JPH06140856A - Voice signal processor - Google Patents
Voice signal processorInfo
- Publication number
- JPH06140856A JPH06140856A JP4311222A JP31122292A JPH06140856A JP H06140856 A JPH06140856 A JP H06140856A JP 4311222 A JP4311222 A JP 4311222A JP 31122292 A JP31122292 A JP 31122292A JP H06140856 A JPH06140856 A JP H06140856A
- Authority
- JP
- Japan
- Prior art keywords
- voice
- signal
- converter
- vowel
- amplitude
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
- Pending
Links
Landscapes
- Tone Control, Compression And Expansion, Limiting Amplitude (AREA)
Abstract
Description
【０００１】[0001]
【産業上の利用分野】本発明は、例えば補聴器、電話、
拡声器、音声通信の分野で用いられる音声信号処理装置
に関するものである。BACKGROUND OF THE INVENTION The present invention relates to hearing aids, telephones,
The present invention relates to a voice signal processing device used in the fields of a loudspeaker and voice communication.
【０００２】[0002]
【従来の技術】本来の音声波形は広いダイナミックレン
ジを持っており、例えば、図６に示す「ねずみ」のよう
な波形をしている。これに対して、例えば、難聴者のよ
うに狭いダイナミックレンジの音声しか聞くことができ
ない場合、あるいは音声を伝送、再生する場合で、伝送
系、再生系のダイナミックレンジが十分広くない場合
に、音声の振幅を圧縮したり、ピーククリッピングした
りすることが多い。2. Description of the Related Art The original voice waveform has a wide dynamic range, and has a waveform such as "mouse" shown in FIG. On the other hand, for example, when hearing only a sound with a narrow dynamic range such as a hearing impaired person, or when transmitting and playing sound and the dynamic range of the transmission system and the reproduction system is not wide enough, Often, the amplitude of is compressed or peak clipping is performed.
【０００３】例として、図６に示す音声波形を増幅した
後にピーククリッピングした音声波形を図７に示す。図
６の波形と図７の波形とを比べると、許容振幅レベルが
同じなら、図７の波形の方がより多くの音声エネルギを
耳に入れることができることが判る。As an example, FIG. 7 shows a voice waveform obtained by peak clipping after amplifying the voice waveform shown in FIG. Comparing the waveform of FIG. 6 with the waveform of FIG. 7, it can be seen that more audio energy can be put into the ear with the waveform of FIG. 7 if the allowable amplitude level is the same.
【０００４】[0004]
【発明が解決しようとする課題】ところで、上述した従
来の音声振幅処理においては、音声のダイナミックレン
ジを単に小さくしているために、振幅の相対的な変化が
小さくなり、次のような問題点があった。By the way, in the above-mentioned conventional voice amplitude processing, since the dynamic range of the voice is simply reduced, the relative change of the amplitude becomes small, and the following problems occur. was there.
【０００５】（１）処理後の音声信号は相対的に大きな
振幅が連続的に続くことが多くなり、聴覚的に疲労す
る。 （２）本来、音声には相対的に大きな振幅を持つ母音中
央部分と比較的小さい振幅を持つ子音部分や母音の最後
の部分があり、聴覚的に見ると音声の区切りが明確であ
るが、従来の振幅処理をすると、上述したように相対的
に大きな振幅が連続することが多くなり、継時マスキン
グを受け易くなったり、聴覚的な音声の区切りがはっき
りしなくなるために言語中枢での解析に混乱が生じる等
の原因で、音声の明瞭度が低下する。(1) The processed audio signal often has a relatively large amplitude continuously, which causes auditory fatigue. (2) Originally, a voice has a vowel central part having a relatively large amplitude, a consonant part having a relatively small amplitude, and the last part of a vowel. When the conventional amplitude processing is performed, relatively large amplitudes are often continuous, as described above, which makes it easier to receive continuous masking and makes it difficult to distinguish auditory speech boundaries. The intelligibility of the voice is reduced due to confusion.
