JPS6033752A - Noise level reduction system - Google Patents

Noise level reduction system

Info

Publication number
JPS6033752A
JPS6033752A JP14196783A JP14196783A JPS6033752A JP S6033752 A JPS6033752 A JP S6033752A JP 14196783 A JP14196783 A JP 14196783A JP 14196783 A JP14196783 A JP 14196783A JP S6033752 A JPS6033752 A JP S6033752A
Authority
JP
Japan
Prior art keywords
noise
time
waveform
waveforms
periodic
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.)
Granted
Application number
JP14196783A
Other languages
Japanese (ja)
Other versions
JPH0336346B2 (en
Inventor
Seishi Suzuki
鈴木 誠史
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
RADIO RES LAB
Original Assignee
RADIO RES LAB
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Filing date
Publication date
Application filed by RADIO RES LAB filed Critical RADIO RES LAB
Priority to JP14196783A priority Critical patent/JPS6033752A/en
Publication of JPS6033752A publication Critical patent/JPS6033752A/en
Publication of JPH0336346B2 publication Critical patent/JPH0336346B2/ja
Granted legal-status Critical Current

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Classifications

    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208Noise filtering

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  • Engineering & Computer Science (AREA)
  • Computational Linguistics (AREA)
  • Quality & Reliability (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Noise Elimination (AREA)

Abstract

PURPOSE:To reduce remarkably random noise and periodic noise components by subtracting the average value of the past M-set of autocorrelation functions S from the S to connect the waveform in a voice processing system where an input voice signal is converted into the autocorrelation function S for a short time. CONSTITUTION:A waveform 10 where a voice waveform and a periodic noise waveform are supplied is added as a time series signal having a sampling period I. A short time autocorrelation function S(j) 11 is calculated. Then the S(j) is averaged for a long time and the averaged value is decreased from the S(j) to eliminate the periodic noise component. Then the maximum value of the subtracted value is obtained, a period T1 is detected from the delay time of the sample to extract the sample corresponding to the T1. The similar processing is executed by using a time t2 (t2=t1+T1) as a reference. The random noise and the periodic noise components are reduced remarkably by repeating this operation.

Description

【発明の詳細な説明】 本発明は、音声波形に重なった雑音のレベルを低減する
方式に関するものである。
DETAILED DESCRIPTION OF THE INVENTION The present invention relates to a method for reducing the level of noise superimposed on a speech waveform.

工場、工事現場などの騒音や、列車、航空機、自動車な
どの交通騒音の激しい環境で話された音声は、騒音のた
めに聞き取り難く、円滑な会話を行うことは困難である
BACKGROUND ART Voices spoken in environments with heavy traffic noise, such as those of factories, construction sites, and trains, airplanes, and automobiles, are difficult to hear due to the noise, making it difficult to have a smooth conversation.

一方、無線通信による音声の伝送では、伝送途中におけ
る障害のためにS/Nが低下し、通話ができなくなるこ
とがある。
On the other hand, when audio is transmitted by wireless communication, the S/N ratio may drop due to a failure during the transmission, making it impossible to make a call.

騒音環境でS/Nのよい音声を得るため、従来は主とし
て接話マイク、咽喉マイクなどのようにマイクロホンや
ピックアップ方式の開発に頼ってきた。しかし、これら
は騒音レベルの高いときに効果が不充分であったシ、原
理的によい品質の音声が得られないなどの問題がある。
In order to obtain sound with a good S/N ratio in a noisy environment, conventional methods have mainly relied on the development of microphones and pickup methods, such as close-talk microphones and throat microphones. However, these methods have problems such as insufficient effectiveness when the noise level is high and, in principle, good quality audio cannot be obtained.

既に雑°音が重畳したり、ひずみの多い音声のS/Nを
改善して会話に対する疲労度を減らし、了解性を高める
ためには、信号源が音声であることを利用した信号処理
が必要であり、これらに関しては「鈴木誠史 SSS/
Hの低い音声信号から雑音を減らす最近の信号処理技術
”1日経エレクトロニクス、No、281. 昭和57
年1月4日」で紹介されている。しかし、紹介されてい
る多くの手法も、動作、性能、価格などに一長一短があ
るため、実用化されているものはない。
In order to improve the S/N ratio of speech that is already overlaid with noise or has a lot of distortion, reduce conversational fatigue, and improve intelligibility, signal processing that takes advantage of the fact that the signal source is speech is necessary. Regarding these, “Masashi Suzuki SSS/
Recent signal processing technology to reduce noise from low H audio signals” 1 Nikkei Electronics, No. 281. 1977
It was introduced on January 4, 2017. However, many of the methods that have been introduced have advantages and disadvantages in terms of operation, performance, price, etc., so none have been put into practical use.

