(19) United States
`(12) Patent Application Publication (10) Pub. No.: US 2008/0152167 A1
`Taenzer
`(43) Pub. Date:
`Jun. 26, 2008
`
`US 2008O152167A1
`
`(54) NEAR-FIELD VECTOR SIGNAL
`ENHANCEMENT
`
`(75) Inventor:
`
`o: Taenzer, Los Altos, CA
`
`Correspondence Address:
`THELEN RED BROWN RAYSMAN &
`STENER LLP
`P. O. BOX 640640
`SAN JOSE, CA 95164-0640
`
`(73) Assignee:
`
`STEP Communications
`Corporation, San Jose, CA (US)
`
`(21) Appl. No.:
`
`11/645,019
`
`(22) Filed:
`
`Dec. 22, 2006
`O
`O
`Publication Classification
`
`(51) Int. Cl.
`H04B I5/00
`
`
`
`(2006.01)
`
`WINDOW
`
`FRAME,
`WINDOW
`& DFT
`
`(52) U.S. Cl. ....................................................... 381A942
`
`ABSTRACT
`(57)
`Near-field sensing of wave signals, for example for applica
`tion in headsets and earsets, is accomplished by placing two
`or more spaced-apart microphones along a line generally
`between the headset and the user's mouth. The signals pro
`duced at the output of the microphones will disagree in ampli
`tude and time delay for the desired signal—the wearer's
`voice but will disagree in a different manner for the ambient
`noises. Utilization of this difference enables recognizing, and
`Subsequently ignoring, the noise portion of the signals and
`passing a clean Voice signal. A first approach involves a
`complex vector difference equation applied in the frequency
`domain that creates a noise-reduced result. A second
`approach creates an attenuation value that is proportional to
`the complex vector difference, and applies this attenuation
`value to the original signal in order to effect a reduction of the
`noise. The two approaches can be applied separately or com
`bined.
`
`OUTPUT
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`(Bp) BONBHEBHICI Hon LIN?VIN
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`DISTANCE (m)
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`FIG. 15
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`DISTANCE (m)
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`FIG.16
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`US 2008/O152167 A1
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`Jun. 26, 2008
`
`NEAR-FIELD VECTOR SIGNAL
`ENHANCEMENT
`
`CROSS-REFERENCE TO RELATED
`APPLICATIONS
`0001 (Not Applicable)
`
`BACKGROUND OF THE INVENTION
`
`0002 1. Field of the Invention
`0003. The invention relates to near-field sensing systems.
`0004 2. Description of the Related Art
`0005. When communicating in noisy ambient conditions,
`a voice signal may be contaminated by the simultaneous
`pickup of ambient noises. Single-channel noise reduction
`methods are able to provide a measure of noise removal by
`using a-priori knowledge about the differences between
`Voice-like signals and noise signals to separate and reduce the
`noise. However, when the “noise' consists of other voices or
`Voice-like signals, single-channel methods fail. Further, as
`the amount of noise removal is increased, some of the Voice
`signal is also removed, thereby changing the purity of the
`remaining Voice signal—that is, the Voice becomes distorted.
`Further, the residual noise in the output signal becomes more
`voice-like. When used with speech recognition software,
`these defects decrease recognition accuracy.
`0006 Array techniques attempt to use spatial or adaptive
`filtering to either: a) increase the pickup sensitivity to signals
`arriving from the direction of the Voice while maintaining or
`reducing sensitivity to signals arriving from other directions,
`b) to determine the direction towards noise sources and to
`steer beam pattern nulls toward those directions, thereby
`reducing sensitivity to those discrete noise sources, or c) to
`deconvolve and separate the many signals into their compo
`nent parts. These systems are limited in their ability to
`improve signal-to-noise ratio (SNR), usually by the practical
`number of sensors that can be employed. For good perfor
`mance, large numbers of sensors are required. Further, null
`steering (Generalized Sidelobe Canceller or GSC) and sepa
`ration (Blind Source Separation or BSS) methods require
`time to adapt their filter coefficients, thereby allowing signifi
`cant noise to remain in the output during the adaptation period
`(which can be many seconds). Thus, GSC and BSS methods
`are limited to semi-stationary situations.
`0007. A good description of the prior art pertaining to
`noise cancellation/reduction methods and systems is con
`tained in U.S. Pat. No. 7,099,821 by Visser and Lee entitled
`“Separation of Target Acoustic Signals in a Multi-Transducer
`Arrangement'. This reference covers not only at-ear, but also
`remote (off-ear) Voice pick-up technologies.
