WO1997011533A1 - A directional acoustic signal processor and method therefor - Google Patents
A directional acoustic signal processor and method therefor Download PDFInfo
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- WO1997011533A1 WO1997011533A1 PCT/US1996/014680 US9614680W WO9711533A1 WO 1997011533 A1 WO1997011533 A1 WO 1997011533A1 US 9614680 W US9614680 W US 9614680W WO 9711533 A1 WO9711533 A1 WO 9711533A1
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- 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
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04B—TRANSMISSION
- H04B3/00—Line transmission systems
- H04B3/02—Details
- H04B3/20—Reducing echo effects or singing; Opening or closing transmitting path; Conditioning for transmission in one direction or the other
-
- 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
- G10K11/18—Methods or devices for transmitting, conducting or directing sound
- G10K11/26—Sound-focusing or directing, e.g. scanning
- G10K11/34—Sound-focusing or directing, e.g. scanning using electrical steering of transducer arrays, e.g. beam steering
- G10K11/341—Circuits therefor
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04B—TRANSMISSION
- H04B15/00—Suppression or limitation of noise or interference
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M9/00—Arrangements for interconnection not involving centralised switching
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M9/00—Arrangements for interconnection not involving centralised switching
- H04M9/08—Two-way loud-speaking telephone systems with means for conditioning the signal, e.g. for suppressing echoes for one or both directions of traffic
- H04M9/082—Two-way loud-speaking telephone systems with means for conditioning the signal, e.g. for suppressing echoes for one or both directions of traffic using echo cancellers
-
- 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
- G10K2210/108—Communication systems, e.g. where useful sound is kept and noise is cancelled
-
- 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
- G10K2210/3028—Filtering, e.g. Kalman filters or special analogue or digital filters
-
- 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
- G10K2210/50—Miscellaneous
- G10K2210/505—Echo cancellation, e.g. multipath-, ghost- or reverberation-cancellation
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
- H04R25/40—Arrangements for obtaining a desired directivity characteristic
- H04R25/407—Circuits for combining signals of a plurality of transducers
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R29/00—Monitoring arrangements; Testing arrangements
- H04R29/004—Monitoring arrangements; Testing arrangements for microphones
- H04R29/005—Microphone arrays
- H04R29/006—Microphone matching
Definitions
- This invention relates to the field of microphone-array signal processing, and more particularly to a two stage processor for extracting one substantially pure sound signal from a mixture of such signals even in the presence of echoes and reverberations
- Microphone-array processing techniques related to separating sources by exploiting spatial separation of their sources are also well known and have been of interest for several decades
- nonlinear processing In each output stream some source direction of arrival, a "look direction " is assumed
- the microphone signals are delay ed to remove differences in time of flight from the look direction
- Signals from any direction other than tht look direction are thus misaligned in time
- the signal in the output stream is formed, in essence, by "gating" sound fragments from the microphones At any given instant, the output is chosen to be equal to one of the microphone signals
- These techniques exemplified by Kaiser and David ( I 960), b Mitchell et al (1971 ), and by Lyon (1983), perform best when the undesired sources consist predominantly of impulse trains, as is the case with human speech While these nonlinear techniques can be very computationally efficient and are of scientific interest as models of human cocktail-part) processing, they do not have practical or commercial significance because of their inherent inability to bring about full suppression of unwanted sources This inability
- gain-based One class of blind active-cancellation techniques may be called “gain-based” it is presumed that the waveform produced by each source is received by the microphones simultaneously, but with varying relative gams (Directional microphones must be employed to produce the required differences in gain )
- the mixtures used in the simulations were partially separated to begin with, i e , the crosstalk between the channels was weak ln practice, the microphone signals must be assumed to contain strong crosstalk unless the microphones are highly directional and the geometry of the sources is constrained
- the present invention employs a two-stage architecture Its two-stage architecture is substantially different from other two-stage architectures found in p ⁇ or art
- a two-stage signal processing architecture is employed in a Griffiths-Jim beamformer (Griffiths and Jim, 1982)
