JPH07101853B2 - Noise reduction method - Google Patents
Noise reduction methodInfo
- Publication number
- JPH07101853B2 JPH07101853B2 JP3031815A JP3181591A JPH07101853B2 JP H07101853 B2 JPH07101853 B2 JP H07101853B2 JP 3031815 A JP3031815 A JP 3031815A JP 3181591 A JP3181591 A JP 3181591A JP H07101853 B2 JPH07101853 B2 JP H07101853B2
- Authority
- JP
- Japan
- Prior art keywords
- component
- frequency
- noise reduction
- power spectrum
- reduction method
- 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.)
- Expired - Fee Related
Links
Landscapes
- Noise Elimination (AREA)
Description
【０００１】[0001]
【産業上の利用分野】本発明は高騒音環境下における通
信システムに用いて好適な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 suitable for use in a communication system in a high noise environment.
【０００２】[0002]
【従来の技術】一般に、航空機内や船舶のエンジン室内
等の高騒音環境下で使用する通信システムでは、通常マ
イクロホンにより音声を検出しても、高騒音に基づく雑
音もそのまま検出されてしまうため、音声が雑音に妨害
され、通話が著しく困難になる場合がある。2. Description of the Related Art Generally, in a communication system used in a noisy environment such as an aircraft or an engine room of a ship, even if a voice is usually detected by a microphone, noise due to the high noise is also detected as it is. The voice may be disturbed by noise, making the call extremely difficult.
【０００３】このため、従来は雑音低減効果の高い骨伝
導マイクロホン（一般的には加速度形ピックアップ）を
用いて音声の検出を行っていた。骨伝導マイクロホンは
人体頭部の骨部、例えば、額等に取付けることにより、
発声した際に生ずる僅かな骨部の振動を電気的信号に変
換し、音声を疑似的に検出するものであり、周囲におけ
る高騒音環境の影響は排除される。Therefore, conventionally, a bone conduction microphone (generally an acceleration type pickup) having a high noise reduction effect is used to detect a voice. By attaching the bone conduction microphone to the bone part of the human head, for example, the forehead,
A slight vibration of the bone portion generated when uttering is converted into an electric signal to detect the voice in a pseudo manner, and the influence of a high noise environment in the surroundings is eliminated.
【０００４】[0004]
【発明が解決しようとする課題】しかし、骨伝導マイク
ロホンにより音声を検出する従来の方法は、本来の音声
を骨振動を介して疑似的に検出するため、音質面で難点
があり、現実には通話音声の明瞭性及び自然性において
普通マイクロホンより大きく劣るなど、実用的な通話手
段にまで確立されていないのが実情である。However, the conventional method of detecting the sound by the bone conduction microphone has a problem in sound quality because it detects the original sound artificially through the bone vibration. In reality, it is not established as a practical means of communication, such as the intelligibility and naturalness of the communication voice being significantly inferior to that of a microphone.
【０００５】本発明はこのような従来の技術に存在する
課題を解決したものであり、高騒音環境下の通信システ
ムにおける通話音声のＳ／Ｎ比を高めると同時に音質を
飛躍的に向上させ得る雑音低減方法の提供を目的とす
る。The present invention solves the problems existing in the prior art as described above, and can improve the S / N ratio of a communication voice in a communication system under a high noise environment and at the same time dramatically improve the sound quality. The purpose is to provide a noise reduction method.
