating the filter may include receiving an actual frequency response analysis result and a target frequency response characteristic of the speaker system in a filter design and correction unit; Comparing the actual frequency response analysis result and the target frequency response characteristic to calculate a characteristic difference; Selecting a frequency region in which the characteristic difference exists and selecting a filter type corresponding to the selected frequency region; Calculating a level difference between the actual frequency response analysis result and the target frequency response characteristic in the selected frequency domain and setting a correction coefficient for compensating the level difference; Designing the filter according to the selected filter type and the set correction coefficient; And applying the designed filter to the speaker system.

And a sound source selection feasibility verification step of determining that the selection of the sound source is valid as the input signal when an even contribution value within a predetermined error is displayed over the audio frequency band using the result of the contribution function analysis.

According to another aspect of the present invention, there is provided an apparatus for evaluating and improving performance of a speaker system in real time in a listening space provided with a speaker system in which a predetermined sound source is output as an input signal, A microphone unit installed to measure an output signal of the speaker system; A signal measuring unit receiving the input signal from the speaker system and receiving the output signal from the microphone unit; Analyzing and correcting a time delay of the output signal from a relationship between the input signal and the output signal, and performing a frequency domain analysis on the output signal with the time delay corrected and the frequency domain analysis part; A screen configuration unit configured to display the frequency domain analysis result on a screen; And a filter designing and correcting unit for designing and correcting a filter to be applied to the speaker system according to a result of the frequency domain analysis, wherein the frequency domain analysis includes a transfer function analysis, an impulse response function analysis and a contribution function analysis A real time performance evaluation and improvement apparatus of a speaker system is provided.

And a sound source selection feasibility verifying unit that determines that the selection of the sound source is valid as the input signal when the contribution signal is within a predetermined error range over the audio frequency band using the result of the contribution function analysis.

Other aspects, features, and advantages will become apparent from the following drawings, claims, and detailed description of the invention.

According to an embodiment of the present invention, a real-time transfer function, an impulse response function, and a contribution function of a speaker are calculated in consideration of a power response in a room acquired and analyzed through a multi-point measurement in a listening room, It is possible to design an efficient speaker system since the filter design corresponding to the targeted frequency response characteristic is automatically performed.

In addition, since the validity of the input signal and the output signal can be confirmed, the validity of the input signal and the output signal can be verified when selecting the sound source used for the speaker performance evaluation.

1 is a block diagram showing a schematic configuration of a real-time performance evaluation and improvement apparatus and a speaker system of a speaker system considering power response in a listening space according to an embodiment of the present invention;
FIG. 2 is a flowchart of a real-time performance evaluation and improvement method of a speaker system according to an embodiment of the present invention;
FIG. 3 is a diagram illustrating a case where a time delay occurs and a case where the time delay is corrected;
4 is an example of calculating a delay time,
5 is an exemplary output of a frequency response function, an impulse response function, a contribution function,
6 is a diagram showing a relationship of a transfer function between an input and an output,
FIG. 7 is an exemplary graph of the analysis result of the transfer function,
8 is a flowchart of a method for obtaining an impulse response function,
9 is a diagram showing an analysis example of the impulse response function,
10 is a diagram showing an analysis example of a contribution function,
11 is a diagram showing an example of a screen configuration of a performance evaluation and improvement apparatus of a speaker system,
12 is a flowchart of a filter design and correction method according to an embodiment of the present invention,
13 is a diagram showing an example in which the frequency response characteristic and the target frequency response characteristic are compared and corrected,
FIG. 14 is a view showing a kind of a filter used in a process of designing and correcting a filter,
15 is a diagram illustrating a difference in a contribution function according to a sound source selection according to an embodiment of the present invention.

The present invention is capable of various modifications and various embodiments, and specific embodiments are illustrated and described in the drawings. It should be understood, however, that the invention is not intended to be limited to the particular embodiments, but includes all modifications, equivalents, and alternatives falling within the spirit and scope of the invention. DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS Hereinafter, the present invention will be described in detail with reference to the accompanying drawings.

