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JPH0738987

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DESCRIPTION JPH0738987
[0001]
BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an
acoustic characteristic correction apparatus for correcting response characteristics such as a
frequency response of a reproduction system including a sound field such as a listening room to
a desired characteristic. To match the desired characteristics.
[0002]
2. Description of the Related Art Conventionally, graphic equalizers have been generally used as
a device for correcting the response characteristics of the entire reproduction system such as a
room or a speaker. This is to divide the voice frequency band into several bands and adjust the
gain for each divided band. However, with this, it was not possible to know how to adjust the
playback sound to have the desired response characteristics.
[0003]
In order to solve the drawbacks of the conventional graphic equalizer and to automatically set
the response characteristic of the whole reproduction system to the desired characteristic, for
example, there is the one described in Japanese Patent Publication No. 61-59004. This allows the
user to set desired characteristics and reproduces measurement signals such as white noise and
impulses in the sound field to be reproduced by the speaker of the reproduction system,
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1
collecting this with a microphone, and responding to the response characteristics Measured and
determined correction characteristics so as to match desired characteristics, set filter
characteristics of the equalizer matching the correction characteristics, and reproduced sound
field signals through the equalizer to adjust to desired characteristics To make music playback
enjoyable.
[0004]
The response characteristic (transmission frequency characteristic) measured using a
measurement signal is a characteristic of time-averaged acoustic power that is a combination of
direct sound, reflected sound, and reverberant sound. Here, the frequency characteristics of the
reverberation time are generally longer as the bass and shorter as the treble, so even if the
frequency characteristic of the direct sound itself is flat, the power of the direct sound, the
reflected sound, and the reverberation is added The response characteristic measured as a value
is a rise in bass. For this reason, when the correction characteristic is calculated based on this
measurement characteristic, the bass becomes too low. However, since the frequency
characteristic in the sense of hearing is dominated by the direct sound, if correction is performed
using the correction characteristic thus obtained, the sound tends to have a lower bass than the
desired characteristic in the sense of hearing. The The present invention has been made in view
of the above-described point, and an object of the present invention is to provide an acoustic
characteristic correction device so that the aural characteristic matches the set desired
characteristic.
[0005]
SUMMARY OF THE INVENTION According to the present invention, the correction characteristic
is compensated according to the reverberation time frequency characteristic or the average
sound absorption coefficient frequency characteristic of the sound field to be reproduced.
[0006]
According to the present invention, the compensation characteristic is compensated according to
the reverberation time frequency characteristic or the average sound absorption coefficient
frequency characteristic of the sound field to be reproduced, so that the auditory characteristic
matches the desired characteristic. Can.
[0007]
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2
An embodiment of the present invention will be described below.
FIG. 2 shows an outline of the hardware configuration of the entire apparatus.
The acoustic characteristic measurement apparatus 10 is composed of a main unit 12 and a
remote control unit 14, and the two companies are connected by a detachable signal cable 16.
[0008]
When measuring the response characteristic, the main unit 12 measures the generation of the
measurement signal, calculates the frequency characteristic based on the microphone pickup
signal, calculates the correction characteristic, calculates the FIR filter coefficient corresponding
to the correction characteristic, etc. When the latter is used as an equalizer, the response
characteristic is corrected by applying a set FIR filter characteristic to the acoustic signal to be
reproduced. The remote control unit 14 instructs the main body unit 12 of various operations at
the time of measurement and desired characteristics setting and displays various response
characteristics (measurement characteristics, desired characteristics, correction characteristics,
etc.).
[0009]
In the main body 12, the measurement microphone is connected to the microphone input
terminal 18 at the time of measurement of the response characteristic, and a microphone sound
collection signal is input. Further, a source device such as a CD player is connected to the source
input terminal 20, and a source signal reproduced from the source device at the time of use as an
equalizer is input. The input unit 22 performs A / D conversion of microphone input and source
input. The output unit 24 D / A converts the equalized source signal and the measurement signal
(test tone signal) and outputs the converted signal from the output terminal 26. The patch bay
unit 28 changes the connections of input and output and various other signals between
measurement and equalizer. The waveform memory output unit 30 reads out and outputs a
measurement signal waveform (band signal waveform, TSP (Time Stretched Pulse) signal
waveform) and TSP inverse filter waveform stored in the ROM.
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[0010]
The input waveform memory unit 32 stores the A / D converted microphone input in the RAM.
The convolution operation unit 34 is composed of a real time convolution circuit (for example, a
circuit in which several stages of cascaded LSIYM 7309s connected by Yamaha Corporation are
connected to form a convolution unit of several stages). By transferring, an equalizer with an FIR
filter is configured. Further, at the time of measurement in the case of using the TSP signal as a
measurement signal, the TSP inverse filter coefficient is transferred to the convolution unit 34 to
configure the TSP inverse filter. Data processing calculation Other control unit 36 is constituted
by a CPU, processing of measurement data (measurement characteristic, desired characteristic,
calculation of correction characteristic, calculation of equalizer filter coefficient corresponding to
correction characteristic (Fourier inverse transform), etc.), patch bay unit 28 And other necessary
control of the main unit 12 and exchange of signals with the CPU 42 of the remote control unit
14.
[0011]
In the remote control unit 14, the operation unit 38 issues all instructions necessary for the main
unit 12 at the time of measurement, desired characteristics setting, and correction characteristics
setting. The display unit 40 displays various response characteristics and displays for operation.
The CPU 42 exchanges data with the CPU 36 of the main unit 42.
[0012]
An example of the panel configuration of the remote control unit 14 is shown in FIG. The display
unit 40 is configured of an LCD display or the like, and various response characteristics are
graphically displayed. That is, in the upper graph display section, the measurement characteristic
is a bar graph 44 on the common graph axis (horizontal axis is frequency, vertical axis is level),
and an example of the desired characteristic (flat characteristic is shown). ) Are displayed
superimposed on the line graph 46. In addition, cursors 62 and 64 for indicating the frequency
range are displayed by vertical lines. In the lower part, a correction graph calculated as a
difference between the desired characteristic and the measurement characteristic is displayed as
a line graph 48. In addition, the correction frequency range set by the operation of the operator
is displayed as a horizontal bar graph 50 between the upper and lower stages. In this case, the
correction characteristic display 48 is not displayed outside the correction frequency range (or
displayed flat on the 0 dB line). In the upper and lower portions of the graph display unit, display
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portions 52 and 54 for displaying current setting items, setting contents and the like are
provided to assist the operator's operation.