【０００６】[0006]
【課題を解決するための手段】本発明に係る雑音低減方
法は、骨伝導マイクロホン（以下、骨伝導マイクと記
す）２等の加速度形ピックアップと普通マイクロホン
（以下、普通マイクと記す）３により音声を同時に検出
し、普通マイク３から得る音声信号Ｓｓを有声音成分
（母音成分）Ｓａと無声音成分（子音成分）Ｓｂに判別
するとともに、有声音成分Ｓａを周波数パワースペクト
ルＰｓに変換し、他方、骨伝導マイク２から得る音声信
号Ｓｆに基づいて有声音成分のピッチ周波数Ｐｆを得、
前記周波数パワースペクトルＰｓから前記ピッチ周波数
Ｐｆに対応する周波数成分を抽出するとともに、抽出し
た周波数パワースペクトルＰｏを音声信号における有声
音処理成分Ｓｃに変換し、この有声音処理成分Ｓｃに対
して無声音成分Ｓｂ、望ましくは無声音成分Ｓｂを所定
の大きさに減衰して得た無声音処理成分Ｓｄを加えて音
声処理信号Ｓｏを得ることを特徴とする。この場合、有
声音成分Ｓａから周波数パワースペクトルＰｓへの変換
はフーリェ変換により、周波数パワースペクトルＰｏか
ら有声音処理成分Ｓｃへの変換は逆フーリェ変換により
行うことが望ましい。また、周波数パワースペクトルＰ
ｓから抽出される周波数成分は、基本周波数乃至所定の
ｎ次高調波成分となるように予め設定する。なお、無声
音処理成分Ｓｄは所定時間遅延させることができ、この
遅延時間は可変可能である。また、音声信号Ｓｓに対す
る有声音成分Ｓａと無声音成分Ｓｂの分離は、骨伝導マ
イク２から得る音声信号Ｓｆに基づいて有声音成分と無
声音成分を判別し、この判別結果により行う。さらにま
た、周波数パワースペクトルＰｓからの周波数成分の抽
出は櫛形フィルタ４を利用できる。A noise reduction method according to the present invention is a method in which an accelerometer such as a bone conduction microphone (hereinafter referred to as a bone conduction microphone) 2 and a normal microphone (hereinafter referred to as a normal microphone) 3 are used for voice Simultaneously, the voice signal Ss obtained from the normal microphone 3 is discriminated into a voiced sound component (vowel component) Sa and an unvoiced sound component (consonant component) Sb, and the voiced sound component Sa is converted into a frequency power spectrum Ps. The pitch frequency Pf of the voiced sound component is obtained based on the voice signal Sf obtained from the bone conduction microphone 2,
A frequency component corresponding to the pitch frequency Pf is extracted from the frequency power spectrum Ps, the extracted frequency power spectrum Po is converted into a voiced sound processing component Sc in a voice signal, and an unvoiced sound component with respect to the voiced sound processing component Sc. Sb, preferably the unvoiced sound processing component Sb obtained by attenuating the unvoiced sound component Sb to a predetermined magnitude is added to obtain the speech processed signal So. In this case, it is preferable that the voiced sound component Sa is converted to the frequency power spectrum Ps by the Fourier transform, and the frequency power spectrum Po is converted to the voiced sound processing component Sc by the inverse Fourier transform. Also, the frequency power spectrum P
The frequency component extracted from s is preset so as to be a fundamental frequency or a predetermined nth harmonic component. The unvoiced sound processing component Sd can be delayed for a predetermined time, and this delay time can be varied. Further, the voiced sound component Sa and the unvoiced sound component Sb are separated from the voice signal Ss by discriminating the voiced sound component and the unvoiced sound component based on the voice signal Sf obtained from the bone conduction microphone 2, and based on the result of this discrimination. Furthermore, the comb filter 4 can be used to extract the frequency component from the frequency power spectrum Ps.
【０００７】[0007]
【作用】本発明に係る雑音低減方法によれば、高騒音環
境下における音声は骨伝導マイク２と普通マイク３によ
って同時に検出される。そして、骨伝導マイク２から得
られる音声信号Ｓｆに基づいて有声音成分と無声音成分
が判別され、この判別結果により、普通マイク３から得
られる音声信号Ｓｓが有声音成分Ｓａと無声音成分Ｓｂ
に判別される。According to the noise reduction method of the present invention, the voice in the high noise environment is detected by the bone conduction microphone 2 and the normal microphone 3 at the same time. Then, the voiced sound component and the unvoiced sound component are discriminated based on the voice signal Sf obtained from the bone conduction microphone 2. Based on the discrimination result, the voice signal Ss obtained from the normal microphone 3 is the voiced sound component Sa and the unvoiced sound component Sb.
Is determined.
【０００８】これにより、無声音成分Ｓｂは、雑音成分
を低減すべく所定の大きさに減衰されるとともに、さら
に、所定時間遅延されて無声音処理成分Ｓｄとなる。As a result, the unvoiced sound component Sb is attenuated to a predetermined size to reduce the noise component, and further delayed for a predetermined time to become the unvoiced sound processing component Sd.
