JPS58218010A - Device for encoding, storing and reproducing signal - Google Patents

Abstract

PURPOSE:To attain the encoding with high compression, by constituting a data with codes representing one of sampling values and the repetition of the sampling values when a 1-frame signal has the same sampling values and using repetitively the sampling values for the number at the reproduction. CONSTITUTION:A sound signal is converted into a 8-bit signal and stored in a memory M1. A control circuit CCT is provided with a sampling counter, a zero point counter and a frame counter. Whether or not a sign of the sampling value read out from the memory M1 is the same as the sign of the sampling value just before the former sampling value is discriminated and this operation is repeated. N-set of sampling values are read out from the memory M1 by taking the count value of every other period Tc and when the values are all the same, a sampling value added with a code representing the repetition, a frame number, the number of sampling N and the period Tc combined into a data, are stored in a memory M2. This data is read out successively one by one and given to a converter DAC1, and the data of the period Tc is given to a converter DAC2 for reproduction.

Description

[Detailed Description of the Invention] When an audio signal is encoded and transmitted or recorded or reproduced as a digital signal, conventional methods for reducing the amount of data as much as possible include logarithmically compressing the signal amplitude, calculating the difference, It is well known that various methods such as modulation or delta modulation have been adopted, but since these conventional methods seek to reduce data in the amplitude direction, the reproduced signal is distorted due to quantization distortion. There was a problem that the quality of the product was deteriorated.

When the amount of data decreases significantly, the applicant's company does not determine the decrease in pits in the amplitude direction, but rather in the time axis direction, and compares the waveform obtained from the sample value sequence with the original. We proposed an approximation compression method for audio signals that can be expected to significantly reduce the amount of data by approximating the waveform so that the degree of difference from the signal waveform is less than a certain ratio (Japanese Patent Laid-Open No. 5
(Refer to Publication No. 6-155998), and achieved some success. However, in this previously proposed method, the sampling period is determined based on the deviation of the signal waveform from the previous sample value, and the signal waveform itself is determined based on the deviation of the signal waveform from the previous sample value. Because the standard was not defined based on the above, it was observed that the reproduction of fine waveforms in the signal tended to be insufficient.

In order to solve the problems with the above-mentioned existing proposal by the applicant company, the applicant company has developed a digital A coding device was proposed. By the way, although the digital encoding device described above can easily encode signals, since the sampling interval is random,
A problem has arisen in that when a distortion component caused by an error from the original signal is included within the reproduced frequency band during signal reproduction, it cannot be removed and is included in the reproduced signal.

In other words, if a signal is encoded with a sampling interval such that the zero point interval is divided into approximately equal parts for each zero point interval of the signal, the sampling period will naturally be shorter in the signal part with the longer zero point interval. However, according to the sampling theorem, the frequency band in which the reproduced signal corresponding to the signal part with a long sampling period (low sampling frequency part) is distortion-free is a frequency band of 1/2 or less of the sampling frequency. However, the passband of the low-pass filter installed in the reproduction system is based on the sampling period determined corresponding to the part with the shortest zero point interval in the signal to be encoded. Because of this setting, distortion components in the reproduced signal corresponding to the signal portion with a long sampling period may not be included in the passband of the low-pass filter installed in the reproduction system. The problem is that the distortion of the endogenous signal inevitably increases.

Therefore, in order to solve the above problem, the applicant company first sets the signal to be encoded as a signal of Tf for each fixed time length (one frame signal), and By determining the number of zero points Zc for each frame and determining the average zero point interval in the signal period of one frame,
Proposed a recording and reproducing apparatus that performs highly compressed encoding and can remove harmonic components in the reproduced signal of each frame during reproduction using a low-pass filter (Japanese Patent Application No. 57-43649) Referenced.

The present invention provides a signal encoding, storage and reproducing device that can realize encoding with a higher degree of compression than the previously proposed devices. The specific contents of the signal encoding, storage and reproducing device of the invention will be explained in detail.

