JPS6058545B2 - Digital signal processing method - Google Patents

Description

[Detailed description of the invention] Only the necessary parts are extracted from a plurality of recorded music signals, and they are connected (connected) in a state that satisfies musical continuity to create a complete It is well known that the process of converting emotional music signals into music signals, so-called music signal editing operations, has been widely carried out in the past as essential for obtaining music signals in desired and complete states.

Figure 1 shows a magnetic tape Ta on which a plurality of music signals #1 to #3 are sequentially recorded, and necessary portions of each of the music signals #1 to #3 are continuously recorded on the magnetic tape. This diagram also explains the outline of the editing work when the editing work is performed. In this figure, the editing work is performed to edit the portion of the music signal #1 recorded on the magnetic tape Ta from Po to P on the magnetic tape Tb. and P in the music signal #2
2→Po part and Po→P5 in the music signal of #3
The example shows a state in which the parts are recorded in such a way that they are sequentially and continuously connected.

By the way, when editing music signals, since musical continuity must not be lost at the connection points of the music signals, waveform processing at the connection points is required.

Figures 2a to 2c are waveform diagrams for explaining the waveform processing at the connection point of two music signals performed when editing music signals, and Figure 2a is a waveform diagram shown in Figure 1. 1 is a diagram illustrating an example of a waveform near a connection point P in a music signal;
Figure 2 shows an example of waveforms near the connection point P in the music signal #2 shown in Figure 1, and furthermore, Figure 2 c
The figure shows connection point P in the music signal #1 shown in Figure 2a,
2 is the connection point Po in the #2 music signal shown in FIG.
FIG. 3 is a waveform diagram in a state in which the The curve shown by the solid line in FIG. In the waveform shown by the solid line in Figure c, a discontinuous state appears at the connection point of the two music signals, which is undesirable. Therefore, when editing, the waveform near the connection point (editing point) is As shown by the dotted line in Figure 2c, waveform processing is performed so that the waveform near the connection point becomes a waveform that satisfies musical continuity. By the way, in recent years, music signals have been recorded and played back as PCM signals in order to reproduce the high fidelity of the lower layer of music signals. The aforementioned editing operations are still required to obtain the desired complete musical signal.

When a music signal is recorded on a magnetic tape in the form of a PCM signal, the two signal parts including the edit point played back from the magnetic tape are in a good continuous state from a musical point of view. PCM of the two signal parts including the above-mentioned edit points so that they are connected at the edit points.
The signals are individually operated so that one of them is faded in and the other one is faded out and then summed. However,
In order to successfully edit a music signal, the editing points in the two signal parts to be edited must be set appropriately, but conventionally, rehearsals to find the appropriate editing points were performed from magnetic tape. Since this was done using the reproduced signal that was being reproduced, it was difficult to find an appropriate editing point.
The present invention provides a digital signal processing method that can solve the conventional problems as described above. The content of the system of the present invention will be specifically explained in detail below.

Now, the two music signals to be edited are music signals #1 and #2, and a predetermined number of -data near the editing point (editing location) of the music signal #1 are WAO, WA,
,...WApi...WAm, and assuming that the predetermined number of data near the editing point of the music signal #2 is WBO, WBl,...WBJ...WBn, the above In the interval where two music signals #1 and #2 have one data, one music signal #1 fades out and the other music signal #2 fades in, resulting in the above-mentioned #1 and #2 music signal. When two music signals are edited,
Fade-in of the two music signals #1 and #2 mentioned above,
Data WCK in the fade-out section (hereinafter sometimes referred to as fade section) (K = 0
, 1, 2...1) is expressed by the following equation (1). In the above equation (1), AK and bK are coefficients when each signal is subjected to fade-in processing and fade-out processing. For example, if music signal #1 is faded in in l data intervals, , #2 is faded out in one data section, the value of the coefficient AK changes in l steps from "゜1゛" to "0゛" in the fade section, and the coefficient The BK value is from 460 to 64 in the fade section.
It changes to F3 in l stages.

The two coefficients AK and bK described above are often related to each other as shown in the following equation (2) during normal editing. Now, the digital signal processing method of the present invention uses a predetermined number of pieces of data in the vicinity of the editing point in two music signals to be edited, for example, data WAO, WAl, WAl, and WAl of one of the music signals. ...WAm and the data WBO, WB, ...WBn of the other music signal are each stored in the memory independently, and each data of the two music signals stored in the memory is individually and arbitrarily stored. A predetermined number (for example, one) of data in the vicinity of the selected editing point is sequentially read in a predetermined order, and a predetermined level control is applied to each of the sequentially read data. After that, the corresponding data in each music signal is added to obtain a continuous digital signal, so the starting point of editing for the two music signals to be edited is now in memory. Data stored in WAO, WApi...WAi
...WAm is data WAi, and data stored in the memory WBO, WBl, WB,
. . . Assuming that W8n is data WBj, the output data string according to the method of the present invention after the necessary fade processing is performed on the data sequentially read from the memory is as shown below. Here, since the above-mentioned output data string has output data other than the fade section added before and after the output data Wc in the fade section shown in equation (1), equation (1) can be expressed as follows. Expand and think about the output data W.

