JPS62110399A - Audio signal processing device - Google Patents

Abstract

PURPOSE:To reproduce stereo sensation of small antiphase feeling by providing two high pass filters HPF, three low pass filters LPF, a audio removing means and two adders, dividing input signals to two frequency bands, and inputting only output signals of the LPF to the audio removing means. CONSTITUTION:The whole audio frequency band F0 is divided into frequency band F1 of 10Hz-5kHz and frequency band F2 of 5-20kHz making about 5kHz a boundary. Antiphase feeling of reproduced sound is lowered by inputting only frequency band F1 as input signals S1 and S2. To reduce antiphase feeling, it is enough to make upper limit of frequency band F1 small. As aural signals do not contain frequency band lower than about 100Hz, input signals S1 and S2 are added by an adder 22. The third LPF 19 having input/output frequency characteristic of a curve Y3 passes only the frequency band F3, and augments lowered sound volume.

Description

DETAILED DESCRIPTION OF THE INVENTION Field of Industrial Application The present invention is applicable to tape media such as audio tapes and video tapes, and 5. An audio signal that removes or attenuates the sound contained in an audio signal recorded on a disc medium such as a video disc or a compact disc, or an audio signal broadcast on television or radio. It relates to a processing device.

Prior Art FIG. 4 shows a block diagram of a conventional audio signal processing device.

In Figure 4, 1 and 2 are input terminals, 3 and 4 are BBD
The first one is represented by (bucket brigade device).
In the second analog delay circuit, 5 is a level variable device (hereinafter referred to as an attenuator), 6 is an adder, and 7 is an output terminal.

The operation of the audio signal processing device having the above configuration will be explained below. First, the signal S1 input to the input terminal 1 and the signal S2 input to the input terminal 2 are LCh and Rch signals of a commercially available music source. The signal S1 input to the input terminal 1 is input to the first analog delay circuit 3,
This results in a time-delayed signal S3. The time delay value of this first analog delay circuit 3 is fixed. On the other hand, the signal S2 input to the input terminal 2 is input to the second analog delay circuit 4 and becomes a time-delayed signal S4. The time delay value of this second analog delay circuit 4 is variable by the delay time control signal S7. The level of this signal Sa'd is varied by the attenuator 5 and becomes a signal S5. Further, the signal S3 and the signal S5 are inverted and added by an adder 6 to obtain a signal S6, which is output to the output terminal 7.

In this way, although the conventional example has two input terminals, it has only one output terminal.

Therefore, FIG. 6 shows a case where the conventional example has a two-channel configuration. Note that the same functions as in the conventional example are given the same numbers.

The analog signal S1 input to the input terminal 1 is branched into two, one of which is input to the first analog delay circuit 3 to obtain the analog signal S3, and the other is input to the fourth analog delay circuit 9 to obtain the analog signal S9. obtain.

On the other hand, the analog signal S2 input to input terminal 2 is 2
After branching, one is input to the third analog delay circuit 8 to obtain an analog signal S8, and the other is input to the second analog delay circuit 4 to obtain an analog signal S4. The analog signal S4 is an analog signal S5 whose amplitude is controlled by the attenuator 5.
On the other hand, the analog signal S9 becomes an analog signal S10 whose amplitude is controlled by the attenuator 10.

The analog signal S3 and the inverted signal of the analog signal S5 are added by an adder 6, and the analog signal S6 is output to the output terminal 7.
On the other hand, analog signal S8 and analog signal S1
An inverted signal of 0 is added by an adder 11, and an analog signal S11 is outputted to an output terminal 12.

If the analog signal S1 input to input terminal 1 is assumed to be the L channel of a general music source and is assumed to be L, and the analog signal S2 input to input terminal 2 is assumed to be the R channel of the above source and is assumed to be R, then the output is The analog signal S6 output to the terminal 7 and the analog signal S11 output to the output terminal 12 are expressed as in equation (1).

Here, (1-1,) is delayed by time t1, 4
・m represents the ratio of amplitude control.

To simplify the formula, the time delay values t1 to t4 are "o"
"', that is, there is no time delay, and assuming that the amplitude control ratio l a m is 1 ', then the (
Equation 1) is expressed as equation (2).

L (t) and R (t) are the L of commercially available music sources.
channel and R channel, so in most sources, the localization of each instrument is performed by level difference control between the L channel and R channel, so there is a difference between analog outputs S6 and 311 at a certain time. ) and the second (
4) There are only two conditions shown in formula.

