(19) United States
`(12) Patent Application Publication (10) Pub. No.: US 2009/0268931 A1
`(43) Pub. Date:
`Oct. 29, 2009
`ANDREA et al.
`
`US 20090268931A1
`
`(54) HEADSET WITH INTEGRATED STEREO
`ARRAY MICROPHONE
`Douglas ANDREA, Sag Harbor,
`(76) Inventors:
`NY (US); Qunsheng Liu, Queens,
`NY (US); John Probst, Hauppauge,
`NY (US)
`Correspondence Address:
`FROMMER LAWRENCE & HAUG
`745 FIFTHAVENUE- 1 OTH FL.
`NEW YORK, NY 10151 (US)
`Appl. No.:
`12/429,623
`Filed:
`Apr. 24, 2009
`Related U.S. Application Data
`(60) Provisional application No. 61/048,142, filed on Apr.
`25, 2008.
`
`(21)
`(22)
`
`Publication Classification
`
`(51) Int. Cl.
`H04R 5/02
`H04B I5/00
`
`(2006.01)
`(2006.01)
`
`(52) U.S. Cl. ......................... 381/311: 381/94.1: 381/309
`
`ABSTRACT
`(57)
`The invention relates to a noise canceling audio transmitting/
`receiving device; a stereo headset with an integrated array of
`microphones utilizing an adaptive beam forming algorithm.
`The invention also relates to a method of using an adaptive
`beam forming algorithm that may be incorporated into a
`Stereo headset. The sensor array used herein has adaptive
`filtering capabilities.
`
`260
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`WS/ 27.
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`C)
`2. Q 7. 24
`2Af 22 N 7
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`260
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`220
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`Page 1 of 18
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`Patent Application Publication
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`Oct. 29, 2009 Sheet 1 of 12
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`20
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`Beam Forming Algorithm
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`Two Channel
`input
`
`
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`Average Fifter
`Calculations
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`22
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`Directional
`Beam
`Steering
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`24
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`Convert to
`Frequency
`Domain
`(DFT)
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`26
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`32
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`Calculate
`thresholds
`and adjust
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`48
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`Update
`Adaptive
`Filter
`Coefficients
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`Filter
`Coefficient
`Calculation
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`Noise
`Calculation
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`Update
`Beam
`Average
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`late
`oise
`Estimator
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`Calculate.
`Beam, Beam
`Reference,
`Reference
`Average,
`Beam
`Average
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`34
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`36
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`Update
`SNR
`Calculation
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`Calculate
`Error from
`Reference
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`Update
`Reference
`Average
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`44
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`Filter the signals
`with the updated
`filter values
`
`Convert back to
`Tirne Domain
`(IDFT)
`
`Two Channel
`Output (Time
`Domain)
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`Adaptive Feedback 50
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`F G. 1
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`AUDIOCOMMANDER
`
`PLAYBACK
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`(2) -š
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`31 Hz 62Hz 125Hz 250Hz 500Hz kHz 2kHz 4kHz 8kHz 16kHz
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`
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`RECORDING
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`
`Microphone
`Andrea PureAuto USB-SA Headset
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`G No enhancements
`O Directional beam
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`Microphone Enhnacements
`No enhancements
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`O Directional beam
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`F. G. 6
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`FIG. 7
`(PRIOR ART)
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`F. G. 8
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`FIG. 9(c)
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`FIG. 9(d)
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`FIG 10
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`US 2009/0268931 A1
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`Oct. 29, 2009
`
`HEADSET WITH INTEGRATED STEREO
`ARRAY MICROPHONE
`
`INCORPORATION BY REFERENCE
`0001. The present application claims the benefit of Provi
`sional Application No. 61/048,142 filed Apr. 25, 2008 and
`co-pending Provisional Application No. 61/012,884 filed
`Dec. 11, 2007. The present application also makes reference
`to U.S. patent application Ser. No. 12/332,959 filed on Dec.
`11, 2008 which claims benefit Provisional Application No.