【０００６】本発明はこのような背景に鑑みてなされた
ものであり、上記従来技術の欠点を解消し、ダイナミッ
クレンジの狭さに対処した振幅レベルの圧縮処理を前提
とし、聴覚的疲労を抑え、かつ、明瞭度の低下を防止す
ることができる音声信号処理装置を提供することを目的
とする。The present invention has been made in view of such a background, and is based on the premise of amplitude level compression processing that solves the above-mentioned drawbacks of the prior art and copes with a narrow dynamic range, and suppresses auditory fatigue. It is also an object of the present invention to provide an audio signal processing device capable of preventing a decrease in clarity.
【０００７】[0007]
【課題を解決するための手段】上記目的を達成するため
本発明は、振幅レベルが圧縮された音声信号をデジタル
信号に変換するＡ／Ｄ変換器と、前記Ａ／Ｄ変換器の出
力信号を取り込んで母音、子音の解析を行い母音区間の
終端を検出する音声解析部と、前記Ａ／Ｄ変換器により
デジタル信号に変換された音声データをそのまま格納す
る音声メモリと、前記音声解析部からの母音区間終端検
出信号を受けて、前記音声メモリに格納された音声デー
タを取り出し、母音区間終了時点から遡って所定時間そ
の振幅を抑圧した音声制御指令信号を出力する制御手段
と、前記制御手段からの音声制御指令信号をアナログ信
号に変換するＤ／Ａ変換器と、前記Ｄ／Ａ変換器の出力
信号に基づいて信号処理済みの再生音声信号を出力する
音声出力部とを備えたことを特徴とする。To achieve the above object, the present invention provides an A / D converter for converting a voice signal whose amplitude level is compressed into a digital signal, and an output signal of the A / D converter. A voice analysis unit that captures and analyzes vowels and consonants to detect the end of a vowel section, a voice memory that directly stores voice data converted into a digital signal by the A / D converter, and a voice analysis unit from the voice analysis unit. From the control means for receiving the vowel section end detection signal, extracting the voice data stored in the voice memory, and outputting the voice control command signal whose amplitude is suppressed for a predetermined period of time from the end point of the vowel section. A D / A converter for converting the audio control command signal of the above into an analog signal, and an audio output section for outputting a reproduced audio signal that has been signal-processed based on the output signal of the D / A converter. Characterized in that was.
【０００８】また、本発明は、振幅レベルが圧縮された
音声信号をデジタル信号に変換するＡ／Ｄ変換器と、前
記Ａ／Ｄ変換器の出力信号を取り込んで母音、子音の解
析を行い母音区間の始端および終端を検出する音声解析
部と、前記Ａ／Ｄ変換器によりデジタル信号に変換され
た音声データをそのまま格納する音声メモリと、前記音
声解析部からの母音区間始端検出信号を受けて、前記音
声メモリに格納された音声データを取り出し、母音区間
開始時点から一定時間経過後に、前記音声解析部からの
母音区間終端検出信号に基づいて、前記一定時間経過後
の時点から母音区間終了時点までその振幅を抑圧した音
声制御指令信号を出力する制御手段と、前記制御手段か
らの音声制御指令信号をアナログ信号に変換するＤ／Ａ
変換器と、前記Ｄ／Ａ変換器の出力信号に基づいて信号
処理済みの再生音声信号を出力する音声出力部とを備え
たことを特徴とする。Further, according to the present invention, an A / D converter for converting a voice signal whose amplitude level has been compressed into a digital signal, and an output signal of the A / D converter are taken in to analyze vowels and consonants and vowels. A voice analysis unit for detecting the beginning and end of a section, a voice memory for directly storing the voice data converted into a digital signal by the A / D converter, and a vowel section start end detection signal from the voice analysis unit. , Taking out the voice data stored in the voice memory, and after a lapse of a fixed time from the start point of the vowel section, based on the vowel section end detection signal from the voice analysis unit, from the point after the lapse of the fixed time to the end point of the vowel section Control means for outputting a voice control command signal whose amplitude has been suppressed, and a D / A for converting the voice control command signal from the control means into an analog signal.
It is characterized by comprising a converter and an audio output section for outputting a reproduced audio signal which has been subjected to signal processing based on an output signal of the D / A converter.
【０００９】さらに、本発明は、前記制御手段は、振幅
抑圧区間の始端と終端に隣接する所定時間の振幅をスロ
ープ状に制御するものとした。Further, according to the present invention, the control means controls the amplitude of a predetermined time adjacent to the start end and the end of the amplitude suppression section in a slope shape.
