Canadian Patents Database / Patent 2798282 Summary

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(12) Patent Application: (11) CA 2798282
(54) English Title: WIND SUPPRESSION/REPLACEMENT COMPONENT FOR USE WITH ELECTRONIC SYSTEMS
(54) French Title: COMPOSANT DE SUPPRESSION/REMPLACEMENT DU VENT A UTILISER AVEC DES SYSTEMES ELECTRONIQUES
(51) International Patent Classification (IPC):
  • G10L 21/0208 (2013.01)
  • G10L 25/78 (2013.01)
  • G10L 19/012 (2013.01)
  • H04R 1/10 (2006.01)
  • H04R 3/04 (2006.01)
(72) Inventors :
  • PETIT, NICOLAS (United States of America)
  • BURNETT, GREGORY (United States of America)
  • GOERTZ, MICHAEL (United States of America)
(73) Owners :
  • ALIPHCOM (United States of America)
(71) Applicants :
  • ALIPHCOM (United States of America)
(74) Agent: CASSAN MACLEAN IP AGENCY INC.
(74) Associate agent: CASSAN MACLEAN IP AGENCY INC.
(45) Issued:
(86) PCT Filing Date: 2011-05-03
(87) Open to Public Inspection: 2011-11-10
Examination requested: 2016-05-03
(30) Availability of licence: N/A
(30) Language of filing: English

(30) Application Priority Data:
Application No. Country/Territory Date
12/772,963 United States of America 2010-05-03
12/772,975 United States of America 2010-05-03

English Abstract

Systems and methods to reduce the negative impact of wind on an electronic system include use of a first detector that receives a first signal and a second detector that receives a second signal. A voice activity detector (VAD) coupled to the first detector generates a VAD signal when the first signal corresponds to voiced speech. A wind detector coupled to the second detector correlates signals received at the second detector and derives from the correlation wind metrics that characterize wind noise that is acoustic disturbance corresponding to at least one of air flow and air pressure in the second detector. The wind detector controls a configuration of the second detector according to the wind metrics. The wind detector uses the wind metrics to dynamically control mixing of the first signal and the second signal to generate an output signal for transmission.


French Abstract

L'invention porte sur des systèmes et sur des procédés qui permettent de réduire l'impact négatif du vent sur un système électronique et qui comprennent l'utilisation d'un premier détecteur qui reçoit un premier signal et d'un second détecteur qui reçoit un second signal. Un détecteur d'activité vocale (VAD) couplé au premier détecteur génère un signal VAD lorsque le premier signal correspond à une parole voisée. Un détecteur de vent couplé au second détecteur établit une corrélation entre les signaux reçus au niveau du second détecteur et déduit de la corrélation des mesures de vent qui caractérisent le bruit du vent qui est une perturbation acoustique correspondant à un écoulement d'air et/ou une pression d'air dans le second détecteur. Le détecteur de vent commande une configuration du second détecteur selon les mesures de vent. Le détecteur de vent utilise les mesures de vent pour commander dynamiquement le mélange du premier signal et du second signal afin de générer un signal de sortie pour une transmission.


Note: Claims are shown in the official language in which they were submitted.




CLAIMS

What is claimed is:

1. A system comprising:
a first detector that receives a first signal and a second detector that
receives a second signal;
a voice activity detector (VAD) coupled to the first detector, the VAD
generating a VAD signal when the first signal corresponds to voiced speech;
and
a wind detector coupled to the second detector, wherein the wind
detector correlates signals received at the second detector and derives from
the
correlation a plurality of wind metrics that characterize wind noise that is
acoustic disturbance corresponding to at least one of air flow and air
pressure
in the second detector, wherein the wind detector controls a configuration of
the second detector according to the plurality of wind metrics, wherein the
wind
detector uses the plurality of wind metrics to dynamically control mixing of
the
first signal and the second signal to generate an output signal for
transmission.
2. The system of claim 1, wherein the first detector is a vibration sensor.

3. The system of claim 2, wherein the first detector is a skin surface
microphone (SSM).

4. The system of claim 2, wherein the second detector is an acoustic
sensor.

5. The system of claim 4, wherein the second detector comprises two
omnidirectional microphones.

6. The system of claim 5, wherein the two omnidirectional microphones are
positioned adjacent one another and are separated by a distance approximately
in a range of 10 millimeters (mm) to 40 mm.



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7. The system of claim 1, wherein the wind detector comprises an adaptive
filter coupled to the second detector, wherein the wind detector correlates
signals by calculating energy of an adaptive filter error.

8. The system of claim 7, wherein the wind detector comprises a first
exponential averaging filter and a second exponential averaging filter coupled

to the adaptive filter, wherein the wind detector applies the energy to the
first
exponential averaging filter and the second exponential averaging filter.

9. The system of claim 8, wherein the wind detector generates an
instantaneous wind level from the energy, wherein the instantaneous wind level

represents an instant wind level of the wind noise.

10. The system of claim 9, wherein the plurality of wind metrics comprise a
wind present metric that characterizes the instantaneous wind level relative
to
a present wind threshold over which the wind noise negatively affects
electronic
operations in a host electronic system.

11. The system of claim 9, wherein the plurality of wind metrics comprise a
wind mode metric that characterizes the instantaneous wind level relative to a

wind high threshold over which the wind noise is considered to have a
relatively
high impact on audio intelligibility in a host electronic system.

12. The system of claim 8, wherein the wind detector generates a current
wind level from the energy, wherein the current wind level represents an
average current wind level of the wind noise.

13. The system of claim 12, wherein the plurality of wind metrics comprise a
wind index metric that characterizes the current wind level relative to a
minimum wind threshold under which the wind noise is considered to have a
negligible impact on noise suppression and audio intelligibility in a host
electronic system.







14. The system of claim 1, wherein the plurality of wind metrics comprise a
wind mode metric that the wind detector generates to control the configuration

of the second detector, wherein the wind mode metric characterizes
instantaneous wind level relative to a wind high threshold over which the wind

noise is considered to have a relatively high impact on audio intelligibility
in a
host electronic system.

15. The system of claim 14, wherein, in response to the wind mode metric
indicating that instantaneous wind level exceeds the wind high threshold, the
wind detector controls the configuration of the second detector by controlling

generation of a summed detector signal by summing signals from each of two
microphones of the second detector.

16. The system of claim 15, wherein the wind detector controls the
configuration of the second detector by controlling application of single-
microphone noise suppression to the summed detector signal.

17. The system of claim 14, wherein the wind detector controls the
configuration of the second detector by controlling separate processing of
signals from each of two microphones of the second detector when the wind
mode metric indicates instantaneous wind level is below the wind high
threshold.

18. The system of claim 17, wherein the wind detector controls the
configuration of the second detector by controlling application of dual-
microphone noise suppression to the signals from the two microphones.

19. The system of claim 1, comprising a gain controller coupled to the first
detector and the wind detector.



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20. The system of claim 19, wherein the gain controller controls gain applied
to the first signal in response to the plurality of wind metrics and the VAD
signal.


21. The system of claim 20, wherein the gain controller adjusts a gain
applied to the first signal when the plurality of wind metrics indicates no
wind is
present.


22. The system of claim 20, wherein the plurality of wind metrics includes a
wind present metric that characterizes an instantaneous wind level derived
from the second signal relative to a present wind threshold over which the
wind
noise negatively affects electronic operations in a host electronic system.


23. The system of claim 20, comprising adjusting the gain to match a first
root mean square (RMS) of the first signal to a second RMS of a noise-
suppressed speech signal.


24. The system of claim 20, comprising generating a VAD signal when the
first signal corresponds to voiced speech, and using the VAD signal to noise
gate the first signal.


25. The system of claim 20, wherein the gain controller adjusts a gain
applied to the first signal when the VAD signal indicates the first signal
corresponds to voiced speech.


26. The system of claim 1, comprising a first filter coupled to the first
detector and a second filter coupled to the second detector.


27. The system of claim 26, wherein the first filter is a low-pass filter and
the
second filter is a high-pass filter.



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28. The system of claim 26, wherein the plurality of wind metrics dynamically
control mixing of the first signal and the second signal.

29. The system of claim 28, wherein the plurality of wind metrics dynamically
adjust a response of the first filter to which the first signal is applied and

dynamically adjust a response of the second filter to which the second signal
is
applied.

30. The system of claim 29, wherein the plurality of wind metrics comprise a
wind index metric that characterizes a current wind level relative to a
minimum
wind threshold under which the wind noise is considered to have a negligible
impact on noise suppression and audio intelligibility in a host electronic
system,
wherein the current wind level represents an average current wind level of the

wind noise.

31. The system of claim 30, comprising estimating a wind frequency
response of the wind noise from the wind index metric.

32. The system of claim 1, comprising a comfort equalizer coupled to the
second detector.

33. The system of claim 32, wherein the comfort equalizer generates a
comfort wind component and adds the comfort wind component to audio
signals, wherein the comfort wind component provides listener awareness of
wind presence.

34. The system of claim 33, wherein the comfort equalizer is coupled to a
transmitter, and adds the comfort wind component to audio signals processed
for transmission.

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35. The system of claim 33, wherein the comfort equalizer is coupled to a
receiver, and adds the comfort wind component to audio signals processed for
reception.

36. The system of claim 33, wherein the comfort equalizer generates the
comfort wind component by subtracting signals from each of two microphones
of the second detector to generate a difference signal.

37. The system of claim 36, comprising modulating the difference signal by a
gain to generate a modulated signal.

38. The system of claim 37, wherein the gain comprises a static gain that
provides an appropriate level of wind noise feedback in a loudspeaker.

39. The system of claim 38, wherein the gain comprises a gating factor
derived from a wind present metric output by the wind detector, wherein the
wind present metric characterizes an instantaneous wind level derived from the

second signal relative to a present wind threshold over which the wind noise
negatively affects electronic operations in a host electronic system.

40. The system of claim 37, comprising filtering the modulated signal to
provide the comfort wind component, the filtering comprising limiting an
amount of low-frequency wind noise and high-frequency wind noise.

