JPS6345140B2 - - Google Patents

Abstract

PURPOSE:To obtain easily a reproducing signal with good quality from digital data having the reduced amount of data with a simple constitution, by changing the pass band of a low pass filter in response to the sampling period set at every one frame signal. CONSTITUTION:A digital signal outputted from an AD converter ADC is a 8- bit digital signal, in which the inputted sound signal is sampled and quantized at a prescribed sampling period (1/8000 sec in the example of this explanation) at all times, and stored in the 1st storage device M1 (the 1st memory of a buffer memory, M1) under the control of a control circuit CCT sequentially. In an encoded storage reproducing device, an analog signal obtained by DA-converting the sampling train corresponding to frame signal read out sequentially from the 2nd memory M2 is given to a low pass filter VLPF of a variable pass band which is variably controlled with a control signal obtained by DA-converting a sampling period TC in the sampling train, allowing to prevent the component of strain being a frequency >=1/2 of the sampling frequency from being produced in the reproducing signal.

Description

[Detailed description of the invention]

When encoding an audio signal and transmitting or recording/playing it as a digital signal, conventional methods to reduce the amount of data as much as possible include logarithmically compressing the signal amplitude, taking the difference, or performing delta modulation. It is well known that various methods such as There was a problem with becoming. When the amount of data decreases significantly, the applicant's company does not determine the decrease in bits in the amplitude direction, but rather in the time axis direction, and compares the waveform obtained from the sample value sequence with the original. 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
(Refer to Publication No. 37660), and implementation of this proposed method has achieved some degree of effectiveness. However, in the previously proposed method mentioned above,
As stated in page 2, column 3, lines 15 to 18 of Publication No. 37660, the sampling period is not determined based on the fine peaks and troughs of the signal waveform, but is determined based on the integral value of the deviation of the signal waveform. In other words, the sampling period is determined in such a manner that waveform approximation is performed such that the degree of difference between the waveform obtained from the previous sample value and the waveform of the original signal is equal to or less than a certain ratio. However, because the sampling period was not determined based on the fine peaks and troughs of the signal waveform, it was found 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, while the above-mentioned digital encoding device can easily encode a signal, since the sampling period is random, distortion components caused by errors from the original signal are generated when the signal is reproduced. A problem has arisen in that when it is included in the reproduced frequency band, it cannot be removed and is included in the reproduced signal. In other words, if a signal is encoded with a sampling period in which 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. become longer. By the way, according to the well-known sampling theorem, the frequency band in which a sampled signal can be reproduced without distortion is 1/2 of the sampling frequency (the reciprocal of the sampling period).
Since the frequency band is as follows, the frequency band in which a signal encoded with a long sampling period as described above is reproduced without distortion is the sampling pulse used to encode the encoded signal. This is a frequency band that is less than 1/2 of the reciprocal of the sampling period (sampling frequency) of ing. Now, as a low-pass filter (low-pass filter) used when reproducing encoded signals, in order to prevent aliasing distortion from being mixed into the reproduced signal, it is necessary to It is well known that a device having a passing characteristic that allows passing only signals in a frequency band of 1/2 or less of the sampling frequency of the sampling pulse used for encoding is used. Now, for example, if the signal to be reproduced is sampled with sampling pulses with a repetition frequency of 10 KHz when encoding it, the low-pass filter used when reproducing the signal is In order to prevent aliasing distortion from being mixed into the signal, a filter having passing characteristics that allows passing only signals in a frequency band of 1/2 or less of the 10 KHz sampling pulse is used, and for example, If the signal to be reproduced is sampled using a sampling pulse with a repetition frequency of 1KHz when encoding it, the low-pass filter used when reproducing the signal is To prevent distortion from entering, 1/2 of the 1KHz sampling pulse mentioned above.