【００１０】[0010]
【作用】請求項１記載の本発明によれば、振幅レベルが
圧縮された音声信号の母音区間が終了した時点から遡っ
て所定時間（数ミリ秒〜十数ミリ秒程度）その振幅を抑
圧（減衰あるいはゼロ）する。請求項２記載の本発明に
よれば、母音区間の途中から振幅抑圧区間を設ける。請
求項３記載の本発明によれば、振幅抑圧をしないところ
と振幅抑圧区間との振幅の不連続の不自然さを緩和する
ため、両者の境界の前後５〜２０ミリ秒の区間をスロー
プ状にして振幅を制御する。According to the first aspect of the present invention, the amplitude of the voice signal whose amplitude level is compressed is suppressed for a predetermined time (several milliseconds to several tens of milliseconds) starting from the end of the vowel section of the audio signal (around a few milliseconds). Decay or zero). According to the second aspect of the present invention, the amplitude suppression section is provided in the middle of the vowel section. According to the present invention as set forth in claim 3, in order to mitigate the unnaturalness of the discontinuity of the amplitude between the non-amplitude suppression area and the amplitude suppression area, a section of 5 to 20 milliseconds before and after the boundary between the two areas is sloped. To control the amplitude.
【００１１】[0011]
【実施例】以下、本発明の実施例を図面に基づいて説明
する。図１は本発明の実施例に係る音声信号処理装置の
ブロック図であり、入力音声信号（アナログ信号）は、
振幅レベルが周知の方式により圧縮されたものを前提と
している。Embodiments of the present invention will be described below with reference to the drawings. FIG. 1 is a block diagram of an audio signal processing device according to an embodiment of the present invention, in which an input audio signal (analog signal) is
It is assumed that the amplitude level is compressed by a known method.
【００１２】図１において、１はその音声信号を入力し
てデジタル信号に変換するＡ／Ｄ変換器、２はＡ／Ｄ変
換器１の出力信号を取り込んで後述する内容の音声解析
を行う音声解析部、３は音声解析部２の出力信号を取り
込むＣＰＵ（制御手段）、４はＣＰＵ３と信号の授受を
行うと共にデジタル変換された音声データを格納する音
声メモリ、５はＣＰＵ３からの音声制御指令信号をアナ
ログ信号に変換するＤ／Ａ変換器、６は本発明による処
理が施された再生音声信号を出力する音声出力部であ
る。In FIG. 1, reference numeral 1 is an A / D converter that inputs the audio signal and converts it into a digital signal, and 2 is a voice that takes in the output signal of the A / D converter 1 and performs audio analysis of the contents described later. An analyzing unit 3, a CPU (control means) for fetching the output signal of the voice analyzing unit 2, 4 a voice memory for transmitting and receiving signals to and from the CPU 3, and storing digitally converted voice data, and 5 a voice control command from the CPU 3. A D / A converter for converting a signal into an analog signal, and 6 is an audio output unit for outputting a reproduced audio signal processed by the present invention.
【００１３】次に、第１の実施例の機能（音声解析部２
の機能）、動作を説明する。この実施例においては、音
声解析部２は線形予測分析法やＰＡＲＣＯＲ分析法等に
よって、振幅レベルが圧縮されたデジタル音声信号の母
音区間を検出し、さらに母音区間の終端を検出して、母
音区間終端検出信号を出力する。一方、音声メモリ４に
はそのデジタル音声信号がそのまま格納される。Next, the function of the first embodiment (speech analysis unit 2
Function) and operation. In this embodiment, the voice analysis unit 2 detects a vowel section of a digital voice signal whose amplitude level is compressed by a linear prediction analysis method, a PARCOR analysis method, or the like, and further detects the end of the vowel section to detect the vowel section. Output the end detection signal. On the other hand, the digital voice signal is stored in the voice memory 4 as it is.