41. A system comprising:
a first detector that receives a first signal and a second detector that
receives a second signal;
a voice activity detector (VAD) coupled to the first detector, the VAD
generating a VAD signal when the first signal corresponds to voiced speech;
and
a wind detector coupled to the second detector, wherein the wind
detector correlates signals received at the second detector and derives from
the



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correlation a plurality of wind metrics that characterize wind noise that is
acoustic disturbance corresponding to at least one of air flow and air
pressure
in the second detector, wherein the wind detector uses the plurality of wind
metrics to dynamically control mixing of the first signal and the second
signal to
generate an output signal for transmission.

42. A system comprising:
a first detector that receives a first signal and a second detector that
receives a second signal;
a voice activity detector (VAD) coupled to the first detector, the VAD
generating a VAD signal when the first signal corresponds to voiced speech;
and
a wind detector coupled to the second detector, wherein the wind
detector correlates signals received at the second detector and derives from
the
correlation a plurality of wind metrics that characterize wind noise that is
acoustic disturbance corresponding to at least one of air flow and air
pressure
in the second detector, wherein the wind detector controls a configuration of
the second detector according to the plurality of wind metrics.




Note: Descriptions are shown in the official language in which they were submitted.


CA 02798282 2012-11-02
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WIND SUPPRESSION/REPLACEMENT COMPONENT FOR USE WITH
ELECTRONIC SYSTEMS

10
RELATED APPLICATIONS
This application claims the benefit of United States (US) Patent
Application number 61/174,606, filed May 1, 2009.
This application is a continuation in part of US Patent Application number
12/139,333, filed June 13, 2008.
This application is a continuation in part of US Patent Application number
12/606,140, filed October 26, 2009.
This application is a continuation in part of US Patent Application number
11/805,987, filed May 25, 2007.
This application is a continuation in part of US Patent Application number
12/243,718, filed October 1, 2008.

TECHNICAL FIELD
The disclosure herein relates generally to noise suppression. In
particular, this disclosure relates to noise suppression systems, devices, and
methods for use in acoustic applications.

BACKGROUND
The ability to correctly identify voiced and unvoiced speech is critical to
many speech applications including speech recognition, speaker verification,
noise suppression, and many others. In a typical acoustic application, speech
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from a human speaker is captured and transmitted to a receiver in a different
location. In the speaker's environment there may exist one or more noise
sources that pollute the speech signal, the signal of interest, with unwanted
acoustic noise. This makes it difficult or impossible for the receiver,
whether
human or machine, to understand the user's speech. Typical methods for
classifying voiced and unvoiced speech have relied mainly on the acoustic
content of single microphone data, which is plagued by problems with noise and
the corresponding uncertainties in signal content. This is especially
problematic
with the proliferation of portable communication devices like mobile
telephones.
There are methods known in the art for suppressing the noise present in the
speech signals, but these generally require a robust method of determining
when speech is being produced.

INCORPORATION BY REFERENCE
Each patent, patent application, and/or publication mentioned in this
specification is herein incorporated by reference in its entirety to the same
extent as if each individual patent, patent application, and/or publication
was
specifically and individually indicated to be incorporated by reference.

BRIEF DESCRIPTION OF THE FIGURES
Figure 1 is a block diagram of the communications system, under an
embodiment.
Figure 2 is a block diagram of a wind detector, under an embodiment.
Figure 3 is a flow diagram for controlling processing of received signals
that include wind noise, under an embodiment.
Figure 4 is a low-pass wind detection filter response, under an
embodiment.
Figure 5 is the magnitude response of the SSM equalization filter, under
an embodiment.
Figure 6 is an example look-up table mapping wind index to cutoff
frequency, under an embodiment.

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Figure 7 is a filter response of a low-pass and corresponding high-pass
filter used in mixing SSM and microphone audio, under an embodiment.
Figure 8 is a magnitude response of a filter used to produce receive
wind comfort noise, under an embodiment.
Figure 9 is a magnitude response of a filter used to produce transmit
wind comfort noise, under an embodiment.
Figure 10 is an example plot comparing the speech response of a
system with no wind, with 10 mph wind, and with 10 mph wind and wind
suppression, under an embodiment.
Figure 11 is a two-microphone adaptive noise suppression system,
under an embodiment.
Figure 12 is an array and speech source (S) configuration, under an
embodiment. The microphones are separated by a distance approximately
equal to 2do, and the speech source is located a distance ds away from the
midpoint of the array at an angle 0. The system is axially symmetric so only
ds
and 0 need be specified.
Figure 13 is a block diagram for a first order gradient microphone using
two omnidirectional elements 01 and 02, under an embodiment.
Figure 14 is a block diagram for a DOMA including two physical
microphones configured to form two virtual microphones V1 and V2, under an
embodiment.
Figure 15 is a block diagram for a DOMA including two physical
microphones configured to form N virtual microphones V1 through VN, where N
is any number greater than one, under an embodiment.
Figure 16 is an example of a headset or head-worn device that includes
the DOMA, as described herein, under an embodiment.
Figure 17 is a flow diagram for denoising acoustic signals using the
DOMA, under an embodiment.
Figure 18 is a flow diagram for forming the DOMA, under an
embodiment.

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Figure 19 is a plot of linear response of virtual microphone V2 to a 1 kHz
speech source at a distance of 0.1 m, under an embodiment. The null is at 0
degrees, where the speech is normally located.
Figure 20 is a plot of linear response of virtual microphone V2 to a 1 kHz
noise source at a distance of 1.0 m, under an embodiment. There is no null
and all noise sources are detected.
Figure 21 is a plot of linear response of virtual microphone V1 to a 1 kHz
speech source at a distance of 0.1 m, under an embodiment. There is no null
and the response for speech is greater than that shown in Figure 19.
Figure 22 is a plot of linear response of virtual microphone V1 to a 1 kHz
noise source at a distance of 1.0 m, under an embodiment. There is no null
and the response is very similar to V2 shown in Figure 20.
Figure 23 is a plot of linear response of virtual microphone V1 to a
speech source at a distance of 0.1 m for frequencies of 100, 500, 1000, 2000,
3000, and 4000 Hz, under an embodiment.
Figure 24 is a plot showing comparison of frequency responses for
speech for the array of an embodiment and for a conventional cardioid
microphone.
Figure 25 is a plot showing speech response for V1 (top, dashed) and V2
(bottom, solid) versus B with d, assumed to be 0.1 m, under an embodiment.
The spatial null in V2 is relatively broad.
Figure 26 is a plot showing a ratio of V1/V2 speech responses shown in
Figure 10 versus B, under an embodiment. The ratio is above 10 dB for all 0.8
< B < 1.1. This means that the physical p of the system need not be exactly
modeled for good performance.
Figure 27 is a plot of B versus actual ds assuming that ds = 10 cm and
theta = 0, under an embodiment.
Figure 28 is a plot of B versus theta with ds = 10 cm and assuming d,
=
10 cm, under an embodiment.
Figure 29 is a plot of amplitude (top) and phase (bottom) response of
N(s) with B = 1 and D = -7.2 psec, under an embodiment. The resulting phase
difference clearly affects high frequencies more than low.

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Figure 30 is a plot of amplitude (top) and phase (bottom) response of
N(s) with B = 1.2 and D = -7.2 psec, under an embodiment. Non-unity B
affects the entire frequency range.
Figure 31 is a plot of amplitude (top) and phase (bottom) response of
the effect on the speech cancellation in V2 due to a mistake in the location
of
the speech source with q1 = 0 degrees and q2 = 30 degrees, under an
embodiment. The cancellation remains below -10 dB for frequencies below 6
kHz.
Figure 32 is a plot of amplitude (top) and phase (bottom) response of
the effect on the speech cancellation in V2 due to a mistake in the location
of
the speech source with q1 = 0 degrees and q2 = 45 degrees, under an
embodiment. The cancellation is below -10 dB only for frequencies below about
2.8 kHz and a reduction in performance is expected.
Figure 33 shows experimental results for a 2do = 19 mm array using a
linear R of 0.83 on a Bruel and Kjaer Head and Torso Simulator (HATS) in very
loud ("85 dBA) music/speech noise environment, under an embodiment. The
noise has been reduced by about 25 dB and the speech hardly affected, with no
noticeable distortion.
Figure 34 is a configuration of a two-microphone array with
speech source S, under an embodiment.
Figure 35 is a block diagram of V2 construction using a fixed R(z),
under an embodiment.
Figure 36 is a block diagram of V2 construction using an adaptive
13(z), under an embodiment.
Figure 37 is a block diagram of Vi construction, under an
embodiment.
Figure 38 is a flow diagram of acoustic voice activity detection,
under an embodiment.
Figure 39 shows experimental results of the algorithm using a
fixed beta when only noise is present, under an embodiment.

