JPH10161694A - Band split type noise reducing method - Google Patents

Band split type noise reducing method

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Publication number
JPH10161694A
JPH10161694A JP8317786A JP31778696A JPH10161694A JP H10161694 A JPH10161694 A JP H10161694A JP 8317786 A JP8317786 A JP 8317786A JP 31778696 A JP31778696 A JP 31778696A JP H10161694 A JPH10161694 A JP H10161694A
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JP
Japan
Prior art keywords
noise
signal
band
input signal
bands
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
JP8317786A
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Japanese (ja)
Other versions
JP3454402B2 (en
Inventor
Jiyunko Sasaki
潤子 佐々木
Yoichi Haneda
陽一 羽田
Junji Kojima
順治 小島
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Nippon Telegraph and Telephone Corp
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Nippon Telegraph and Telephone Corp
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Priority to JP31778696A priority Critical patent/JP3454402B2/en
Publication of JPH10161694A publication Critical patent/JPH10161694A/en
Application granted granted Critical
Publication of JP3454402B2 publication Critical patent/JP3454402B2/en
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Expired - Lifetime legal-status Critical Current

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  • Noise Elimination (AREA)

Abstract

PROBLEM TO BE SOLVED: To prevent deterioration of quality from being accompanied even when the band split number is minimized. SOLUTION: An input signal is band-divided, and with respect to a signal XK(n) every band, a power PX.k (n) is determined (24), whereby a noise average power PavN.k (n) is estimated (51). SNR)k (n) is estimated by the PX.k (n), PavN.k (n) (27), and a gain factor G(SNRk (n)) is determined by SNRk (n) (30). The addition ratio αaccording to the ratio of voice to noise is determined by SNRk (n) so that it is large when S/N is poor, Xk (n) is multiplied by G(SNRk (n)) to provide a signal Pk (n)' reduced in noise (28), Xk (n) is added to Yk (n)' according to αby αX.k (n)+(1-α)Yk (n)' to provide Yk (n) (53), and all the bands are composed and converted into time area.

Description

【発明の詳細な説明】DETAILED DESCRIPTION OF THE INVENTION

【0001】[0001]

【発明の属する技術分野】この発明は、音声会議装置・
TV会議装置等の音声/音響装置等において、目的とな
る音声信号と不要な雑音等の信号が混在する入力信号か
ら、雑音を低減した音声信号を出力する雑音低減方法に
関する。
BACKGROUND OF THE INVENTION 1. Field of the Invention
The present invention relates to a noise reduction method for outputting a noise-reduced voice signal from an input signal in which a target voice signal and a signal such as unnecessary noise are mixed in a voice / sound device such as a TV conference device.

【0002】[0002]

【従来の技術】音声会議・TV会議等の拡声通話系で
は、マイクロホンで受音し、相手側に送出される送話信
号に目的となる音声以外の周囲雑音等が混入すると、音
声の明瞭性が損なわれ通話品質が著しく劣化する。この
為、送話信号に含まれる目的音声以外の周囲雑音を低減
する事が強く求められている。
2. Description of the Related Art In a voice communication system such as a voice conference or a TV conference, if a surrounding signal other than a target voice is mixed in a transmission signal received by a microphone and transmitted to a partner side, the clarity of the voice is increased. And the call quality is significantly degraded. For this reason, there is a strong demand for reducing ambient noise other than the target voice included in the transmission signal.

【0003】雑音低減方法とは、目的となる音声信号と
不要な周囲雑音等の信号が混在する入力信号から、雑音
を低減した信号を出力する技術である。図2Aは収音シ
ステムを示すもので、これを用いて従来の雑音低減方法
を説明する。この明細書においては、信号の時間表現は
離散時間を表わす整数値nを用いて、例えばX(n)と
表わす。今発声者12が発声した目的とする音声信号1
3をS(n)、空調などの不要な周囲雑音14をN
(n)、これら音声信号13と雑音14とがマイクロホ
ン11で受音されて雑音低減装置16へ入力される入力
信号15をX(n)、雑音低減装置16の出力信号17
をY(n)とする。雑音低減装置16への入力信号X
(n)には、目的となる音声信号S(n)以外に周囲雑
音N(n)が混入している。即ち X(n)=S(n)+N(n) (1) と表わされる。この時、入力信号X(n)中の雑音N
(n)を低減し、目的となる音声信号S(n)に近い信
号を出力信号Y(n)として取り出す装置を雑音低減装
置と呼ぶ。
[0003] The noise reduction method is a technique of outputting a noise-reduced signal from an input signal in which a target voice signal and a signal such as unnecessary ambient noise are mixed. FIG. 2A shows a sound collection system, which will be used to explain a conventional noise reduction method. In this specification, the time representation of a signal is represented by, for example, X (n) using an integer value n representing a discrete time. The target audio signal 1 uttered by the speaker 12 now
3 is S (n) and unnecessary ambient noise 14 such as air conditioning is N
(N) The input signal 15 received by the microphone 11 when the voice signal 13 and the noise 14 are received by the microphone 11 and the input signal 15 is input to the noise reduction device 16 is X (n), and the output signal 17 of the noise reduction device 16 is output.
Is Y (n). Input signal X to noise reduction device 16
(N) contains ambient noise N (n) in addition to the target audio signal S (n). That is, X (n) = S (n) + N (n) (1) At this time, the noise N in the input signal X (n)
A device that reduces (n) and extracts a signal close to the target audio signal S (n) as an output signal Y (n) is called a noise reduction device.