The terms first, second, etc. may be used to describe various components, but the components should not be limited by the terms. The terms are used only for the purpose of distinguishing one component from another.

The terminology used in this application is used only to describe a specific embodiment and is not intended to limit the invention. The singular expressions include plural expressions unless the context clearly dictates otherwise. In the present application, the terms "comprises" or "having" and the like are used to specify that there is a feature, a number, a step, an operation, an element, a component or a combination thereof described in the specification, But do not preclude the presence or addition of one or more other features, integers, steps, operations, elements, components, or combinations thereof.

The terms "part", "unit", "unit", "module", "block", and the like described in the specification mean units for processing at least one function or operation. Software or a combination of hardware and software.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT Hereinafter, embodiments of the present invention will be described in detail with reference to the accompanying drawings. Referring to the accompanying drawings, the same or corresponding components are denoted by the same reference numerals, .

1 is a block diagram showing a schematic configuration of a real-time performance evaluation and improvement apparatus and a speaker system of a speaker system considering power response in a listening space according to an embodiment of the present invention.

The apparatus 20 for real-time performance evaluation and improvement of a speaker system according to an embodiment of the present invention includes a fast Fourier transform (FFT), a power spectrum, a frequency response function, a coherence function, , Impulse response function, etc., and real-time performance evaluation and improvement of the speaker system are performed.

1, the speaker system 10 includes a signal reproducing unit 12, a signal amplifying unit 14, and a speaker unit 16, and the real-time performance evaluating and improving apparatus 20 of the speaker system includes one or more A microphone 21, a signal measuring unit 22, a signal analyzing unit 23, a screen forming unit 24, a filter designing and calibrating unit 25, and a sound source selection feasibility verifying unit 26.

In this embodiment, the speaker unit 16 is installed in a listening room corresponding to the actual listening space, and one or more microphone units 21 are installed in the vicinity of the listener's position expected in the listening room, So that the characteristics of the speaker can be accurately grasped in real time.

The signal reproducing unit 12 decodes and reproduces the encoded sound source file. The sound source file is encoded with one of the audio codecs of WAV, MPEG1 Layer 1/2/3, WMA, FLAC, OGG Vorbis, AAC / AAC +, AC3, BSAC, True Audio, WavPack, G.726, And the signal reproducing unit 12 may be an audio codec capable of decoding a sound source file.

The signal amplifying unit 14 receives the signal reproduced by the signal regenerating unit 12, increases the signal magnitude by a predetermined gain, and provides the output to the speaker unit 16 at the subsequent stage.

The speaker unit 16 converts the output signal from the signal amplifying unit 14 into a sound that the listener can hear. Examples of the sound converting method of the speaker unit 16 include a method of directly vibrating the air by vibrating the diaphragm, or receiving sound by receiving the vibration as a horn.

The microphone unit 21 is installed at a point (listening expectation position) at which the listener is expected to be positioned for listening to the sound source in the actual listening space provided in the speaker system, It is a device for monitoring. The microphone unit 21 may convert a sound wave inputted thereto into an electrical signal, and may have a structure in which a listener is located in a listening space and is easily installed and released at a position where listening is expected.

The signal measuring unit 22 is connected to the signal regenerating unit 12 and the microphone unit 21 via separate channels Ch1 and Ch2 so that an input ) Signal and a converted output (measurement) signal through the microphone section 21, respectively.

The signal analyzing unit 23 stores the input signal and the output signal inputted through the signal measuring unit 22 in a memory (not shown), and performs time delay analysis and frequency domain analysis in real time.

From the time delay analysis result, the delay time in the current listening space is calculated and reflected in the performance evaluation and improvement of the speaker system 10.

The screen configuration unit 24 displays input / output signals in the time domain, input / output signals in the frequency domain, and frequency domain analysis results on a screen in a graph form so that data can be measured and analyzed in real time, A portion (for example, the lower left portion) is set as a setting area so that the setting of data measurement and analysis results can be changed intuitively.

Frequency domain analysis includes transfer function analysis, impact response function analysis, and contribution function analysis. The filter designing and correcting unit 25 designs filters suitable for the listening space in which the speaker system 10 is installed or corrects the designed filters from the analysis results of these functions.