【００１４】ＣＰＵ３は、音声解析部２からの母音区間
終端検出信号を受けると、音声メモリ４内に予め格納さ
れている音声データを読み出して、母音区間の終端から
遡ってｔミリ秒の間の振幅を減少させるかまたはゼロに
する制御を行った後、Ｄ／Ａ変換器５に音声制御指令信
号を出力する。この信号はＤ／Ａ変換器５でアナログ信
号に変換され、音声出力部６から信号処理済みの再生音
声信号として出力される。Upon receiving the vowel section end detection signal from the voice analysis unit 2, the CPU 3 reads out the voice data stored in advance in the voice memory 4 and traces back from the end of the vowel section for t milliseconds. After controlling the amplitude to decrease or to zero, a voice control command signal is output to the D / A converter 5. This signal is converted into an analog signal by the D / A converter 5, and is output from the audio output unit 6 as a reproduced audio signal that has undergone signal processing.
【００１５】図２は第１の実施例に係る再生音声信号波
形図であり、この音声波形は図７に示す振幅処理済みの
音声波形を基にしている。この図から明らかなように、
母音区間の最後の部分に振幅がゼロになる部分が存在し
ている。FIG. 2 is a reproduced voice signal waveform diagram according to the first embodiment, and this voice waveform is based on the voice waveform after amplitude processing shown in FIG. As you can see from this figure,
There is a part where the amplitude is zero in the last part of the vowel section.
【００１６】なお、音声データが音声メモリ４に格納さ
れてから少なくともｔミリ秒より後に、再びＣＰＵ３に
よりこの音声データを外部に読み出すことになるので、
音声メモリ４は最低この時間だけ音声を記憶できるメモ
リ容量が必要になる。ここで、ｔは前述したように数ミ
リ秒〜十数ミリ秒の範囲に設定される。Since at least t milliseconds after the voice data is stored in the voice memory 4, the voice data is read out to the outside again by the CPU 3.
The voice memory 4 needs to have a memory capacity capable of storing voice for at least this time. Here, t is set in the range of several milliseconds to ten and several milliseconds as described above.
【００１７】また、本装置を適用すると、母音部分がｔ
ミリ秒だけ短くなるが、母音部分の長さは多くの場合１
００ミリ秒程度あり、本来冗長であるので、最後の部分
が数ミリ秒欠損したからといって音声言語解析に悪い影
響を与えることはなく、むしろ無音区間が生じることに
よる効果の方が大きい。When this device is applied, the vowel part is t
Shortened by milliseconds, but the length of the vowel part is often 1
Since it is about 00 milliseconds and is inherently redundant, the fact that the last part is missing for a few milliseconds does not have a bad influence on the spoken language analysis, but rather has the effect of producing a silent section.
【００１８】次に第２の実施例について説明する。この
実施例においては、音声解析部２は母音の始端および終
端を検出し、ＣＰＵ３に対して母音区間始端検出信号お
よび母音区間終端検出信号を出力する。ＣＰＵ３は母音
区間開始時点から一定時間が経過すると、その時点から
母音区間終了時点まで、その振幅を抑圧した音声制御指
令信号を出力する。そして音声出力部６からはこの指令
信号に基づいた再生音声信号が出力される。Next, a second embodiment will be described. In this embodiment, the voice analysis unit 2 detects the start and end of a vowel and outputs a vowel section start detection signal and a vowel section end detection signal to the CPU 3. When a certain time elapses from the start point of the vowel section, the CPU 3 outputs a voice control command signal whose amplitude is suppressed from that point to the end point of the vowel section. Then, the audio output unit 6 outputs a reproduced audio signal based on the command signal.
【００１９】図３は第２の実施例の再生音声信号波形の
基になる原音波形（振幅レベル圧縮前の音声波形）であ
る。この原音波形に振幅処理を施した後、第２の実施例
に係る処理を行うと、図４に示すような再生音声信号波
形が得られる。この図から明らかなように、母音区間の
中間部から終了時点に渡って振幅がゼロの部分が存在す
る。FIG. 3 shows the original sound waveform (sound waveform before amplitude level compression) which is the basis of the reproduced sound signal waveform of the second embodiment. When amplitude processing is applied to the original sound waveform and then the processing according to the second embodiment is performed, a reproduced voice signal waveform as shown in FIG. 4 is obtained. As is clear from this figure, there is a portion where the amplitude is zero from the middle part of the vowel section to the end point.
