EP0285632B1 - Limitation active des vibrations - Google Patents

Limitation active des vibrations Download PDF

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Publication number
EP0285632B1
EP0285632B1 EP87906587A EP87906587A EP0285632B1 EP 0285632 B1 EP0285632 B1 EP 0285632B1 EP 87906587 A EP87906587 A EP 87906587A EP 87906587 A EP87906587 A EP 87906587A EP 0285632 B1 EP0285632 B1 EP 0285632B1
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Prior art keywords
filter
control system
vibration control
signal
active vibration
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English (en)
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EP0285632A1 (fr
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Stephen John Elliott
Philip Arthur Nelson
Ian Mcgregor Stothers
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Adaptive Control Ltd
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Adaptive Control Ltd
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Priority claimed from GB08624053A external-priority patent/GB2203016A/en
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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K11/00—Methods or devices for transmitting, conducting or directing sound in general; Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/16—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/175—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
    • G10K11/178—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound by electro-acoustically regenerating the original acoustic waves in anti-phase
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K2210/00—Details of active noise control [ANC] covered by G10K11/178 but not provided for in any of its subgroups
    • G10K2210/10—Applications
    • G10K2210/128—Vehicles
    • G10K2210/1282—Automobiles
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K2210/00—Details of active noise control [ANC] covered by G10K11/178 but not provided for in any of its subgroups
    • G10K2210/30—Means
    • G10K2210/301—Computational
    • G10K2210/3023—Estimation of noise, e.g. on error signals
    • G10K2210/30232—Transfer functions, e.g. impulse response
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K2210/00—Details of active noise control [ANC] covered by G10K11/178 but not provided for in any of its subgroups
    • G10K2210/30—Means
    • G10K2210/301—Computational
    • G10K2210/3032—Harmonics or sub-harmonics
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K2210/00—Details of active noise control [ANC] covered by G10K11/178 but not provided for in any of its subgroups
    • G10K2210/30—Means
    • G10K2210/301—Computational
    • G10K2210/3046—Multiple acoustic inputs, multiple acoustic outputs
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K2210/00—Details of active noise control [ANC] covered by G10K11/178 but not provided for in any of its subgroups
    • G10K2210/30—Means
    • G10K2210/301—Computational
    • G10K2210/3048—Pretraining, e.g. to identify transfer functions
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K2210/00—Details of active noise control [ANC] covered by G10K11/178 but not provided for in any of its subgroups
    • G10K2210/30—Means
    • G10K2210/301—Computational
    • G10K2210/3049—Random noise used, e.g. in model identification
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K2210/00—Details of active noise control [ANC] covered by G10K11/178 but not provided for in any of its subgroups
    • G10K2210/30—Means
    • G10K2210/321—Physical
    • G10K2210/3212—Actuator details, e.g. composition or microstructure

Definitions

  • the invention relates to active vibration control.
  • vibration includes sound or noise, and the invention is particularly concerned with active noise control.
  • an active vibration control system for reducing vibration generated by a primary source, is defined in Claim 1. It is characterised in that at least one reference signal containing selected harmonics of the said primary source vibration is supplied to means driving a plurality of secondary vibration sources, such that vibration energy detected by sensor means operable to sense the vibration field established by the primary and secondary sources is reduced.
  • control system is operable in accordance with an algorithm which adjusts the outputs from the secondary sources so as to substantially reduce a cost function on a time scale comparable with the delays associated with the propagation of vibration from the secondary sources to the sensor means.
  • the present invention is particularly concerned with an active noise reduction system which can control the sound throughout an enclosure of a car, or at one or a number of "quiet zones" within it, and which can quickly adapt to changes in the excitation of the sound field due to changes in, for example, engine load or speed.
  • a signal related to the engine crankshaft rotation rate for example a signal emitted by the engine ignition system, is used to generate a reference signal containing a number of sinusoids at harmonics (or subharmonics) of the engine crankshaft rotation frequency. These are known as engine order frequencies. These sinusoids may be obtained using a variety of methods outlined below. Rather than attempt to control all harmonics, only a selected set of engine order frequencies may, in accordance with the invention, be generated as the reference signal. For example, only the firing frequency (second engine order in a four cylinder car) and its second harmonic (fourth engine order) are used if the spectrum of the sound in the car is dominated by these components.
  • a signal containing all the engine order frequencies may be fed to a band pass filter which isolates only the particular frequency or frequencies exciting a particular resonance in the enclosure interior which could cause a "boom" to be excited.
  • the advantage of reducing the number of frequencies fed to the filter is that an adaptive filter having fewer coefficients than would otherwise be the case can be used. This makes implementation more efficient and allows faster adaption time. The faster adaption time is particularly important in automotive applications, for example, in which the active control system has to adapt sufficiently quickly to track changes in engine speed which may occur on a very short timescale.
  • reference signals may be obtained from transducers mounted on a road wheel hub or the suspension system of the car. Such reference signals would contain the harmonics of road wheel rotation or road noise. Transducers placed outside a vehicle may provide reference signals representative of wind noise. If the reference signals are periodic (deterministic), control at individual harmonics can be exercised as described hereinafter.
  • the secondary sources may be loudspeakers used as the low frequency drives of a car audio system.
  • a signal is obtained from the primary vibration source which contains components at all harmonics prevalent in the sound in a vehicle powered by an engine.
  • This signal is filtered so as only to leave the most important or dominant harmonics. Filtering is carried out by a filter whose centre frequency can be controlled by an external signal in such a manner that the critical filter frequencies have a constant ratio compared with the engine crankshaft rotation rate. This can be achieved by using, for example, charge coupled devices whose switching frequency is locked to the crankshaft rotation frequency, but can, also be implemented as a program running on a microprocessor, as described hereinafter.
  • a primary source signal rich in harmonics is filtered by a band pass filter, having a centre frequency fixed at that of a pronounced "boom" in the car enclosure and a characteristic such that the reference signal only contains the harmonic(s) which are particularly exciting the boom.
  • This may be extended such that the filter contains a number of resonances at a number of boom frequencies of the car, or even such that the filter has a frequency response which models the acoustic response of the car interior to the primary excitation.
  • the input signal to the filter may be a signal from the engine containing all important harmonics and may be in the form of a pulse train.
  • This method may be accomplished by using phase lock loops to generate sinusoidal signals, with frequencies bearing an integer relationship to a square wave signal from the engine, which frequencies are then added together to form the reference signal.
  • the signal derived from the engine can be used to control a number of tunable oscillators, each producing a sinusoid at a selected harmonic frequency.
  • ⁇ (n) is a unit sample sequence to initiate the oscillators formed by the difference equations above.
  • the second and fourth harmonic may be generated for example.
  • sample frequency (f s ) is derived from the counter frequency (f c ) by a frequency division circuit for example, so that the ratio f s /f c is exactly an integer number.
  • An alternative difference equation which may be used to implement the digital oscillator has the form of a series approximation to a trigonometric function, for example :-
  • the series approximation above can be used for y(n) in the range - ⁇ /2 ⁇ y(n) ⁇ /2 . For values of y(n) outside this range the symmetry properties of the cosine waveform are utilised, until y(n)> ⁇ .
  • the calculations of the coefficients used in the difference equations forming the digital oscillators, together with the difference equations themselves may be implemented on a dedicated processor, or may form part of the program which also implements a controller that generates the outputs used to drive the secondary sources from the reference signals described above.
  • the controller is designed to be adaptive so as to quickly track changes in engine speed and load.
  • the outputs of the secondary sources are adaptively controlled so that some measurable cost function is minimised.
  • This cost function would typically be the sum of the mean square outputs from a number of microphones in the enclosed space.
  • the controller can be implemented as a digital adaptive FIR filter, using the basic update algorithm described by S. Elliott and P. Nelson in “Electronics Letters", at pp. 979-981, 1985. A number of additions must be made to this basic algorithm, however, to enable it to work quickly and efficiently in this particular example.
  • the basic algorithm referred to hereinafter as the stochastic gradient algorithm, is presented below, in order to highlight the necessary alterations.
  • each of these coefficients should be adjusted at every sample according to the equation where ⁇ is a convergence coefficient, e l (n) is the sampled output from the l'th sensor and r l m (n) is a sequence formed by filtering the reference signal discussed above ( x(n), say) with a digital filter which models the response of the l'th sensor to excitation of the m'th secondary source.
  • ⁇ is a convergence coefficient
  • e l (n) is the sampled output from the l'th sensor
  • r l m (n) is a sequence formed by filtering the reference signal discussed above ( x(n), say) with a digital filter which models the response of the l'th sensor to excitation of the m'th secondary source.
  • the digital filters must model the relevant response over a range of frequencies, governed by the frequency range which the active system is attempting to control. It has been found that under certain circumstances a digital filter need only model the overall delay in the response to ensure the stability of an adaptive filter. It is more common, however, to have the digital filters incorporate a delay and then some reverberant response. This may be implemented using either digital FIR or IIR filters whose coefficients are adjusted adaptively during an initialisation phase, so as to accurately match the desired responses. It is also possible to continue this initial adaption process during the operation of the active control system by feeding training signals to each secondary source which are suitably uncorrelated with each other and with the primary excitation.
  • this change may be detected with mechanical transducers and the information used to switch between a cariety of filters modelling the response of the enclosure under a variety of conditions.
  • a number of other adaptive algorithms may also be implemented to adjust the coefficients of the digital filters in the controller driving the secondary sources. These alternative algorithms are best described in matrix form.
  • y m (n) may be obtained by passing this reference signal through a digital filter whose i'th coefficient is w mi (n) at the n'th sample, so that
  • the sampled output from the l'th error sensor, e l (n) is equal to the sum of the contributions from the primary source, d l (n), and each of the secondary sources.
  • an enclosure 10 which is the interior of the passenger or driver compartment of an internal combustion engine driven vehicle, in this example, a motor car 100, is represented schematically together with an active sound control system 1 according to the invention.
  • the system 1 employs two secondary sound sources 11, comprising two low frequency loudspeakers of a stereo audio system fitted to the car, and three acoustic sensors, comprising microphones 12.
  • the loudspeakers 11 are driven by a controller circuit 13 which comprises a pair of adaptive filters 14.
  • Each adaptive filter 14 drives a respective one of the loudspeakers 11 with an output signal 3 which the filter 14 produces as a result of its action on a reference signal 4 supplied thereto by a reference signal generator 15.
  • the reference signal 4 is generated by the generator 15 from an input signal 16 which is periodic at the crankshaft rotation rate of the internal combustion engine 2.
  • the signal generator 15 may comprise a tracking filter.
  • the purpose of the outputs from the loudspeakers 11 driven by the controller 13 is to reduce the sound vibration field established by the primary and secondary sources, experienced within the enclosure 10. Since the primary source (engine 2) of the noise to be reduced is periodic, the reference signal 4 generated by the generator 15 is, in accordance with the invention, arranged to contain one or more sinusoidal components at harmonics (or subharmonics) of the crankshaft rotation rate of the engine 2.
  • the adaptive filters 14 are adjusted automatically by output signals 5 from the sensor-microphones 12, corresponding adjustment being made simultaneously to the outputs of the loudspeakers 11 so as to substantially minimise a cost function on a time scale comparable with the delays associated with the propagation of sound vibrations from the loudspeakers 11 to the microphones 12.
  • the cost function may comprise the sum of the mean square outputs of the microphones 12.
  • the control system 1 does not employ a stored solution. Instead, it makes use of a plurality of closed loops, each loop comprising a microphone 12, the controller 13, and a loudspeaker 11, whereby signals from the microphone 12 are used to adapt the filters 14 controlling the loudspeaker 11, which has an influence on the output of the microphone as a result of acoustic response within the enclosure 10.
  • the loudspeakers 11 and the microphones 12 are distributed in the enclosure 10 in spaced relationship. The distribution, which varies from car to car, is adjusted to get substantial sound reductions throughout the enclosure 10.
  • system 1 employs as many closed loops as the number of sensors (12) multiplied by the number of secondary sources (11).
  • system 1 employs more sensors (12) than secondary sources (11), whereby a controlled reduction of primary source vibration is achieved.
  • sensors (12) employ more sensors (12) than secondary sources (11), whereby a controlled reduction of primary source vibration is achieved.
  • the reference signal 4 contains harmonics of the input signal 16, which signal is periodic at the engine crankshaft rotation rate
  • the reference signal 4 contains engine order frequencies.
  • the signal generator 15 is arranged to select engine order frequencies that ensure that the sound produced by the loudspeakers 11 is of the same frequency or frequencies as the sound produced in the enclosure 10 by the engine 2, even during changes in engine conditions such as load or speed.
  • the number of engine order frequencies in the reference signal 4 is restricted so that the adaptive filters 14 have a relatively small number of coefficients and can therefore adapt quickly.
  • the input signal 16 can be obtained from another moving part of the engine or part of the ignition circuitry; for example.
  • Figure 2(a) illustrates graphically the response of the reference signal generator 15 when in the form of a tracking filter. That is to say, a filter having a centre frequency so controlled that the filter output frequencies have a constant ratio to the dominant input frequencies, so that in Figure 2(a), the frequency f o is N x (engine crankshaft rotation rate), and the reference signal 4 ( Figure 1) contains only the first N harmonics of the engine rotation rate, where N is an integer.
  • the filter input signal is a voltage pulse train as represented by Figure 2(b), where 8T is the periodic time of the engine rotation rate, the first eight harmonics of the engine rotation rate are present in the reference signal 4.
  • the spectrum, by Fourier analysis, of the reference signal 4 is then illustrated by Figure 2(c), in which A is amplitude. With the response of Figure 2(a), only the first six harmonics would be usable.
  • the tracing filter comprising the signal generator, can be in the form of charge coupled devices having a switching frequency locked to the engine crankshaft rotation rate.
  • FIG. 3 An alternative form of reference signal generator, which employs a plurality of tracking band pass filters is illustrated by Figure 3 in which the input signal 16, a square wave at, for example, 128 times the engine drive shaft rotation rate, is divided first by 32 and then by 2. Division by 32 is achieved by a divider 6 which produces a square wave signal 7 at four times the engine crankshaft rotation rate, which signal is supplied to a bandpass filter 17. The filter 17 has a centre frequency f4 which is arranged to track the fundamental frequency of the square wave signal 7 supplied thereto. The further division by 2 is achieved by a divider 8 which produces a square wave signal 9 at twice the engine drive shaft rotation rate.
  • the signal 9 is then supplied to a bandpass filter 18 having a centre frequency f2 and which is arranged to track the fundamental frequency of the square wave signal 9 supplied thereto.
  • the bandpass filters 17 and 18 produce respectively sinusoidal output signals 7 a , 9 a at f2 and f4 which are linearly summed at an adder 19 to produce the required reference signal 4.
  • reference signal generator may comprise a fixed frequency filtering circuit which selects harmonics and/or subharmonics from an input signal rich in the harmonics of the engine crankshaft rotation rate or firing rate.
  • the filtering circuit may comprise a bandpass filter having a centre frequency fixed at the frequency of a pronounced resonance excited in the enclosure 10 ( Figure 1) by the engine 2 or other primary source of vibration.
  • a bandpass filter may be arranged to have a response that models the acoustic response of a motor car passenger compartment to the engine.
  • a further form of reference signal generator may comprise a plurality of phase lock loops used to generate sinusoidal signals having respective frequencies with integer relationships to a square wave input signal from the engine 2 or other primary vibration source. The sinusoidal signals may then be added together to form the required reference signal. Thus a reference signal 4 comprising specific harmonics and/or harmonics locked to the primary source fundamental, such as engine crankshaft rotation rate, is generated.
