JPH01151836A - Digital data transmission method - Google Patents

Abstract

PURPOSE:To realize highly efficient coding and data transmission by sending a predicted residual for each sample and a prediction coefficient and a normalized gain for each block. CONSTITUTION:An error caused by compressing a decoded data Dt is minimized because the prediction coefficient of prediction filters 19, 43 is controlled to an optimum value according to an input data Xt even when there is a limit in the coefficient and the work length of arithmetic operation. Moreover, in case of the transmission of the predicted residual Dt, since the number of bits of the residual Dt is decreased by requantization and normalization is applied before the requantization, the number of bits of the data Dt.G to be sent is few and the data has less error. Thus, highly efficient coding and data transmission is attained.

Description

[Detailed description of the invention] (Industrial application field) The present invention relates to a method for transmitting digital data.

[Summary of the invention]

In a digital data transmission method, the present invention realizes highly efficient encoding and data transmission by transmitting prediction residuals for each sample and prediction coefficients and normalization gains for each block.

[Conventional technology]

For example, in the case of 8 mm video, an optional function is to digitize the audio signal into a PCM fit number during recording, record this PCM (W number) in the overscan section of the tape, and perform the reverse process during playback. In particular, it is allowed to obtain the original Oday No. 114.

In this case, audio signals can be recorded and played back with better characteristics by increasing the sumblink frequency and number of quantization bits of the PCM signal, but doing so increases the number of bits to be recorded and played back, making recording and playback impossible. It's gone.

Therefore, when recording, the number of bits of the PCM signal is compressed.
It has been proposed to expand the number of bits during playback so that excellent recording and playback characteristics can be obtained even if the number of bits on the tape is small.

And, as a method of bit compression/expansion, AD
There is a method called PCM.

Literature: [Fundamentals of Speech Information Processing] Published by Ohmsha, patent application 1986
Specification and Drawings of No. 299285 [Problems to be Solved by the Invention] The present invention aims to provide a transmission method that is superior to the PCM system.

[Means for solving problems]

Therefore, in the present invention, each predetermined number of samples converted to a digital signal at a predetermined sampling frequency is treated as one block, and for each block, prediction coefficients and normalization are performed from the samples included in the block. Calculate the gain, based on this prediction coefficient, calculate the prediction residual for each sample of the block for which this prediction coefficient was calculated, and based on the normalized gain, calculate the prediction for the block for which this normalized gain was calculated. Normalize the residuals,
In this digital data transmission method, the normalized prediction residual is transmitted at the rate of the samples, and the prediction coefficient and the normalized gain are transmitted for each block.

(Operation) Cool digital audio data is compressed to about 30% and transmitted.

〔Example〕

FIG. 1 shows an example of a transmission system according to the present invention,
In this example, every 64 consecutive samples of input data are treated as one block, and the prediction coefficients of the prediction filter are controlled to optimal values for each block. At this time, bit-compressed main data is output for each sample of the human data, and auxiliary data related to the bit compression is output for each l block.

That is, in FIG. 1, (10) is an encoder, (
30) is a signal transmission system, and (40) is a decoder. For example, if it is used for a PCM audio system in 8 mm video, an encoder (10) is provided in a recording system,
The decoder (40) is provided on the recycled yarn, and the transmission system (30) includes an error correction processing circuit, a single rotation magnetic head, and the like.

In the encoder (10), the digital data Xt is transmitted in parallel every l samples from the input terminal (11) through the delay circuits (12) and (13) to the subtraction circuit (
14). In this case, the input data Xt is a linear A/D converted analog audio signal.
It is a CM signal, for example, the sampling frequency is 48k.
Hz, and the number of quantization bits is 16 bits. Also,
As shown in FIG. 2, it is assumed that the data Xt is expressed as a fixed decimal point with −1≦Xt<1 and as a two's complement (the same applies to other values).

Furthermore, the delay circuits <12) and (13) are for synchronizing the timing of the main data and the auxiliary data, and each has a delay time of one block period (
Therefore, strictly speaking, if the human power value of the terminal (11) is Xt, the output of the delay circuit (13) will be Xt-12@, but for the sake of complexity, it will simply be written as Xt).

Further, the predicted value ×[ for the data Xt is taken out from the prediction filter (19), and this value
) and from the subtraction circuit (14), the values Xt and ×t
The difference between Dt ot=xt - t is extracted. This value DL is an error (prediction residual) of the predicted value x t with respect to the human power value Xt. Therefore, the value D
Ideally, t)t=o, which is a small value even for boat fishing, so even if the word length of the value Dt is, for example, 16 bits, for example, Dt-"0.000... -011
011”, the significant bits on the MSB side are
All become “O” (except for the sign bit), and the remaining LSB
Several bits on the side become θ' or "1" corresponding to the difference between the value Xt and the null L. Further, when the value Dt becomes a large value, the lower bits can be ignored.

Therefore, this value L)t is supplied to the gain control circuit (15) and multiplied by G (G≧1) to become a normalized value Dt-G, and this value G-1)t is re-doubled. Quantization circuit (1
6) and requantized, for example, into a 4-bit value t5t-c.

