JPS60214490A - Editing method of recording and reproducing device of digital signal - Google Patents

Abstract

PURPOSE:To prevent the deterioration of a signal and the occurrence of strange sounds before and behind an editing point by editing signals after numbers of sampled sound signals of simultaneous fields of signals reproduced by two units of VTR are controlled with the aid of sample number information of the field and caused to be equal. CONSTITUTION:In order to sample a sound signal at 48kHz and record it in a VTR of an NTSC system, samples are distributed by five fields, that is, 800, 801, 801, 801 and 801, which are recorded. While two units of VTR are simultaneously operated in the upstream of an editing point, sample number information of the field is taken out from a signal reproduced by a VTR2, and the field of 800 sample is retrieved. Simultaneously the sample, number information is taken out from the signal reproduced by a VTR1. In case of 801 sample, 801-numbered sample signal is provisionally stored in a memory, and each sample is delayed by one sample in the next field. After the processing is repeated and sample numbers of sound signals in respective simultaneous fields of the signals reproduced by the VTR1 and the VTR2 are caused to be equal, editing is carried out.

Description

DETAILED DESCRIPTION OF THE INVENTION Field of the Invention The present invention relates to an editing method for an apparatus capable of recording and reproducing digital audio signals or video signals and audio digital signals on a VTR.

Conventional configuration and its problems Conventionally, audio signals are sampled at a certain sampling frequency,
When recording a digital audio signal obtained by quantization on a VTR, the sampling frequency is selected to be an integral multiple of the vertical synchronization signal, and a fixed number of digital samples are recorded in one vertical scanning period (hereinafter referred to as a field). Consideration is given to recording audio signals. For example, in the case of the EIAI (Japan Electronics Industries Association) standard NTSC system used in PCM processors, the sampling frequency is selected to be 736 times the vertical synchronization signal, and 736 times the vertical synchronization signal is used.
35 samples are recorded.

By the way, recently, the sampling frequency of the audio signal in mode β of satellite broadcasting has been changed to 4s KHz, and
48 KHz has become an extremely important frequency, with the sampling frequency of commercial recording machines becoming fixed at 48 KHz. However, this frequency is NT
It is not an integral multiple of the SC system vertical synchronization signal (60 x 1001 H2). However, since there are 4004 samples in units of 5 fields, it is only necessary to divide the data into 800 samples for one field and 801 samples for the other 4 fields. A recording and reproducing method when the number of recording samples differs between fields in this way will be described. FIG. 1(a) shows an audio signal converted into a digital signal 5iy1.

WK indicates the second sample signal. The digitized audio signal is first divided into fields. Now, W1
~W800 8o○ sample in the first field, W8o
The 801 samples of 1 to W1601 are the second field, hereinafter 801.801.801.800,801...
・Divide in the order of. Next, the signals are rearranged in field units (hereinafter referred to as interleaving), error correction codes are added, and this signal sequence is divided into synchronization signals (S
YNC), an address signal (ADD), and an error detection code (shortened cyclic code, CRCC) are added to form a block. In FIG. 1, W' indicates the second sample after interleaving, and Pl and Ql indicate error correction codes. By the way, if one block is composed of 6 samples, 8
Fields of oO sample and 801 sample are B1 to B.
Although 134 blocks are required, there is a shortage of samples for the 134th block. Therefore, before performing the above-mentioned interleaving process, etc., a dummy signal (shaded area) is added as shown in FIG. 1(b) to create a signal l1i22 in which each field consists of 804 samples. After adding the dummy signal, the above-described processing is performed to form a block configuration as shown in FIG. 1(C).

FIG. 2 shows the conversion from 5i21 in FIG. 1(a) to 5iy2 in FIG. 1(b). 1 is a memory section, and 2 is an address generation section. First, load the sea address generator 2.
The write address of f1 is returned to the initial value, and the write clock f1
The address is increased and 5ip1 is written to the memory in the memory section 1. If the audio signal is a SOO sample field, after writing 800 samples, write 4 samples,
If it is a field of 801 samples, after writing 801 samples, dummy signals for 3 samples are written. When reading a signal, the read address of the address generator 2 is returned to its initial value in Load2, and the address is increased in read clock f2, and 804 samples including the dummy signal are read out regardless of the field.

FIG. 3 shows the timing of Load 1 and Load 2.

The vertical synchronization signal of Load 2 and the signal of Load 1 are synchronized in units of 5 fields, and the pulse interval (T) of Load 2 and the pulse interval of Load 1 (field T1 of 800 samples, field T2 of 801 samples) are as follows. There is a relationship.

5T2T1+4T2 (T1\T2) ...(1)
Therefore, in Figures 1) and (b), the samples at the beginning of the fields are aligned every 5 fields as shown in Figure 3, but in other fields they are shifted by ΔT, 2ΔT, 3ΔT, and 4ΔT, respectively.

ΔT, = T-T ・・・・・・・・・・・・・・・−
(2) Conversion from 5ip1 to 5122 is performed as described above. Thereafter, blocks are constructed as described above and recorded on tape.

