EP1202253A2 - Estimateur actif de niveau de bruit - Google Patents

Estimateur actif de niveau de bruit Download PDF

Info

Publication number: EP1202253A2
Authority: EP; European Patent Office
Prior art keywords: value; input signal; noise level; determined; estimate
Prior art date: 2000-10-24
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

EP01440295A

Other languages

German (de)

English (en)

Other versions

EP1202253B1 (fr

EP1202253A3 (fr

Inventor

Michael Walker

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.)

Alcatel Lucent SAS

Nokia Inc

Original Assignee

Alcatel SA

Nokia Inc

Priority date (The priority date 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 date listed.)

2000-10-24

Filing date

2001-09-12

Publication date

2002-05-02

2001-09-12 Application filed by Alcatel SA, Nokia Inc filed Critical Alcatel SA

2002-05-02 Publication of EP1202253A2 publication Critical patent/EP1202253A2/fr

2004-01-02 Publication of EP1202253A3 publication Critical patent/EP1202253A3/fr

2005-04-20 Application granted granted Critical

2005-04-20 Publication of EP1202253B1 publication Critical patent/EP1202253B1/fr

2021-09-12 Anticipated expiration legal-status Critical

Status Expired - Lifetime legal-status Critical Current

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Classifications

- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Speech 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
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
- G10L21/0208—Noise filtering

Definitions

the MAM value often also reflects a too high one Noise level compared to the actual noise level.
noise estimators In general, the value of a signal's noise level is common to many all algorithms of great importance for signal processing as a threshold or control value. The reliability and temporal behavior of a noise estimator have a great influence on the achievable quality of a signal processing algorithm. This is especially true in the field of speech recognition, to improve the detection rate in the field of echo cancellation and for noise reduction. Areas of application for noise estimators are, for example, switching systems, conference facilities, but also conventional telephones or cell phones.
a disadvantage of known estimation methods is the relatively slow behavior the averaging in the noise estimator. Especially with speech activity with only short ones Speech breaks in periods of ⁇ 100 ms are often not enough time to to capture the "noise floor”.
noise threshold Another problem with the noise threshold is that of successful speech level estimation performed noise update in the event of changes in time Environmental conditions.
the estimated noise level thus fluctuates in certain, sometimes relatively large limits.
the object of the present invention is a method of kind described above with the simplest possible means to further develop the most exact possible determination of the current noise level is achieved with the fastest possible adaptation times, which is considerable are lower than in known methods, and in addition that is as low as possible Computational effort is required.
a short-term interval length ts is a maximum value from the samples of the input signal x (k) determined and for the estimation of the current noise level from the set of several maximum values found one after the other Minimum n1 (x) is used as an estimate n (x) for the current noise level n.
an initialization value n0 is specified.
the time length ts should in each case be chosen such that a Adaptation of low-frequency signals in the range ⁇ 100 Hz excluded becomes.
the lower limit frequencies are usually in a range fug ⁇ 500 Hz.
the lower is Cutoff frequency 330 Hz.
a value of around 10 Hz as the lower limit for the lower Cut-off frequency fug corresponds to the value of a conventional hi-fi amplifier and is therefore reasonable.
n1 (x) is set to a predeterminable or fixed lower limit value n min for determining the estimated value n (x) if a value n1 (x) ⁇ n min is determined.
the value n1 (x) is set to a predefinable or fixed upper limit value n max for determining the estimated value n (x) if a value n1 (x)> n max is determined becomes.
n max is chosen to be less than or equal to the initialization value n0, preferably n max n n0 -16dB.
this upper limit is predetermined by the statistically determined speech dynamics of human speech.
Another advantageous embodiment of the method according to the invention provides that the maximum values found within the short-term intervals of the input signal x (k) multiplied by a scaling factor S ⁇ 1 in the Determine the value n1 (x). The majority of the actual level values is actually below the maximum value determined in each case within the relevant short-term interval.
the scaling factor S ⁇ 0.5 corresponds approximately to the position the maximum value of a statistical distribution, for example a Gaussian distribution of the samples in relation to the location of the maximum found Level value. This will average the actual current noise level n hit much better than by using the unscaled Maximum value.
n (x) is a measure of one currently estimated noise level is scaled by a factor D> 1.
⁇ ⁇ (x) adaptive with the Roughness of the level of the input signal x (k) can be changed. With that leaves an optimal and extremely fast update and adjustment of the reach the estimated level value to the actual noise conditions.
a server unit also falls within the scope of the present invention Processor assembly and a gate array assembly to support the The inventive method described above and a computer program to carry out the procedure.
the method can be used as a hardware circuit, as well as in the form of a computer program.
software programming for powerful DSP's preferred because new insights and additional functions are easier through a Software changes can be implemented on existing hardware basis are.
methods can also be used as hardware modules, for example in IP or TK terminals or conventional telephone systems implemented become.
the invention is shown in the drawing and is based on exemplary embodiments explained in more detail.
the figure shows a highly schematic principle diagram of how a Estimating device for carrying out the method according to the invention.
a sampled input signal x (k) becomes a first estimated value n1 (x) for the noise level n, the background noise superimposed on a useful signal in the input signal x (k), in a first short-term interval of time length ts ⁇ 1 ms calculated according to the following equation:
K fs / fug is the quotient of the sampling frequency of the sampled input signal x (k) and the lower cut-off frequency of the transmitting telecommunications system.
n1 (x) assumes a value dependent on the speech level, since the Speech level is louder than the noise. For example, a is acceptable S / N ratio of 6 dB.
n1 (x) found in this way still changes with the language, responds but on noise reduction and during speech breaks with extremely short Adaptation time.
n1 (x) is only adopted as the actual estimate n (x) for the current noise level n if the dynamic variations of the input signal x (k) fall below a predeterminable threshold value ⁇ , that is if dx (i) ... dx (i-ts) ⁇
the envelope of the incoming input signals x (i) is now "stable", that is with a probability bordering on certainty that there are no speech signals, can assign the current level values directly to the background noise become. Otherwise, if the envelope "wobbles", there is a high probability Speech, i.e. predominant useful signal in the input signal x (i) before, see above that the peaks of the input signal are not for the estimation of the background noise can be used. In this case, as above described, a scaled noise value is obtained from the speech signal itself become.
the drawing now shows this process in a schematic manner, in particular the maximum formation from the input signal x (k), the scaling with a Scaling factor S and the minimum formation for obtaining the value n1 (x), the acceptance of this value depending on a speech pause detector (SPD), the initial value of which may be based on an application-specific Factor D is scaled, as well as the threshold value estimation of the dynamic Variations of the input signal x (k) that in the example shown the temporal change in the short-term mean value dsam (x) / dt.
SPD speech pause detector