[0013]
In the operation unit 38, a cursor key 56, a shuttle key (rotary encoder) 58, various key switches
60, and the like are provided. The cursor key 56 is composed of an up key 56a, a down key 56b,
a left cursor selection key 56c, and a right cursor selection code 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 a desired characteristic or setting a correction
frequency range. When the left cursor selection key 56c is pressed and the shuttle 58 is turned,
the left cursor 62 is moved in the turning direction to set the lower limit value of the frequency
range. When the right cursor selection key 56d is pressed and the shuttle 58 is turned, 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 position of the selected one of the cursors 62 and
64, which makes it possible to know which one is selected. The up and down keys 56a and 56b
are used, for example, to correct the desired characteristic. When the up key 56a is pressed for
the designated frequency range, the level of the desired characteristic is gradually raised in a
curve, and the down key 56b is used. When pressed, the level of the desired characteristic is
gradually lowered in a curve. The key switch 60 is used for selection of setting items, selection of
measurement data, execution instructions, and various other instructions.
[0014]
An outline of a procedure from measurement of frequency characteristics to use as an equalizer
using the acoustic characteristic correction device 10 of FIG. 2 is shown in FIG. Each process is
sequentially advanced by mode progress operation by the operator (for example, every time one
key switch is pressed, it progresses to the next process). Each process will be briefly described.
[0015]
Test As shown in FIG. 5 (a), a microphone 72 is disposed at a listening position 71 in a room 70
where music is reproduced, and a speaker used for reproduction via a power-up 74 by outputting
a measurement signal from the unit 10 The sound is reproduced from 76 and 78, collected by
the microphone 72, and the sound-wave form is taken into the memory in the machine 10. This
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measurement is performed at each position by moving the microphone 72 to a plurality of points
(for example, five points) centered on the receiving position 71 as shown on the right of FIG. 5A,
as necessary.
[0016]
The response characteristic is calculated based on the collected sound signal captured in the
calculation memory of the measurement characteristic. The obtained response characteristic
(measurement characteristic) is displayed on the display unit 40 of the remote control unit 14 as
a bar graph as shown in FIG. 6A, for example.
[0017]
Setting of Desired Characteristics The remote controller 14 operates the operation unit 38 while
looking at the display unit 40 to set desired characteristics. The desired characteristic selected or
set is displayed on the same graph axis as the display 44 of the measurement characteristic on
the display unit 40 as shown in FIG. As a desired characteristic, for example, as shown to this FIG.
6 (a), when setting it as the characteristic which smoothed and made the measurement
characteristic 44 flat, since both characteristic display is superimposed and displayed on the
same graph axis, You can see at a glance what kind of desired characteristics you want to be flat
and it is easy to set.
[0018]
Calculation desired characteristics of the correction characteristics are set, the correction
characteristics are automatically calculated as the difference between the desired characteristics
and the measurement characteristics, and displayed on the display unit 40 as a line graph 50 as
shown in FIG. Even when the desired characteristic is corrected, the correction characteristic is
calculated at any time and displayed. The compensation of the correction characteristic based on
the reverberation time frequency characteristic or the average sound absorption coefficient
frequency characteristic is also performed here.
[0019]
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If the peak of the correction characteristic of the correction characteristic is large, the sense of
incongruity may occur, so the upper and lower limit values of the level of the correction
characteristic are regulated as necessary. Further, when the correction range is limited due to the
limitation of the reproduction frequency characteristics of the speaker to be used, the correction
frequency range is restricted as needed (that is, the correction amount outside the correction
frequency range is made 0 dB).
[0020]
When the calculation correction characteristic of the equalizer filter coefficient is determined, it
is subjected to inverse Fourier transform to obtain a corresponding impulse response. In this
case, it is arbitrarily selected and used from linear phase processing inverse Fourier transform,
minimum phase processing inverse Fourier transform, or Fourier inverse transform of another
algorithm according to the use condition etc. As a result, an impulse response as shown in FIG. 6
(d) or (e) is obtained. An equalizer (FIR) filter coefficient is given as a level value at each position
on the time axis of this impulse response. In this way, the equalizer characteristics over the entire
frequency band are set.
[0021]
Confirmation of correction effect Confirm the correction effect as necessary. This sets the
obtained equalizer filter coefficient in the convolution unit 34 to form an equalizer, applies
correction characteristics to the measurement signal with this equalizer, reproduces from the
speaker, measures the response characteristics again, and displays The measured characteristic
and the desired characteristic are displayed superimposed on the part 40 to confirm the
correction effect. As the two characteristics match, the correction according to the desired
characteristics is performed. If the expected correction state can not be obtained due to the
limitations of the speaker characteristics, etc., the desired characteristics are re-corrected as
necessary.
[0022]
When the music reproduction equalizer filter characteristics are finally determined, as shown in
FIG. 5B, the source device 80 such as a CD player is connected and the main unit 12 of the main
unit 10 is used as an equalizer to achieve the final purpose. Play music.
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[0023]
FIG. 1 shows a control block configuration in the acoustic characteristic correction device 10 for
realizing each step of the above procedure.
FIG. 1 shows the connection state at the time of measurement. The measurement microphone 72
and the source device 80 are connected to the microphone input terminal 18 and the source
input terminal 20, respectively. The measurement signal input from the microphone input
terminal 18 is amplified by the microphone amplifier 82. The switch 84 is switched between
measurement and calculation (the above steps) and regeneration (the above steps). A / D
converter 86 converts the microphone input or analog source input into a digital signal. The
switch 88 is for passing the digital source input to the bypass 90, and is switched between the
digital source input reproduction and the other modes (analog input reproduction,
measurement). The switch 92 is switched between measurement and reproduction. The
waveform memory 32 takes in the microphone input at the time of test. The measurement signal
generator 30 is configured of a ROM that stores the waveform of the measurement signal. In this
embodiment, a band signal of band signal method (to be described later) and a TSP signal of TSP
method (to be described later) are stored as measurement signals, and any one of them can be
read out according to the selection operation of the operator. It has been
[0024]
The switch 94 is switched at the time of reproduction, at the time of response characteristic
calculation, and at the time of test. The switch 96 switches between the route through the
convolution unit 34 and the route 98 for bypassing the same. In the test and in the response
characteristic calculation in the band signal method, the bypass path 98 is selected, and the
response characteristic calculation in the TSP method is corrected At the time of confirmation of
the effect and at the time of music reproduction, a route passing through the convolution unit 34
is selected. The application of the convolution operator 34 can be 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, time-compacts the collected TSP signal as a TSP inverse filter, and impulse
response Ask. In addition, at the time of confirmation of the correction effect and at the time of
music reproduction, an equalizer filter coefficient corresponding to the correction characteristic
obtained by the calculation is set as a filter coefficient to operate as an equalizer. As a result,
since the convolution operation unit 34 is used both as an inverse filter in the TSP method at the
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time of response characteristic operation and at the time of correction effect confirmation and at
the time of music reproduction, the hardware configuration is simplified. There is no problem at
all in this way because the response characteristic calculation, the correction effect confirmation
and the music reproduction are not performed simultaneously.