  • FIG. 4 An alternative generator for such a reference signal is illustrated in Figure 4 in which a square wave 20 at the primary source fundamental is used to control a plurality of tunable digital oscillators 25, 26 each producing a sinusoidal signal at a chosen harmonic or subharmonic frequency to be added at the adder 19 which produce the reference signal 4 by simple addition of the sinusoidal signals.
  • the square wave signal 20 which is at an engine crankshaft rotation rate, is supplied to a bistable circuit 21 which divides the rate signal by two and thereby produces a pulse train signal 20 a in which the duration of each pulse is equal to the prevailing periodic time of the square wave signal 20.
  • This periodic time is then measured by a counter 22 which is enabled throughout the duration of each positive pulse from the bistable circuit 21 and counts clock pulses supplied by a clock pulse generator 23.
  • the clock pulses are generated at a fixed, suitably high rate f c .
  • the contents of the counter 22 are read at the end of each positive pulse from the bistable circuit 21 by a trigonometric function generator 24.
  • the digital sinusoidal outputs from the two oscillators 25 and 26 are superposed by a digital adder 191 which supplies the reference signal 4 as a digital signal.
  • the trigonometric function generator 24, oscillators 25 and 26 and adder 191 can be implemented by a microprocessor with a suitable program.
  • Figure 5 represents in block form an active sound control system 30 for reducing the level of engine generated noise in the passenger compartment of a motor car.
  • the car is provided with an ignition circuit including a low tension coil 31 from which a voltage signal 32 at the firing rate of the engine is taken and supplied to a waveform shaper 33 which in response thereto produces a pulse train at the engine firing rate.
  • the engine firing rate is twice the engine crankshaft rotation rate f o .
  • the shaper 33 provides a signal having a fundamental frequency which is a single harmonic, (2f0), of the crankshaft rate.
  • a reference signal generator is provided in the form of a proprietory tracking filter 34, manufactured by Bruel and Kjaer under type number 1623.
  • the tracking filter 34 receives the output of the shaper 33 as an input signal and as a trigger signal and produces a sinusoidal output signal at the selected harmonic 2f o .
  • This sinusoidal signal is sampled with an analog to digital converter 35 to produce a reference sequence x(n) of digitised samples which are supplied as data to a processor and memory unit 36.
  • the loudspeakers 371 and 372 are mounted within the motor car passenger compartment 10 (not shown in Figure 5) are two loudspeakers 371 and 372, which are in positions normally used for car stereo reproduction.
  • the loudspeakers 371, 372 are driven by a multiplexer 38 through respective low pass filters 39 and output amplifiers 40.
  • the filters 39 have a cut off frequency of 460 Hz and are provided to prevent aliasing.
  • the multiplexer 38 which contains sample and hold circuits for each output, is controlled by the processor and memory unit 36, through a control line 55, and receives a single input signal 57 from a digital to analog converter 41.
  • the purpose of the loudspeakers 371 and 372 is to generate, in the passenger compartment, audio waves that will cancel those set up directly by mechanical transmission from the engine to the compartment.
  • the digital to analog converter 41 is supplied by the processor and memory unit 36 with output data 58 which consists of two interleaved sequences of digitised samples y1(n) and y2(n).
  • the data 58 is converted by the converter 41 into interleaved sequences of analog samples and separated into respective sequences by the multiplexer 38 for application to the low pass filters 39.
  • the loudspeaker 371 is driven by the sequence y1(n) and the loudspeaker 372 is driven by the sequence y2(n).
  • each sequence of data 58 is represented by the expression y m (n), so that in this example m may be 1 or 2.
  • error signals are picked up from the passenger compartment and utilised by the processor and memory unit 36.
  • Acoustic error signals if present, are sensed by four microphones 421, 422, 423 and 424, which are placed respectively either side of a driver headrest and a passenger headrest, there being only two seats in the compartment in the present example.
  • the electrical outputs from the microphones 421 etc. are respectively amplified by amplifiers 43 and passed through low pass filters 44 to a four-input multiplexer 45 which supplies a single analog output to an analog to digital converter 46.
  • the filters 44 are provided to prevent aliasing and have a cut-off frequency of 460 Hz.
  • the multiplexer 45 is controlled by the processor unit 36 by way of control line 56.
  • the multiplexer 45 and the converter 46 convert the four filtered microphone outputs into a data stream 59 comprising four interleaved sequences of digitised samples e1(n), e2(n), e3(n) and e4(n), which correspond respectively to the filtered outputs of the microphones 421, 422, 423 and 424.
  • each sequence is represented by e (n) so that in this example l may be 1, 2, 3 or 4.
  • the processor and memory unit 36 receives a square wave signal 60 at 1.2 kilohertz from a sample rate oscillator 47 which determines the rate at which the converters 35, 41 and 46 convert samples and the frame duration of processing carried out by the unit 36.
  • the unit 36 completes its processing frame within 833 milliseconds.
  • a crystal clock oscillator 61 with a frequency of 10 Megahertz is included in the unit 36.
  • :- y1(n) w10x(n) + w11x(n - 1)
  • y2(n) w20x(n) + w21x(n - 1).
  • the filtered reference signal r lm is a sequence formed by filtering the reference signal x(n) with a filter that models the effect of the acoustic coupling between the m th loudspeaker and the l th microphone.
  • the unit 36 simulates this filtering as digital FIR (Finite Impulse Response) filtering. Coefficients for the digital FIR filtering are adjusted adaptively during an initialisation program in which a white noise generator 48 is energised.
  • a white noise signal is generated by the generator 48, which is then filtered by a low pass filter 49 to prevent aliasing, the filter 49 having a cut-off frequency of 460 Hz.
  • the signal is subsequently sampled and converted by an analog to digital converter 50.
  • the unit 36 performs a 35 coefficient FIR modelling of the impulse response between the m th loudspeaker and the l th microphone at the j th sample. Such modelling is described in "Adaptive Signal Processing" by B. Widrow and S.D. Stearns, published in 1985 by Prentice Hall.
  • the filtered reference sequence is then given by :-
  • the operation of the unit 36 is such that, having obtained the error samples e l (n) and the filtered reference signal r lm (n), each adaptive filter coefficient w mi for each output y m (n) is updated by a quantity proportional to the sum of the computed products of e l (n) and r lm (n - i) in accordance with the equation ;-
  • the new set of adaptive coefficients w mi is then stored and used to filter the next sample of the reference signal, x(n + 1).
  • the unit 36 includes RAM for temporary storage and computation, and EPROM for program storage. Calculated coefficients w mi and c lmj , and reference sequences r lm (n) are held in RAM. The convergence coefficient is entered at a set of manually operable switches (not shown).
  • the unit 36 includes a Texas Instruments TMS 32010 microprocessor.
  • the input signal rate from the ignition circuit, including the low tension coil 31, is 100 Hz to 200 Hz, and the waveform shaper 33 is a monostable circuit triggered by the leading edge of the input signal to produce pulses of a constant width which is small relative to the sample period set by the sample rate of 1.2 kilohertz.
  • the low pass filters 39, 44 and 49 are active filter modules supplied by Kemo Limited under No. 1431/L.
  • a family of algorithms which are alternatives to the SNM algorithm uses a bank of adaptive digital filters working in parallel for each secondary source.
  • Each individual filter is fed by a reference signal containing a subset of the harmonics or subharmonics to be controlled.
  • Figure 6 of the accompanying drawings shows two parallel FIR filters 70 each fed by pure tone reference signals 71, at the second and fourth engine order frequencies in this case. The outputs 72 of these filters are added together by an adder 73 to form an output 74 to the secondary source.
  • Each of the parallel filters 70 may be updated by any of the algorithms discussed above.
  • the said stochastic gradient algorithm may be modified so that :- where w Imi is the i'th coefficient of the FIR filter fed from the I'th harmonic of the engine, driving the m'th secondary source.
  • each harmonic frequency is controlled independently and the convergence of an harmonic does not couple with the convergence of any other harmonic, as is the case when a 2I coefficient filter is used to filter I harmonics simultaneously.
  • I filters each with 2 coefficients could be used to filter I harmonics individually, and their response combined afterwards.
  • the disadvantage of this algorithm is that a filtered reference signal needs to be generated for each source (m), sensor (l) and harmonic (I), to give each r lmI (n).
  • Another approach to controlling a number of harmonics is to take the Fourier transform of each of the error signals and to update a set of coefficients controlling each harmonic of each secondary output independently. The outputs of each of these filters are then combined together, for each secondary source, via an inverse Fourier transform, to generate the output waveform for this source, as indicated in Figure 7.
  • the complex value of the 'th error signal will be given by where A l is the value of E l with no active control, w m is the complex amplitude of the voltage to the m'th secondary source and c lm is the complex transfer function between the l'th sensor and m'th source at the frequency of the harmonic of interest.
  • the algorithm also reduces to the iterative matrix algorithm described in White and Cooper (1984, Applied Acoustics 17, 99-109. "An adaptive controller for multivariable active control”). See also U.K. Patent Specification No. 2,122,052A.
  • Figure 9 illustrates application of the invention to non-noise, i.e. mechanical, vibration control.
  • Figure 9 comprises a modification of the Figure 1 arrangement, wherein the microphone sensors have been replaced by accelerometers 90 and the loudspeakers by mechanical vibrators 91.
  • the accelerometers 90 and vibrators 91 are mounted on surface portions of the enclosure 10.
  • a sensing combination of microphones 12 and accelerometers 90 is used, and/or a source combination of loudspeakers 11 and vibrators 91.

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
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Abstract

Afin de réduire le bruit à l'intérieur du compartiment des passagers d'un véhicule à moteur, on y dispose deux hauts-parleurs (371, 372) qui sont attaqués par des signaux obtenus à partir d'un signal de référence x(n) par filtrage adaptatif effectué par une unité composée d'un microprocesseur programmé et d'une mémoire (36) qui adapte le filtrage en fonction des signaux d'erreur (el(n) provenant de quatre microphones (421, 422, 423 et 424) répartis dans le compartiment des passagers. Des coefficients de filtrage de référence sont initialement déterminés par analyse des réponses d'un nombre fini d'impulsions, lorsqu'il se produit un couplage acoustique du bruit blanc entre les hauts-parleurs (37) et les microphones (42), un générateur de bruit blanc (48) étant couplé à l'unité (36). Le signal de référence x(n) est limité à un ou plusieurs harmoniques ou sous-harmoniques sélectionnés de la fréquence de bruit fondamentales par un filtre (34) qui recherche la fréquence sélectionnée. La fréquence sélectionnée peut être obtenue à partir d'une bobine (31) se trouvant dans le circuit d'allumage du véhicule.

Claims (22)

  1. Un système actif de limitation de vibration destiné à réduire la vibration qui est produite par une source primaire de vibration possédant des fréquences harmoniques dominantes qui peuvent changer rapidement, comprenant des moyens de traitement (36); des premiers moyens de filtrage passe-bas (39) connectés à une sortie des moyens de traitement (36) et ayant une fréquence de coupure fixe; des moyens de référence (31, 33, 34, 35) qui sont destinés à appliquer aux moyens de traitement (36) au moins un signal de référence représentant au moins un harmonique sélectionné de la source de vibration primaire; les moyens de traitement (36) générant au moins un signal d'attaque en utilisant le signal de référence précité, et émettant le signal d'attaque ou chacun d'eux vers un ensemble de sources de vibration secondaires (37), par l'intermédiaire des premiers moyens de filtrage passe-bas (39); des seconds moyens de filtrage passe-bas (44), connectés à une entrée des moyens de traitement (36), et ayant une fréquence de coupure fixe; des moyens détecteurs (42) placés à un ou plusieurs emplacements, et fonctionnant de façon à détecter un champ de vibration qui est établi à l'un au moins de ces emplacements par les sources primaire et secondaire (37), et à émettre au moins un signal d'erreur vers les moyens de traitement (36), par l'intermédiaire des seconds moyens de filtrage passe-bas (44); un oscillateur de cadence d'échantillonnage (47) qui applique aux moyens de traitement un signal de cadence d'échantillonnage constante, de façon que le signal de référence et le signal d'erreur, ou chaque signal d'erreur, soient échantillonnés à une cadence constante; les moyens de traitement (36) comprenant des moyens de filtrage à réponse adaptative (14, 70) ayant des premiers coefficients de filtre pour modéliser la réponse de retard et de réverbération des moyens détecteurs (42) à au moins un signal de sortie des sources de vibration secondaires (37) sur une gamme de fréquence étendue; les moyens de filtrage à réponse adaptative (14, 70) fonctionnant de manière à déterminer des seconds coefficients de filtre sous la dépendance du signal d'erreur ou de chaque signal d'erreur, et à ajuster le signal d'attaque ou chaque signal d'attaque en utilisant les premiers et seconds coefficients de filtre, pour réduire le champ de vibration que détectent les moyens détecteurs (42).
  2. Un système de limitation active de vibration selon la revendication 1, dans lequel les moyens détecteurs comprennent un certain nombre de détecteurs, et ce nombre de détecteurs est supérieur au nombre de sources de vibration secondaire.
  3. Un système de limitation active de vibration selon la revendication 1 ou la revendication 2, dans lequel les moyens de référence (31, 33, 34, 35) appliquent aux moyens de traitement (36) un signal de référence représentant au moins deux harmoniques de la source de vibration primaire.
  4. Un système de limitation active de vibration selon l'une quelconque des revendications précédentes, dans lequel les moyens de filtrage à réponse adaptative comprennent un réseau de filtres ayant chacun 35 premiers coefficients (Clmj) qui modélisent la réponse des moyens détecteurs (42) à au moins un signal de sortie des sources de vibration secondaires (37).
  5. Un système de limitation active de vibration selon l'une quelconque des revendications 1 à 3, dans lequel les moyens de filtrage à réponse adaptative (14, 70) comprennent un nombre I de filtres (70) ayant chacun deux seconds coefficients (wmi), en désignant par I le nombre d'harmoniques dans le signal de référence.
  6. Un système de limitation active de vibration selon la revendication 1 ou la revendication 2, dans lequel les moyens de filtrage à réponse adaptative (14, 70) comprennent des filtres (70) qui reçoivent un ensemble de signaux de référence (71), représentant chacun un seul harmonique, ces filtres (70) ayant leurs sorties (72) combinées pour former une sortie (74) des sources secondaires, et étant réglables indépendamment.
  7. Un système de limitation active de vibration selon l'une quelconque des revendications 1 à 3, dans lequel les moyens de filtrage à réponse adaptative (14, 70) comprennent un réseau de filtres et ils peuvent fonctionner de façon à prendre la transformée de Fourier du signal d'erreur, ou de chaque signal d'erreur, à actualiser un ensemble de seconds coefficients complexes (W k) pour le réseau de filtres qui commandent l'harmonique ou chaque harmonique du signal d'attaque, et à combiner les signaux de sortie du réseau de filtres, par l'intermédiaire d'une transformation de Fourier inverse, pour générer le signal d'attaque ou chaque signal d'attaque.
  8. Un système de limitation active de vibration selon l'une quelconque des revendications précédentes, dans lequel les moyens de référence (31, 33, 34, 35) comprennent un filtre de signal de référence (34) qui est destiné à filtrer un signal d'entrée périodique dont la fréquence fondamentale est verrouillée sur une fréquence prédominante de la source de vibration primaire.
  9. Un système de limitation active de vibration selon la revendication 8, dans lequel le filtre de signal de référence (34) consiste en un filtre à poursuite.
  10. Un système de limitation active de vibration selon l'une quelconque des revendications 1 à 7, dans lequel les moyens de référence (31, 33, 34, 35) comprennent au moins un oscillateur accordable (25, 26) dont la fréquence est commandée par un signal (20) qui est représentatif de la fréquence fondamentale de la source de vibration primaire.
  11. Un système de limitation active de vibration selon l'une quelconque des revendications précédentes, dans lequel les moyens de filtrage à réponse adaptative (14, 70) comprennent un réseau de filtres ayant chacun un ensemble de premiers coefficients (Clmj) qui sont ajustés de façon adaptative pendant une phase d'initialisation du fonctionnement du système.
  12. Un système de limitation active de vibration selon l'une quelconque des revendications précédentes, dans lequel les moyens de filtrage à réponse adaptative (14, 70) comprennent un réseau de filtres ayant chacun un ensemble de premiers coefficients (Clmj), qui sont ajustés de façon adaptative pendant le fonctionnement du système, par l'application à chaque source secondaire de signaux d'apprentissage couvrant une gamme de fréquence étendue, ces signaux étant avantageusement non corrélés entre eux et avec la source de vibration primaire.
  13. Un système de limitation active de vibration selon l'une quelconque des revendications précédentes, dans lequel les moyens de filtrage à réponse adaptative ajustent le signal d'attaque ou chaque signal d'attaque, conformément à un algorithme, de façon à minimiser pratiquement une fonction de coût sur une échelle de temps comparable aux retards qui sont associés à la propagation jusqu'aux moyens détecteurs (42) de la vibration qui provient des sources de vibration secondaires (37).
  14. Un système de limitation active de vibration selon la revendication 13, dans lequel les moyens de filtrage à réponse adaptative ajustent le signal d'attaque, ou chaque signal d'attaque, conformément à un algorithme de la forme :

