JP2943143B2 - Digital data transmission method - Google Patents

Description

Description: TECHNICAL FIELD The present invention relates to a method for transmitting digital data. [Summary of the Invention] The present invention realizes highly efficient encoding and data transmission by transmitting a prediction residual for each sample, a prediction coefficient and a normalization gain for each block in a digital data transmission method. It was done. [Prior Art] For example, in an 8 mm video, as an optional function, at the time of recording, an audio signal is digitized into a PCM signal, and this PCM signal is recorded in an overscan section of a tape. It has been recognized that the original audio signal can be obtained by performing the processing. In this case, if the sampling frequency and the number of quantization bits of the PCM signal are increased, the audio signal can be recorded and reproduced with more excellent characteristics.However, the number of bits to be recorded and reproduced increases, and the recording and reproduction cannot be performed. Would. Therefore, during recording, the number of bits of the PCM signal is compressed,
At the time of reproduction, it has been considered to expand the number of bits so that excellent recording and reproduction characteristics can be obtained even if the number of bits on the tape is small. And as such a bit compression / expansion method, AD
There is a method called PCM. Literature: "Basics of Speech Information Processing", Japanese Patent Application No. Sho 61-issued by Ohmsha
Description and drawings of Japanese Patent No. 299285 [Problems to be Solved by the Invention] The present invention aims to provide a better transmission method for the ADPCM system. [Means for Solving the Problems] Therefore, the digital data transmission method of the present invention is:
For each data block composed of a predetermined number of sample data, a prediction coefficient is obtained from the sample data, for each of the data blocks, a normalized gain is obtained from the sample data, and the sample data of the data block corresponding to the prediction coefficient is obtained. Previously, a prediction residual is obtained based on the prediction coefficient, and for each sample data of the data block corresponding to the normalization gain, the prediction residual is normalized based on the normalization coefficient. And outputting data representing the prediction residual for each sample data, and outputting data representing the prediction coefficient and the normalized gain for each data block. [Operation] Original digital audio data is transmitted after being compressed to about 30%. [Embodiment] Fig. 1 shows an example of a transmission system according to the present invention. In this example, every 64 consecutive samples of input data, the 64 samples are regarded as one block, and a prediction filter is used for each block. This is a case where the prediction coefficient is controlled to an optimum value. At this time, the main data which is bit-compressed is output for each sample of the input data, and the auxiliary data relating to the bit compression is output for each block. That is, in FIG. 1, (10) is an encoder,
(30) indicates a signal transmission system, and (40) indicates a decoder. For example, if it is applied to a PCM audio system in 8 mm video, the encoder (10) is provided in a recording system, and the decoder (40) is In addition to being provided in the playback system, the transmission system (3
0) includes a processing circuit for error correction, a rotating magnetic head, and the like. Then, in the encoder (10), the digital data
Xt is supplied from the input terminal (11) to the subtraction circuit (14) through the delay circuits (12) and (13) in parallel for each sample. In this case, the input data Xt is a PCM signal obtained by linearly A / D converting an analog audio signal.
The sampling frequency is 48 kHz and the number of quantization bits is 16 bits. The data Xt has a value of -1 as shown in FIG.
It is assumed that the value is represented by a fixed point of Xt <1 and is represented by a two's complement (the same applies to other values). Further, the delay circuits (12) and (13) are for adjusting the timing of the main data and the auxiliary data.
Each has a delay time of one block period. (Strictly speaking, if the input value of the terminal (11) is Xt, the output of the delay circuit (13) is Xt _-128 . ,
Simply Xt). Further, a prediction value t for the data Xt is taken out from the prediction filter (19), and this value t is supplied to the subtraction circuit (14), and the difference Dt Dt = Xt between the value Xt and t is output from the subtraction circuit (14). -T is retrieved. This value Dt is an error (prediction residual) of the predicted value t with respect to the input value Xt. Therefore, the value Dt is
Ideally, Dt = 0 and generally a small value. Even if the word length of the value Dt is, for example, 16 bits, for example, Dt = “0.000 ‥‥ 011011”; All of the bits are “0”
(Excluding the sign bit), and the remaining few bits on the LSB side become “0” or “1” corresponding to the difference between the values Xt and t. When the value Dt becomes a large value, the lower bits can be ignored. Therefore, this value Dt is supplied to the gain control circuit (15) and is multiplied by G (G1) to be a normalized value Dt ·
G, and this value G · Dt is supplied to the requantization circuit (16), and is requantized into, for example, a 4-bit value t · G. Further, this value t · G is supplied to the gain control circuit (17) and multiplied by 1 / G, so that it is an unnormalized value t in the same order as the value Dt, and this value t is added to the addition circuit ( 18), the predicted value t from the filter (19) is supplied to the addition circuit (18), and the sum tt = t + t of the values t and t is extracted from the addition circuit (18). This value t is supplied to the filter (19). In this case, the value t is a predicted value for the value Xt, and the value t is a value obtained by truncating or rounding the lower bits of the error Dt at the time of the prediction.
T, which is the sum of t and t, is approximately equal to the input value Xt.
Then, since this value t is supplied to the filter (19), the value t, which is the filter output, can be a value that predicts the input value _{Xt + 1} at the next sample time. Then, the value t · G from the requantization circuit (16) is supplied to the decoder (40) through the transmission system (30). In the decoder (40), the value t · G is multiplied by 1 / G by a gain control circuit (41) to obtain a value t, and this value t is supplied to an addition circuit (42), and the added output is output to an output terminal. At the same time as the filter (19), the signal is supplied to a prediction filter (43) configured in the same manner as the filter (19), and the filter output is supplied to the addition circuit (42). Accordingly, the output of the filter (43) becomes the value t, and the terminal (44) extracts the data t in which the lower bits of the input data Xt are rounded, that is, the digital data t substantially equal to the input data Xt. . Further, the following circuit is provided to make the prediction coefficients in the filters (19) and (43) optimal values for each block. That is, the prediction filters (19) and (43) are tertiary filters that use, for example, Percoll coefficients (PARCOR coefficients) as prediction coefficients, and have first to fourth coefficients a _{1 to} a _3. Can be changed to any value. Further, the input data Xt from the terminal (11) is supplied to a time window circuit (21), and after a predetermined weighting, supplied to an autocorrelation circuit (22) to calculate a correlation coefficient. coefficient PARCOR coefficients k ₁ to k ₃ are calculated as prediction coefficients to the tertiary per block are supplied to the prediction coefficient circuit (23) data Xt. Further, the data Xt from the delay circuit (12) is supplied to the prediction error filter (24), and the output of the filter is supplied to the intra-block maximum value detection circuit (25). In this case, the filter (24) includes a tertiary prediction filter (241) configured in the same manner as the prediction filter (19), and a subtraction circuit (242), and the prediction coefficient k from the coefficient circuit (23). _{1 to} k ₃ are supplied to the filter (241), and the predicted value (prediction error) t of the error Dt with respect to the input data Xt is
