JPS63152061A - Sound signal recorder - Google Patents

Abstract

PURPOSE:To broaden the recording and reproducing band of a sound signal and to reduce quantization noises by executing coding with an ADPCM (adaptive difference PCM) system. CONSTITUTION:The sound signal can be satisfactorily recorded and reproduced on a magnetic tape 2 by executing with the PCM system, but the recording and reproducing band of the sound signal is narrow such as 15.75kHz and the quantization noise becomes large since the sampling frequency in an A/D converter is set in 31.5kHz and one sample is set in 10 pits. Then by executing coding in the ADPCM system, the number of quantization pits can be reduced. Thus, the sampling frequency and the number of the quantization pits in the A/D converter 12 can be made large and the recording and reproducing band of the sound signal can be broadened and also the quantization noises can be reduced.

Description

DETAILED DESCRIPTION OF THE INVENTION [Field of Industrial Application] The present invention relates to an audio signal recording device that encodes an audio signal and records it on a recording medium as a digital signal.

[Summary of the invention]

The present invention provides an audio signal recording device that encodes an audio signal and records it on a recording medium as a digital signal.
By using the DPCM method, it is possible to record and reproduce a wider band audio signal, and at the same time, it is possible to reduce quantization noise.

[Conventional technology]

As one of the rotary head VTRs, as shown in FIG.
Two rotating magnetic heads H arranged with an angular spacing of °
A and HB are scanned, and as shown in FIG. 6, one part of the video signal is located in the bone area AP in an angular range of about 36 degrees from the point at which the rotating magnetic fields HA and HB start scanning in the recording trunk.
It is proposed to convert the audio signal related to one field into PCM and record it in a compressed time-axis state, and then record the video signal for one field in the 180° angular range bone area AV. has been done.

FIG. 7 shows an audio signal recording and reproducing system of such a VTR.

In the figure, the left signal SL and right signal SR of the audio signal are sent to the A/D converter (33) through input terminals (31L) and (31R) and a pre-emphasis circuit (32).
and converted into a digital signal. In this case, the sampling frequency is, for example, 31.5K) Iz, and one sample is 10 bits, and the left signal SL and right signal SR are sampled.

The 10-pin digital data from this A/D converter (33) is converted into 8-bit digital data by a 10-8 conversion circuit (34), and then sent to a signal processing circuit (35).
supplied to

In the signal processing circuit (35), processing such as interleaving and addition of error correction/detection codes is performed.

In this case, a CRC code is used as an error detection code, and a simple parity (P, Q) code is used as an error correction method.
A cross-interleave method using

Furthermore, as shown in Figure 8, the data per field consists of 132 blocks #0 to #131, and these 132 blocks are interleaved by being distributed and arranged in three zones. There is. That is, the ninth
The figure shows the data arrangement of data for one field.

One word has 8 bits, and m x n = 1056 words (m = 8 words, n = 132 broncs).

NT when sampling frequency is 31.5KHz
The digital data for one field in the SC method is 105
Since it is 0 word, the discrimination data IDo is 6 words.

IDz, . . . IDs will be added. In other words, (Lo, Ro, LL, R1, L2.R2,...

Ls2i, R523, L524, R624
) and the above-mentioned 6-word discrimination words IDo to IDs are added to the beginning of one field of continuous digital data. 10 including 6 words of discrimination data IDo=IDs
The 56 words of data are arranged at intervals of 44 blocks in the horizontal direction every two words. In hardware, data is written to addresses separated by 44 blocks by RAM address control. Apart from control data or parity data, (Li, Ri) (i=,
Two words (0 to 524) are arranged horizontally. The reason why the digital data is interleaved by dividing into three in the horizontal direction is to increase the burst error length that can be corrected, for example, by interpolating the average value. In particular, (Li, Ri)
By arranging them in the horizontal direction, the correction length can be made longer than by arranging them in the vertical direction.

Two parities, for example, even parity, are added to this one field of audio data and discrimination data. The digital data series of each row of the above configuration is 'iJ
6 + W 1+ ..., W? Then, a first parity sequence P is formed from 8 words belonging to each data sequence separated by a horizontal distance of 14 or 15 blocks. In addition, the digital data series W o ” W
? and the parity series P! A second parity sequence Q is formed from nine words taken at a distance of 12 blocks from each of the sequences of Ill[I. This first
The parity sequence P of is placed in the center within the l block, and
The parity series Q is arranged at the end of one block. In other words, the data at the center position within one block is
Since there is a high probability that errors cannot be corrected, a parity sequence P is arranged that is less important than the audio data, and the parity sequence P is set to maximize the distance between two words that generate this parity sequence P. Q is placed at the end of one block.

Each of the 132 blocks includes 8 words of digital data and 2 words of parity data, and a 16-bit CRC code for error detection, for example, is added to the data of each block, and a 3-bit CRC code is added to the data of each block. A block synchronization code of 8 pins and a block address code of 8 pins are added.