`61/012.884. All of these applications are incorporated herein
`by reference.
`0002 Each document cited in this text (“application cited
`documents') and each document cited or referenced in each
`of the application cited documents, and any manufacturer's
`specifications or instructions for any products mentioned in
`this text and in any document incorporated into this text, are
`hereby incorporated herein by reference; and, technology in
`each of the documents incorporated herein by reference can
`be used in the practice of this invention.
`0003 Citation or identification of any document in this
`application is not an admission that such document is avail
`able as prior art to the present invention.
`
`BACKGROUND OF THE INVENTION
`0004. 1. Field of the Invention
`0005. The invention generally relates to noise canceling
`audio transmitting/receiving devices such as headsets with
`microphones, and particularly relates to stereo headsets inte
`grated with an array of microphones for use in internet gam
`1ng.
`0006 2. Description of Prior Art
`0007. There is a proliferation of mainstream PC games
`that Support Voice communications. Team chat communica
`tion applications are used such as Ventrilo(R). These commu
`nication applications are being used on networked computers,
`utilizing Voice over Internet Protocol (VOIP) technology. PC
`game players typically utilize PC headsets to communicate
`via the internet and the earphones help to immerse themselves
`in the game experience.
`0008. When gamers need to communicate with team part
`ners or taunt their competitors, they typically use headsets
`with close talking boom microphones, for example as shown
`in FIG. 7. The boom microphone may have a noise cancella
`tion microphone, so their voice is heard clearly and any
`annoying background noise is cancelled. In order for these
`types of microphones to operate properly, they need to be
`placed approximately one inch in front of the user's lips.
`0009 Gamers are, however, known to play for many hours
`without getting up from their computer terminal. During pro
`longed game sessions, the gamers like to eat and drink while
`playing for these long periods of time. If the gamer is not
`communicating via VoIP, he may move the boom microphone
`with his hand into an upright position to move it away from in
`front of his face. If the gamer wants to eat or drink, he would
`also need to use one hand to move the close talking micro
`phone from in front of his mouth. Therefore if the gamer
`wants to be unencumbered from constantly moving the
`annoying close talking boom microphone and not to take his
`hands away from the critical game control devices, an alter
`native microphone solution would be desirable.
`0010. Accordingly, there is a need for a high fidelity far
`field noise canceling microphone that possesses good back
`
`ground noise cancellation and that can be used in any type of
`noisy environment, especially in environments where a lot of
`music and speech may be present as background noise (as in
`a game arena or internet cafe), and a microphone that does not
`need the user to have to deal with positioning the microphone
`from time to time. Therefore, an object of the present inven
`tion is to provide for a device that integrates both these fea
`tures. A further object of the invention is to provide for a
`Stereo headset with an integrated array of microphones uti
`lizing an adaptive beam forming algorithm. This preferred
`embodiment is a new type of “boom free' headset, which
`improves the performance, convenience and comfort of a
`game player's experience by integrating the above discussed
`features.
`
`SUMMARY OF THE INVENTION
`0011. The present invention relates to a noise canceling
`audio transmitting/receiving device; a stereo headset with an
`integrated array of microphones utilizing an adaptive beam
`forming algorithm. The invention also relates to a method of
`using an adaptive beam forming algorithm that can be incor
`porated into a stereo headset.
`0012. One embodiment of the present invention may be a
`noise canceling audio transmitting/receiving device which
`may comprise at least one audio outputting component, and at
`least one audio receiving component, wherein each of the
`receiving means may be directly mounted on a Surface of a
`corresponding outputting means. The noise canceling audio
`transmitting/receiving device may be a stereo headset or a ear
`budset. At least one audio outputting means may be a speaker,
`headphone, or an earphone, and at least one audio receiving
`means may be a microphone. The microphone may be a unior
`omni-directional electret microphone, or a microelectrome
`chanical systems (MEMS) microphone. The noise canceling
`audio transmitting/receiving device may also include a con
`necting means to connect to a computing device oran external
`device, and the noise canceling audio transmitting/receiving
`device may be connected to the computing device or the
`external device via a stereo speaker/microphone input or
`Bluetooth R) or a USB external sound card device. The posi
`tion of at least one audio receiving means may be adjustable
`with respect to a user's mouth.
`0013 For a better understanding of the invention, its oper
`ating advantages and specific objects attained by its uses,
`reference is made to the accompanying descriptive matter in
`which preferred, but non-limiting, embodiments of the inven
`tion are illustrated.
`
`BRIEF DESCRIPTION OF THE DRAWINGS
`0014. The accompanying drawings, which are included to
`provide a further understanding of the invention, are incor
`porated in and constitute a part of this specification. The
`drawings presented herein illustrate different embodiments
`of the invention and together with the description serve to
`explain the principles of the invention. In the drawings:
`0015 FIG. 1 is a schematic depicting a beam forming
`algorithm according to one embodiment of the invention;
`0016 FIG. 2 is a drawing depicting a polar beam plot of a
`2 member microphone array, according to one embodiment of
`the invention;
`0017 FIG. 3 shows an input wave file that is fed into a
`Microsoft(R) array filter and an array filter according to one
`embodiment of the present invention;
`
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`Oct. 29, 2009
`
`0018 FIG. 4 depicts a comparison between the filtering of
`Microsoft(R) array filter with an array filter according to one
`embodiment of the present invention;
`0019 FIG. 5 is a depiction of an example of a visual
`interface that can be used in accordance with the present
`invention;
`0020 FIG. 6 is a portion of the visual interface shown in
`FIG. 5;
`FIG. 7 is a photograph of a headset from prior art;
`0021
`0022 FIG. 8 is a photograph of a headset with micro
`phones on either side, according to one embodiment of the
`invention;
`0023 FIG. 9(a)-9(d) are illustrations of the headset,
`according to one embodiment of the invention; and
`0024 FIG. 10 is an illustration of the functioning of the
`headset with microphones, according to one embodiment of
`the invention.
`
`DETAILED DESCRIPTION OF THE INVENTION
`0025. According to an embodiment of the present inven
`tion, a sensor array, receives signals from a source. The digi
`tized output of the sensors may then transformed using a
`discrete Fourier transform (DFT).
`0026. The sensors of the sensor array preferably are
`microphones. In one embodiment the microphones are
`aligned on a particular axis. In the simplest embodiment the
`array comprises two microphones on a straight line axis.
`Normally, the array consists of an even number of sensors,
`with the sensors, according to one embodiment, at a fixed
`distance apart from each adjacent sensor. However, arrange
`ments with sensors arranged along different axes or in differ
`ent locations, with an even or odd number of sensors may be
`within the scope of the present invention.
`0027. According to an embodiment of the invention, the
`microphones generally are positioned horizontally and sym
`metrically with respect to a vertical axis. In Such an arrange
`ment there are two sets of microphones, one on each side of
`the vertical axis corresponding to two separate channels, a left
`and right channel, for example.
`0028. In one embodiment, the microphones are digital
`microphones such as unior omni-directional electret micro
`phones, or micro machined microelectromechanical systems
`(MEMS) microphones. The advantage of using the MEMS
`microphones is they have silicon circuitry that internally con
`verts an audio signal into a digital signal without the need of
`an A/D converter, as other microphones would require. In any
`event, after the signals are digitized, according to an embodi
`ment of the present invention, the signals travel through
`adjustable delay lines that act as input into a microprocessor
`ora digital signal processor (DSP). The delay lines are adjust
`able, such that a user may control the direction in which the
`sensors or microphones receive sound signals or audio signals
`from, generally referred to hereinafter as a beam. In one
`embodiment, the delay lines are fed into the microprocessor
`of a computer. In such an embodiment, as well as others, there
`may be a graphical user interface (GUI) that provides feed
`back to a user. For example, the interface may tell the user
`how narrow the beam produced from the array, the direction
`of the beam, and how much sound it is picking up from a
`source. Based on input from a user of the electronic device
`containing the microphone array, the user may vary the delay
`lines that carry the output of the digitizer or digital micro
`phone to the microprocessor or DSP
`
`0029. The invention, according to one embodiment as pre
`sented in FIG. 1, produces substantial cancellation or reduc
`tion of background noise. After the steerable microphone
`array produces a two-channel input signal that may be digi
`tized 20 and on which beam steering may be applied 22, the
`output may then be transformed using a DFT 24. It well
`known there are many algorithms that can perform a DFT. In
`particular a fast Fourier transform (FFT) maybe used to effi
`ciently transform the data so that it may be more amenable for
`digital processing. As mentioned before, the DFT processing
`may take place in a general microprocessor, or a DSP. After
`transformation, the data may be filtered according to the
`embodiment of FIG. 1.
`0030 This invention, in particular, applies an adaptive
`filter in order to efficiently filter out background noise. The
`adaptive filter may be a mathematical transfer function. The
`filter coefficients of such adaptive filters help determine the
`performance of the adaptive filters. In the embodiment pre
`sented, the filter coefficients may be dependent on the past
`and present digital input.
`0031. An embodiment as shown in FIG. 1 discloses an
`averaging filter that may be applied to the digitally trans
`formed input 26 to Smooth the digital input and remove high
`frequency artifacts. This may be done for each channel. In
`addition the noise from each channel may also determined 28.
`Once the noise is determined, different variables may be
`calculated to update the adaptive filter coefficients 30. The
`channels are averaged and compared against a calibration
`threshold 32. If the result falls below a threshold, the values
`are adjusted, by a weighting average function so as to reduce
`distortion by a phase mismatch between the channels.
`0032. Another parameter that may be calculated, accord
`ing the embodiment in FIG. 1, is the signal to noise ratio
`(SNR). The SNR may be calculated from the averaging filter
`output and the noise calculated from each channel 34. The
`result of the SNR calculation, if it reaches a certain threshold,
`triggers modifying the digital input using the filter coeffi
`cients of the previously calculated beam. The threshold,
`which may be set by the manufacturer, may be a value in
`which the output may be sufficiently reliable for use in certain
`applications. In different situations or applications, a higher
`SNR may be desired, and the threshold may be adjusted by an
`individual.
`0033. The beam for each input may be continuously cal
`culated. A beam may be calculated as the average of the two
`signals from the left and right channels, the average including
`the difference of angle between the target source and each of
`channels. Along with the beam, a beam reference, reference
`average, and beam average may also calculated 36. The beam
`reference may be a weighted average of a previously calcu
`lated beam and the adaptive filter coefficients. A reference
`average may be the weighted Sum of the previously calculated
`beam references. Furthermore, there may also a calculation
`for beam average, which may be calculated as the running
`average of previously calculated beams. All these factors are
`used to update the adaptive filter.
`0034. Using the calculated beam and beam average, an
`error calculation may be performed by Subtracting the current
`beam from the beam average 42. This error may then used in
`conjunction with an updated reference average 44 and
`updated beam average 40 in a noise estimation calculation 46.
`The noise calculation helps predict the noise from the system
`including the filter. The noise prediction calculation may be
`
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`Oct. 29, 2009
`
`used in updating the coefficients of the adaptive filter 48 such
`as to minimize or eliminate potential noise.
`0035. After updating the filter and applying the digital
`input to it, the output of the filter may then be processed by an
`inverse discrete Fourier transform (IDFT). After the IDFT, the
`output then may be used in digital form as input into an audio
`application, such as, audio recording, VoIP speech recogni
`tion in the same computer, or perhaps sent as input to another
`computing system for additional processing.
`0036. According to another embodiment, the digital out
`put from the adaptive filter may be reconverted by a D/A
`converter into an analog signal and sent to an output device. In
`the case of an audio signal, the output from the filter may be
`sent as input to another computer or electronic device for
`processing. Or it may be sent to an acoustic device Such as a
`speaker system, or headphones, for example.
`0037. The algorithm, as disclosed herein, may advanta
`geously be able to produce an effective filtering of noise,
`including filtering of non-stationary or Sudden noise Such as a
`door slamming. Furthermore, the algorithm allows Superior
`filtering, at lower frequencies while also allowing the micro
`phone spacing Small, such as little as 5 inches in a two element
`microphone embodiment. Previously microphones array
`would require Substantially more amount of spacing, such as
`a foot or more to be able to have the same amount filtering at
`the lower frequencies.
`0038 Another advantage of the algorithm as presented is
`that it, for the most part, may require no customization for a
`wide range of different spacing between the elements in the
`array. The algorithm may be robust and flexible enough to
`automatically adjust and handle the spacing in a microphone
`array system to work in conjunction with common electronic
`or computer devices.
`0039 FIG. 2 shows a polar beam plot of a 2 member
`microphone array according to an embodiment of the inven
`tion when the delays lines of the left and right channels are
`equal. If the speakers are placed outside of the main beam, the
`array then attenuates signals originating from Such sources
`which lie outside of the main beam, and the microphone array
`acts as an echo canceller with there being no feedback distor
`tion. The beam typically will be focused narrowly on the
`target source, which is typically the human voice. When the
`target moves outside the beam width, the input of the micro
`phone array shows a dramatic decrease in signal strength.
`0040. A research study comparing Microsoft(R)'s micro
`phone array filters (embedded in the new Vista R) operating
`system) and the microphone array filter according to the
`present invention is discussed herein. The comparison was
`made by making a stereo recording using the Andrea R. Super
`beam array. This recording was then processed by both the
`Microsoft(R) filters and the microphone array filter according
`to the present invention using the exact same input, as shown
`in FIG. 3. The recording consisted of:
`0041
`1. A voice counting from 1 to 18, while moving in a
`180 degree arc in front of the array.
`0042. 2. A low level white noise generator was positioned
`at an angle of 45 degrees to the array.
`0043. 3. The recording was at a sampling rate of 8000 Hz,
`16-bit audio, which is the most common format used by VoIP
`applications.
`0044) For the Microsoft(R) filters test, their Beam Forming,
`Noise Suppression and Array Pre-Processing filters were
`
`turned on. For the instant filters test, the DSDA(RR3 and
`PureAudio(R) filters were turned on, thus given the best com
`parison of the two systems.
`004.5
`FIG. 4 shows the output wave files from both the
`filters. While the Microsoft(R) filters do improve the audio
`input quality, they use a loose beam forming algorithm. It was
`observed that it improves the overall voice quality, but it is not
`as effective as the instant filters, which are designed for envi
`ronments where a user wants all sound coming from the side
`removed, such as Voices or sound from multimedia speakers.
`The Microsoft(R) filters removed 14.9 dB of the stationary
`background noise (white noise), while the instant filters
`removed 28.6 dB of the stationary background noise. Also
`notable is that the instant beam forming filter has 29 dB more
`directional noise reduction of non-stationary noise (voice/
`music etc.) than the Microsoft(R) filters. The Microsoft(R) fil
`ters take a little more than a second before they start removing
`the stationary background noise. However, the instant filters
`start removing it immediately.
`0046. As shown in FIG. 4, the 12,000 mark on the axis
`represents when a target Source or input source is directly in
`front of the microphone array. The 10,000 and 14,000 marks
`correspond to the outer parts of the beam as shown in FIG. 2.
`FIG. 4 shows, for example, a comparison between the filter
`ing of Microsoft(R) array filter with an array filter disclosed
`according to an embodiment of the present invention. As soon
`as the target source falls outside of the beam width, or the
`10,000 or 14,000 marks, there is very noticeably and dramatic
`roll off in signal strength in the microphone array using an
`embodiment of the present invention. By contrast, there is no
`such roll off found in Microsoft(R) array filter.
`0047. As someone in the art would recognize, the inven
`tion as disclosed, the sensor array could be placed on or
`integrated within different types of devices such as any
`devices that requires or may use an audio input, like a com
`puter system, laptop, cellphone, gps, audio recorder, etc. For
`instance in a computer system embodiment, the microphone
`array may be integrated, wherein the signals from the micro
`phones are carried through delay lines directly into the com
`puter's microprocessor. The calculations performed for the
`algorithm described according to an embodiment described
`herein may take place in a microprocessor, Such as an Intel(R)
`Pentium(R) or AMDR Athlon R. Processor, typically used for
`personal computers. Alternatively the processing may be
`done by a digital signal processor (DSP). The microprocessor
`or DSP may be used to handle the user input to control the
`adjustable lines and the beam steering.
`0048 Alternatively in the computer system embodiment,
`the microphone array and possibly the delay lines may be
`connected, for example, to a USB input instead of being
`integrated with a computer system and connected directly to
`a microprocessor. In such an embodiment, the signals may
`then be routed to the microprocessor, or it maybe routed to a
`separate DSP chip that may also be connected to the same or
`different computer system for digital processing. The micro
`processor of the computer in Such an embodiment could still
`run the GUI that allows the user to control the beam, but the
`DSP will perform the appropriate filtering of the signal
`according to an embodiment of an algorithm presented
`herein.
`0049. In some embodiments, the spacing of the micro
`phones in the sensor array maybe adjustable. By adjusting the
`spacing, the directivity and beam width of the sensor may be
`modified. FIGS. 5 and 6 show different aspects of embodi
`
`Page 16 of 18
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`Oct. 29, 2009
`
`ments of the microphone array and different visual user inter
`faces or GUIs that may be used with the invention as dis
`closed. FIG. 6 is a portion of the visual interface as shown in
`FIG.S.
`0050. The invention according to a preferred embodiment
`may be an integrated headset system 200, a highly directional
`Stereo array microphone with reception beam angle pointed
`forward from the earphone to the corner of a user's mouth, as
`shown in FIG.8. The pick-up angles or the angles in which the
`microphones 250 pick up sound from a sound source 210 is
`shown in FIG. 9(d), for example, in front of the array, while
`cancellation of all sounds occurs from side and back direc
`tions. Different views of this pick-up area 220 are shown in
`FIGS. 9(a)-9(c). Cancellation is approximately 30 dB of
`noise, including speech noise.
`0051. According to this embodiment, left and right micro
`phones 250 are mounted on the lower font surface of the
`earphone 260. They are, preferably, placed on the same hori
`Zontal axis. The user's head may be centered between the two
`earphones 260 and act as additional acoustic separation of the
`microphone elements 250. The spacing of microphones may
`range anywhere from 5 to 7 inches, for example.
`0052 By adjusting the microphone 250 spacing, the beam
`width may be adjusted. The closer the microphones are, the
`wider the beam becomes. The farther apart the microphones
`are, the narrower the beam becomes. It is found that approxi
`mately 7 inches achieves a more narrow focus on to the corner
`of the user's mouth, however, other distances are within the
`scope of the instant invention. Therefore, any acoustic signals
`outside of the array microphones forward pick up angle are
`effectively cancelled.
`0053. The stereo microphone spacing allows for determin
`ing different time of arrival and direction of the acoustic
`signals to the microphones. From the centered position of the
`mouth, the voice signal 310 will look like a plain wave and
`arrive in-phase at same time with equal amplitude at both the
`microphones, while noise from the sides will arrive at each
`microphone in different phase/time and be cancelled by the
`adaptive processing of the algorithm. Illustration of Such an
`instance is clearly shown in FIG. 10, for example, where noise
`coming from a speaker 300 on one side of the user is cancelled
`due to varying distances (X, 2X) of the sound waves 290 from
`either microphone 250. However, the voice signal 310 travels
`an equidistant (Y) to both microphones 250, thus providing
`for a high fidelity far field noise canceling microphone that
`possesses good background noise cancellation and that may
`be used in any type of noisy environment, especially in envi
`ronments where a lot of music and speech may be present as
`background noise (as in a game arena or internet cafe).