【0004】図3は、Spectrum Subtraction(S.F.Bol
l, IEEE Trans. on ASSP, vol.27, no.2, pp.113-120,
Apr(1979)). Wiener Filer(J.S.Lim. & A.V.Oppenheim,
in Proc. IEEE, vol.67, no.12, pp.1586-1604, Dec(1
979)). Maximum Likelihood Envelop(R.J.McAulay &
M.L.Malpass, IEEE Trans. on ASSP, vol.28, no.2, p
p.137-145, Apr(1980)). minimum mean squared error
method(MMSE)(Y.Ephraim& D.Malah, IEEE Trans. on AS
SP, vol.32, no.6, pp.1109-1121, Dec(1984)).等の短
時間スペクトラル振幅(STSA)評価(Short Time S
pectral Amplitude(STSA) Estimation)を基礎とした雑
音低減方式で、従来使用されている方法の機能構成を示
すものである。これを用いて従来の雑音低減方法を説明
する。図2Aと同一の要素には共通の記号を用いた。
FIG. 3 shows Spectrum Subtraction (SFBol
l, IEEE Trans. on ASSP, vol.27, no.2, pp.113-120,
Apr (1979)). Wiener Filer (JSLim. & AVOppenheim,
in Proc.IEEE, vol.67, no.12, pp.1586-1604, Dec (1
979)). Maximum Likelihood Envelop (RJMcAulay &
MLMalpass, IEEE Trans.on ASSP, vol.28, no.2, p
p.137-145, Apr (1980)). minimum mean squared error
method (MMSE) (Y.Ephraim & D.Malah, IEEE Trans.on AS
SP, vol.32, no.6, pp.1109-1121, Dec (1984)) etc. (Short Time S)
This is a noise reduction method based on Spectral Amplitude (STSA) Estimation, and shows the functional configuration of a conventionally used method. Using this, a conventional noise reduction method will be described. The same symbols are used for the same elements as in FIG. 2A.

【0005】まず、マイクロホン11により受音され
た、目的信号と不要な雑音とが混入する入力信号15を
A/D変換部21においてデジタル化し、周波数帯域分
割部22と雑音判別部23に転送する。周波数帯域分割
部22では、転送された信号が複数の周波数帯域に分割
される。周波数帯域への分割は、例えば離散的フーリエ
変換等を用いて行う。ここで、帯域分割された信号は一
般に複素数であるが、分割方法によっては実数となる場
合もある。ここでは一般的に、複素数を仮定して議論す
るが、実数の場合も同じ議論が可能である。周波数帯域
に分割されたk番目の周波数帯域の信号を Xk (n) =Xk,r (n) +jXk,i (n) (2) (Xk,r ,Xk,i はそれぞれXk (n) の実数部分と虚数
部分)とすると、Xk (n) は、入力信号パワー計算部2
4、入力信号位相計算部25、雑音パワー計算部26に
転送される。入力信号パワー計算部24では、各帯域ご
との入力信号のパワーレベル PX,k (n) =Xk,r (n)2+Xk,i (n)2 (3) が、入力信号位相計算部25では各帯域ごとの位相 Φk (n) =tan -1[Xk,i (n) /Xk,r (n) ] (4) それぞれが計算される。その後PX,k (n) はS/N比推
定部27およびゲインファクター挿入部28に転送さ
れ、Φk (n) は時間領域変換部29に転送される。一
方、雑音判別部23ではA/D変換部21から転送され
たきたX(n)に対して、まずパワーレベル PX (n) =Σ{X(n-k) }2 (5) Σはk=0からL−1までが計算される。ここで、Lは
積分時間を表わす。次に例えば予め決められたしきい値
thに対し、 PX (n) <Pth (6) の判定が行われ、この条件式を満たした場合には、雑音
であると判別する。雑音パワー計算部26では、雑音判
別部23において入力信号X(n)が雑音であると判定
された時のみ、雑音の各帯域ごとのパワーレベルを PN,k (n) =Xk,r (n)2+Xk,i (n)2 (7) として計算し、その時間平均PavN,k (n) をS/N比推
定部27に転送する。時間平均は、例えば PavN,k (n) =(1/A)Σm γm N,k (n−m) (8) と計算される。ここでγm は例えば、 γm =(γ)m (9) と表わされるような指数重みづけの係数で(γ<1)、
Aは (1/A)Σm γm =1 (10) となる正規化の為の定数である。
First, an A / D converter 21 digitizes an input signal 15 mixed with a target signal and unnecessary noise received by a microphone 11, and transfers the digitized signal to a frequency band division unit 22 and a noise discrimination unit 23. . In the frequency band dividing unit 22, the transferred signal is divided into a plurality of frequency bands. Division into frequency bands is performed using, for example, a discrete Fourier transform or the like. Here, the band-divided signal is generally a complex number, but may be a real number depending on the division method. Here, in general, discussion will be made assuming a complex number, but the same argument can be made for a real number. X k (n) = X k, r (n) + jX k, i (n) (2) where X k, r , X k, i are X k (n), the real part and the imaginary part of X (n)), X k (n) is
4. The signal is transferred to the input signal phase calculator 25 and the noise power calculator 26. The input signal power calculator 24 calculates the input signal power level P X, k (n) = X k, r (n) 2 + X k, i (n) 2 (3) for each band. The unit 25 calculates the phase Φ k (n) = tan -1 [X k, i (n) / X k, r (n)] (4) for each band. Thereafter, P X, k (n) is transferred to the S / N ratio estimating unit 27 and the gain factor inserting unit 28, and Φ k (n) is transferred to the time domain transforming unit 29. On the other hand, in the noise discrimination unit 23, first, for X (n) transferred from the A / D conversion unit 21, the power level P X (n) = {X (nk) 2 (5)} is k = 0 to L-1 are calculated. Here, L represents the integration time. Next, for example, a determination of P X (n) <P th (6) is performed for a predetermined threshold value P th , and if this conditional expression is satisfied, it is determined that the noise is present. The noise power calculator 26 sets the power level of each noise band to P N, k (n) = X k, r only when the noise discriminator 23 determines that the input signal X (n) is noise. (n) 2 + X k, i (n) 2 (7) and the time average Pav N, k (n) is transferred to the S / N ratio estimating unit 27. Time average, for example Pav N, k (n) = (1 / A) Σ m γ m P N, is calculated as k (n-m) (8 ). Here, γ m is, for example, an exponentially weighted coefficient expressed as γ m = (γ) m (9) (γ <1),
A is a constant for normalization to be (1 / A) Σ m γ m = 1 (10).