Also, the sound source selection feasibility verifying unit 26 verifies the feasibility of the sound source selection used for the performance evaluation and improvement of the speaker system 10 from the analysis result of the contribution function. The validity of sound source selection is judged based on whether or not it represents a uniform contribution value over the entire frequency band, for example, pink noise, which is noise including components of all frequencies.

The real-time performance evaluation and improvement apparatus 20 of the speaker system according to the embodiment of the present invention is applicable to any product including a speaker system 10 such as a digital TV.

Hereinafter, the operation of the real-time performance evaluation and improvement apparatus of the speaker system will be described in detail with reference to the related drawings.

FIG. 2 is a flowchart illustrating a method of evaluating and improving a real-time performance of a speaker system according to an embodiment of the present invention. FIG. 3 illustrates a case where a time delay occurs and a case where the time delay is corrected. FIG. 6 is a diagram showing a relationship of a transfer function between an input and an output, and FIG. 7 is a graph showing an analysis result of a transfer function Fig. 10 is a diagram showing an example of analysis of the contribution function, Fig. 11 is a graph showing an example of the analysis of the impact response function, Fig. FIG. 12 is a flowchart of a filter designing and correcting method according to an embodiment of the present invention, and FIG. 13 is a diagram illustrating a frequency response characteristic FIG. 14 is a view showing a kind of a filter used in a process of designing and correcting a filter, and FIG. 15 is a flowchart illustrating a method of selecting a sound source according to an embodiment of the present invention. In the case of the second embodiment of the present invention.

Referring to FIG. 2, the real-time performance evaluation and improvement method of the speaker system is performed in the real-time performance evaluation and improvement apparatus 20 of the speaker system shown in FIG. 1, And is performed by the respective components of the evaluation and improvement apparatus 20.

A real-time performance evaluation and improvement method of a speaker system includes a signal measurement step S110, a signal storage step S120, a time delay analysis step S130, a frequency domain analysis step S140, a screen configuration step S150, A correction step (S160), and a sound source selection feasibility verification step (S170). The frequency domain analysis step S140 may include a transfer function analysis step S142, an impulse response function analysis step S144, and a contribution function analysis step S146. Hereinafter, each step will be described in detail.

In the signal measuring step S110, the signal measuring unit 22 receives an input signal, that is, a sound source (excitation) signal, from the signal reproducing unit 12 of the speaker system 10, That is, a measurement (mastication) signal.

In the signal storage step S120, the signal analysis unit 23 stores the input signal and the output signal measured by the signal measurement unit 22 in a memory built in or externally connected to the real-time performance evaluation and improvement apparatus 20. [

In the time delay analysis step S130, the signal analysis unit 23 calculates a delay time, which is one of the characteristics of the listening space from the input signal and the output signal, and compensates the delay time for the corresponding time, To be performed.

The time delay is caused by the difference in the distance between the speaker unit 16 and the microphone unit 21 and the difference in the spaced space. When the output signal is compared with the input signal, when a time delay of 3.4 seconds occurs, the delay time compensation is performed by eliminating the time delay by preceding the output signal by 3.4 seconds as illustrated in FIG. 3 (b) Thereby suppressing the occurrence of an error due to the time delay between the input signal and the output signal.

The calculation process of the delay time is as follows.

First, after calculating the cross-correlation function between the input signal and the output signal, the following process is performed.

에 창함수(Optimal Maximum Likelihood)

를 적용한다. As shown in Equation 1,

A window function (Optimal Maximum Likelihood)

Is applied.

는 지연 시간을 계산하기 위한 시간 영역에서의 개선된 상호상관함수이다.here,

Is an improved cross-correlation function in the time domain for calculating the delay time.

는 수학식 2와 같은 입력 신호와 출력 신호의 기여도 함수를 적용하여 수학식 3과 같이 정의된다. Window function

Is defined as Equation (3) by applying the contribution function of the input signal and the output signal as shown in Equation (2).