【０００９】他方、音声信号Ｓｓの有声音成分Ｓａは、
例えば、フーリェ変換法により周波数パワースペクトル
Ｐｓに変換され、この周波数パワースペクトルＰｓは櫛
形フィルタ４の入力側に供給される。一方、骨伝導マイ
ク２側の信号処理系では、音声信号Ｓｆから有声音成分
のピッチ周波数Ｐｆを得、このピッチ周波数Ｐｆは櫛形
フィルタ４の制御端子に付与される。この結果、櫛形フ
ィルタ４の出力には周波数パワースペクトルＰｓからピ
ッチ周波数Ｐｆに対応する周波数成分が抽出され、この
周波数成分に基づく周波数パワースペクトルＰｏは、例
えば、逆フーリェ変換され、これにより、雑音成分の除
去された有声音処理成分Ｓｃを得る。なお、周波数パワ
ースペクトルＰｓから抽出されるピッチ周波数Ｐｆに対
応する周波数成分は、基本周波数乃至予め設定する所定
のｎ次高調波成分（例えば、第三次高調波成分程度）ま
でで十分である。On the other hand, the voiced sound component Sa of the voice signal Ss is
For example, it is converted into a frequency power spectrum Ps by the Fourier transform method, and this frequency power spectrum Ps is supplied to the input side of the comb filter 4. On the other hand, in the signal processing system on the bone conduction microphone 2 side, the pitch frequency Pf of the voiced sound component is obtained from the audio signal Sf, and this pitch frequency Pf is given to the control terminal of the comb filter 4. As a result, a frequency component corresponding to the pitch frequency Pf is extracted from the frequency power spectrum Ps at the output of the comb filter 4, and the frequency power spectrum Po based on this frequency component is subjected to, for example, an inverse Fourier transform, whereby a noise component is generated. A voiced sound processing component Sc from which is removed is obtained. It should be noted that the frequency component corresponding to the pitch frequency Pf extracted from the frequency power spectrum Ps is sufficient from the fundamental frequency to a predetermined n-th harmonic component (for example, about the third harmonic component) set in advance.
【００１０】よって、無声音処理成分Ｓｄと成分有声音
処理成分Ｓｃを加えれば、目的の音声処理信号Ｓｏが得
られる。この音声処理信号Ｓｄは普通マイク３から得ら
れる音声信号Ｓｓに対して雑音低減処理した信号であ
る。Therefore, if the unvoiced sound processing component Sd and the component voiced sound processing component Sc are added, the target voice processing signal So can be obtained. The audio processing signal Sd is a signal obtained by subjecting the audio signal Ss obtained from the normal microphone 3 to noise reduction processing.
【００１１】[0011]
【実施例】次に、本発明に係る好適実施例を挙げ、図面
に基づき詳細に説明する。DESCRIPTION OF THE PREFERRED EMBODIMENTS Next, preferred embodiments according to the present invention will be described in detail with reference to the drawings.
【００１２】まず、本発明に係る雑音低減方法を実施で
きる信号処理装置１について、図１を参照して説明す
る。First, a signal processing apparatus 1 capable of implementing the noise reduction method according to the present invention will be described with reference to FIG.
【００１３】図１において、２は人体の額等に付設する
骨伝導マイク（加速度形ピックアップ）であり、この骨
伝導マイク２はローパスフィルタ１１の入力側に接続す
る。ローパスフィルタ１１の出力側はアナログ−デジタ
ル変換器１２の入力側に接続するとともに、同変換器１
２の出力側は発声状態認識部１３に接続する。さらに、
同認識部１３の出力側は有声音成分と無声音成分を判別
する音声成分判別部１４の入力側に接続するとともに、
同判別部１４はピッチ周波数を検出するピッチ周波数検
出部１５に接続する。そして、同検出部１５は櫛形フィ
ルタ４の制御端子に接続する。In FIG. 1, reference numeral 2 denotes a bone conduction microphone (acceleration type pickup) attached to the forehead of the human body, and this bone conduction microphone 2 is connected to the input side of a low pass filter 11. The output side of the low-pass filter 11 is connected to the input side of the analog-digital converter 12, and the converter 1
The output side of 2 is connected to the utterance state recognition unit 13. further,
The output side of the recognition unit 13 is connected to the input side of a voice component determination unit 14 that determines a voiced sound component and an unvoiced sound component, and
The discrimination unit 14 is connected to a pitch frequency detection unit 15 that detects the pitch frequency. Then, the detection unit 15 is connected to the control terminal of the comb filter 4.
【００１４】一方、３はダイナミックマイクロホン、コ
ンデンサマイクロホン等で代表される発声音を直接検出
する普通マイクであり、この普通マイク３はローパスフ
ィルタ１６の入力側に接続する。同フィルタ１６の出力
側はアナログ−デジタル変換器１７の入力側に接続する
とともに、同変換器１７の出力側は第一切換スイッチ１
８の一方の接点部に接続する。第一切換スイッチ１８は
前記発声状態認識部１３の認識結果に基づいて切換えら
れる。また、同切換スイッチ１８の他方の接点部は第二
切換スイッチ１９の可動接点部に接続する。第二切換ス
イッチ１９は前記音声成分判別部１４の判別結果により
制御回路２０を介して切換えられる。さらにまた、第二
切換スイッチ１９における一方の固定接点部は高速フー
リェ変換器２１の入力側に接続し、同変換器２１の出力
側は櫛形フィルタ４の入力側に接続する。そして、櫛形
フィルタ４の出力側は逆フーリェ変換器２２の入力側に
接続する。On the other hand, 3 is a normal microphone for directly detecting a vocal sound represented by a dynamic microphone, a condenser microphone, etc. The normal microphone 3 is connected to the input side of the low-pass filter 16. The output side of the filter 16 is connected to the input side of the analog-digital converter 17, and the output side of the converter 17 is connected to the first changeover switch 1.