FIG. 1 is a block diagram of an example configuration of a signal encoding storage/reproduction device according to the present invention.
is a microphone microphone, LPF is a low-pass filter, ADC is an AD converter, OG is a clock pulse generator, CCT is a control circuit including a microcomputer, OP is an operation unit, Br is a recording button, Bp is a play button, and Bs is a stop button.

In addition, in FIG. 1, M1 is a first storage device (first memory), M2 is a second storage device (second memory), and DA
C1 and DAC2 are DA converters, VLPF is a variable passband type low-pass filter, AMP is an amplifier, and SP is a speaker.

FIG. 2 is a waveform example diagram of the signal Sa before encoding, and the 0-0 line in FIG. 2 is an AC axis line shown for reference. In the waveform diagram shown in FIG. 2, Tf is the time length T of a signal portion obtained by dividing the signal Sa into every fixed time length on the time axis.
f, and each signal portion having the above-mentioned time length Tf is called a signal of one frame. The digital data to be decoded in the signal encoding/storage device of the present invention has a sampling period for each frame of the signal according to the state of change on the time axis of the signal in each frame of the signal. The method of encoding is such that the number of zero points in each frame is determined in a specific relationship with the number of zero points in the signal of one frame. A sampling period is determined.

In the signal Sa shown in the second figure, a predetermined time length T
Each one-frame signal having f normally crosses the AC axis 0-0 line multiple times within the time length Tf, that is, it has multiple zero points within the time length Tf. However, the number of zero points in the signal of each frame differs depending on the frequency components of the signal in each frame, and in some cases, the number of zero points in the signal of each frame differs depending on the frequency components of the signal, and in some cases, the number of zero points in the signal There are periods when there is no.

For example, to explain the waveform Sa shown in FIG. 2, the zero point is 8 in the signal of one frame from time t1 to t2.
There are 4 zero points in one frame signal from time t2 to t3, and 1 zero point from time t3 to time t4.
There is no zero point in the frame signal, and from time t4 to time t
In the signal of one frame up to 5, there are 3 zero points, and at time t5
It can be seen that the number of zero points is different for successive frames of the signal on the time axis, such as in the signal of one frame from time t6 to time t6, there is no zero point that occurs when the signal intersects the AC axis. Karu.

Now, in the existing proposal, a signal part of a predetermined constant time length Tf in the signal Sa, that is, for each one-frame signal, the time is expressed as a number related to the number of zero points existing in the one-frame signal. In order to reduce the amount of data by performing encoding such that a sequence of sample values is obtained for the signal of one frame using a sampling period Tc whose length is divided into equal parts, it is necessary to reduce the amount of data one after another on the time axis. When the number of zero points of the signal of a frame is Zc, for each signal of one frame, the sample value sequence from the signal of each frame is obtained at a sampling period such that the time length Tf is equally divided by (Zc × K). Recording and reproducing is performed with the amount of data reduced so as to obtain , during a no-signal period, such as from time t5 to t6 in FIG. 2, unnecessary encoding will be performed.

Therefore, in the present invention, when the signals of one frame all have the same sample value, the data is composed of one of the sample values and a code representing the repetition of the sample value, and when the signal is reproduced, the sample value is repeated. For a single frame signal in which there is a code expressing In the case of a signal with many silent periods, encoding with a high degree of compression can be performed particularly effectively.In the present invention, when the number of zero points in one frame of the signal is Zc, the time of each frame of the signal is A sequence of sample values from each frame of the signal is obtained at a sampling period Tc such that the length Tf is divided into {(Zc+1)K} equal parts, and each sampling period Tc within the signal of one frame is Determine whether all the sampled values are the same or not, and determine the sampling period Tc within one frame of the signal.
When all sample values are the same, data is obtained by combining one sample value and a code representing the repetition of that sample value, and the sampling period Tc and the signal within one frame are In order to distinguish each frame from Create data in pairs with the assigned frame numbers for each successive frame, store them sequentially in a storage device, and when playing back, read out the data from the storage device and convert the data for each successive frame into a signal. This is done so that the playback is carried out.

Next, the above-mentioned encoding operation will be explained with reference to the block diagram of FIG.