【0006】S/N比推定部27では、各帯域ごとに入
力信号パワー計算部24で計算されたPX,k (n) 及び雑
音パワー計算部26で計算された雑音パワーPav
N,k (n) を用いて目的音声信号対雑音信号の比率である
S/N比が推定される。S/N比推定部27について
は、処理の流れ図を図4Aに記した。詳しい処理の説明
はこの図を用いて行う。
The S / N ratio estimator 27 calculates P X, k (n) calculated by the input signal power calculator 24 and the noise power Pav calculated by the noise power calculator 26 for each band.
The S / N ratio, which is the ratio of the target speech signal to the noise signal, is estimated using N, k (n). Regarding the S / N ratio estimating unit 27, a flowchart of the processing is shown in FIG. 4A. Detailed description of the processing will be made using this figure.

【0007】まず、ステップ31において、 SNRk (n) ′=PX,k (n) /PN,k (n) (11) で定義されるSNRk (n) ′を計算する。次にステップ
32において、決定したSNRk (n) ′に対して、(1
3)式で表わされる一時刻前の推定値(雑音低減された
パワー)PY,k (n−1)を用いて平均化してSNRk
(n) とする。即ち SNRk (n) =(1−β)P[SNRk (n) ′−1] +β[PY,k (n−1)/PN,k (n−1)] (12) とする。P[*]は*が正なら*を、*が負なら0をと
る。このSNRk (n) 、必要に応じてSNRk (n) ′
は、ゲインファクター計算部30に転送される。
[0007] First, in step 31, calculates the SNR k (n) '= P X, k (n) / P N, k (n) SNR k defined in (11) (n)'. Next, in step 32, for the determined SNR k (n) ′, (1
SNR k by averaging using the estimated value (noise reduced power) P Y, k (n-1) one time ago represented by the expression (3)
(n). That is, SNR k (n) = (1−β) P [SNR k (n) ′ − 1] + β [P Y, k (n−1) / P N, k (n−1)] (12) . P [*] takes * if * is positive and 0 if * is negative. This SNR k (n) and, if necessary, SNR k (n) ′
Is transferred to the gain factor calculation unit 30.

【0008】ゲインファクター計算部30では、S/N
比推定部27より転送されてきたSNRk (n) 、場合に
よると、これとSNRk (n) ′を用いて、各雑音低減方
式で定義されている各周波数におけるゲインファクター
G(SNRk (n) )を計算する。このゲインファクター
G(SNRk (n) )は、ゲインファクター挿入部28に
転送される。図5に各手法によるゲインファクターを表
わす。つまり図5中の上3つの手法ではSNRk (n) の
みを用いてゲインファクターを求めるが、最も下のMM
SE法による時は、SNRk (n) の他にSNRk (n) ′
を用いる。
[0008] In the gain factor calculation section 30, S / N
Using the SNR k (n) transferred from the ratio estimator 27 and possibly the SNR k (n) ′, the gain factor G (SNR k ( n)) is calculated. The gain factor G (SNR k (n)) is transferred to the gain factor insertion unit 28. FIG. 5 shows a gain factor according to each method. That is, in the upper three methods in FIG. 5, the gain factor is obtained using only SNR k (n), but the lowermost MM
When using the SE method, SNR k (n) ′ in addition to SNR k (n)
Is used.

【0009】ゲインファクター挿入部28では、各帯域
ごとに、ゲインファクター計算部30において計算され
たゲインファクターを用いて雑音低減を行う。即ち入力
信号パワー計算部24より転送されてきた帯域信号P
X,k (n) に対して、 PY,k (n) =G(SNRk (n) )×PX,k (n) (13) を行い雑音を低減した帯域出力PY,k (n) を出力する。
Y,k (n) は時間領域変換部29に転送され、入力信号
位相計算部25から送られてきたΦk (n) を用いて、 Yk (n) =Yk,r (n) +jYk,i (n) 但し、Yk,r (n) =PY,k (n) cos [Φk (n) ] Yk,i (n) =PY,k (n) sin [Φk (n) ] (14) に変換され、全帯域信号に合成され、更に例えば逆離散
的フーリエ変換により時間領域信号に変換される。この
結果を、D/A変換部34でアナログ信号にして雑音を
低減した信号17、Y(n)を出力する。なお、Spectr
al subtractionでは(13)式の演算で入力信号パワー
X,k (n) から雑音パワーPN,k (n) を引算した結果と
なる。
The gain factor insertion unit 28 performs noise reduction for each band using the gain factor calculated by the gain factor calculation unit 30. That is, the band signal P transferred from the input signal power calculator 24
With respect to X, k (n), P Y, k (n) = G (SNR k (n)) × P X, k (n) (13), and a noise-reduced band output P Y, k ( n) is output.
P Y, k (n) is transferred to the time domain transformation unit 29, and Y k (n) = Y k, r (n) using Φ k (n) sent from the input signal phase calculation unit 25. + JY k, i (n) where Y k, r (n) = P Y, k (n) cos [Φ k (n)] Y k, i (n) = P Y, k (n) sin [Φ k (n)] (14), is synthesized into a full-band signal, and further converted into a time-domain signal by, for example, an inverse discrete Fourier transform. The result is converted into an analog signal by the D / A converter 34, and the signal 17, Y (n), in which noise is reduced, is output. Spectr
The al subtraction is a result of subtracting the noise power P N, k (n) from the input signal power P X, k (n) by the calculation of the equation (13).