これらの中で、入力音声信号を短時間自己相関関数に変
換する音声処理方式S P A C(SpeechPr
ocessing system by use of
 Auto−Corre−1a日on functio
n 、特許第1045102号、XX音声処理方式“、
昭和56年5月28日;鈴木誠史、5自己相関関数を利
用した音声処理方式S P A C//。
Among these, the speech processing method SPAC (Speech Pr
ocessing system by use of
Auto-Corre-1a day on function
n, Patent No. 1045102, XX audio processing method",
May 28, 1981; Masashi Suzuki, Speech processing method using 5 autocorrelation functions SPAC//.

電子通信学会、技術研究報告EA75−25.昭和50
年7月)は、評価実験も行われ、ハードウェアも試作さ
れていて有望々方法である。しかし、5PACは、ラン
ダム性雑音の低減能力は大きいが、周期波形を強調する
性質をもっているため、周期性雑音に対しては効果がな
い。
Institute of Electronics and Communication Engineers, Technical Research Report EA75-25. Showa 50
(July 2016), evaluation experiments have been conducted, and hardware has been prototyped, so it is a promising method. However, although 5PAC has a large ability to reduce random noise, it has a property of emphasizing periodic waveforms and is therefore ineffective against periodic noise.

本発明は、音声波形の1周期ごとに短時間自己相関関数
をめ、その相関関数の1周期の波形を接続して音声波形
を合成する過程において、短時間自己相関関数を平均し
た波形を差し引くことが特徴であって、その目的は雑音
が加わった音声波形信号からランダム性雑音とともに周
期性雑音のレベルを低減することである。
The present invention calculates a short-time autocorrelation function for each period of a speech waveform, and in the process of connecting the waveforms of one period of the correlation function to synthesize a speech waveform, subtracts the waveform obtained by averaging the short-time autocorrelation functions. Its purpose is to reduce the level of both random noise and periodic noise from a speech waveform signal to which noise has been added.

以下、図に従って本発明を説明する。第1図は本発明の
方式の原理を説明するだめの、種々の波形とその短時間
自己相関数の関係の例、第2図は本発明の方式の処理過
程の流れ図、第3図は第2図の各段階の波形を示した図
である。第1図で1は音声波形(母音)、3はランダム
雑音波形、5最初に、第1図によって種々の波形とその
短時間自己相関関数Sの関係を説明する。■は母音のよ
うガ周期的音声波形であり、1のSは2のようになり、
1と2の周波数成分は等しい。力お、標本化周期Iで標
本化された入力信号をalとすると、Sは(1)式で計
算される。
The present invention will be explained below with reference to the drawings. Fig. 1 is an example of the relationship between various waveforms and their short-time autocorrelation numbers, which is useful for explaining the principle of the method of the present invention, Fig. 2 is a flowchart of the processing process of the method of the present invention, and Fig. 3 is a flowchart of the processing process of the method of the present invention. FIG. 2 is a diagram showing waveforms at each stage in FIG. 2; In FIG. 1, 1 is a speech waveform (vowel), 3 is a random noise waveform, and 5 is a random noise waveform. First, the relationship between various waveforms and their short-time autocorrelation function S will be explained with reference to FIG. ■ is a periodic speech waveform like a vowel, and S in 1 becomes like 2,
The frequency components of 1 and 2 are equal. Furthermore, if the input signal sampled at the sampling period I is set to al, then S is calculated using equation (1).

Jは遅延時間に対応する標本番号、積分の時間はN1で
ある。ところで、ランダム雑音3のSは4のようになり
、Nを大きくするとインパルスに近づく。また、周期性
雑音を5とすると、どの時刻に関してもそのSは6であ
る。
J is the sample number corresponding to the delay time, and the integration time is N1. By the way, S of random noise 3 becomes 4, and as N becomes larger, it approaches an impulse. Further, if the periodic noise is 5, its S is 6 at any time.