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Figure 40 shows experimental results of the algorithm using a
fixed beta when only speech is present, under an embodiment.
Figure 41 shows experimental results of the algorithm using a
fixed beta when speech and noise is present, under an embodiment.
Figure 42 shows experimental results of the algorithm using an
adaptive beta when only noise is present, under an embodiment.
Figure 43 shows experimental results of the algorithm using an
adaptive beta when only speech is present, under an embodiment.
Figure 44 shows experimental results of the algorithm using an
adaptive beta when speech and noise is present, under an embodiment.
Figure 45 is a block diagram of a NAVSAD system, under an embodiment.
Figure 46 is a block diagram of a PSAD system, under an embodiment.
Figure 47 is a block diagram of a denoising system, referred to herein as the
Pathfinder system, under an embodiment.
Figure 48 is a flow diagram of a detection algorithm for use in detecting
voiced and unvoiced speech, under an embodiment.
Figure 49A plots the received GEMS signal for an utterance along with the
mean correlation between the GEMS signal and the Mic 1 signal and the
threshold
for voiced speech detection.
Figure 49B plots the received GEMS signal for an utterance along with the
standard deviation of the GEMS signal and the threshold for voiced speech
detection.
Figure 50 plots voiced speech detected from an utterance along with the
GEMS signal and the acoustic noise.
Figure 51 is a microphone array for use under an embodiment of the PSAD
system.
Figure 52 is a plot of AM versus di for several Ad values, under an
embodiment.
Figure 53 shows a plot of the gain parameter as the sum of the absolute
values of H,(z) and the acoustic data or audio from microphone 1.
Figure 54 is an alternative plot of acoustic data presented in Figure 53.
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Figure 55 is a cross section view of an acoustic vibration sensor, under
an embodiment.
Figure 56A is an exploded view of an acoustic vibration sensor, under
the embodiment of Figure 55.
Figure 56B is perspective view of an acoustic vibration sensor, under
the embodiment of Figure 55.
Figure 57 is a schematic diagram of a coupler of an acoustic vibration
sensor, under the embodiment of Figure 55.
Figure 58 is an exploded view of an acoustic vibration sensor, under an
alternative embodiment.
Figure 59 shows representative areas of sensitivity on the human head
appropriate for placement of the acoustic vibration sensor, under an
embodiment.
Figure 60 is a generic headset device that includes an acoustic vibration
sensor placed at any of a number of locations, under an embodiment.
Figure 61 is a diagram of a manufacturing method for an acoustic
vibration sensor, under an embodiment.

DETAILED DESCRIPTION
Systems and methods to reduce the negative impact of wind on a
communications headset are described below. The communications headset
example used is the Jawbone Prime Bluetooth headset, produced by AliphCom
in San Francisco, CA. This headset uses two omnidirectional microphones to
form two virtual microphones using the system described below (see section
"Dual Omnidirectional Microphone Array (DOMA)" below) as well as a third
vibration sensor to detect human speech inside the cheek on the face of the
user. Although the cheek location is preferred, any sensor that is capable of
detecting vibrations reliably (such as an accelerometer or radiovibration
detector (see section "Detecting Voiced and Unvoiced Speech Using Both
Acoustic and Nonacoustic Sensors" below) can be used as. well. Any italicized
text herein generally refers to the name of a variable in an algorithm
described
herein.

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In the following description, numerous specific details are introduced to
provide a thorough understanding of, and enabling description for,
embodiments. One skilled in the relevant art, however, will recognize that
these embodiments can be practiced without one or more of the specific
details,
or with other components, systems, etc. In other instances, well-known
structures or operations are not shown, or are not described in detail, to
avoid
obscuring aspects of the disclosed embodiments.
Unless specifically stated, the following acronyms and terms are defined
as follows.
The term ADC represents analog to digital converter.
The term AEC represents acoustic echo cancellation.
The term DAC represents digital to analog converter.
The term EQ represents equalization, generally in terms of frequency.
Microphone is a physical acoustic sensing element.
Normalized Least Mean Square (NLMS) adaptive filter is a common
adaptive filter used to determine correlation between the microphone signals.
Any similar adaptive filter may be used.
The term 01 represents the first physical omnidirectional microphone
The term 02 represents the second physical omnidirectional microphone
Skin Surface Microphone (SSM) is a microphone adapted to detect
human speech on the surface of the skin (see section "Acoustic Vibration
Sensor" below). Any similar sensor that is capable of detecting speech
vibrations in the skin of the user yet immune to wind noise can be
substituted.
The term VAD represents Voice Activity Detection, and may be used as
the name of an algorithm or as a signal, depending on the context.
Virtual microphone is a microphone signal comprised of combinations of
physical microphone signals.
Wind is the movement of air.
Wind comfort noise is wind or wind-like noise that is included in either
the transmitted or received signal to alert the user and the person to whom
they are speaking to the presence of wind without unduly affecting the
communication intelligibility.

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Wind noise is unwanted acoustic disturbances from air pressure and/or
air flow in the microphone signal of interest.
In the following description, numerous specific details are introduced to
provide a thorough understanding of, and enabling description for,
embodiments. One skilled in the relevant art, however, will recognize that
these embodiments can be practiced without one or more of the specific
details,
or with other components, systems, etc. In other instances, well-known
structures or operations are not shown, or are not described in detail, to
avoid
obscuring aspects of the disclosed embodiments.
Generally, the enclosed wind solution takes advantage of the SSM
transducer's high resistance to wind noise and acoustic noise and its
relatively
acceptable speech fidelity below 1 kHz in windy conditions. Figure 1 is a
block
diagram of the communications system, under an embodiment.
The system of an embodiment generally comprises five system
components including, but not limited to, wind detection, SSM equalization,
wind mode audio, dynamic mixing, and comfort wind. Each of these system
components is described in detail below.
The detection of wind presence and intensity in either or both
microphones is used to achieve good wind mitigation. The wind detection
algorithm takes advantage of the fact that the wind noises in each of the
microphones are uncorrelated. Indeed, since wind physically displaces air
molecules when it flows, it independently moves the diaphragm of each
microphone in a non-correlated fashion. Even acoustic wind noise (caused by
turbulence near the microphone) is not highly correlated. Thus, the effect on
the microphone from wind is chaotic and non-linear. It is true that the
intensities of wind in each microphone are slightly correlated, but their
waveforms cannot be easily represented by linear transfer functions even if
the
microphones are only a few millimeters apart.
Consequently a linear adaptive filter can be used to monitor the degree
of correlation (or lack thereof) between the two microphone signals by
measuring the energy of the adaptive filter error from which a metric of wind
intensity can be derived. A Normalized Least Mean Square (NLMS) or similar

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adaptive filter is used to carry out this task. Figure 2 is a block diagram of
a
wind detector, under an embodiment.
More specifically, the system of an embodiment includes a first detector
that receives a first signal and a second detector that receives a second
signal.
A voice activity detector (VAD) is coupled to the first detector. The VAD
generates a VAD signal when the first signal corresponds to voiced speech. The
system includes a wind detector coupled to the second detector. The wind
detector correlates signals received at the second detector and derives or
generates from the correlation wind metrics that characterize wind noise that
is
acoustic disturbance corresponding to at least one of air flow and air
pressure
in the second detector. The wind metrics are used as control signals as
described in detail herein. For example, the wind detector controls a
configuration of the second detector according to the wind metrics, by using
the
wind metrics to dynamically control mixing of the first signal and the second
signal to generate an output signal for transmission.
The wind detector comprises an adaptive filter coupled to the second
detector. The wind detector correlates signals by calculating energy of an
adaptive filter error of the adaptive filter. The error is large when the
signals
are uncorrelated, which is the case for wind noise. Normal acoustic speech and
noise are highly correlated between the microphones, which are typically 10-40
mm apart from one another. The wind detector of an embodiment comprises a
first exponential averaging filter and a second exponential averaging filter
coupled to the adaptive filter. The wind detector applies the energy to the
first
exponential averaging filter and the second exponential averaging filter. The
system of an embodiment comprises a gain controller coupled to the first
detector and the wind detector.
Figure 3 is a flow diagram for controlling processing of received signals
that include wind noise 300, under an embodiment. The signal processing
receives a first signal at a first detector and a second signal at a second
detector 302. A correlation is determined between signals received at the
second detector, and wind metrics are derived from the correlation that
characterize wind noise that is acoustic disturbance corresponding to at least



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one of air flow and air pressure in the second detector 304. An embodiment
controls configuration of the second detector according to the wind metrics
306.
An embodiment generates an output signal for transmission by dynamically
mixing the first signal and the second signal according to the wind metrics
308.
In the description below, many variable values are presented. The
algorithm of an embodiment does not require that the variables take on exactly
the values presented, and some variation is allowable such that the
embodiments are not restricted to the values presented herein.
Prior to NLMS adaptive filtering, a fifth order low-pass Butterworth filter
with a cutoff frequency of 40Hz is used on both microphone signals. Figure 4
is a low-pass wind detection filter response, under an embodiment. This region
is normally dominated by wind (and not acoustic) noise, making detection more
accurate and robust.
The filtered signals are then decimated by a factor of 21 to limit the LMS
computational workload and provide faster adaptation with fewer adaptive taps
on the "whiter" decimated signals. Note that the reference LMS signal (in this
case low-passed and decimated O1) is delayed prior to calculating the error to
allow causal models of sound from all angles of incidence on the microphone
array. For this LMS filter, a delay of 2 (decimated) samples and 7 adaptive
LMS
taps are used.
The energy of the LMS residual is then calculated. It is sent to two
smoothing exponential averaging filters E1(z) and E2(z):