【0010】これらの従来の方式では、256〜102
4の周波数帯域に分割して処理が行われる事が多いが、
これを会議システムにそのまま用いると大きな遅延が生
じる。会議を行う際に信号に遅延が生じると、通話性能
が劣化するという問題がある。一方、遅延を小さくする
為に周波数分割の数を減らすと、以下の(15)式で定
義されているS/N比、 S/N=10 log10(PS /PN ) ただし、PS :目的信号の平均パワー PN :雑音信号の平均パワー (15) が15dB程度以上ある場合には、雑音低減によって音
声の品質が向上するが、10dB程度以下の場合には、
雑音は低減されるが、それに伴い音声信号に歪みが生じ
たり、消し残された雑音が時間的に変化するのが原因で
聴感的に悪くなり品質が劣化することが判明した。これ
は、従来の雑音低減方式では、少数の帯域に分割された
信号で処理を行うと、非線形処理に起因する歪が大きく
なる為である。
In these conventional systems, 256 to 102
In many cases, processing is performed by dividing into four frequency bands,
If this is used for the conference system as it is, a large delay occurs. When a signal is delayed during a conference, there is a problem that the call performance deteriorates. On the other hand, reducing the number of frequency division in order to reduce the delay, the following (15) Defined S / N ratio by the formula, S / N = 10 log 10 (P S / P N) , however, P S : Average power of the target signal P N : Average power of the noise signal (15) When the average power is about 15 dB or more, the quality of voice is improved by noise reduction, but when the average power is about 10 dB or less,
It has been found that the noise is reduced, but the sound signal is distorted in accordance with the noise, and the remaining noise is temporally changed. This is because, in the conventional noise reduction method, when processing is performed on a signal divided into a small number of bands, distortion due to nonlinear processing increases.

【0011】即ち、従来法では、遅延を小さいままで、
効果的に雑音低減する事が出来ないという問題がある。
この為、音声会議装置・TV会議装置等、受聴を目的と
し音質が重要であり、またリアルタイム性が要求される
収音においては、この方法をそのまま適用する事はでき
ない。
That is, in the conventional method, while keeping the delay small,
There is a problem that noise cannot be effectively reduced.
For this reason, the sound quality is important for the purpose of listening, such as a voice conference device and a TV conference device, and this method cannot be applied as it is to a sound collection that requires a real-time property.

【0012】[0012]

【発明が解決しようとする課題】周囲雑音が混入した入
力信号の雑音を低減する方法として雑音の振幅分を音声
信号から減算する従来の方法では、周波数帯域分割数が
多いと遅延が大きくなり、周波数帯域分割数を減らすと
音声信号に歪が生じたり、引き残された雑音が時間的に
変化する為に聴感上好ましくない音をたてるという問題
があった。この発明の目的は、処理遅延が少なく、聴感
上の音質の劣化が少ない雑音低減処理方法を提供する事
である。
As a method for reducing the noise of an input signal mixed with ambient noise, the conventional method of subtracting the amplitude of the noise from a speech signal increases the delay when the number of frequency band divisions is large. If the number of frequency band divisions is reduced, there is a problem in that distortion occurs in the audio signal, and undesired sound is produced because the remaining noise changes with time. SUMMARY OF THE INVENTION It is an object of the present invention to provide a noise reduction processing method which has a small processing delay and a small deterioration in sound quality in audibility.

【0013】[0013]

【課題を解決するための手段】この発明は、目的となる
音声信号と周囲雑音などの混在したマイクロホンでの受
音信号を複数の帯域に分割し、各々の帯域別の信号に対
し雑音パワーを推定し、推定された雑音パワーと実際に
入力されてきた入力信号パワーとを比較して音声信号と
雑音信号の比率を推定すると共に音声信号の振幅を雑音
低減された信号として推定し、推定された音声信号と雑
音信号の比率に基づいて計算された帯域ごとの雑音抑圧
の為のゲインファクターを、対応する帯域の入力信号に
掛け合わせる事によって得る。雑音低減された信号の推
定はその際生じる歪を少量の各帯域別の信号を、音声信
号と雑音信号の比率に応じて雑音が多い程、多く加算す
る事によってマスキングし、その結果歪が少ない効果的
な雑音低減を可能にする。
SUMMARY OF THE INVENTION The present invention divides a signal received by a microphone in which a target voice signal and ambient noise are mixed into a plurality of bands, and reduces noise power for each band signal. Estimating, comparing the estimated noise power with the input signal power that has actually been input, estimating the ratio of the audio signal to the noise signal, and estimating the amplitude of the audio signal as a noise-reduced signal. The gain factor for noise suppression for each band calculated based on the ratio of the speech signal to the noise signal is multiplied by the input signal of the corresponding band. The estimation of the noise-reduced signal is performed by masking the resulting distortion by adding a small amount of the signal for each band according to the ratio of the voice signal to the noise signal as the noise increases, thereby increasing the noise. As a result, the distortion is reduced. Enables effective noise reduction.