Further, from the signal processing circuit (35), a first block,
The digital data is outputted in the order of the second block, . . . , the 132nd block, and is supplied to the time axis compression circuit (36). is time-axis compressed to the area AP mentioned above. In this case, the time compression rate is, for example, 1/6
．． 84, and the digital data output from this time axis compression circuit (36) is 5.8 ps.

The digital data output from this time axis compression circuit (36) is subjected to, for example, bi-phase modulation in a modulator (37), and then supplied to rotating magnetic nodes HA and HB via a recording amplifier (38). , recorded on the magnetic tape (2).

Also, from the magnetic tape (2), the rotating magnetic heads HA and H
The digital data reproduced by B is sent to the reproduction amplifier (39).
The signal is supplied to the demodulator (40) via the . This demodulator (4
The time axis of the digital data adjusted to 11L by 0) is expanded to one field time in a time axis expansion circuit (41), and then supplied to a signal processing circuit (42). This signal processing circuit (42) includes dinterleaving, error detection,
Processing such as error correction is performed.

The 8-bit digital data output from the signal processing circuit (42) is converted into 10-bit digital data by an 8-10 conversion circuit (43). The output of this 8-10 conversion circuit (43) is converted into an analog signal by a D/A converter (44) and then supplied to a de-emphasis circuit (45). The left signal SL and right signal SR of the audio signal are obtained at the output terminals (46L) and (46R) derived from this de-emphasis circuit (45), respectively.

[Problem that the invention seeks to solve]

In this way, in the example shown in FIG. 7, the audio signal can be converted into PCM and can be recorded and reproduced satisfactorily on the magnetic tape (2).

However, according to the example in FIG. 7, the sampling frequency in the A/D converter (33) is 31.5 KHz.
In addition, since each sample is 10 bits, the audio signal recording/reproducing band is as narrow as 15.75KH2, and the quantization noise is large.

In view of these points, the present invention aims to widen the recording/reproduction band of audio signals and reduce quantization noise.

[Means for solving problems]

In order to solve the above-mentioned problems, the present invention performs encoding using ADPC.
This is performed using the M (adaptive differential PCM) method.

[Effect]

By using the ADPCM encoding method, the number of quantization bits can be reduced. Therefore, the sampling frequency and the number of quantization bits in the A/D converter can be increased, the recording/reproduction band of audio signals can be widened, and quantization noise can be reduced.

〔Example〕

An embodiment of the present invention will be described below with reference to FIG. This example is an example that is applicable to a rotary head type VTR that records audio signals as digital signals on a magnetic tape. In FIG. 1, parts corresponding to those in FIG. 7 are designated by the same reference numerals, and detailed explanation thereof will be omitted.

In the same figure, the left signal SL and right signal SR of the audio signal are input to the A/D through input terminals (ILL) and (IIR).
The signal is supplied to a converter (12) and converted into a digital signal. In this case, the sampling frequency is 48KH2, 1 sample is 16 bits, and the left signal SL
and the right signal SR are sampled. The 16-bit digital data from this A/D converter (12) is supplied to a band compression circuit (13) to compress the band.

This band compression circuit (13) is composed of, for example, an ADPCM encoder as shown in FIG. This ADPCM
Since the encoder has a conventionally well-known configuration, a detailed explanation will be omitted, but the data x input to the input terminal (131)
(n) is processed block by block.

In this case, in a VTR like this example, it is desirable to complete the data processing for each track, and the blocks are divided so that there is no excess data for one track, that is, one field, and evenly across all blocks. It is said to be a number that can be divided into That is, as mentioned above, since the sampling frequency in the A/D converter (12) is 48KH2, the amount of data for one field is 800 samples for each of the left signal SL and right signal SR, and a total of 1600 samples.
This is a sample. Therefore, in this example, one block of data is 40 samples. This allows data processing to be completed for each dock.

16-bit digital digital input terminal (131)
(n) is supplied, the prediction coefficient calculation circuit (132) uses digital data x (
Data correlation is determined from nl, and prediction coefficients are calculated. Then, the digital data Then, it is rounded to a certain number of bits, in this example 4 bits.

In this case, the error due to rounding is taken into account by the circuit (135), and rounding is performed in a manner appropriate to the range of data within the block using the range information from the intra-block maximum value detection circuit (136). After all, the output terminal (137)
A residual error (≧) rounded to 4 bits is obtained. In addition, the output terminal (138) has a 4×8 bit prediction coefficient (
When a fourth-order prediction filter is used, 4 coefficients are obtained, and 8-bit range information is obtained at the output terminal (139).

△ Therefore, the residual data d (11 is 4 bits, the prediction coefficient is 4×8 focus, and the range information is 8 bits, so
The average bit length is 4υ-5 feet/word, and 16-bit data is compressed to 5 bits.

