JP2778418B2 - Acoustic characteristic correction device - Google Patents

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an acoustic characteristic correcting apparatus for correcting a response characteristic (frequency response, etc.) of a reproduction system including a sound field such as a listening room to a desired characteristic. The calculation of the filter coefficient corresponding to the above can be arbitrarily selected from among a plurality of algorithms of the filter characteristics, so that an optimum correction characteristic can be given according to the purpose of use and the like.

[0002]

2. Description of the Related Art Conventionally, a graphic equalizer has been generally used as a device for correcting the response characteristics of the entire reproduction system including a room and a speaker. In this method, an audio frequency band is divided into several bands, and the gain is adjusted for each divided band. However, it was not possible to know how to adjust the reproduced sound to have a desired response characteristic.

To solve the drawbacks of the conventional graphic equalizer, the response characteristics of the entire reproduction system can be automatically set to desired characteristics.
There was one described in JP-A-59004. This means that the user sets the desired characteristics, reproduces the measurement signal such as white noise and impulse in the sound field to be reproduced using a reproduction system speaker, collects this with a microphone, and measures the response characteristics. Measure, determine the correction characteristics so that they match the desired characteristics, set the filter characteristics of the equalizer that matches the correction characteristics, and play the music signal through this equalizer to adjust the desired characteristics. It is intended to enable music playback.

[0004]

The calculation of a filter coefficient of an equalizer that matches the correction characteristic is performed by performing an inverse Fourier transform of the correction characteristic to obtain a corresponding impulse response. The impulse response obtained in this way is set as a filter coefficient in an equalizer (FIR filter), and a correction characteristic is given to the music signal.

By the way, there are various kinds of Fourier inverse transform algorithms. When they are used for calculating FIR filter coefficients for correcting the response characteristics of a music signal, there are advantages and disadvantages. It turns out that there are times when it can not be used.

SUMMARY OF THE INVENTION The present invention has been made in view of the above points, and has as its object to provide a music characteristic correction device capable of providing an optimum correction characteristic depending on a purpose of use and the like.

[0007]

According to the first aspect of the present invention,
At least a linear phase filter (Linear Phase Filter)
Filter) , minimum phase filter (Minimum)
Phase Filter) and the bandwidth of the playback system
The low band has the minimum phase characteristic and the high band has the linear phase characteristic.
The midrange is continuous or stepped from minimum phase to linear phase.
The filter coefficient is calculated using a filter characteristic selected from a plurality of filter characteristics including a composite phase filter characteristic that changes in a stepwise manner.

[0008]

[0009]

The linear phase filter and the minimum phase filter have the following advantages and disadvantages, respectively.

[0010] According to this, the linear phase filter has good transmission characteristics and the filter coefficient calculation is easy, but the delay is too large and the PA
It cannot be used when real-time performance is required, such as when mixing or mixing down (because the raw sound and the equalized sound are shifted in time). Further, the minimum phase filter is inferior to the linear phase filter in terms of transmission characteristics and ease of calculating filter coefficients, but has little delay and is suitable for a case where real-time properties are required .

Further, according to the double engagement phase filter, the following characteristics are obtained.

[0012] According to the composite phase filter , even when real-time properties are required, the transmission characteristics can be made closer to the linear phase while suppressing the delay amount to a practically acceptable level. did
Therefore, according to the present invention, an arc suitable for the purpose of use is provided.
By selecting the algorithm, the optimal
Characteristic correction can be performed.

[0013]

An embodiment of the present invention will be described below. FIG. 2 shows an outline of a hardware configuration of the entire apparatus. The acoustic characteristic correction device 10 includes a main body 12 and a remote controller 14, and both are connected by a detachable signal cable 16.

When measuring the response characteristics, the main body 12 performs generation of a measurement signal, calculation of frequency characteristics based on microphone pick-up signals, calculation of correction characteristics, calculation of FIR filter coefficients corresponding to the correction characteristics, and the like. When used as an equalizer after response characteristic measurement, the response characteristic is corrected by adding a set FIR filter characteristic to the audio signal to be reproduced. The remote controller 14 instructs the main body 12 on various operations at the time of measurement and setting of desired characteristics, and displays various response characteristics (measurement characteristics, desired characteristics, correction characteristics, etc.).

In the main body 12, a microphone input terminal 18
Has a measurement microphone connected when measuring the response characteristics,
A microphone pickup signal is input. Also, source input terminal 2
To 0, a source device such as a CD player is connected, and a source signal reproduced from the source device when used as an equalizer is input. The input unit 22 performs A / D conversion of a microphone input or a source input. The output unit 24 is a source signal and a measurement signal (test tone signal) that have been subjected to the equalizer processing.
Is output from the output terminal 26 after D / A conversion. The patch bay unit 28 changes connection of input / output and other various signals at the time of measurement and at the time of equalization. The waveform memory output unit 30 reads and outputs a measurement signal waveform (band signal waveform, TSP (Time Stretched Pulse) signal waveform) and a TSP inverse filter waveform stored in the ROM.

The input waveform memory unit 32 stores the A / D converted microphone input in the RAM. Convolution operator 34
Is a real-time convolution circuit (for example, LS manufactured by Yamaha Corporation)
IYM 7309 is connected in cascade to form a convolution unit having several thousands of stages. When the equalizer is used, an equalizer using an FIR filter is formed by transferring the filter coefficient of the equalizer. Further, at the time of measurement when the TSP signal is used as a measurement signal, a TSP inverse filter is configured by transferring the TSP inverse filter coefficient to the convolution operation unit 34. The data processing calculation and other control unit 36 is constituted by a CPU, and performs processing of measured data (calculation of measurement characteristics, desired characteristics, correction characteristics, calculation of an equalizer filter coefficient corresponding to the correction characteristics (Inverse Fourier transform), etc.) and the patch bay unit 28. Connection control, and other controls required by the main unit 12 and the CP of the remote control unit 14.
It exchanges signals with U42.

In the remote control unit 14, the operation unit 38 gives all necessary instructions to the main unit 12 at the time of measurement, setting of desired characteristics, and setting of correction characteristics. The display unit 40 displays various response characteristics and displays for operation. CPU4
2 exchanges data with the CPU 36 of the main body 12.

