JPH01251900A - Acoustic system - Google Patents

Abstract

(57) [Summary] This bulletin contains application data before electronic filing, so abstract data is not recorded.

Description

[Detailed description of the invention] [Purpose of the invention] (Industrial application field) TECHNICAL FIELD The present invention relates to an acoustic system.

(Prior Art) As a conventional sound system, there is one shown in FIG. 6, for example. As shown in the figure, in the listening room 67, a speaker 51 is placed in front of the listener 65.
．． 53 and a television (TV) 59 are arranged, and speakers 55 and 57 are arranged behind them.

The reproducing device 61 sends a video signal to a television 59, and sends a right channel signal and a left channel signal to speakers 51 and 53. The surround decoder 63 is the playback device 61
The rear sound sent from the decoder is decoded, delayed by several tens of meters, and sent to the speaker 55.57.

The sound emitted from speakers 55 and 57 is from speaker 51
．． Since it is emitted later than the sound emitted from 5B,
The listener 65 can hear audio with a sense of realism.

There is also an audio system that creates a deeper sense of presence by adjusting the delay and volume of the four speakers 51, 53, 55, and 57.

Furthermore, like a sound field reproduction device, six speakers are used,
There are also sound systems that simulate as if you were listening in a concert hall.

In such an audio system, it is necessary to correctly adjust the amount of delay and volume of each speaker. If it is not adjusted correctly, for example, in a scene where an airplane flies from back to front, the sound from back to front will not be continuous, and the sound will cut off overhead.

However, adjusting parameters such as speaker delay and volume in such an audio system requires skill.
Adjusting it takes time.

(Problems to be Solved by the Invention) As described above, in the conventional acoustic system, there is a problem in that it is difficult to adjust the parameters of each speaker, and it takes a lot of time for adjustment.

The present invention has been made in view of these problems, and an object of the present invention is to provide an audio system in which parameters of each speaker can be easily adjusted.

[Structure of the Invention] (Means for Solving the Problems) In order to achieve the above object, the present invention includes a plurality of speakers, a microphone for collecting sound, and a plurality of speakers emitted from each of the speakers in response to a specific signal. The apparatus includes means for calculating an adjustment amount of a parameter of each speaker from an audio signal collected by the microphone, and means for correcting a parameter of each speaker according to the calculated adjustment amount.

(Function) In the present invention, a microphone collects audio signals emitted from a plurality of speakers in response to a specific signal, calculates the adjustment amount of each speaker's parameter from the collected signal, and according to the calculated adjustment amount. Correct the parameters of each speaker.

(Example) Hereinafter, an example of the present invention will be described in detail based on the drawings.

FIG. 1 is a diagram showing an audio system according to a first embodiment of the present invention, and this audio system includes speakers 1.3.5.
7. Delay circuit/amplifier (delay and amplifier) 9.1
1.13.15, microphone 17, analog-to-digital converter (AD converter) 19, delay amount/volume detection circuit 21
, a sound field reproduction device 23.

The speaker 1.3.5.7 converts the electrical signal sent from the sound field reproduction device 23 into an audio signal and outputs the audio signal. Delay circuit/amplifier 9.11.13.15 is connected to corresponding speaker 1.
The delay amount and volume, which are the parameters of 3.5.7, are set according to the signal sent from the delay amount/volume detection circuit 21. The microphone 17 is placed at the listener's position and collects the audio signals emitted from each speaker 1.3.5.7.

The AD converter 19 converts the analog signal sent from the microphone 17 into a digital signal. The delay amount/volume detection circuit 21 detects each speaker 1.
Calculate the delay amount and volume of 3.5.7, and send the calculated values to delay circuit/amplifier 9.11.13.15. The sound field reproducing device 23 sends the reproduced audio signal to the speaker 1.3.5.7 via the delay circuit/amplifier 9.11.13.15.
send to

Next, the operation of the embodiment will be explained based on FIG. 2.

FIG. 2 shows the impulse signal and the signal waveform received by the microphone 17.

A microphone 17 is placed at the listener's position, and each speaker 1
．． 3. Make sounds in order from 5.7. That is, Fig. 2 (a
), the speaker 1.3.5.7 outputs an impulse signal. This impulse signal may be a sine wave or a frequency sweep signal. The signals received by the microphone 17 from each speaker 1.3.5.7 are shown in FIG. 2(b).
The waveforms shown in ~(e) are obtained. That is, each speaker 1.
Depending on the positional relationship between 3.5.7 and the microphone 17, the signal received by the microphone 17 is subject to predetermined attenuation and delay. For example, when an impulse signal with a volume of AO is emitted from the speaker 1, a signal with a volume of APR is received by the microphone 17 after a delay time DPR. The signal received by the microphone 17 is converted into a digital signal by the AD converter 9 and sent to the delay amount/volume detection circuit 21.

The delay amount/volume detection circuit 21 detects the delay ffi D PR1 sound 1AFR from the signal sent from the AD converter 19. Further, when the impulse signal of the sound ff1Ao is emitted from the speaker 3, the delay amount/volume detection circuit 21 detects the delay ffi D PL and the sound 1AFL. Similarly, the delay amount/volume detection circuit 21 determines whether the delay amount Dl? l? ,
DI? Detect L, volume ARP, and APL.