ところで、S PACでは、準周期的とみなされる音声
波形のSは2のような波形になるが、ランダム雑音のS
が4のようなインパルス状になることに着目し、」=0
近傍を除いてSの1周期の波形を切り出して出力信号と
している。その結果、1に3が加わった波形の場合、ラ
ンダム雑音成分を大幅に減少した出力信号を得ている。
By the way, in S PAC, the S of the speech waveform that is considered to be quasi-periodic is a waveform like 2, but the S of random noise is
Paying attention to the fact that it becomes an impulse like 4, ``=0
One cycle of the waveform of S is cut out, excluding the vicinity, and used as an output signal. As a result, in the case of a waveform in which 3 is added to 1, an output signal with significantly reduced random noise components is obtained.

一方、1に5が加わった場合、そのSは2と6の和にな
り、5の成分を除くことはできない。しかし、音声信号
な長時間観測すると定常な部分はわずかであり、波形を
区切って重ね合わせ平均すると、3のようなランダム雑
音に近くなることはよく知られている。しだがって、音
声のSを数多く平均すると、4に近くなることが期待で
きる。一方、5のような周期信号のSは、常に同じ形を
しているので、平均してもその形は変わらない。そこで
、1周期ごとに計算されるSから、Sを平均した波形S
を差し引くと、Sに含まれている周期性雑音成分を大幅
にとり除くことができる。
On the other hand, when 5 is added to 1, S becomes the sum of 2 and 6, and the 5 component cannot be removed. However, it is well known that when an audio signal is observed for a long time, there are only a few stationary parts, and when the waveforms are segmented and averaged over each other, it becomes close to random noise like 3. Therefore, if you average S of many voices, you can expect it to be close to 4. On the other hand, since S of a periodic signal such as 5 always has the same shape, its shape does not change even if it is averaged. Therefore, from the S calculated every cycle, the waveform S which is the average of S
By subtracting , the periodic noise component included in S can be largely removed.

本発明の方式を第2図の流れ図と、第3図の波形によシ
説明する。第2図で、入力には音声波形1と周期性雑音
波形5が加わった波形10が、標本化周期■の時系列信
号として加えられる。10に関して、時刻t1を基準に
して(1)式により短時間自己相関関数S+(J)11
を計算する。なお、(1)式では、tlに対応する標本
を21としている。次に 11から5(J)を長時間に
わたって平均した51(D 12を減じ、Gl (j)
 13をめる。この処理の結果、13では11に含まれ
ていた周期性雑音成分がほとんど取り除かれている。次
に13の極大値をめ、その標本の遅延時間から周期Il
l、を決定し、13から1゛1に相当する標本を切り出
して出力信号16とする。なお、この標本の切出しに際
しては、」=0近傍の標本は使用しない。第2図では、
JがOに近く、振幅が0から立ち上る標本から、1周期
の標本を切り出している。これは、10に含捷れるラン
ダム雑音成分が、4で示したようにJ−0の付近に集ま
り、その影響が13にも残る可能性があるので行う処理
である。
The method of the present invention will be explained with reference to the flowchart of FIG. 2 and the waveforms of FIG. 3. In FIG. 2, a waveform 10 in which a speech waveform 1 and a periodic noise waveform 5 are added is added to the input as a time-series signal with a sampling period ■. 10, the short-time autocorrelation function S+(J)11 is calculated using equation (1) based on time t1.
Calculate. Note that in equation (1), the sample corresponding to tl is set to 21. Next, 51 (D, which is the average of 5 (J) from 11 over a long period of time, subtract 12, and Gl (j)
Get 13. As a result of this processing, in 13, most of the periodic noise components contained in 11 have been removed. Next, find the maximum value of 13 and calculate the period Il from the delay time of that sample.
1 is determined, and a sample corresponding to 1゛1 is cut out from 13 and used as an output signal 16. Note that when cutting out this sample, samples near "=0" are not used. In Figure 2,
A sample of one period is cut out from a sample where J is close to O and the amplitude rises from 0. This process is performed because the random noise components included in 10 may gather near J-0 as shown in 4, and their influence may remain on 13 as well.

次に、10について、時刻t2(t2=’l +TI 
)を基準にして短時間自己相関関数82(J)14を計
算する。14からS (Dを長時間にわたって平均した
52(D ; (12とほとんど同じである。)を減じ
、Gz(D15をめる。15から周期T2を決定し、T
2に相当する標本を切り出し、13がら切り出しだ標本
に接続して出力信号16とする。次に、1oについて、
時刻t3(t3= t2+T2 )を基準にして同様の
処理を行う。この操作を反復することにより、16には
10の中に含まれているランダム雑音と周期性雑音成分
を大幅に低減した信号を得ることができる。
Next, for 10, time t2 (t2='l +TI
) is used as a reference to calculate the short-time autocorrelation function 82(J)14. From 14, subtract 52 (D; which is almost the same as 12), which is the average of D over a long period of time, and subtract Gz (D15. From 15, determine the period T2, and calculate T
A sample corresponding to 2 is cut out and connected to the cut out sample 13 to provide an output signal 16. Next, regarding 1o,
Similar processing is performed using time t3 (t3=t2+T2) as a reference. By repeating this operation, it is possible to obtain a signal in which the random noise and periodic noise components contained in signal 10 are significantly reduced.

