EP1016314B1 - Verfahren zur optimierung des empfangs akustischer signale und elektrisches gerät - Google Patents
Verfahren zur optimierung des empfangs akustischer signale und elektrisches gerät Download PDFInfo
- Publication number
- EP1016314B1 EP1016314B1 EP98916975A EP98916975A EP1016314B1 EP 1016314 B1 EP1016314 B1 EP 1016314B1 EP 98916975 A EP98916975 A EP 98916975A EP 98916975 A EP98916975 A EP 98916975A EP 1016314 B1 EP1016314 B1 EP 1016314B1
- Authority
- EP
- European Patent Office
- Prior art keywords
- speech
- electrical
- source
- sound source
- signals
- 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 - Lifetime
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Classifications
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R3/00—Circuits for transducers, loudspeakers or microphones
- H04R3/005—Circuits for transducers, loudspeakers or microphones for combining the signals of two or more microphones
Definitions
- the invention relates to a method according to the genus independent claim 1 and of an electrical device according to the genus of independent claim 5.
- Telephone terminals There are already electrical devices in the form of Telephone terminals known that have a voice input enable.
- the voice input takes place, for example via a hands-free microphone.
- the inventive method with the features of independent claim 1 and the invention electrical device with the characteristics of independent Claim 5 have the advantage that the phase-shifted superimposition of the electrical Microphone output signals have a characteristic directivity is achieved.
- the sensitivity can increase a predetermined point in the room can be increased so that a sound source arranged there particularly well through the Microphones can be recorded and a fade out Interference signal sources at other points in the room becomes.
- the Word recognition probability of a Speech recognition system is increased accordingly and the Word analysis simplified. The signal gets less from Background noise falsified.
- phase delays of the at least one phase delay element can be set. In this way, a location-independent setting is a Maximum reception for the beat signal possible.
- a Signal processing unit is provided, the electrical signals of the microphones are supplied and depending on the amplitudes of the electrical signals Coordinates of at least one sound source determined.
- the microphone from the recorded signals two or calculate three-dimensional images of the sound environment so that all sound sources can be determined locally.
- this information can then be the phase delay of the at least one phase delay element set in this way be that for a desired sound source Received maximum for the beat signal results.
- the Signal processing unit a speech analysis device is assigned and that the speech analysis device one Comparison of parameters of the electrical signals with in an assigned storage unit Language parameters carried out and with one depending of the comparison result Probability value of a sound source as a speech source identified.
- the phase delay of the at least one phase delay element be set to be at the location of the language source for the beat signal gives a maximum reception. Consequently are speech signals from this speech source with a high Sensitivity received, whereas interference signals from other sound sources are hidden.
- a particular advantage is that the Signal processing unit the phase delay of the at least one phase delay element depending the location of the identified language source so that yourself in the place of the language source for that Superposition signal results in a maximum reception.
- the phase delay of the at least one phase delay element automatically without User intervention, the location of the largest Sensitivity also adaptively to the location of the Language source can be updated. This represents for the User a significant improvement in ease of use represents.
- FIG. 1 shows a block diagram of a electrical input device with two voice inputs Microphones whose electrical output signals without Phase shift are superimposed
- Figure 2 a Block diagram of an electrical device with at least three microphones, their electrical output signals can be superimposed out of phase.
- 1 denotes a telecommunication terminal trained voice-input-capable electrical device.
- the Telecommunication terminal 1 comprises an adder 20 and a speech processing unit 70.
- An output 97 of the Adder 20 is with an input 107 of Speech processing unit 70 connected.
- An output 108 of the Speech processing unit 70 is not connected to one in FIG Telecommunications network shown connected.
- To the Telecommunications terminal 1 is a first microphone 5 and a second microphone 10 is connected.
- FIG. 1 At the same distance from the two microphones 5, 10 is a as in FIG Loudspeaker trained sound source 55 arranged Emits speech signals.
- the speech signals are from the two microphones 5, 10 according to that in FIG. 1 received dashed arrows.
- Speech source trained sound source 55 can be for example, the voice organ of a user of the Act telecommunications terminal 1.
- the microphones 5, 10 convert the received speech signals into electrical signals around and forward them to adder 20, where they can be superimposed by simple addition. Since the Speech source 55 of the two microphones 5, 10 is the same is spaced, the speech signal it emits by superimposing the electrical output signals of the Microphones 5, 10 scored twice in adder 20.