- the first stage of a Griffiths-Jim beamformer is delay-based two microphone signals are delayed to remove time-of-flight differences with respect to a given look direction, and in contrast with classical beamforming, these delayed microphone signals are subtracted to create a reference noise signal ln a separate channel, the delayed microphone signals are added, as in classical beamforming, to create a signal in which the desired source is enhanced relative to the noise
- the first stage of a Griffiths-Jim beamformer produces a reference noise signal and a signal that is predominantly desired source
- the noise reference is then employed in the second stage, using standard active noise- cancellation techniques, to improve the signal-to-noise ratio in the output
- the Griffiths-Jim beamformer suffers from the flaw that under reverberant conditions, the delay - based first stage cannot construct a reference noise signal devoid of the desired signal, whereas the second stage relies
- An audio signal processing system for processing acoustic waves from a plurality of sources, comprising a plurality of spaced apart transducer means for receiving acoustic waves from the plurality of sources, including echoes and reverberations thereof
- the transducer means generates a plurality of acoustic signals in response thereto
- Each of the plurality of transducer means receives acoustic waves from the plurality of sources including echoes and reverberations thereof, and generates one of the plurality of acoustic signals
- a first processing means receives the plurality of acoustic signals and generates a plurality of first processed acoustic signals in response thereto In the absence of echoes and reverberations of the acoustic waves from the plurality of sources, each of the first processed acoustic signals represent acoustic waves from only one different source
- a second processing means receives the plurality of first processed acoustic signals and generates a plurality
- Figure 1 is a schematic block diagram of an embodiment of an acoustic signal processor of the present invention, using two microphones
- Figure 2 is a schematic block diagram of an embodiment of the direct-signal separator portion, I e the first stage of the processor shown in Figure 1
- Figure 3 is a schematic block diagram of an embodiment of the crosstalk remover portion, i e . the second stage of the processor shown in Figure 1
- Figure 4 is an overview of the delay in the acoustic waves arriving at the direct signal separator portion of the signal processor of Figure 1 , and showing the separation of the signals
- Figure 5 is an overview of a portion of the crosstalk remover of the signal processor of Figure I showing the removal of the crosstalk from one of the signal channels
- Figure 6 is a detailed schematic block diagram of an embodiment of a direct-signal separator using three microphones
- FIG. 7 is a detailed schematic block diagram of an embodiment of a crosstalk remover suitable using three microphones Detailed Description of The Invention
- the present invention is a device that mimics the cocktail-party effect using a plurality of microphones with as many output audio channels, and a signal-processing module When situated in a complicated acoustic environment that contains multiple audio sources with arbitrary spectral characteristics, it supplies output audio signals, each of which contains sound from at most one of the original sources These separated audio signals can be used in a variety of applications, such as hearing aids or voice-activated devices
- Figure 1 is a schematic diagram of a signal separator processor of one embodiment of the present invention
- the signal separator processor of the present invention can be used with any number of microphones
- the signal separator processor receives signals from a first microphone 10 and a second microphone 12, spaced apart by about two centimeters
- the microphones 10 and 12 include transducers (not shown), their associated pre-amplifiers (not shown), and A/D converters 22 and 24 (shown in Figure 2)
- the microphones 10 and 12 in the preferred embodiment are omnidirectional microphones, each of which is capable of receiving acoustic wave signals from the environment and for generating a first and a second acoustic electrical signal 14 and 16 respectively
- the microphones 10 and 12 are either selected or calibrated to have matching sensitivity
- the use of matched omnidirectional microphones 10 and 12, instead of directional or other microphones leads to simplicity in the direct-signal separator 30, described below
- two Knowles EM-3046 omnidirectional microphones were used, with a separation of 2 centimeters
- the pair was mounted at least 25 centimeters from any large surface in order to preserve the omnidirectional nature of the microphones
- Matching was achieved by connecting the two microphone outputs to a stereo microphone preamplifier and adjusting the individual channel gains so that the preamplifier outputs were closely matched
- the preamplifier outputs were each digitally sampled at 22,050 samples per second, simultaneously These sampled electrical signals 14 and 16 are supplied to the direct signal separator 30 and to
- the direct-signal separator 30 employs information from a DOA estimator 20 which derives its estimate from the microphone signals
- DOA information could come from an source other than the microphone signals, such as direct input from a user via an input device