`0054) The two elements or microphones 250 of the stereo
`headset-microphone array device may be mounted on the left
`and right earphones of any size/type of headphone. The
`microphones 250 may be protruding outwardly from the
`headphone, or may be adjustably mounted Such that the tip of
`the microphone may be moved closer to a user's mouth, or the
`distance thereof may be optimized to improve the sensitivity
`and minimize gain. Acoustic separation may be considered
`between the microphones and the output of the earphones, as
`not to allow the microphones to pickup much of the received
`playback audio (known as crosstalk or acoustic feedback).
`Any type of microphone may be used. Such as for example,
`uni-directional or omni-directional microphones.
`0055. The above described embodiment may be inexpen
`sively deployed because most of today’s PCs have integrated
`
`audio systems with stereo microphone input or utilize Blue
`tooth R) or a USB external sound card device. Behind the
`microphone input connector may be an analog to digital con
`verter (A/D Codec), which digitizes the left and right acoustic
`microphone signals. The digitized signals are then sent over
`the data bus and processed by the audio filter driver and
`algorithm by the integrated host processor. The algorithm
`used herein may be the same adaptive beam forming algo
`rithm as described in the previous embodiments of the inven
`tion. Once the noise component of the audio data is removed,
`clean audio/voice may then be sent to the preferred voice
`application for transmission.
`0056. This type of processing may be applied to a stereo
`array microphone system that may typically be placed on a
`PC monitor with distance of approximately 12-18 inches
`away from the user's the mouth. In the present invention,
`however, the same array System may be placed on the persons
`head to reduce the microphone sensitivity and points the two
`microphones in the direction of the person's mouth.
`0057 Although the embodiments described herein relate
`to a stereo headset, the scope of the invention is not limited
`thereto. The invention may be integrated into smaller devices
`Such as an ear bud, for example. The figures used herein are
`purely exemplary and are strictly provided to enable a better
`understanding of the invention. Accordingly, the present
`invention is not confined only to product designs illustrated
`therein.
`Accordingly, one embodiment of the present inven
`0.058
`tion may be a noise canceling audio transmitting/receiving
`device comprising at least one audio outputting component,
`and at least one audio receiving component, wherein each of
`the receiving means may be directly mounted on a surface of
`a corresponding outputting means. The noise canceling audio
`transmitting/receiving device may be a stereo headset or a ear
`budset. At least one audio outputting means may be a speaker,
`headphone, or an earphone, and at least one audio receiving
`means may be a microphone. The microphone may be a unior
`omni-directional electret microphone, or a microelectrome
`chanical systems (MEMS) microphone. The noise canceling
`audio transmitting/receiving device may also include a con
`necting means to connect to a computing device oran external
`device, and the noise canceling audio transmitting/receiving
`device may be connected to the computing device or the
`external device via a stereo speaker/microphone input or
`Bluetooth R) or a USB external sound card device. The posi
`tion of at least one audio receiving means may be adjustable
`with respect to a user's mouth.
`0059. Thus by the present invention its objects and advan
`tages are realized and although preferred embodiments have
`been disclosed and described in detail herein, its scope should
`not be limited thereby rather its scope should be determined
`by that of the appended claims.
`
`1. A noise canceling audio transmitting/receiving device,
`said device comprising:
`at least one audio outputting means; and
`at least one audio receiving means, each for receiving an
`acoustic signal and outputting an electrical signal repre
`senting the received acoustic signal;
`wherein each of said receiving means is directly mounted
`on a Surface of a corresponding outputting means and
`processing means connected to the audio receiving means
`and operable to apply a frequency domain adaptive filter
`
`Page 17 of 18
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`US 2009/0268931 A1
`
`Oct. 29, 2009
`
`to the electrical signal output by the audio receiving
`means, said processing means operable to carry out pro
`cessing comprising:
`applying an averaging filter on the digitized electrical
`signa