- the Language source 55 is thus in a place for which the beat signal at output 97 of the adder 20 results in a sensitivity or reception maximum.
- the Locations of the reception maxima are repeated with the Distance of the wavelength of the signal. Because language one represents statistically distributed frequency mix on average, only one reception maximum in the geometric Middle between the two microphones 5, 10 according to one dashed line 200 in Figure 1.
- FIG 2 is a telecommunications terminal trained speech input according to the invention electrical device 1 shown. It includes a first one Phase delay 30, a second Phase delay 35, a third Phase delay 40 and a fourth Phase delay element 45.
- the telecommunications terminal 1 also has a signal processing unit 50, a Speech analysis device 60 and a storage unit 65. Furthermore, there is a first one in the telecommunications terminal 1 Adder 20 and a second adder 25 as well a speech processing unit 70 is provided.
- To the Telecommunications terminal 1 is a first microphone 5, a second microphone 10 and a third microphone 15 connected. There is an output 104 of the first Microphone 5 to a first input 85 of the first Phase delay element 30 and to a first input 75 connected to the signal processing unit 50.
- An exit 86 of the first phase delay element 30 is with a first non-inverting input 87 of the first Addition member 20 connected.
- An output 105 of the second Microphone 10 is at a first input 88 of the second Phase delay element 35 and to a second input 76 connected to the signal processing unit 50.
- An exit 89 of the second phase delay element 35 is connected to a second non-inverting input 90 of the first Addition member 20 connected.
- An output 106 of the third Microphone 15 is at a first input 94 of the fourth Phase delay element 45 and to a third input 77 connected to the signal processing unit 50.
- An exit 95 of the fourth phase delay element 45 is with a first non-inverting input 96 of the second Addition member 25 connected.
- the microphone can have its output via a connecting line shown in dashed lines in Figure 2 to a first input 91 of the third Phase delay element 40 and to a fourth input 78 the signal processing unit 50 are connected.
- On Output 92 of the fourth phase delay element 40 is with a second non-inverting input 93 of the second Addition member 25 connected.
- An exit 97 of the first Adder 20 is with a third non-inverting input 98 of the second adder 25 connected.
- An output 99 of the second adder 25 is with an input 107 of the speech processing unit 70 connected.
- An output 108 of the speech processing unit 70 is to a not shown in Figure 2 Telecommunications network connected.
- the Has speech processing unit 70 according to Figure 1 and Figure 2 the task, the superimposed electrical voice signals for prepare the transmission in the telecommunications network and to deliver to this. If necessary, others can Microphones connected to the telecommunications terminal 1 and via appropriate phase delay and Addition elements with the other electrical voice signals superimposed and fed to the speech processing unit 70 become. In addition, the output signals of these are further Microphones also the signal processing unit 50 supply.
- the Signal processing unit 50 has a fifth input 79, which is connected to an output 110 of the memory unit 65 is.
- a first output 81 is the Signal processing unit 50 with a second input 100 of the first phase delay element 30 connected.
- On second output 82 of the signal processing unit 50 is connected a second input 101 of the second Phase delay element 35 connected.
- a third exit 83 of the signal processing unit 50 is with a second Input 102 of the third phase delay element 40 connected.
- a fourth exit 84 of the Signal processing unit 50 has a second input 103 of the fourth phase delay element 45 connected.
- Figure 2 is again a speaker trained sound source 55 shown, the speech signals emits and for example the speech organ of a user can represent.
- the speech signals the sound source 55 designed as a speech source Figure 2 is the language source 55 on a a dashed line 200 depicting the geometric location, in contrast to the arrangement according to FIG. 1 no longer the geometric center between the three microphones 5, 10, 15 forms, so that the three microphones 5, 10, 15 of the Speech source 55 are spaced differently.
- a non-central directional effect can be the location for which the beat signal of the electrical speech signals gives a maximum of reception appropriate choice of phase delay of each Phase delay elements 30, 35, 40, 45 can be specified. This can also for the non-central arrangement Speech source 55 according to FIG. 2 reaches a reception maximum become. Depending on the location of the language source 55, it can already be sufficient, only a single microphone output signal delay in the phase, so limited to this Use only one phase delay element required would. By using a phase delay element for however, each microphone is more flexible for the specification of the location for the language source 55, at which the Beat signal at input 107 of Speech processing unit 70 results in a maximum reception.
- the reception sensitivity of the Telecommunication terminal 1 for certain areas reinforce or reduce so that disturbing sound sources in Areas of low sensitivity essentially hidden and useful sound sources in the area increased Sensitivity can be received better. For every Sound source can be the reception sensitivity in one specified range.