1 -
E1(z) a1
_ Gi- )
a1z-1
G 1 - a2
E2(z) =
a2z-1I
- I

where a1 = 0.895 and a2 = 0.97375 so that the time constant of El using an 8
kHz sampling rate decimated by 21 is 25 msec and E2 is 100 msec. The time
constants can be varied somewhat, but El should react significantly faster
than
E2. The output of El yields a variable called instantWindLevel and the output
of E2 variable currentWindLevel. For the sake of clarity, these variables will
be
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valued in dBFS; that is measured in decibels relative to a full scale residual
input (0dB).
A first output variable windlndex of this module is obtained by
subtracting the minWindLevel threshold (-86 dBFS in this enablement) from
currentWindLevel and limiting its values to the 0-30dB range. This variable is
used later as a metric of the wind level compared to a minimum level under
which wind is considered to have negligible impact on noise suppression,
intelligibility, and listening experience.
Another output variable windPresent (binary) is obtained by comparing
instantWindLevel to a windPresentThreshold (e.g., -74 dBFS) constant
threshold yielding a binary variable equal to 1 when the variable exceeds the
threshold. The binary variable is then followed by a hold block that maintains
a
binary output of 1 for 20 msec whenever the input is 1. This variable is used
by other components of the system to suspend activities that would otherwise
be negatively affected in the presence of wind.
A final binary variable windMode is produced by comparing
instantWindLevel to windHighLevel, a constant threshold of -69 dBFS in this
enablement over which wind is deemed to have high impact on intelligibility
and
comfort. The resulting binary output is then filtered by a 2.5 second moving
average filter whose output effectively indicates the portion of time that was
windy during the last 2.5 seconds. When windMode=O, this percentage is
compared against a 35% threshold. If exceeded, windMode is switched to 1.
When windMode=1, this percentage is compared against an 8% threshold. If
under, windMode is switched to 0. This hysteresis approach prevents windMode
from rapidly switching states and is used in lieu of windPresent in scenarios
where such fast changes are undesirable.
Turning to the SSM equalization component of an embodiment, in order
to substitute SSM audio for microphone audio in the transmitted audio
spectrum, the SSM level and its frequency response should be adjusted to
match the transmit audio in the absence of wind as closely as possible. The
SSM or similar signal captured at the skin is filtered in order to match as
closely
as possible speech captured by the primary microphone (01) in the absence of
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wind or noise. Unfortunately, this technique cannot be used to extract high-
fidelity speech for several reasons. One reason is that SSM audio has too
little
speech content beyond 1 kHz, where it is near or under the sensor's noise
floor.
Another reason is that there is not a unique transfer function mapping SSM to
01 responses for all phonemes; the transform is specific to each phoneme. Yet
another reason is that the SSM response to speech is affected by facial
features
near the SSM pickup location, specifically the layer of soft tissues
separating
SSM from cheekbone. As a result, mic-to-SSM response (even for stationary
speech production) varies across users.
For these reasons, the optimal SSM-microphone speech response
transform can only be approximated. In this implementation, it is achieved in
two consecutive stages: a one-size-fit-all static equalization filter and an
adaptive gain control stage (AGC) to match the RMS of regular speech.
Figure 5 is the magnitude response of the SSM equalization filter, under
an embodiment. It is implemented as a cascade of 3 biquad IIR filters, but is
not so limited. The filter attempts to match the responses up to about 1 kHz
where the SSM speech response becomes too small. The region beyond 1kHz is
treated as a stop-band and filtered out to avoid amplifying what is mostly
acoustic noise and sensor self-noise.
The subsequent AGC stage adjusts its gain to match the root mean
square (RMS) of the equalized SSM signal to the RMS of the noise-suppressed
speech from 0 to 1 kHz. The gain adjustment only occurs when two conditions
are met. The first condition is no wind present, as indicated by windPresent.
The second condition involves speech activity. A conservative VAD is used
indicating speech activity with high level of confidence at the expense of
potentially many false negatives. The idea is that the AGC gain should not be
adapted when speech is not occurring. Also, short-lived VAD pulses of less
than 60ms are rejected from this binary VAD waveform. Furthermore to
increase robustness, the AGC gain is limited to a +/ 25dB range.
Finally the conservative, pulse-rejected VAD used above is also used to
noise gate the SSM audio before mixing. This eliminates the static noise that
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would otherwise be perceived when no speech is present in the SSM. This
noise gate reduces noise during no-speech sections by 15 dB.
The omnidirectional microphone array used in Jawbone Prime (DOMA)
(see section "Dual Omnidirectional Microphone Array (DOMA)" below) provides
relatively good noise suppression performance. Unfortunately, due to the
combination of omnidirectional microphones to form virtual microphones, it is
more sensitive to wind than a single omnidirectional microphone. Therefore, in
sustained wind conditions it can be preferable to switch to a microphone array
configuration that presents better wind immunity even at the expense of lower
noise suppression performance.
If the wind detector reports sufficient sustained wind, the wind detector's
windMode variable turns on. At this time, the benefit of further reducing the
wind in the microphones outweighs the noise reduction advantage offered by
the microphone array. In wind mode, the DOMA array is therefore turned off
and the noise suppression algorithm bypassed. However, instead of using
audio from a single microphone, a simple way to further reduce the wind-to-
speech ratio is to add the signals from both omnidirectional microphones
together. While the resulting speech content increases by 6dB, the wind RMS is
only increased by about 3dB as wind signals in 01 and 02 are uncorrelated.
Note that 01 audio is delayed by a fractional delay that accounts for the
travel
time of speech from 01 to 02 plus any ADC sampling time difference between
02 and 01 audio channels. Naturally, the resulting signal needs to be scaled
by
a correction gain factor to match the speech response outside wind mode.
A problem with this technique is the absence of any noise suppression.
To mitigate this performance drop, a basic single-microphone noise suppression
algorithm such as spectral subtraction is used to attenuate stationary noise
in
16 frequency bands evenly distributed across the spectrum (0-4kHz). These
algorithms work by selectively attenuating bands where speech-to-noise (and -
wind) ratio is lower than 12dB. The maximum attenuation used here is 8dB
when the SNR is lower than 3dB.
Regarding the dynamic mixing component of an embodiment, there is an
optimal mix of low-passed SSM and high-passed microphone audio (DOMA or
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Wind Mode Audio) that can be achieved for each level of wind intensity. As the
wind continuously changes, the mixer adjusts (dynamic) filters' responses to
obtain the desired mix. Now that SSM and microphone audio signals have been
processed in previous stages, they must be combined in a seamless fashion for
varying amount of wind. The technique used here relies on the observation
that the microphones' responses to wind drops as the frequency increases. In
fact, the wind detector windlndex variable provides a rough but reliable
metric
of the amount of wind at any time from which an estimate of the wind
frequency response can be derived. Another important characteristic of the
wind frequency response curve is that it tends to decrease with frequency at a
constant rate through the spectrum and for varying level of winds until the
wind
response eventually reaches the noise floor. Note however that this is true
only
when wind is moderate enough such as not to saturate the microphone(s)
and/or ADC converter(s).
In very low or no-wind conditions (windlndex = 0dB), no SSM mixing is
required. In very high wind conditions (windlndex = 30dB), only equalized
SSM audio is used for frequencies up to 1 kHz. In-between these two
extremes, a set of two dynamic filters is used to high-pass the microphone
signal (either DOMA/noise-suppressed or wind mode output) and low-pass the
equalized SSM audio.
For Jawbone Prime, advantage is taken of the fact that noise-suppressed
or wind mode audio is available in 16-band frequency analysis format to
implement the low- and high-pass filters as sub-band equalizers characterized
by one real weight per sub-band. A look-up table is used to find the starting
band index B corresponding to the -70dB stop-band of the filter used to low-
pass the SSM. Figure 6 is an example look-up table mapping wind index to
cutoff frequency, under an embodiment. This index is then used to retrieve the
gain GLP used to multiply each of the bands, for example:

SSM band index GLP
<_ B-4 0dB
B - 4 -5dB


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B - 3 -12dB
B-2 -22dB
B - 1 -35dB
B -70dB

The high-pass equalizer weights GHP used for the microphone audio are
obtained by calculating

GLP
GHP=20*loglo(1-1020)
for each band. For a full-band implementation, a 32-tap linear-phase low-pass
FIR filter can be stored in memory and retrieved for each of the 31 wind
indexes and the corresponding high-pass filter can be derived by subtracting 1
from the central tap. Figure 7 is a filter response of a low-pass and
corresponding high-pass filter used in mixing SSM and microphone audio for
windlndex = 12dB, under an embodiment.
Regarding the comfort wind noise component of an embodiment, a
limited amount of wind noise is added to both receive and transmit audio to
increase both near- and far-end users' awareness to wind impact on the
conversation with little negative effect on intelligibility and listening
comfort. A
complementary approach to wind reduction is to get the near-end user to take
a pro-active role in limiting wind exposure. By adding a limited amount of
wind
noise to the receive audio in the form of a side tone, the user will tend to
subconsciously change his position relative to the wind to minimize the
feedback in the speaker.
Generation of the comfort wind noise of an embodiment begins by
subtracting 02 from 01. This reduces much of the non-wind components of the
signal. This difference is modulated by a gain that combines two factors. The
first factor is a static gain to guarantee an appropriate level of wind
feedback in
the speaker. The second factor is a gating factor derived from the binary
windPresent variable going through a binary pulse-rejection block rejecting
positive pulses shorter than 20ms followed by a hold block with a hold
duration
of 10ms. Upon modulation of the signal, a filter is applied that limits the

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amount of low-frequency wind reaching the headset receiver (whose low-
frequency response is poor given its small size) and to scale down the higher
frequency content of wind that can be responsible for uncomfortable noise in
high wind. The resulting signal is designed to sound like a rumble
characteristic
of wind heard through speakers without overdriving the receiver.
Figure 8 is a magnitude response of a filter used to produce receive
wind comfort noise, under an embodiment. Note that this filter was designed
based on the specific characteristics of the microphones and speaker used in
Jawbone Prime and there may be some changes required for different
implementations. The important part is to add enough wind noise to be audible
to the headset user but not enough to disrupt the conversation.
Likewise, transmit comfort wind noise is added to the transmitted audio
to alert the far-end user that wind is present, providing an explanation for
the
difference in speech response due to SSM mixing and/or degradation of noise
suppression performance. However, due to differences in Bluetooth
transmission and phone/network responses, a number of changes are made. A
first change is the use of a different static gain to ensure appropriate level
of
wind feedback on the other end of the line; this gain is set experimentally.
Another change is the use of a different filter, where Figure 9 is a magnitude
response of a filter used to produce transmit wind comfort noise, under an
embodiment. In addition, the resulting signal is delayed so as to be
synchronous with the processed transmit audio it is added to before
transmission.
As an example of performance obtained under an embodiment, Figure
10 shows a plot of a male English speaker speaking in silence (left), in a
moderate wind (10 mph, center), and in the same wind with the wind
suppression algorithm active (right), under an embodiment. The top is the
time series, the middle the spectrogram, and the bottom the energy vs. time.
Clearly the wind overwhelms the vast majority of the speech, significantly
disrupting the quality and intelligibility. The wind suppression algorithm
significantly reduces the wind noise and restores the speech quality and
intelligibility.