以上の説明は有声音に対するものであるが、無声音の場
合は5(J)の極大値から適当に周期を定め、有声音と
同様の処理を行えばよい。
Although the above explanation is for voiced sounds, in the case of unvoiced sounds, the period may be determined appropriately from the maximum value of 5 (J) and the same processing as for voiced sounds may be performed.

なお、本発明の方式において、■は標本化定理を満足す
るように設定する。まだ、積分の時間N1’ u 20
 ms前後、LIモzQms程iK&ル、J:つK、N
やLを設定すればよい。
Note that in the method of the present invention, (2) is set so as to satisfy the sampling theorem. There is still time for integration N1' u 20
Around ms, LIMozQms, iK&L, J:tsuK,N
or L may be set.

一方、S (Dの長時間平均値S (Dには幾つかの計
算法が考えられる。第1の方式は、時刻tkに関して(
2)式でS (Dをめ、処理を行う。
On the other hand, several calculation methods can be considered for the long-term average value S (D of S (D). The first method is that (
2) Determine S (D) in the equation and perform the process.

この方式は、常にtkよりも過去のM個のS (j)の
平均値を使用する。これは移動平均であり、周期性雑音
がゆるやかに変動する場合でも、比較的容易に追従する
ことができる。ただし、平均操作により音声のみのSを
4のような形にするためにはMを大きくとる必要がある
。例えばM= 200 とすると、常に200のS (
Dを記憶するメモリが必要である。
This method always uses the average value of M S (j) past tk. This is a moving average and can be tracked relatively easily even if the periodic noise fluctuates slowly. However, in order to make the voice-only S into a shape like 4 by averaging, it is necessary to set M large. For example, if M = 200, then S (
A memory is required to store D.

ところで、周期性雑音が定常な場合、あるいは極めてゆ
るやかに変化する場合は、M個の平均のS (Dを、次
のM周期の間使用しても実用上差し支えない。換言する
と、時刻ipに関して (3)式%式%( を計算する。5k(J)からは、kに最も近いp(k≧
p)のSp (Dを減じて処理を行う。これを第2の方
式とする。この方式は、第1の方式の場合よシも、周期
性雑音の変動に対する追従が悪く、また、処理を開始し
て最初のM周期の時間が経過するまで、周期性雑音の低
減は行われ々いが、2種類の5p(j) ;現在使って
いるものと次に使うために計算中のものを記憶しておく
だけでよい。
By the way, if the periodic noise is stationary or changes very slowly, there is no practical problem in using M averages of S(D) for the next M periods.In other words, with respect to time ip, (3) Formula % Formula % ( Calculate. From 5k (J), p (k≧
Processing is performed by subtracting Sp (D) of p).This is the second method.This method has poor tracking of periodic noise fluctuations than the first method. Periodic noise is not reduced until the first M periods have elapsed, but two types of 5p(j) are used: the one currently used and the one being calculated for next use. Just remember it.

第3の方式として、(2)式と(3)式の茗(」)の中
間的な平均値として、時定数をもった荷重平均が考えら
れる。これを(4)式で示す。
As a third method, a weighted average with a time constant can be considered as an intermediate average value between the expressions (2) and (3). This is shown by equation (4).

Sk (i) = B :A −8k−m (Dmに1 −B (5k−1(D+A−百に−+ (D ) (4
)ここで、Aは減衰定数(0<A< 1 )、Bは周期
信号のときに、S (DとS (Dを等しくするだめの
定数である。このS (Dでは、時間の経過とともに古
いS (Dの寄与が小さくなり、周期性雑音の変動にも
追従しゃすい。(4)式の計算には、毎回積和計算を必
要とするが、メモリは(3)式を使用する方式と同様で
少ない。
Sk (i) = B :A -8k-m (1 to Dm -B (5k-1(D+A-100-+ (D) (4
) Here, A is the attenuation constant (0<A<1), and B is a constant to make S (D and S (D) equal when the signal is periodic. In this S (D, as time passes, The contribution of the old S (D becomes smaller and it is easier to follow the fluctuations of periodic noise. Calculation of equation (4) requires sum-of-products calculation every time, but the memory method uses equation (3). Same as and less.