- the signal processing unit 50 can optionally also calculate a three-dimensional image of the sound environment, so that all sound sources can be determined locally. At Only two microphones can be used determine two-dimensional image of the sound environment. at The use of more than three microphones can Accuracy in determining the location of sound sources increased be, but also require more computing effort is.
- the speech analysis device 60 is a Comparison of parameters of electrical Microphone output signals with in the memory unit 65 stored language parameters possible.
- the Signal processing unit 50 depending on the Comparative result for everyone in the sound environment detected sound source a value that the Probability indicates with which the respective Sound source was recognized as a speech source. The Sound source with the highest probability value then identified as a language source.
- phase delays with the microphones 5, 10, 15 connected phase delay elements 30, 35, 45 so be set at the location of the as the language source identified sound source for the beat signal Maximum reception results.
- the other sound sources will be thus essentially hidden as sources of interference.
- the appropriate adjustment of the phase delays can be done by the signal processing unit 50 also take place automatically, so that an adaptation of the phase shifts with the microphones 5, 10, 15 connected Phase delay elements 30, 35, 45 to a changing Location of the sound source identified as the language source is possible, so that despite a relative movement between the Voice source 55 and the telecommunications terminal 1 or the microphones 5, 10, 15 at the location of the speech source 55 for the beat signal maintains a receive maximum becomes.
- Sound sources can also be considered a sound source by the user Language source can be specified. For example, this is then advantageous if the telecommunications terminal 1 in a Car radio is integrated and as both the language source Driver as well as a passenger can be considered. It can then by changing the phase delays accordingly the one connected to the microphones 5, 10, 15 Phase delay elements 30, 35, 45 the driver or a Passenger can be selected as the language source 55, so that for the location of the selected language source is a maximum reception for the beat signal is set.
- the invention is not based on a telecommunications terminal 1 limited, but for all speech input electrical devices applicable. For example also be devices that have voice control.
- the speech processing unit 70 serves Evaluation and initiation of voice commands. Since it is at the evaluation of voice commands interference-free reception arrives separation of useful and interference signals according to the invention detection of the voice commands as error-free as possible without special mechanical aids such as Directional microphones or special filter algorithms for Elimination of the interference signals are required.
- the voice-input capable electrical device 1 When executing the voice-input capable electrical device 1 as a telecommunications terminal, it is due to the adaptive tracking of the reception maximum for the Superposition signal for a relative movement between the Telecommunications terminal 1 and the language source 55 not required that the telecommunications terminal 1 fixed is arranged. Therefore, the invention is also applicable to radio devices, Mobile phones, cordless phones and the like are applicable. The same applies to mobile voice-input electrical Devices with voice control. Electric voice input Devices with voice control can, for example, car radios, Personal computers, and the like, however wired or wireless telecommunications terminals.
- Signal processing unit 50 can be, for example, a digital one Signal processor are used.
- the inventive method and the inventive electrical equipment can be used for general optimization of receiving any acoustic signals, so that no limitation to electrical input Equipment is required.
- a speech analysis is then also not mandatory.
- To select a sound source as Useful sound sources are then appropriate criteria to be selected by the signal processing unit 50 are to be taken into account accordingly.
- Can also be provided be on an input unit by a user To have the sound source selected as the useful sound source.
- the not be selected as a useful sound source then faded out using suitable phase delays.
- the phase delays from the Signal processing unit 50 set so that a adaptive tracking of reception sensitivity in Dependence of the location of the useful sound source takes place, whereby the noise sources adaptive depending on their location be hidden.
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- Health & Medical Sciences (AREA)
- General Health & Medical Sciences (AREA)
- Otolaryngology (AREA)
- Physics & Mathematics (AREA)
- Engineering & Computer Science (AREA)
- Acoustics & Sound (AREA)
- Signal Processing (AREA)
- Circuit For Audible Band Transducer (AREA)
Description
Claims (9)
- Verfahren zur Optimierung des Empfangs akustischer Signale, wobei von mindestens zwei Mikrofonen (5, 10, 15) empfangene akustische Signale in elektrische Ausgangssignale umgewandelt werden, wobei von den elektrischen Ausgangssignalen abgeleitete Signale überlagert werden und wobei mindestens ein elektrisches Signal vor der Überlagerung in seiner Phase verzögert wird, dadurch gekennzeichnet, daß von den elektrischen Ausgangssignalen Sprachparameter abgeleitet werden, daß die Sprachparameter mit vorgegebenen Sprachparametern verglichen werden und daß mit einem in Abhängigkeit des Vergleichsergebnisses ermittelten Wahrscheinlichkeitswert mindestens eine Schallquelle (55) als Sprachquelle identifiziert wird, daß eine als Sprachquelle identifizierte Schallquelle (55) ausgewählt wird und daß die Phase des mindestens einen elektrischen Signals vor der Überlagerung in Abhängigkeit des Ortes der angeschlossenen Mikrofone (5, 10, 15) so verzögert wird, daß sich an der Stelle der ausgewählten Sprachquelle für das Überlagerungssignal ein Empfangsmaximum ergibt.
- Verfahren nach Anspruch 1, dadurch gekennzeichnet, daß die Schallquelle (55) mit dem höchsten Wahrscheinlichkeitswert als Sprachquelle ausgewählt wird.
- Verfahren nach Anspruch 1, dadurch gekennzeichnet, daß eine Schallquelle (55) durch Benutzervorgabe als Sprachquelle vorgegeben wird.
- Verfahren nach Anspruch 1, 2 oder 3, dadurch gekennzeichnet, daß in Abhängigkeit der Amplituden der elektrischen Ausgangssignale Ortskoordinaten mindestens einer Schallquelle (55) ermittelt werden.
- Elektrisches Gerät (1), insbesondere Telekommunikationsendgerät, wobei mindestens zwei Mikrofone (5, 10, 15) an das elektrische Gerät (1) anschließbar sind, die empfangene akustische Signale in elektrische Ausgangssignale umwandeln, wobei mindestens ein Mittel (20, 25), insbesondere ein Additionsglied, vorgesehen ist, das von den elektrischen Ausgangssignalen angeschlossener Mikrophone (5, 10, 15) abgeleitete Signale überlagert, und wobei mindestens ein Phasenverzögerungsglied (30, 35, 40, 45) vorgesehen ist, das ein elektrisches Signal vor der Überlagerung in seiner Phase verzögert, dadurch gekennzeichnet, daß eine Sprachanalysevorrichtung (60) vorgesehen ist, die einen Vergleich von Parametern der elektrischen Signale mit in einer zugeordneten Speichereinheit (65) abgelegten Sprachparametern durchführt und mit einem in Abhängigkeit des Vergleichsergebnisses ermittelten Wahrscheinlichkeitswert mindestens eine Schallquelle (55) als Sprachquelle identifiziert, wobei eine Auswahl einer als Sprachquelle identifizierten Schallquelle (55) erfolgt, und daß eine Signalverarbeitungseinheit (50) vorgesehen ist, die der Sprachanalysevorrichtung (60) zugeordnet ist und die die Phasenverzögerung des mindestens einen Phasenverzögerungsgliedes (30, 35, 40, 45) in Abhängigkeit des Ortes der angeschlossenen Mikrofone (5, 10, 15) so einstellt, daß sich an der Stelle der ausgewählten Sprachquelle für das Überlagerungssignal ein Empfangsmaximum ergibt.
- Elektrisches Gerät (1) nach Anspruch 5, dadurch gekennzeichnet, daß verschiedene Phasenverzögerungen des mindestens einen Phasenverzögerungsgliedes (30, 35, 40, 45) einstellbar sind.
- Elektrisches Gerät (1) nach Anspruch 5 oder 6, dadurch gekennzeichnet, daß eine Signalverarbeitungseinheit (50) vorgesehen ist, der die elektrischen Signale der Mikrophone (5, 10, 15) zugeführt sind und die in Abhängigkeit der Amplituden der elektrischen Signale Ortskoordinaten mindestens einer Schallquelle (55) ermittelt.
- Elektrisches Gerät (1) nach Anspruch 5, 6 oder 7, dadurch gekennzeichnet, daß die Signalverarbeitungseinheit (50) die Schallquelle (55) mit dem höchsten Wahrscheinlichkeitswert als Sprachquelle auswählt.
- Elektrisches Gerät (1) nach einem der Ansprüche 5 bis 8, dadurch gekennzeichnet, daß die Auswahl einer Schallquelle (55) durch Benutzervorgabe erfolgt.
