JPS5850812A - Transmitting circuit for audio signal - Google Patents

Abstract

PURPOSE:To obtain variable transmission characteristics and reproduced source sound quality with fidelity, by providing a convolutional integrating circuit to a transmission line for audio signals, and calculating and controlling the weighting coefficient of the convolutional integrating circuit. CONSTITUTION:The convolutinal integrating circuit (convolver) 5 which varies transmission characteristics consists of a delay circuit 1 with a number of output terminals, multipliers 2a-2n which control weighting coefficients for plural outputs of the delay circuit 1, and an adder 3. An audio signal applied to an input terminal 4 has transmission characteristics varied by the convolver 5 and is listened to at a listening position 8 through a power amplifier 6 and a speaker 7. To vary the transmission characteristics, the values of the multipliers 2a-2n are varied and the weighting coefficients are controlled so that the square of the difference in impulse response between the input terminal 4 of the convolver 5 and the listening position 8 is minimized at the listening position 8.

Description

DETAILED DESCRIPTION OF THE INVENTION The present invention provides a recording system for audio signals such as music signals and voice signals on a recording medium such as a magnetic tape and a record, and/or an acoustic space for audio signals pre-recorded on a recording medium. The purpose of this invention is to provide an audio signal transmission circuit that can record and/or reproduce the original sound with high fidelity to the listener. .

In general, Bohr (Bode) is used in the transmission line in the audio signal reproduction system as a means to change the sound quality.
Control means are employed to vary the frequency amplitude characteristics of the shape. However, with such conventional control methods, the sound pressure at the listening position is flat, but the structure of the speaker itself is not a minimum phase shift system due to various resonances, etc., so it becomes necessary to correct the level characteristics. Even if this level characteristic is corrected, the waveform will not be transmitted correctly due to phase distortion.

There was a problem in that the original sound quality could not be reproduced with high fidelity for the listener.

The present invention solves these conventional drawbacks,
A convolution integration circuit is provided in the audio signal transmission path, and the transmission characteristics are varied by calculating and controlling the weighting coefficients of both the convolution and integration paths. This makes it possible to record and/or reproduce the original sound quality with high fidelity for the listener.

The present invention will be described below with reference to drawings of embodiments.

FIG. 1 shows an entire embodiment of the present invention. In FIG. 1, 1 is a delay circuit having a large number of output terminals, 2a to 2n
3 is a multiplier that controls weighting coefficients for a plurality of outputs from the delay circuit 1, and 3 is an adder, which are connected to the input terminal 4.
A convolution/integration circuit (hereinafter referred to as a convolver) 6 is configured to vary the transmission characteristics of an audio signal applied to the convolver.

6 is a power amplifier, and 7 is a speaker.

In a transmission circuit having such a configuration, in order to obtain a weighting coefficient that minimizes the square of the difference between the impulse responses obtained at the input terminal 4 of the convolver 6 and the listening position 8 at the listening position 8,
The multipliers 2&~2n are configured to change their values. In such a state, if the discrete coefficients of the impulse responses of the convolver 5 are ql, q2...''m-1, then the discrete responses at the listening position 8 are 1°, f,...・・・
−+m−2 is expressed by the following formula.

However, hi is the transfer characteristic p=o −・・φ−・n 10m −2・・・・・・
Q) Next door.

Equation 2 can be further expressed as F=HG.

Here, let us take the square of the difference between the input impulse F0 and the impulse response F at the listening position 8 and set it as an evaluation function p.

P= (F-Fo)T(F-Fo) = (HG-Fo)T(HG-Fo) = (G"HT-FO") (HG-Fo)=G"H"
HG-Fo"HG-G"H"F 10F"Fo
o− ・・・・・・ Plow, and convolver 5 to minimize the evaluation function P
The impulse response G'ft- is determined, where T represents the transposed matrix.

And from formula 4 HHG-H"F . (It will be four.)

That is, the transmission characteristics are changed by setting the coefficients of the multipliers 2a to 2n as shown in Equation 6 above.