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Systems and methods of wind noise detection, suppression, and speech
replacement using data from a vibration sensor have been presented. The
embodiments described herein take advantage of the vibration sensor's wind
immunity and acoustic noise resistance to not only remove wind noise, but
restore a significant amount of speech presence and intelligibility. The
method
of wind suppression involves an adaptive, filtered combination of vibration
sensor signal, combined omnidirectional signal, and normal virtual microphone
noise suppressed signal. This allows for achievement of significant wind
reduction with limited speech distortion despite the severe impact of wind on
the microphone signals.

DUAL OMNIDIRECTIONAL MICROPHONE ARRAY (DOMA)
A dual omnidirectional microphone array (DOMA) that provides improved
noise suppression is described herein. Compared to conventional arrays and
algorithms, which seek to reduce noise by nulling out noise sources, the array
of an embodiment is used to form two distinct virtual directional microphones
which are configured to have very similar noise responses and very dissimilar
speech responses. The only null formed by the DOMA is one used to remove
the speech of the user from V2. The two virtual microphones of an embodiment
can be paired with an adaptive filter algorithm and/or VAD algorithm to
significantly reduce the noise without distorting the speech, significantly
improving the SNR of the desired speech over conventional noise suppression
systems. The embodiments described herein are stable in operation, flexible
with respect to virtual microphone pattern choice, and have proven to be
robust
with respect to speech source-to-array distance and orientation as well as
temperature and calibration techniques.
In the following description, numerous specific details are introduced to
provide a thorough understanding of, and enabling description for,
embodiments of the DOMA. One skilled in the relevant art, however, will
recognize that these embodiments can be practiced without one or more of the
specific details, or with other components, systems, etc. In other instances,

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well-known structures or operations are not shown, or are not described in
detail, to avoid obscuring aspects of the disclosed embodiments.
Unless otherwise specified, the following terms have the corresponding
meanings in addition to any meaning or understanding they may convey to one
skilled in the art.
The term "bleedthrough" means the undesired presence of noise during
speech.
The term "denoising" means removing unwanted noise from Micl, and
also refers to the amount of reduction of noise energy in a signal in decibels
(dB).
The term "devoicing" means removing/distorting the desired speech from
Micl.
The term "directional microphone (DM)" means a physical directional
microphone that is vented on both sides of the sensing diaphragm.
The term "Mica (Ml)" means a general designation for an adaptive noise
suppression system microphone that usually contains more speech than noise.
The term "Mic2 (M2)" means a general designation for an adaptive noise
suppression system microphone that usually contains more noise than speech.
The term "noise" means unwanted environmental acoustic noise.
The term "null" means a zero or minima in the spatial response of a
physical or virtual directional microphone.
The term "01" means a first physical omnidirectional microphone used to
form a microphone array.
The term "02" means a second physical omnidirectional microphone used
to form a microphone array.
The term "speech" means desired speech of the user.
The term "Skin Surface Microphone (SSM)" is a microphone used in an
earpiece (e.g., the Jawbone earpiece available from Aliph of San Francisco,
California) to detect speech vibrations on the user's skin.
The term "V1" means the virtual directional "speech" microphone, which
has no nulls.

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The term "V2" means the virtual directional "noise" microphone, which
has a null for the user's speech.
The term "Voice Activity Detection (VAD) signal" means a signal
indicating when user speech is detected.
The term "virtual microphones (VM)" or "virtual directional microphones"
means a microphone constructed using two or more omnidirectional
microphones and associated signal processing.
Figure 11 is a two-microphone adaptive noise suppression system 1100,
under an embodiment. The two-microphone system 1100 including the
combination of physical microphones MIC 1 and MIC 2 along with the
processing or circuitry components to which the microphones couple (described
in detail below, but not shown in this figure) is referred to herein as the
dual
omnidirectional microphone array (DOMA) 1110, but the embodiment is not so
limited. Referring to Figure 11, in analyzing the single noise source 1101 and
the direct path to the microphones, the total acoustic information coming into
MIC 1 (1102, which can be an physical or virtual microphone) is denoted by
mi(n). The total acoustic information coming into MIC 2 (1103, which can also
be an physical or virtual microphone) is similarly labeled m2(n). In the z
(digital frequency) domain, these are represented as M1(z) and M2(z). Then,

M, (z) = S(z) + N 2 (z)
M2(z)=N(z)+S2(Z)
with

N 2 (z) = N(z)H 1(z)
,
S2(z)=S(z)H2(Z)
so that
M1(z) = S(z) + N(z)H 1(z)
M 2 (z) = N(z) + S(z)H 2 (z) . Eq. 1
This is the general case for all two microphone systems. Equation 1 has four
unknowns and only two known relationships and therefore cannot be solved
explicitly.
However, there is another way to solve for some of the unknowns in
Equation 1. The analysis starts with an examination of the case where the


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speech is not being generated, that is, where a signal from the VAD subsystem
1104 (optional) equals zero. In this case, s(n) = S(z) = 0, and Equation 1
reduces to

MIN (z)=N(z)H, (z)
M2N(z)=N(z),
where the N subscript on the M variables indicate that only noise is being
received. This leads to

MIN (z) = M2N (z)HI (z)

H I (z) = M N ~Z) . Eq. 2
The function H1(z) can be calculated using any of the available system
identification algorithms and the microphone outputs when the system is
certain that only noise is being received. The calculation can be done
adaptively, so that the system can react to changes in the noise.
A solution is now available for H1(z), one of the unknowns in Equation 1.
The final unknown, H2(z), can be determined by using the instances where
speech is being produced and the VAD equals one. When this is occurring, but
the recent (perhaps less than 1 second) history of the microphones indicate
low
levels of noise, it can be assumed that n(s) = N(z) - 0. Then Equation 1
reduces to

MIS(z)=S(z)
M 2S (z)= S(z)H2 (z) ,
which in turn leads to

M2s (z)=Mis (z)H2 (z)
H 2 (z) = M 2s (z)
MIS (z)

which is the inverse of the H1(z) calculation. However, it is noted that
different
inputs are being used (now only the speech is occurring whereas before only
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the noise was occurring). While calculating H2(z), the values calculated for
H1(z) are held constant (and vice versa) and it is assumed that the noise
level
is not high enough to cause errors in the H2(z) calculation.
After calculating H1(z) and H2(z), they are used to remove the noise from
the signal. If Equation 1 is rewritten as

S(z) = M 1(z) - N(z)H 1(z)
N(z)=M2 (z)-S(z)H2 (z)
S(z) = Ml (z) - [M 2 (z) - S(z)H 2 (z)]H 1(z)
S(z)[ 1- H 2 (z)H 1(z)] = M1(z) - M2 (z)H 1(z) ,
then N(z) may be substituted as shown to solve for S(z) as
S(z)=MI(z)-M2(z)Hl(z) Eq. 3
1-H1(z)H2 (z)

If the transfer functions H1(z) and H2(z) can be described with sufficient
accuracy, then the noise can be completely removed and the original signal
recovered. This remains true without respect to the amplitude or spectral
characteristics of the noise. If there is very little or no leakage from the
speech
source into M2, then H2 (Z) Z 0 and Equation 3 reduces to

S(z) M 1(z) - M 2 (z)H I (z) . Eq. 4
Equation 4 is much simpler to implement and is very stable, assuming
H1(z) is stable. However, if significant speech energy is in M2(z), devoicing
can
occur. In order to construct a well-performing system and use Equation 4,
consideration is given to the following conditions:

R1. Availability of a perfect (or at least very good) VAD in noisy conditions
R2. Sufficiently accurate H1(z)
R3. Very small (ideally zero) H2(z).
R4. During speech production, H1(z) cannot change substantially.
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R5. During noise, H2(z) cannot change substantially.

Condition R1 is easy to satisfy if the SNR of the desired speech to the
unwanted noise is high enough. "Enough" means different things depending on
the method of VAD generation. If a VAD vibration sensor is used, as in Burnett
7,256,048, accurate VAD in very low SNRs (-10 dB or less) is possible.
Acoustic-only methods using information from O1 and 02 can also return
accurate VADs, but are limited to SNRs of N3 dB or greater for adequate
performance.
Condition R5 is normally simple to satisfy because for most applications
the microphones will not change position with respect to the user's mouth very
often or rapidly. In those applications where it may happen (such as hands-
free conferencing systems) it can be satisfied by configuring Mic2 so that
H2(z) Z 0 .