`0008 Prior art technologies for at-ear voice pickup sys
`tems recently have been driven by the availability and public
`acceptance of wired and wireless headsets, primarily for use
`with cellular telephones. A boom microphone system, in
`which the microphone's sensing port is located very close to
`the mouth, long has been a solution that provides good per
`formance due to its close proximity to the desired signal. U.S.
`Pat. No. 6,009,184 by Tate and Wolff entitled “Noise Control
`Device for a Boom Mounted Noise-canceling Microphone'
`describes an enhanced version of Such a microphone. How
`ever, demand has driven a reduction in the size of headset
`devices so that a conventional prior art boom microphone
`Solution has become unacceptable.
`
`0009 Current at-ear headsets generally utilize an omni
`directional microphone located at the very tip of the headset
`closest to the user's mouth. In current devices this means that
`the microphone is located 3" to 4" away from the mouth and
`the amplitude of the Voice signal is Subsequently reduced by
`the 1/r spreading effect. However, noise signals, which are
`generally arriving from distant locations, are not reduced so
`the result is a degraded signal-to-noise ratio (SNR).
`0010 Many methods have been proposed for improving
`SNR while preserving the reduced size and more distant
`from-the-mouth location of modern headsets. Relatively
`simple first-order microphone systems that employ pressure
`gradient methods, either as “noise canceling microphones or
`as directional microphones (e.g. U.S. Pat. Nos. 7,027,603;
`6,681,022: 5,363,444; 5,812,659; and 5,854,848) have been
`employed in an attempt to mitigate the deleterious effects of
`the at-ear pick-up location. These methods introduce addi
`tional problems: the proximity effect, exacerbated wind noise
`sensitivity and electronic noise, frequency response colora
`tion of far-field (noise) signals, the need for equalization
`filters, and if implemented electronically with dual micro
`phones, the requirement for microphone matching. In prac
`tice, these systems also suffer from on-axis noise sensitivity
`that is identical to that of their omni-directional brethren.
`0011. In order to achieve better performance, second-or
`der directional systems (e.g. U.S. Pat. No. 5,473.684 by Bar
`tlett and Zuniga entitled “Noise-canceling Differential
`Microphone Assembly’) have also been attempted, but the
`defects common to first-order systems are also greatly mag
`nified so that wind noise sensitivity, signal coloration, elec
`tronic noise, in addition to equalization and matching require
`ments, make this approach unacceptable.
`0012. Thus, adaptive systems based upon GSC, BSS or
`other multi-microphone methods also have been attempted
`with some success (see for example McCarthy and Boland,
`“The Effect of Near-field Sources on the Griffiths-Jim Gen
`eralized Sidelobe Canceller. Institution of Electrical Engi
`neers, London, IEE conference publication ISSN 0537-9989,
`CODEN IECPB4, and U.S. Pat. Nos. 7,099,821; 6,799,170:
`6,691,073; and 6,625,587). Such systems suffer from
`increased complexity and cost, multiple sensors requiring
`matching, slow response to moving or rapidly changing noise
`Sources, incomplete noise removal and Voice signal distortion
`and degradation. Another drawback is that these systems
`operate only with relatively clean (positive SNR) input sig
`nals, and actually degrade the signal quality when operating
`with poor (negative SNR) input signals. The Voice degrada
`tion often interferes with Automatic Speech Recognition
`(ASR), a major application for Such headsets.
`0013 Another, multi-microphone noise reduction tech
`nology applicable to headsets is disclosed by Luo, et al. in
`U.S. Pat. No. 6,668,062 entitled “FFT-based Technique for
`Adaptive Directionality of Dual Microphones’. In this
`method, developed for use in hearing aids, two microphones
`are spaced approximately 10-cm apart within a behind-the
`ear or BTE hearing aid case. The microphone input signals are
`converted to the frequency domain and an output signal is
`created using the equation
`
`Y(co)
`Zo) = x(a)-x(oxyi
`
`(1)
`
`Page 15 of 26
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`
`Jun. 26, 2008
`
`where X(c)), Y(CD) and Z(CD) are the frequency domain trans
`forms of the time domain input signals X(t) and y(t), and the
`time domain output signal Z(t). In hearing aids the goal is to
`help the user to clearly hear the conversations of other indi
`viduals and also to hear environmental sounds, but not to hear
`the user him/herself. Thus, this technology is designed to
`clarify far-field sounds. Further, this technology operates to
`produce a directional sensitivity pattern that “cancels noise.