次に本発明の実施例について述べる。3.4 kHzに
帯域制限した男声に、妨害波として800Hzの正弦波
を加え、S/Nがは′ぼOdBになるような資料を作成
した。この資料を標本化周期0.1 msで標本化し、
第2図の方法で処理を行った。
Next, examples of the present invention will be described. I created a material in which an 800Hz sine wave was added as an interference wave to a male voice whose band was limited to 3.4kHz, and the S/N ratio was almost OdB. This material was sampled at a sampling period of 0.1 ms,
The treatment was carried out using the method shown in FIG.

(1)式のNは200、Lは170である。(2)式を
使用する第1の方式で、Mを30から250″!!で変
えて出力信号をめたところ、いずれの場合も妨害波の8
00I−1zはほとんど完全に除去され検知できなくな
った。ただ、習には音声のSの平均値が成分として含ま
れ、これはMを無限大にすれば0になるものである。本
実施例のようにMが有限の場合は雑音となり、検知され
るが入力信号に比較するとはるかにききやすい。Mが2
00程度(平均の周期はsms位)のとき、出力音声信
号に対してこの雑音レベルは−13〜−14dBであり
、実質的な妨害の度合は少ない。なお、更にMを大きく
とれば雑音レベルは低くなるが、妨害波の周期波を除去
するまでに時間がかかるため、通常は1〜3秒の区間で
百をめるようにMを設定すればよい。
In formula (1), N is 200 and L is 170. In the first method using formula (2), we measured the output signal by changing M from 30 to 250''!!
00I-1z was almost completely removed and could no longer be detected. However, Xi includes the average value of S of the voice as a component, and this becomes 0 if M is set to infinity. When M is finite as in this embodiment, it becomes noise and is detected, but it is much easier to hear than the input signal. M is 2
00 (average period is about SMS), this noise level is -13 to -14 dB with respect to the output audio signal, and the degree of substantial interference is small. Note that if M is made larger, the noise level will be lowered, but since it takes time to remove the periodic wave of the interference wave, normally it is best to set M to the nearest hundred in an interval of 1 to 3 seconds. good.

次に(3)式を使用する第2の方式について同様の実験
を行った。この場合、最初のSρが計算される(M周期
の時間が経過する。)丑で、周期性雑音の減少は行われ
ないが、その後は第1の方式と同様の効果をあげた。
Next, a similar experiment was conducted regarding the second method using equation (3). In this case, the periodic noise was not reduced when the first Sρ was calculated (time of M periods elapsed), but thereafter the same effect as the first method was obtained.

次に(4)式を使用する第3の方式について同様の実験
を行った。ここで、A二0.98、B=1749とした
ところ、周期性雑音は徐々に減衰し、約2秒経過してか
らは、はとんど検知できなくなった。その後は、第1の
方式と同様の効果をあげた。
Next, a similar experiment was conducted regarding a third method using equation (4). Here, when A2 was set to 0.98 and B was set to 1749, the periodic noise gradually attenuated and became almost undetectable after approximately 2 seconds. After that, the same effect as the first method was obtained.

以上のように、本発明によれば、従来のランダム性雑音
の低減に加えて、音声信号に加わった周期性雑音を大幅
に低減することができるから、本発明は騒音環境下の音
声通信や、無線通信における通話品質の改善に広く利用
することができる。
As described above, according to the present invention, in addition to the conventional reduction of random noise, periodic noise added to voice signals can be significantly reduced. , can be widely used to improve call quality in wireless communications.