8 is connected to one contact portion. The first changeover switch 18 is changed over based on the recognition result of the utterance state recognition unit 13. The other contact of the changeover switch 18 is connected to the movable contact of the second changeover switch 19. The second changeover switch 19 is changed over via the control circuit 20 according to the discrimination result of the voice component discrimination section 14. Furthermore, one fixed contact portion of the second changeover switch 19 is connected to the input side of the high-speed Fourier converter 21, and the output side of the converter 21 is connected to the input side of the comb filter 4. The output side of the comb filter 4 is connected to the input side of the inverse Fourier transformer 22.
【００１５】他方、第二切換スイッチ１９における他方
の固定接点部は減衰回路２３の入力側に接続するととも
に、同回路２３の出力側は遅延回路２４の入力側に接続
する。なお、減衰回路２３には係数器２５を接続する。On the other hand, the other fixed contact portion of the second changeover switch 19 is connected to the input side of the attenuation circuit 23, and the output side of the circuit 23 is connected to the input side of the delay circuit 24. A coefficient unit 25 is connected to the attenuation circuit 23.
【００１６】そして、遅延回路２４の出力側と前記逆フ
ーリェ変換器２２の出力側は、共にデジタル−アナログ
変換器２６の入力側に接続するとともに、同変換器２６
の出力側はローパスフィルタ２７を介して出力部２８の
入力側に接続し、出力部２８の出力側は音声を出力する
スピーカ２９に接続する。The output side of the delay circuit 24 and the output side of the inverse Fourier converter 22 are both connected to the input side of the digital-analog converter 26 and the converter 26
The output side of is connected to the input side of the output unit 28 via the low-pass filter 27, and the output side of the output unit 28 is connected to the speaker 29 that outputs sound.
【００１７】次に、信号処理装置１を用いた本発明に係
る雑音低減方法について、図１及び図２を参照して説明
する。Next, a noise reduction method according to the present invention using the signal processing device 1 will be described with reference to FIGS. 1 and 2.
【００１８】まず、音声は骨伝導マイク２と普通マイク
３により同時に検出される。First, the voice is simultaneously detected by the bone conduction microphone 2 and the ordinary microphone 3.
【００１９】骨伝導マイク２から得る音声信号Ｓｆはロ
ーパスフィルタ１１により高域周波数成分が除去され、
アナログ−デジタル変換器１２によりデジタル信号に変
換される。そして、発声状態認識部１３により発声状態
であるか否かを認識する。発声状態の認識は音声信号Ｓ
ｆに基づく電圧が所定のしきい値以上発生しているか否
かによって認識できる。The high-frequency component of the audio signal Sf obtained from the bone conduction microphone 2 is removed by the low-pass filter 11,
It is converted into a digital signal by the analog-digital converter 12. Then, the uttered state recognition unit 13 recognizes whether or not the uttered state. Speech signal S
It can be recognized by whether or not the voltage based on f is generated above a predetermined threshold value.
【００２０】他方、普通マイク３から得る音声信号Ｓｓ
はローパスフィルタ１６により高域周波数成分が除去さ
れ、アナログ−デジタル変換器１７によりデジタル信号
に変換されるとともに、第一切換スイッチ１８の一方の
接点部に付与される。On the other hand, the voice signal Ss obtained from the ordinary microphone 3
The high-pass frequency component is removed by the low-pass filter 16, converted into a digital signal by the analog-digital converter 17, and applied to one contact portion of the first changeover switch 18.
【００２１】これにより、発声状態認識部１３が音声信
号Ｓｆを認識すれば、第一切換スイッチ１８はＯＮに切
換制御され、音声信号Ｓｓは第一切換スイッチ１８を通
して第二切換スイッチ１９の可動接点部に付与される。
なお、発声状態認識部１３が発声を認識しなければ、第
一切換スイッチ１８はＯＦＦに切換制御され、音声信号
Ｓｓの入力が遮断される。As a result, when the utterance state recognition unit 13 recognizes the voice signal Sf, the first changeover switch 18 is controlled to be turned on, and the voice signal Ss is passed through the first changeover switch 18 to the movable contact of the second changeover switch 19. Given to the department.
If the utterance state recognition unit 13 does not recognize the utterance, the first changeover switch 18 is controlled to be turned off, and the input of the voice signal Ss is cut off.