`... when the noise and the target signal are not in the same
`direction from the apparatus”. The downsides are that this
`technology significantly distorts the desired target signal and
`requires excellent microphone array element matching.
`0014. Others have developed technologies specifically for
`near-field sensing applications. For example, Goldin (U.S.
`Publication No. 2006/0013412 A1 and “Close Talking Auto
`directive Dual Microphone'. AES Convention, Berlin, Ger
`many, May 8-11, 2004) has proposed using two microphones
`with controllable delay-&-add technology to create a set of
`first-order, narrow-band pick-up beam patterns that optimally
`steer the beams away from noise sources. The optimization is
`achieved through real-time adaptive filtering which creates
`the independent control of each delay using LMS adaptive
`means. This scheme has also been utilized in modern DSP
`based hearing aids. Although essentially GSC technology, for
`near-field Voice pick-up applications this system has been
`modified to achieve non-directional noise attenuation. Unfor
`tunately, when there is more than a single noise source at a
`particular frequency, this system can not optimally reduce the
`noise. In real situations, even if there is only one physical
`noise source, room reverberations effectively create addi
`tional virtual noise sources with many different directions of
`arrival, but all having the identical frequency content thereby
`circumventing this method’s ability to operate effectively. In
`addition, by being adaptive, this scheme requires substantial
`time to adjust in order to minimize the noise in the output
`signal. Further, the rate of noise attenuation vs. distance is
`limited and the residual noise in the output signal is highly
`colored, among other defects.
`
`BRIEF SUMMARY OF THE INVENTION
`0.015. In accordance with one embodiment described
`herein, there is provided a voice sensing method for signifi
`cantly improved voice pickup in noise applicable for example
`in a wireless headset. Advantageously it provides a clean,
`non-distorted Voice signal with excellent noise removal,
`wherein Small residual noise is not distorted and retains its
`original character. Functionally, a Voice pickup method for
`better selecting the user's voice signal while rejecting noise
`signals is provided.
`0016. Although discussed in terms of voice pickup (i.e.
`acoustic, telecom and audio), the system herein described is
`applicable to any wave energy sensing system (wireless radio,
`optical, geophysics, etc.) where near-field pick-up is desired
`in the presence of far-field noises/interferers. An alternative
`use gives Superior far-field sensing for astronomy, gamma
`ray, medical ultrasound, and so forth.
`0017 Benefits of the system disclosed herein include an
`attenuation of far-field noise signals at a rate twice that of
`prior art systems while maintaining flat frequency response
`characteristics. They provide clean, natural Voice output,
`highly reduced noise, high compatibility with conventional
`transmission channel signal processing technology, natural
`Sounding low residual noise, excellent performance in
`extreme noise conditions—even in negative SNR condi
`
`tions—instantaneous response (no adaptation time prob
`lems), and yet demonstrate low compute power, memory and
`hardware requirements for low cost applications.
`0018 Acoustic voice applications for this technology
`include mobile communications equipment Such as cellular
`handsets and headsets, cordless telephones, CB radios,
`walkie-talkies, police and fire radios, computer telephony
`applications, stage and PA microphones, lapel microphones,
`computer and automotive voice command applications, inter
`coms and so forth. Acoustic non-voice applications include
`sensing for active noise cancellation systems, feedback detec
`tors for active Suspension systems, geophysical sensors,
`infrasonic and gunshot detector systems, underwater warfare
`and the like. Non-acoustic applications include radio and
`radar, astrophysics, medical PET scanners, radiation detec
`tors and Scanners, airport security systems and so forth.
`0019. The system described herein can be used to accu
`rately sense local noises, so that these local noise signals can
`be removed from mixed signals that contain desired far-field
`signals, thereby obtaining clean sensing of the far-field sig
`nals.
`0020. Yet another use is to reverse the described attenua
`tion action so that near-field Voice signals are removed and
`only the noise is preserved. Then this resulting noise signal,
`along with the original input signals, can be sent to a spectral
`subtraction, Generalized Sidelobe Canceller, Weiner filter,
`Blind Source Separation system or other noise removal appa
`ratus where a clean noise reference signal is needed for accu
`rate noise removal.
`0021. The system does not change the purity of the
`remaining voice while improving upon the signal-to-noise
`ratio (SNR) improvement performance of beam forming
`based systems and it adapts much more quickly than do GSC
`or BSS methods. With these other systems, SNR improve
`ments are still below 10-dB in most high noise applications.