Then, the delay amount/volume detection circuit 21 detects the sound ffi A F
The volumes are set so that R to ARL have appropriate volume distribution (for example, all four are at the same level), and sent to the delay circuit/amplifier 9.11.13.15.

Also, compare the delay amounts DPR to DRL with the appropriate delay amount, set the delay amount for each speaker, and set the delay amount for each speaker.
．． Send on 13.15.

Thus, in this embodiment, each speaker 1.3.
5.7 volume and delay amount can be adjusted.

In addition, as shown in Figure 3, each speaker 1.3.5.7
It is also possible to provide a circuit for correcting the frequency of the signal sent to. Such a frequency correction circuit includes a fast Fourier transformer (CFFT) 25, a frequency characteristic generation circuit for correction 27, an inverse fast Fourier transformer (FFT) 29, and a convolution circuit 3.
Consists of 1. The fast Fourier transformer 25 generates a signal 52 (FIG. 4(b)) indicating the frequency characteristics of the impulse signal Sl (FIG. 4(a)). Correction frequency characteristic generation circuit 2
7 generates a signal S3 (FIG. 4(C)) showing the insufficient frequency characteristics of the signal S2.

The inverse fast Fourier transformer 29 generates correction coefficients from the signal S3 and sends them to the convolution circuit 31. The convolution circuit 31 performs correction indicated by the correction coefficient on the signal S1 before correction, and outputs the corrected signal S to each speaker 1.3.5.7.
Send 4. FIG. 4(d) is a diagram showing the frequency characteristics of such a signal S4.

If the signal S1 before correction lacks low frequency components, the correction frequency characteristic generation circuit 27 generates a signal S3 that increases the low frequency. The signal S4 sent to speaker 1.3.5.7 is corrected to increase the low frequency component.

In this embodiment, a delay circuit/amplifier is provided for each speaker, but if a delay circuit is provided in the sound field reproducing device, it may be used.

Further, in this embodiment, only one microphone is used, but a plurality of microphones may be used to obtain the distance and direction to each speaker for more detailed adjustment.

Furthermore, only the volume adjustment can be done manually. For example, you can measure the distance between the speaker and the microphone using a measuring tape, manually adjust the volume based on that distance, and then automatically adjust only the amount of delay. In this case, a cheaper sound system can be provided.

FIG. 5 shows another embodiment of the invention.

That is, in this embodiment, instead of the microphone 17 and AD converter 19 of the first embodiment, each speaker 1, 3, .
5.7 and sends it to the delay amount/volume detection circuit 12.

As shown in FIG. 5(b), the remote control 33 includes a left microphone 35, a right microphone 37, a low-pass filter and AD converter (LPF-ADC) 39,
It consists of a buffer memory 41, a transmission encoder 43, and a transmitter 45.

Microphones 35 and 37 receive the sound output from the speakers. The LPF-ADC 39 extracts only the low frequency components of the signals sent from the microphones 35 and 37, and converts the extracted signals into digital signals. Buffer memory 41 temporarily stores digitized signals. This buffer memory 41 temporarily stores data in consideration of the case where the data transfer from the remote control 33 is slower than the capture speed. The transmitting encoder 4B encodes the data stored in the buffer memory 41. The transmitter 45 sends the signal sent from the transmitting encoder 43 to the delay amount/volume detection circuit 21 as a wireless signal.

In this way, as in the first embodiment, the microphone 1
7. Since it is not necessary to connect the AD converter 19 to the delay/volume detection circuit 21, the parameters of each speaker can be adjusted more easily.

[Effects of the Invention] As described above in detail, according to the present invention, it is possible to provide an audio system in which parameters of each speaker can be easily adjusted.

[Brief explanation of the drawing]

FIG. 1 is a diagram showing an acoustic system according to a first embodiment of the present invention, FIG. 2 is a signal waveform diagram of the first embodiment, FIG. 3 is a block diagram showing the configuration of a frequency correction circuit, and FIG. The figure shows the waveform diagram of the signal input to the frequency correction circuit and the frequency characteristics of each part, Figure 5 shows the sound collection part of another embodiment of the present invention, and Figure 6 shows the conventional acoustic system. FIG. 1.3.5.7...Speaker, 9.11.13.15
...Delay circuit/amplifier, 17...Microphone, 1
9...Analog-digital converter, 21...Delay amount/
Sound volume detection circuit, 23..., Sound field reproduction device agent Patent attorney Nori Chika Ken Yudo Uji Hiroshi (a) 51-to-1 Figure 3 No. 4LJ

Claims (1)

[Scope of Claims] A plurality of speakers, a microphone that collects sound, and an adjustment amount of a parameter of each speaker is calculated from a signal collected by the microphone of an audio signal emitted from each of the speakers in response to a specific signal. and means for correcting parameters of each speaker according to the calculated adjustment amount.