A device having passing characteristics that allows passing only signals in the following frequency bands will be used. Now, when the signal reproduction system reproduces a signal sampled with a high repetition frequency sampling pulse and a signal sampled with a low repetition frequency sampling pulse, for example, as in the above example, 10KHz Trying to reproduce a signal sampled with a sampling pulse, i.e., a reproduction signal with a frequency band up to 5KHz, and a signal sampled with a 1KHz sampling pulse, i.e. a reproduction signal with a frequency band up to 0.5KHz. In this case, it goes without saying that the low-pass filter to be installed in the reproduction system should be one that can pass signals up to 5KHz. That is, for example, in the case of the above example,
This is because if a device that can pass a 0.5KHz signal is used, a signal with a frequency band up to 5KHz will become a signal with a frequency band up to 0.5KHz. Therefore, in order to be able to reproduce the 5KHz signal well with the above-mentioned reproduction system, we have to
If a low-pass filter that can pass signals up to 5KHz is provided, aliasing distortion will be introduced into the 0.5KHz signal. In this way, the passband of the low-pass filter provided in the reproduction system is determined based on the sampling period that corresponds 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 be included in the passband of the low-pass filter installed in the reproduction system. The problem is that signal distortion inevitably increases. The present invention has been made for the purpose of providing an encoding storage/reproduction device capable of solving the above-mentioned problems, and provides a signal for every predetermined time length in a signal to be encoded. means for detecting the number Z of zero points for each one-frame signal using a signal obtained by sampling each one-frame signal at a predetermined sampling period; The number Z of zero points obtained by the detection means is multiplied by a predetermined coefficient K, which corresponds to the time length obtained by dividing the time length of one frame signal into equal parts by the value Z.K. Each state is a set of a sample value indicating the amplitude value of the original signal obtained by encoding with a new sampling cycle and data indicating the sampling cycle for obtaining the sample value.
An encoding storage/reproduction device that stores digital data for each frame signal in a storage device, reads out and decodes the stored digital data, and includes a sample value indicating the amplitude value of the original signal, and a sample value of the sample value. Regarding the sample value indicating the amplitude value of the original signal in the digital data that is paired with the data indicating the sampling period for obtaining the value, when reading from the storage device described above, zeros in the signal of each one frame are The number of points Z is multiplied by a predetermined coefficient K, which is the value Z.K, and the signal is read out at time intervals that approximately correspond to the time length obtained by equally dividing the time length of one frame signal. means to perform DA conversion from the source signal and provide it as an input signal to a variable passband type low-pass filter, and a sample value indicating the amplitude value of the original signal, and sampling for obtaining the sample value. For the data indicating the sampling period in the digital data that is paired with the data indicating the period, means is provided for converting the data from DA to the above-mentioned variable pass band type low-pass filter as a control signal. By changing the cutoff frequency of the variable passband type low-pass filter described above using a control signal obtained by DA converting the data indicating the sampling period in the digital data described above, a reproduced signal without harmonic distortion can be obtained. DESCRIPTION OF THE PREFERRED EMBODIMENTS The specific content of the encoding storage/reproduction device of the present invention will be explained in detail below with reference to the accompanying drawings. FIG. 1 is a block diagram of an example of a storage/reproduction device including the encoding storage/reproduction device of the present invention. In FIG. 1, MIC is a microphone, LPF is a low-pass filter, and ADC is a microphone. is an AD converter, CG is a clock pulse generator, CCT is a control circuit including a microcomputer, OP is an operation unit, Br is a record button, Bp is a playback button,
Bs is a stop button, M1 is a first storage device (first memory), M2 is a second storage device (second memory), DAC1 and DAC2 are DA converters,
VLPF is a variable passband 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 of a signal portion obtained by dividing the signal Sa into each fixed time length on the time axis, and each signal portion of each time length Tf described above is one frame. This is called the signal. The digital data to be decoded in the encoding storage/reproduction device of the present invention has a sampling period for each one-frame signal depending on the state of change in the signal on the time axis in each one-frame signal. The encoded signal is encoded as determined by Z. Thus, a new sampling period corresponding to the time length obtained by equally dividing the time length of one frame signal is determined. Next, a case will be described in which encoding is performed using the encoding mode described above. In the signal Sa shown in the second diagram, each one-frame signal having a predetermined time length Tf crosses the AC axis 0-0 line multiple times within the time length Tf, that is, the time length Tf has multiple zero points, but the number of zero points in each frame of the signal varies depending on the frequency components of the signal in each frame. For example, to explain the waveform Sa shown in Fig. 2, there are eight zero points in one frame signal from time t1 to t2, and four zero points in one frame signal from time t2 to t3. Hereinafter, it will be seen that the numbers of zero points are 6, 3, and 4 for signals of one frame successively on the time axis. Now, as mentioned above, the signal portion of the signal Sa with a predetermined constant time length Tf, that is,