【0014】この発明は、次の様な特徴を持つ。まず、
雑音低減によって生じた歪を、少量の各帯域別の信号を
加算しマスキングする事によって音質の劣化を防ぐ。ま
た、各帯域別の信号の加算率を音声信号と雑音信号の比
率によって変化させる事によって、雑音が少ない、つま
り、雑音低減による処理歪が小さい場合には、各帯域別
信号の加算量を小さくし、雑音低減効果を最大限にする
事が可能である。
The present invention has the following features. First,
The distortion caused by noise reduction is masked by adding a small amount of signal for each band to prevent deterioration of sound quality. Also, by changing the addition rate of the signal for each band according to the ratio of the audio signal to the noise signal, when the noise is small, that is, when the processing distortion due to the noise reduction is small, the addition amount of the signal for each band is reduced. However, it is possible to maximize the noise reduction effect.

【0015】[0015]

【発明の実施の形態】図1にこの発明の実施例を適用し
た雑音低減装置の機能構成を示し、図3と対応するもの
については共通の記号を用いた。雑音パワー推定には、
図3に示したものでもよいがここでは、特願平8−68
548や佐々木、羽田“損失制御を用いた帯域分割型雑
音低減方式について”春季日本音響学会予稿515〜5
16(1996)で提案された方式を用いて推定を行う
事にして、以下の説明を行う。この方法によると、音声
と雑音とが混在している状態とみなす区間でも時間的に
音声がない区間があれば、雑音パワーを検出することが
できる。
FIG. 1 shows a functional configuration of a noise reduction apparatus to which an embodiment of the present invention is applied, and common symbols are used for those corresponding to FIG. For noise power estimation,
Although it may be the one shown in FIG. 3, here, Japanese Patent Application No. Hei 8-68
548, Sasaki, Haneda, "On band-division noise reduction using loss control," Spring Meeting of the Acoustical Society of Japan, 515-5.
16 (1996), the estimation is performed using the method described below. According to this method, the noise power can be detected if there is a temporally non-voice section even in a section in which voice and noise are regarded as being mixed.

【0016】まず、マイクロホン11により受音され
た、目的信号と不要な雑音等の混入する入力信号15を
A/D変換部21においてデジタル化し、周波数帯域分
割部22に転送する。周波数帯域分割部22では、転送
された信号が周波数帯域に分割される。分割された各帯
域信号は、入力信号パワー計算部24、ゲインファクタ
ー挿入部28に転送される。以降、入力信号のk番目の
帯域信号をXk (n) として、Xk (n) に対する処理の流
れを説明する。
First, an input signal 15 mixed with a target signal and unnecessary noise received by the microphone 11 is digitized in an A / D converter 21 and transferred to a frequency band divider 22. In the frequency band dividing section 22, the transferred signal is divided into frequency bands. Each of the divided band signals is transferred to the input signal power calculator 24 and the gain factor inserter 28. Hereinafter, the k-th band signal of the input signal as X k (n), for explaining the flow of processing for X k (n).

【0017】入力信号パワー計算部24では、転送され
てきたXk (n) のパワーレベルを前記(7)式で計算
し、S/N比推定部27、雑音パワー推定部51に転送
される。雑音パワー推定部51では転送されてきたP
X,k (n) を用いて雑音パワーPav N,k (n) の推定が行わ
れる。雑音パワー推定部51の処理の流れ図を図2Aに
示す。
The input signal power calculation unit 24 transfers the
XkCalculate the power level of (n) by the above equation (7)
Then, the data is transferred to the S / N ratio estimation unit 27 and the noise power estimation unit 51.
Is done. In the noise power estimating unit 51, the transferred P
X, knoise power Pav using (n) N, k(n) is estimated
It is. FIG. 2A is a flowchart of the processing of the noise power estimation unit 51.
Show.

【0018】まずステップ61において、転送されてき
たPX,k (n) の時間平均パワーレベルPX,k (n) を前記
(8)式、(9)式、(10)式を用いて計算する。こ
の場合の平均時間としては、例えば5〜6msecをと
る。次に、ステップ62である一定時間におけるPav
X,k (n) のレベル分布のヒストグラムをとる。つまりP
avX,k (n) が属するパワー区間の数を1加算、即ち、 hk (int PavX,k (n))=hk (int PavX,k (n) )+1 (16) を行う。int(*)は小数点以下を切り捨て整数化す
る事を示す。更にステップ63でヒストグラムhk (i)
のピーク区間が検出され記憶される。即ち前後の値に対
して hk (i′)≦hk (i) (17) となるiを求める。このiの中で最も小さなiを雑音の
パワーPavN,x (n) とする。つまりピーク値が複数得ら
れた時、最小の値のピーク値を雑音パワーPavN, k (n)
とし、これはS/N比推定部27に転送される。
First, in step 61, the time average power level P X, k (n) of the transferred P X, k (n) is calculated by using the above equations (8), (9) and (10). calculate. The average time in this case is, for example, 5 to 6 msec. Next, at step 62, Pav for a certain period of time.
Take a histogram of the level distribution of X, k (n). That is, P
Add 1 to the number of power sections to which av X, k (n) belongs, that is, h k (int Pav X, k (n)) = h k (int Pav X, k (n)) + 1 (16) . int (*) indicates that the decimal part is truncated to an integer. Further, in step 63, the histogram h k (i)
Are detected and stored. That is, i that satisfies h k (i ′) ≦ h k (i) (17) with respect to values before and after is obtained. The smallest i of the i is the noise power Pav N, x (n). That is, when a plurality of peak values are obtained, the peak value of the minimum value is determined by the noise power Pav N, k (n)
This is transferred to the S / N ratio estimation unit 27.