L=R...(3) L'qR...(4) Substituting the condition of equation (3) into equation (2) results in the following.

56=O, 511=0 In other words, the output signals of both channels disappear.

Next, the condition of equation (4) is temporarily set as the condition of equation (5).

L=2R (5) Substituting this equation (5) into equation (2) yields the following.

56=R 311=-R In other words, the output signals have an opposite phase relationship. In other words, the fifth
In the case of the configuration shown in the figure, the output signals have an antiphase relationship with each other, and the reproduced sound gives a strong sense of discomfort.

Problems to be Solved by the Invention However, in the conventional example shown in FIG.
Since the output signal is 1 channel even though it is 5 channels, stereo playback is impossible, and if the conventional example shown in Figure 4 shown in Figure 5 is changed to 2 channel playback, it will feel strange The drawback was that I felt a strong sense of discomfort.

In view of the above problems, the present invention provides audio signal processing that enables two-channel stereo reproduction by using a high-pass filter (hereinafter abbreviated as HPF), a low-pass filter (hereinafter abbreviated as 1.PF), an audio removal means, and an adder. It provides equipment.

Means for Solving the Problems In order to solve the above problems, the present invention provides a first and a second HP
F, first, second and third LPFs, and the first LPF
audio signal removing means receiving the output signal of the first HPF and the output signal of the second LPF; phase inverting means receiving the output signal of the audio signal removing means; and an output signal of the first HPF and the audio signal removing means. a first adder for adding the output signal of the means and the output signal of the third LPF; and a second adder for adding the output signal of the second LPF, the output signal of the phase inverting means, and the output signal of the third LPF. It consists of an adder.

Operation The present invention has the above-described configuration, and input signals are input to LPF and HP.
F into two frequency bands, and only the output signal of the LPF is input to the audio removal means. Therefore, the sense of discomfort caused by the audio removal means is limited to the output signal of the LPF, and is not caused by the output signal of the HPF. does not occur, so
It has the advantage of being able to reproduce stereo sound with less discomfort.

Embodiment An embodiment of the present invention will be described with reference to FIGS. 1 to 3. FIG. 1 is a block diagram of an audio signal processing device according to an embodiment of the present invention. In Figure 1, 1.2 is the input terminal, 7
．． 12 is the output terminal, 13° and 14 are the first. second HPF,
15, 16.19 is the 1st. Second. 3rd glue, PF, 17
20, 21.2 are audio signal removing means; 18 are phase inverting means for inverting the phase of the output signal of the audio signal removing means 17;
2 is an adder. Note that the same components as in the conventional example are given the same numbers.

First, the analog signal S1 input to input terminal 1 is 3
After being branched, the first signal is input to the first HPF 13 to obtain an analog signal S12, and the second signal is input to the second LPF.
16 to obtain an analog/gu signal 513, and a third signal is input to an adder 22. On the other hand, after the analog signal S2 inputted to the input terminal 2 is branched into three, the first signal is inputted to the second HPF 14 to obtain the analog signal S14,
The second signal is input to the second LPF 16 to obtain an analog signal S15, and the third signal is input to the adder 22 to obtain an analog signal S16.

After branching the analog signals 813 and 315 into two, the first signal of the analog signal S13 and the analog signal 315
The first signal of 317 is input to the audio signal removing means 17 to obtain an analog signal S17, and the first signal after branching the analog signal 317 into two is input to the phase inverting means 18 to invert the phase with respect to the input signal. An analog signal 318 is obtained.

The output signal sis of the adder 22 is input to the third LPF 19 to obtain an analog signal 319. The analog signal 312, the second signal after the analog signal is divided into two, and the first signal after the analog signal 319 is divided into two are input to the adder 20, and an analog signal S2o is obtained at the output terminal 7.

On the other hand, the analog signal 814, the analog signal 818, and the second signal of the analog signal 319 are input to the adder 21 to obtain the analog signal 321 at the output terminal 12.

FIG. 3 shows the 1st section shown in FIG. Second LPF15, f
6 and 1st. Second HPF13.14 and third LPF
The input/output frequency characteristics of 19 are shown. That is, the curve Y1 is the first. The second HPF is 13.14, and the curve Y2 is the first. Second LPFls.

16. Curve Y3 is the input/output frequency characteristic of the third LPF 19. Since the main idea of the invention is based on these filters, they will be explained in more detail below.

