JPH0588573B2 - - Google Patents

Description

[Detailed description of the invention]

[Industrial Application Field] The present invention relates to a signal transmission device that transmits a signal obtained by digitizing an analog signal such as a PCM signal by dividing it into blocks of a certain number of words, and in particular, the present invention relates to a signal transmission device that transmits a signal obtained by digitizing an analog signal such as a PCM signal by dividing it into blocks of a certain number of words. The present invention relates to a signal transmission device which aims to reduce the transmission bit rate. [Prior Art] In recent years, analog audio signals, video signals, etc. are sampled, quantized and encoded, and then transmitted, recorded, and played back as so-called PCM (pulse code modulation) signals. Things are becoming more and more common. Transmitting or recording/recording such PCM signals, etc.
When playing, for example, a band of about 20KHz and
In order to obtain an S/N of about 90 dB or higher, the sampling frequency s is set to 44.1 KHz and linear quantization of 16 bits per word is generally adopted, but the transmission rate in this case is 700 KBPS (700 K bits per second). It is extremely high, reaching even more than that. By the way, digital signals obtained by A/D conversion of analog signals such as audio and video signals as mentioned above have biased statistical properties and parts that are less important from the viewpoint of audiovisual phenomena. It is possible to compress the amount of information by taking advantage of the fact that there is a ing. Taking these points into consideration, the applicant first developed a digital PCM signal, for example, by dividing it into blocks for a certain time unit or by a certain number of words, and applying predictive processing such as differential processing and companding to each block. processing and transmission or recording/
Japanese Patent Application No. 58-97687-9, Japanese Patent Application No. 163054-1982, Japanese Patent Application No. 166-267-1982, Japanese Patent Application No. 210382-1983, etc. propose the reproduction. In these techniques, each block contains at least one word of reference data, e.g.
PCM data is provided, and by performing arithmetic processing such as sequentially adding difference data based on this reference data, it is possible to restore all of the original sampling data (straight PCM data) in a block. Furthermore, as the companding process mentioned above, it has been proposed to perform noise shaping processing by requantizing the input data and feeding back the predicted value of the quantization error at this time (so-called error feedback). It is desirable to perform this quantization error prediction process separately from the signal prediction process in order not to degrade the instantaneous S/N ratio. In this case, the extraction position of the requantized bits relative to the original data bits during the requantization, the so-called ranging position, is determined based on the data before the noise shaping process. [Problems to be Solved by the Invention] By the way, in order to further increase the bit rate reduction efficiency, when transmitting, recording, or reproducing data without providing reference data for each block, the input signal level may be lowered near the boundaries of the blocks. When there is a sudden change in the noise shaping, the error fed back from the last word of the previous block may be superimposed on the first word of the next block, causing an overflow in the requantized data. This overflow causes adverse effects such as distortion on the transmitted signal. In view of these circumstances, the present invention has been developed to divide input signals into blocks and transmit them without providing a reference word for each block, and to perform signal and noise prediction processing separately. In a transmission device, there is a method that suppresses the overflow of requantization bits caused by feedback of errors from the previous block when the signal level changes near a block boundary, and reduces the negative effects of the overflow. The purpose is to provide signal transmission equipment. [Means for Solving the Problems] In order to solve the above-mentioned problems, the signal transmission device of the present invention divides the input digital signal into blocks of a certain number of words along the time axis, and divides the signal of each block into blocks. and a means for requantizing the predictively processed signal and feeding back the quantization error to perform noise shaping processing. Based on the maximum absolute value within the block of the requantized signal, the requantization bit extraction position, so-called ranging position, is determined during the requantization, and this ranging position is the data before requantization.
It is characterized in that the amount of movement when moving to the LSB side of a word is limited to, for example, about 1 bit. [Function] In this way, during requantization in the companding process for each block, by limiting the amount of movement when the ranging position, that is, the requantization bit extraction position moves toward the LSB side, the previous block Errors fed back from the system can be suppressed to a small level, and the negative effects of overflow can be reduced. [Embodiment] Schematic Configuration First, the overall schematic configuration of an audio bitrate reduction system, which is an example of a signal transmission device to which the present invention is applied, will be described with reference to FIG. The system shown in FIG. 1 consists of an encoder 10 on the transmitting side (or recording side) and a decoder 30 on the receiving side (or reproducing side).
