JPS6216364B2 - - Google Patents

Description

DETAILED DESCRIPTION OF THE INVENTION This invention relates to a method for measuring transient characteristics of a transmission system, such as a reverberation waveform, and in particular a method for correcting the influence of noise occurring in a transmission system and measuring highly accurate transient characteristics using digital technology. It is related to.

In general, it has often been necessary to carry out targeted characteristic measurements in order to learn various characteristics of a particular room from an acoustic point of view.

For example, when a lecture is given in a single hall or a musical performance is performed by a band, the acoustic effects of the speech or music become important. In this case, if matters such as the attenuation characteristics for the signal from one signal source or the reverberation time are clearly understood, it is important to understand the structure and acoustic characteristics of the venue where the event will be held. This means that you can aim for high acoustic effects.

Among various acoustic characteristics, several methods have been considered in the past for measuring transient characteristics or attenuation characteristics, including, for example, reverberation time characteristics.

One method is to measure the reverberation time by exciting the room using a tone burst wave and integrating the impulse response of the tone burst wave, but the accuracy with which the reverberation time can be determined from the decay curve is It was limited by random changes in the decay curve (transient curve). One way to minimize the effect of attenuation curve variations on measured reverberation time values is to repeat the measurements many times and average them. However, this method is not only inefficient, but also does not reveal the true characteristics of the attenuation.

By the way, as a method to measure transient characteristics,
MR Schroeder's so-called impulse square integration method is known. The principle is that when the sound is interrupted from a steady state, the essential transient response characteristic <S ² (t)>, which corresponds to the average of ∞ times of the transient characteristics of the sound receiving point, is expressed as an impulse between the sound source and the sound receiving point. According to this, the sound pressure level S(t) at a certain point in time t of the transient characteristic is expressed as follows.

<S ² (t)>=N∫ ^∞ _t r ² (x) dx ...(1) where N: noise power r(x): impulse response Therefore, the impulse response in the integral interval [t, +∞] By squaring r(x) and integrating it, an infinite set average of the squares of the sound pressure level S(t) at time t can be obtained.

However, in a transmission system that includes a chamber, etc., in addition to the impulse response r(x) from the sound source, there is noise such as background noise in the room.
(x) exists, this will be superimposed on the impulse response r(x), making the measurement inaccurate. In other words, accurate transient characteristics and reverberation characteristics of the impulse response from the sound source cannot be measured, resulting in results that differ from the original characteristics. In particular, for impulse responses that attenuate over time, noise such as indoor background noise greatly influences the shape of the attenuation curve depending on how the integration interval is taken.

For example, if the integral interval is too long, the impulse response will be affected too much by the noise superimposed on it, making it impossible to measure the transient characteristics accurately, and conversely, if the integral interval is too short, the integral interval of the impulse response itself will become too large. Since there is a large interval that is too short to be integrated, it is impossible to measure the transient characteristics properly. Therefore, if the integration interval is set too long or too short, accurate measurements cannot be expected, so it is important to set an appropriate integration interval, especially when the S/N ratio is insufficient. It has meaning.

If such an integral interval is not appropriate, the transient characteristics obtained as described above, such as the slope of the reverberation waveform itself, will change, the reliability of the characteristic curve will decrease, and the usable range will be small and limited. I become confused.

From the above viewpoint, the present invention expands the confidence interval of the reverberation waveform by removing the influence of noise, and also sets the integration end point T in the integration interval, especially ∫ ^T _t r ² (x) dx, to a sufficiently large value. The purpose of the present invention is to provide a method of measuring the transient characteristics of a transmission system using digital technology, which does not require strict settings.

That is, the method according to the present invention is based on the impulse response r(x) and the noise n existing in the transmission system.
The square of the effective value of noise n(x), n ² _eff , is subtracted from the square of the sum of (x), and the results of this subtraction are sequentially accumulated to measure the transient characteristics. It is characterized by what has been achieved.

Embodiments of the present invention will be described below with reference to the drawings. FIG. 1 is a block diagram showing the configuration of a measuring device according to an embodiment of the present invention. In this figure, the output of a short sound generator 1 is installed in a room 2 in which the transient characteristics of a room 2, in particular the reverberant waveform, are measured. It is designed to be input to the sound source 3 that was created. A microphone 4 is installed in this chamber 2, and the output of this microphone 4 is passed through an amplifier circuit 5, a low-pass filter 6, and a hold circuit 7 to an analog/digital conversion circuit (hereinafter referred to as an A/D conversion circuit). 8), where it is converted into a digital signal, and then
The squared signal is squared by a squaring circuit 9 and inputted to a subtracted input terminal of a subtracting circuit 10. The output of this subtraction circuit 10 is input to one input terminal of an accumulator 11, and the output of this accumulator 11 is input to the other input terminal of this accumulator 11. The output of the register 14 is input to one input terminal of an AND gate 16 via a division circuit 15, and the output of the AND gate 16 is input to a memory (hereinafter referred to as RAM) 12, a register 13, and a register 14, respectively. The output of the gate control circuit 17 is input to the input terminal of the gate control circuit 17. Note that the division circuit 15 divides the output of the register 14 into
1/τ (τ will be explained later), and the gate control circuit 17 is
It generates a control signal to open and close the AND gate 16. The output of the AND gate 16 is input to the subtraction input terminal of the subtraction circuit 10. The output of the RAM 12 is input to the subtraction input terminal of the subtraction circuit 18, and the output of the register 13 is input to the subtracted input terminal of the subtraction circuit 18 via the latch circuit 19. The output is input to a display/recording device 22 via a read-only memory (hereinafter referred to as ROM) 20 with a function of logarithmic calculation (10logX) and an interface circuit 21, where it is displayed/recorded as a reverberation waveform. It is becoming more and more common.