- the direct signal separator 30 generates a plurality of output signals 40 and 42
- the direct signal separator 30 generates as many output signals 40 and 42 as there are microphones 10 and 12 generating as many input signals 14 and 16 as are supplied to the direct signal separator 30
- each of the microphones 10 and 12 would detect acoustic waves from both sources
- each of the electrical signals 14 and 16, generated by the microphones 10 and 12 respectively contains components of sound from sources A and B
- the direct-signal separator 30 processes the signals 14 and 16 to generate the signals 40 and 42 respectively, such that in anechoic conditions (i e , the absence of echoes and reverberations), each of the signals 40 and 42 would be of an electrical signal representation of sound from only one source ln the absence of echoes and reverberations, the electrical signal 40 would be of sound only from source A, with electrical signal 42 being of sound only from source B, or vice versa
- the direct-signal separator 30 can bring about full separation of the sounds represented in signals 14 and 16
- the separation is only partial
- the output signals 40 and 42 of the direct signal separator 30 are supplied to the crosstalk remover 50
- the crosstalk remover 50 removes the crosstalk between the signals 40 and 42 to bring about fully separated signals 60 and 62 respectively
- the direct-signal separator 30 and the crosstalk remover 50 play complementary roles in the system 8
- the direct-signal separator 30 is able to bring about full separation of signals mixed in the absence of echoes and reverberation, but produces only partial separation when echoes and reverberation are present
- the crosstalk remover 50 when used alone is often able to bring about full separation of sources mixed in the presence of echoes and reverberation, but is most effective when given inputs 40 and 42 that are partially separated
- each output 60 and 62 of the crosstalk remover 50 contains the signal from only one sound source A or B
- these outputs 60 and 62 can be connected individually to post filters 70 and 72, respectively, to remove known frequency coloration produced by the direct signal separator 30 or the crosstalk remover 50 Practitioners skilled in the
- the filters 70 and 72 generate output signals 80 and 82, respectively, which can be used in a variety of applications For example, they may be connected to a switch box and then to a hearing aid
- the microphone transducers generate input signals 1 1 and 13, which are sampled and digitized, by clocked sample-and-hold circuits followed by analog-to- digital converters 22 and 24, respectively, to produce sampled digital signals 14 and 16 respectively
- the digital signal 14 is supplied to a first delay line 32
- the delay line 32 delays the digitally sampled signal 14 by a non-integral multiple of the sampling interval T, which was 45 35 microseconds given the sampling rate of 22,050 samples per second
- the integral portion of the delay was implemented using a digital delay line, while the remaining subsample delay of less than one sample interval was implemented using a non-causal, truncated sine filter with 41 coefficients
- the output of the first delay line 32 is supplied to the negative input of a second combiner 38
- the first digital signal 14 is also supplied to the positive input of a first combiner 36 Similarly, for the other channel, the second digital signal 16 is supplied to a second delay line 34, which generates a signal which is supplied to the negative input of the first combiner 36
- the sample-and-hold and A/D operations were implemented by the audio input circuits of a Silicon Graphics Indy workstation, and the delay lines and combiners were implemented in software running on the same machine
- other delay lines such as analog delay lines, surface acoustic wave delays, digital low - pass filters, or digital delay lines with higher sampling rates, may be used in place of the digital delay line 32, and 34
- other combiners such as analog voltage subtractors using operational amplifiers, or special purpose digital hardware, may be used in place of the combiners 36 and 38
- the function of the direct signal separator 30 may be seen by referring to Figure 4 Assuming that there are no echoes or reverberations, the acoustic wave signal received by the microphone 12 is the sum of source B and a delayed copy of source A (For clarity in presentation here and in the forthcoming theory section, time relationship between the sources A and B and the microphones 10 and 12 are described as if the electrical signal 14 generated by the microphone 10 were simultaneous with source A and the electrical signal
- the electrical signal supplied to the negative input of the combiner 38 would represent a delayed copy of source A plus a further delayed copy of source B
- the subtraction of the signal from the delay line 32 and digital signal 16 would remove the signal component representing the delayed copy of sound source A, leaving only the pure sound B (along with the further delayed copy of B)
- the amount of delay to be set for each of the digital delay lines 32 and 34 can be supplied from the DOA estimator 20 Numerous methods for estimating the relative time delays have been described in the prior art (for example, Schmidt, 1981 , Roy et al . 1988, Morita, 1991.