Furthermore, this value f5t-G is supplied to the gain control circuit (17) and is converted to 1/G(W), so that it is made into a non-normalized value 5t in the same order as the value L)t, and this value t5t is At the same time, the predicted value 7t from the filter (19) is supplied to the addition circuit (18), and from the addition circuit (18), the sum of the values 3t and ×t, t ×tllll t+5t is taken out and this value is fed to the filter (19).

In this case, the value t is the predicted value for the value Xt, and the value t5t is the value obtained by truncating or rounding the lower bits of the error Dt at the time of prediction, so the sum of these values t and 5t The value father t is approximately equal to the input value Xt. Since this value difference is supplied to the filter (19), the value 5<t which is the filter output
can be a value that predicts the human power value Xtφ1 at the time of the next sample.

The other bt-G from the requantization circuit (16) is then supplied to the decoder (40) through the transmission system (30).

In this decoder (40), the value eit-G is multiplied by 1/G by the gain control circuit (41) to give the value 5t, and this value 5t is supplied to the addition circuit (42), whose addition output is output from the output terminal. (44), and a prediction filter (43) configured similarly to filter (19).
), and its filter output is supplied to an adder circuit (42).

Therefore, the output of the filter (43) becomes the value nuL, and the terminal (44) receives data 5<t, in which the lower bits of the human data Xt are rounded, that is, digital data that is approximately equal to the human data Xt. The material is removed.

Further, in order to set the prediction coefficients in the filters (19) and (43) to optimal values for each block, the following circuit is provided.

That is, the prediction filters (19) and (43) are
As a prediction coefficient, for example, the Percor coefficient (PAIICOR
The third-order filter uses coefficients), and the first to fourth coefficients a1 to a3 can be changed to arbitrary values.

Input data Xt from the terminal (11) is supplied to a time window circuit (21), subjected to predetermined weighting, and then supplied to an autocorrelation circuit (22) to calculate a correlation coefficient. The coefficients are supplied to a prediction coefficient circuit (23), and Percoll coefficients of 1 to 3 are calculated as prediction coefficients up to the third order for each block of data Xt.

Further, data Xt from the delay circuit (12) is supplied to a prediction error filter (24), and the filter output is supplied to an intra-block maximum value detection circuit (25).

In this case, the filter (24) is the prediction filter (19)
a third-order prediction filter (241) configured in the same manner as
a subtraction circuit (242), and a coefficient circuit (242).
The prediction coefficients 1 to 3 from 3) are supplied to the filter (241), and a predicted value (prediction error) 5t of the error Vt with respect to the human data Xt is generated for each sample. Further, the detection circuit (25) detects 1 of the input data Xt.
For each block, the absolute value i5 tmax of the prediction error having the maximum absolute value is detected among the prediction i14 differences e5t (there are 64 of them) within that block.

Then, this maximum value j5*a is determined by the normalized gain calculation circuit (
26) and the data of the gain G during normalization, Q w
b/5max b is a safety factor of Q<b<1, and is converted to, for example, b=0.9, and this data G is used in the gain control circuit (15).

(17) and is also supplied to the gain control circuit (41) through the latch (52). In this case, D
t-G=DL-b/15+ax=pt
Since -b/(maximum value of 15t) bit#Dt, the value Dt-G is normalized to -1≦Dt-G<1.

Furthermore, considering the amount of data transmitted from the encoder (10) through the transmission system (30) to the decoder (40), data bt-c, which is raw data, is transmitted for each sample with 4 bits, for example, and auxiliary data For example, the prediction coefficient 1 to 3 and the data G are 16 bits,
Since 12 pints, 12 bits, and 8 hits are transmitted in one block, the amount of data in one block period is 4 bits x 64 samples + 16 bits + 12 bits +
It becomes 12 bits + 8 bits - 304 bits. and,
The amount of data in one flock period without data compression is 6 bits x 64 samples - 1024 bits. Therefore, the amount of data is compressed to 304 bits / 1024 bits - 29.7% and transmitted. That means that.

Thus, according to the present invention, it is possible to perform data compression of digital audio data, but in this case, in particular according to the present invention, the prediction filter (19) can be used even if there are restrictions on the word length of the coefficients and operations.

Since the prediction coefficient in (43) is controlled to a consistent value according to the input data Xt, errors caused by compression of the decoded data 6t can be minimized.

In addition, when transmitting the prediction residual 1)t, this residual L)t
Since the number of bits is reduced by requantization and normalization is performed before the requantization, the transmitted data e5t-c has a small number of bits and has little error.

FIGS. 3 and 4 show an example in which the encoder (1o) and decoder (40) described above are used in a recording system and a reproducing system for 8 mm video audio signals.

That is, in the recording thread shown in FIG. 3, a color video signal of, for example, the NTSC system is supplied to the recording video circuit (12) through the terminal (61), the luminance signal is converted into an FM signal, and the carrier color signal is converted into an FM signal. , the lower frequency side than the FM luminance signal, that is, the carrier frequency fc is f c =
47.25f h ('+743kHz.

th is the horizontal frequency), and these F
Addition signal S of the M luminance signal, the low-frequency converted carrier color signal, and the pilot signal for tracking servo during playback
v is taken out, and this signal Sv is supplied to a switch circuit (64) through a recording amplifier (63).