It is generated for each sample. The detection circuit (25) detects, for each block of the input data Xt, the absolute value max of the prediction error having the maximum absolute value among the prediction errors t (there are 64) in the block. Things. The maximum value max is calculated by the normalized gain calculation circuit (2
The data of the gain G at the time of normalization supplied to 6), G = b / maxb is a safety coefficient of 0 <b <1, for example, converted to b = 0.9, and this data G is converted to a gain control circuit ( 15), (1
7) and to the gain control circuit (41) through the latch (52). In this case, since Dt · G = Dt · b / max = Dt · b / (maximum value of t) t ≒ Dt, the value Dt · G is normalized to −1Dt · G <1. Considering the amount of data transmitted from the encoder (10) to the decoder (40) through the transmission system (30),
The data G, which is the main data, is transmitted, for example, by 4 bits for each sample, and the prediction coefficients k ₁ to
k ₃ and the data G are for example 16-bit, 12-bit, because it is transmitted per block of 12 bits and 8 bits, the data amount in one block period, 4 bits × 64 samples + 16 bits + 12 bits + 12 bits +8 bits = 304 bits. Then, 1 when no data compression is performed
The data amount during the block period is 16 bits × 64 samples = 1024 bits. Therefore, the data amount was compressed and transmitted to 304 bits / 1024 bits9.729.7%. Thus, according to the present invention, data compression of digital audio data can be performed. In this case, in particular, according to the present invention, the prediction filter (19), Since the prediction coefficient of (43) is controlled to an optimum value according to the input data Xt, an error caused by compression of the decoded data t can be minimized. When transmitting the prediction error Dt, the residual Dt is requantized to reduce the number of bits, and normalization is performed before the requantization. Data having a small number of bits and a small error is obtained. FIGS. 3 and 4 show an example in which the above-described encoder (10) and decoder (40) are applied to a recording system and a reproduction system of an audio signal of 8 mm video. That is, in the recording system of FIG. 3, for example, an NTSC color video signal is supplied to a recording video circuit (12) through a terminal (61), a luminance signal is converted into an FM signal, and a carrier chrominance signal is converted into an FM signal. Lower frequency side than FM luminance signal,
That is, if the carrier frequency fc is fc = 47.25fh (≒ 743 kHz; f
h is a horizontal frequency), and an addition signal Sv of the FM luminance signal, the low-frequency-converted carrier color signal, and a pilot signal for tracking servo during reproduction is extracted. It is supplied to the switch circuit (64) through the recording amplifier (63). Also, the audio signals L and R of the left and right channels of the stereo are supplied to the A / D converters (72L) and (72R) through the terminals (71L) and (71R). 48kHz, 16 qubits
A / D-converted to digital data Xt, Xt of bits, and these data Xt, Xt are supplied to encoders (10L), (10R) configured similarly to the encoder (10) described above, t · G, k _{1 to} k ₃ , G are extracted. These data are supplied to a recording encoder (73), and for each field period, recording encoding such as addition of error correction data, interleaving, and time axis compression to about 1/5 field period at the end of each field period. The processed digital signal Sa is assumed to be
Sa is supplied to the modulation circuit (74) to be a bi-phase mark signal Sb, for example, and this signal Sb is supplied to the switch circuit (64) through the recording amplifier (75). Then, the switch circuit (64) is controlled at a predetermined timing, and the signal Sv is alternately supplied to the rotating magnetic heads (1A) and (1B) every one field period.
Are supplied to the heads (1A) and (1B) in a reverse relationship to the signal Sv. Further, the heads (1A) and (1B) have an image interval of 180 ° from each other, are rotated at a frame frequency in synchronization with the color video signal of the terminal (61), and have a rotating circumference of just over 216 °. The magnetic tape (2) is caused to run obliquely at a constant speed over the angular range of. The head (1A),
(1B) has different slit angles, so-called azimuth angles. Therefore, as shown in FIG. 5, tracks (2T) are successively formed adjacent to each other on the tape (2), and a signal of 36 ° from the beginning of the track (2T) is formed.
Sb is recorded, and the signal Sv is recorded in the remaining 180 ° section. The recording system and recording format described above are the same as the current 8-mm video except for the signal Sa in the signal Sb. In the reproduction system shown in FIG. 4, signals Sv and Sb are sequentially reproduced from every other track (2T) by the head (1A), and the other tracks (2
T), the signals Sv and Sb are sequentially reproduced, and these reproduced signals are passed through the reproduction amplifiers (81A) and (81B) to the switch circuit (82).
To the head (1A) from the switch circuit (82),
The reproduced signals Sv, Sv of (1B) are connected and taken out, and the reproduced signals Sb, Sb of the heads (1A), (1B) are taken out approximately every 1/5 field period at the end of each field period. . Then, the signal Sv from the switch circuit (82) is supplied to the reproduction video circuit (83), and the processing reverse to that at the time of recording is performed, so that the original color video signal is taken out to the terminal (84). The signal Sb from the switch circuit (82) is supplied to a demodulation circuit (91) to demodulate the signal Sa, and the signal Sa is supplied to a reproduction decoder (92) to expand the time axis, correct an error, and correct an error. By performing such operations, the original data t · G, k _{1 to} k ₃ , G are decoded for each channel, and these data are decoded by a decoder (40L) configured similarly to the above-described decoder (40). , (40R) supplied data
t, t are decoded, and these data are supplied to the D / A converters (93L) and (93R) to be supplied to the terminals (94L) and (94R).
Original audio signals L and R are extracted. The audio signals L and R are recorded / reproduced as described above. In this case, the amount of data t · G, k _{1 to} k ₃ , G recorded on the tape (2) per second (error correction data excluding etc.) is (48000 samples / 64 samples) block × 304 bits × 2 channels = 456 × 10 ³ bit. And, in the PCM audio in the current 8mm video, the sampling frequency is 2fh ≒ 31.468k
Since Hz and the number of quantization bits are 8 bits, the amount of data recorded on the tape (2) per second is 31468 samples × 8 bits × 2 channels ≒ 503 × 10 ³ bits. Therefore, the data t · G, k _{1 to} k ₃ , G
Can be sufficiently recorded and reproduced. According to the present invention, the prediction coefficients of the prediction filters (19) and (43) are controlled to the optimum values in accordance with the input data Xt even if the coefficient and the word length of the operation are limited. In addition, errors caused by the compression of the decoded data t can be minimized. When transmitting the prediction residual Dt, the residual Dt is requantized to reduce the number of bits, and normalization is performed before the requantization. , The number of bits is small and the error is small.