0 input terminal 11 receives an audio PCM signal x(n) obtained by sampling an analog audio signal at frequency s, quantizing and encoding it.
is supplied. This input signal x(n) is sent to a predictor 12 and an adder 13, respectively.
The predicted signal x〓(n) from the predictor 12 is sent to the adder 13 as a subtraction signal. Therefore, in the adder 13, by subtracting the prediction signal x〓(n) from the input signal x(n), the prediction error signal or (in a broad sense) difference output d(n), That is, d(n)=x(n)−x〓(n)... is output. Here, the predictor 12 generally calculates the predicted value x〓(n) by a linear combination of p past inputs x(n-p), x(n-p+1)...x(n-1). x〓(n)= _p〓k ^=1αk _・x(n−k)…where _αk (k−1, 2…p) is a coefficient. Therefore, the above prediction error output or difference output (in a broad sense) d(n) can be expressed as d(n)=x(n)− _p 〓 ^k=1 α _k ×(n−k) . In addition, in the present invention, data within a certain period of time of the input digital signal, that is, the input data is divided into blocks every certain number of words l, and the coefficient α _k The set is selected. As will be described later, this can be considered as having multiple filters for obtaining differential outputs (prediction error outputs), including predictors with different characteristics or adders, and these multiple differentials The optimal filter among the processing filters is selected for each block. The selection of this optimal filter is
This is performed by comparing the maximum absolute value (peak value) within a block or the value obtained by multiplying the maximum absolute value (peak value) by a coefficient of the output from each of the plurality of difference processing filters with each other in the prediction/range adaptation circuit 21. Specifically, the differential processing filter whose value is the minimum among the maximum absolute values (or their coefficient multiplication values) is selected as the optimal filter for the block. The optimal filter selection information at this time is output from the prediction/range adaptation circuit 21 and sent to the predictor 12 as mode selection information. Next, the difference output d(n) as the above prediction error is
The signal is sent via an adder 14 to a bit compression means consisting of a shifter 15 for gain G and a quantizer 16. Compression processing or ranging processing is applied to each output from the converter 16. That is, the shifter 15 switches the so-called range by shifting the digital binary data by the number of bits corresponding to the gain G (arithmetic shift), and the quantizer 16 changes the range of the bit-shifted data. Requantization is performed to extract the number of bits. Next, the noise shaping circuit (noise shaper) 17 uses an adder 18 to obtain a so-called quantization error corresponding to the error between the output and input of the quantizer 16, and converts this quantization error into a gain G ^-1
The predicted signal of the quantization error is sent to the predictor 20 via the shifter 19, and so-called error feedback is performed in which the predicted signal of the quantization error is fed back to the adder 14 as a subtraction signal. Next, the prediction/range adaptation circuit 21 outputs range information based on the maximum absolute value within the block of the differential output from the filter of the selected mode, and sends this range information to each shifter 15 and 19 for each block. However, in the present invention, the range is changed from large (coarse requantization) ^to small (fine requantization) because the signal level decreases near the block boundary. )
In other words, when the gain G increases and the bit extraction position (ranging) of the requantized data moves to the LSB side of the original data, it is possible to limit the range change or the movement of the ranging position. There is. In other words, the prediction/range adaptation circuit 21 does not impose any restrictions when increasing the range (moving the ranging position to the MSB side: decreasing G), but when increasing the range (moving the ranging position to the MSB side: decreasing G).
When moving to the LSB side (increasing G), control is performed to limit it to only one bit, for example. Further, the prediction/range adaptation circuit 21 sends the mode information to the predictor 20 to select the optimum filter characteristic. The range information from the prediction/range adaptation circuit 21 is taken out from the output terminal 23, and the mode selection information is taken out from the output terminal 24. Next, to explain the basic operation of the noise prediction process after the adder 14, the output d''(n) from the adder 14 is determined by the quantization error from the noise shaper 17 than the difference output d(n). Subtracting the predicted signal e〓(n) gives d′(n)=d(n)−e〓(n)..., and the output d″(n) from the shifter with gain G is d″(n)=G・d′(n) … Also, the output d^(n) from the quantizer 16 is
Letting the quantization error in the quantization process be e(n), e^(n)=d''(n)+e(n)..., and the adder 18 of the noise shaper 17 calculates the quantization error e(n). ) is taken out and the gain G ^-1
The predicted signal e〓(n) of the _quantization error obtained through the shifter 19 ^of β _k・e(n−k)・G ⁻¹ …. This equation has a similar form to the above equation, and predictors 12 and 20 each have a system function

〔Effect of the invention〕

According to the signal transmission device according to the present invention, when the input signal level suddenly decreases near the block boundary and the ranging position, that is, the requantization bit extraction position is about to rapidly move toward the LSB side,
Since this amount of movement is limited, errors caused by overflow of requantization bits can be kept small. Furthermore, when the requantization bits overflow in this way, clipping is performed at the maximum positive or negative value, and the error at this time is fed back and noise shaping processing is performed to suppress error propagation and eliminate abnormal noise. This makes it possible to effectively prevent adverse effects such as the occurrence of. In addition, multiple differential processing filters that output high-order differential PCM, first-order differential PCM, and straight PCM data are used, and these are adaptively switched and selected, allowing efficient bit rate reduction.
Signal transmission at extremely low bit rates is possible without deteriorating signal quality. In addition, since the straight PCM data output mode can be switched and selected, it is possible to solve problems such as S/N deterioration when inputting high-frequency signals and excessive error power when an error occurs. Furthermore, a coefficient γ (γ≧
1) and determines the ranging position, that is, the requantization bit extraction position based on this multiplied value, even if feedback errors due to noise shaping processing are superimposed,
Overflow is less likely to occur.

[Brief explanation of drawings]

FIG. 1 is a block circuit diagram showing a schematic configuration of the entire system to which the signal transmission device according to the present invention is applied, and FIGS. 2 and 3 show more specific configuration examples of the encoder and decoder shown in FIG. The respective block circuit diagrams are shown, FIG. 4 is a graph showing the frequency characteristics of multiple differential processing filters, FIG. 5 is a diagram showing an example of the transmission word structure within one block, and FIG. 6 is used for other specific examples. A graph showing the frequency characteristics of multiple differential processing filters, Figure 7 is a graph showing the frequency characteristics of multiple differential processing filters.
A graph showing the spectral distribution of noise subjected to shaping processing. Fig. 8 is a diagram for explaining the movement of the ranging position during requantization. Fig. 9 is a block circuit diagram showing the main parts of the encoder. Fig. 10. 1 is a block circuit diagram showing the main part of the decoder, and FIGS. 11 and 12 are diagrams for explaining error propagation due to overflow during requantization. 10... Encoder, 12, 12A to 12D,
20, 34... Predictor, 15, 19, 32... Shifter, 16... Quantizer, 17... Noise shaper, 21... Prediction/range adaptation circuit, 30...
decoder.

Claims (1)

[Claims] 1. A means for dividing the input digital signal into blocks of a certain number of words along the time axis, performing predictive processing on the signal for each block, requantizing the predictively processed signal, and feeding back the quantization error. means for performing noise shaping processing based on the maximum absolute value within the block of the signal subjected to the prediction processing, and determining the requantization bit extraction position in the requantization, and A signal transmission device characterized in that a limit is placed on the amount of movement when a quantization bit extraction position moves toward the LSB side.