For each one-frame signal, the sample value sequence for that one-frame signal is determined by a sampling period Tc such that the time length is equally divided by a number related to the number of zero points present in the one-frame signal. A reduction in the amount of data is achieved when encoding is performed to obtain .This point will be explained below using the signal Sa shown in the second figure as an example. That is, when the number of zero points of the signal of one frame successively on the time axis is 8, 4, 6, 3, or 4 as described above, as in the signal Sa shown in the second figure. For example, for a signal of one frame with 8 zero points, the time length Tf is (8
×K) In order to obtain a sequence of sample values from the signal of one frame with a sampling period that is divided into equal parts,
For example, for a signal of one frame with four zero points, the sequence of sampled values from the signal of one frame will be Similarly, for each one-frame signal where the number of zero points is 6 or 3, the time length Tf is divided equally into (6 x K) or (3 x K). A sequence of sample values from each frame of the signal is obtained at a sampling period such that
When the number of zero points in one frame signal is Z (in the device of the present invention that digitally encodes an AC signal, the signal to be processed is an AC signal, and the DC signal is not a DC signal to be processed). (and there is no case where Z continues to be zero for a longer period than the period corresponding to the lowest frequency of the AC signal being processed during signal processing),
For the signal of one frame, a sequence of sample values is obtained at a sampling period such that the time length Tf is equally divided by (Z x K), and the encoding means described above is used. For example, recording and reproducing operations can be easily realized with a reduced amount of data. The data obtained by the above-mentioned encoding means, that is, the sample value sequence from each one frame signal having a predetermined time length Tf, is specified as the number Z of zero points in one frame signal. The data obtained by sampling with the sampling period obtained by equally dividing the time length Tf by the number (Z × K) having the following relationship is the data and the sampling period Tc. , the number N of sample values in one frame signal, and information such as the frame number are used for transmission or recording in pairs. Next, the aforementioned encoding operation will be explained with reference to the block diagram of FIG. The microphone MIC shown in Figure 1 converts sound waves into electrical signals (audio signals).
Convert it to and feed it to the low pass filter LPF. In the recording and reproducing apparatus shown in FIG. 1, a microphone MIC is used as a signal source, but the signal source may be a generator of other types of audio signals or a generator of other signals. In the following example description, the low-pass filter LPF is as follows:
Its cutoff frequency is said to be 3KHz.
The audio signal converted into a frequency band signal of 3KHz 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, and then sent to the microcomputer. The above-mentioned AD converter ADC performs AD conversion using pulses with a repetition frequency of 8 KHz from a clock pulse generator CG. In the digital signal output from the AD converter ADC, the input audio signal is always sampled at a constant sampling period (1/8000 seconds in the example).
This is a quantized 8-bit digital signal, which is sequentially stored in the first storage device M1 (first memory M1 or buffer memory M1) under the control of the control circuit CCT. The buffer memory M1 described above is assumed to have a storage capacity of 512 bytes in the following explanation example, and it is divided into two parts each having half the storage capacity, and the two parts are sequentially and alternately storing data. used for writing and reading data. Now, the encoding/recording operation of the apparatus shown in FIG. 1 is performed according to the program shown in the flowchart shown in FIG. 3 by operating the recording button Br on the operation section OP. Operation section
When the record button Br in the OP is operated, the program starts ("Start" in Figure 3),
9 provided in the control circuit CCT in step (1)
The bit sample counter, 8-bit zero point counter, 16-bit frame counter, etc. are reset. Before the record button Br is operated, that is, the third
Even before the "beginning" in the illustrated flowchart, the control circuit of the recording and reproducing apparatus illustrated in the first figure
The CCT performs the interrupt operation in step (10) by receiving pulses from the clock pulse generator CG, and sequentially stores the digital signal output from the AD converter ADC in the buffer memory M1.
It also counts up a 9-bit sample counter. In step (2), the stored sample value is read from the buffer memory M1, and the 9-bit sample counter is counted up by 1. 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 assumed that the point is not zero and the process is continued. Return to step 2), and when 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 by 1. rise. In step (5), the count value of the 9-bit sample counter is used to check whether the number of sample values sequentially read from the buffer memory M1 has reached 256.
When the number of sample values read from the buffer memory M1 has not reached 256 (that is, after repeating steps (2) to (4) 256 times), proceed to step (6). Return to (2). Here, the number of sample values 256 read out from the buffer memory M1 as described above is AD for one frame of signal Sa of time length Tf shown in FIG.