는 주파수 영역에서의 입력 신호와 출력 신호의 기여도 함수이며,

는 주파수 영역에서의 입력 신호의 자기상관함수이고,

는 주파수 영역에서의 출력 신호의 자기상관함수이며,

는 주파수 영역에서의 입력 신호와 출력 신호의 상호상관함수이다. here,

Is a function of the contribution of the input signal and the output signal in the frequency domain,

Is an autocorrelation function of the input signal in the frequency domain,

Is an autocorrelation function of the output signal in the frequency domain,

Is a cross-correlation function of the input signal and the output signal in the frequency domain.

는 주파수 영역에서의 창함수이다. here,

Is a window function in the frequency domain.

에 곱하면 수학식 4와 같고, 이를 역푸리에 변환(IFFT)하면 개선된 상호상관함수

를 구할 수 있다. The window function above is a cross-correlation function

(IFFT) is performed as shown in Equation (4), and an improved cross-correlation function

Can be obtained.

As shown in FIG. 4, it can be seen that the maximum peak value appears at a point corresponding to the actually delayed portion (amount), and the value becomes the delay time.

Referring again to FIG. 2, after the time delay compensation is performed, the signal analysis unit 23 analyzes the frequency domain characteristics of the input signal and the output signal in the frequency domain analysis step S140.

5 (a) (size) and (c) of FIG. 5 show examples of the output of the frequency response function. In the analysis of the frequency domain characteristic, a frequency response function (transfer function), an impulse response function, (Phase), an example of the output of the impulse response function is shown in FIG. 5 (d), and an example of the output of the contribution function is shown in FIG. 5 (b).

In the transfer function analysis step S142, the signal analysis unit 23 analyzes the transfer function between the input signal and the output signal in the following process.

The transfer function is calculated by the relationship and the equation as shown in Fig. The transfer function H (f) represents the relationship between the input x (t) and the output y (t) and is expressed as the ratio of the output's Laplace transform (X (f)) to the input's Laplace transform (Y (f)).

Here, H (f) is the transfer function, X (f) is the input signal, X * (f) is the complex conjugate, Y (f) is the output signal, S _xx (f) of the input signal is the autocorrelation function of the input signal , And S _xy (f) are cross-correlation functions of the input signal and the output signal.

The calculated transfer function is shown as a graph of magnitude and phase and is illustrated in Figures 7 (a) and 7 (b). FRF stands for a frequency response function. This transfer function graph can be used to determine the distortion of the signal.

In the impulse response function analysis step S144, the signal analysis unit 23 can calculate the impulse response function h (t) through the inverse transformation of the transfer function, and can know the delay of the entire signal phase by analyzing the impulse response function .

Referring to Fig. 8, the step for analyzing the impulse response function includes a complex number transformation step S144a, a data mirroring step S144b, and an inverse Fourier transform step S144c.

In the complex number conversion step S144a, a transfer function divided into a magnitude r and a phase Φ is converted into a complex number type divided into a real part and an imaginary part. In the data mirroring step S144b, a symmetric signal is generated so that N / 2 signals are increased to N signals by two times. Then, the inverse Fourier transform step S144c can calculate the impulse response function through the inverse Fourier transform.

That is, from the frequency response function, the impulse response function can be obtained by going through the steps described above, and the total phase delay time can be known from this. Referring to FIG. 9, a frequency response function (see FIG. 9A) and an impulse response function (see FIG. 9B) obtained therefrom are illustrated.

In the contribution function analysis step S146, the signal analysis unit 23 calculates a contribution function indicating the similarity between the input signal and the output signal, and the analysis result is represented by the degree of contribution of the input to the output.

The calculation formula of the contribution function is as shown in Equation 7, and the result is illustrated in FIG.

는 주파수 영역에서의 입력 신호와 출력 신호의 기여도 함수이며,

는 주파수 영역에서의 입력 신호의 자기상관함수이고,

는 주파수 영역에서의 출력 신호의 자기상관함수이며,

는 주파수 영역에서의 입력 신호와 출력 신호의 상호상관함수이다. Equation (7) corresponds to Equation (2)

Is a function of the contribution of the input signal and the output signal in the frequency domain,

Is an autocorrelation function of the input signal in the frequency domain,

Is an autocorrelation function of the output signal in the frequency domain,

Is a cross-correlation function of the input signal and the output signal in the frequency domain.