[0025]
The output of the convolution unit 34 or the output through the bypass 98 is input to the switch
106 through the summing point 104. The switch 106 is switched between a test, a correction
effect confirmation, a music reproduction and a response characteristic calculation. At the time of
test, at the time of correction effect confirmation, at the time of music reproduction, the
measurement signal or music signal passed through the switch 106 is converted into an analog
signal by the D / A converter 108 and the low pass filter 110 and output from the output
terminal 26 It is reproduced by the speakers 76 and 78 in the room 70 through the power
amplifier 74.
[0026]
The signal led from the switch 106 to the line 112 at the time of response characteristic
calculation is distributed by the switch 114 according to the measurement method. That is, in the
case of the TSP method, after the impulse response signal is Fourier-transformed by the
frequency conversion means 116 and converted into frequency information, the band division
means 118 divides it 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 obtained in the
state of being originally divided into frequency bands (for example, every 1/3 octave band), it is
passed through the bypass path 120 as it is. The signals of both paths pass through a summing
point 122 and a band power averaging circuit 124 obtains a power average for each divided
band. The band power data of the entire frequency band determined is stored in the band data
memory 126. The band data memory 126 can store measurement data of a plurality of times (for
example, eight times). The measurement data of each time is displayed as a bar graph according
to the display selection operation of the operator (measurement characteristic display 44 of FIG.
3).
[0027]
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The selection / weighting means 128 selects and outputs a plurality of measurement data stored
in the band data memory 126 which are selected and instructed by the operator's selection
operation. The measurement data is weighted according to the positions of the measurement
points P1 to P5 (FIG. 5A) with respect to the listening position 71 as required. The collective
averaging means 130 calculates a collective average of the plurality of selected and weighted
measurement data. Interpolation means 132 treats the values of each band averaged in a group
as the value at the center frequency of each band, interpolates between the center frequencies of
each band, and connects all frequency bands with continuous smooth curve data Find the
characteristics. The interpolation data thus obtained is stored in the RAM 134 as a final
measurement characteristic.
[0028]
The ROM 136 stores average characteristics and other characteristics as desired characteristics,
and the one selected by the key switch 60 is read out. The selected desired characteristic is
corrected to the desired characteristic by the computing 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 backup power-supply RAM 138, and can be read out and used as
needed as the characteristics of the ROM 136.
[0029]
The calculating means 142 calculates the correction characteristic from the set desired
characteristic and the measured characteristic. As the correction characteristics, corrections such
as upper / lower limit regulation of the correction level and regulation of the correction
frequency range are added based on the operation of the operator as necessary.
[0030]
The reverberation time or average sound absorption coefficient frequency characteristic output
means 143 outputs information on the reverberation time frequency characteristic or the
average sound absorption coefficient frequency characteristic of the room 70 where the
reproduction is performed. The reverberation time or average sound absorption coefficient
frequency characteristic output unit 143 can be configured as a reverberation time frequency
characteristic measurement device or an average sound absorption coefficient frequency
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characteristic measurement device incorporated in the acoustic characteristic correction device
10. Or means for inputting information on reverberation time frequency characteristics or
average sound absorption coefficient frequency characteristics measured by an external
reverberation time frequency characteristic measurement device or average sound absorption
coefficient frequency characteristics measurement device (for example, these external
measurement devices) And an input terminal for inputting a measurement result, a RAM card
storing the measurement result, and the like.
[0031]
The characteristic compensation means 145 is a correction characteristic based on the
information of the reverberation time frequency characteristic or the average sound absorption
coefficient frequency characteristic so as to relatively raise the level of the correction
characteristic for a long reverberation time frequency or a low average sound absorption
coefficient frequency. On the other hand, the compensation characteristic is given so that the
auditory characteristic after correction agrees with the set desired characteristic.
[0032]
The equalizer filter coefficient calculation means 144 calculates an equalizer filter coefficient
corresponding to the set correction characteristic.
The calculated filter coefficient is set in the convolution unit 34, and the equalizer characteristic
at the time of correction effect confirmation is set at the time of music reproduction. Also, the
calculated filter coefficient is stored in the backup power supply RAM 146 and can be read out
and used as needed. It is also stored in the RAM card 148, and this filter coefficient can be
shared by inserting the RAM card 148 into another music characteristic correction device.
[0033]
The display control means 150 performs control to display the calculated measurement
characteristics, desired characteristics, correction characteristics and the like on the display unit
40 of the remote control unit 14. The CPU 36 (FIG. 2) of the main unit 12 executes various
operations other than the switching control of each switch in FIG. 1 and the convolution
operation unit 34.
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[0034]
Next, control of each process of the procedure of FIG. 4 by the control block of FIG. 1 described
above will be described in detail. When measuring the response characteristics indoors, the
characteristics differ considerably depending on the location. This is because the reflected waves
from the ceiling, floor, wall and the like in the room interfere with each other to disturb the
frequency characteristics. In addition, this phenomenon is more pronounced at short highfrequency short wavelength differences. Therefore, if the correction coefficient is determined
based on the data of one measurement point to obtain the filter coefficient of the equalizer, the
best result is given at that point, but the area including the periphery (the range where the head
of the listener moves etc. In some cases, extreme peak dips may occur and the best result may
not be obtained.
[0035]
Therefore, in this embodiment, as shown on the right of FIG. 5A, a measurement area 73
centered on the listening position 71 in the room 70 is set, and the listening position 71 is
included in the area 73. A plurality of measurement points P1 to P5 are set, the microphone 72
is moved to each of the points P1 to P5 to perform measurement, and a correction characteristic
is obtained from the spatial average of them. As a result, good correction characteristics can be
obtained on average at any position within the area, and the effective area of correction can be
expanded.
[0036]
Further, in this embodiment, as the test method, either one of the band signal method and the
TSP method can be selected according to the selection operation of the operator as described
above. The TSP method has an advantage that measurement time can be 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 operator 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. The power of the entire TSP signal is limited, and when used
for measurement in a noisy environment, the SN ratio of the measurement result may be
degraded.