FIG. 3 shows an example of a panel configuration of the remote controller 14. The display unit 40 is configured by an LCD display or the like, and various response characteristics are displayed in a graph. That is, a common graph axis (horizontal axis is frequency, vertical axis is level) is displayed on the upper graph display section.
The measurement characteristic is displayed on a bar graph 44 and the desired characteristic (an example of a flat characteristic is shown) is displayed on a line graph 46 on the upper side.
Also, cursors 62 and 64 indicating the frequency range are displayed as vertical lines. In the lower part, a correction characteristic calculated as a difference between the desired characteristic and the measurement characteristic is displayed by a line graph 48. Further, between the upper and lower rows, the correction frequency range set by the operation of the operator is displayed as a horizontal bar graph 50. In this case, the correction characteristic display 48 is not displayed outside the correction frequency range (or is displayed flat on the 0 dB line). Display portions 52 and 54 for displaying current setting items, setting contents, and the like are provided at the upper and lower portions of the graph display portion to assist the operator's operation.

The operation unit 38 is provided with a cursor key 56, a shuttle key (rotary encoder) 58, various key switches 60, and the like. Cursor keys 56 include an up key 56a, a down key 56b, a left cursor selection key 5
6c, a right cursor selection key 56d. The left and right cursor selection keys 56c and 56d are used to select one of the left and right cursors 62 and 64 of the display unit 40, for example, when correcting desired characteristics or setting a correction frequency range. When the shuttle 58 is turned by pressing the left cursor selection key 56c, the left cursor 6 is turned in the turned direction.
2 moves to set the lower limit of the frequency range. When the shuttle 58 is turned by pressing the right cursor selection key 56d, the right cursor 64 moves in the turning direction, and the upper limit value of the frequency range is set. For example, a mark 65 is displayed at the selected position of the cursors 62 and 64, so that it is possible to know which is selected. The up and down keys 56a and 56b are used, for example, when correcting desired characteristics. When the up key 56a is pressed in a specified frequency range, the level of the desired characteristics is gradually and smoothly increased in a curve, and the down key 56b is pressed. When pressed, the level of the desired characteristic is smoothly and gradually reduced in a curve. The key switch 60 is used to select setting items, select measurement data,
Used for execution instructions and other various instructions.

FIG. 4 shows an outline of a procedure from the measurement of the frequency characteristic to the use as an equalizer using the acoustic characteristic correction apparatus 10 of FIG. Each process is sequentially advanced by the mode progress operation by the operator (for example, the process proceeds to the next process each time one key switch is pressed). The outline of each step will be described.

Test As shown in FIG. 5A, a microphone 72 is arranged at a listening position 71 in a room 70 where music is reproduced, and a measurement signal is output from the main unit 10 to be reproduced via a power amplifier 74. Are reproduced from the speakers 76 and 78 used for
And captures the collected sound waveform into a memory in the apparatus 10.
This measurement is performed at each position by moving the microphone 72 to a plurality of points (for example, five points) centering on the receiving position 71 as shown on the right of FIG.

Calculation of Measurement Characteristics The response characteristics are calculated based on the picked-up sound signals taken into the memory. The obtained response characteristics (measurement characteristics) are displayed in a bar graph on the display unit 40 of the remote control unit 14, for example, as shown in FIG.

Setting of Desired Characteristics Using the remote controller 14, the user operates the operation unit 38 while viewing the display unit 40.
Use to set the desired characteristics. The selected or set desired characteristic is displayed as a line graph 46 on the display unit 40 as shown in FIG. For example, FIG.
As shown in (b), when the measurement characteristic 44 is set to a characteristic that is flattened, since both characteristic displays are superimposed on the same graph axis, what kind of desired characteristic should be set. You can see at a glance if it is flat and it is easy to set up.

Calculation of Correction Characteristics When the desired characteristics are set, the correction characteristics are automatically calculated as the difference between the desired characteristics and the measured characteristics, and are displayed on the display unit 40 as shown in FIG.
It is displayed in a line graph 50 as shown in FIG. Even when the desired characteristics are being corrected, the correction characteristics are calculated and displayed as needed.

Correction of Correction Characteristic If the peak of the correction characteristic is large, a sense of incongruity may be generated in the sense of hearing. When the correction range is limited due to the limitation of the reproduction frequency characteristic of the speaker to be used, the correction frequency range is regulated as necessary (that is, the correction amount outside the correction frequency range is set to 0 dB).

Calculation of Equalizer Filter Coefficients Once the correction characteristics have been determined, they are inverse Fourier transformed to determine the corresponding impulse response. In this case, an arbitrary selection is made from linear phase processing Fourier inverse transform, minimum phase processing Fourier inverse transform, or Fourier inverse transform of another algorithm, depending on the use situation or the like. Other algorithms include, for example, the characteristic that the low band of the reproduction system band has the minimum phase characteristic, the high band has the linear phase characteristic, and the middle band changes continuously or stepwise from the minimum phase to the linear phase. Can be prepared. As a result, an impulse response as shown in FIGS. 6D and 6E, further another impulse response, and further another impulse response are obtained. Equalizer (FI
R) The filter coefficient is given as a level value at each position on the time axis of the impulse response. In this way, the equalizer characteristics over the entire frequency band are set.

Confirmation of Correction Effect Confirmation of the correction effect is performed as necessary. That is, the equalizer filter coefficient thus obtained is set in the convolution operation unit 34 to constitute an equalizer, and the equalizer is provided with a correction characteristic for the measurement signal, reproduced from the speaker, and the response characteristic is measured again. The measurement characteristic and the desired characteristic are superimposed and displayed on the section 40 to confirm the correction effect. The more the two characteristics match, the more the correction as desired is made. If the expected correction state cannot be obtained due to the limitation of the speaker characteristics or the like, the desired characteristics are re-corrected as necessary.

Music Playback Once the equalizer filter characteristics are finally determined, FIG.
As shown in (b), a source device 80 such as a CD player
To perform music playback as the final purpose by using the main unit 12 of the main unit 10 as an equalizer.

FIG. 1 shows a control block configuration in the acoustic characteristic correcting apparatus 10 for realizing each step of the above procedure.
FIG. 1 shows a connection state at the time of measurement. A microphone 72 for measurement is connected to the microphone input terminal 18 and the source input terminal 20.
Source devices 80 are respectively connected. The measurement signal input from the microphone input terminal 18 is amplified by the microphone amplifier 82. The switch 84 is used for measurement and calculation (the above-described?
) And during reproduction (the above-described step). The A / D converter 86 converts a microphone input or an analog source input into a digital signal. The switch 88
This is for passing the digital source input through the bypass path 90, and can be switched between digital source input reproduction and other modes (analog input reproduction and measurement). The switch 92 is switched between measurement and reproduction. The waveform memory 32 captures a microphone input during a test. The measurement signal generator 30 is configured by a ROM that stores the waveform of the measurement signal. In this embodiment, a band signal (to be described later) and a TSP (to be described later) TSP signal are stored as measurement signals, and one of them can be read out according to an operator's selection operation. Has been.