【0019】S/N比推定部27では、入力信号パワー
計算部24で計算されたPX,k (n)及び雑音パワー推定
部51で推定された雑音パワーPavN,k (n) を用いて、
図4Aで示した方法でSNRk (n)′及びSNRk (n)
が推定される。S/N比推定部27で推定されたSNR
k (n)はゲインファクター計算部30及び入力信号加算
率決定部52に転送される。ゲインファクター計算部3
0で用いる計算法によってはSNRk (n)′も入力され
る。
The S / N ratio estimator 27 uses P X, k (n) calculated by the input signal power calculator 24 and the noise power Pav N, k (n) estimated by the noise power estimator 51. hand,
In the method shown in FIG. 4A, SNR k (n) ′ and SNR k (n)
Is estimated. SNR estimated by S / N ratio estimating section 27
k (n) is transferred to the gain factor calculation unit 30 and the input signal addition rate determination unit 52. Gain factor calculator 3
Depending on the calculation method used at 0, SNR k (n) ′ is also input.

【0020】ゲンイファクター計算部30では、S/N
比推定部201より転送されてきたSNRk (n)、必要
に応じてSNRk (n)′を用いて、ゲインファクターG
(SNRk (n))が決定される。ここで、ゲインファク
ターの具体的な計算は、従来技術の項で図5に示した方
法などが用いられる。ゲインファクター計算部30で推
定されたゲインファクターG(SNRk (n))は、ゲイ
ンファクター挿入部28に転送される。なお、S/N比
推定部27の方法およびゲインファクター計算部30の
方法は、この明細書に記載したもの以外の方法でもよ
い。
In the gain factor calculating section 30, the S / N
Using the SNR k (n) transferred from the ratio estimating unit 201 and, if necessary, the SNR k (n) ′, the gain factor G
(SNR k (n)) is determined. Here, the specific calculation of the gain factor uses the method shown in FIG. The gain factor G (SNR k (n)) estimated by the gain factor calculation unit 30 is transferred to the gain factor insertion unit 28. Note that the method of the S / N ratio estimating unit 27 and the method of the gain factor calculating unit 30 may be methods other than those described in this specification.

【0021】ゲインファクター挿入部28では、ゲイン
ファクター計算部30において計算されたゲインファク
ターG(SNRk (n))を用いて雑音低減を行う。即ち
周波数帯域分割部22より転送されてきた帯域信号Xk
(n) に対して(18)式の演算を行う。 Y′k (n) =G(SNRk (n))×Xk (n) (18) この雑音を低減した信号Y′k (n) を入力信号加算部5
3に転送する。
The gain factor insertion unit 28 performs noise reduction using the gain factor G (SNR k (n)) calculated by the gain factor calculation unit 30. That is, the band signal X k transferred from the frequency band dividing unit 22
The operation of equation (18) is performed on (n). Y ′ k (n) = G (SNR k (n)) × X k (n) (18) The noise-reduced signal Y ′ k (n) is input to the input signal adder 5.
Transfer to 3.

【0022】入力信号加算率決定部52では、転送され
てきたSNRk (n)を用いて、S/N比に基づいた入力
信号の加算率α′を決定する。α′は、SNRk (n)が
大きい時は小さい値をとる事が望ましい。例えば、 15≦10 log10[SNRk (n)] ⇒ α′=0 5<10 log10[SNRk (n)]<15 ⇒ α′=0.1 10 log10[SNRk (n)]<5 ⇒ α′=0.3(19) の様に決定する。更に、この例では周波数帯域分割部2
2において帯域分割数Mに応じて決定される帯域分割数
ファクターβをα′に掛け合わせて入力信号加算率αと
した場合である。例えば α=β×α′ 1024≦M ⇒ β=0.01 M<1024 ⇒ β=1 (20) とする。つまり帯域分割数Mが大きい場合は加算率αを
小さくする。入力信号加算率αは、入力信号加算部53
に転送される。
The input signal addition rate determination section 52 uses the transferred SNR k (n) to determine the input signal addition rate α ′ based on the S / N ratio. α ′ desirably takes a small value when SNR k (n) is large. For example, 15 ≦ 10 log 10 [SNR k (n)] ⇒ α ′ = 0 5 <10 log 10 [SNR k (n)] <15 ⇒ α ′ = 0.10 log 10 [SNR k (n)] <5 ⇒ α '= 0.3 (19) Further, in this example, the frequency band dividing unit 2
In FIG. 2, the input signal addition rate α is obtained by multiplying α ′ by a band division number factor β determined according to the band division number M. For example, α = β × α ′ 1024 ≦ M⇒β = 0.01 M <1024 → β = 1 (20) That is, when the number M of band divisions is large, the addition rate α is reduced. The input signal addition rate α is determined by the input signal addition unit 53
Is forwarded to

【0023】入力信号加算部53では、周波数帯域分割
部22、ゲインファクター挿入部28、入力信号加算率
決定部52から転送されてきた帯域信号Xk (n) 、雑音
を低減した信号Y′k (n) 及び入力信号加算率αを用い
て帯域出力信号Yk (n) を出力する。即ち、 Yk (n) =α×Xk (n) +(1−α)×Y′k (n) (21) とする。Yk (n) は時間領域変換部29に転送され、全
体域が合成された後時間領域に変換される。この結果
を、D/A変換部34でアナログ信号にして雑音を低減
した信号17、Y(n) を出力する。
In the input signal adding section 53, the band signal X k (n) transferred from the frequency band dividing section 22, the gain factor inserting section 28, and the input signal addition rate determining section 52, the signal Y ′ k with reduced noise (n) and outputs the band output signal Y k (n) using the input signal addition rate α. That is, Y k (n) = α × X k (n) + (1−α) × Y ′ k (n) (21) Y k (n) is transferred to the time domain conversion unit 29, where the entire area is synthesized and then converted to the time domain. The result is converted into an analog signal by the D / A converter 34 to output a signal 17, Y (n), in which noise is reduced.