【００２２】一方、発声状態認識部１３を通過した音声
信号Ｓｆは音声成分判別部１４に付与され、有声音成分
と無声音成分とが判別される。この場合、判別は精度の
高い自己相関法により行われる。即ち、有声音成分は声
帯の基本周波数（ピッチ周波数）の高調波からなり、母
音成分として特徴づけられるとともに、無声音成分はラ
ンダムな雑音成分からなり、子音成分として特徴づけら
れる。また、有限な離散信号における自己相関関数Φ
（ｋ）の一般式に基づいて、音声信号Ｓｆの自己相関を
求めると、ｋ＝０において最大値をとる周期関数とな
り、ｋ≠０の部分での極大値Φ（ｔ1）を見付けること
により、音声信号Ｓｆ中、最も優勢な周期が得られる。
よって、優勢な周期とΦ（ｔ1）の値により有声音成分
と無声音成分の判別を行うことが可能となる。なお、具
体的には、しきい値を設け、Φ（ｔ1）の値がしきい値
より大きく、かつ優勢な周期がある周波数の範囲内の値
である場合、優勢な周期は音声信号の基本周期と判断
し、その音声信号は有声音成分とみなすとともに、それ
以外の場合は無声音成分とみなしている。On the other hand, the voice signal Sf that has passed through the utterance recognition unit 13 is given to the voice component discriminating unit 14 to discriminate a voiced sound component and an unvoiced sound component. In this case, the discrimination is performed by the highly accurate autocorrelation method. That is, the voiced sound component is composed of harmonics of the fundamental frequency (pitch frequency) of the vocal cord and is characterized as a vowel component, and the unvoiced sound component is composed of random noise components and characterized as a consonant component. Also, the autocorrelation function Φ for a finite discrete signal
When the autocorrelation of the audio signal Sf is obtained based on the general expression of (k), it becomes a periodic function which takes a maximum value at k = 0, and by finding the maximum value Φ (t 1 ) in the part of k ≠ 0, , The most dominant cycle of the audio signal Sf is obtained.
Therefore, the voiced sound component and the unvoiced sound component can be discriminated based on the dominant cycle and the value of Φ (t 1 ). Specifically, if a threshold value is set, the value of Φ (t 1 ) is larger than the threshold value, and the dominant cycle is a value within a range of frequencies, the dominant cycle is the voice signal. The voice signal is determined to be the fundamental period, and the voice signal is regarded as the voiced sound component, and otherwise, it is regarded as the unvoiced sound component.
【００２３】そして、音声信号Ｓｆが無声音成分の場合
には、第二切換スイッチ１９は減衰回路２３側に切換え
られ、入力する音声信号Ｓｓ、つまり、無声音成分Ｓｂ
は減衰回路２３に供給される。無声音成分Ｓｂは減衰回
路２３により「１」以下の係数（望ましくは１／５〜１
／１０）に基づいて減衰され、さらに、遅延回路２４を
介して遅延される。これにより、無声音処理成分Ｓｄを
得、同成分Ｓｄはデジタル−アナログ変換器２６の入力
側に付与される。なお、減衰回路２３の減衰量は係数器
２５によって任意に設定でき、望ましくは、雑音成分を
低減できる減衰量を選定する。無声音成分Ｓｂをこのよ
うに処理する理由は次の通りである。高騒音環境下で発
声した音声であっても有声音成分であれば、自己相関法
によりピッチ周波数を検出し、それに従ってスペクトル
・サブトラクションを行えば、Ｓ／Ｎ比の高い音声が得
られる。ところが、無声音成分ランダム雑音のような非
周期的な雑音成分が重なった場合、無声音成分と雑音成
分は性質がよく似ていることからＳ／Ｎ比の向上は容易
なことではない。そこで、音声信号が無声音成分と判別
された場合には、上述した適当な係数（任意に可変可
能）を無声音成分Ｓｂに乗じて減衰させ、疑似子音成分
である無声音処理成分Ｓｄとして出力する。なお、遅延
回路２４は後述する有声音成分Ｓａ側の処理速度に対す
るタイミングを調整するものである。When the voice signal Sf is the unvoiced sound component, the second changeover switch 19 is switched to the attenuation circuit 23 side, and the input voice signal Ss, that is, the unvoiced sound component Sb.
Is supplied to the attenuation circuit 23. The unvoiced sound component Sb is reduced by the attenuation circuit 23 by a coefficient of "1" or less (preferably 1/5 to 1).