The switch 94 is switched between at the time of reproduction, at the time of calculating response characteristics, and at the time of testing. The switch 96 switches between a route that passes through the convolution operation unit 34 and a route 98 that bypasses the route.
When calculating the response characteristics in the SP method, when confirming the correction effect, and when playing music, a route that passes through the convolution calculator 34 is selected. The use of the convolution operation unit 34 is switched by switching the switch 102. That is, at the time of response characteristic calculation in the TSP method, the TSP inverse filter waveform read out from the TSP inverse filter waveform memory 100 is set as a filter coefficient, and the TSP inverse filter performs time compression of the collected TSP signal to obtain an impulse response. Ask. Also, when confirming the correction effect and playing back music, an equalizer filter coefficient corresponding to the correction characteristic obtained by the operation is set as a filter coefficient and operates as an equalizer. As a result, the convolution operation unit 34 is also used as an inverse filter in the TSP method at the time of response characteristic calculation and as an equalizer at the time of checking the correction effect and at the time of music reproduction, so that the hardware configuration is simplified. Even if they are used in this way, there is no problem since the response characteristic calculation, the correction effect confirmation, and the music reproduction are not performed simultaneously.

The output of the convolution unit 34 or the output of the bypass 98 is input to the switch 106 through the addition point 104. The switch 106 is switched between at the time of a test, at the time of checking a correction effect, at the time of music reproduction, and at the time of response characteristic calculation. During the test, when confirming the correction effect, and during music playback, the measurement signal or music signal passed through the switch 106 is applied to the D
The signal is converted into an analog signal by the / A converter 108 and the low-pass filter 110, output from the output terminal 26, and reproduced by the speakers 76 and 78 in the room 70 via the power amplifier 74.

The signal guided from the switch 106 to the line 112 at the time of calculating the response characteristic is distributed by the switch 114 according to the measurement method. That is, in the case of the TSP method,
After the Fourier transform of the impulse response signal into frequency information by the frequency conversion means 116, the band division means 11
At 8, the signal is divided into predetermined frequency bands (for example, every 1/3 octave band). Further, in the case of the band signal method, since the measurement data is originally obtained in a state of being divided into frequency bands (for example, every 1/3 octave band), the data is passed through the bypass 120 as it is. The signals of both paths pass through an addition point 122, and a band power average calculation circuit 124 calculates a power average for each divided band. The obtained band power data of all the frequency bands is stored in the band data memory 12.
6 is stored. The band data memory 126 can store multiple points of measurement data (for example, eight times). Each measurement data is displayed as a bar graph according to the display selection operation of the operator (measurement characteristic display 44 in FIG. 3).

The selection and weighting means 128 selectively outputs the data selected and instructed by the operator's selection operation from the multi-point multiple-time measurement data stored in the band data memory 126. Further, the measurement data is weighted according to the positions of the measurement points P1 to P5 with respect to the listening position 71 (FIG. 5A) as necessary. Collective averaging means 130
Calculates an aggregate average of a plurality of selected and weighted measurement data. The interpolating means 132 treats the value of each band subjected to collective averaging as a value at the center frequency of each band, interpolates between the center frequencies of each band, and connects all frequency bands with continuous and smooth curve data. Find characteristics. The interpolation data thus obtained is stored in the RAM 13
4 is stored as the final measured characteristic.

The ROM 136 stores an average characteristic and some other characteristics as desired characteristics, and the characteristic selected by the key switch 60 is read out. The selected desired characteristic is corrected to a desired characteristic by the arithmetic means 140 based on the operation of the cursor key 56, the shuttle key 58 and the like by the operator. The corrected desired characteristics are stored in the RAM 138 with the backup power supply, and can be read out and used at any time in the same manner as the characteristics of the ROM 136.

The calculating means 142 calculates a correction characteristic from the set desired characteristic and measured characteristic. The correction characteristics may be modified as needed based on the operation of the operator, such as upper and lower limit values of the correction level and the restriction of the correction frequency range. The equalizer filter coefficient calculating means 144 calculates an equalizer filter coefficient corresponding to the set correction characteristic. The calculated filter coefficients are set in the convolution calculator 34, and the equalizer characteristics at the time of music reproduction and at the time of checking the correction effect are set. The calculated filter coefficient is stored in the RAM 146 with a backup power supply, and can be read and used at any time. The filter coefficients are also stored in the RAM card 148 and can be shared by inserting the RAM card 148 into another music characteristic correction device.

The display control means 150 displays the calculated measurement characteristics, desired characteristics, correction characteristics and the like on the display unit 4 of the remote control unit 14.
Control for displaying at 0 is performed. The switching control of each switch in FIG. 1 and various operations other than the convolution operation unit 34 are executed by the CPU 36 (FIG. 2) of the main body 12.

Next, control of each step of the procedure of FIG. 4 by the control block of FIG. 1 described above will be described in detail. When response characteristics are measured in a test room, the characteristics vary considerably depending on the location. This is because reflected waves from ceilings, floors, walls, and the like in the room interfere with each other and disturb the frequency characteristics. In addition, this phenomenon is more remarkable even at a slight difference in location at a higher frequency with a shorter wavelength. Therefore, when the correction characteristic is obtained based on the data of one measurement point and the filter coefficient of the equalizer is obtained, the best result is obtained at that point, but the area including the surrounding area (the range in which the listener's head moves, etc.) In some cases, the best results cannot be obtained due to extreme peak dips.

Therefore, in this embodiment, FIG.
As shown on the right of FIG. 7, a measurement area 73 is set around the listening position 71 in the room 70, and a plurality of measurement points P1 to P5 including the listening position 71 are set in this area 73. The microphone 72 is moved to the points P1 to P5 to perform measurement, and a correction characteristic is obtained from a spatial average thereof. As a result, good correction characteristics can be obtained on average at any position in the area, and an effective area for correction can be enlarged.

In this embodiment, as a test method,
As described above, one of the band signal method and the TSP method can be selected according to the selection operation of the operator. The TSP method has the advantage that the measurement time is short and continuous measurement data can be obtained instead of discrete measurement data for each divided band. However, in this embodiment, as described above, since the convolution operation unit 34 for the equalizer is also used as the TSP inverse filter used in the TSP method, the length of the TSP signal for measurement is limited. There is a limit to the power of the entire TSP signal, and if it is used for measurement in a noisy environment, the S / N ratio of the measurement result may deteriorate.

Therefore, the band signal method is used in an environment with a lot of noise or when there is no restriction on the measurement time. There are many (speaker systems), and the band signal method requires time for measurement.)
, The TSP method is used, and both methods are used properly.

The test methods using the band signal method and the TSP method will be described. (A) Band signal method In the band signal method, band signals obtained by dividing a plurality of frequency bands are sequentially emitted at different times, and the response of each band is measured. Here, the bandwidth of each band uses a 1/3 octave band method (that is, a division method in which each band has a 1/3 octave band width) which is said to be relatively close to the audibility characteristics. In this case, it is possible to obtain continuous data having a high division ability if the division pitch is made fine, but it takes an enormous amount of time to generate a band signal of the entire band. Therefore, in this case, the division pitch is set to every 1/3 octave in FIG. 7A or every 1/6 octave in FIG. Continuous data is obtained by interpolation. If the division pitch is 1/3 octave pitch, the bandwidth will not overlap,
If the pitch is 1/6 octave, the bandwidth changes while overlapping by 1/3 octave. The overlap improves the connection between the bands in the measurement data.

FIG. 8 shows an example in the case of dividing into １ ／ octave pitches in １ ／ octave bands. (A) is the band signal waveform center frequency, (b) is the band signal waveform (when the center frequency is 100 Hz), and (c) is the band signal waveform output flow. FIG. 1 shows the band signal waveform of FIG.
Is stored in the measurement signal generator 30 (ROM) of
By changing the reading speed, a measurement signal for each band is generated. Speakers 7 sequentially at different times
The band signals emitted from 6, 78 are collected by microphone 72 for each band, and the collected sound waveform is stored in waveform memory 32 in FIG.

(B) TSP method In general, a single pulse is used to measure the impulse response of a hall or the like. However, since the signal power is small, even if a technique such as synchronous addition is used together, a sufficient SN ratio cannot be obtained. Often. On the other hand, when the TSP signal is used, the signal power is large and the S / N ratio is easily obtained. Also, in order to easily obtain an inverse filter and convert the response of the TSP signal into an impulse response, a convolution operation with the inverse filter may be performed. Therefore, when a convolution device can be used, the conversion is easy. Therefore, the TSP signal has convenient characteristics for measurement.

FIG. 9A shows a TSP signal used in the TSP method.
It has a waveform as shown in FIG. This TSP waveform is stored in the measurement signal generator 30 of FIG. 1, and is read once by one measurement and reproduced from the speakers 76 and 78.
The reproduced TSP signal is picked up by the microphone 72, and the collected sound waveform is stored in the waveform memory 32.

Calculation of Measurement Characteristics The calculation of the response characteristics based on the sound waveform stored in the waveform memory 30 is performed as follows according to the test method.

(A) Band signal method In the band signal method, the sound wave form for each divided band stored in the waveform memory 30 of FIG.
4, 96, bypass 98, addition point 104, switch 1
06, 114, the bypass path 120, and the addition point 122, the band power average calculator 124 calculates the average band power for each divided band, and stores the average in the band data memory 126. The band data memory 126 can store measurement data for a plurality of times.
The measurement data of the five points P1 to P5 shown on the right of (a) is stored. The selection weighting means 128 discriminates data by, for example, excluding data that is extremely different from the others by looking at individual measurement characteristics on the display unit 40. The remaining data is weighted as necessary. Specifically, when the measurement points are, for example, the five points P1 to P5 shown on the right in FIG. 5A, the point P1 at the center position (mainly the position where the head is located) is set to one.
The other points P2 to P5 are each set to 0.5, or the point P1 at the center position is set to 1 and the other points P2
2 to P5 are summed up to 1.

The selected and weighted measurement data is subjected to collective averaging by a collective averaging means 130. As a result, average measurement data of the region where the measurement is performed is obtained. Since the collectively averaged measurement data is discrete data for each divided band, it is interpolated by the interpolation means 132 to convert the data into continuous smooth curve data. As an interpolation method, a spline interpolation method capable of performing interpolation in a short time is suitable. In the interpolation, as shown in FIG. 10, data obtained as a power average for each divided band is treated as a value at the center frequency of each band, and a spline interpolation is performed between the points based on values of several points before and after. And for example 40
Interpolation data of 96 points is obtained and used as measurement characteristics.

As described above, the power average for each divided band is obtained, and the average is used as the value at the center frequency, and the spline interpolation is performed between the respective points, whereby useful and practical averaging of the obtained measurement characteristic results is achieved. As a result, large peak dips due to phase interference are prevented from occurring in the measurement characteristics, as in the past. Is prevented. The data of the measurement characteristics obtained in this manner is stored in the RAM 134 of FIG. 1 and is displayed in a bar graph on the display unit 40 (the measurement characteristic display 4 of FIG. 3).
4) is done.

(B) TSP Method In the TSP method, the sound waveform stored in the waveform memory 32 of FIG. 1 was immediately stored in the TSP inverse filter coefficient memory 100 by the convolution calculator 34 via the switches 94 and 96. The convolution operation (time compression) with the inverse TSP waveform (FIG. 9B) yields an impulse response (FIG. 9C). The inverse TSP waveform is a waveform obtained by temporally inverting the TSP waveform (FIG. 9A). When the number of stages of the convolution operation unit 34 is insufficient as a time compression filter for the TSP signal, time compression can be performed in a divided manner.

The impulse response output from the convolution calculator 34 is Fourier-transformed by the frequency conversion means 116 through the addition point 104 and the switches 106 and 114, and the frequency response characteristic (FIG. 9D) is obtained. The obtained frequency response characteristic is band-divided by the band dividing means 18 into a state similar to the band signal method (1/3 or 1/6 octave pitch with 1/3 octave band width). The band-divided measurement data undergoes the same processing as in the case of the band signal method. That is, the measurement data for each band divided by the band dividing means 118 passes through the addition point 122, the band power average for each divided band is calculated by the band power average calculating means 124, and is stored in the band data memory 126. You. The band data memory 126 stores measurement data for multiple points and multiple times. The selection weighting means 128 discriminates data by, for example, excluding data that is extremely different from the others by looking at individual measurement characteristics on the display unit 40. The remaining data is weighted as necessary. The selected and weighted measurement data is subjected to a collective average by the collective averaging means 130. As a result, average measurement data of the region where the measurement is performed is obtained. Since the collectively averaged measurement data is discrete data for each divided band, the data is spline-interpolated by the interpolation means 132 to convert the data into continuous smooth curve data. The interpolated measurement data is stored in the RAM 34 as measurement characteristics, and is displayed on the display unit 40 as a bar graph.

As described above, even in the TSP method, the measured data is once divided into bands, the power average is obtained for each band, and interpolation is performed to obtain continuous data. Since the occurrence of the peak dip is prevented, it is possible to prevent a sense of incongruity due to an extreme correction when the correction characteristic is obtained by using the measurement characteristic as it is and used for the characteristic correction.

FIG. 11 shows an example of an operation procedure in the calculation of the test and measurement characteristics described above. First, the microphone position is set (S1), and one of the band signal method and the TSP method is selected as a test method (S2).
Further, either a 1/3 octave band pitch or a 1/6 octave band pitch is selected as the band division pitch (S3). Thereafter, when the test start button is pressed (S4), the test sound is reproduced from the speakers 76 and 78, collected by the microphone 72, and stored in the waveform memory 32 (S5). The measurement result is immediately displayed on the display unit 40 as a bar graph (S6), and the operator can see and confirm the result. If the measurement result is considered abnormal (for example, a large noise is included), a retest is performed at that point (S7, S8). If the measurement result is good, move the microphone position to another point and repeat the test (S
9).

When the test is completed for all points (S10), the collected data is sequentially displayed on the display unit 40, and the data is selected as necessary (S11). The selected data is weighted for each measurement point automatically or manually as necessary (S1).
2). Then, a set average value and an interpolation value are automatically calculated for the data of each weighted point, and R
The measurement data is stored in the AM 134 as one final measurement characteristic data (S13), and the measurement is completed.

FIG. 12 shows an example of a desired characteristic setting flow. When the desired characteristic setting mode is selected and operated on the remote controller 14 (FIG. 3), a graph scale is displayed on the display 40 (S2).
2), the measurement characteristics stored in the RAM 134 are displayed as a bar graph 44 (S23). Next, when a desired characteristic is selected (S24), the corresponding desired characteristic is stored in the ROM.
136 or the RAM 138 and the display unit 4
0 is displayed as a line graph 46 (S25).

By the way, the transmission characteristics of a speaker in a listening room or a hall vary depending on the directivity of the speaker and the reverberation characteristics of the room, and desired characteristics in terms of audibility do not always coincide with flattening the measurement characteristics. do not do. Therefore, it would be convenient if the desired characteristics in the room could be easily set. For example, by preparing in advance a desired characteristic as a characteristic for PA (Public Address) in a hall and a desirable characteristic when listening with a small speaker in a home listening room in a large speaker system, it is easy to prepare in advance. Correction to that characteristic becomes possible.

Therefore, in the ROM 136, as a general pattern of the desired characteristic, for example, as shown in FIG. 13, in addition to the flat characteristic C1 over the entire band, a characteristic C2 in which the low and high regions of the flat characteristic are attenuated, It is convenient to prepare in advance the characteristic C3, the mid-sound emphasis characteristic C4, and the bass / treble emphasis characteristic C5. In this case, by displaying the characteristic pattern name on the display section 40, the operator refers to the character pattern, moves the cursor to a desired characteristic pattern, performs a selection operation, reads the corresponding characteristic data from the ROM 136, and reads the corresponding characteristic data. Can be used as In addition, characteristic data classified by various speakers (for a PA in a hall, for outdoor PA, small speakers for a studio monitor, etc.) and various rooms (Japanese listening room, Western listening room, etc.) are stored in the ROM 136, and the display unit 40 is provided. By displaying the speaker type name and room type name on the display, the operator can refer to this and move the cursor according to the speaker type or room to be used to select and operate the speaker type or room type. Characteristic data to be read out from the ROM 136 and used as desired characteristics. Once the desired characteristics are set,
The calculation of [measurement characteristic]-[desired characteristic] is automatically performed by the calculating means 142, and the correction characteristic is obtained.
Is displayed as a line graph 48 (S27). ROM
The characteristic data read from 136 can be used as desired characteristics as it is, but can also be used after being partially modified.

FIG. 14 shows a characteristic adjustment method in the case of a conventional graphic equalizer or parametric equalizer.
, The center frequency F, the gain G, and the sharpness Q
Was generally adjusted by changing the value of. In this case, as the order of adjustment, first determine the center frequency F,
Next, the value of Q is set, and finally the gain G is increased or decreased. However, the adjustment operation is not easy because it is necessary to adjust the three parameters independently and to match the target characteristics. Also, if Q is changed, the effect will be over the entire frequency range, so it is difficult to grasp how the characteristics actually change when Q is changed,
It was difficult to adjust.

Therefore, here, instead of determining the center frequency, a frequency range from everywhere is set,
By raising and lowering the characteristics within the specified range while maintaining the smooth connection of the characteristics at both ends, it is possible to easily set a characteristic curve of desired characteristics that is smooth and close to human sense. Step S28 of FIG. 12 showing this setting procedure
The following steps will be described.

When the desired characteristics are initially set, the display unit 40
One of the lower limit or the upper limit of the frequency range indicated by the upper cursors 62 and 64 is selected and can be corrected (the mark 65 is displayed on the selected one). When the shuttle key 58 is operated in this state (S28), the value selected from the upper limit value and the lower limit value of the frequency range changes in the direction in which the shuttle key 58 is turned (S29, S30, S31). Display unit 4
The cursor 62 or 64 with the @ mark 65 above 0 also moves in the same direction (S32).

When an operation to switch to the other cursor is performed by pressing the left or right cursor key 56c or 56d (S3
4), the switched value of the frequency range lower limit value or upper limit value can be corrected, and the position of the ▽ mark 65 on the display unit 40 also moves to the other cursor side. When the shuttle key 58 is operated in this state (S28), the corresponding value changes in the direction in which the shuttle key 58 is turned (S28).
29, S30, S31), the Δ mark 65 on the display unit 40 also moves in the same direction (S32).

When the up key 56a or the down key 56b is pressed after setting the frequency range in this way (S39), the level of the desired characteristic is set for the set frequency range as shown in FIG. In accordance with the number of times of pressing or the pressing time, while maintaining continuity outside the set frequency range, the center position of the frequency range is increased or decreased in a curve with a peak (S40,
S41), the display of the desired characteristic on the display unit 40 also changes accordingly. According to such a correction method, the operation is simple because only the designation of the frequency range and the designation of the level increase / decrease amount are required. In addition, since the influence of the increase or decrease of the level does not reach outside the designated frequency range, it is easy to grasp how the characteristic actually changes by the increase or decrease operation, and it is easy to correct the characteristic as desired. The calculation for correcting the desired characteristic is performed by the calculation means 10 in FIG.

As a specific algorithm of the correction processing in the arithmetic means 10, for example, it is examined what kind of correction curve should be increased or decreased in accordance with the frequency range so that the operational sensation and the change in the actual characteristic coincide with each other. A correction curve corresponding to the range can be set in a table in advance, and the correction curve corresponding to the set frequency range can be read from the table, and a gain corresponding to the increase / decrease instruction amount can be added and used. . By doing so, the sense of the level increase / decrease operation matches the actual change state of the characteristic, and the correction operation to the desired desired characteristic is facilitated.

When the up key 56a or the down key 56b is pressed in a state where the lower limit of the frequency range is set to the lowest frequency of the entire frequency band, the desired characteristic becomes one-sided in the low frequency side as shown in FIG. Or it changes to a one-sided state. Similarly, when the up key 56a or the down key 56b is pressed while the upper limit of the frequency range is set to the highest frequency in the entire frequency band, the desired characteristic becomes one-sided up or one-sided as shown in FIG. It changes to a down state. In these cases, for example, a one-sided or one-sided correction curve corresponding to the frequency range and the increase / decrease amount is set in a table in advance, and the correction curve corresponding to the set frequency range is read out from the table and the increase / decrease instruction is issued. A gain corresponding to the amount (the number of times the up / down keys 56a and 56b are pressed) can be applied.

After correcting the desired characteristics as described above,
By depressing the key switch 60 (S42), the process exits the characteristic setting routine, and at this time, the characteristic determination and the setting are completed (S43). By instructing storage of the determined characteristics as necessary, the characteristics can be stored in the designated area of the RAM 138 with the backup power supply as correction desired characteristic information, and can be read and used at any time. Therefore, it is not necessary to adjust again each time the desired characteristic is switched.

Another modification method for modifying the desired characteristics will be described. When the desired characteristic is set, if the correction characteristic is obtained and equalized as it is, the user may feel that the correction is excessive. Therefore, as shown in FIG. 16, the intermediate characteristic between the initially set desired characteristic (or the desired characteristic modified as shown in FIG. 15) and the measured characteristic is calculated by the calculating means 140.
, And this can be newly set as a desired correction characteristic and used. Specifically, for example, the difference between the desired characteristic and the measurement characteristic set at the initial frequency at each frequency (that is, the correction value of each frequency) is divided into 20 equal parts. Calculate and display a characteristic change that gradually brings the characteristic closer to the measured characteristic, or conversely, gradually returns to the original desired characteristic, and sets it as the new desired characteristic when it becomes the desired characteristic I do. FIG. 17 shows the calculation process at this time. First, the difference between the measured characteristic Nb and the initially set desired characteristic Db is determined (S51), and the difference Eb
Is multiplied by [the number of times the up key 56a or the down key 56b is pressed] ÷ 20 to determine the correction amount ΔEb of the desired characteristic (S52), and the correction amount ΔEb is added to the desired characteristic Db to obtain Db + ΔEb (S53). This is used as a new desired characteristic (S54). By doing so, an intermediate correction value that is not corrected as much as the initial desired characteristic Db can be set in a well-balanced manner in all frequency bands by a simple operation. The intermediate characteristics thus created can also be stored in the RAM 136.

Calculation of Correction Characteristics The correction characteristics are calculated by setting desired characteristics.
At 42, the difference from the measurement characteristic is automatically calculated and displayed on the display unit 40.

Correction of Correction Characteristics If, for example, a desired characteristic of 0 dB flat is set for the measurement characteristics shown in FIG.
A large peak dip occurs as shown in FIG. This peak dip is often caused by a slight change in the measurement environment. If equalization is performed using such a correction characteristic as it is, a greatly corrected portion (a portion indicated by a circle in (b)) ), A slight change in the environment (for example, a slight shift in the frequency characteristic due to the influence of the temperature and humidity of the air) makes the correction no longer a true correction, and conversely, a correction that cannot normally occur. As shown in (c), the correction error becomes large, resulting in a rather habitous characteristic. Therefore, the upper and lower limits of the level of the correction characteristic are set to arbitrary values (for example, ± 10 dB) by the operation of the operator.
Set to. Thereby, the correction characteristic calculating means 14 of FIG.
2 prevents the correction error from increasing by restricting the upper and lower limit values of the correction characteristic to the set value as shown in FIG. In addition, since the maximum value on the + side of the correction characteristic is limited, the maximum input can be suppressed, and distortion of the entire system such as a power amplifier and a speaker can be suppressed.

When the correction range is limited due to the limitation of the reproduction frequency characteristic of the speaker to be used, if the speaker is driven based on the calculated correction characteristic, the speaker may be overloaded. The frequency range is set by the user's operation, and the correction characteristics are used only within the range, and the correction is not performed by making the outside of the range 0 dB flat. The frequency range to be corrected is displayed on the display unit 40 of FIG. 3 as a corrected frequency range display 50 in a horizontal bar graph.

FIG. 19 shows an example of a specific calculation process at each stage from when the measurement data is obtained to when the final correction characteristic is determined as described above. The data to be used for calculating the measurement characteristics is selected out of a plurality of measurement data stored in the band data memory 126 in FIG. 1 (S61) and weighted. The singled out the data and _(i) M _b. Here, i is a measurement number, and i = 1 to N. b is a band number obtained by dividing the entire frequency band, and b = 1
To B. B is 31 or 61 in this embodiment.

When a plurality of data are selected, the set averaging means 130 calculates a set average for each band. (S62). And as the average value of all the bands of this collective average (S63). Furthermore, as normalized average measurement data (S64), and this is displayed on the display unit 40 as a measurement characteristic. This measurement characteristic Nb is subjected to spline interpolation to obtain continuous data. By normalization, the average value of the measurement characteristics Nb is adjusted to be always 0 dB, and even if the sound pickup level is small, the display of the measurement characteristics on the display unit 40 always comes to substantially the same level. It becomes easy to compare with the characteristic display.

[0071] If desired characteristic Db is set by operation of the operator _{(S65), E b = N} b -D b is obtained as the correction characteristic in the calculation unit 142 (S66). The measurement characteristic Nb here is data after spline interpolation. Then, as an average value of all bands of this correction characteristic, (S67). Further, the calculating means 142 calculates the normalized correction characteristic as (S68). By normalization, the average value of the correction characteristic Fb is adjusted so as to be always 0 dB, so that the sound before and after the correction as a whole only changes the sound quality but does not change the sound volume.

For the obtained correction characteristic Fb, a process for regulating the upper limit value and the lower limit value of the level shown in FIG. 18D is performed (S69). Steps S66 to S6
Step 9 is performed only within the specified frequency range of FIG. Outside the designated frequency range, a process for flattening the correction characteristic to 0 dB is separately performed (S70). The correction characteristics finally determined in this way go to a routine for calculating a convolutional (equalizer) filter coefficient (S71).

Calculation of Equalizer Filter Coefficients The algorithm of the FIR filter for acoustic characteristic correction includes:
As described above, each has advantages and disadvantages, and may not be usable depending on the purpose of use. Therefore, here, as a FIR filter, a linear phase filter, a minimum phase filter,
One of the composite phase filters can be selected according to the selection operation of the operator. The impulse responses of the linear phase filter and the minimum phase filter are, for example, as shown in FIG.
This is as shown in (d) and (e).