In FIG. 1, broken lines indicate control signals from the timing control circuit 23 and the short tone generator 1, and the signal C-1 indicates the control signals from the hold circuit 7,
The signal C-2 is input to the A/D conversion circuit 8 and the accumulator 11, the signal C-2 is input to the register 14, the signal C-3 is input to the RAM 12, the register 13, and the latch circuit 19, and the signal C-4 is input to the RAM 1.
The signal C-5 is also input to the gate control circuit 17 and the accumulator 11.

Next, the operation of the above device will be explained with reference to the timing chart shown in FIG.

First, before an impulse is generated by the short sound generator 1, the microphone 4 installed in the room 2 picks up the noise n(x) in the room 2. Noise n picked up by microphone 4
(x) is amplified by the amplifier circuit 5 and then input to the hold circuit 7 via the low-pass filter 6. A control signal C-1 from the timing control circuit 23 is input to the hold circuit 7.
This control signal C-1 holds the signal input to the hold circuit 7 as shown in FIG.
This is for supplying this held signal to the circuits below the A/D conversion circuit 8, and is used for shifting registers or calculations in the A/D conversion circuit 8, squaring circuit 9, etc., which will be described later. be. The noise n(x) held by the hold circuit 7 is converted into a digital signal by the A/D conversion circuit 8, squared by the squaring circuit 9, and then supplied to the subtracted input terminal of the subtraction circuit 10. Ru. On the other hand, the output of the gate control circuit 17 is
Before the impulse is generated by the short tone generator 1, the signal "0" at the binary logic level is output, so the output of the AND gate 16 holds the signal "0", and this signal "0" is output. The subtraction circuit 10
is supplied to the subtraction input terminal of . As a result, the subtraction circuit 10 outputs [(output of squaring circuit 9=n ² (x))−
"0"], and the result of this operation, that is, the square n ² (x) of the noise n(x), is sequentially accumulated in the accumulator 11. The square of the accumulated noise n(x) (∫n ² (x)dx) is sequentially input to the register 14 at a period τ by the signal C-2 shown in FIG. The output is converted to 1/τ by the division circuit 15 and supplied to one input terminal of the AND gate 16. Note that the accumulator 11 is cleared each time its contents are input to the register 14.
According to the above result, the output of the division circuit 15 contains the calculation result of 1/τ ∫〓 _p n ² (x) dx, that is, the noise n
The square n ² _eff of the effective value of (x) is obtained. Therefore, before an impulse is generated from the short tone generator 1, the square n ² _eff of the effective value of the noise n(x) is always prepared at the output of the divider circuit 15 while being updated at the period τ. be.

Next, the operation when an impulse is generated from the short sound generator 1 will be explained. First, at the same time as the short sound generating device 1 generates an impulse, the short sound generating device 1 generates a signal C-5 shown in FIG.
is the gate control circuit 17 and the accumulator 1
1, and the output of the gate control circuit 17 becomes a binary logic level signal "1", opening the AND gate 16 and clearing the accumulator 11. Next, the impulse generated by the short sound generator 1 and the noise in the room 2 are transmitted to the microphone 4.
The sound is picked up by the impulse response r
(x) and noise n(x) [r(x)+n
(x)=R(x)] is input to the hold circuit 7 via the amplifier 5 and the low-pass filter 6. The impulse response r held in this hold circuit 7 by the signal C-1 shown in FIG.
The sum R(x) of (x) and noise n(x) is A/D
After being converted into a digital signal by the conversion circuit 8, it is squared by the squaring circuit 9 and input to the subtracted input terminal of the subtraction circuit 10. At the subtraction input terminal of this subtraction circuit 10, the output of the above-mentioned division circuit 15,
In other words, the square n ² _eff of the effective value of the noise n(x) is input through the AND gate 16, and the subtraction circuit 10 calculates [R ² (x) − n ² _eff ], and the result of this calculation is are sequentially accumulated in the accumulator 11 at the timing of the signal C-1 of the timing control circuit 23. That is, in the accumulator 11, the calculation ∫{R ² (x)-n ² _eff }dx is performed. The result of this calculation is the signal C shown in FIG.
-3, the data is sequentially written into the RAM 12 starting from address 1, and is sequentially input into the register 13. It is assumed that the RAM 12 has an address configuration as shown in FIG.
3 (shown in FIG. 2).