In the embodiment of the configuration shown in FIG.
The frequency bands of 1 and S2 are from 10Hz in Fig. 3.
This is a frequency band indicated by FO of 20 KHz, and since it is an audio frequency band, a strong anti-phase window occurs in the entire frequency band.

Therefore, we divide the entire audio frequency band FO into frequency bands F1 and F6 from 10Hz to 5kHz, with the boundary at approximately 5kHz.
KHz ~ 20 KH2 (7) The input signal S shown in FIG.
1 and S2, it is possible to reduce the negative phase window of the reproduced sound.

That is, if we explain the case where equation (6) is substituted into equation (2), the results will be different in the case of frequency band F1 and in the case of frequency band F2, as shown in the table below.

As mentioned above in the table, it can be seen that in the case of the frequency band F1, the phase is reversed, but in the case of the frequency band F2, the phase is not reversed.

Now, in order to reduce the number of negative phase windows, the upper limit of one frequency band F1 can be made smaller, but at the cost of this, the effect of sound removal by the sound signal removal means becomes smaller. Through experiments, the inventor found that the upper limit is between 6 and 6 KH2.

That's the first thing. Second HPF13,14. 1st ° 2nd L
This is an explanation of PF15.18.

Next, a description of the function of the third LPF 19 will be given below. When substituting the condition of equation (3) into equation (2), 56=
We have already mentioned that O, Sl 1=O, but then,
Not only the audio signal but also the bass instrument signal contained in the input signals S1 and S2 will disappear. Since the audio signal does not include a frequency band below about 10oHz, the input signals S1 and S2 are added by the adder 22, and the curve Y3 shown in the figure is added to the third LPF 1 having input/output frequency characteristics.
9 has a function of passing only the frequency band F3 and enhancing the volume of the low range.

Figure 2 shows 1 the audio signal removal means 1 shown in the embodiment of Figure 1;
7 is constructed from the analog delay circuit, attenuator, and adder shown in FIGS. 4 and 5. Note that the first analog delay circuit can be configured not only by a BBI) but also by a phase shifter circuit. Also, it goes without saying that the audio signal removing means 17 shown in the embodiment of FIG.
It is also possible to configure it with a digital memory such as an AM (random access memory) or a shift register, an attenuator, and an adder. Furthermore, it is also possible to ignore the time control function and configure the audio signal removing means 17 with an attenuator and an adder without an analog delay circuit. In this case, the effect of removing or attenuating the sound may be reduced.

Effects of the Invention According to the present invention, it is possible to provide an audio signal processing device that outputs a 2-channel signal obtained by removing or attenuating the audio signal contained in a 2-channel music source.

[Brief explanation of drawings]

FIG. 1 is a block diagram of an audio signal processing device according to an embodiment of the present invention, FIG. 2 is a detailed block diagram of main parts of the same embodiment, FIG. 3 is an input/output frequency characteristic diagram of LPF and HPF, and FIG.
The figure is a block diagram of a conventional example, and FIG. 6 is a block diagram of a two-channel configuration of the conventional example. 1.2...Input terminal, 7.12...Output terminal, 13...First HPF, 14.・・・・・・
-Second HPF. 15...1st LPF, 16...Whistle 2 LPF
. 17...Audio signal removal means, 18...
Phase inversion means, 19...Third LPF, 20.2
1.22-・・・Adder.

Claims (1)

[Claims] first and second input terminals, a first branching means for branching the input signal of the first input terminal into three, a second branching means for branching the input signal of the second input terminal into three; a first high-pass filter means that receives as an input the first output signal of the branching means; a first low-pass filter means that receives the second output signal of the first branching means; and a second branching means. the first of
a second high-pass filter means to which the output signal of the second branching means is input; a second low-pass filter means to which the second output signal of the second branching means is input; and a third output signal of the first branching means. and a third output signal of the second branching means, an output signal of the first low-pass filter means, and an output signal of the second low-pass filter means. a third branching means for branching the output signal of the audio signal removing means into two; a phase inverting means receiving the first output signal of the third branching means;
A third low-pass filter means that receives the output signal of the first addition means, a fourth branching means that branches the output signal of the third low-pass filter means into two, and an output signal of the first high-pass filter means. a second adding means for adding the second output signal of the third branching means and the first output signal of the fourth branching means and outputting the added signal to the first output terminal; third adding means for adding the output signal of the second high-pass filter means, the output signal of the phase inverting means, and the second output signal of the fourth branching means, and outputting the added signal to the second output terminal; An audio signal processing device comprising: