WO2002100287A2

WO2002100287A2 - Dental treatment noise suppression

Info

Publication number: WO2002100287A2
Application number: PCT/IL2002/000446
Authority: WO
Inventors: Uri Zilberman; Nissim Saban
Original assignee: Uri Zilberman; Nissim Saban
Priority date: 2001-06-10
Filing date: 2002-06-09
Publication date: 2002-12-19
Also published as: AU2002302962A1; IL143664A0; WO2002100287A3

Abstract

A system and method for dental treatment noise suppression including a first microphone (20) and a first speaker (24) arranged to be located adjacent a first ear of a patient, a second microphone and a second speaker arranged to be located adjacent a second ear of a patient. A first quiet zone generator (22) is operative to receive an input from the first microphone (20) and to provide canceling noise at the first speaker (24), thereby creating a quite zone adjacent the first ear of the patient and a second quiet zone generator operative to receive an input from the second microphone and to provide canceling noise at the second speaker, thereby creating a quite zone adjacent the second ear of the patient.

Description

DENTAL TREATMENT NOISE SUPPRESSION FIELD OF THE INVENTION The present invention relates to dentistry generally and more particularly to noise reduction during dental treatment.

BACKGROUND OF THE INVENTION Drilling noise and vibration during dental treatment is known to be a significant source of patient discomfort. Japanese Kokai No. 4-242638 describes a noise reduction device for dental cutters which senses dental treatment noise at a location near a patient, determines the amplitude and phase characteristics thereof and produces an interference sound having the same amplitude and phase characteristics as the sensed dental treatment noise. The device described in Japanese Kokai No. 4-242638 employs a digital filter whose amplitude and phase characteristics can be set.

U.S. Patent 5,692,056 describes a method and apparatus for intracranial noise suppression, which produces active cancellation of vibrational noise produced as by a dental drill.

U.S. Patents 5,305,387, 4,977,600, 4,644,581 and 4,455,675 relate to other types of active noise cancellation systems.

SUMMARY OF THE INVENTION The present invention seeks to provide highly effective dental treatment noise suppression. There is thus provided in accordance with a preferred embodiment of the present invention a system for dental treatment noise suppression including a first microphone and a first speaker arranged to be located adjacent a first ear of a patient, a second microphone and a second speaker arranged to be located adjacent a second ear of a patient, a first quiet zone generator operative to receive an input from the first microphone and to provide canceling noise at the first speaker, thereby creating a quite zone adjacent the first ear of the patient and a second quiet zone generator operative to receive an input from the second microphone and to provide canceling noise at the second speaker thereby creating a quite zone adjacent the second ear of the patient

In accordance with a preferred embodiment of the present invention, both the first and the second quiet zone generators include adaptive digital sound processors

Preferably the first and second microphones and the first and second speakers are located in a patient headrest forming part of or attached to a dental treatment chair

There is additionally provided in accordance with a preferred embodiment of the present invention a method for suppressing dental treatment noise including placing a first microphone and a first speaker adjacent a first ear of a patent, placing a second microphone and a second speaker adjacent a second ear of a patient. receiving an input from the first microphone and providing canceling noise at (he first speaker, thereby creating a quite zone adjacent the first ear of the patient, and receiving an input from the second microphone and providing canceling noise at the second speaker, thereby creating a quite zone adjacent the second ear of the patient

Preferably, the canceling noise is provided by employing an adaptive digital speech processing algorithm

There is also provided a dental chair comprising a noise suppression system of the type described hereinabove

BRIEF DESCRIPTION OF THE DRAWINGS The present invention will be understood and appreciated more fully from the following detailed description, taken in conjunction with the drawings m which Fig. 1 is a simplified pictorial illustration of a dental treatment noise suppression system and functionality constructed and operative in accordance with a preferred embodiment of the present invention;

Fig. 2 is a simplified functional block diagram of circuitry employed in one channel of the system and functionality shown in Fig. 1;

Fig. 3 is an electrical schematic of a preferred embodiment of one channel of the system of Fig. 2; and

Fig. 4 is a flow chart illustrating operation of the circuitry of Fig. 3.

DETAILED DESCRIPTION OF A PREFERRED EMBODIMENT

Reference is now made to Fig. 1, which is a simplified pictorial illustration of a dental treatment noise suppression system and functionality constructed and operative in accordance with a preferred embodiment of the present invention and to Fig. 2. which is a simplified functional block diagram of circuitry employed in one channel of the system and functionality shown in Fig. 1.

As seen in Figs. 1 and 2, there is provided a dental treatment noise suppression system and functionality, which may be integrally formed with a conventional dental chair 10, as illustrated in Fig. 1, or alternatively retrofitted thereto. In accordance with a preferred embodiment of the present invention, a headrest 12 of the dental chair is equipped with a pair of individual ear noise suppression assemblies 14. It is a particular feature of the present invention that the individual ear noise suppression assemblies 14, although located in the close vicinity of each ear of a patient, do not block sound from reaching the patient.

In accordance with a preferred embodiment of the present invention, a dental practitioner is supplied with a microphone 16, preferably associated with a wireless transmitter 18 for enabling instructions or other verbal communications from the dental practitioner to reach the patient without substantial suppression or distortion thereof. Preferably, the microphone 16 is a throat microphone, such as that commercially available from Earmark, Inc. of Hamden, Connecticut, U.S.A. As seen particularly in Fig. 2, each individual ear noise suppression assembly 14 preferably comprises a microphone 20, located in the vicinity of the patients, ear..which. senses the_so.und_in the-vicinity-of-the-patient's-ear- and -outputs-to- sound processing circuitry 22 Circuitry 22 is preferably located in the same general location as microphone 20 but alternatively may be remote therefrom and communicate with microphone 20 in a wireless manner

Circuitry 22 provides a noise canceling speaker driving output to a speaker 24 which is preferably located in the same general location as microphone 20 Alternatively speaker 24 may be located elsewhere As a further alternative, speaker 24 may comprise a plurality of mutually spatially distributed speakers

Turning to Fig 3 , it is seen that in sound processing circuitry 22, the outputs of microphones 16 and 20 are each preferably supplied to a low pass filter, respectively designated by reference numerals 26 and 28 The outputs of low pass filters 26 and 28 are converted to digital form in an A/D-D/A converter 30, preferably embodied in a TLV320AIC27 chip, commercially available from Texas Instruments Preferably, the bandwidth threshold of low pass filter 26 is 4 KHz, while that of low pass filter 28 is 7KHz The A/D-D/A converter 30 preferably provides a serial output to a digital signal processor 32, typically embodied in a chip belonging to the TMS320C5000 family of DSPs commercially available from Texas Instruments Digital signal processor 32 is operative to provide a digital noise cancellation signal which causes the sound sensed by microphone 20, adjacent each ear of the patient, to be identical to that sensed by microphone 16, which is preferably only the voice of the dental practitioner

According to an alternative embodiment of the present invention, microphone 16 may be eliminated In such a case, digital signal processor 32 is operative to provide a digital noise cancellation signal which causes the sound sensed by microphone 20 to be zero, insofar as possible Typically there is associated with digital signal processor 32, suitable conventional power and memory circuitry 34 In accordance with a preferred embodiment of the present invention, the digital noise cancellation signal output of digital signal processor 32 is supplied to speaker 24 via an analog output of A/D-D/A converter 30 which here functions as a D/A converter The sound processing circuitry 22 also typically compπses DC voltage blocking devices such as capacitors Cl, C2 and C3 It is further noted that one or both individual ear noise suppression assemblies 14 (Fig 1) may simultaneously be embodied in the circuitry shown m Fig 3

It is a particular feature of the present invention that the individual ear noise suppression assembly 14 in general, and the digital signal processor 32 in particular provide an adaptive filtering functionality, thus providing highly effective noise cancellation This functionality is provided typically by means of suitable software resident in digital signal processor 32, which expresses an algorithm, which is now described in detail with reference to the flow chart of Fig 4

Turning now to Fig 4, it is seen that following conventional hardware initialization in which parameters and sampling rates are set, various system coefficients are set These may include setting of operational parameters of A/D-D/A converter 30 and of circuitry 34

Following setting of system coefficients, a suitable amount of phase shift is set for the speech of the dental practitioner, which is picked up via microphone 16 This phase shift ensures that the speech of the dental practitioner is not cancelled by the system

An estimation of the sound transfer function operative in the environment of the system is then made, preferably by emitting white noise at speaker 24 and employing a least mean square algorithm The estimation function may be expressed as

Est = ∑H(ή*Xι(N-ή

where N is the magnitude of the vector H(ι),

H(ι) represents the sound transfer function, and

X\(n ) is the input signal from the microphone at the patient's ear and n = (N - l)

X is shifted with every sample so the oldest data is discarded and the newest data is entered at the first element of the vector

A difference function is now calculated, using the current sample which represents the difference between the real signal and the calculated signal The difference function preferably is expressed as follows

Thereafter the coefficients of the transfer function are updated in accordance -with the following:

Hn _* i (/) = k * Diff *

+ H„(z) where k is a constant that determines the speed of convergence. 5 This process is preferably carried out at least 100 times in order to obtain acceptable convergence.

At this stage, run time processing is begun. This process uses the function H„(i) calculated above.

Digital samples are acquired by microphones 16 and 20. 10 A current interim output OUT(n) is then calculated in accordance with the following equation:

OW(n) = W(l)* D(l -ϊ)

;=ι where W(i) are weighting coefficients employed in the least mean square algorithm; and 1 is the vector size of the weights W(i). 15 The calculation of D(i) is described hereinbelow.

The output to the speaker 24 delivered by the D/A-A/D converter 30 is combined with the input from microphone 16 as indicated by the following equation: OUTPUT = OUT(n) + Xτ(n) where X2 is the input from microphone 16. 20 The output is fed to the- D/A-A/D converter 30 and output to the speaker

24.

A filtered reference is then calculated as follows:

Df = ∑D(j)*H(i-N)

where N is the magnitude of the vector H(i). 25 H(i) represents the sound transfer function.

The above filtered reference is then employed together with the combined microphones' input and a convergence coefficient to calculate W(i) in accordance with the following equation: rV, + x(n) = J * X * Df + Wι(n) jυ wnere j is a convergence coemcient X is the input to the microphone 20 combined with the signal from microphone 16 as described hereinbelow.

The system output as heard by the patient is then estimated as follows:

S,,; = ∑OUT(i) *H(i -N)

The current element of the D vector is calculated as sst forth hereinbelow This vector includes the difference between the inputs of the microphones 16 and 20 and the predicted system output:

where X is an input from the digital sample of the sound picked up by microphones 16 and 20 as shown hereinbelow.

The vector D(l) is shifted so the new difference becomes the first element of the vector D(l) and the last difference is discarded .

The process described above is carried out at a sampling rate which preferably exceeds 14 KHz. The input X which is fed to the system algorithm is preferably calculated in the following manner:

X = Xι(n) - Xι(n - d) * r where X2 is the input from microphone 16; d is a delay factor; and r is a volume adjustment factor.

Coefficients d and r are preferably set so the input of microphone 16 is not be cancelled by the system and will reach the patient's ear.

It will be appreciated by persons skilled in the art that the present invention is not limited by that which is specifically described hereinabove and in the drawings. Rather the scope of the present invention includes both combinations and sub-combinations of various features described hereinabove and further developments and various thereof which are not in the prior art.

Claims

C L A I M S

1 A system for dental treatment noise suppression including a first microphone and a first speaker arranged to be located adjacent a first ear of a patient, a second microphone and a second speaker arranged to be located adjacent a second ear of a patient a first quiet zone generator operative to receive an input from said first microphone and to provide canceling noise at said first speaker, thereby creating a quite zone adjacent said first ear of said patient and a second quiet zone generator operative to receive an input from said second microphone and to provide canceling noise at said second speaker, thereby creating a quite zone adjacent said second ear of said patient

2 A system according to claim 1 and wherein both said first and said second quiet zone generators include an adaptive digital sound processor

3 A system according to claim 1 or claim 2 and wherein said first and second microphones and said first and second speakers are located in a patient headrest forming part o or attached to a dental treatment chair

4 A system according to any of the preceding claims and also comprising a third microphone arranged to be located adjacent a dental practitioner and wherein said first and second quite zone generators are operative in response to an output from a third microphone to allow speech of said dental practitioner to be heard by at least one of said first and second ears of said patient

5 A system according to any of claims 1 to 3 and wherein at least one of said first and second quite zone generators comprises sound processing circuitry which receives an input from at least one of said first and second microphones

6 A system according to any of the preceding claims and wherein said sound processing circuitry is located m the same general location as the microphone

7. A system according to any of the preceding claims and wherein said sound processing circuitry is located remotely from the microphone.

8. A system according to claim 5 and wherein said sound processing circuitry provides a noise canceling speaker driving output to at least one of said first and second speakers.

9. A system according to claim 1 and wherein said first and second speakers are each located in the same general location as a corresponding one of said first and second microphones.

10. A system according to claim 1 and wherein at least one of said first and second speakers comprises a plurality of mutually spatially distributed speakers.

1 1. A system according to claim 5 and wherein said sound processing circuitry comprises: at least one low pass filter receiving an output from at least one microphone adjacent an ear of a patient; an A/D converter which receives an input from said at least one low pass filter and provides a serial output; a digital signal processor operative to provide a digital noise cancellation signal which causes cancellation of sound sensed by said at least one microphone.

12. A system according to claim 5 and wherein said sound processing circuitry comprises: at least one low pass filter receiving an output from at least one microphone adjacent an ear of a patient; an A/D converter which receives an input from said at least one low pass filter and provides a serial output; a digital signal processor operative to provide a digital noise cancellation signal which causes cancellation of sound sensed by said at least one microphone .without-canceling speech.of a dental practitioner-picked-up by-another microphone.

13. A system according to claim 5, claim 11 or claim 12 and wherein said sound processing circuitry provides adaptive filtering functionality.

14. A system according to claim 13 and wherein said adaptive filtering functionality is provided by software resident in said sound processing circuitry which expresses an adaptive filtering algorithm.

15. A system according to claim 14 and wherein said software resident in said sound processing circuitry is operative to provide the following functionality: setting a suitable phase shift for speech of the dental practitioner, which is picked up via a microphone in order to ensure that the speech of the dental practitioner is not cancelled; estimating a sound transfer function operative in the environment of the system; calculating a difference function which represents the difference between sound actually picked up by a microphone adjacent said patient's ear and a calculated value thereof; updating the estimated sound transfer function using the difference function; run time processing of the sound transfer function; and delivering a sound canceling output to a speaker adjacent said patient's ear.

16. A system according to claim 15 and wherein said sound canceling output is combined with an input from said microphone which picks up the speech of the dental practitioner.

17. A system according to claim 4 and comprising functionality for applying a phase shift to speech of the dental practitioner, which is picked up via said third microphone, thereby to ensure that the speech of the dental practitioner is not cancelled by-the-system.-

18. A system according to claim 2 and wherein said adaptive digital sound processor provides an estimation of the sound transfer function operative in the environment of the system.

19. A system according to claim 18 and wherein said estimation is provided by emitting white noise at said speakers and employing a least mean square algorithm.

20. A system according to claim 18 or claim 19 and wherein said estimation is expressed as:

Est =∑H(i)* Xι(N-i) ι=l where N is the magnitude of the vector H(i);

H(i) represents the sound transfer function; and

X\(n) is the input signal from the microphone at the patient's ear and n = (N - i), and

X is shifted with every sample so the oldest data is discarded and the newest data is entered at the first element of the vector.

21. A system according to claim 18 and wherein said adaptive digital sound processor also calculates a difference function using a current sample which represents the difference between a real signal and a calculated signal.