Applications Claiming Priority (3)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
DE19741596 | 1997-09-20 | ||
DE19741596A DE19741596A1 (de) | 1997-09-20 | 1997-09-20 | Verfahren zur Optimierung des Empfangs akustischer Signale und elektrisches Gerät |
PCT/EP1998/001537 WO1999016285A1 (de) | 1997-09-20 | 1998-03-17 | Verfahren zur optimierung des empfangs akustischer signale und elektrisches gerät |
Publications (2)
Publication Number | Publication Date |
---|---|
EP1016314A1 EP1016314A1 (de) | 2000-07-05 |
EP1016314B1 true EP1016314B1 (de) | 2002-06-19 |
Family
ID=7843098
Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
---|---|---|---|
EP98916975A Expired - Lifetime EP1016314B1 (de) | 1997-09-20 | 1998-03-17 | Verfahren zur optimierung des empfangs akustischer signale und elektrisches gerät |
Country Status (5)
Country | Link |
---|---|
US (1) | US6529869B1 (de) |
EP (1) | EP1016314B1 (de) |
JP (1) | JP2001517916A (de) |
DE (2) | DE19741596A1 (de) |
WO (1) | WO1999016285A1 (de) |
Families Citing this family (7)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
DE19854373B4 (de) | 1998-11-25 | 2005-02-24 | Robert Bosch Gmbh | Verfahren zur Steuerung der Empfindlichkeit eines Mikrofons |
DE19943875A1 (de) * | 1999-09-14 | 2001-03-15 | Thomson Brandt Gmbh | System zur Sprachsteuerung mit einem Mikrofonarray |
DE19943872A1 (de) * | 1999-09-14 | 2001-03-15 | Thomson Brandt Gmbh | Vorrichtung zur Anpassung der Richtcharakteristik von Mikrofonen für die Sprachsteuerung |
SI2436012T1 (sl) | 2009-05-27 | 2016-09-30 | Rolls-Royce Nuclear Field Services Inc. | Pregledovalno orodje za zgornji snop parnega generatorja |
DE112014003443B4 (de) * | 2013-07-26 | 2016-12-29 | Analog Devices, Inc. | Mikrophonkalibrierung |
US9685730B2 (en) | 2014-09-12 | 2017-06-20 | Steelcase Inc. | Floor power distribution system |
US9584910B2 (en) | 2014-12-17 | 2017-02-28 | Steelcase Inc. | Sound gathering system |
Family Cites Families (7)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US4279027A (en) | 1979-09-13 | 1981-07-14 | Honeywell Inc. | Acoustic sensor |
US4802227A (en) * | 1987-04-03 | 1989-01-31 | American Telephone And Telegraph Company | Noise reduction processing arrangement for microphone arrays |
US5737431A (en) * | 1995-03-07 | 1998-04-07 | Brown University Research Foundation | Methods and apparatus for source location estimation from microphone-array time-delay estimates |
US5828997A (en) * | 1995-06-07 | 1998-10-27 | Sensimetrics Corporation | Content analyzer mixing inverse-direction-probability-weighted noise to input signal |
DE19540795C2 (de) | 1995-11-02 | 2003-11-20 | Deutsche Telekom Ag | Verfahren zur Sprecherlokalisierung mit Hilfe eines Mikrofonarrays |
JP3522954B2 (ja) * | 1996-03-15 | 2004-04-26 | 株式会社東芝 | マイクロホンアレイ入力型音声認識装置及び方法 |
JP3541339B2 (ja) * | 1997-06-26 | 2004-07-07 | 富士通株式会社 | マイクロホンアレイ装置 |
-
1997
- 1997-09-20 DE DE19741596A patent/DE19741596A1/de not_active Withdrawn
-
1998
- 1998-03-17 JP JP2000513442A patent/JP2001517916A/ja active Pending
- 1998-03-17 DE DE59804536T patent/DE59804536D1/de not_active Expired - Lifetime
- 1998-03-17 WO PCT/EP1998/001537 patent/WO1999016285A1/de active IP Right Grant
- 1998-03-17 EP EP98916975A patent/EP1016314B1/de not_active Expired - Lifetime
- 1998-03-17 US US09/509,135 patent/US6529869B1/en not_active Expired - Lifetime
Also Published As
Publication number | Publication date |
---|---|
US6529869B1 (en) | 2003-03-04 |
EP1016314A1 (de) | 2000-07-05 |
DE59804536D1 (de) | 2002-07-25 |
WO1999016285A1 (de) | 1999-04-01 |
JP2001517916A (ja) | 2001-10-09 |
DE19741596A1 (de) | 1999-03-25 |
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