Now, at time t _k shown in FIG.
To explain the case where the accumulated value ∫ ^tk _p {R ² (x) − n ² _eff }dx=S _k , which is the output of the accumulator 11, is input to address k of 2, this accumulated value S _k is , is also input to the register 13 at the same time, and the output of this register 13 is supplied to the subtracted input terminal of the subtraction circuit 18 via the latch circuit 19. ). On the other hand, at addresses 1 to k of the RAM 12, address 1→S ₁ =∫ ^t1 _p {R ² (x)−n ² _eff }dx address 2 → S ₂ =∫ ^t2 _p {R ² (x)− n ² _eff } dx 〓 address k → S _k = ∫ ^tk _p {R ² (x) − n ² _eff } dx is stored, and each accumulated value is
The data are sequentially written until the next write time t _k+1 of the RAM 12 arrives, and are sequentially supplied to the subtraction input terminal of the subtraction circuit 18 . ( _{Changes in} the address of the RAM ₁₂ during this period are shown in _FIG .
......[S _k −S _k ] is performed, and the result of this calculation is logarithmically compressed by the ROM ₂₀ having a logarithmic calculation function ( _10log The display/recording device 22 displays/records the information over time. Note that this display device is made of, for example, a CRT, liquid crystal, or light emitting diode, and this recording device is, for example, an electromagnetic oscilloscope.

In this way, each time the output of the accumulator 11 is written to the RAM 12 by the signal C-3, the display/recording device 22 displays the progress of the attenuation characteristic of the impulse response r(x) while being updated each time. / recorded. Then, at the end of the accumulation (t=t _o ), the attenuation characteristic of the impulse response r(x) is displayed on the display/recording device 22.
It is obtained by In other words, this accumulation end time t
_o , each address of the RAM 12 has 1st address→S ₁ =∫ ^t1 _p {R ² (x)−n ² _eff }dx 2nd address→S ₂ =∫ ^t2 _p {R ² (x)−n ² _eff }dx 〓 address n→S _o =∫ ^tn ₀ {R ² (x) − n ² _eff } Each cumulative value of dx is stored, and the calculation [S _o −S ₁ ] is performed using each cumulative value. , [S _o −S ₂ ], ……… [S _o −
S _o ], the attenuation characteristic of the impulse response r(x) is obtained. However,
4A to 4E show the waveforms of each part in the above device.
8 is shown in an analog display for convenience, d shows an example of the display on the display device 22, and e shows the signal C-3 of the timing control circuit again. Furthermore, FIG. 5 shows that according to the present invention, even if the accumulation end time t _o is not strictly determined,
This is a diagram showing that the attenuation characteristics can be obtained correctly and that the confidence interval (utilization interval) of the reverberant waveform is expanded. O shows the attenuation characteristics in the case where the present invention is not used (input signal R(x) with noise n(x) still included).
The figure shows the attenuation characteristics obtained by accumulating the attenuation characteristics. (For the principle by which the influence of noise n(x) can be removed by this invention, please refer to "Japanese Patent Application No. 1983-37165" filed by the applicant of this invention.) As explained above, this invention According to the above, since the transient characteristics are measured after removing the influence of noise from the input signal, the confidence interval of the curve is expanded, and at the same time, if the measurement end point T is set sufficiently large, the high Measurements can be made with precision,
It becomes easy to determine the confidence interval in the display/recording device, and it is possible to obtain a measuring device that is extremely easy to measure. On the other hand, since digital processing is used, there is an advantage that the entire device can be constructed more easily and at a lower cost than when analog processing is used.

[Brief explanation of the drawing]

FIG. 1 is a block diagram showing the configuration of an embodiment of the present invention, FIG. 2 is a time chart for explaining the operation of the embodiment shown in FIG. 1, and FIG.
FIG. 4 is a diagram showing the address structure of the RAM, FIG. 4 is a diagram showing waveforms of various parts of the embodiment shown in FIG. 1, and FIG. 5 is a graph showing the effects of the present invention. 8...Analog/digital conversion circuit, 9...
Square circuit, 10...subtraction circuit, 11...accumulation circuit, 15...arithmetic circuit (divider circuit), 22...display/recording device.

Claims (1)

[Claims] 1. In a method of measuring the transient characteristics of a transmission system by analyzing the impulse response r(x), noise n(x) is converted into a digital signal and then squared,
This square value n ² (x) is cumulatively averaged to calculate the square of the effective value of the noise n ² eff, while the sum of the impulse response r(x) and the noise n(x) [r
(x)+n(x)] is converted into a digital signal, squared, and the squared value R ² (x)=[r(x)+n
(x)] ² to the square of the effective value of the noise n(x)
Subtract n ² eff and accumulate the result of this subtraction to get ∫
^ti(i=1 , ² ,…… ⁿ⁾ _p {R ² (x)−n ² eff
}dx, and the transient characteristics are measured based on the calculated results.