22. A system according to claim 21 and wherein said difference function is expressed as: Dijf = Xι - Est

23. A system according to claim 21 or claim 22 and wherein said adaptive digital sound processor also updates coefficients of a transfer function in accordance with the following: Hi. ÷ ι( = k * Diff * Xι(i) + Hn(t) wnere K IS a constant mat determines the speed of convergence.

24 A system according to claim 23 and wherein said adaptive digital sound processor updates said coefficients of said transfer function at least 100 times in order to obtain acceptable convergence

25 A system according to claim 23 and wherein said adaptive digital sound processor also provides run time processing employing the function H_n+ι(ι)

26 A system according to claim 23 and wherein said adaptive digital sound processor also calculates a current interim output OUT(n) in accordance with the following equation

OUT(n) = Y_j W(ι)* D(l -ι)

; = 1 where W(ι) are weighting coefficients employed in the least mean square algorithm, and 1 is the vector size of the weights W(ι), and

D/ = ^ (,)*H(z-N) ι=l where N is the magnitude of the vector H(ι), and H(ι) represents sound the transfer function

27 A s)'stem according to claim 26 and wherein said adaptive digital sound processor also combines the output of said speaker with the output of said third microphone in accordance with the following equation

OUTPUT = OUT(n) + X2(n) where Xi is the input from said third microphone

28 A system according to claim 27 and wherein said adaptive digital sound processor also calculates W(ι) in accordance with the following equation

W_{l +} ι(n) = J * X * Df + W₁ n) — where- is a convergence coefficient, and" X is an input to said first or second microphones combined with an input from said third microphone

29 A system according to claim 28 and wherein said adaptive digital sound processor also calculates an estimated system output as heard by the patient as follows

30 A system according to claim 29 and wherein said adaptive digital sound processor also calculates a current element of the D vector is calculated as set forth hereinbelow

where X is an input from the digital sample of the sound picked up by at least one of said first and second microphones and by said third microphone

31 A system according to claim 30 and wherein said adaptive digital sound processor also calculates a shift of the vector D is shifted so the new difference becomes the first element of the vector D and the last difference is discarded

32 A system according to claim 2 and wherein said adaptive digital sound processor operates a sampling rate which exceeds 14 KHz

33 A system according to claim 31 and wherein said adaptive digital sound processor calculates X in the following manner

where X≥ is the input from said third microphone, d is a delay factor, and r is a volume adjustment factor, where d and r are set so the input of said third microphone is not canceled by the system and reaches the patient's ear

34 A method for suppressing dental treatment noise including placing a first microphone and a first speaker adjacent a first ear of a patient; placing a second microphone and a second speaker adjacent a second ear of a patient; receiving an input from said first microphone and providing canceling noise at said first speaker, thereby creating a quiet zone adjacent said first ear of said patient; and receiving an input from said second microphone and providing canceling noise at said second speaker, thereby creating a quiet zone adjacent said second ear of said patient.

35. A method according to claim 34 and wherein said canceling noise is provided by employing an adaptive digital speech processing algorithm.

36. A method according to claim 34 or claim 35 and wherein said first and second microphones and said first and second speakers are located in a patient headrest forming part of or attached to a dental treatment chair.

37. A method according to any of claims 34 to 36 and also employing a third microphone arranged to be located adjacent a dental practitioner to allow speech of said dental practitioner to be heard by at least one of said first and second ears of said patient.

38. A method according to any of claims 34 to 37 and wherein said providing canceling noise comprises: providing an output from at least one microphone adjacent an ear of a patient to at least one low pass filter; providing an output from said at least one low pass filter to an A/D converter which provides a serial output; providing said serial output to a digital signal processor which provides a digital noise cancellation signal which causes cancellation of sound sensed by said at least jonejmicrophone_

39. A method according to claim 38 and wherein said digital noise cancellation signal causes cancellation of sound sensed by said at least one microphone without canceling speech of a dental practitioner picked up by another microphone.

40. A method according to claim 38 and wherein said digital signal processor provides adaptive filtering functionality.

41. A method according to claim 40 and wherein said adaptive filtering functionality is provided by software resident in said digital signal processor which expresses an adaptive filtering algorithm.