Referring again to FIG. 2, in the screen configuration step S150, the screen configuration unit 24 analyzes the result of the transfer function analysis performed by the signal analysis unit 23, the result of the impulse response function analysis, Output signals in the frequency domain are displayed in real time on a screen provided in the real-time performance evaluating and improving device 20 of the speaker system in the form of a graph, thereby real-time analysis, i.e., data measurement and intuitive Allow interpretation.

Referring to FIG. 11, a real-time analysis result screen of a speaker system constituted by LabVIEW is illustrated. The user interface of the real time analysis result screen includes a time domain signal zone 1, a frequency domain signal zone 2, a frequency response function zone 3, a contribution function zone 4, an impact response function zone 5, 6).

In the time domain signal area 1, signals in the time domain of the input signal and the output signal are displayed. In the frequency domain signal domain 2, a conversion graph of the input signal and the output signal, Is displayed. In the frequency response function section 3, frequency response analysis results, that is, transfer function analysis results are displayed divided into magnitude and phase. In the contribution function section 4, the result of the contribution function analysis is displayed. In the response function area (5), the result of the impulse response function analysis is displayed.

The information displayed in each zone is displayed in a graph form, and the setting zone 6 is configured to receive a direct input from the user about the input signal path, signal type, frequency, size, sampling frequency, It has a graphical user interface configuration that allows intuitively changing settings for data measurement and analysis results.

Referring to FIG. 2 again, the filter designing and correcting unit 25 in the filter designing and correcting step (S160) grasps the frequency response characteristic from the transfer function analysis result transmitted from the signal analyzing unit 23, It is possible to newly design the filter of the speaker system by calculating the difference from the frequency response, or to calibrate the designed filter to output a sound suitable for the listening space in which the current speaker system is installed.

12, the filter design and correction method performed by the filter designing and correcting unit 25 includes a frequency response characteristic receiving step S210, a destination frequency response receiving step S220, a characteristic difference calculating step S230, A filter type selection step S245, a level difference calculation step S250, a correction coefficient setting step S255, a filter design / correction step S260, and a filter application step S270.

In the frequency response characteristic reception step S210, the filter design and correction unit 25 receives the analysis result on the frequency response characteristic, that is, the transfer function analysis result, from the signal analysis unit 23. The transfer function analysis results include information on size and phase.

In the destination frequency response reception step S220, the filter design and correction unit 25 reads the destination frequency response function stored in advance in the memory or receives the destination frequency response function received through communication with the external device. The objective frequency response function is a transfer function that indicates the relationship between the input signal and the output signal in a state in which the optimum sound to be heard by the listener is output to the speaker system.

Steps S210 and S220 are not limited to being performed in the order described in this specification, but may be performed in any order or simultaneously.

In the characteristic difference calculation step S230, the filter design and correction unit 25 compares the actual frequency response characteristic of the current speaker system received in the step S210 and the target frequency response characteristic received in the step S220 to calculate the characteristic difference. The design and / or correction of the filter is made based on the characteristic difference calculated at this stage.

Referring to FIG. 13, an actual frequency response function 310 and an objective frequency response function 300 for an arbitrary speaker system are illustrated. (F1, F3, F5) in the case of the actual frequency response function 310 as compared with the target frequency response function 300, F2, and F4) are lower than the target frequency response characteristics.

Therefore, in order to correct the characteristic of the actual frequency response function 310 to be the same as or similar to the characteristic of the target frequency response function 300, it is necessary to lower the level at the first, third and fifth points F1, F3 and F5, (F2, F4), it is necessary to raise the level.

In the frequency domain selection step S240, the filter designing and correcting unit 25 selects a point requiring correction of the comparison result between the actual frequency response characteristic and the target frequency response characteristic, that is, a frequency region requiring correction.