Converter ADC has a constant sampling period (1/8000 seconds)
This is the number of sample values obtained by sampling. 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/Zc・K Number of samples N = 256/Tc Next, in step (7), the above-mentioned sampling period Tc is applied and the sample value sequence is to be extracted from the buffer memory M1 for one frame of the signal. In order to sequentially read out the sampled values for each sampling period Tc mentioned above, N sampled values are read out sequentially from the buffer memory M1 using the count value of the 9-bit sampling counter (address counter) every Tc as an address signal. , Also, create data in which the frame number of the count value Fc of the 16-bit frame counter, the number of samples N, the sampling period Tc, and the above-mentioned N sample values are combined, and store it in the second storage device. Store it in M2 (second memory M2) and proceed to step (8) {In addition, if a remainder is generated by the calculation of Tc = Tf / Zc・K, the remainder may be truncated, rounded up, rounded off, etc. , just process the remainder appropriately to produce an integer answer. In step (8), the 16-bit frame counter is incremented by one. In step (9), check whether the 16-bit frame counter has reached full count or the stop button has been pressed.
Check whether Bs is being operated and if the frame counter has reached a full count or if the stop button Bs is being operated, the process is over; if not, return to step (2) and repeat each of the steps above. Repeat. The buffer memory M1 has a storage capacity of 512 bytes, as described above, and is divided into two parts with a storage capacity of 1/2, and the two parts are sequentially and alternately used for writing and reading. The time length of the signal is
Assume that a signal with 256 samples in one frame is stored and read out in 32 milliseconds.
Now, for example, if the number K mentioned above is set to 2, then 1
Assuming that the number Zc of zero points in the frame signal is 32, the sampling period Tc is Tc=256/32×2=4, that is, 4/8000=0.5 (milliseconds). In the above example, the number of samples N is 64, which is 256
A signal of one frame with a time length of 32 milliseconds is made up of 64 samples. This 1-frame signal with 64 samples is processed by a low-pass filter installed in the reproduction system, whose cut-off frequency is
If a frequency of 1 KHz or less, for example 750 Hz, is used, aliasing distortion will not occur in the reproduced signal due to the well-known sampling theorem. Also, if the average zero point interval Zc in the entire signal is 32, then 2 bytes correspond to the count number Fc of the frame counter, 1 byte corresponds to the number of samples N, and 1 byte corresponds to the sampling period Tc. Since the second memory M2 with a storage capacity of 68 bytes is required for one frame of signal due to the sample value string of 1 byte and 64 bytes, the second memory M2 with a storage capacity of 68 bytes is now required.
If you use 64K bytes of memory as the second
memory M2 contains 963 frames, or about 30
A signal lasting just over a second will be memorized. As is clear from the above explanation, the second memory M2 stores, for each frame of the signal, data with a sampling period Tc, a sequence of sample values, data with a number of samples N, and a frame number Fc (frame counter). A digital signal containing a set of calculated values Fc), etc. is stored, but the amount of data is greatly reduced compared to the digital signal stored in the first memory M1 (buffer memory M1). The above-described encoding performed during recording reduces the amount of data, making it possible to record and reproduce audio signals over a long period of time using a small-capacity memory. Next, the case where the signal stored in the second memory M2 as described above is read and the audio signal is reproduced by the apparatus of the present invention will be described with reference to the flowchart shown in FIG. 4. Playback button Bp of the operating section OP in the device shown in the first diagram
is operated, the program shown in Figure 4 starts ("Start" in Figure 4), and the first step (1P) is started.
At step (2P), the frame counter and sample counter are reset, and at step (2P) 1 is read from the second memory M2.
The number of samples N in the signal of the frame and the data of the sampling period Tc are read out, and then, in step (3P), the sample values are read out from the second memory M2 in the order in which they were stored.
The data of the sampling period Tc is read out and given to the AD converter DAC1, and the data of the sampling period Tc is sent to the DA converter DAC2.
given to. In step (4P), step (2P) ~
Waiting is performed so that the time required for one round (5P) matches the time length indicated by the sampling period Tc. In step (5P), it is determined whether reading of the N sample values has been completed. If the reading has not been completed yet, the process returns to step (2P), and if it has been completed, the process proceeds to step (6P). In step (6P), check whether the frame counter has reached a full count or whether the stop button Bs has been operated. If the frame counter has reached a full count or the stop button Bs has been operated, the process is over, or not. Return to step (2P). DA converter in the above step (3P)
By supplying digital data to DAC1 and 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. Also, from the DA converter DAC2, the sampling period of the sample value in one frame signal is
An analog signal corresponding to Tc is given to a 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. The higher cutoff frequency in the variable range of cutoff frequency is the resistance, R5, R
8 and capacitors C1 and C2, and the lower cutoff frequency in the variable range of cutoff frequency is determined by resistors R6 and R7 and capacitors C1 and C2.
2, and furthermore, the intermediate frequency value of the cut-off frequency variable range described above is determined by the DA converter.
an analog switch ASW that is switched by data input with a sampling period Tc input to DAC2;
variable resistors R2, each connected in series;
By varying R3, it is adjusted to a predetermined frequency value corresponding to the data of the sampling interval Tc. 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.

[Table] The variable passband type low-pass filter VLPF shown in Fig. 5 is selected by the analog switch ASW according to the data value of the sampling period Tc given to the DA converter DAC2. Amplifier A1 by resistor
The gain of the photocoupler PC changes accordingly.
1, PC2 (Photo couplers PC1 and PC2 may have the same configuration, and it is a variable resistor.
A photoresistor (e.g.
Cds) VR, the cutoff frequency changes by changing the resistance value of VR, and the pass band is variably controlled. In addition, in FIG. 5, R1, R4, R5
~R8 is a resistor, R2 and R3 are variable resistors, A1,
A2 is an amplifier, and C1 and C2 are capacitors. As is clear from the above description, the encoding storage/reproduction device of the present invention treats each predetermined time length signal in the signal to be encoded as one frame signal, and each one frame signal. Detect the zero point in each frame of the signal using the signal obtained by sampling at a predetermined sampling period,
Also, in order to obtain a sample value indicating the amplitude value of the original signal obtained by encoding with a sampling period having a specific relationship with the number of zero points in the signal of each one frame described above, and the sample value thereof. An encoding storage/reproduction device which stores digital data for each signal of each frame in a storage device, and reads and decodes the stored digital data. Regarding the sample value indicating the amplitude value of the original signal in digital data in which the sample value indicating the amplitude value of the original signal described above and the data indicating the sampling period for obtaining the sample value are set, the sample value indicating the amplitude value of the original signal is as follows. When reading data from a storage device, data is read out at time intervals corresponding to a sampling period that has a specific relationship with the number of zero points in each frame of the signal, and then DA conversion is performed and converted into a variable passband type. digital data that includes a sample value indicating the amplitude value of the original signal and data indicating a sampling period for obtaining the sample value; As for the data indicating the sampling period in , it is provided with means for DA converting the data and then supplying it as a control signal to the variable pass band type low pass filter described above,
By changing the cutoff frequency of the variable passband type low-pass filter described above using a control signal obtained by DA converting the data indicating the sampling period in the digital data described above, a reproduced signal without harmonic distortion can be obtained. In the encoding storage and reproducing apparatus of the present invention, a signal to be encoded is a signal of a certain time length as a signal of one frame, and each frame is divided into one frame. The zero point in each one frame signal is detected using the signal obtained by sampling the signal at a predetermined sampling period, and the specific relationship with the number of zero points in each one frame signal is detected. For each frame, a sample value indicating the amplitude value of the original signal obtained by encoding with a sampling cycle having Store digital data for each signal in a storage device,
An analog signal obtained by AD converting each one-frame signal sequentially read from the storage device and the corresponding sample value string is a control signal obtained by DA converting the sampling period Tc in the sample value string. is applied to a variable pass band type low pass filter VLPF whose pass band is variably controlled by Since the passband of the device is changed in accordance with the sampling period set for each frame signal, the device of the present invention can successfully solve all of the above-mentioned problems. Distortion components with a frequency value higher than 1/2 of the sampling frequency are prevented from occurring in the signal. In other words, as mentioned above, the frequency band in which a sampled signal can be reproduced without distortion is a frequency band below 1/2 of the sampling frequency (the reciprocal of the sampling period), according to the well-known sampling theorem. The frequency band that is reproduced without distortion is a frequency band that is less than or equal to 1/2 of the reciprocal of the sampling period (sampling frequency) of the sampling pulse used to encode the encoded signal. As is well known, in frequency bands above the above-mentioned frequency bands, there are signals that cause aliasing distortion in the reproduced signal. Therefore, in order to prevent aliasing distortion from being mixed into the reproduced signal, a low-pass filter (low-pass filter) used when