42. A method according to claim 41 and wherein said software resident in said sound processing circuitry is operative to provide the following functionality: setting a suitable phase shift for speech of the dental practitioner, which is picked up via a microphone in order to ensure that the speech of the dental practitioner is not cancelled; estimating a sound transfer function operative in the environment of the method; calculating a difference function which represents the difference between sound actually picked up by a microphone adjacent said patient's ear and a calculated value thereof; updating the estimated sound transfer function using the difference function; run time processing of the sound transfer function; and delivering a sound canceling output to a speaker adjacent said patient's ear.

43. A method according to claim 42 and wherein said sound canceling output is combined with an input from said microphone which picks up the speech of the dental practitioner.

44. A method according to claim 42 or claim 43 and comprising functionality for applying said phase shift to speech of the dental practitioner thereby to ensure that the speech of the dental practitioner is not cancelled.

45. A method according to claim 35 and wherein said adaptive digital speech processing algorithm employs an estimation of the sound transfer function operative in the environment of the method.

46. A method according to claim 45 and wherein said estimation is provided by emitting while noise at said speakers and employing a least mean square algorithm.

47. A method according to claim 45 or claim 46 and wherein said estimation is expressed as:

Est =∑H(i)*Xι(N-i) ι=l where N is the magnitude of the vector H(i);

H(i) represents the sound transfer function; and

(N - i), and

48. A method according to claims 45 to 47 and wherein said adaptive digital speech processing algorithm also calculates a difference function using a current sample which represents the difference between a real signal and a calculated signal.

49. A method according to claim 48 and wherein said difference function is expressed as:

Diff = Xι ~ Est

50. A method according to claim 48 or claim 49 and wherein said adaptive digital speech processing algorithm also updates coefficients of a transfer function in accordance with the following:

Hn ÷ i (/) = * Diff * Xι(i) + H„(i) where k is a constant that determines the speed of convergence.

51. A method according to claim 50 and wherein said adaptive digital speech processing algorithm processor updates said coefficients of said transfer function at least 100 times in order to obtain acceptable convergence.

52. A method according to claim 50 and wherein said adaptive digital speech processing algorithm also provides run time processing employing the function H_n(i).

53. A method according to claim 50 and wherein said adaptive digital speech processing algorithm also calculates a current interim output OUT(n) in accordance with the following equation:

OUT(n) = ∑ W(i)* D(l -i)

where W(i) are weighting coefficients employed in the least mean square algorithm; 1 is the vector size of the weights W(i); and Df = ∑D(f)*H(j-N)

where N is the magnitude of the vector H(i).

H(i) represents the sound transfer function.

54. A method according to claim 53 and wherein said adaptive digital sound processor also combines the output of said speaker with the output of said third microphone in accordance with the following equation: OUTPUT = OUT(n) + X2()i) where X2 is the input from said third microphone.

55. A method according to claim 54 and wherein said adaptive digital sound processor also calculates W(i) in accordance with the following equation:

where J is a convergence coefficient, and X is an input to said first or second microphones combined with an input from said third microphone.

56. A method according to claim 55 and wherein said adaptive digital speech processing algorithm also calculates an estimated method output as heard by the patient as follows:

S«, = ∑OUT(i) * H(! - N)

57. A method according to claim 56 and wherein said adaptive digital speech processing algorithm also calculates a current element of the D vector is calculated as set forth hereinbelow:

where X is an input from the digital sample of the sound picked up by at least one of said first and second microphones and by said third microphone.

58. A method according to claim 57 and wherein said adaptive digital speech processing algorithm also calculates a shift of the vector D is shifted so the new difference becomes the first element of the vector D and the last difference is discarded .

59. A method according to claim 58 and wherein said adaptive digital speech processing algorithm calculates X in the following manner:

X = Xι(n) - X₂(n - d) * r where X is the input from said third microphone; d is a delay factor; and r is a volume adjustment factor, where d and r are set so the input of said third nticroph mejs not canceled _by_ the method.and_reaches-the-.patientls ear,-