During playback, error detection is first performed in units of pronks using CRCC, errors are corrected using an error correction code based on the error detection information, and then converted to 5iy2 as shown in FIG. 1(b). Conversion from 5iy2 to 5iy1 can be accomplished by reversing the writing and reading operations in the memory in FIG. However, since it is not known which field is the 8o○ sample or the 801 sample of the audio signal, it is necessary to input sample information of the field in the dummy signal in advance, for example, at the time of recording. And from 5iy2 to 5iy1 during playback
During conversion, a Loadl signal is generated from the sample information and a dummy signal is removed, and the signal returned to 5i71 is converted into an analog signal.

Next, the editing method will be explained using FIG.

Consider a case where two VTRs (VTR1, VTR2) are used to record the 5th field and subsequent fields of tape #1 after the 100th field of tape #2. Without switching, the VTRs were played separately in advance and tape #1. Determine the editing point for tape #2. After the determination, the VTR 1 is operated in synchronization with the vertical synchronization signal of the VTR 2 so that each edit point matches the editing point, and both VTRs reproduce the signal using the advance reproduction head. The distance between the advance playback head and the recording head is such that the playback signal passes through the signal processing circuit of the playback system and is reproduced in its original time sequence, and then the signal passes through the signal processing circuit of the recording system again and is recorded on the same tape. has been adjusted. Therefore, even if the reproduced signal is recorded while being reproduced by the advance reproduction head, there will be no problem with the sound connection 9. At the editing point, a certain section, tape #1. The reproduced signals from tape #2 are individually weighted and added together (hereinafter referred to as cross-fade), and at the same time, this signal is recorded on the VTR 2. At this time, as shown in Figure 4, the cross-faded signal is recorded on tape #2 after the first o○ field, and the signal of tape #1 played back on VTR 1 is then recorded and edited. It ends.

Now, the two VTRs are operated synchronously and the signal sequence with the dummy signal added shown in Fig. 1(b) is reproduced as shown in Fig. 5(a).
Shown below. If the number of audio signal samples is the same in each field, there is no problem with editing at this point, but as shown in Figure 5(a)
If the number of samples is different as shown in , if the 5th field of tape #1 continues after the 100th field of tape #2, 800, 800, 801 will be generated before and after the edit point.
, 801, --, and a temporal disturbance occurs. Furthermore, in order to prevent temporal disturbances, interpolation of one sample in the fifth field is considered, but this is not particularly desirable as it degrades the signal. It is also conceivable to edit after returning to the signal sequence after removing the dummy signal in Fig. 1(a), but as shown in Fig. 5(b), VTR1 and VT
Load 1 of R2 is ΔT and 4ΔT, and the timing difference differs depending on the field. Therefore, a time lag occurs before and after the editing point, and abnormal noise may also occur at the editing point.

As described above, even when editing when audio signal samples are different in each field, occurrence of temporal disturbances, signal deterioration, etc. before and after the editing point poses serious problems. In particular, the greater the number of edits, the greater the problem.

OBJECTS OF THE INVENTION The present invention solves the above-mentioned problems, and provides an editing method that prevents signal deterioration and abnormal noise caused by editing, and makes the connections between sounds before and after the editing point natural.

Structure of the Invention The present invention converts an audio signal into a digital signal and converts it into a VTR.
If the sampling frequency of the audio signal is not an integer multiple of the vertical synchronization signal and the number of recording samples in each field is different, the number of samples in the field will be displayed in the audio signal area. The information is input and recorded together with the audio signal, and when editing, the two VTRs are operated synchronously and one of the reproduced signals is appropriately shifted using the sample information, so that the number of samples of the audio signal in each field is equal to the number of samples of the audio signal in each field. By editing so that the signal is equal to , it is possible to perform editing without producing abnormal sounds or deteriorating the signal before and after the editing point.

DESCRIPTION OF EMBODIMENTS The present invention is applied to an apparatus for converting an audio signal into a digital signal at a sampling frequency of 48 KHz and recording and reproducing it on a VTR. An embodiment of the present invention will be described below.

When sampling an audio signal at 48 KHz and recording it on an NTSC VTR, the sampling frequency is VT.
Since it is not an integral multiple of the R vertical synchronization signal, 800, 801, 801, 801 is generated in 6-field units.
, 801 samples are distributed and recorded. The signal is interleaved in field units, an error correction code is added, the signal is divided into a certain length, and a synchronization signal, an address signal, and an error detection code are added to form a block structure.

At this time, if one block is composed of nine samples, dummy signals of four samples are added to the field of 8o○ samples, and three samples of dummy signals are added to the field of 8o1 sandals because fractions occur. Field sample number information is added to this dummy signal. For example, using 1 bit in the dummy signal, a field of 800 samples is 11
The sample number information may be ", 801 samples field is 0". The block-configured signals are the same as those shown in FIG. Then, this signal is recorded on a tape, and during reproduction, first, errors in blocks are detected by an error detection code, and based on the error information, the errors are corrected by an error correction code.

After correction, the signal is dinterleaved and the first
Returning to the time series shown in Figure (b). Here, since dummy signals are still included, in order to remove the dummy signals and restore the original time axis, writing and reading are performed using a memory at different frequencies as shown in FIG.