`
`BRIEF DESCRIPTION OF THE SEVERAL
`VIEWS OF THE DRAWINGS
`0022. Many advantages of the present invention will be
`apparent to those skilled in the art with a reading of this
`specification in conjunction with the attached drawings,
`wherein like reference numerals are applied to like elements,
`and wherein:
`0023 FIG. 1 is a schematic diagram of a type of a wearable
`near-field audio pick-up device;
`0024 FIG. 1A is a block diagram illustrating a general
`pick-up process;
`0025 FIG. 2 is generalized block diagram of a system for
`accomplishing noise reduction;
`0026 FIG. 3 is a block diagram showing processing
`details;
`0027 FIG. 4 is a block diagram of a signal processing
`portion of a direct equation approach;
`0028 FIG. 5 shows on-axis sensitivity relative to the
`mouth sensitivity vs. distance from the headset:
`0029 FIG. 6 shows the attenuation response of a system at
`seven different arrival angles from 0 to 180°:
`0030 FIG. 7 is a plot of the directionality pattern of a
`system using two omni-directional microphones and mea
`sured at a source range of 0.13 m (5");
`0031
`FIG. 8 shows attenuation created by Equation (7) as
`a function of the magnitude difference between the front
`microphone signal and the rear microphone signal for the 3
`dB design example;
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`0032 FIG. 9 shows the attenuation characteristics pro
`duced by Equations (8) and (9) as compared with that pro
`duced by Equation (7);
`0033 FIG. 10 shows a block diagram of how an attenua
`tion technique can be implemented without the need for the
`real-time calculation of Equation (7);
`0034 FIG. 11 shows a block diagram of a processing
`method employing full attenuation to the output signal;
`0035 FIG. 12 demonstrates a block diagram of a calcula
`tion approach for limiting the output to expected signals;
`0036 FIG. 13 is an example limit table:
`0037 FIGS. 14A and 14B show a set of limits plotted
`Versus frequency;
`0038 FIG. 15 shows a graph of sensitivity as a function of
`the Source distance away from the microphone array along the
`major axis and that of a prior art system; and
`0039 FIG. 16 shows the data of FIG. 15 graphed on a
`logarithmic distance scale to better demonstrate the improved
`performance.
`
`DETAILED DESCRIPTION OF THE INVENTION
`
`0040 Embodiments of the present invention are described
`herein in the context of near-field pick-up systems. Those of
`ordinary skill in the art will realize that the following detailed
`description of the present invention is illustrative only and is
`not intended to be in any way limiting. Other embodiments of
`the present invention will readily suggest themselves to Such
`skilled persons having the benefit of this disclosure. Refer
`ence will now be made in detail to implementations of the
`present invention as illustrated in the accompanying draw
`ings. The same reference indicators will be used throughout
`the drawings and the following detailed description to refer to
`the same or like parts.
`0041. In the interest of clarity, not all of the routine fea
`tures of the implementations described herein are shown and
`described. It will, of course, be appreciated that in the devel
`opment of any such actual implementation, numerous imple
`mentation-specific decisions must be made in order to
`achieve the developer's specific goals, such as compliance
`with application- and business-related constraints, and that
`these specific goals will vary from one implementation to
`another and from one developer to another. Moreover, it will
`be appreciated that such a development effort might be com
`plex and time-consuming, but would nevertheless be a routine
`undertaking of engineering for those of ordinary skill in the
`art having the benefit of this disclosure.
`0042. The system described herein is based upon the use
`of a controlled difference in the amplitude of two detected
`signals in order to retain, with excellent fidelity, signals origi
`nating from nearby locations while significantly attenuating
`those originating from distant locations. Although not con
`strained to audio and sound detection apparatus, presently the
`best application is in head worn headsets, in particular wire
`less devices known as Bluetooth R) headsets.
`0043 Recognizing that energy waves are basically spheri
`cal as they spread out from a source, it can be seen that Such
`waves originating from nearby (near-field) source locations
`are greatly curved, while waves originating from distant (far
`field) source locations are nearly planar. The intensity of an
`energy wave is its power/unit area. As energy spreads out, the
`intensity drops off as 1/r, wherer is distance from the source.
`Magnitude is the square root of intensity, so the magnitude
`drops off as 1/r. The greater the difference in distance of two
`
`detectors from a source, the greater is the difference in mag
`nitude between the detected signals.