BRIEF DESCRIPTION OF THE DRAWINGS FIG. 1 is a system diagram of an example of the present invention, and FIGS. 2 to 5 are diagrams for explaining the same. (10) is an encoder, (30) is a signal transmission system, and (40) is a decoder.

Continued on the front page (72) Inventor Hiromi Takano 6-7-35 Kita-Shinagawa, Shinagawa-ku, Tokyo Sony Corporation (72) Inventor Kenzo Akairi 6-35-35 Kita-Shinagawa, Shinagawa-ku, Tokyo Sony Stock In-company (56) References JP-A-62-42621 (JP, A) JP-A-61-158218 (JP, A) JP-A-61-158219 (JP, A) JP-A-61-158220 (JP, A) (58) Fields surveyed (Int.Cl. ⁶ , DB name) H04B 14/04-14/06 H03M 7/38

Claims (1)

(57) [Claims] For each data block composed of a predetermined number of sample data, a prediction coefficient is obtained from the sample data. For each data block, a normalization gain is obtained from the sample data. Sample data of a data block corresponding to the prediction coefficient In each case, a prediction residual is obtained based on the prediction coefficient, and for each sample data of the data block corresponding to the normalization gain, the prediction residual is normalized based on the normalization coefficient. Transmitting data representing the prediction residual for each sample data, and outputting data representing the prediction coefficient and the normalized gain for each data block.