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[0037]
Therefore, band signal method is used when there is no restriction in noise environment or
measurement time, and when noise environment is small or measurement time is limited (for
example, reproduction system (speaker system in hall etc. If there are many cases, and if it takes
a lot of time to measure with band signal method, etc.), use TSP method and use both methods
properly.
[0038]
The band signal method and the test method using the TSP method will be respectively
described.
(A) Band signal method The band signal method is to measure the response of each band by
sequentially emitting a band signal having a plurality of frequency bands divided at different
times. Here, as the bandwidth of each band, a 1/3 octave band method (that is, a division method
in which each band has a 1/3 octave bandwidth) which is said to be relatively close to the
auditory characteristic is used. In this case, although it is possible to obtain continuous data with
high division ability if the division pitch is finely taken, it takes a huge amount of time to emit
band signals of the entire band. Therefore, here, the division pitch is set by the operator's
selection operation to either every 1/3 octave in FIG. 7A or every 1/6 octave in FIG. 7B, and the
measurement data is measured. Interpolation is performed to obtain continuous data. If the
division pitch is 1/3 octave pitch, the bandwidth does not overlap, and if it is 1/6 octave pitch,
the bandwidth shifts while overlapping by 1/3 octave. If overlapping is made, the connection
between the bands in the measurement data becomes good.
[0039]
FIG. 8 shows an example in the case of dividing into 1/3 octave pitch by 1/3 octave band. (A) is a
band signal waveform center frequency, (b) is a band signal waveform (when the center
frequency is 100 Hz), and (c) is an output flow of the band signal waveform. The band signal
waveform of (b) is stored in the measurement signal generator 30 (ROM) of FIG. 1, and the
measurement signal of each band is generated by changing the reading speed. Band signals
sequentially emitted from the speakers 76 and 78 at different times are collected by the
microphone 72 for each band, and the sound wave form is stored in the waveform memory 32 of
FIG.
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[0040]
(B) TSP method In general, a single pulse is used to measure an impulse response such as a hole,
but since the signal power is small, a sufficient signal-to-noise ratio often can not be obtained
even by using methods such as synchronous addition. . On the other hand, when the TSP signal is
used, the signal power is large and the SN ratio can be easily obtained. In addition, since an
inverse filter can be easily obtained and the response of the TSP signal can be converted into an
impulse response, the convolution operation with the inverse filter may be performed. Therefore,
the TSP signal has convenient characteristics for measurement.
[0041]
The TSP signal used in the TSP method has a waveform as shown in FIG. 9 (a). The TSP waveform
is stored in the measurement signal generator 30 shown in FIG. 1, read out once from one
measurement, and reproduced from the speakers 76 and 78. The reproduced TSP signal is
collected by the microphone 72, and the sound wave form is stored in the waveform memory 32.
[0042]
The calculation of the response characteristic based on the sound wave form stored in the
arithmetic waveform memory 30 of the measurement characteristic is performed as follows
according to the test method.
[0043]
(A) Band signal method In the band signal method, the sound absorption type for each divided
band stored in the waveform memory 30 of FIG. 1 is instantly switched to switches 94 and 96,
bypass 98, summing point 104, switches 106 and 114, After passing through the bypass path
120 and the addition point 122, the band power average calculation unit 124 calculates the band
power average for each divided band and stores it in the band data memory 126.
The band data memory 126 can store measurement data for a plurality of times, and stores, for
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example, measurement data of five points P1 to P5 shown on the right of FIG. 5A. In the selection
weighting means 128, the operator looks at individual measurement characteristics on the
display unit 40, and discards data, for example, by excluding data extremely different from the
others. Also, the remaining data is weighted as necessary. Specifically, in the case where the
measurement points are, for example, 5 points P1 to P5 shown on the right in FIG. 5A, weighting
is performed using the point P1 at the center position (position where the head is mainly) as 1
Each of P2 to P5 is set to 0.5, and the point P1 at the center position is set to 1, and the other
points P2 to P5 are totaled to 1 and so on.
[0044]
The selected and weighted measurement data is subjected to collective averaging in collective
averaging means 130. This gives an average measurement data of the area where the
measurement was made. Since the aggregate averaged measurement data is discrete data for
each divided band, it is interpolated by the interpolation means 132 to be converted into
continuous smooth curve data. As the interpolation method, a spline interpolation method
capable of interpolation in a short time is suitable. Interpolation treats data obtained as power
average for each divided band as a value at the center frequency of each band as shown in FIG.
10, and spline interpolates between each point based on the values of several points before and
after Then, for example, interpolation data of 4096 points is obtained and used as a
measurement characteristic.
[0045]
In this way, by obtaining the power average for each divided band and using it as the value at the
center frequency, spline interpolation is performed between each point to achieve useful and
practical averaging of the obtained measurement characteristic results. As described above, it is
possible to prevent the occurrence of a large peak and dip due to phase interference in the
measurement characteristics, so that the sense of incongruity due to extreme correction is
prevented when the correction characteristics are obtained using the measurement
characteristics as they are. Ru. The data of the measurement characteristic thus obtained is
stored in the RAM 134 of FIG. 1 and displayed on the display unit 40 as a bar graph
(measurement characteristic display 44 of FIG. 3).
[0046]
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(B) TSP method The sound wave form stored in the waveform memory 32 of FIG. 1 in the TSP
method is the reverse TSP waveform stored in the TSP inverse filter coefficient memory 100 by
the convolution operator 34 via the switches 94 and 96 immediately. A convolution operation
(time compression) is performed with (FIG. 9 (b)) to obtain an impulse response (FIG. 9 (c)). The
reverse 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 of the TSP signal, time compression can be performed by division.
[0047]
The impulse response output from the convolution unit 34 passes through the summing point
104 and the switches 106 and 114, and is Fourier-transformed by the frequency converter 116
to obtain the frequency response characteristic (FIG. 9 (d)). The determined frequency response
characteristic is band-divided by the band division means 18 into the same state (1/3 octave
bandwidth, 1/3 or 1/6 octave pitch) as the band signal method. The band-divided measurement
data is subjected to the same processing as in the case of the band signal method. That is, the
measurement data of each band divided by the band dividing means 118 passes through the
addition point 122, the band power average of each divided band is calculated by the band
power average calculating means 124, and stored in the band data memory 126. Ru. The band
data memory 126 stores measurement data of multiple points and a plurality of times. In the
selection weighting means 128, the operator looks at individual measurement characteristics on
the display unit 40, and discards data, for example, by excluding data extremely different from
the others. Also, the remaining data is weighted as necessary. The selected and weighted
measurement data is subjected to collective averaging in collective averaging means 130. This
gives an average measurement data of the area where the measurement was made. Since the
aggregate averaged measurement data is discrete data for each divided band, this is splineinterpolated by the interpolation means 132 to be converted into continuous smooth curve data.