    w ̲ (n+1) = w ̲ (n) - 2µ R ̲ ​ T (n) e ̲ (n)
    Figure imgb0058

    avec les notations suivantes :
       w(n+1) représente un vecteur de valeurs des seconds coefficients de filtre pour le (n+1)-ième échantillon;
       w(n) représente un vecteur de valeurs des seconds coefficients de filtre le n-ième échantillon;
       µ représente un facteur de convergence;
       R T(n) représente une matrice de signaux obtenus par filtrage du signal de référence, en utilisant les premiers coefficients de filtre; et
       e(n) représente un vecteur de valeurs des signaux d'erreur pour le n-ième échantillon.
  15. Un système de limitation active de vibration selon la revendication 13, dans lequel les moyens de filtrage à réponse adaptative ajustent le signal d'attaque, ou chaque signal d'attaque, conformément à un algorithme de la forme :

    w ̲ (n+1) = w ̲ (n) - 2µ Q ̲ ​ T ( n ̲ ) e(n)
    Figure imgb0059

    avec les notations suivantes :
       w(n+1) représente un vecteur de valeurs des seconds coefficients de filtre pour le (n+1)-ième échantillon;
       w(n) représente un vecteur de valeurs pour les coefficients de filtre pour le n-ième échantillon;
       µ représente un facteur de convergence;
       Q T(n) représente une matrice modifiée de signaux de référence filtrés; et
       e(n) représente les valeurs complexes pour les signaux d'erreur pour le n-ième échantillon.
  16. Un système de limitation active de vibration selon la revendication 13, dans lequel les moyens de filtrage à réponse adaptative ajustent le signal d'attaque, ou chaque signal d'attaque, pour un seul harmonique, conformément à un algorithme de la forme :

    W ̲ ​ k+1 = W ̲ ​ k - 2µ C ̲ ​ H E ̲ ​ k
    Figure imgb0060

    avec les notations suivantes :
       W k+1 représente un vecteur de valeurs complexes de la réponse du filtre à la (k+1)-ième itération;
       W k représente un vecteur de valeurs complexes de la réponse du filtre à la k-ième itération;
       µ représente un facteur de convergence;
       E k représente un vecteur de valeurs complexes de la transformée de Fourier des signaux d'erreur à la k-ième itération;
       C représente la matrice de fonctions de transfert; et
       H désigne le conjugué complexe du vecteur ou de la matrice transposé.
  17. Un système de limitation active de vibration selon la revendication 13, dans lequel les moyens de filtrage à réponse adaptative ajustent le signal d'attaque ou chaque signal d'attaque conformément à un algorithme de la forme :

    W ̲ ​ k+1 = W ̲ ​ k - 2µ( C ̲ ​ H C ̲ )⁻¹ C ̲ ​ H E ̲ ​ k
    Figure imgb0061

    avec les notations suivantes :
       W k+1 représente un vecteur de valeurs complexes de la réponse du filtre à la (k+1)-ième itération;
       W k représente un vecteur de valeurs complexes de la réponse du filtre à la k-ième itération;
       µ représente un facteur de convergence;
       C représente la matrice de fonctions de transfert;
       E k représente un vecteur de valeurs complexes de la transformée de Fourier des signaux d'erreurs à la k-ième itération; et
       H désigne le conjugué complexe du vecteur ou de la matrice transposé.
  18. Un système de limitation active de vibration selon l'une quelconque des revendications précédentes, dans lequel les sources de vibration secondaires (37) sont constituées par des haut-parleurs, et dans lequel les moyens détecteurs (42) sont constitués par au moins un microphone.
  19. Un système de limitation active de vibration selon l'une quelconque des revendications 1 à 17, dans lequel les sources de vibration secondaires (37) sont constituées par des vibrateurs, et dans lequel les moyens détecteurs (42) sont constitués par des accéléromètres.
  20. Un système de limitation active de vibration selon la revendication 18 et la revendication 19, dans lequel les sources de vibration secondaires (37) comprennent une combinaison de haut-parleurs et de vibrateurs, et les moyens détecteurs (42) comprennent une combinaison de microphones et d'accéléromètres.
  21. Un véhicule propulsé par un moteur à combustion interne, comprenant le système de limitation active de vibration de l'une quelconque des revendications 1 à 20.
  22. Un véhicule propulsé par un moteur à combustion interne selon la revendication 21, rattachée à l'une quelconque des revendications 1 à 18 et 20, dans lequel les haut-parleurs sont constitués par des haut-parleurs d'un système audio stéréophonique qui est installé dans le véhicule.
EP87906587A 1986-10-07 1987-10-06 Limitation active des vibrations Expired - Lifetime EP0285632B1 (fr)

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GB08624053A GB2203016A (en) 1986-10-07 1986-10-07 Active sound control apparatus
GB8624053 1986-10-07
GB08630422A GB2201858A (en) 1986-10-07 1986-12-19 Active noise control
GB8630422 1986-12-19

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JP2890196B2 (ja) 1999-05-10
DE3786165T2 (de) 1993-10-14
DE3786165D1 (de) 1993-07-15
JPH01501344A (ja) 1989-05-11
WO1988002912A1 (fr) 1988-04-21

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