/ 10) and is further delayed via the delay circuit 24. As a result, the unvoiced sound processing component Sd is obtained, and the same component Sd is added to the input side of the digital-analog converter 26. The attenuation amount of the attenuation circuit 23 can be arbitrarily set by the coefficient unit 25, and preferably, the attenuation amount that can reduce the noise component is selected. The reason for processing the unvoiced sound component Sb in this way is as follows. If a voiced component is a voiced component even in a high-noise environment, a pitch frequency is detected by the autocorrelation method, and spectral subtraction is performed according to the detected pitch frequency, whereby a voice with a high S / N ratio can be obtained. However, when a non-periodic noise component such as unvoiced sound component random noise overlaps, the unvoiced sound component and the noise component are very similar in nature, so that it is not easy to improve the S / N ratio. Therefore, when the voice signal is determined to be an unvoiced sound component, the unvoiced sound component Sb is multiplied by the above-described appropriate coefficient (which can be changed arbitrarily) to be attenuated and output as the unvoiced sound processing component Sd which is a pseudo consonant component. The delay circuit 24 adjusts the timing with respect to the processing speed on the side of the voiced sound component Sa described later.
【００２４】他方、音声信号Ｓｆが有声音成分の場合に
は、第二切換スイッチ１９は高速フーリェ変換器２１側
に切換えられ、入力する音声信号Ｓｓ、つまり、有声音
成分Ｓａは高速フーリェ変換器２１に供給される。な
お、第二切換スイッチ１９の切換制御は制御部２０を介
して行われ、なめらかな高速切換が実現される。そし
て、有声音成分Ｓａは高速フーリェ変換器２１によりフ
ーリェ変換されるとともに、フーリェ変換により得る周
波数パワースペクトルＰｓは、櫛形フィルタ４の入力側
に供給される。On the other hand, when the voice signal Sf is a voiced sound component, the second changeover switch 19 is switched to the high speed Fourier converter 21 side, and the input voice signal Ss, that is, the voiced sound component Sa, is a high speed Fourier converter. 21. The switching control of the second changeover switch 19 is performed via the control unit 20 to realize smooth high speed switching. Then, the voiced sound component Sa is subjected to Fourier transform by the high-speed Fourier transform unit 21, and the frequency power spectrum Ps obtained by the Fourier transform is supplied to the input side of the comb filter 4.
【００２５】一方、音声成分判別部１４の判別結果が有
声音成分の場合にはピッチ周波数検出部１５によりピッ
チ周波数Ｐｆが求められ、このピッチ周波数Ｐｆは櫛形
フィルタ４の制御端子に付与される。On the other hand, when the discrimination result of the voice component discriminating unit 14 is the voiced sound component, the pitch frequency detecting unit 15 obtains the pitch frequency Pf, and the pitch frequency Pf is given to the control terminal of the comb filter 4.
【００２６】櫛形フィルタ４はピッチ周波数Ｐｆに制御
され、ピッチ周波数Ｐｆとピッチ周波数の高調波成分を
中心周波数とした一定の通過帯域幅を有するフィルタを
構成する。また、櫛形フィルタ４はピッチ周波数Ｐｆの
変化に対応した通過帯域幅が任意に設定される。The comb filter 4 is controlled to the pitch frequency Pf and constitutes a filter having a constant pass band width with the pitch frequency Pf and the harmonic component of the pitch frequency as the center frequency. Further, the comb filter 4 has an arbitrary pass band width corresponding to a change in the pitch frequency Pf.
【００２７】この結果、図２に示すように、櫛形フィル
タ４によって、周波数パワースペクトルＰｓからピッチ
周波数Ｐｆに対応する周波数成分が抽出され、抽出され
た周波数パワースペクトルＰｏは逆フーリェ変換器２２
により逆フーリェ変換される。これにより、雑音成分の
除去された有声音処理成分Ｓｃを得、デジタル−アナロ
グ変換器２６の入力側に付与される。なお、周波数パワ
ースペクトルＰｓから抽出される周波数成分は、基本周
波数乃至予め設定した所定のｎ次高調波成分までであ
り、通常は第三次高調波成分までで十分となる。図２
中、ｆｐは基本周波数成分、２ｆｐは二次高調波成分、
３ｆｐは三次高調波成分を示す。As a result, as shown in FIG. 2, the comb filter 4 extracts the frequency component corresponding to the pitch frequency Pf from the frequency power spectrum Ps, and the extracted frequency power spectrum Po is the inverse Fourier transformer 22.
By the inverse Fourier transform. As a result, the voiced sound processing component Sc from which the noise component has been removed is obtained and applied to the input side of the digital-analog converter 26. The frequency components extracted from the frequency power spectrum Ps are from the fundamental frequency to a predetermined n-th order harmonic component set in advance, and normally the third-order harmonic component is sufficient. Figure 2
Where fp is the fundamental frequency component, 2fp is the second harmonic component,
3fp shows a third harmonic component.