Digital data from the band compression circuit (13) is interleaved by an interleaving circuit (14) and then supplied to an error correction encoder (15). In this encoder (15), processing such as adding an error correction code is performed.

In this case, as an error correction code, for example, GF (2@
) is used. At this time, 1
Word data is 8 bits.

As mentioned above, the digital data from the band compression circuit (13) is residual data, prediction coefficients for each block, and range information (hereinafter referred to as "side chain").Currently, the residual data is 4 bits, so 2 1 with residual data
The data for one track, and therefore for one field, is residual data: 800 words, side chain: 200 words. Here, since the side chain is more important, it needs to be protected more strongly, and is protected by writing 2ii. In the end, the format for data for one field is the third one.
What is shown in the figure is 2 (tri).

C1 is a (36,32) Reed-Solomon code, C2 is a (22,18) Reed-Solomon code, and C1 and 02 are a product code. Further, C3 performs error correction on the double-written portion of the side chain, and is a (22, 18) Reed-Solomon code.

The correction code redundancy in this case is as follows: The digital data to which the error correction code has been added by the encoder (15) is supplied to the time axis compression circuit (36) and is time axis compressed to 1/6.84. The digital data output from the time axis compression circuit (36) is modulated by a modulator (37) and then supplied to the rotary magnetic heads HA and (B) via a recording amplifier (38), and is then applied to the magnetic tape (2). ) is recorded.

In addition, from the magnetic tape (2), the rotating magnetic disks HA and H
The digital data reproduced by B is sent to the time axis expansion circuit (41
), the time axis is expanded to one field time and then supplied to the error correction decoder (16).

In this decoder (16), error detection and correction processing is performed using the above-mentioned Reed-Solomon code. The decoder (16) outputs the residual data and side chain as digital data, which is then passed through the dinterleave circuit (17) to the band expansion circuit (18).
).

This band expansion circuit (18) is composed of, for example, an ADPCM decoder as shown in FIG. Since this ADPCM decoder has a conventionally well-known configuration, a detailed explanation will be omitted, but the input terminal (181) receives 4-bit residual data Ω.
) is supplied to the input terminal (182), a 4×8 bit prediction coefficient is supplied to the input terminal (183), and an 8-bit prediction coefficient is supplied to the input terminal (183).
Bit range information is provided. Then, the residual data rounded by range information is converted back to range-independent data, and further predictions are made! (184) predicts the original data using the prediction coefficient, and outputs 1 to the output terminal (185).
6-bint digital audio data is output.

The 16-bit digital audio data from this band expansion circuit (18) is sent to the D/A converter (19).
A left signal SL and a right signal SR of the audio signal are obtained at output terminals (20L) and (2OR) derived from this D/A converter (L9), respectively.

This example is configured as described above, and includes a time axis compression circuit (36).
The number of bits of digital data output from is: Number of bits = sampling frequency x 2 x average bit length x correction code redundancy x time compression rate = 48 x 10' x 2 x 5 x 1.76x
6.84# 5.778Mbps.

Therefore, also in this example, digital audio data can be recorded and reproduced satisfactorily.

In this way, according to this example, since the encoding is performed using the It is possible to reduce noise.

In addition, according to this example, the side chain, that is, the prediction coefficient and range information, is written in 2M and is protected by adding an error correction code, so the side chain data, which is important in the ADPCM method, is sufficiently protected. can do.

In addition, according to this example, when dividing blocks in ADPCM processing, the data for one track, that is, one field, is divided equally among all blocks without any surplus, so there is no problem in dividing the continuous audio signal. can be played.

In the above embodiment, one block is made up of 40 samples, but it can also be made, for example, 32 samples.

In short, it is sufficient if the number can be divided evenly among all blocks.

〔Effect of the invention〕

According to the present invention described above, since encoding is performed using the ADPCM method, sampling frequency and via)
The number can be increased, the recording/reproduction band of audio signals can be widened, and quantization noise can be reduced.

[Brief explanation of the drawing]

Figure 1 is a configuration diagram showing one embodiment of the present invention, Figures 2 to 4
5 is a diagram for explaining the rotary head device, FIG. 6 is a diagram showing a recording track pattern, FIG. 7 is a diagram of a conventional example, and FIGS. 8 and 9 are diagrams thereof. It is a figure for explanation. (IIL) and (IIR) are input terminals, (12) is A
/D converter, (13) is a band compression circuit, (14) is an interleave circuit, (15) is an error correction encoder,
(16) is an error correction decoder, (17) is a dinterleave circuit, (18) is a band expansion circuit, (19) is a D/A
converter, (20L'') and (20R) are output terminals, (36) is a time-base compression circuit, (37) is a modulation circuit, (
40) is a demodulation circuit, and (41) is a time axis expansion circuit.

Claims (1)

[Claims] In an audio signal recording device that encodes an audio signal and records it on a recording medium as a digital signal, the encoding is performed using ADP.
An audio signal recording device characterized in that it uses a CM system.