Sound pressure frequency characteristics at listening position? Become a flat carrier.

It is also possible to reproduce the source sound quality with high fidelity.

Incidentally, in the above explanation, the case where the characteristics of the transmission path in the reproduction system are changed is explained, but this transmission characteristic variable means 5
This can also be implemented in the same way when provided in a transmission path in a recording system that records audio signals from a microphone on a recording medium, and the same effects can be achieved. In this case, the weighting coefficient of the convolver provided in the transmission path in the recording system may be calculated and controlled by comparing the sound from the microphone with the output sound of the recording system composed of a recording medium.

FIG. 2 shows another embodiment of the present invention. In FIG. 2, 11 is a left speaker, 12 is a right speaker, and 13 is a convolver provided in the signal transmission path of the left speaker 11.
14 is a convolver provided on the signal transmission path of the right speaker 12, 15 is an input end, and 16 is a listener. Here, the convolvers 13 and 14 are constructed in the same manner as in the case of FIG. 1 described above.

In such a configuration, the front left and right speakers 11
, 12 is the input waveform to 11. , IB, impulse response 4. h2, and the input waveform f i B impulse response to the virtual speaker 17 at the location is h3. h4, from the front left and right speakers 11°12 to the listener 16
0The sound that reaches both ears is 'eqL' that reaches the left ear, and the sound that reaches the right ear is qR.

'J L=h1 yellow 1 H+h2-% i L
・・・・・・ (6) q R=h 2 * i R
+ h , + i L . . . Mountain (7), and furthermore, the sound reaching both ears of the listener 16 from the virtual speaker 17 is the sound reaching the left ear '(i-fL).

Assume that the sound reaching the right ear is 1tH.

'L"h4" lS @fH
=h3+is (becomes 91.

Then, the impulse responses of the convolvers 13 and 14 installed on the signal transmission paths of the left and right speakers 1 and 12 are respectively a.
, b, the sound at the left and right ears of the listener 16 is gl=h2+b+h1%"#0."-'
(10) qR=h vb+h2+a ,,,
,, (11). Also, assuming that is at this time is an impulse response.

fL=h4・(12)
fR=h3 (13)
From equations 10, 11, 12, and 13, g L 'w
fL.

Convolver 13, 14'i so that CIR→fR
5 adjustment, the two speakers 1.12 can reproduce the sound so that the sound source (virtual speaker 7) is located at the desired position.

here.

(1) h4. h2. h3. The impulse responses of h4 are each n discrete impulses, the impulse responses of (It)a and b are m discrete impulses, and the evaluation function pl impulse h1. h2 and h3
．． Let it be the square of the sound pressure difference of h4. In other words, the evaluation function P is (14).

Here, Equation 10.11 is changed to the following equation.

Below, go to the margin axis and quality 3 1゜eases as *L,
, toQ A A -Q “A””
A: Formula 16 can also be manipulated as shown in the following formula.

g: HX (17) Also impulse h3. Vector df consisting of h4: Defined as follows.

Therefore, the evaluation function P is P=(q-d)(q-d)...,...(1
8) This gives P = (XtHt-dt) (HX-d) =) [HHX - (X”Htd + d 'HX) + d
``d'' Here, the constants of the convolution integration circuits a and b are determined by finding the extreme value of X that makes the evaluation function P 0. That is.

-”-=2HtHX-2Hd=.

δX follows, HHX=Ha, --------(1
9) By finding X, the constants of a and b are determined.

It is also possible to reproduce the fundamental sound quality with high fidelity.