【００２８】そして、デジタル−アナログ変換器２６の
入力側において所定時間だけ遅延された無声音処理成分
Ｓｄと有声音処理成分Ｓｃが加えられ、これにより、普
通マイク３から得る音声信号Ｓｓに対して雑音低減処理
及び時間調整処理を行った音声処理信号Ｓｏが得られる
とともに、この音声処理信号Ｓｏは同変換器２６により
アナログ信号に変換され、ローパスフィルタ２７、出力
部２８を介し、スピーカ２９から音声として出力され
る。Then, the unvoiced sound processing component Sd and the voiced sound processing component Sc delayed by a predetermined time are added at the input side of the digital-analog converter 26, whereby noise is added to the voice signal Ss obtained from the ordinary microphone 3. The audio processing signal So that has been subjected to the reduction processing and the time adjustment processing is obtained, and this audio processing signal So is converted into an analog signal by the converter 26, and is output as audio from the speaker 29 via the low-pass filter 27 and the output unit 28. Is output.
【００２９】以上、実施例について詳細に説明したが、
本発明はこのような実施例に限定されるものではない。
例えば、音声信号は切換スイッチに分離して有声音成分
と無声音成分を得たが、分離しなくてもよい。また、周
波数パワースペクトルへの変換はフーリェ変換以外の方
法を用いても勿論よい。その他、細部の回路構成、手法
等において本発明の要旨を逸脱しない範囲で任意に変更
できる。The embodiment has been described in detail above.
The present invention is not limited to such an embodiment.
For example, the voice signal is separated into the changeover switch to obtain the voiced sound component and the unvoiced sound component, but it is not necessary to separate them. In addition, it is of course possible to use a method other than the Fourier transform for the conversion into the frequency power spectrum. In addition, the detailed circuit configuration and method may be arbitrarily changed without departing from the scope of the present invention.
【００３０】[0030]
【発明の効果】このように、本発明に係る雑音低減方法
は加速度形ピックアップと普通マイクにより音声を同時
に検出し、普通マイクから得る音声信号を有声音成分と
無声音成分に判別するとともに、有声音成分を周波数パ
ワースペクトルに変換し、他方、加速度形ピックアップ
から得る音声信号に基づいて有声音成分のピッチ周波数
を得、前記周波数パワースペクトルからピッチ周波数に
対応する周波数成分を抽出するとともに、抽出した周波
数パワースペクトルを音声信号における有声音処理成分
に変換し、この有声音処理成分と無声音成分又は無声音
成分を減衰して得た無声音処理成分を加えて音声処理信
号を得るようにしたため、高騒音環境下の通信システム
における通話音声のＳ／Ｎ比を高めることができるとと
もに、同時に通話音声の音質、即ち、音声の明瞭性及び
自然性を飛躍的に向上できる。As described above, according to the noise reduction method of the present invention, the voice is simultaneously detected by the acceleration type pickup and the ordinary microphone, and the voice signal obtained from the ordinary microphone is discriminated into the voiced sound component and the unvoiced sound component. The component is converted to a frequency power spectrum, while the pitch frequency of the voiced sound component is obtained based on the voice signal obtained from the acceleration pickup, and the frequency component corresponding to the pitch frequency is extracted from the frequency power spectrum, and the extracted frequency Since the power spectrum is converted to the voiced sound processing component in the voice signal, and the voiced sound processing component and the unvoiced sound processing component obtained by attenuating the unvoiced sound component or the unvoiced sound component are added to obtain the voice processing signal, it is possible to operate in a high noise environment. Can improve the S / N ratio of the call voice in the communication system of Voice of sound quality, that is, can dramatically improve the clarity and naturalness of sound.
【図１】本発明に係る雑音低減方法を実施できる信号処
理装置のブロック回路図、FIG. 1 is a block circuit diagram of a signal processing device capable of implementing a noise reduction method according to the present invention.
【図２】周波数パワースペクトルを示す作用説明図、FIG. 2 is an operation explanatory view showing a frequency power spectrum,
２ 骨伝導マイク ３ 普通マイク ４ 櫛形フィルタ Ｓｓ 音声信号 Ｓｆ 音声信号 Ｓａ 有声音成分 Ｓｂ 無声音成分 Ｓｃ 有声音処理成分 Ｓｄ 無声音処理成分 Ｓｏ 音声処理信号 2 Bone conduction microphone 3 Normal microphone 4 Comb filter Ss Voice signal Sf Voice signal Sa Voiced sound component Sb Unvoiced sound component Sc Voiced sound processing component Sd Unvoiced sound processing component So Voice processing signal
Claims (9)
ンにより音声を同時に検出し、普通マイクロホンから得
る音声信号を有声音成分と無声音成分に判別するととも
に、有声音成分を周波数パワースペクトルに変換し、他
方、加速度形ピックアップから得る音声信号に基づいて
有声音成分のピッチ周波数を得、前記周波数パワースペ
クトルから前記ピッチ周波数に対応する周波数成分を抽
出するとともに、抽出した周波数パワースペクトルを音
声信号におけ有声音処理成分に変換し、この有声音処理
成分と無声音成分又は無声音成分を処理した無声音処理
成分を加えて音声処理信号を得ることを特徴とする雑音
低減方法。1. A voice is simultaneously detected by an acceleration type pickup and an ordinary microphone, a voice signal obtained from the ordinary microphone is discriminated into a voiced sound component and an unvoiced sound component, and the voiced sound component is converted into a frequency power spectrum. Form the pitch frequency of the voiced sound component based on the voice signal obtained from the pickup, and extract the frequency component corresponding to the pitch frequency from the frequency power spectrum, the extracted frequency power spectrum in the voice signal voiced sound processing component And a voiceless signal is obtained by adding a voiced sound processing component and an unvoiced sound component or an unvoiced sound processing component obtained by processing the unvoiced sound component.