Satisfying conditions R2, R3, and R4 are more difficult but are possible
given the right combination of V1 and V2. Methods are examined below that
have proven to be effective in satisfying the above, resulting in excellent
noise
suppression performance and minimal speech removal and distortion in an
embodiment.
The DOMA, in various embodiments, can be used with the Pathfinder
system as the adaptive filter system or noise removal. The Pathfinder system,
available from AliphCom, San Francisco, CA, is described in detail in other
patents and patent applications referenced herein. Alternatively, any adaptive
filter or noise removal algorithm can be used with the DOMA in one or more
various alternative embodiments or configurations.
When the DOMA is used with the Pathfinder system, the Pathfinder
system generally provides adaptive noise cancellation by combining the two
microphone signals (e.g., Mic1, Mic2) by filtering and summing in the time
domain. The adaptive filter generally uses the signal received from a first
microphone of the DOMA to remove noise from the speech received from at
least one other microphone of the DOMA, which relies on a slowly varying
linear
transfer function between the two microphones for sources of noise. Following
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processing of the two channels of the DOMA, an output signal is generated in
which the noise content is attenuated with respect to the speech content, as
described in detail below.
Figure 12 is a generalized two-microphone array (DOMA) including an
array 1201/1202 and speech source S configuration, under an embodiment.
Figure 13 is a system 1300 for generating or producing a first order gradient
microphone V using two omnidirectional elements O1 and 02, under an
embodiment. The array of an embodiment includes two physical microphones
1201 and 1202 (e.g., omnidirectional microphones) placed a distance 2do apart
and a speech source 1200 is located a distance ds away at an angle of 8. This
array is axially symmetric (at least in free space), so no other angle is
needed.
The output from each microphone 1201 and 1202 can be delayed (z1 and z2),
multiplied by a gain (A1 and AZ), and then summed with the other as
demonstrated in Figure 13. The output of the array is or forms at least one
virtual microphone, as described in detail below. This operation can be over
any frequency range desired. By varying the magnitude and sign of the delays
and gains, a wide variety of virtual microphones (VMs), also referred to
herein
as virtual directional microphones, can be realized. There are other methods
known to those skilled in the art for constructing VMs but this is a common
one
and will be used in the enablement below.
As an example, Figure 14 is a block diagram for a DOMA 1400 including
two physical microphones configured to form two virtual microphones V1 and
V2, under an embodiment. The DOMA includes two first order gradient
microphones V1 and V2 formed using the outputs of two microphones or
elements O1 and 02 (1201 and 1202), under an embodiment. The DOMA of an
embodiment includes two physical microphones 1201 and 1202 that are
omnidirectional microphones, as described above with reference to Figures 12
and 13. The output from each microphone is coupled to a processing
component 1402, or circuitry, and the processing component outputs signals
representing or corresponding to the virtual microphones V1 and V2.
In this example system 1400, the output of physical microphone 1201 is
coupled to processing component 1402 that includes a first processing path
that
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includes application of a first delay z11 and a first gain All and a second
processing path that includes application of a second delay z12 and a second
gain A12. The output of physical microphone 1202 is coupled to a third
processing path of the processing component 1402 that includes application of
a third delay z21 and a third gain A21 and a fourth processing path that
includes
application of a fourth delay z22 and a fourth gain A22. The output of the
first
and third processing paths is summed to form virtual microphone V1, and the
output of the second and fourth processing paths is summed to form virtual
microphone V2.
As described in detail below, varying the magnitude and sign of the
delays and gains of the processing paths leads to a wide variety of virtual
microphones (VMs), also referred to herein as virtual directional microphones,
can be realized. While the processing component 1402 described in this
example includes four processing paths generating two virtual microphones or
microphone signals, the embodiment is not so limited. For example, Figure 15
is a block diagram for a DOMA 1500 including two physical microphones
configured to form N virtual microphones V1 through VN, where N is any number
greater than one, under an embodiment. Thus, the DOMA can include a
processing component 1502 having any number of processing paths as
appropriate to form a number N of virtual microphones.
The DOMA of an embodiment can be coupled or connected to one or
more remote devices. In a system configuration, the DOMA outputs signals to
the remote devices. The remote devices include, but are not limited to, at
least
one of cellular telephones, satellite telephones, portable telephones,
wireline
telephones, Internet telephones, wireless transceivers, wireless communication
radios, personal digital assistants (PDAs), personal computers (PCs), headset
devices, head-worn devices, and earpieces.
Furthermore, the DOMA of an embodiment can be a component or
subsystem integrated with a host device. In this system configuration, the
DOMA outputs signals to components or subsystems of the host device. The
host device includes, but is not limited to, at least one of cellular
telephones,
satellite telephones, portable telephones, wireline telephones, Internet



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telephones, wireless transceivers, wireless communication radios, personal
digital assistants (PDAs), personal computers (PCs), headset devices, head-
worn devices, and earpieces.
As an example, Figure 16 is an example of a headset or head-worn
device 1600 that includes the DOMA, as described herein, under an
embodiment. The headset 1600 of an embodiment includes a housing having
two areas or receptacles (not shown) that receive and hold two microphones
(e.g., O1 and 02). The headset 1600 is generally a device that can be worn by
a speaker 1602, for example, a headset or earpiece that positions or holds the
microphones in the vicinity of the speaker's mouth. The headset 1600 of an
embodiment places a first physical microphone (e.g., physical microphone O1)
in a vicinity of a speaker's lips. A second physical microphone (e.g.,
physical
microphone 02) is placed a distance behind the first physical microphone. The
distance of an embodiment is in a range of a few centimeters behind the first
physical microphone or as described herein (e.g., described with reference to
Figures 11-15). The DOMA is symmetric and is used in the same configuration
or manner as a single close-talk microphone, but is not so limited.
Figure 17 is a flow diagram for denoising 1700 acoustic signals using
the DOMA, under an embodiment. The denoising 1700 begins by receiving
1702 acoustic signals at a first physical microphone and a second physical
microphone. In response to the acoustic signals, a first microphone signal is
output from the first physical microphone and a second microphone signal is
output from the second physical microphone 1704. A first virtual microphone is
formed 1706 by generating a first combination of the first microphone signal
and the second microphone signal. A second virtual microphone is formed
1708 by generating a second combination of the first microphone signal and the
second microphone signal, and the second combination is different from the
first combination. The first virtual microphone and the second virtual
microphone are distinct virtual directional microphones with substantially
similar responses to noise and substantially dissimilar responses to speech.
The denoising 1700 generates 1710 output signals by combining signals from
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the first virtual microphone and the second virtual microphone, and the output
signals include less acoustic noise than the acoustic signals.
Figure 18 is a flow diagram for forming 1800 the DOMA, under an
embodiment. Formation 1800 of the DOMA includes forming 1802 a physical
microphone array including a first physical microphone and a second physical
microphone. The first physical microphone outputs a first microphone signal
and the second physical microphone outputs a second microphone signal. A
virtual microphone array is formed 1804 comprising a first virtual microphone
and a second virtual microphone. The first virtual microphone comprises a
first
combination of the first microphone signal and the second microphone signal.
The second virtual microphone comprises a second combination of the first
microphone signal and the second microphone signal, and the second
combination is different from the first combination. The virtual microphone
array including a single null oriented in a direction toward a source of
speech of
a human speaker.
The construction of VMs for the adaptive noise suppression system of an
embodiment includes substantially similar noise response in V1 and V2.
Substantially similar noise response as used herein means that H1(z) is simple
to model and will not change much during speech, satisfying conditions R2 and
R4 described above and allowing strong denoising and minimized bleedthrough.
The construction of VMs for the adaptive noise suppression system of an
embodiment includes relatively small speech response for V2. The relatively
small speech response for V2 means that H2(z) 0, which will satisfy conditions
R3 and R5 described above.
The construction of VMs for the adaptive noise suppression system of an
embodiment further includes sufficient speech response for V1 so that the
cleaned speech will have significantly higher SNR than the original speech
captured by 01.
The description that follows assumes that the responses of the
omnidirectional microphones 01 and 02 to an identical acoustic source have
been normalized so that they have exactly the same response (amplitude and
phase) to that source. This can be accomplished using standard microphone
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array methods (such as frequency-based calibration) well known to those
versed in the art.
Referring to the condition that construction of VMs for the adaptive noise
suppression system of an embodiment includes relatively small speech
response for V2, it is seen that for discrete systems V2(z) can be represented
as:

V2(Z)=02(z)-Z-yf301(Z)
where

1
d
d2
y= d, - d, . f, (samples)
C

dl = ds 2d, do cos(O) + d2
d2 = ds +2dsdo cos(h)+do

The distances d1 and d2 are the distance from 01 and 02 to the speech source
(see Figure 12), respectively, and y is their difference divided by c, the
speed
of sound, and multiplied by the sampling frequency f5. Thus y is in samples,
but
need not be an integer. For non-integer y, fractional-delay filters (well
known to
those versed in the art) may be used.
It is important to note that the R above is not the conventional P used to
denote the mixing of VMs in adaptive beamforming; it is a physical variable of
the system that depends on the intra-microphone distance do (which is fixed)
and the distance ds and angle 0, which can vary. As shown below, for properly
calibrated microphones, it is not necessary for the system to be programmed
with the exact R of the array. Errors of approximately 10-15% in the actual R
(i.e. the R used by the algorithm is not the R of the physical array) have
been
used with very little degradation in quality. The algorithmic value of R may
be
calculated and set for a particular user or may be calculated adaptively
during
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speech production when little or no noise is present. However, adaptation
during use is not required for nominal performance.
Figure 19 is a plot of linear response of virtual microphone V2 with R =
0.8 to a 1 kHz speech source at a distance of 0.1 m, under an embodiment.
The null in the linear response of virtual microphone V2 to speech is located
at 0
degrees, where the speech is typically expected to be located. Figure 20 is a
plot of linear response of virtual microphone V2 with R = 0.8 to a 1 kHz noise
source at a distance of 1.0 m, under an embodiment. The linear response of V2
to noise is devoid of or includes no null, meaning all noise sources are
detected.
The above formulation for V2(z) has a null at the speech location and will
therefore exhibit minimal response to the speech. This is shown in Figure 19
for an array with do = 10.7 mm and a speech source on the axis of the array (0
= 0) at 10 cm (R = 0.8). Note that the speech null at zero degrees is not
present for noise in the far field for the same microphone, as shown in Figure
20 with a noise source distance of approximately 1 meter. This insures that
noise in front of the user will be detected so that it can be removed. This
differs from conventional systems that can have difficulty removing noise in
the
direction of the mouth of the user.
The V1(z) can be formulated using the general form for V1(z):

V1(Z)=aA01(z)= Z-dA -aB02(z)=Z-dB
Since

V2 (z) = 02 (z) - z-'P O1(z)

and, since for noise in the forward direction

02N(z)=01N(z). Z-Y
i

then l
V2N (z) = O1N (Z)' Z-' - Z-TO1N (z)
V2N(Z)=(1-P)`O1N(z).z-r)

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If this is then set equal to Vi(z) above, the result is

V1N(Z)-a'AO1N(z).Z dA -aBO1N(Z)-z Y =z dB =(1-NXOIN(Z).z Y)
thus we may set

dA=Y
dB = 0

aA= 1
aB=
to get

Vl(z)=01(z)=z-' -(302(z)

The definitions for V1 and V2 above mean that for noise H1(z) is:
H1(z) V1(Z) P02(Z)+01(Z) z Y
V2(Z) 02(Z)-z YRO1(Z)
which, if the amplitude noise responses are about the same, has the form of an
allpass filter. This has the advantage of being easily and accurately modeled,
especially in magnitude response, satisfying R2.