The interpolated measurement data is stored in the RAM 34 as a measurement characteristic and
displayed on the display unit 40 as a bar graph.
[0048]
As described above, in the TSP method as well, measurement data is divided once into bands,
power averaging is performed for each band, and interpolation is performed to obtain continuous
data. As a result, it is possible to prevent a sense of incongruity due to extreme correction when
the correction characteristic is obtained using the measurement characteristic as it is and used
for the characteristic correction.
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[0049]
An example of the operation procedure in the calculation of the test and the measurement
characteristic described above is shown in FIG.
First, the microphone position is set (S1), and either the band signal method or the TSP method is
selected as a test method (S2). Further, either 1/3 octave band pitch or 1/6 octave band pitch is
selected as the band division pitch (S3). Thereafter, when the test start button is pressed (S4), a
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 results are immediately displayed on
the display unit 40 as a bar graph (S6), and the operator can see and confirm this. If the
measurement result seems abnormal (for example, large noise etc.), re-testing is performed at
that point (S7, S8). If the measurement result is good, the microphone position is moved to
another point and the test is repeated (S9).
[0050]
When the test is completed for all points (S10), collected data are sequentially displayed on the
display unit 40, and data selection is performed as needed (S11). The selected data is weighted
for each measurement point by automatic or manual setting as required (S12). Then, the
aggregate average value and further the interpolation value are automatically calculated for the
data of each point that has been weighted, and stored as one final measurement characteristic
data in the RAM 134 (S13), and the measurement is ended.
[0051]
Setting of Desired Characteristics One example of a setting flow of desired characteristics is
shown in FIG. When the desired characteristic setting mode is selected and operated by the
remote control unit 14 (FIG. 3), a graph scale is displayed on the display unit 40 (S22), and the
measurement characteristic stored in the RAM 134 is displayed by the bar graph 44 (S23). Next,
when a desired characteristic selection operation is performed (S24), the corresponding desired
characteristic is read from the ROM 136 or the RAM 138 and displayed on the display unit 40 by
the line graph 46 (S25).
08-05-2019
17
[0052]
By the way, the transmission characteristic of the speaker in the listening room or the hall
changes depending on the directivity of the speaker and the reverberation characteristic of the
room, and the desired characteristic of the sense of hearing does not always coincide with the
flattening of the measurement characteristic. Therefore, it is convenient if the desired
characteristics in the room can be easily set. For example, in a large speaker system, it is easily
possible by preparing in advance a desired characteristic desired as a characteristic for PA
(Public Address) in a hall or a desired characteristic when listening with a small speaker in a
home listening room. Correction to that characteristic is possible.
[0053]
Therefore, the ROM 136 has, as a general pattern of desired characteristics, for example, a flat
characteristic C1 over the entire band as shown in FIG. It is convenient to prepare in advance the
medium sound emphasis characteristic C4, the low / high sound emphasis characteristic C5 and
the like. In this case, by displaying the characteristic pattern name on the display unit 40, the
operator refers to it and moves the cursor to the desired characteristic pattern to perform
selection operation, and reads out the corresponding characteristic data from the ROM 136 to
obtain the desired characteristic. It can be used as In addition, the characteristic data classified in
various speakers (in-hall PA, outdoor PA, small speaker for studio monitor, etc.) and various
rooms (Japanese-room listening room, Western-room listening room, etc.) are stored in the ROM
136. By displaying the speaker type name and the room type name on the screen, the operator
refers to this and moves the cursor according to the speaker type and room to be used to select
and operate the speaker type and the room type. Characteristic data can be read from the ROM
136 and used as a desired characteristic. When the desired characteristic is set, the calculation
means 142 automatically calculates [measurement characteristic] − [desired characteristic] to
obtain the correction characteristic, which is displayed on the display unit 40 by the line graph
48 (see FIG. S27). The characteristic data read out from the ROM 136 can be used as it is as a
desired characteristic, but can also be partially corrected and used.
[0054]
As a characteristic adjustment method in the case of the conventional graphic equalizer and
parametric equalizer, as shown in FIG. 14, it is common to adjust by changing the values of the
08-05-2019
18
center frequency F, the gain G and the sharpness Q. In this case, as the order of adjustment, the
center frequency F is determined first, then the value of Q is set, and finally the gain G is
increased or decreased. The adjustment operation was not easy because it had to be adjusted to
the characteristics. In addition, when Q is changed, the influence extends to the entire frequency
range, so it was difficult to grasp how the characteristics actually change when Q was changed,
and it was difficult to adjust.
[0055]
Therefore, here, instead of determining the center frequency, the frequency range is set from
where to where, and by keeping the smooth connection of the characteristics at both ends up and
down the characteristics within the specified range, it is smooth and human sense It is possible to
easily set the characteristic curve of the desired characteristic close to. The steps following step
S28 in FIG. 12 showing this setting procedure will be described.
[0056]
Initially, one of the frequency range lower limit value or the upper limit value designated by the
cursors 62 and 64 on the display unit 40 is selected and can be corrected (the selected one is
selected). ▽ mark 65 is displayed. When the shuttle key 58 is operated in this state (S28), the
selected one of the frequency range upper limit value and the lower limit value changes in the
direction in which the shuttle key 58 is turned (S29, S30, S31), The cursor 62 or 64 having the
▽ mark 65 on the display unit 40 is also moved in the same direction (S32).
[0057]
When the left or right cursor key 56c or 56d is pressed to switch to the other cursor (S34), the
switched value of the lower limit value or the upper limit value of the frequency range can be
corrected, and the position of the ▽ mark 65 on the display 40 Also move to the other cursor
side. In this state, when the shuttle key 58 is operated (S28), the corresponding value is changed
in the direction of turning the shuttle key 58 (S29, S30, S31), and the マ ー ク mark 65 on the
display 40 is also in the same direction accordingly. Move to (S32).
[0058]
08-05-2019
19
Thus, when the frequency range is set and then the up key 56a or the down key 56b is pressed
(S39), as shown in FIG. 15A, the level of the desired characteristic is pressed for the set
frequency range. Depending on the number of times or pressing time, while maintaining
continuity with the outside of the set frequency range, the central position of the frequency
range is peaked and increased or decreased in a curve (S40, S41), the display unit 40 The display
of the desired characteristics in the will also change accordingly. According to such a correction
method, the operation is easy since it is only necessary to specify the frequency range and the
amount of increase or decrease of the level. In addition, since the influence of level increase /
decrease does not reach outside the designated frequency range, it is easy to grasp how the
characteristic actually changes by the increase / decrease operation, and it is easy to correct it to
the desired characteristic. The calculation for correcting the desired characteristic is performed
by the calculation means 10 of FIG.