When K is expressed, it becomes as shown in the following equation (3). WOK=AK
? WAK+BKIwB(K+α) 160 (3) And in equation (3), α is represented by α=j-1, and the conditions for coefficients AK and bK in equation (3) are K
The value of can be determined as follows. (a) The value of K is in the range of 0 to (1-1) (b) The value of K is in the range of i to (1-1)
For the range (1-1+l), AK is a value from 1 to 0, BK is a value from 0 to 1, and (c) K is a value from (1+
I to P), AK=0, bK=1
FIG. 3 is a block diagram of one embodiment of the digital signal processing method of the present invention, in which 1 is a control circuit, 2 and 3 are memories, 4 and 9 are data selectors,
5 is a multiplier, 10 and 11 are registers, 12 is an adder 13
is a coefficient memory, and the control circuit 1 supplies control signals to each component of the device at predetermined timings to control the operation of the device.

Memory 2 stores data of one of the two music signals to be edited, and memory 3 stores data of the other of the two music signals to be edited. be done. In FIG. 3, for ease of explanation, two memories 2 and 3 are shown as being used as memories, but each data of the two music signals to be edited is , may be independently stored in different address areas in one memory.

The data selector 4 sequentially and alternately sends the data of the two music signals read individually from the memory 2 and the memory 3 to the multiplier 5 under the control of the control signal from the control circuit 1.
A control operation is performed to send the multiplicand signal to the multiplicand signal input terminal 6.

The multiplier 5 multiplies the data of the music signal supplied to the multiplicand signal input terminal 6 by the coefficients AK and bK from the coefficient memory 13 supplied to the multiplier signal input terminal 7, and sends the output signal to the output terminal 8. The data is applied to one of the registers 10 and 11 via the data selector 9.

Since the data selector 4 and the data selector 9 switch signals in synchronization, for example, when the data in the memory 2 is given to the multiplier 5, the output signal obtained as a result of multiplying it by a coefficient is When stored in the register 10 and applied to the data multiplier 5 of the memory 3, the output signal obtained as a result of multiplying it by a coefficient can be stored in the register 11.

The data stored in register 10 and register 11 is
The signals are added by the adder 12 and sent to the output terminal 14. In the apparatus shown in the third figure described above, data WAO, WAl, . . . Wam of one of the two music signals to be edited are stored in the memory 2, and Data WBO, WB of the other of the two music signals targeted for editing in memory 3
l...WBn is stored, and the data of two different music signals read from the memories 2 and 3 are sequentially and alternately selected by the data selector 4 and supplied to the multiplier 5 as a multiplicand signal. The control circuit 1 supplies control signals to each component so that the multiplier 5 is supplied with the required coefficient values AK, bK from the coefficient memory 13, and the output from the multiplier 5 is The signals are sent to specific registers 10 and 11, respectively, under the control of the control circuit 1.
When the data are stored individually and then added by the adder 12, the output data according to the equation (3) already described is outputted to the output terminal 14 by the device shown in the third diagram. It is clear that it can be done.

Incidentally, in actual implementation, the components consisting of the multiplier 5, data selector 9, registers 10, 11, and adder 12 in FIG. 3 can be replaced by a so-called ALU.

In the digital signal processing method of the present invention, two digital signals to be edited are individually stored in a memory, and edited output data is obtained by processing the data read from the memory. Therefore, the position of the edit point and the length of the fade section can be changed arbitrarily by simply changing the address for reading data from the memory, and the manner in which the coefficients AK and bK change can be freely controlled using the control circuit. This makes it easy to set editing points and set waveform processing methods near the connection points so that the two music signals to be edited can be well connected in a musically continuous state. According to the method of the present invention, before the music signal is recorded on the magnetic tape in an edited form, the appropriate editing point can be determined by easily repeating the trial listening (rehearsal) using electronic operations. It is also easy to find out, and according to the method of the present invention, the problems of the conventional method described above can be satisfactorily solved.

[Brief explanation of the drawing]

Fig. 1 is a plan view of a magnetic tape for explaining editing work, Figs. 2 a-c are explanatory waveform diagrams, and Fig. 3 is a block diagram of one embodiment of the digital signal processing method of the present invention. be. 1... Control circuit, 2, 3...
Memory, 4, 9... Data selector, 10, 1
1...Register, 12...Adder, 1
3...Coefficient memory.

Claims (1)

[Claims] 1 After converting the music signal from analog to digital, the first information signal and the second information signal, which have been encoded in a predetermined encoding format and are to be connected to each other, are each independently stored in a memory. and sequentially read a predetermined number of data in the vicinity of each arbitrarily selected editing point of the first and second information signals stored in the memory in a predetermined order. , each of the sequentially read data is subjected to predetermined level control, and then mutually corresponding data of the first information signal and the second information signal are added,
A digital signal processing method that allows continuous digital signals to be obtained.