As described above, the linear phase filter has good transmission characteristics and is easy to calculate the filter coefficient. However, when the delay is too large (see FIG. 6D), when the real-time property such as PA or mixdown is required. Cannot be used (because the raw sound and the equalized sound will be offset in time). Further, the minimum phase filter is inferior to the linear phase filter in terms of transmission characteristics and ease of calculating filter coefficients, but has little delay (see FIG. 6E), and is suitable for a case where real-time properties are required. In addition, the composite phase filter can obtain a characteristic in which the transmission characteristic is close to the linear phase and the amount of delay is suppressed to a practically acceptable level even when real-time properties are required. Therefore, the operator can select one of the algorithms according to the purpose of use, so that one device can be used in various situations.

In any case, since the FIR filter using the digital convolution operation is used for the correction characteristics, a linear phase filter, a minimum phase filter, or other special characteristics can be provided only by switching the algorithm. It is very easy to change the specifications according to the purpose, and if necessary, the calculation accuracy can be set arbitrarily if the calculation accuracy is arbitrarily increased. The effect is great.

An example of a method of calculating the impulse response of the linear phase filter and the impulse response of the minimum phase filter using the inverse Fourier transform or the like from the correction characteristics in the equalizer filter coefficient calculating means 144 will be described.

(A) Calculation of Impulse Response of Linear Phase Filter It is assumed that the phase = linear phase = 0 at all frequencies. Specifically, the following procedure is used. i) Once the correction characteristic is divided into bands (for example, 1/3 to 1
/ 12 octave pitch), and obtain the average power of each band. ii) Using the obtained power average value as the value at the center frequency of each band, the data is interpolated into 4096 points of data that can be Fourier-transformed by spline interpolation or the like. iii) The data obtained in ii) is taken as the real part (corresponding to the amplitude term), and the imaginary part (corresponding to the phase term) is subjected to the inverse Fourier transform on the complex format data in which all zeros are set to zero. iv) The real part of the complex data obtained as a result becomes a linear phase impulse response as it is.
This is set as a coefficient of the filter (convolution calculator 34).

(B) Calculation of impulse response of minimum phase filter A phase characteristic corresponding to the frequency characteristic is obtained by the Hilbert transform. Specifically, the following procedure is used. i) Once the correction characteristic is divided into bands (for example, 1/3 to 1
/ 12 octave pitch), and obtain the average power of each band. ii) Using the obtained power average value as the value at the center frequency of each band, the data is interpolated into 4096 points of data that can be Fourier-transformed by spline interpolation or the like. iii) Perform the Hilbert transform on the complex form data in which the data obtained in ii) is the real part and the imaginary part is all 0, and calculate the complex form data that matches the correction characteristic curve and becomes the minimum phase transition system. I do. This complex data has a necessary phase component added to the imaginary part. iv) Inverse Fourier transform of the complex format data obtained in iii). v) Since the real part of the resulting complex data has the minimum phase impulse response, these are set as the coefficients of the FIR filter (convolution unit 34).

(C) Calculation of Impulse Response of Composite Phase Filter The composite phase filter is set to have the following characteristics. (A) The high frequency range (f2 or more) is linearized within a predetermined delay range. (B) The low band (f1 or less) has the minimum phase characteristic. (C) The intermediate range (f1 to f2) is shifted from the minimum phase to the linear phase.

Such characteristics are specifically determined by the following procedure. i) Once the correction characteristic is divided into bands (for example, 1/3 to 1
/ 12 octave pitch), and obtain the average power of each band. ii) Using the obtained power average value as the value at the center frequency of each band, the data is interpolated into 4096 points of data that can be Fourier-transformed by spline interpolation or the like. iii) Perform the Hilbert transform on the complex form data in which the data obtained in ii) is the real part and the imaginary part is all 0, and calculate the complex form data that matches the correction characteristic curve and becomes the minimum phase transition system. I do. This complex data has a necessary phase component added to the imaginary part. iv) For only the phase component obtained in iii), the frequency below the lower switching frequency f1 is I'll do it. The phase is set to 0 at the switching frequency f2 or higher. v)) Inverse Fourier transform of the complex format data obtained in iii). vi) Since the real part of the resulting complex data has the minimum phase impulse response, these are set as coefficients of the FIR filter (convolution operation unit 34). The values of the switching frequencies f1 and f2 are set to, for example, the following values according to the allowable delay amount.

When the delay is 10 msec: 1 kHz or more linear phase, 20
0 Hz or less Minimum phase delay of 5 msec: 2 kHz or more linear phase, 40
Here, the actual calculation result of the impulse response of the linear phase filter, the minimum phase filter, and the composite phase filter by the above method will be described. FIG. 20 shows the measurement characteristics (dotted line).
And a characteristic (solid line) obtained by spline-interpolating it.
FIG. 21 shows the characteristic (dotted line) and the desired characteristic (solid line) obtained by spline interpolation in FIG. FIG. 22 shows the correction characteristics obtained as the difference between the two characteristics in FIG.

FIG. 23 shows a calculation result when the correction characteristic of FIG. 22 is obtained by a linear phase filter. FIG. 24 shows the calculation result when the minimum phase filter is used. Also,
FIG. 25 (in the case of a delay of 5 msec) and FIG. 26 (in the case of a delay of 10 msec) obtained by using the composite phase filter
Shown in Note that a linear phase filter, a minimum phase filter,
In addition to the composite phase filter, another filter having another characteristic can be prepared, and an arbitrary filter can be selected from the filter.

[0083] FIG. 27 shows a result of performing the confirmation of the correction effect coefficients of the FIR filter 34 on the confirmation above procedure sets calculated by any algorithm of correction characteristics. (A) shows each measurement point P1.
It is an initial (that is, without equalizing) measurement result at P5 (see FIG. 5). (B) Each point P1
It is a desired characteristic which is arbitrarily set by measuring characteristics and operator who ensemble average measured data ~P5 in Installing the same weight. (C) is a correction characteristic obtained as a difference from the desired characteristic of (b). FI calculated based on this correction characteristic
The R filter coefficient is set in the convolution calculator 34 to form an equalizer, and the measurement signal (band signal or TSP
Signal) is reproduced through this equalizer and the measurement is performed again. The results measured at each measurement point P1~P5 shown in FIG. 2 7 (d). Compared with the previous correction, according to (a), which have been made corresponding characteristic correction even in the measurement point, it was confirmed that the area containing these points are optimal correction was made.

Music Reproduction By inputting a music source instead of the measurement signal and passing it through an equalizer (convolution unit 34), it is reproduced.
Music appreciation can be enjoyed with the desired reproduction characteristics.