reproducing the encoded signal is used to filter the signal that is being reproduced. A device with a pass characteristic that allows passing only signals in a frequency band of 1/2 or less of the sampling frequency of the sampling pulse used for encoding is used. If the signal being encoded is sampled using a sampling pulse with a repetition rate of 10KHz, the low-pass filter used when reproducing that signal will not introduce aliasing distortion into the reproduced signal. In order to do this, a filter is used that has a pass characteristic that allows only signals in a frequency band of 1/2 or less of the 10KHz sampling pulse mentioned above to pass, and for example, if the signal to be reproduced is , if the signal was sampled using a sampling pulse with a repetition frequency of 1KHz during encoding, the low-pass filter used when reproducing the signal was designed to prevent aliasing distortion from being mixed into the reproduced signal. In this case, a filter having pass characteristics that allows passing only signals in a frequency band of 1/2 or less of the 1 KHz sampling pulse described above is used. By the way, when the signal reproduction system reproduces a signal sampled with a sampling pulse of a high repetition frequency and a signal sampled with a sampling pulse of a low repetition frequency, for example,
As in the example above, a signal sampled with a 10KHz sampling pulse, i.e., a reproduction signal with a frequency band up to 5KHz, and a signal sampled with a 1KHz sampling pulse, i.e.
When trying to reproduce a reproduction signal having a frequency band up to 0.5KHz, the low-pass filter that should be installed in the reproduction system should be one that can pass signals up to 5KHz. Of course, in the above-mentioned playback system, in order to be able to reproduce the 5KHz signal well, it is necessary to
If a low-pass filter that can pass signals up to 5KHz is provided, aliasing distortion will be mixed into the 0.5KHz signal, but in the present invention, the passband of the low-pass filter is Since the sampling period is changed in accordance with the sampling period set for each frame signal, a high-quality reproduction signal can be easily obtained from digital data with a reduced amount of data with a simple configuration. It is possible.

[Brief explanation of drawings]

FIG. 1 is a block diagram of an embodiment of a recording and reproducing apparatus including the encoding storage and reproducing apparatus of the present invention, FIG. 2 is an explanatory waveform example diagram, and FIGS.
The figure is an explanatory flowchart, and FIG. 5 is a block circuit diagram 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 memory equipment, DAC
1, DAC2...DA converter, VLPF...variable passband type low-pass filter, ASW...analog switch, PC
1, PC2...Photocoupler, Pd...Photodiode, VR...Variable resistor, R1, R4, R5 to R8
...Resistor, R2, R3...Variable resistor, A1, A2...
Amplifier, C1, C2...capacitor.

Claims (1)

[Claims] 1 A signal for each predetermined time length in the signal to be encoded is considered to be one frame signal, and a signal obtained by sampling each one frame signal at a predetermined sampling period is used. means for detecting the number Z of zero points for each frame of signals;
The time length obtained by equally dividing the time length of one frame signal described above by the value Z・K obtained by multiplying the number Z of zero points obtained by the above detection means by a predetermined coefficient K. Each of the states in which a sample value indicating the amplitude value of the original signal obtained by encoding with a corresponding new sampling cycle and data indicating the sampling cycle for obtaining the sample value are paired. An encoding storage/reproduction device that stores digital data for each frame signal in a storage device, reads out and decodes the stored digital data, and includes a sample value indicating the amplitude value of the original signal, and a sample value of the sample value. Regarding the sample value indicating the amplitude value of the original signal in the digital data that is paired with the data indicating the sampling period for obtaining the value, when reading from the storage device described above, zeros in the signal of each one frame are Number of points Z
is read out at time intervals approximately corresponding to the time length obtained by equally dividing the time length of one frame signal described above by a value Z K multiplied by a predetermined coefficient K, and then DA conversion is performed. and means for providing the same as an input signal to a variable passband type low-pass filter;
In addition, a sample value indicating the amplitude value of the original signal described above,
Data indicating the sampling period in digital data that is paired with data indicating the sampling period for obtaining the sample value is converted from DA to the above-mentioned variable passband type low-pass filter. by changing the cut-off frequency of the variable passband type low-pass filter by the control signal obtained by DA converting the data indicating the sampling period in the digital data. , an encoding storage and reproducing device which can obtain a reproduced signal free of harmonic distortion.