【００２０】最後に第３の実施例について説明する。こ
の実施例においては、振幅抑圧をしないところと振幅抑
圧区間との振幅の不連続の不自然さを緩和するため、両
者の境界の前後５〜２０ミリ秒の区間をスロープ状にし
て振幅を制御するようにしたものである。Finally, a third embodiment will be described. In this embodiment, in order to mitigate the unnaturalness of the discontinuity of the amplitude between the place where the amplitude is not suppressed and the amplitude suppression section, the amplitude is controlled by making a section of 5 to 20 milliseconds before and after the boundary between them into a slope shape. It is something that is done.
【００２１】図４に示す波形に適用すると図５に示すよ
うな波形となる。図５の例では十ミリ秒のスロープ区間
を設けている。なお、第１の実施例に示す波形にも適用
できることは言うまでもない。When applied to the waveform shown in FIG. 4, a waveform as shown in FIG. 5 is obtained. In the example of FIG. 5, a slope section of 10 milliseconds is provided. Needless to say, it can be applied to the waveforms shown in the first embodiment.
【００２２】[0022]
【発明の効果】請求項１記載の本発明によれば、振幅レ
ベルが圧縮された音声信号の母音区間が終了した時点か
ら遡って所定時間（数ミリ秒〜十数ミリ秒程度）その振
幅を抑圧（減衰あるいはゼロ）し、また請求項２記載の
本発明によれば、母音区間の途中から振幅抑圧区間を設
けるようにしたので、音声の明瞭度の低下を防止するこ
とができ、また、大振幅の波形が連続して続くことがな
くなるので、聴覚的疲労が少なくなる効果がある。また
請求項３記載の本発明によれば、振幅抑圧をしない区間
とする区間の境界の前後所定区間の振幅をスロープ状に
したので、上記の効果を一層高めることができる。According to the first aspect of the present invention, the amplitude of the voice signal whose amplitude level is compressed is traced back for a predetermined time (several milliseconds to ten and several milliseconds) from the end of the vowel section. According to the present invention described in claim 2, the amplitude suppression section is provided in the middle of the vowel section, so that it is possible to prevent a decrease in the intelligibility of the voice, and Since the large-amplitude waveform does not continue continuously, it is effective in reducing auditory fatigue. Further, according to the present invention as set forth in claim 3, since the amplitude of the predetermined section before and after the boundary of the section where the amplitude is not suppressed is made into a slope shape, the above effect can be further enhanced.
【図１】本発明の実施例に係る音声信号処理装置のブロ
ック図である。FIG. 1 is a block diagram of an audio signal processing device according to an embodiment of the present invention.
【図２】第１の実施例に係る処理を施した再生音声信号
波形図である。FIG. 2 is a reproduced audio signal waveform diagram subjected to the processing according to the first embodiment.
【図３】原音波形図である。FIG. 3 is an original sound wave diagram.
【図４】図３に示す原音に対して第２の実施例に係る処
理を施した再生音声信号波形図である。FIG. 4 is a waveform diagram of a reproduced audio signal obtained by performing the processing according to the second embodiment on the original sound shown in FIG.
【図５】図３に示す原音に対して第３の実施例に係る処
理を施した再生音声信号波形図である。5 is a reproduced voice signal waveform diagram in which the process according to the third embodiment is applied to the original sound shown in FIG.
【図６】他の原音波形図である。FIG. 6 is another original sound wave diagram.
【図７】図６に示す原音波形に本発明の前提となる振幅
処理を施した波形図である。7 is a waveform diagram in which the original sound waveform shown in FIG. 6 is subjected to amplitude processing which is a premise of the present invention.