Landscapes

Engineering & Computer Science (AREA)
Acoustics & Sound (AREA)
Signal Processing (AREA)
Health & Medical Sciences (AREA)
Audiology, Speech & Language Pathology (AREA)
Human Computer Interaction (AREA)
Physics & Mathematics (AREA)
Computational Linguistics (AREA)
Quality & Reliability (AREA)
Multimedia (AREA)
Cable Transmission Systems, Equalization Of Radio And Reduction Of Echo (AREA)
Noise Elimination (AREA)
Telephonic Communication Services (AREA)
Monitoring And Testing Of Transmission In General (AREA)
Fittings On The Vehicle Exterior For Carrying Loads, And Devices For Holding Or Mounting Articles (AREA)
Telephone Function (AREA)
Monitoring And Testing Of Exchanges (AREA)
Control Of Amplification And Gain Control (AREA)

EP01440295A 2000-10-24 2001-09-12 Estimateur actif de niveau de bruit Expired - Lifetime EP1202253B1 (fr)

Applications Claiming Priority (2)

Application Number	Priority Date	Filing Date	Title
DE10052626A DE10052626A1 (de)	2000-10-24	2000-10-24	Adaptiver Geräuschpegelschätzer
DE10052626		2000-10-24

Publications (3)

Publication Number	Publication Date
EP1202253A2 true EP1202253A2 (fr)	2002-05-02
EP1202253A3 EP1202253A3 (fr)	2004-01-02
EP1202253B1 EP1202253B1 (fr)	2005-04-20

Family

ID=7660840

Family Applications (1)

Application Number	Title	Priority Date	Filing Date
EP01440295A Expired - Lifetime EP1202253B1 (fr)	2000-10-24	2001-09-12	Estimateur actif de niveau de bruit

Country Status (5)

Country	Link
US (1)	US6842526B2 (fr)
EP (1)	EP1202253B1 (fr)
JP (1)	JP2002198918A (fr)
AT (1)	ATE293828T1 (fr)
DE (2)	DE10052626A1 (fr)