[0059]
As a specific correction processing algorithm in the calculation means 10, for example, it is
examined according to the frequency range whether or not the correction of the operation sense
and the actual characteristic match if the correction curve is increased or decreased according to
the frequency range. The correction curve may be set in advance in a table, and the
corresponding correction curve may be read out from the table according to the set frequency
range, and a gain may be added and used according to the increase / decrease indication amount.
By doing so, the sense of the level increase / decrease operation matches the actual change of the
characteristic, and the correction operation to the desired desired characteristic becomes easy.
[0060]
If the up key 56a or the down key 56b is pressed with the frequency range lower limit value set
to the lowest frequency of the entire frequency band, the desired characteristics are as shown in
FIG. 15 (b). Change to the state of Similarly, if the up key 56a or the down key 56b is pressed
with the frequency range upper limit value set to the highest frequency of all frequency bands,
the desired characteristics are as shown in FIG. 15 (c). It changes to the state of going down. Also
in these cases, for example, a correction curve of one side up or one side down corresponding to
the frequency range and the increase / decrease amount is set in advance in the table, and the
corresponding correction curve is read out from the table according to the set frequency range A
gain corresponding to the amount (the number of times the up and down keys 56a and 56b are
08-05-2019
20
pressed) can be given and used.
[0061]
When the desired characteristic is corrected as described above, the key setting button 60 is
pressed (S42) to exit the characteristic setting routine, and at this time, the characteristic
determination and setting are completed (S43). Note that the determined characteristic can be
stored and instructed as needed to be stored in the designated area of the backup power-supply
RAM 138 as correction desired characteristic information and read out and used at any time.
Therefore, it is not necessary to adjust again each time the desired characteristic is switched.
[0062]
Describes another modification method of modifying the desired characteristics. When the
desired characteristics are set, it may be felt that the correction is excessive if the correction
characteristics are obtained as it is and equalized. Therefore, as shown in FIG. 16, the
intermediate characteristic between the initially set desired characteristic (the desired
characteristic corrected as shown in FIG. 15 may be used) and the measurement characteristic is
automatically calculated by the operation means 140. Can be newly set and used as a correction
desired characteristic. Specifically, for example, the difference (that is, the correction value of
each frequency) at each frequency between the initially set desired characteristic and the
measured characteristic is equally divided into 20, and in accordance with this step, each time
the up key 56a or the down key 56b is pressed The characteristic change is calculated and
displayed so as to gradually bring the characteristic closer to the measurement characteristic or
vice versa, and when the desired characteristic is obtained, this is set as a new desired
characteristic. 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 correct the desired characteristic ΔEb Is obtained (S52), this correction
amount .DELTA.Eb is added to the desired characteristic Db to obtain Db + .DELTA.Eb (S53), and
this is used as a new desired characteristic (S54). By doing this, it is possible to set an
intermediate correction value that is not corrected as in the initial desired characteristic Db in a
well-balanced manner by a simple operation in all frequency bands. Intermediate characteristics
created in this manner can also be stored in the RAM 136.
[0063]
08-05-2019
21
Calculation Characteristic of Correction Characteristic The correction characteristic of the
correction characteristic is automatically calculated as the difference from the measurement
characteristic by the calculation means 142 by setting the desired characteristic, and displayed
on the display unit 40.
[0064]
The compensation of the correction characteristic by the characteristic compensation means 145
is performed as follows.
Assuming that the power P of a certain frequency is the sum of the power Pd of the direct sound
and the power Pr of the reflected sound, the power Pr of the reflected sound is expressed by Pr =
(1-α) Pd using the average sound absorption coefficient α. Therefore, P = Pd + PrP = Pd + (1 +
α) Pd∴P = (2-α) Pd Also, the reverberation time RT60 and the average sound absorption
coefficient α are as follows: RT60 is V / Sα (V is the volume of a room, S is a room There is a
proportional relationship to the surface area of Therefore, if the total power P of a certain
frequency and the reverberation time RT60 or the average sound absorption coefficient α are
known, the power Pd of the direct sound can be calculated.
[0065]
Therefore, the characteristic compensation means 145 uses, for example, the measurement
characteristic (total power P) stored in the RAM 134 and the reverberation time frequency
characteristic or the average sound absorption coefficient frequency characteristic outputted
from the reverberation time or average sound absorption coefficient frequency characteristic
output unit 143 The power Pd of the direct sound at each frequency is calculated to compensate
the correction characteristics. That is, since the frequency characteristic of the power Pd of the
direct sound is close to the characteristic on the audibility, the correction characteristic is
determined so that the frequency characteristic of the power Pd of the direct sound matches the
desired characteristic. Specifically, the compensation characteristic is obtained from [frequency
characteristic of full power]-[frequency characteristic of direct sound power] = [compensation
characteristic], and this compensation characteristic is given to any of the measurement
characteristic, the desired characteristic and the correction characteristic. As a result, the
compensation characteristic is given to the correction characteristic as a result. That is, in the
case where a process of giving a compensation characteristic to the measurement characteristic
is performed, {[measurement characteristic]-[compensation characteristic]}-[desired
08-05-2019
22
characteristic] = [compensated correction characteristic]. In addition, in the case of performing
processing for imparting a compensation characteristic to the desired characteristic,
[measurement characteristic] − {[desired characteristic] + [compensation characteristic]} =
[compensated correction characteristic]. In addition, in the case where the compensation
characteristic is added to the correction characteristic, [correction characteristic] +
[compensation characteristic] = [compensated correction characteristic]. In addition, since the
frequency characteristic of the direct sound power should match the desired characteristic, the
direct characteristic (that is, the compensation characteristic is calculated independently)
according to [frequency characteristic of direct sound power] − [desired characteristic] =
[compensated correction characteristic]. Compensation) can be performed without passing
through the following steps.
[0066]
A specific example of the compensation characteristic will be described. FIG. 21 shows an
example of the reverberation time frequency characteristics of a certain room. The bass has a
long reverberation time, and the treble has a short reverberation time. Therefore, for example,
the frequency characteristic of the direct sound power Pd is obtained from the measured
characteristic (frequency characteristic of the total power P) of the room stored in the RAM 134
and the reverberation time frequency characteristic, and the difference between the frequency
characteristic of the total power P If (P-Pd) is calculated | required, it will become like FIG. This is
the compensation characteristic, and if this is used to apply any of the measurement
characteristic, the desired characteristic, and the correction characteristic as described above, the
compensation of the correction characteristic is performed as a result, and this compensation is
performed. By applying the correction characteristic to the music signal, the auditory
characteristic matches the desired characteristic. The display of each characteristic in the display
unit 40 may be the same as the characteristic before compensation.