This formulation assures that the noise response will be as similar as
possible
and that the speech response will be proportional to (1-l32). Since 03 is the
ratio
of the distances from O1 and 02 to the speech source, it is affected by the
size
of the array and the distance from the array to the speech source.
Figure 21 is a plot of linear response of virtual microphone V1 with R =
0.8 to a 1 kHz speech source at a distance of 0.1 m, under an embodiment.
The linear response of virtual microphone V1 to speech is devoid of or
includes
no null and the response for speech is greater than that shown in Figure 14.
Figure 22 is a plot of linear response of virtual microphone V1 with R =
0.8 to a 1 kHz noise source at a distance of 1.0 m, under an embodiment. The


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linear response of virtual microphone V1 to noise is devoid of or includes no
null
and the response is very similar to V2 shown in Figure 15.
Figure 23 is a plot of linear response of virtual microphone V1 with R =
0.8 to a speech source at a distance of 0.1 m for frequencies of 100, 500,
1000, 2000, 3000, and 4000 Hz, under an embodiment. Figure 24 is a plot
showing comparison of frequency responses for speech for the array of an
embodiment and for a conventional cardioid microphone.
The response of V1 to speech is shown in Figure 21, and the response to
noise in Figure 22. Note the difference in speech response compared to V2
shown in Figure 19 and the similarity of noise response shown in Figure 20.
Also note that the orientation of the speech response for V1 shown in Figure
21
is completely opposite the orientation of conventional systems, where the main
lobe of response is normally oriented toward the speech source. The
orientation of an embodiment, in which the main lobe of the speech response of
V1 is oriented away from the speech source, means that the speech sensitivity
of V1 is lower than a normal directional microphone but is flat for all
frequencies
within approximately +-30 degrees of the axis of the array, as shown in Figure
23. This flatness of response for speech means that no shaping postfilter is
needed to restore omnidirectional frequency response. This does come at a
price - as shown in Figure 24, which shows the speech response of V1 with (3 =
0.8 and the speech response of a cardioid microphone. The speech response of
V1 is approximately 0 to -13 dB less than a normal directional microphone
between approximately 500 and 7500 Hz and approximately 0 to 10+ dB
greater than a directional microphone below approximately 500 Hz and above
7500 Hz for a sampling frequency of approximately 16000 Hz. However, the
superior noise suppression made possible using this system more than
compensates for the initially poorer SNR.
It should be noted that Figures 19-22 assume the speech is located at
approximately 0 degrees and approximately 10 cm, R = 0.8, and the noise at all
angles is located approximately 1.0 meter away from the midpoint of the array.
Generally, the noise distance is not required to be 1 m or more, but the
denoising is the best for those distances. For distances less than
approximately
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1 m, denoising will not be as effective due to the greater dissimilarity in
the
noise responses of V1 and V2. This has not proven to be an impediment in
practical use - in fact, it can be seen as a feature. Any "noise" source that
is
"-10 cm away from the earpiece is likely to be desired to be captured and
transmitted.
The speech null of V2 means that the VAD signal is no longer a critical
component. The VAD's purpose was to ensure that the system would not train
on speech and then subsequently remove it, resulting in speech distortion. If,
however, V2 contains no speech, the adaptive system cannot train on the
speech and cannot remove it. As a result, the system can denoise all the time
without fear of devoicing, and the resulting clean audio can then be used to
generate a VAD signal for use in subsequent single-channel noise suppression
algorithms such as spectral subtraction. In addition, constraints on the
absolute value of H1(z) (i.e. restricting it to absolute values less than two)
can
keep the system from fully training on speech even if it is detected. In
reality,
though, speech can be present due to a mis-located V2 null and/or echoes or
other phenomena, and a VAD sensor or other acoustic-only VAD is
recommended to minimize speech distortion.
Depending on the application, R and y may be fixed in the noise
suppression algorithm or they can be estimated when the algorithm indicates
that speech production is taking place in the presence of little or no noise.
In
either case, there may be an error in the estimate of the actual R and y of
the
system. The following description examines these errors and their effect on
the
performance of the system. As above, "good performance" of the system
indicates that there is sufficient denoising and minimal devoicing.
The effect of an incorrect R and y on the response of V1 and V2 can be
seen by examining the definitions above:

V1(Z)=01(Z).Z-YT -PT02\Z)
V2 (z) _ 02 (Z) - Z_1 T PT01(Z)

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where RT and YT denote the theoretical estimates of f3 and y used in the noise
suppression algorithm. In reality, the speech response of 02 is

('
2S (Z) = PRO1S (Z)' Z-Y R
where RR and yR denote the real (3 and y of the physical system. The
differences
between the theoretical and actual values of p and y can be due to mis-
location
of the speech source (it is not where it is assumed to be) and/or a change in
air
temperature (which changes the speed of sound). Inserting the actual
response of 02 for speech into the above equations for V1 and V2 yields

V1S(Z)=O1S(z)[z-YT -PTPRZ-YR
V2S(Z)-01S(Z)[PRZ !R - PTZYT J1

If the difference in phase is represented by

YR =YT +YD
And the difference in amplitude as
PR = BRT
then

V1S (Z) = 01S (z)z-YT [1 - BRTZ-YD
V2s (z) = PT01s (z)z-YT [Bz-YD _ 11' Eq. 5
The speech cancellation in V2 (which directly affects the degree of
devoicing) and the speech response of V1 will be dependent on both B and D.
An examination of the case where D = 0 follows. Figure 25 is a plot showing
speech response for V1 (top, dashed) and V2 (bottom, solid) versus B with ds
assumed to be 0.1 m, under an embodiment. This plot shows the spatial null in
V2 to be relatively broad. Figure 26 is a plot showing a ratio of V1/V2 speech
responses shown in Figure 20 versus B, under an embodiment. The ratio of
V1/V2 is above 10 dB for all 0.8 < B < 1.1, and this means that the physical R
of
the system need not be exactly modeled for good performance. Figure 27 is a

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plot of B versus actual ds assuming that ds = 10 cm and theta = 0, under an
embodiment. Figure 28 is a plot of B versus theta with ds = 10 cm and
assuming ds = 10 cm, under an embodiment.
In Figure 25, the speech response for V1 (upper, dashed) and V2 (lower,
solid) compared to 01 is shown versus B when ds is thought to be
approximately 10 cm and 0 = 0. When B = 1, the speech is absent from V2. In
Figure 26, the ratio of the speech responses in Figure 20 is shown. When 0.8
< B < 1.1, the V1/V2 ratio is above approximately 10 dB - enough for good
performance. Clearly, if D = 0, B can vary significantly without adversely
affecting the performance of the system. Again, this assumes that calibration
of the microphones so that both their amplitude and phase response is the
same for an identical source has been performed.
The B factor can be non-unity for a variety of reasons. Either the
distance to the speech source or the relative orientation of the array axis
and
the speech source or both can be different than expected. If both distance and
angle mismatches are included for B, then

B= RR dsR - 2dsRdo cosOOR )+ do dsT + 2dSTdo cos(OT )+ do
PT jd2R + 2dsRdo cos(OR ) + do dsT - 2dsTdo cos(OT)+ do

where again the T subscripts indicate the theorized values and R the actual
values. In Figure 27, the factor B is plotted with respect to the actual ds
with
the assumption that ds = 10 cm and 0 = 0. So, if the speech source in on-axis
of the array, the actual distance can vary from approximately 5 cm to 18 cm
without significantly affecting performance - a significant amount. Similarly,
Figure 28 shows what happens if the speech source is located at a distance of
approximately 10 cm but not on the axis of the array. In this case, the angle
can vary up to approximately +-55 degrees and still result in a B less than
1.1,
assuring good performance. This is a significant amount of allowable angular
deviation. If there is both angular and distance errors, the equation above
may
be used to determine if the deviations will result in adequate performance. Of
course, if the value for PT is allowed to update during speech, essentially

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tracking the speech source, then B can be kept near unity for almost all
configurations.
An examination follows of the case where B is unity but D is nonzero.
This can happen if the speech source is not where it is thought to be or if
the
speed of sound is different from what it is believed to be. From Equation 5
above, it can be sees that the factor that weakens the speech null in V2 for
speech is

N(z) = Bz-YD -1
or in the continuous s domain

N(s) = Be-D5 -1.

Since y is the time difference between arrival of speech at V1 compared to V2i
it
can be errors in estimation of the angular location of the speech source with
respect to the axis of the array and/or by temperature changes. Examining the
temperature sensitivity, the speed of sound varies with temperature as
c = 331.3 +(0.606T) m/s

where T is degrees Celsius. As the temperature decreases, the speed of sound
also decreases. Setting 20 C as a design temperature and a maximum
expected temperature range to -40 C to +60 C (-40 F to 140 F). The design
speed of sound at 20 C is 343 m/s and the slowest speed of sound will be 307
m/s at -40 C with the fastest speed of sound 362 m/s at 60 C. Set the array
length (2do) to be 21 mm. For speech sources on the axis of the array, the
difference in travel time for the largest change in the speed of sound is
VtMAX = d - d =0.021m 1 - 1 =-7.2x10-6 sec
cl c2 343m/s 307m/s

or approximately 7 microseconds. The response for N(s) given B = 1 and D =
7.2 sec is shown in Figure 29. Figure 29 is a plot of amplitude (top) and