FIG. 3 shows a specific embodiment of the present invention. In FIG. 3, the output of the pulse generator 21 is passed through the switch 22 as the output of the terminal a or b, and is input to the amplifier 23 or the convolution integral Connected to circuits 24 and 25. The convolution integration circuits 24, 26 are connected to speakers 28, 29 via amplifiers 26, 27, and the audio signals are transmitted to microphones 31, 32 placed at the entrances of both ears of the listener 30. Here, when the contact point of the switch 22 is set to the a side, the 3 pulse signal is emitted from the speaker 33 via the amplifier 2rai, and the listener 30 hears both! = detected from 31.32. The signal is then recorded as data into the computer 36 via the microphone amplifier 34 and the analog-to-digital converter 36f. On the other hand, when the switch 227&: is set to the contact side, the pulse signal passes through a convolution and integration circuit (hereinafter referred to as a convolver) 24.25, and then passes through an amplifier 26.27.
The sound is output from the speakers 28 and 29 through the microphones 31°3 set in both ears of the listener 30 as described above.
2, and the data is set to the computer 36 via the amplifier 34 and analog-to-digital converter 36. Here, the computer 36 calculates the square of the difference between the sounds of the speaker 33 and the speaker 28.29 as described above, and the address and data are sent to the buffer circuit 37f via the convolver 24.25 as the computer output.
connected to. At this time, this convolver 24, 26
For example, the tris is configured so that multiple output signals can be extracted after a certain delay time with respect to an analog input signal.

4 The multi-output terminals are each configured to have a digital converter and an analog converter. In this case, the digital-to-analog converter of each tap output may be selected using the address signal described above, and the level of the analog delay signal may be controlled using the data signal. Another method may be to convert the analog input into digital data and then use two, for example, a shift register and a digital multiplier.

Moreover, the same thing can be considered even if the listener has a condensed head for sound collection.

FIG. 4 shows another embodiment of the present invention, and in FIG. 6, 41 is a voice band generator such as a tape recorder;
42 is a preamplifier and %43.44 is a convolver. Here, the audio signal passes through a preamplifier 42, and through a convolver 43, 44i, a power amplifier 45, 46 obtains power for driving the speakers 47, 48. At this time, the convolvers 43 and 44 are connected to the speakers 47 and 44, as described above.
48 and the listener 49, in advance, 1
6. The load of each tap output of the convolver is recorded in the memory 60, and this memory 50 also records the load coefficient of each convolver 43, 44 at the position of an arbitrary sound image to be synthesized. Here, for example, a three-dimensionally changing panbot 51 detects the potential of VRef with respect to the reference voltage, and is connected to an analog-to-digital converter 62 and then to a decoder 53. The decoder 63 stores in memory the position corresponding to the digital signal detected by the analog-to-digital converter 62 and the weighting coefficients of the convolvers 43 and 44 for obtaining the same synthesized sound image.

The sound image is detected at a desired position by changing the tap coefficients of each convolver 43 and 44.

Note that the three-dimensional panpot 51 can also be achieved by rotating the joystick bar of a normal two-dimensional panpot.

Furthermore, the same problem may occur even if headphones are used in place of the speaker J of this embodiment.

As described above, according to the present invention, the audio signal
8-50812 (5) A convolutional integrator circuit is provided in the transmission line, and the weighting coefficient of this convolutional integrator circuit is set to the impulse response of a specific characteristic at the listening position and the weighting coefficient of the convolutional integrator circuit. Since it is configured to perform calculation control to minimize the square of the difference with the position impulse response, it has the advantage of being able to reproduce the original sound quality with high fidelity by leveling out the sound pressure frequency characteristics without any phase distortion. It is something.

[Brief explanation of drawings]

FIG. 1 is a block diagram showing one embodiment of the audio signal transmission circuit of the invention, FIG. 2 is a block diagram showing another embodiment of the audio signal transmission circuit of the invention, and FIGS.
The figure is a block diagram showing still another embodiment of the audio signal transmission circuit of the present invention. . 5.13', 14, 24. ! 5, 43, 44...Medium/convolution integral circuit.

Claims (1)

[Claims] A convolution and integration circuit is installed in the audio signal transmission system,
By calculating and controlling the weighting coefficient of the convolutional integrator circuit so as to minimize the square of the difference between the impulse response of a specific characteristic at the listening position and the listening position impulse response determined by the setting of the weighting coefficient of the convolutional integrator circuit. An audio signal transmission circuit characterized by being configured to change transmission characteristics.