１ Ａ／Ｄ変換器 ２ 音声解析部 ３ ＣＰＵ（制御手段） ４ 音声メモリ ５ Ｄ／Ａ変換器 ６ 音声出力部 1 A / D converter 2 Voice analysis unit 3 CPU (control means) 4 Voice memory 5 D / A converter 6 Voice output unit
Claims (3)
タル信号に変換するＡ／Ｄ変換器と、 前記Ａ／Ｄ変換器の出力信号を取り込んで母音、子音の
解析を行い母音区間の終端を検出する音声解析部と、 前記Ａ／Ｄ変換器によりデジタル信号に変換された音声
データをそのまま格納する音声メモリと、 前記音声解析部からの母音区間終端検出信号を受けて、
前記音声メモリに格納された音声データを取り出し、母
音区間終了時点から遡って所定時間その振幅を抑圧した
音声制御指令信号を出力する制御手段と、 前記制御手段からの音声制御指令信号をアナログ信号に
変換するＤ／Ａ変換器と、 前記Ｄ／Ａ変換器の出力信号に基づいて信号処理済みの
再生音声信号を出力する音声出力部と、 を備えたことを特徴とする音声信号処理装置。1. An A / D converter for converting a voice signal whose amplitude level is compressed into a digital signal, and an output signal of the A / D converter is taken in to analyze a vowel and a consonant to determine the end of a vowel section. A voice analysis unit for detecting, a voice memory for directly storing the voice data converted into a digital signal by the A / D converter, and a vowel section end detection signal from the voice analysis unit,
The voice data stored in the voice memory is taken out, and the voice control command signal from the control unit is converted into an analog signal by outputting a voice control command signal whose amplitude is suppressed for a predetermined time dating back from the end point of the vowel section. An audio signal processing device comprising: a D / A converter for converting; and an audio output unit for outputting a reproduced audio signal that has been signal-processed based on an output signal of the D / A converter.
タル信号に変換するＡ／Ｄ変換器と、 前記Ａ／Ｄ変換器の出力信号を取り込んで母音、子音の
解析を行い母音区間の始端および終端を検出する音声解
析部と、 前記Ａ／Ｄ変換器によりデジタル信号に変換された音声
データをそのまま格納する音声メモリと、 前記音声解析部からの母音区間始端検出信号を受けて、
前記音声メモリに格納された音声データを取り出し、母
音区間開始時点から一定時間経過後に、前記音声解析部
からの母音区間終端検出信号に基づいて、前記一定時間
経過後の時点から母音区間終了時点までその振幅を抑圧
した音声制御指令信号を出力する制御手段と、 前記制御手段からの音声制御指令信号をアナログ信号に
変換するＤ／Ａ変換器と、 前記Ｄ／Ａ変換器の出力信号に基づいて信号処理済みの
再生音声信号を出力する音声出力部と、 を備えたことを特徴とする音声信号処理装置。2. An A / D converter for converting a voice signal whose amplitude level is compressed into a digital signal, and an output signal of the A / D converter is taken in to analyze a vowel and a consonant to start and end a vowel section. A voice analysis unit that detects the end, a voice memory that directly stores the voice data converted into a digital signal by the A / D converter, and a vowel section start end detection signal from the voice analysis unit,
Taking out the voice data stored in the voice memory, after a lapse of a fixed time from the start point of the vowel section, based on the vowel section end detection signal from the voice analysis unit, from the time point after the lapse of the fixed time to the end point of the vowel section Based on a control unit that outputs a voice control command signal whose amplitude is suppressed, a D / A converter that converts the voice control command signal from the control unit into an analog signal, and an output signal of the D / A converter. An audio signal processing device comprising: an audio output unit that outputs a reproduced audio signal that has undergone signal processing.
終端に隣接する所定時間の振幅をスロープ状に制御する
請求項１または２記載の音声信号処理装置。3. The audio signal processing device according to claim 1, wherein the control means controls the amplitude of a predetermined time adjacent to the start end and the end of the amplitude suppression section in a slope shape.
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| JP4311222A JPH06140856A (en) | 1992-10-26 | 1992-10-26 | Voice signal processor |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| JP4311222A JPH06140856A (en) | 1992-10-26 | 1992-10-26 | Voice signal processor |
Publications (1)
| Publication Number | Publication Date |
|---|---|
| JPH06140856A true JPH06140856A (en) | 1994-05-20 |
Family
ID=18014572
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| JP4311222A Pending JPH06140856A (en) | 1992-10-26 | 1992-10-26 | Voice signal processor |
Country Status (1)
| Country | Link |
|---|---|
| JP (1) | JPH06140856A (en) |
Cited By (1)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| US7890323B2 (en) | 2004-07-28 | 2011-02-15 | The University Of Tokushima | Digital filtering method, digital filtering equipment, digital filtering program, and recording medium and recorded device which are readable on computer |
-
1992
- 1992-10-26 JP JP4311222A patent/JPH06140856A/en active Pending
Cited By (1)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| US7890323B2 (en) | 2004-07-28 | 2011-02-15 | The University Of Tokushima | Digital filtering method, digital filtering equipment, digital filtering program, and recording medium and recorded device which are readable on computer |
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