[0067]
Correction of correction characteristics If, for example, a desired characteristic of 0 dB flat is set
with respect to the measurement characteristics shown in FIG. 18A, the correction characteristics
have large peak dips as shown in FIG. 18B. This peak-dip often results from a slight change in the
measurement environment, and when equalizing using such a correction characteristic as it is, a
portion largely corrected (indicated by ○ in the same (b)) ), With slight changes in the
environment (eg slight deviations in the frequency characteristics due to the influence of air
temperature and humidity), so that the correction can no longer be a true correction and
08-05-2019
23
conversely usually not As shown in c), the correction error becomes large, resulting in rather
peculiar characteristics. Therefore, the upper and lower limit values of the level of the correction
characteristic are set to an arbitrary value (for example, ± 10 dB) by the operation of the
operator. Thus, as shown in FIG. 18 (d), the correction characteristic calculation means 142 of
FIG. 1 restricts the upper and lower limit values of the correction characteristic to the set value to
prevent the correction from being performed more than necessary. Prevent the increase. Further,
since the maximum value on the + side of the correction characteristic is limited by this, the
maximum input can be suppressed, and distortion of the entire system such as the power
amplifier and the speaker can be suppressed.
[0068]
Also, if 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 as it is based on the calculated
correction characteristic, the speaker may be overloaded. The frequency range is set according to
the above, and the correction characteristic is utilized only within that 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 in the form of a horizontal bar graph as the corrected frequency range
display 50 on the display unit 40 of FIG.
[0069]
As described above, FIG. 19 shows an example of a specific calculation process at each stage
from the measurement data being obtained to the final determination of the correction
characteristic. Data to be used for calculation of measurement characteristics is selected (S61)
and weighted from among a plurality of times of measurement data stored in the band data
memory 126 of FIG. 1 and selected. It is assumed that b = 1 to B by the band number obtained by
dividing the selected data. B is 31 or 61 in this embodiment.
[0070]
When a plurality of data are selected, the group averaging means 130 obtains the group average
for each band (S62). Then, an average value of all the bands of the group average is obtained
(S63). Further, normalized average measurement data is obtained (S64), and this is displayed on
the display unit 40 as a measurement characteristic. The measured characteristic Nb is spline-
08-05-2019
24
interpolated to be continuous data. By normalization, the average value of the measurement
characteristics Nb is adjusted to be always 0 dB, and even if the sound collection level is small,
the display of the measurement characteristics on the display unit 40 is always approximately on
the same level. It becomes easy to compare with the characteristic display.
[0071]
When the desired characteristic Db is set by the operation of the operator (S65), Eb = Nb−Db is
obtained as the correction characteristic in the calculation means 142 (S66). The measurement
characteristic Nb here is data after spline interpolation. Then, the average value of all the bands
of this correction characteristic is obtained (S67). Further, the calculating means 142 obtains a
normalized correction characteristic (S68). By normalization, the average value of the correction
characteristic Fb is adjusted to be always 0 dB, whereby the sound before and after correction as
a whole is not changed in sound volume but only in sound quality.
[0072]
A process of restricting the upper limit value and the lower limit value of the level shown in FIG.
18D is performed on the obtained correction characteristic Fb (S69). In addition, the processes of
steps S66 to S69 are performed only within the designated frequency range of FIG. 18 (e).
Outside the designated frequency range, processing to flatten the correction characteristic by 0
dB is separately performed (S70). The correction characteristic finally determined in this manner
goes to a routine for calculating a convolution (equalizer) filter coefficient (S71).
[0073]
Calculation of Equalizer Filter Coefficients The FIR filter algorithm for acoustic characteristic
correction has advantages and disadvantages, and may not be usable depending on the purpose
of use. Therefore, here, as the FIR filter, one of a linear phase filter and a minimum phase filter
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 FIGS. 6 (d) and
6 (e), and their advantages and disadvantages are as follows.
[0074]
08-05-2019
25
Transmission characteristics Delay amount Ease of calculation of filter coefficients Linear phase
filter × × (large) ○ Minimum phase filter Δ ◎ (small) Δ According to this, the linear phase
filter has good transmission characteristics and filter coefficient calculation is easy The delay is
too large (see FIG. 6 (d)), and can not be used when real-time characteristics such as PA and
mixdown are required (since the raw sound and the equalized sound deviate in time). In addition,
although the minimum phase filter is inferior to the linear phase filter in terms of transmission
characteristics and easiness of filter coefficient calculation, since there is almost no delay (see
FIG. 6E), it is suitable when real time property is required. Therefore, one device can be used in
various situations by allowing the operator to select one of the algorithms according to the
purpose of use.
[0075]
In any case, since the correction characteristic assignment utilizes an FIR filter using digital
convolution operation, it is possible to impart a linear phase filter, a minimum phase filter, or
other special characteristics simply by switching the algorithm, The specification change is
extremely easy, and if the calculation accuracy is arbitrarily increased as necessary, the
correction accuracy can be set arbitrarily, and the practical effect of using the FIR correction
means in this kind of acoustic characteristic correction device is large .
[0076]
An example of a procedure for calculating the impulse response of the linear phase filter and the
impulse response of the minimum phase filter by using the inverse Fourier transform or the like
from the correction characteristic in the equalizer filter coefficient calculation means 144 will be
described.
[0077]
(A) Calculation of Impulse Response of Linear Phase Filter i) The correction characteristic is
temporarily divided into bands (for example, every 1/3 to 1/12 octave pitch), and the power
average for each band is determined.
ii) The obtained power average value is used as a value at the center frequency of each band to
interpolate into 4096-point data that can be Fourier-transformed by spline interpolation or the
like.
08-05-2019
26
iii) Inverse Fourier transform is performed on complex-form data in which the data obtained in ii)
is the real part (corresponding to the amplitude term) and the imaginary part (corresponding to
the phase term) is all zero. iv) The real part of the resulting complex format data is directly used
as a linear phase impulse response, so these are set as the coefficients of the FIR filter
(convolution operator 34).
