EP1202253A2 - Adaptiver Geräuschpegelschätzer - Google Patents
Adaptiver Geräuschpegelschätzer Download PDFInfo
- Publication number
- EP1202253A2 EP1202253A2 EP01440295A EP01440295A EP1202253A2 EP 1202253 A2 EP1202253 A2 EP 1202253A2 EP 01440295 A EP01440295 A EP 01440295A EP 01440295 A EP01440295 A EP 01440295A EP 1202253 A2 EP1202253 A2 EP 1202253A2
- Authority
- EP
- European Patent Office
- Prior art keywords
- value
- input signal
- noise level
- determined
- estimate
- 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
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Classifications
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- 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
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- 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)
Abstract
Description
dass in einem ersten Schritt ein vorgebbarer Initialisierungswert n0 als Schätzwert n(x) für einen aktuellen Geräuschpegel n angenommen wird;
dass im nächsten und ggf. in weiteren Schritten der Schätzwert n(x) des Geräuschpegels n für ein in vorzugsweise äquidistanten Zeitschritten T jeweils zu Zeitpunkten k mit einer Abtastfrequenz fs = 1/T abgetastetes Eingangssignal x(k) als ein Wert n1(x) bestimmt wird, der durch den Minimalwert aus der Menge aller sukzessive jeweils innerhalb eines Kurzzeitintervalls mit einer zeitlichen Länge ts ≥ 1 ms, vorzugsweise ts ≥ 3ms gefundenen Maximumwerte des Eingangssignals x(k) ermittelt wird;
dass der Wert n1(x) als Schätzwert n(x) für den aktuellen Geräuschpegel n übernommen wird, wenn die dynamischen Variationen des Eingangssignals x(k) einen vorgebbaren Schwellwert ε unterschreiten; und
dass der im vorhergehenden Schritt ermittelte Schätzwert n(x) ungeändert als neuer Schätzwert n(x) für den aktuellen Geräuschpegel n übernommen wird, wenn die dynamischen Variationen des Eingangssignals x(k) einen vorgebbaren Schwellwert ε überschreiten.
Claims (10)
- Verfahren zur Ermittlung eines Schätzwerts für den Geräuschpegel n eines Hintergrundgeräusches, das einem über ein Telekommunikations (=TK)-System übertragenen akustischen Nutzsignal, insbesondere einem menschlichen Sprachsignal überlagert ist,
dadurch gekennzeichnet, dass in einem ersten Schritt ein vorgebbarer Initialisierungswert n0 als Schätzwert n(x) für einen aktuellen Geräuschpegel n angenommen wird;
dass im nächsten und ggf. in weiteren Schritten der Schätzwert n(x) des Geräuschpegels n für ein in vorzugsweise äquidistanten Zeitschritten T jeweils zu Zeitpunkten k mit einer Abtastfrequenz fs = 1/T abgetastetes Eingangssignal x(k) als ein Wert n1(x) bestimmt wird, der durch den Minimalwert aus der Menge aller sukzessive jeweils innerhalb eines Kurzzeitintervalls mit einer zeitlichen Länge ts ≥ 1ms, vorzugsweise ts ≥ 3ms gefundenen Maximumwerte des Eingangssignals x(k) ermittelt wird;
dass der Wert n1(x) als Schätzwert n(x) für den aktuellen Geräuschpegel n übernommen wird, wenn die dynamischen Variationen des Eingangssignals x(k) einen vorgebbaren Schwellwert ε unterschreiten;
und dass der im vorhergehenden Schritt ermittelte Schätzwert n(x) ungeändert als neuer Schätzwert n(x) für den aktuellen Geräuschpegel n übernommen wird, wenn die dynamischen Variationen des Eingangssignals x(k) einen vorgebbaren Schwellwert ε überschreiten. - Verfahren nach Anspruch 1, dadurch gekennzeichnet, dass ts = 1/fug, wobei fug die untere Grenzfrequenz des übertragenden TK-Systems ist.
- Verfahren nach Anspruch 2, dadurch gekennzeichnet, dass fug ≤ 500Hz, vorzugsweise fug ≤ 330Hz und fug ≥ 10Hz.
- Verfahren nach Anspruch 1, dadurch gekennzeichnet, dass als Initialisierungswert n0 der maximal darstellbare Wert des Zielsystems für die Signalübertragung innerhalb des TK-Systems gewählt wird.
- Verfahren nach Anspruch 1, dadurch gekennzeichnet, dass für die Bestimmung des Schätzwerts n(x) der Wert n1(x) auf einen vorgebbaren oder festen unteren Grenzwert nmin gesetzt wird, falls ein Wert n1(x) < nmin ermittelt wird.
- Verfahren nach Anspruch 1, dadurch gekennzeichnet, dass für die Bestimmung des Schätzwerts n(x) der Wert n1(x) auf einen vorgebbaren oder festen oberen Grenzwert nmax gesetzt wird, falls ein Wert n1(x) > nmax ermittelt wird.
- Verfahren nach Anspruch 1, dadurch gekennzeichnet, dass die innerhalb der Kurzzeitintervalle gefundenen Maximumwerte des Eingangssignals x(k) mit einem Skalierungsfaktor S < 1 multipliziert in die Ermittlung des Werts n1(x) eingehen.
- Verfahren nach Anspruch 1, dadurch gekennzeichnet, dass ein Schwellwert ε = ε(x) adaptiv mit der Rauhigkeit des Pegels des Eingangssignals x(k) verändert wird.
- Prozessorbaugruppe, insbesondere digitaler Signalprozessor (=DSP) zur Unterstützung des Verfahrens nach einem der vorhergehenden Ansprüche.
- Programmierbare Gate-Array-Baugruppe zur Unterstützung des Verfahrens nach einem der Ansprüche 1 bis 8.
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 (de) | 2002-05-02 |
| EP1202253A3 EP1202253A3 (de) | 2004-01-02 |
| EP1202253B1 EP1202253B1 (de) | 2005-04-20 |
Family
ID=7660840
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| EP01440295A Expired - Lifetime EP1202253B1 (de) | 2000-10-24 | 2001-09-12 | Adaptiver Geräuschpegelschätzer |
Country Status (5)
| Country | Link |
|---|---|
| US (1) | US6842526B2 (de) |
| EP (1) | EP1202253B1 (de) |
| JP (1) | JP2002198918A (de) |
| AT (1) | ATE293828T1 (de) |
| DE (2) | DE10052626A1 (de) |
Cited By (1)
| 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)
| 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 |
Family Cites Families (8)
| 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 (en) * | 1988-03-11 | 1989-09-21 | British Telecommunications Public Limited Company | Voice activity detection |
| WO1992015986A1 (en) * | 1991-03-05 | 1992-09-17 | Picturetel Corporation | Variable bit rate speech encoder |
| 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/de 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)
| 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 (de) | 2005-04-20 |
| DE50105947D1 (de) | 2005-05-25 |
| ATE293828T1 (de) | 2005-05-15 |
| EP1202253A3 (de) | 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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