【図面の簡単な説明】[Brief explanation of the drawing]

第1図と第3図は本発明の詳細な説明する波形、第2図
は本発明の実施例の流れ図である。 特許出願人 郵政省電波研究所長 …71桔号 第 2 図
FIGS. 1 and 3 are waveforms illustrating detailed explanations of the present invention, and FIG. 2 is a flowchart of an embodiment of the present invention. Patent applicant Director of Radio Research Institute, Ministry of Posts and Telecommunications…71 No. 2 Figure

Claims (3)

【特許請求の範囲】[Claims] (1)音声波形信号を1周期ごとに短時間自己相関関数
Sに変換し、Sの1周期の波形を逐次接続して出力信号
とする音声処理方式において、過去のM(M≧2の整数
)個のSの平均値をSから減じてから波形を接続するこ
とによシ、入力音声波形信号に重なったランダム雑音成
分と周期性雑音成分を大幅に減少した音声波形信号を得
ることを特徴とする雑音レベル低減方式。
(1) In an audio processing method in which an audio waveform signal is converted into a short-time autocorrelation function S every cycle, and the waveforms of one cycle of S are successively connected to produce an output signal, the past M (an integer of M≧2) is used. ) by subtracting the average value of S from S and then connecting the waveforms, a voice waveform signal with significantly reduced random noise components and periodic noise components superimposed on the input voice waveform signal is obtained. Noise level reduction method.
(2)音声波形信号を1周期ごとに短時間自己相関関数
Sに変換し、Sの1周期の波形を逐次接続して出力信号
とする音声処理方式において、M個のSの平均値をめ、
この平均値を次のM個のSからそれぞれ減じてから波形
を接続して出力信号とし、この過程をM周期ごとに反復
することにより、入力音声波形信号に重なったランダム
雑音成分と周期性雑音成分を大幅に減少した音声波形信
号を得ることを特徴とする雑音レベル低減方式。
(2) In an audio processing method in which an audio waveform signal is converted into a short-time autocorrelation function S every cycle, and the waveforms of one cycle of S are successively connected to produce an output signal, the average value of M S is calculated. ,
This average value is subtracted from each of the next M S's, and the waveforms are connected to obtain the output signal. By repeating this process every M periods, random noise components and periodic noise superimposed on the input speech waveform signal are A noise level reduction method characterized by obtaining an audio waveform signal with significantly reduced components.
(3)音声波形信号を1周期ごとに短時間自己相関関数
Sに変換し、Sの1周期の波形を逐次接続して出力信号
とする音声処理方式において、m周期(m≧1)前のS
に、(m−1)回係数A(0<A<1 )を乗じて累積
加算してめたSの荷重平均値を、Sがら減じてから波形
を接続することによシ、入力音声波形に重なったランダ
ム雑音成分と周期性雑音成分を大幅に減少した音声波形
信号を得ることを特徴とする雑音レベル低減方式。
(3) In an audio processing method in which an audio waveform signal is converted into a short-time autocorrelation function S every cycle, and the waveforms of one cycle of S are sequentially connected to produce an output signal, S
By multiplying (m-1) times by a coefficient A (0<A<1) and cumulatively adding the weighted average value of S, subtracting it from S and then connecting the waveforms, the input speech waveform can be obtained. A noise level reduction method characterized by obtaining a speech waveform signal with significantly reduced random noise components and periodic noise components superimposed on the noise.
JP14196783A 1983-08-04 1983-08-04 Noise level reduction system Granted JPS6033752A (en)

Priority Applications (1)

Application Number Priority Date Filing Date Title
JP14196783A JPS6033752A (en) 1983-08-04 1983-08-04 Noise level reduction system

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
JP14196783A JPS6033752A (en) 1983-08-04 1983-08-04 Noise level reduction system

Publications (2)

Publication Number Publication Date
JPS6033752A true JPS6033752A (en) 1985-02-21
JPH0336346B2 JPH0336346B2 (en) 1991-05-31

Family

ID=15304287

Family Applications (1)

Application Number Title Priority Date Filing Date
JP14196783A Granted JPS6033752A (en) 1983-08-04 1983-08-04 Noise level reduction system

Country Status (1)

Country Link
JP (1) JPS6033752A (en)

Cited By (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2016099312A1 (en) * 2015-02-10 2016-06-23 Szechniuk Sławomir Phase filter and method for interference and noise reduction in systems with two signal paths.
US9384756B2 (en) 2014-03-07 2016-07-05 JVC Kenwood Corporation Cyclic noise reduction for targeted frequency bands

Cited By (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US9384756B2 (en) 2014-03-07 2016-07-05 JVC Kenwood Corporation Cyclic noise reduction for targeted frequency bands
WO2016099312A1 (en) * 2015-02-10 2016-06-23 Szechniuk Sławomir Phase filter and method for interference and noise reduction in systems with two signal paths.
US10158386B2 (en) 2015-02-10 2018-12-18 Szechniuk Sławomir Phase filter and method for interference and noise reduction in systems with two signal paths

Also Published As

Publication number Publication date
JPH0336346B2 (en) 1991-05-31

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