[0078]
(B) Calculation of Impulse Response of Minimum Phase Filter i) The correction characteristic is
temporarily divided into bands (for example, every 1/3 to 1/12 octave pitch), and the power
average for each band is determined. ii) The obtained power average value is used as a value at
the center frequency of each band to interpolate into 4096-point data that can be Fouriertransformed by spline interpolation or the like. iii) The data obtained in ii) is used as the real part
and the Hilbert transform is applied to the complex format data in which all imaginary parts are
set to 0, and the complex format data that conforms to the correction characteristic curve and
has a minimum phase shift Do. In this complex type data, necessary phase components are added
to the imaginary part. iv) Inverse Fourier transform the complex format data obtained in iii). v)
Since the real part of the resulting complex format data is the minimum phase impulse response,
these are set as the coefficients of the FIR filter (convolution calculator 34).
[0079]
In addition to the linear phase filter and the minimum phase filter, a filter having an intermediate
characteristic may be prepared, and an arbitrary filter may be selected therefrom.
[0080]
Confirmation of Correction Characteristics The coefficient of the FIR filter 34 is set in the abovedescribed procedure to confirm the correction effect, and the results are shown in FIG.
(A) is an initial (i.e., without equalization) measurement result at each measurement point P1 to
P5 (see FIG. 5). (B) shows a measurement characteristic in which measurement data of each of
the points P1 to P5 are collectively averaged with the same weighting and a desired
characteristic arbitrarily set by the operator. (C) is a correction characteristic obtained as a
08-05-2019
27
difference from the desired characteristic of (b). The FIR filter coefficients calculated based on
the correction characteristics are set in the convolution unit 34 to configure an equalizer, and the
measurement signal (band signal or TSP signal) is reproduced through the equalizer to perform
measurement again. The result measured at each measurement point P1 to P5 is shown in FIG.
According to this, it was confirmed that the corresponding characteristic correction was made at
every measurement point and the optimum correction was made for the area including these
points, as compared with that before the correction of (a).
[0081]
By inputting a music source instead of the music reproduction measurement signal and
reproducing it through an equalizer (convolution calculator 34), it is possible to enjoy music
appreciation with reproduction characteristics as desired.
[0082]
In the embodiment described above, in the calculation of the measurement characteristics, for
example, spline interpolation is performed on the average value obtained for each divided band
in any case of the band signal method and the TSP method to obtain the measurement
characteristics, and Although the band division is performed again when calculating the FIR filter
coefficients that realize the characteristics, the present invention is not particularly limited
thereto.
[0083]
That is, when displaying the obtained measurement characteristic with high accuracy or trying to
use the measurement characteristic separately, it is much more difficult to use the divided band
data as it is, but in other cases the correction characteristic is By performing spline interpolation
at the time of calculation, it is possible to omit or simplify interpolation processing at the time of
calculation of measurement characteristics before that.
For example, the correction information is calculated as a correction value for each frequencydivided band based on the measured characteristics calculated for each frequency-divided band
and the desired characteristics set for each frequency-divided band, and calculated for each band
The correction value can also be calculated by interpolation as a value at about the center
frequency of each band by interpolation to obtain a correction characteristic.
08-05-2019
28
In this way, it is possible to prevent the occurrence of a large peak-dip in the correction
characteristics and to prevent the sense of incongruity due to extreme correction as a matter of
course. The correction characteristics are calculated as they are based on the measurement
characteristics interpolated by 4096 points. The amount of computation can be significantly
reduced as compared with the case and the like, and it is effective because the correction
characteristic to be finally obtained does not cause significant deterioration in accuracy.
[0084]
As described above, according to the present invention, the compensation characteristic is
compensated according to the reverberation time frequency characteristic or the average sound
absorption coefficient frequency characteristic of the sound field to be reproduced. Properties
can be matched to desired properties.
[0085]
Brief description of the drawings
[0086]
FIG. 1 is a block diagram showing an embodiment of the present invention, showing a control
configuration of the acoustic characteristic correction device 10 of FIG.
[0087]
FIG. 2 is a block diagram showing a hardware configuration in the acoustic characteristic
correction device 10 to which the present invention is applied.
[0088]
3 is an external view showing a panel configuration of the remote control unit 14 of FIG. 2;
[0089]
4 is a flowchart showing an outline of a procedure from measurement of characteristics by the
music characteristic correction device of FIG. 1 to use as an equalizer.
[0090]
FIG. 6 is a diagram showing the connection state of the device and the microphone arrangement
when measuring the acoustic characteristics using the acoustic characteristic correction device of
08-05-2019
29
FIG. 2 and the connection state of the device when used for music reproduction as an equalizer.
[0091]
6 is a diagram showing various characteristics in the process of FIG.
[0092]
FIG. 7 is a diagram showing a band division state at the time of measurement.
[0093]
8 is a diagram showing a band signal used in the band signal method.
[0094]
FIG. 9 is a diagram showing an outline of the TSP method.
[0095]
FIG. 10 is a diagram for explaining a method of obtaining continuous measurement
characteristics by interpolating between bands based on data of each divided band.
[0096]
FIG. 11 is a flowchart showing an example of an operation procedure in calculation of test and
measurement characteristics.
[0097]
FIG. 12 is a flow chart showing a setting procedure of desired characteristics.
[0098]
FIG. 13 is a view showing various patterns of desired characteristics prepared in the ROM.
[0099]
FIG. 14 is a diagram showing a characteristic adjustment method in a conventional graphic
equalizer or parametric equalizer.
08-05-2019
30
[0100]
FIG. 15 is a diagram showing a correction method of desired characteristics.
[0101]
FIG. 16 is a diagram showing another method of correcting the desired characteristic.
[0102]
17 is a flowchart showing an operation process for realizing the method of FIG.
[0103]
FIG. 18 is a diagram showing a method of correcting the correction characteristic.
[0104]
FIG. 19 is a flow chart showing an example of an operation process at each stage from when
measurement data is obtained to when a correction characteristic is determined.
[0105]
FIG. 20 is a diagram showing measured values, correction characteristics, and measured values
of correction effects.
[0106]
FIG. 21 is a diagram showing an example of reverberation time frequency characteristics.
[0107]
22 is a diagram showing an example of the compensation characteristic obtained from the
reverberation time frequency characteristic and the measurement characteristic of FIG. 21. FIG.
[0108]
Explanation of sign
[0109]
08-05-2019
31
DESCRIPTION OF SYMBOLS 10 Acoustic characteristic correction apparatus 34 Convolution
operation unit (correction characteristic provision means) 38 Operation part (desired
characteristic setting means) 70, 76, 78 Room, speaker (reproduction system including sound
field) 134 RAM (measurement characteristic output means) 143 Reverberation time or average
sound absorption coefficient frequency characteristic output means 145 characteristic
compensation means
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32
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