An editing method when two VTRs are used to record the signal of tape #1 of VTR1 from the middle of tape #2 of VTR2 as shown in FIG. 4 will be described. First, two VTRs
are played separately to determine the edit point. After determining the editing point,
Going back from the editing point, the VTR 1 is run in synchronization using the vertical synchronizing signal of the VTR 2 so that each editing point coincides with the VTR 1. At this time, both of the two VTRs reproduce the audio signal recorded on the tape by the advance reproduction head provided upstream of the recording head. At this time, the reproduced signal after dinterleaving is the same as that shown in FIG. 5(a).

Here, the editing method according to the present invention will be explained with reference to FIG. Two VTRs from a position further back than the editing point
While operating synchronously, output the field sample number information from the playback signal of tape #2 from VTR2.
”, that is, the field with 8oo samples.

This is the case here for the 100th field. In FIG. 6, the numbers in parentheses in each field are the number of samples of the audio signal. At this time, the VTR 1 simultaneously outputs field sample number information in the reproduced signal.
If it is a sample, no special processing is performed below, and if the field where the sample number information is "0", that is, 801 samples, the 801st sample signal is temporarily stored in the memory, and the sample information of that field is stored in "O". ” to 1°” and insert a dummy signal at the 801st sample location. In the case of FIG. 6(a), the sample number information in the fourth field is changed to "1" and the 801st sample is temporarily stored in the memory.

It is then read out from memory as the first sample of the next field. Therefore, in the fifth field, each sample is delayed by one sample.

And if this field is 801 samples, 801
The th sample is also stored in the memory and this process is repeated sequentially. Then, when the field of 800 samples arrives, the sample information of that field is changed from 1'' to "0", the sample is delayed one sample at a time, and the previous 800th sample is placed in the 801st dummy signal area. It should be avoided to include information such as sample number information in the dummy signal.Next, the VTR2
Search the field of 800 samples from the playback signal of
Repeat the above process. The results are shown in FIG. 6(b). After making the number of audio signal samples in each field at the same time of the signals reproduced from pVTRl and VTR2 the same as described above, by editing,
Editing can be performed in the same way as when the sampling frequency is selected as an integral multiple of the vertical synchronization signal.

In the above manner, using an NTSC VTR, 48
Audio signals sampled at KHz can be edited without signal deterioration, temporal disturbance, or abnormal noise before and after the editing point.

Further, in the above-described embodiment, one block is composed of 6 samples of the audio signal, but the present invention can also be applied to cases where the block is composed of 7 samples, 8 samples, etc. 1
If the block is composed of 7 samples, the dummy signal area is the SOO sample field, which is 5 samples, 8.0
One sample field may correspond to 4 samples, and each field may have 806 samples. When one block is composed of 8 samples, the number of samples in each field is 808 samples, and only the size of the dummy signal area changes. Therefore, in the present invention, one block is
It is applicable not only to samples but also to any number of samples.

Effects of the Invention The present invention operates two VTRs in synchronization, and when cross-fading and editing, uses the signal sequence before removing the dummy signal to compare fields at the same time of the signals reproduced by the two VTRs. By controlling the number of samples of the audio signal using the sample number information in the field to make the same number, and then editing, it is possible to prevent signal deterioration and generation of abnormal sounds before and after the editing point.

[Brief explanation of the drawing]

Figure 1 is a signal diagram showing an example of the block configuration when an audio signal is sampled at 4aKHz, Figure 2 is a block diagram for explaining the conversion of the signal in Figure 1, and Figure 3 is a diagram showing the operation of Figure 2. Figure 4 is an explanatory diagram that explains the conventional editing method. Figure 6 is an explanatory diagram that explains the problems with the conventional editing method. Figure 6 is the book. FIG. 3 is an explanatory diagram for explaining an editing method according to the invention. Name of agent Patent attorney Satoshi Nakao (1 male) Figure 1 Figure 5FI4 Figure 6

Claims (1)

[Claims] When converting an audio signal into a digital signal and recording it on a video tape recorder, or when recording a video signal together with the digital signal on a video tape recorder, the sampling frequency of the audio signal is not a frequency that is an integral multiple of the vertical synchronization signal of the video signal. , after distributing the number of samples of the audio signal to each field (vertical scanning period), adding a dummy signal containing sample information of each field to make the number of samples in each field the same, rearranging the audio signal in field units, An error correction code is added, and a synchronization signal and an error 9 detection code are added at intervals of a certain length to determine the block structure, and then recorded on tape. At the time of editing, after determining the editing point,
Two video tape recorders are operated synchronously starting from the edit point, and the two video tape recorders are run so that the edit points coincide, and at that time, the number of samples in the field is used to calculate the number of samples of the two video tape recorders. When the number of samples of the audio signal in the field at the same time is not always the same, the audio signal played back from one video tape recorder is sequentially delayed, and at the same time, the sample number information in the field is rewritten, An editing method for a digital signal recording and reproducing device that performs editing by always keeping the same number of samples of reproduced audio signals at the same time.