In the filter type selection step S245, the filter designing and correcting unit 25 determines the type of the filter that includes or does not include the frequency region corresponding to the frequency region selected in Step S240, for example, A band-pass filter (b), a high-pass filter (c), and a low-pass filter (d).

In the level difference calculation step S250, the filter design and correction unit 25 calculates a level difference occurring at an arbitrary point as a result of comparing the actual frequency response characteristic and the target frequency response characteristic. The point at which the level difference is calculated may be the frequency region selected in step S240.

In the correction coefficient setting step S255, the filter designing and correcting unit 25 corrects the calculated level difference so that the level of the actual frequency response characteristic at an arbitrary point is substantially equal to the level of the target frequency response characteristic. Set the correction factor. The filter for setting the correction coefficient may be a filter of the kind selected in step S245.

In the filter design / correction step (S260), the filter design and correction unit (25) designs a filter corresponding to the type of the filter selected in step S245. It is possible to design various filters having different characteristics even if the filters of the same kind are adjusted by adjusting the cutoff frequency at the time of designing the filter. In addition, in order to reflect the correction coefficient set in step S255 in the designing process of the filter or to adjust the cut-off characteristic of the designed filter, the correction coefficient set in step S255 may be reflected.

In the filter applying step (S270), the filter designing and correcting unit (25) applies the finally designed / corrected filter to the speaker system installed in the current listening space to be analyzed.

Then, steps S210 to S270 may be repeated until the actual frequency response characteristic is corrected to be substantially equal to the target frequency response characteristic.

As described above, according to the filter design and correction method according to the present embodiment, an efficient speaker system design is possible because filter design for analysis and tuning during measurement is performed automatically, rather than analysis and tuning after measurement.

2, in the sound source selection validity verification step S170, the sound source selection validity verification unit 26 compares the input signal used for performance evaluation and improvement of the speaker system according to the result of the contribution function analysis performed in step S146, that is, Verify that the sound source has been properly selected.

The result of the contribution function analysis is illustrated in FIG. 15, and when a uniform contribution value within a predetermined error is indicated over a predetermined frequency band (refer to FIG. 15A), an appropriate sound source It is determined that the selected sound source is selected, and if not (see (b) of FIG. 15), it can be determined that the sound source selection is wrong.

The predefined frequency band means a frequency band which is equal to or higher than a predetermined cutoff frequency or the following frequency band and includes an audible frequency band. If the frequency band has a uniform contribution value within a predetermined error, Is about 80% or more when compared with the maximum value.

This allows us to verify the validity of the input and output signals, so it can be validated when selecting the sources used to evaluate speaker performance.

For example, in order to reflect the influence of the actual listening space, white noise having the same size in the entire frequency band, pink noise having a strength in which the intensity of the component included in the unit frequency band is inversely proportional to the frequency, and the like When selected as this sound source, the appropriate sound source can be analyzed as selected.

Also, in this embodiment, even if the noise source is not the conventional noise source, the analysis result of the contribution function among the sound sources actually output through the speaker system has a uniform contribution over the entire frequency band as illustrated in FIG. 15 (a) The performance of the speaker system can be evaluated and improved by analyzing the relationship between the input signal and the output signal for the corresponding sound source.

It is apparent that the real-time performance evaluation and improvement method of the speaker system described above can be performed by an automated procedure in a time-series sequence by a software program built in the real-time performance evaluation and improvement apparatus of the speaker system. The codes and code segments that make up the program can be easily deduced by a computer programmer in the field. In addition, the program is stored in a computer readable medium, readable and executed by a computer, thereby implementing the method. The information storage medium includes a magnetic recording medium, an optical recording medium, and a carrier wave medium.

It will be apparent to those skilled in the art that various modifications and variations can be made in the present invention without departing from the spirit or scope of the invention as defined in the following claims And changes may be made without departing from the spirit and scope of the invention.

10: Speaker system
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16: Speaker section
20: Real-time performance evaluation and improvement of speaker system
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22: Signal measurement section
23: Signal Analysis Section
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25: Filter design and correction section
26: Sound source selection validation unit

Claims (9)

A method for evaluating and improving performance of a speaker system in real time in a listening space provided with a speaker system for outputting an input signal corresponding to a predetermined sound source file as an audible sound,
Measuring an output signal of the speaker system at a microphone section installed at one or more expected listening positions in the listening space;
Receiving the input signal from the speaker system at a signal measurement unit;
Analyzing and correcting a time delay of the output signal from a relationship between the input signal and the output signal in a signal analysis unit;
Performing a frequency domain analysis on the output signal and the input signal in which the time delay is corrected by the signal analyzing unit;
Displaying a result of the frequency domain analysis on a screen constituent unit; And
Designing and correcting a filter to be applied to the speaker system according to a result of the frequency domain analysis in a filter designing and correcting unit,
Wherein performing the frequency domain analysis comprises:
Performing a transfer function analysis, performing an impulse response function analysis, and performing a contribution function analysis.
The method according to claim 1,
The step of analyzing and correcting the time delay comprises:
Calculating a cross-correlation function between the input signal and the output signal, and determining a portion in which a maximum peak value appears as a delay time by applying a window function (Optimal Maximum Likelihood) .
delete The method according to claim 1,
Wherein performing the impulse response function analysis comprises:
A complex number conversion step of converting the transfer function into a complex number type divided into a real part and an imaginary part; a data mirroring step of generating a symmetric signal to increase the number of signals by two times; and performing an inverse Fourier transform And a method for evaluating and improving real-time performance of a speaker system.
The method according to claim 1,
Wherein the step of performing the contribution function analysis computes a contribution function indicative of a similarity between the input signal and the output signal and the analysis result is expressed as the degree of contribution of the input signal to the output signal, Real-time performance evaluation and improvement of system

here,

Is a function of the contribution of the input signal and the output signal in the frequency domain,

Is an autocorrelation function of the input signal in the frequency domain,

Is an autocorrelation function of the output signal in the frequency domain,

Is a cross-correlation function of the input signal and the output signal in the frequency domain.
The method according to claim 1,
The step of designing and calibrating the filter comprises:
Receiving the actual frequency response analysis result and the target frequency response characteristic of the speaker system from the filter design and correction unit;
Comparing the actual frequency response analysis result and the target frequency response characteristic to calculate a characteristic difference;
Selecting a frequency region in which the characteristic difference exists and selecting a filter type corresponding to the selected frequency region;
Calculating a level difference between the actual frequency response analysis result and the target frequency response characteristic in the selected frequency domain and setting a correction coefficient for compensating the level difference;
Designing the filter according to the selected filter type and the set correction coefficient; And
And applying the designed filter to the speaker system. ≪ Desc / Clms Page number 20 >
The method according to claim 1,
And a sound source selection feasibility verification step of determining that the selection of the sound source file is valid as the input signal when an even contribution value within a predetermined error is represented over the audio frequency band using the result of the contribution function analysis A method for evaluating and improving real - time performance of a speaker system.
An apparatus for evaluating and improving performance of a speaker system in real time in a listening space provided with a speaker system for outputting an input signal corresponding to a predetermined sound source file as an audible sound,
A microphone installed at one or more expected listening positions in the listening space to measure an output signal of the speaker system;
A signal measuring unit receiving the input signal from the speaker system and receiving the output signal from the microphone unit;
A signal analyzer for analyzing and correcting a time delay of the output signal from a relationship between the input signal and the output signal and performing a frequency domain analysis on the output signal and the input signal with the time delay corrected;
A screen configuration unit configured to display a result of the frequency domain analysis on a screen; And
And a filter designing and correcting unit for designing and correcting a filter to be applied to the speaker system according to a result of the frequency domain analysis,
Wherein the frequency domain analysis includes a transfer function analysis, an impulse response function analysis, and a contribution function analysis.
9. The method of claim 8,
And a sound source selection feasibility verifying unit that determines that the selection of the sound source file is valid as the input signal when the result of the contribution function analysis is used to indicate a uniform contribution value within a predetermined error over the audio frequency band A real - time performance evaluation and improvement device of a speaker system.

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