【0024】この本発明の有効性を、実際のTV会議シ
ステムの状況を想定したオピニオン評価法を用いて評価
した。評価実験は、体積87m3 、残響時間300ms
ec、暗騒音レベルは46dB(A)の可変残響室で行
った。帯域分割数Mは64とした(これはサブバンド方
式エコーキャンセラで好ましいとされている分割数であ
る)。評価は30名の被験者数で行った。作成した刺激
音を、スピーカから流し、それを拡声会議における受話
音声として、被験者に品質評価を行ってもらった。8d
B、6dB、1dBの3種類のS/N比の雑音付加音声
を図5に示した4種類の雑音低減方式で処理し、雑音低
減処理の前後で評価がどの様に変るかを調べた。
The effectiveness of the present invention was evaluated using an opinion evaluation method assuming the situation of an actual TV conference system. The evaluation experiment was performed with a volume of 87 m 3 and a reverberation time of 300 ms.
ec, the background noise level was set in a variable reverberation room of 46 dB (A). The number M of band divisions was set to 64 (this is a preferable number of divisions in the subband echo canceller). The evaluation was performed on 30 subjects. The created stimulus sound was played from a speaker, and the quality was evaluated by the subject as the received voice in a public conference. 8d
B, 6 dB, and 1 dB, three types of noise-added voices with S / N ratios were processed by the four types of noise reduction schemes shown in FIG.

【0025】図2B(a)は、各雑音低減方式におけ
る、MOS値の変化を示したものである。雑音低減処理
前と処理後を比較すると、S/N比の良い時は、ほぼ、
処理によってMOS値が雑音付加音声(原声)のそれよ
り上がるのに対し、S/N比が悪い時は、下がってしま
う。これは、帯域分割数が少ない場合、雑音低減による
品質改善よりも、処理による音声の歪や、消し残り雑音
による劣化が激しく、MOS値を下げてしまう為だと考
えられる。この事から、S/N比が悪い場合には、従来
法では雑音低減による音質向上がみられないばかりか、
却って低下している事がわかる。
FIG. 2B (a) shows the change in the MOS value in each noise reduction method. Comparing before and after the noise reduction processing, when the S / N ratio is good, almost
While the MOS value is higher than that of the noise-added voice (original voice) by the processing, when the S / N ratio is poor, the MOS value is lowered. This is considered to be because when the number of band divisions is small, voice distortion due to processing and degradation due to residual noise are more severe than quality improvement by noise reduction, and the MOS value is lowered. From this fact, when the S / N ratio is poor, not only does the conventional method not improve the sound quality due to noise reduction,
It turns out that it is decreasing.

【0026】図2B(b)は、この発明により雑音低減
音に対し、原音を足し合わせた結果である。ここでは、
S/N比によらずα=0.3とした。S/N比が悪い場
合に原音声を付加すると、付加前に比べMOS値がほぼ
0.5〜1上がり、その結果、雑音低減処理による音質
向上が見られる。また、S/N比に応じて加算率αを変
化させ、S/N比がいい場合には、付加率αを小さくす
ることで、更に効果的な音質向上が期待出来る事がわか
る。
FIG. 2B (b) shows the result of adding the original sound to the noise reduction sound according to the present invention. here,
Α was set to 0.3 regardless of the S / N ratio. When the original sound is added when the S / N ratio is poor, the MOS value is increased by approximately 0.5 to 1 as compared with before the addition, and as a result, the sound quality is improved by the noise reduction processing. Further, it can be understood that the addition rate α is changed according to the S / N ratio, and when the S / N ratio is good, the sound quality can be more effectively improved by reducing the addition rate α.

【0027】ゲインファクター挿入を図3に示したよう
に、つまり(13)式により行い、これより(14)式
によりYk (n) ′を得てもよい。先の実施例から理解さ
れるように、各帯域ごとの入力信号パワーと、対応帯域
ごとの雑音パワーとから、その帯域の目的音声信号に対
する雑音の比率を推定すると共に、その帯域の前記目的
音声信号の振幅を雑音低減された信号として推定すれば
よい。
The gain factor may be inserted as shown in FIG. 3, that is, by the equation (13), and Y k (n) ′ may be obtained from the equation (14). As can be understood from the above embodiment, from the input signal power for each band and the noise power for each corresponding band, the ratio of noise to the target audio signal in that band is estimated, and the target voice in that band is estimated. What is necessary is just to estimate the signal amplitude as a noise-reduced signal.

【0028】[0028]

【発明の効果】以上述べたようにこの発明は、目的音声
と雑音が混在した信号を周波数帯域に分割し、帯域毎の
目的信号に対する雑音信号の比率を推定し、この推定結
果に基づいて帯域毎に雑音を低減するが、周波数帯域分
割数が少ない場合に生じる目的音声の歪を低減する為
に、雑音と音声の比率に応じて雑音低減音声に原信号を
その雑音が多い程多く加算することによって、歪の少な
い雑音低減音声が得られることができる。その際に帯域
分割数が少ない程、加算する比率を大とする。
As described above, the present invention divides a signal in which the target voice and noise are mixed into frequency bands, estimates the ratio of the noise signal to the target signal for each band, and based on the estimation result, The noise is reduced every time, but in order to reduce the distortion of the target sound that occurs when the number of frequency band divisions is small, the original signal is added more to the noise-reduced sound according to the ratio of the noise to the sound as the noise increases. As a result, noise-reduced speech with little distortion can be obtained. At this time, the smaller the number of band divisions, the larger the ratio to be added.