- the crosstalk remover 50 comprises a third combiner 56 for receiving the first output signal 40 from the direct signal separator 30
- the third combiner 56 also receives, at its negative input, the output of a second adaptive filter 54
- the output of the third combiner 56 is supplied to a first adaptive filter 52
- the output of the first adaptive filter 52 is supplied to the negative input of the fourth combiner 58, to which the second output signal 42 from the direct signal separator 30 is also supplied
- the outputs of the third and fourth combiners 56 and 58 respectively, are the output signals 60 and 62, respectively of the
- the microphones 10 and 12 are omnidirectional and matched in sensitivity
- time-delay units 32 and 34 As depicted in Figure 2, these operations are accomplished by time-delay units 32 and 34, and combiners 36 and 38
- k, ] '(t)*w,(t) represents the echoes and reverberations of source 1 (w ⁇ (t)) as received at input 1 (microphone 10)
- kp'(t)*w 2 (t) represents the echoes and reverberations of source 2 (w-,(t)) as received at input 1 (microphone 10)
- k 2j '(t)*W ] (t) represents the echoes and reverberations of source 1 (w j (f)) as received at input 2 (microphone 12)
- k 2 '(t)*w-,(t) represents the echoes and reverberations of source 2 (w 7 (t)) as received at input 2 (microphone 12)
- the adaptation rule used for the filter coefficients in the preferred embodiment is a variant of the LMS rule ("Adaptive Signal Processing,” Bernard Widrow and Samuel D Steams, Prentice-Hall, Englewood Cliffs, N , 1985, p 99)
- the filter coefficients are updated at every time-step n.
- m is a constant that determines the rate of adaptation of the filter coefficients, e g 0 15 if the input signals 10 and 12 were normalized to lie in the range -1 ⁇ x(t) ⁇ +1
- the filters h j and h can be implemented in a variety of ways, including FIRs and lattice IIRs
- the direct-signal separator 30 and crosstalk remover 50 adaptively bring about full separation of two sound sources mixed in an echoic, reverberant acoustic environment
- filters 70 and 72 may be desirable to append filters 70 and 72 to the network
- the purpose of these filters is to undo the effects of filters ⁇ (t) and ⁇ 2 (t)
- the invention is able to separate an arbitrary number M of simultaneous sources as long as they are statistically independent if there are at least M microphones
- v (t) be the j'th source signal and x,(t) be the l'th microphone (mic) signal
- t be the time required for sound to propagate from source j to mic l
- d(t ) be the impulse response of a filter that delays a signal by t
- d(t ) is the unit impulse delayed by t that is whe
- the i'th mic signal x ⁇ t can be expressed as a sum of the appropriately delayed j source signals
- X(t) is an V -element column vector whose i'th element is the i'th mic signal x (t)
- D(t) is an MxM element square matrix whose ij'th element (ie , the element in the i'th row and j'th column) is d(t )
- W(t) is an M-element column vector whose j'th element is the j'th source signal w (t)
- y (t) is a "colored" or filtered version of w (t) because of the convolution by the filter impulse response
- this coloration can be undone by post filtering the outputs by a filter that is the inverse of
- the filter 'D(t) may have zeroes at certain frequencies, these make it impossible to exactly realize the inverse of the filter
- any one of the numerous techniques available for approximating filter inverses may be used to derive an approximate filter with which to do the post filtering
- the delays t can be estimated from the statistical properties of the mic signals, up to a constant additive factor that can be different for each source This is the subject of a co- pending patent application by the same inventors, filed even date herewith Alternatively, if the position of each microphone and each source is known, then the delays t can be calculated exactly For any source that is distant, i e , many times farther than the greatest separation between the mics, only the direction of the source is needed to calculate its delays to each mic, up to an arbitrary additive constant
- the first stage of the processor 8 namely the direct signal separator 30, uses the estimated delays to construct the adjugate matrix adjD(t), which it applies to the microphone signals X(t) to obtain the outputs Y(t) of the first stage, given by
- each output y (t) contains energy from a single but different source w (t)
- each mic receives the direct signals from the sources as well as echoes and reverberations from each source
- e (t) is the impulse response of the echo and reverberation path from the j 'th source to the i'th mic
- each y (t) contains a colored direct signal from a single source, as in the case with no echoes, and differently colored components from the echoes and reverberations of every source, including the direct one
- the echoes and reverberations of the other sources are removed by the second stage of the network, namely the crosstalk remover 50, which generates each output as follows
- An optional third stage can use any one of numerous techniques available to reduce the amount of coloration on any individual output.
- the output of the first stage namely the direct signal separator 30, is formed by convolving the mic signals with the adjugate matrix.
- the outputs of the first stage are the individual sources, each colored by the determinant of the delay matrix
- each output of the first stage also contains echoes and reverberations from each source
- the second stage namely the cross talk remover 50, consisting of a feedback network of adaptive filters, removes the effects of these unwanted echoes and reverberations to produce outputs that each contain energy only one different source, respectively.
- the number of microphones and associated channels of signal processing need not be as large as the total number of sources present, as long as the number of sources emitting a significant amount of sound at any given instant in time does not exceed the number of microphones For example, if during one interval of time only sources A and B emit sound, and in another interval of time only sources B and C emit sound, then during the first interval the output channels will correspond to A and B respectively, and during the second interval the output channels will correspond to B and C respectively
- the invention has been implemented in software, as set forth in the microfiche appendix
- the software code is written in the C++ language for execution on a workstation from Silicon Graphics
- hardware implementation of the present invention is also contemplated to be within the scope of the present invention
- the direct signal separator 30 and the crosstalk remover 50 can be a part of a digital signal processor, or can be a part of a general purpose computer, or can be a part of analog signal processing circuitry
- the present invention is not limited to the processing of acoustic waves It can be used to process signals, having a delay with the problem of separation of the signals from the sources
- the present invention is similar in objective to sound separation based on auditory scene analysis, it differs from them in principle and in technique 2 In contrast with approaches based on auditory scene analysis, this invention separates sounds by exploiting the separation of their sources in space and their statistical independence
- the present invention differs from directional microphones in that few presuppositions are made with regard to the locations of sound sources relative to the device
- the present device need not be pointed at or mounted close to a source of interest
- the necessary selectivity is brought about by processing of signals captured by a microphone array, I e , a collection of microphones
- the selectivity attained is much greater a directional microphone cannot completely suppress any source of sound 4
- the p ⁇ or art nonlinear techniques can be very computationally efficient and are of scientific interest as models of human cocktail -party processing, they are of less practical or commercial significance than the present invention because of their inherent inability to bring about full suppression of unwanted sources This inability originates from the incorrect assumption that at every instant in time, at least one microphone contains only the desired signal
- the present invention differs from these nonlinear techniques in that linear operations
- the present invention is an example of "active cancellation"
- active cancellation In contrast with classical beamforming, which aligns copies of a desired source in time and adds them together, active cancellation matches copies of undesired sources and subtracts them to bring about cancellation ("Matching" the copies of the undesired sources generally entails more than simply re-aligning them in time, usually, it involves re-shaping them by filtering )
- the degree of selectivity achieved by active cancellation is determined by factors rather different from those important in classical beamforming
- the degree of selectivity is determined by the exactness with which identical copies of unwanted sources can be created from different microphones for subsequent cancellation
- a sound- separation device that employs active cancellation can in principle remove one undesired source completely using just two microphones
- the present invention also does not need a reference sound 7
- the present invention operates "blindly" it accommodates the more difficult case in which all of the signals directly available for processing, I e , the microphone signals, are assumed to be mixtures of sources
- the present invention is a method for bringing about blind separation of sources, given the direction of arrival (DOA) of each source
- DOA information may be obtained in a variety of ways, for example by direct specification from a human user or by statistical estimation from the microphone signals Precisely how DOA information is obtained is immaterial in the context of the present invention, what is important is how DOA information is used to bring about source separation
- the present invention differs from gain-based active cancellation in that it requires no assumption of simultaneity of signal reception at all of the microphones
- the present invention is based on an acoustic model that includes the effects of echoes and reverberation
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- void SS2 For Output ( float* z ) Form0ut2( z[0], zd) ) ; void SS2::Setup( int argc, char** argv, SS** aa, RepnSS** repn.
- Signal* ya Signal* yb, char** filel, char** fil ⁇ 2, double* lo, double* hi, long* aampa, double* freq, int* delaya, int* croaaacho, int* nrep )
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- void RepnSS ::Copy( const Representation* that. )
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- Searcher :-Searcher() ( ) void Searcher::Initialize! ⁇ ) void Searcher: :Iterate! short ) ()
- RepnSS2 : ⁇ et_bounds( lo, hi, lo, hi ) ;
- SoKeyboardEvent :H ) ;
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- SrchSSpop :-SrchSSpop() ( for! int i * 0; i ⁇ npop; i** ) delete pop(i); delete pop; delete eval; delete income; delete aaaeta; delete income.invrank; delete aaaeta.invrank; delete eolorfrac; ) void SrchSSpop: initialized ( rlcheat • -1; double rlcheat.aeaet ⁇ • 0; for!
- MAX_ ⁇ SSETS min.iteration ⁇ .to.max.as ⁇ ets * max.income.dif f_ er_iteration
- INVJ INVJ ⁇ AX kSSETS (1. / MAX VSSETS) ahort* SrchSSpop : ; aort.invrank * 0; double* S chSSpop : : sort.value * 0; int SrchSSpop: :invrank_d ⁇ creasing( const void* one., const void* two.)
- Hake a ⁇ sets have zero median, and apply limit; find richest and poorest richest.assets * 0 ; riche ⁇ t * -1 ; for! lp * 0; ip ⁇ npop; ip** ) ( aaaeta dp) -» median.aaaeta ; if ! aa ⁇ et ⁇ dp] > HAX_ ⁇ SSETS ) aaaeta (ip) e MAX.ASSETS; if !
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- Engineering & Computer Science (AREA)
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- Acoustics & Sound (AREA)
- Computer Networks & Wireless Communication (AREA)
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Abstract
Description
Claims
Priority Applications (3)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
AU73611/96A AU7361196A (en) | 1995-09-18 | 1996-09-13 | A directional acoustic signal processor and method therefor |
EP96935819A EP0806092A1 (en) | 1995-09-18 | 1996-09-13 | A directional acoustic signal processor and method therefor |
JP9512787A JPH10510127A (en) | 1995-09-18 | 1996-09-13 | Directional sound signal processor and method |
Applications Claiming Priority (2)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
US08/531,143 US6002776A (en) | 1995-09-18 | 1995-09-18 | Directional acoustic signal processor and method therefor |
US08/531,143 | 1995-09-18 |
Publications (1)
Publication Number | Publication Date |
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WO1997011533A1 true WO1997011533A1 (en) | 1997-03-27 |
Family
ID=24116419
Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
---|---|---|---|
PCT/US1996/014680 WO1997011533A1 (en) | 1995-09-18 | 1996-09-13 | A directional acoustic signal processor and method therefor |
Country Status (6)
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---|---|
US (1) | US6002776A (en) |
EP (1) | EP0806092A1 (en) |
JP (1) | JPH10510127A (en) |
KR (1) | KR970707648A (en) |
AU (1) | AU7361196A (en) |
WO (1) | WO1997011533A1 (en) |
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Also Published As
Publication number | Publication date |
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KR970707648A (en) | 1997-12-01 |
EP0806092A1 (en) | 1997-11-12 |
US6002776A (en) | 1999-12-14 |
AU7361196A (en) | 1997-04-09 |
JPH10510127A (en) | 1998-09-29 |
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