`0044. The system employs a unique combination of a pair
`of microphones located at the ear, and a signal process that
`utilizes the magnitude difference in order to preserve a voice
`signal while rapidly attenuating noise signals arriving from
`distant locations. For this system, the drop off of signal sen
`sitivity as a function of distance is double that of a noise
`canceling microphone located close to the mouth as in a high
`end boom microphone system, yet the frequency response is
`still zeroth-order that is, inherently flat. Noise attenuation is
`not achieved with directionally so all noises, independent of
`arrival direction, are removed. In addition, due to its zeroth
`order sensitivity response, the system does not suffer from the
`proximity effect and is wind noise-resistant, especially using
`the second processing method described below.
`0045. The system effectively provides an appropriately
`designed microphone array used with proper analog and A/D
`circuitry designed to preserve the signal “cues required for
`the process, combined with the system process itself. It
`should be noted that the input signals are often "contami
`nated with significant noise energy. The noise may even be
`greater than the desired signal. After the system's process has
`been applied, the output signal is cleaned of the noise and the
`resulting output signal is usually much Smaller. Thus, the
`dynamic range of the input signal path should be designed to
`linearly preserve the high input dynamic range needed to
`encompass all possible input signal amplitudes, while the
`dynamic range requirement for the output path is often
`relaxed in comparison.
`Microphone Array
`0046. A microphone array formed of at least two separated
`microphones preferably positioned along a line (axis)
`between the headset location and the user's mouth in par
`ticular the upper lip is a preferred target so that both oral and
`nasal utterances are detected is shown in FIG. 1. Only two
`microphones are shown, but a greater number can be used.
`The two microphones are designated 10 and 12 and are
`mounted on or in a housing 16. The housing may have an
`extension portion 14. Another portion of the housing or a
`Suitable component is disposed in the opening of the earcanal
`of the wearer such that the speaker of the device can be heard
`by wearer. Although the microphone elements 10 and 12 are
`preferably omni-directional units, noise canceling and uni
`directional devices and even active array systems also may be
`compatibly utilized. When directional microphones or micro
`phone systems are used, they are preferably aimed toward the
`user's mouth to thereby provide an additional amount of noise
`attenuation for noise sources located at less sensitive direc
`tions from the microphones.
`0047. The remaining discussion will focus primarily on
`two omni-directional microphone elements 10 and 12, with
`the understanding that other types of microphones and micro
`phone systems can be used. For the remaining description, the
`microphone closest to the mouth that is, microphone
`10 will be called the “front microphone and the micro
`phone farthest from the mouth (12) the “rear microphone.
`0048. In simple terms, using the example of two spaced
`apart microphones located at the ear of the user and on a line
`approximately extending in the direction of the mouth, the
`two microphone signals are detected, digitized, divided into
`time frames and converted to the frequency domain using
`conventional digital Fourier transform (DFT) techniques. In
`
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`Jun. 26, 2008
`
`the frequency domain, the signals are represented by complex
`numbers. After optional time alignment of the signals, 1) the
`difference between pairs of those complex numbers is com
`puted according to a mathematical equation, or 2) their
`weighted Sum is attenuated according to a different math
`ematical equation, or both. Since in the system described
`herein there is no inherent restriction on microphone spacing
`(as long as it is not Zero), other system considerations are the
`driving factors on the choice of the time alignment approach.
`0049. The ratio of the vector magnitudes, or norms, is used
`as a measure of the “noisiness” of the input data to control the
`noise attenuation created by each of the two methods. The
`result of the processing is a noise reduced frequency domain
`output signal, which is Subsequently transformed by conven
`tional inverse Fourier means to the time domain where the
`output frames are overlapped and added together to create the
`digital version of the output signal. Subsequently, D/A con
`version can be used to create an analog output version of the
`output signal when needed. This approach involves digital
`frequency domain processing, which the remainder of this
`description will further detail. It should be recognized, how
`ever, that alternative approaches include processing in the
`analog domain, or digital processing in the time domain, and
`so forth.
`0050 Normalizing the acoustic signals sensed by the two
`microphones 10 and 12 to that of the front microphone 10,
`then the front microphone's frequency domain signal is, by
`definition, equal to “1,” That is,
`(2)
`S(0,0.dr)=1
`where () is the radian frequency, 0 is the effective angle of
`arrival of the acoustic signal relative to the direction toward
`the mouth (that is, the array axis), d is the separation distance
`between the two microphone ports and r is the range to the
`sound source from the front microphone 10 in increments of
`d. Thus, the frequency domain signal from the rear micro
`phone 12 is
`
`S(0, 0, d, r) = ye", where
`
`y = 1 + cos(0)+ 5,
`
`(3)
`
`4
`(4)
`
`c is the effective speed of sound at the array, and i is the
`imaginary operator -1. The term rd(y-1)/c represents the
`arrival time difference (delay) of an acoustic signal at the two
`microphone ports. It can be seen from these equations that
`when r is large, in other words when a sound Source is far
`away from the array, the magnitude of the rear signal is equal
`to “1, the same as that of the front signal.
`0051. When the source signal is arriving on-axis from a
`location along a line toward the user's mouth (0–0), the
`magnitude of the rear signal is
`
`(5)
`
`0052. As an example of how this result is used in the
`design of the array, assume that the designer desires the
`
`magnitude of the voice signal to be 3 dB higher in the front
`microphone 10 than it is in the rear microphone 12. In this
`Case,
`
`= 1032 = 0.708
`
`and thus r=2.42. Therefore, the front microphone 10 should
`be located 2.42d away from the mouth, and, of course, the
`rear microphone 12 should be located a distanced behind the
`front microphone. If the distance from the mouth to the front
`microphone 10 will be, for example, 12-cm (434-in) in a
`particular design, then the desired port-to-port spacing in the
`microphone array—that is the separation between the micro
`phones 10 and 12 will be 4.96-cm (about 5-cm or 2-in). Of
`course, the designer is free to choose the magnitude ratio
`desired for any particular design.
`Microphone Matching
`0053 Some processing steps that may be initially applied
`to the signals from the microphones 10 and 12 are described
`with reference to FIG. 1A. It is advantageous to provide
`microphone matching, and using omni-directional micro
`phones, microphone matching is easily achieved. Omni-di
`rectional microphones are inherently flat response devices
`with virtually no phase mismatch between pairs. Thus, any
`simple prior art level matching method suffices for this appli
`cation. Such methods range from purchasing pre-matched
`microphone elements for microphones 10 and 12, factory
`selection of matched elements, post-assembly test fixture
`dynamic testing and adjustment, post-assembly mismatch
`measurement with matching “table' insertion into the device
`for operational on-the-fly correction, to dynamic real-time
`automatic algorithmic mismatch correction.
`Analog Signal Processing
`0054 As shown in FIG. 1A, analog processing of the
`microphone signals may be performed and typically consists
`of pre-amplification using amplifiers 11 to increase the nor
`mally very Small microphone output signals and possibly
`filtering using filters 13 to reduce out-of-band noise and to
`address the need for anti-alias filtering prior to digitization of
`the signals ifused in a digital implementation. However, other
`processing can also be applied at this stage. Such as limiting,
`compression, analog microphone matching (15) and/or
`squelch.
`0055. The system described herein optimally operates
`with linear, undistorted input signals, so the analog process
`ing is used to preserve the spectral purity of the input signals
`by having good linearity and adequate dynamic range to
`cleanly preserve all parts of the input signals.
`
`A/D-D/A Conversion
`0056. The signal processing conducted herein can be
`implemented using an analog method in the time domain. By
`using a bank of band-split filters, combined with Hilbert
`transformers and well known signal amplitude detection
`means, to separate and measure the magnitude and phase
`components within each band, the processing can be applied
`on a band-by-band basis where the multi-band outputs are
`then combined (added) to produce the final noise reduced
`analog output signal.
`
`Page 18 of 26
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`US 2008/O152167 A1
`
`Jun. 26, 2008
`
`0057 Alternatively, the signal processing can be applied
`digitally, either in the time domain or in the frequency
`domain. The digital time-domain method, for example, can
`perform the same steps and in the same order as identified
`above for the analog method, or may be any other appropriate
`method.
`0058 Digital processing can also be accomplished in the
`frequency domain using Digital Fourier Transform (DFT),
`Wavelet Transform, Cosine Transform, Hartley transform or
`any other means to separate the information into frequency
`bands before processing.
`0059 Microphone signals are inherently analog, so after
`the application of any desired analog signal processing, the
`resulting processed analoginput signals are converted to digi
`tal signals. This is the purpose of the A/D converters (22, 24)
`shown in FIGS. 1A and 2-one conversion channel per input
`signal. Conventional A/D conversion is well known in the art,
`so there is no need for discussion of the requirements on
`anti-aliasing filtering, sample rate, bit depth, linearity and the
`like since standard good practices suffice.
`0060. After the noise reduction processing, for example by
`circuit 30 in FIG. 2, is complete, a single digital output signal
`is created. This output signal can be utilized in a digital
`system without further conversion, or alternatively can be
`converted back to the analog domain using a conventional
`D/A converter system as known in the art.
`
`T