The microphone MIC shown in FIG. 1 converts a sound wave into an electrical signal (audio signal) and supplies it to a low-pass filter LPF. In the recording/reproducing apparatus shown in FIG. 1, a microphone MI is used as a signal source.
Although C is used, the signal source may be a generator of other forms of audio signals or other signal generators.

In the following description of the embodiment, the low-pass filter LPF is assumed to have a cut-off frequency of 3 KHz.

The audio signal converted into a signal in a frequency band of 3 KHz or less by the low-pass filter LPF is converted into a digital signal with the required number of bits (8 bits in the following explanation) by the AD converter ADC, which is configured to include a microcomputer. is applied to the control circuit CCT which is
The converter ADC receives 8 from the clock pulse generator OG.
AD conversion is performed using pulses with a repetition frequency of KHz.

The digital signal output from the AD converter ADC is an 8-bit digital signal in which the input audio signal is always sampled at a constant sampling period (1/8000 seconds in the example) and quantized. , which is the control circuit CC
A first storage device M1 (first memory M1
, or sequentially stored in the buffer memory M1).

The buffer memory M1 mentioned above is 512 in the following explanation example.
It is said to have a storage capacity of bytes, which is divided into two parts each having half the storage capacity, and the two parts are used alternately for data writing and data reading.

Now, the encoding and recording operation of the apparatus shown in FIG.
When the record button Br in the operation section OP is operated, the program is executed according to the flowchart shown in FIG. Then, in step (1), the 9-bit sample counter, 8-bit zero point counter, 16-bit frame counter, etc. provided in the control circuit CCT are reset.

Even before the record button Br is operated, that is, before the "beginning" in the flowchart shown in the third figure, the first
The control circuit CCT of the illustrated recording and reproducing apparatus performs the interrupt operation of step (10) by receiving pulses from the clock pulse generator CG, and transfers the digital signal output from the AD converter ADC to the buffer memory. The data are sequentially stored in M1, and a 9-bit sample counter is counted up.

Also, before the beginning, the 9-bit sample counter is repeatedly reset each time it reaches a full count.

In step (2), the sample value stored in the buffer memory M1 is read out, and the 9-bit sample counter is set to 1.
Count up only.

In step (3), it is checked whether the sign of the sample value read in step (2) is the same as the sign of the sample value immediately before it, and if there is no change in sign, it is determined that the point is not zero and step ( Return to step 2), and if there is a change in sign, the sample value read in step (2) is assumed to be the zero point, and the process proceeds to step (4). In step (4), the 8-bit zero point counter is counted up by 1. do.

In step (5), it is checked whether the number of sample values sequentially read from the buffer memory M1 has reached 256 (or 512) using the count value of the 9-bit sample counter, and the sample value read from the buffer memory M1 is checked. When the number of sample values reaches 256 (that is, after repeating steps (2) to (4) 256 times), proceed to step (6), and if the number of sample values read from the buffer memory M1 has not reached 256. Then return to step (2).

Here, the number 256 of sample values read out from the buffer memory M1 as described above is calculated by the AD converter ADC for one frame of the signal Sa shown in FIG.
is the number of sample values obtained by sampling at a constant sampling period (1/8000 seconds), and the number 256 is 1
This corresponds to the time length Tf of the frame signal.

In step (6), using the count value Zc of the 8-bit zero point counter, the predetermined number K, and the number 256 representing the time length Tf of the signal of one frame, the sample in the signal of one frame is calculated. The sampling period Tc required to obtain the value sequence is calculated, and the number of samples N is also calculated.

Sampling period=Tc=256/(Ze+1)K Number of samples N=
256/Tc Next, in step (7), sample values for each sampling period Tc are extracted from the buffer memory M1 for one frame of the signal to which the sampling period Tc is applied and a sample value sequence is to be taken out. In order to read out sequentially, N sample values are sequentially read from the buffer memory M1 using the count value of the 9-bit sample counter (address counter) at intervals of Tc as an address signal, and all of the N sample values that have been read out are read out sequentially from the buffer memory M1. In the case of the same value, a code indicating that the sample value is repeated is added to one sample value, the frame number indicated by the count value Fc of the 16-bit frame counter, and the number of samples N. and the sampling period Tc, and store it in the second storage device M2 (second memory M2
) and proceed to step (8). If the read N sample values are not the same, the N sample values and 1
Data is created in which the frame number indicated by the count value Fc of the 6-bit frame counter, the number of samples N, and the sampling period Tc are combined, and the data is stored in the second storage device M2 (second memory M2).
2) and proceed to step (8).

In step (8), the 16-bit frame counter is counted up by one.

In step (9), it is checked whether the 16-bit frame counter has reached a full count or whether the stop button Bs has been operated. If so, it is over, and if not, return to step (2) and repeat each of the above steps.

As is clear from the explanation so far, the second memory M2
For each one-frame signal, data of sampling period Tc, data of number of samples N, a sample value sequence or one sample value and a code representing its repetition, and a frame number Fc are included.
(frame counter count value Fc), etc., is stored, but this is a much wider data set than the digital signal stored in the first memory M1 (buffer memory M1). Due to the above-mentioned encoding performed during recording, the amount of data is reduced, making it possible to record and play back audio signals over long periods of time with a small capacity memory. do.

Next, regarding the case where the signal stored in the second memory M2 as described above is read out and the audio signal is reproduced,
This will be explained with reference to the flowchart shown in FIG.

When the playback button Bp of the operation unit OP in the device shown in FIG. 1 is operated, the program shown in FIG. 4 starts ("
First, in step (1P), the frame counter and sample counter are reset, and in step (2P), the number of samples N in the signal of one frame, the data of the sampling period Tc, and the sample value are retrieved from the second memory Mc. Then, in step (3P), the sample values are read out from the second memory M2 one by one in the order in which they were stored, or one sample value is read out repeatedly for the required number of turns. Data with a sampling period Tc is applied to the DA converter DAC2.

In step (4P), a time wait is performed so that the time required for one round of steps (2P) to (5P) matches the time length indicated by the sampling period Tc.

1/8000=(Tc-1) (program loop time)
+ {Step (2P), Step (3P), Step (
5P) and the fixed part of step (4P)}, in step (5P), it is determined whether reading of N sample values has been completed, and if it has not been completed yet, the process proceeds to step (2P). Return, and if completed, proceed to step (6P).

In step (6P), check whether the frame counter has reached a full count or whether the stop button Br has been operated, and if the frame counter has reached a full count or the stop button Bs has been operated, it is over, or not. If so, return to step (2P).

DA converter DAC1 in the above step (4P)
, by supplying digital data to DAC2, analog signals corresponding to successive sample values within one frame signal are supplied as input signals from the DA converter DAC1 to the variable passband type low-pass filter VLPF, and , D.A.
From the converter DAC2, an analog signal corresponding to the sampling period Tc of the sample value in one frame signal is given to the variable pass band type low pass filter VLPF as its control signal.

FIG. 5 is a block circuit diagram showing a configuration example of the DA converters DAC1 and DAC2 and a variable pass band type low pass filter VLPF, and the variable pass band type low pass filter VL
In the PF, the higher cut-off frequency in the variable range of cut-off frequency is determined by resistors R5, R6 and capacitors C1, C2, and the lower cut-off frequency in the variable range of cut-off frequency is determined by resistors R6, R7 and capacitor C
1 and C2, and furthermore, the intermediate frequency value of the cut-off frequency variable range described above is determined by the DA converter DAC.
The analog switch ASW, which is switched in response to the data input of the sampling period Tc input to 2, and the variable resistors R2 and R3, which are connected in series, are varied, respectively, in response to the data of the sampling period Tc. The frequency is adjusted to a predetermined frequency value.

An example of the correspondence between the sampling period Tc, the sampling frequency fs corresponding to the sampling period Tc, and the cutoff frequency fc of the low-pass filter is shown in Table 1 below.

Variable passband type low-pass filter VL shown in FIG.
PF is the sampling period T given to the DA converter DAC2
Depending on the data value of c, the gain of amplifier A1 changes depending on the resistance selected by analog switch ASW, and the amount of light emitted by photodiodes Pd and Pd in photocouplers PC1 and PC2 changes accordingly. ) The cutoff frequency changes depending on the change in the resistance value of VR, and the passband is variably controlled. In addition, in FIG. 5, R1, R4, R5 to R
6 is a resistor, R2 and R3 are variable resistors, A1 and A2 are amplifiers, C1 and C2 are capacitors, and PC1.

PC2 is a photocoupler.

As is clear from the above description, the signal encoding, storage and reproducing apparatus of the present invention includes means for sampling and quantizing a signal at a predetermined sampling period, and a first storage for storing the digital signal obtained by the above means. a means for storing a signal section of a predetermined time length Tf as a signal of one frame, a means for measuring the number of zero points Zc in the signal of one frame, and a time of the signal of one frame as described above. Means for obtaining the sampling period Tc by further dividing the quotient obtained by dividing the length Tf by the numerical value of {measured value of the number of zero points Zc) + 1), means for obtaining the number of samples N in one frame of the signal by dividing the signal time length Tf by the sampling interval Tc; and a sampling period T in one frame of the signal.
means for obtaining sample values for each sampling period Tc; means for determining whether all sample values for each sampling period Tc within one frame signal are the same;
means for obtaining data in which one sample value is combined with a code representing repetition of the sample value when all sample values for each c are the same; a sampling period Tc; Distinguish each frame from data that combines the number of samples N in a signal and a sample value for each sampling period Tc in a signal of one frame or one sample value and a code representing the repetition of that sample value. means for creating data in pairs with frame numbers assigned for each successive frame for each successive frame and sequentially storing the data in a second storage means; A signal encoding storage/reproduction device comprising means for performing DA conversion and reproducing the signal, and in the device of the present invention, when the signals of one frame all indicate the same sample value, one sample value and its repetition are used. Since the data of the sample value sequence is composed of codes representing In this regard, recording and reproduction can be performed satisfactorily by a recording device having a simple configuration, and the above-mentioned conventional problems can be satisfactorily solved by the present invention.

[Brief explanation of the drawing]

FIG. 1 is a block diagram of an embodiment of a recording/reproducing apparatus including the encoding storage device of the present invention, FIG. 2 is a waveform example diagram for explanation, and FIGS. 3 and 4 are diagrams for explanation. The flowchart and FIG. 5 are block circuit diagrams of a variable passband type low-pass filter and its related parts. MIC...Microphone, LPF...Low pass filter,
ADC...AD converter, CG...clock pulse generator, CCT...control circuit including a microcomputer, OP...operation unit, Br...record button, Bp... Play button, Bs...stop button, M1...first storage device (buffer memory), M2...second storage device, DAC1, DAC2...DA...converter,
VLPF...Variable pass band type low pass filter 4 Closed

Claims (1)

[Claims] means for sampling and quantizing the signal at a predetermined sampling period;
A means for storing the digital signal obtained by the above means in the first storage means, and a signal section of a predetermined time length Tf is defined as one frame signal, and the number of zero points Zc in the one frame signal is determined. Measuring means and the above-mentioned 1
Means for obtaining the sampling period Tc by further dividing the quotient obtained by dividing the time length Tf of the frame signal by the numerical value of {(measured value Zc of the number of zero points) + 1} by a predetermined number. and means for dividing the time length Tf of the signal of one frame by the sampling interval Tc to obtain the number of samples N in the signal of one frame;
means for obtaining sample values at each sampling period Tc within a frame signal; and a sampling period Tc within one frame signal.
means for determining whether all sample values for each c are the same; 1.
means for obtaining data that is a combination of one sample value and a code representing repetition of the sample value when all the sample values for each sampling period Tc in the signal of the frame are the same; sampling period Tc, number of samples N in one frame signal, and sampling period Tc in one frame signal
For each successive frame, the data is a set of data that is a combination of each sample value or one sample value and a code representing the repetition of that sample value, and a frame number assigned to distinguish each frame. A method for encoding, storing and reproducing a signal, which comprises a means for creating and sequentially storing it in a second storage means, and a means for reproducing the signal by DA converting the data of each successive frame read from the second storage means. Device