【0029】この発明方法による受聴を目的とした雑音
低減により、遅延が少なく、聞き易い目的信号を得る事
が可能になる。その結果、音声会議・TV会議等の拡声
通話系において、マイクロホンで受音し、相手側に送出
される送話信号に、目的となる音声以外の周囲雑音が混
入した場合でも、この発明方法による雑音の低減により
音声の明瞭性を保つ事が可能になり、通信品質が向上す
る。
The noise reduction for the purpose of listening according to the method of the present invention makes it possible to obtain an easy-to-listen target signal with a small delay. As a result, even in a voice communication system such as a voice conference or a TV conference, even if ambient noise other than the target voice is mixed in a transmission signal received by the microphone and transmitted to the other party, the method of the present invention can be used. The reduction of the noise makes it possible to maintain the clarity of the voice, and the communication quality is improved.

【図面の簡単な説明】[Brief description of the drawings]

【図1】この発明の実施例を適用した雑音低減装置の機
能構成を示すブロック図。
FIG. 1 is a block diagram showing a functional configuration of a noise reduction device to which an embodiment of the present invention is applied.

【図2】Aは図1中の雑音パワー推定部51における処
理流れ図、B(a)は従来法による雑音低減効果を主観
評価した結果を示す図、(b)はこの発明による雑音低
減効果を主観評価した結果を示す図である。
2A is a flowchart illustrating the processing performed by the noise power estimating unit 51 in FIG. 1, FIG. 2B is a diagram illustrating the result of a subjective evaluation of the noise reduction effect according to the conventional method, and FIG. 2B is a diagram illustrating the noise reduction effect according to the present invention; It is a figure showing the result of subjective evaluation.

【図3】従来の雑音低減方法を適用した装置の機能構成
を示すブロック図。
FIG. 3 is a block diagram showing a functional configuration of a device to which a conventional noise reduction method is applied.

【図4】Aはゲインファクター計算部における処理の流
れ図、Bは雑音低減装置の原理説明図である。
FIG. 4A is a flowchart of a process in a gain factor calculation unit, and FIG. 4B is a diagram illustrating the principle of a noise reduction device.

【図5】代表的なゲインファクターの計算式を示す図で
ある。
FIG. 5 is a diagram showing a calculation formula of a typical gain factor.

Claims (3)

【特許請求の範囲】[Claims] 【請求項1】 入力信号から雑音信号を除去した音声信
号を出力する雑音低減処理方法において、 前記入力信号を複数の周波数帯域に分割する帯域分割過
程と、 前記各帯域に分割された入力信号のパワーを計算する入
力信号パワー計算過程と、 前記各帯域に分割された入力信号中の雑音パワーの求め
る雑音パワー計算過程と、 前記各帯域毎の入力信号パワーと、前記各帯域毎の雑音
パワーとから前記各帯域毎の目的音声信号に対する雑音
の比率を推定すると共に、前記各帯域毎の前記目的音声
信号の振幅を雑音低域された信号として推定する推定過
程と、 前記推定された前記目的音声信号対雑音比率に基づい
て、前記帯域毎の前記雑音低減された信号と前記各帯域
に分割された入力信号との加算割合を決定する過程と、 前記決定された前記加算割合に基づいて、前記帯域毎の
前記雑音低減された信号と前記各帯域に分割された入力
信号を加算し、帯域出力信号を出力する過程と、 前記帯域出力信号を時間領域に変換し全帯域信号に合成
する過程と、 を有することを特徴とする帯域分割型雑音低減方法。
1. A noise reduction method for outputting an audio signal obtained by removing a noise signal from an input signal, comprising: a band dividing step of dividing the input signal into a plurality of frequency bands; An input signal power calculating step of calculating power, a noise power calculating step of calculating noise power in the input signal divided into the respective bands, an input signal power of each of the bands, and a noise power of each of the bands. An estimation step of estimating the ratio of noise to the target audio signal for each band from the above, and estimating the amplitude of the target audio signal for each band as a signal with low noise, and the estimated target voice A step of determining an addition ratio of the noise-reduced signal for each band and the input signal divided into the respective bands, based on a signal-to-noise ratio; and Adding the noise-reduced signal for each band and the input signal divided into each band based on the addition ratio, and outputting a band output signal; Combining a band signal with a band signal.
【請求項2】 請求項1記載の雑音低減方法において、 前記推定過程は前記各帯域毎の入力信号パワーと前記各
帯域毎の雑音パワーとを用いて、前記各帯域毎の前記目
的音声信号対雑音の比率を推定する過程と、 前記目的音声信号対雑音比率に基づいて前記各帯域毎の
ゲインファクターを決定するゲインファクター計算過程
と、 前記各帯域に分割された入力信号に対して前記各帯域毎
に決定された前記ゲインファクターを挿入して、前記各
帯域毎の前記雑音低減された信号を得るゲインファクタ
ー挿入過程とよりなることを特徴とする帯域分割型雑音
低減方法。
2. The noise reduction method according to claim 1, wherein the estimation step uses the input signal power for each of the bands and the noise power for each of the bands to generate the target audio signal pair for each of the bands. Estimating a noise ratio; calculating a gain factor for each of the bands based on the target audio signal-to-noise ratio; and calculating each of the bands with respect to the input signal divided into the respective bands. A gain factor inserting step of inserting the gain factor determined for each band to obtain the noise-reduced signal for each band.
【請求項3】 請求項1又は2記載の雑音低減方法にお
いて、 前記帯域分割過程で分割する周波数帯域の数に応じて、
前記帯域毎の前記雑音低減された信号と前記各帯域に分
割された入力信号の加算割合を決定することを特徴とす
る帯域分割型雑音低減方法。
3. The noise reduction method according to claim 1, wherein the number of frequency bands to be divided in the band division step is:
A band division type noise reduction method, wherein an addition ratio of the noise-reduced signal and the input signal divided into each band is determined for each band.
JP31778696A 1996-11-28 1996-11-28 Band division type noise reduction method Expired - Lifetime JP3454402B2 (en)

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