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number	Priority date	Publication date	Assignee	Title
US20220319532A1 (en) *	2019-08-30	2022-10-06	Dolby Laboratories Licensing Corporation	Pre-conditioning audio for machine perception

Families Citing this family (8)

* Cited by examiner, † Cited by third party
Publication number	Priority date	Publication date	Assignee	Title
JP4601970B2 (ja) *	2004-01-28	2010-12-22	株式会社エヌ・ティ・ティ・ドコモ	有音無音判定装置および有音無音判定方法
JP4490090B2 (ja) *	2003-12-25	2010-06-23	株式会社エヌ・ティ・ティ・ドコモ	有音無音判定装置および有音無音判定方法
US20050226442A1 (en) *	2004-04-12	2005-10-13	Landon Michael D	Method and apparatus for achieving temporal volume control
JP4551817B2 (ja) *	2005-05-20	2010-09-29	Ｏｋｉセミコンダクタ株式会社	ノイズレベル推定方法及びその装置
US8275153B2 (en) *	2007-04-16	2012-09-25	Evertz Microsystems Ltd.	System and method for generating an audio gain control signal
US8894316B2 (en) *	2009-07-22	2014-11-25	Music Express, Llc	Adjustable joint for microphone
CN103238180A (zh) *	2010-11-25	2013-08-07	日本电气株式会社	信号处理设备、信号处理方法及信号处理程序
US10978096B2 (en) *	2017-04-25	2021-04-13	Qualcomm Incorporated	Optimized uplink operation for voice over long-term evolution (VoLte) and voice over new radio (VoNR) listen or silent periods

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Publication number	Priority date	Publication date	Assignee	Title
US3855423A (en) *	1973-05-03	1974-12-17	Bell Telephone Labor Inc	Noise spectrum equalizer
US4000369A (en) *	1974-12-05	1976-12-28	Rockwell International Corporation	Analog signal channel equalization with signal-in-noise embodiment
DE3243232A1 (de) *	1982-11-23	1984-05-24	Philips Kommunikations Industrie AG, 8500 Nürnberg	Verfahren zur erkennung von sprachpausen
US4885790A (en) *	1985-03-18	1989-12-05	Massachusetts Institute Of Technology	Processing of acoustic waveforms
WO1989008910A1 (fr) *	1988-03-11	1989-09-21	British Telecommunications Public Limited Company	Detection d'activite vocale
WO1992015986A1 (fr) *	1991-03-05	1992-09-17	Picturetel Corporation	Codeur vocal a debit binaire variable
US5341456A (en) *	1992-12-02	1994-08-23	Qualcomm Incorporated	Method for determining speech encoding rate in a variable rate vocoder
US5485522A (en) *	1993-09-29	1996-01-16	Ericsson Ge Mobile Communications, Inc.	System for adaptively reducing noise in speech signals

2000
- 2000-10-24 DE DE10052626A patent/DE10052626A1/de not_active Withdrawn
2001
- 2001-09-12 DE DE50105947T patent/DE50105947D1/de not_active Expired - Fee Related
- 2001-09-12 EP EP01440295A patent/EP1202253B1/fr not_active Expired - Lifetime
- 2001-09-12 AT AT01440295T patent/ATE293828T1/de not_active IP Right Cessation
- 2001-10-11 US US09/973,828 patent/US6842526B2/en not_active Expired - Fee Related
- 2001-10-18 JP JP2001320228A patent/JP2002198918A/ja active Pending

Cited By (2)

* Cited by examiner, † Cited by third party
Publication number	Priority date	Publication date	Assignee	Title
US20220319532A1 (en) *	2019-08-30	2022-10-06	Dolby Laboratories Licensing Corporation	Pre-conditioning audio for machine perception
US12080317B2 (en) *	2019-08-30	2024-09-03	Dolby Laboratories Licensing Corporation	Pre-conditioning audio for echo cancellation in machine perception

Also Published As

Publication number	Publication date
EP1202253B1 (fr)	2005-04-20
DE50105947D1 (de)	2005-05-25
ATE293828T1 (de)	2005-05-15
EP1202253A3 (fr)	2004-01-02
JP2002198918A (ja)	2002-07-12
US20020064288A1 (en)	2002-05-30
US6842526B2 (en)	2005-01-11
DE10052626A1 (de)	2002-05-02

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