In addition, stereo left and right channel audio signals R and A/L converters (72L') and (72R) are connected through terminals (71L) and (71R), respectively.
The signals are supplied to the encoder and converted into digital data xt, xt with a sampling frequency of 48 kHz and a number of bits of 16 for each H, for example, and these data xt, xt are converted into digital data xt, Configured encoder (IOL), (IOR)
For each channel, data 5t・G,
kr ~ 3. G is taken out.

These data are then supplied to the recording encoder (73), and are subjected to processing such as addition of error correction data, interleaving, and time axis compression to approximately 115 field periods at the end of each field period for each field period. The recording encoded digital signal Sa is supplied to a modulation circuit (74) and is converted into, for example, a pie-phase mark signal sb.This signal sb is passed through a recording amplifier (75) to a switch circuit (64). ).

Then, the switch circuit (64) is controlled at a predetermined timing so that the signal Sv is alternately supplied to the rotating magnetic heads (IA) and (IB) every one field period, and the signal sb is different from the signal Sv. Inverse relationship Tehet (IA
), (1B).

Also, the heads (18) and (1B) are each %2180"
The magnetic tape (
2) is made to run diagonally at a constant speed. Note that the heads (LA) and (IB) have different slit angles, so-called azimuth angles.

Therefore, as shown in FIG.
Tracks (2T) are formed adjacently and sequentially,
The signal sb is recorded in a 36° section from the beginning of the two tracks, and the signal Sv is recorded in the remaining 180° section.

Note that the recording system and recording format described above are the same as those of the current 8 mm video except for the signal Sa in the signal sb.

In the raw silk shown in FIG. 4, the head (IA) outputs a signal Sv from every other track (two tracks).

sb is sequentially reproduced, and the head (IB) sends the signals Sν, sb7! from every other remaining track (2T). l< are sequentially reproduced, and these reproduced signals are sent to a reproduction amplifier (81A).

(81B) to the switch circuit (82),
From the switch circuit (82), the head (IA), (
The reproduced signals Sv and Sv of 1B) are taken out continuously, and the reproduced signals sb and sb of heads (LA) and (IB) are taken out approximately at the end of each field period.
It is retrieved every 15 field periods.

Then, the signal Sv from the switch circuit (82) is supplied to the reproduction video circuit (83), where the reverse processing to that during recording is performed and the original color video signal is taken out to the terminal (84).

Further, the signal sb from the switch circuit (82) is supplied to the demodulation circuit (91) to demodulate the signal Sa, and this signal Sa is supplied to the reproduction decoder (92) for time axis expansion and
Due to error correction and error correction,
Original data 5t-G, kt~ka for each channel
, G are decoded and these data are sent to a decoder (40L) configured similarly to the above-described decoder (40),
(4011) and data×t, 9. t is decoded and these data are sent to the D/A converter (93L
), (93R) is supplied to the terminal (94L)
, (94R), the original audio signal R is extracted.

As described above, the audio signals R and R are recorded and reproduced, but in this case, the data 5t-G, kt~ recorded on the tape (2) per second are 3. The data amount of G (excluding error stop data, etc.) is (48000 samples/64 samples) block x 304 bits x 2 channels = 4
It becomes 56 x 10' bits. In the current PCM audio for 8mm video, the sampling frequency is 2 f h # 31.468 kHz and the number of quantization bits is 8 bits, so tape (2)
The amount of data recorded in
03X10' bits. Therefore, the data 5t-G mentioned above.

k1 ~ 3. G can be recorded and reproduced satisfactorily.

〔Effect of the invention〕

According to this invention, even if there are restrictions on the coefficients and word lengths of calculations, the prediction coefficients of the prediction filters (19) and (43) are controlled to optimal values according to the input data Xt, so that the decoded Errors caused by compression of data 5t can be minimized.

In addition, when transmitting the prediction residual Dt, the residual Dt is requantized to reduce the number of bits and normalized before the requantization, so the transmitted data tst-c is , resulting in data with fewer bits and fewer errors.

[Brief explanation of the drawing]

FIG. 1 is a system diagram of an example of the present invention, and FIGS. 2 to 5 are diagrams for explaining the same. (lO) is the encoder, (30) is the signal transmission system, (40
) is a decoder.

Claims (1)

[Claims] A predetermined number of samples converted into digital signals at a predetermined sampling frequency is set as one block, and a prediction coefficient and a normalization gain are calculated from the samples included in the block for each block. Based on this prediction coefficient, calculate the prediction residual for each sample of the block for which this prediction coefficient was calculated, and based on the normalization gain, calculate the prediction residual for the block for which this normalization gain was calculated. A method for transmitting digital data, wherein the normalized prediction residual is transmitted at the rate of the samples, and the prediction coefficient and the normalized gain are transmitted for each block.