ワースペクトルを得るとともに、周波数パワースペクト
ルを逆フーリェ変換して有声音処理成分を得ることを特
徴とする請求項１記載の雑音低減方法。2. The noise reduction method according to claim 1, wherein the voiced sound component is subjected to a Fourier transform to obtain a frequency power spectrum, and the frequency power spectrum is subjected to an inverse Fourier transform to obtain a voiced sound processing component.
周波数成分は、基本周波数乃至予め設定した所定のｎ次
高調波成分であることを特徴とする請求項１記載の雑音
低減方法。3. The noise reduction method according to claim 1, wherein the frequency component extracted from the frequency power spectrum is a fundamental frequency or a predetermined n-th harmonic component set in advance.
きさに減衰して得ることを特徴とする請求項１記載の雑
音低減方法。4. The noise reduction method according to claim 1, wherein the unvoiced sound processing component is obtained by attenuating the unvoiced sound component to a predetermined size.
とを特徴とする請求項４記載の雑音低減方法。5. The noise reduction method according to claim 4, wherein the unvoiced sound processing component is delayed for a predetermined time.
特徴とする請求項５記載の雑音低減方法。6. The noise reduction method according to claim 5, wherein the delay time is variable.
に基づいて有声音成分と無声音成分を判別し、この判別
結果により普通マイクロホンから得る音声信号を有声音
成分と無声音成分に選択的に分離することを特徴とする
請求項１記載の雑音低減方法。7. A voiced sound component and an unvoiced sound component are discriminated based on a voice signal obtained from an acceleration pickup, and a voice signal obtained from a normal microphone is selectively separated into a voiced sound component and an unvoiced sound component based on the discrimination result. The noise reduction method according to claim 1, which is characterized in that.
ホンを用いることを特徴とする請求項１記載の雑音低減
方法。8. The noise reduction method according to claim 1, wherein the acceleration type pickup uses a bone conduction microphone.
分の抽出は、櫛形フィルタを用いることを特徴とする請
求項１記載の雑音低減方法。9. The noise reduction method according to claim 1, wherein a comb filter is used to extract the frequency component from the frequency power spectrum.
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| JP3031815A JPH07101853B2 (en) | 1991-01-30 | 1991-01-30 | Noise reduction method |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| JP3031815A JPH07101853B2 (en) | 1991-01-30 | 1991-01-30 | Noise reduction method |
Publications (2)
| Publication Number | Publication Date |
|---|---|
| JPH04245720A JPH04245720A (en) | 1992-09-02 |
| JPH07101853B2 true JPH07101853B2 (en) | 1995-11-01 |
Family
ID=12341592
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| JP3031815A Expired - Fee Related JPH07101853B2 (en) | 1991-01-30 | 1991-01-30 | Noise reduction method |
Country Status (1)
| Country | Link |
|---|---|
| JP (1) | JPH07101853B2 (en) |
Families Citing this family (5)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| US7447630B2 (en) * | 2003-11-26 | 2008-11-04 | Microsoft Corporation | Method and apparatus for multi-sensory speech enhancement |
| KR100778143B1 (en) * | 2005-08-13 | 2007-11-23 | 백다리아 | A Headphone with neck microphone using bone conduction vibration |
| JP2007267331A (en) * | 2006-03-30 | 2007-10-11 | Railway Technical Res Inst | Combination microphone system for speaking voice collection |
| EP2458586A1 (en) * | 2010-11-24 | 2012-05-30 | Koninklijke Philips Electronics N.V. | System and method for producing an audio signal |
| CN107910011B (en) * | 2017-12-28 | 2021-05-04 | 科大讯飞股份有限公司 | Voice noise reduction method and device, server and storage medium |
-
1991
- 1991-01-30 JP JP3031815A patent/JPH07101853B2/en not_active Expired - Fee Related
Also Published As
| Publication number | Publication date |
|---|---|
| JPH04245720A (en) | 1992-09-02 |
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