[0085]

[Modification] In the above-described embodiment, in the calculation of the measurement characteristic, in both the band signal method and the TSP method, for example, spline interpolation or the like is performed on the average value obtained for each divided band to obtain the measurement characteristic. In addition to the calculation, the band is divided again when calculating the FIR filter coefficient for realizing the correction characteristic. However, the present invention is not limited to this.

That is, when the obtained measurement characteristics are displayed with high precision or when the measurement characteristics are to be used separately, it is difficult to use the divided band data as it is. By performing the spline interpolation when calculating the correction characteristic, the interpolation processing for calculating the measurement characteristic before that can be omitted or simplified.
For example, correction information is calculated as a correction value for each frequency-divided band based on a measurement characteristic calculated for each frequency-divided band and a desired characteristic set for each frequency-divided band, and is calculated for each band. The correction characteristic can also be obtained by calculating the correction value as a value at the substantially center frequency of each band and calculating the value between them by interpolation. By doing so, it is possible to prevent a large peak dip from occurring in the correction characteristic, and it is possible to prevent a sense of incongruity due to extreme correction, and to calculate the correction characteristic as it is based on the measurement characteristic interpolated by 4096 points. This is effective because the amount of calculation can be significantly reduced as compared with the case and the like, and the accuracy characteristic finally obtained does not cause much significant deterioration in accuracy.

[0087]

As described above, according to the present invention, a filter coefficient can be obtained by selecting an algorithm of a filter characteristic suitable for a purpose of use and the like, so that optimum characteristic correction can be performed in various situations. Can be performed.

[Brief description of the drawings]

FIG. 1 is a block diagram showing an embodiment of the present invention, showing a control configuration of an acoustic characteristic correction apparatus 10 of FIG.

FIG. 2 is an acoustic characteristic correction device 10 to which the present invention is applied.
FIG. 2 is a block diagram showing a hardware configuration in the apparatus.

FIG. 3 is an external view showing a panel configuration of a remote controller 14 of FIG. 2;

4 is a flowchart showing an outline of a procedure from characteristic measurement by the music characteristic correction device of FIG. 1 to use as an equalizer.

5 is a diagram showing a connection state of devices and a microphone arrangement when measuring acoustic characteristics using the acoustic characteristic correction device of FIG. 2, and a connection state of the devices when the device is used for music reproduction as an equalizer.

FIG. 6 is a diagram showing various characteristics in the process of FIG.

FIG. 7 is a diagram showing a division state of a band at the time of measurement.

FIG. 8 is a diagram showing a band signal used in the band signal method.

FIG. 9 is a diagram showing an outline of the TSP method.

FIG. 10 is a diagram illustrating a method of obtaining continuous measurement characteristics by interpolating between bands based on data for each divided band.

FIG. 11 is a flowchart illustrating an example of an operation procedure in calculation of test and measurement characteristics.

FIG. 12 is a flowchart showing a procedure for setting a desired characteristic.

FIG. 13 is a diagram showing various patterns of desired characteristics prepared in a ROM.

FIG. 14 is a diagram showing a characteristic adjustment method in a conventional graphic equalizer or parametric equalizer.

FIG. 15 is a diagram showing a method for correcting a desired characteristic.

FIG. 16 is a diagram showing another method for correcting a desired characteristic.

FIG. 17 is a flowchart showing a calculation process for realizing the method of FIG. 16;

FIG. 18 is a diagram illustrating a method of correcting a correction characteristic.

FIG. 19 is a flowchart illustrating an example of an arithmetic process at each stage from when measurement data is obtained to when a correction characteristic is determined.

FIG. 20 shows a measurement characteristic (dotted line) and a characteristic obtained by spline-interpolating the measurement characteristic (solid line).

FIG. 21 is a characteristic (dotted line) obtained by performing spline interpolation in FIG. 20;
And the desired characteristics (solid line).

FIG. 22 shows a correction characteristic obtained as a difference between the two characteristics in FIG. 21.

FIG. 23 is a calculation result when the correction characteristics of FIG. 22 are obtained by a linear phase filter.

FIG. 24 is a calculation result when the correction characteristics in FIG. 22 are obtained by a minimum phase filter.

FIG. 25 is a calculation result (in the case of a delay of 5 msec) when the correction characteristics of FIG. 22 are obtained by a composite phase filter.

26 is a calculation result (in the case of a delay of 10 msec) when the correction characteristics in FIG. 22 are obtained by the composite phase filter.

FIG. 27 is a diagram showing measured values of a measurement characteristic, a correction characteristic, and a correction effect.

[Explanation of symbols]

Reference Signs List 10 acoustic characteristic correction device 34, 144 convolution operation unit, equalizer filter coefficient operation means (correction characteristic provision means) 38 operation unit (desired characteristic setting means) 70, 76, 78 room, speaker (reproduction system including sound field) 134 RAM (measurement characteristic output means) 142 correction characteristic calculation means 144 equalizer filter coefficient calculation means (filter coefficient calculation means)

 ──────────────────────────────────────────────────続 き Continuation of the front page (56) References JP-A-63-234699 (JP, A) JP-A-63-42599 (JP, A) JP-A-63-243635 (JP, A) JP-A-63-234635 125098 (JP, A) JP-A-2-122798 (JP, A) JP-A-5-88677 (JP, A) JP-A-63-187417 (JP, U) JP-B-61-59004 (JP, B1)

Claims (1)

(57) [Claims] A desired characteristic setting means for setting a desired response characteristic of a reproduction system including a sound field based on an operation of an operator, and information on a measurement characteristic of a response characteristic of the reproduction system including a sound field. Output means for outputting a measured characteristic; correction characteristic calculating means for calculating a correction characteristic for realizing the desired characteristic based on the desired characteristic and the measured characteristic; and the calculated characteristic for an audio signal to be reproduced. an acoustic characteristic correction device comprising; and a correction characteristic imparting means for imparting the correction characteristic, said correction characteristic imparting means, said even without least as a filter coefficient of the convolution operation corresponding to the correction characteristic of the reproduction system all A linear phase filter whose band has linear phase characteristics, a minimum phase filter whose entire band of the reproduction system has no delay characteristics, and a reproduction system
The low frequency band has the minimum phase characteristic, and the high frequency band is
Phase characteristic, the middle range is continuous from minimum phase to linear phase
Phase filter with characteristic changing stepwise or stepwise
A filter coefficient calculating means for calculating a filter coefficient corresponding to the filter characteristic selected from among a plurality of filter characteristics including characteristics, using a filter coefficient of the selected filter characteristic with respect to the sound field signal to be reproduced An acoustic characteristic correction apparatus comprising: a convolution operation unit that performs a convolution operation to provide a correction characteristic corresponding to the filter characteristic.