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phase (bottom) response of N(s) with B = 1 and D = -7.2 sec, under an
embodiment. The resulting phase difference clearly affects high frequencies
more than low. The amplitude response is less than approximately -10 dB for
all frequencies less than 7 kHz and is only about -9 dB at 8 kHz. Therefore,
assuming B = 1, this system would likely perform well at frequencies up to
approximately 8 kHz. This means that a properly compensated system would
work well even up to 8 kHz in an exceptionally wide (e.g., -40 C to 80 C)
temperature range. Note that the phase mismatch due to the delay estimation
error causes N(s) to be much larger at high frequencies compared to low.
If B is not unity, the robustness of the system is reduced since the effect
from non-unity B is cumulative with that of non-zero D. Figure 30 shows the
amplitude and phase response for B = 1.2 and D = 7.2 sec. Figure 30 is a
plot of amplitude (top) and phase (bottom) response of N(s) with B = 1.2 and
D = -7.2 sec, under an embodiment. Non-unity B affects the entire frequency
range. Now N(s) is below approximately -10 dB only for frequencies less than
approximately 5 kHz and the response at low frequencies is much larger. Such
a system would still perform well below 5 kHz and would only suffer from
slightly elevated devoicing for frequencies above 5 kHz. For ultimate
performance, a temperature sensor may be integrated into the system to allow
the algorithm to adjust YT as the temperature varies.
Another way in which D can be non-zero is when the speech source is not
where it is believed to be - specifically, the angle from the axis of the
array to
the speech source is incorrect. The distance to the source may be incorrect as
well, but that introduces an error in B, not D.
Referring to Figure 12, it can be seen that for two speech sources (each
with their own ds and 0) that the time difference between the arrival of the
speech at 01 and the arrival at 02 is

OtI(d12-dii-d22+d21)
C
where

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d11 = d51 - 2dS1do cos(0l ) + do

d12 = d21 +2dsldo cos(01)+d2
d21 = d2 S2 -2dS2do cos(02)+do
2 2
2 + 2dS2do cos(02 )+ do
d22 = dS

The V2 speech cancellation response for 01 = 0 degrees and 02 = 30
degrees and assuming that B = 1 is shown in Figure 31. Figure 31 is a plot
of amplitude (top) and phase (bottom) response of the effect on the speech
cancellation in V2 due to a mistake in the location of the speech source with
q1
= 0 degrees and q2 = 30 degrees, under an embodiment. Note that the
cancellation is still below -10 dB for frequencies below 6 kHz. The
cancellation
is still below approximately -10 dB for frequencies below approximately 6 kHz,
so an error of this type will not significantly affect the performance of the
system. However, if 02 is increased to approximately 45 degrees, as shown in
Figure 32, the cancellation is below approximately -10 dB only for frequencies
below approximately 2.8 kHz. Figure 32 is a plot of amplitude (top) and phase
(bottom) response of the effect on the speech cancellation in V2 due to a
mistake in the location of the speech source with q1 = 0 degrees and q2 = 45
degrees, under an embodiment. Now the cancellation is below -10 dB only for
frequencies below about 2.8 kHz and a reduction in performance is expected.
The poor V2 speech cancellation above approximately 4 kHz may result in
significant devoicing for those frequencies.
The description above has assumed that the microphones 01 and 02 were
calibrated so that their response to a source located the same distance away
was identical for both amplitude and phase. This is not always feasible, so a
more practical calibration procedure is presented below. It is not as
accurate,
but is much simpler to implement. Begin by defining a filter a(z) such that:

01C(Z) =a (Z)02C(Z)

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where the "C" subscript indicates the use of a known calibration source. The
simplest one to use is the speech of the user. Then

015(z) =a (Z)02c(Z)
The microphone definitions are now:

V1 (z) = 01(z) = z-y - J3(z)a(z)02 (z)
V2 (z) = a(z)02 (z) - z-'"(3i z)O1(z)

The 0 of the system should be fixed and as close to the real value as
possible. In practice, the system is not sensitive to changes in R and errors
of
approximately +-5% are easily tolerated. During times when the user is
producing speech but there is little or no noise, the system can train a(z) to
remove as much speech as possible. This is accomplished by:

1. Construct an adaptive system as shown in Figure 11 with (3O1S(z)z `' in
the "MIC1" position, 02S(z) in the "MIC2" position, and a(z) in the H1(z)
position.
2. During speech, adapt a(z) to minimize the residual of the system.
3. Construct V1(z) and V2(z) as above.

A simple adaptive filter can be used for a(z) so that only the relationship
between the microphones is well modeled. The system of an embodiment
trains only when speech is being produced by the user. A sensor like the SSM
is invaluable in determining when speech is being produced in the absence of
noise. If the speech source is fixed in position and will not vary
significantly
during use (such as when the array is on an earpiece), the adaptation should
be
infrequent and slow to update in order to minimize any errors introduced by
noise present during training.
The above formulation works very well because the noise (far-field)
responses of V1 and V2 are very similar while the speech (near-field)
responses
are very different. However, the formulations for V1 and V2 can be varied and

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still result in good performance of the system as a whole. If the definitions
for
V1 and V2 are taken from above and new variables 131 and B2 are inserted, the
result is:

v1(Z)_01(Z)= Z-" -B1RT02(Z)

V2(Z)=02(Z)-Z YTB2NTO1(z)

where 131 and B2 are both positive numbers or zero. If 131 and B2 are set
equal
to unity, the optimal system results as described above. If 131 is allowed to
vary from unity, the response of V1 is affected. An examination of the case
where B2 is left at 1 and 131 is decreased follows. As 131 drops to
approximately
zero, V1 becomes less and less directional, until it becomes a simple
omnidirectional microphone when 131 = 0. Since B2 = 1, a speech null remains
in V2, so very different speech responses remain for V1 and V2. However, the
noise responses are much less similar, so denoising will not be as effective.
Practically, though, the system still performs well. B1 can also be increased
from unity and once again the system will still denoise well, just not as well
as
withal=1.
If B2 is allowed to vary, the speech null in V2 is affected. As long as the
speech null is still sufficiently deep, the system will still perform well.
Practically values down to approximately B2 = 0.6 have shown sufficient
performance, but it is recommended to set B2 close to unity for optimal
performance.
Similarly, variables E and 0 may be introduced so that:

V1(Z)=( - N)02N(Z) + (1 + i)O1N(Z)Z-Y
V2(Z)=(1 + A)O2N(Z) + ( - R)01N(Z)Z-Y
This formulation also allows the virtual microphone responses to be varied but
retains the all-pass characteristic of H1(z).
In conclusion, the system is flexible enough to operate well at a variety
of 131 values, but B2 values should be close to unity to limit devoicing for
best
performance.

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Experimental results for a 2do = 19 mm array using a linear R of 0.83
and B1 = B2 = 1 on a Bruel and Kjaer Head and Torso Simulator (HATS) in
very loud (-85 dBA) music/speech noise environment are shown in Figure 33.
The alternate microphone calibration technique discussed above was used to
calibrate the microphones. The noise has been reduced by about 25 dB and the
speech hardly affected, with no noticeable distortion. Clearly the technique
significantly increases the SNR of the original speech, far outperforming
conventional noise suppression techniques.
The DOMA can be a component of a single system, multiple systems,
and/or geographically separate systems. The DOMA can also be a
subcomponent or subsystem of a single system, multiple systems, and/or
geographically separate systems. The DOMA can be coupled to one or more
other components (not shown) of a host system or a system coupled to the
host system.
One or more components of the DOMA and/or a corresponding system or
application to which the DOMA is coupled or connected includes and/or runs
under and/or in association with a processing system. The processing system
includes any collection of processor-based devices or computing devices
operating together, or components of processing systems or devices, as is
known in the art. For example, the processing system can include one or more
of a portable computer, portable communication device operating in a
communication network, and/or a network server. The portable computer can
be any of a number and/or combination of devices selected from among
personal computers, cellular telephones, personal digital assistants, portable
computing devices, and portable communication devices, but is not so limited.
The processing system can include components within a larger computer
system.

ACOUSTIC VOICE ACTIVITY DETECTION (AVAD) FOR ELECTRONIC SYSTEMS
Acoustic Voice Activity Detection (AVAD) methods and systems are
described herein. The AVAD methods and systems, which include
algorithms or programs, use microphones to generate virtual directional



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microphones which have very similar noise responses and very dissimilar
speech responses. The ratio of the energies of the virtual microphones is
then calculated over a given window size and the ratio can then be used
with a variety of methods to generate a VAD signal. The virtual
microphones can be constructed using either a fixed or an adaptive filter.
The adaptive filter generally results in a more accurate and noise-robust
VAD signal but requires training. In addition, restrictions can be placed
on the filter to ensure that it is training only on speech and not on
environmental noise.
In the following description, numerous specific details are
introduced to provide a thorough understanding of, and enabling
description for, embodiments. One skilled in the relevant art, however,
will recognize that these embodiments can be practiced without one or
more of the specific details, or with other components, systems, etc. In
other instances, well-known structures or operations are not shown, or
are not described in detail, to avoid obscuring aspects of the disclosed
embodiments.
Figure 34 is a configuration of a two-microphone array of the
AVAD with speech source S, under an embodiment. The AVAD of an
embodiment uses two physical microphones (O1 and 02) to form two
virtual microphones (V1 and V2). The virtual microphones of an
embodiment are directional microphones, but the embodiment is not so
limited. The physical microphones of an embodiment include
omnidirectional microphones, but the embodiments described herein are
not limited to omnidirectional microphones. The virtual microphone (VM)
V2 is configured in such a way that it has minimal response to the speech
of the user, while V1 is configured so that it does respond to the user's
speech but has a very similar noise magnitude response to V2, as
described in detail herein. The PSAD VAD methods can then be used to

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determine when speech is taking place. A further refinement is the use
of an adaptive filter to further minimize the speech response of V2,
thereby increasing the speech energy ratio used in PSAD and resulting in
better overall performance of the AVAD.
The PSAD algorithm as described herein calculates the ratio of the
energies of two directional microphones M1 and M2:

)2
M
R = l~Z)2
i

where the "z" indicates the discrete frequency domain and "i" ranges
from the beginning of the window of interest to the end, but the same
relationship holds in the time domain. The summation can occur over a
window of any length; 200 samples at a sampling rate of 8 kHz has been
used to good effect. Microphone M1 is assumed to have a greater speech
response than microphone M2. The ratio R depends on the relative
strength of the acoustic signal of interest as detected by the
microphones.
For matched omnidirectional microphones (i.e. they have the same
response to acoustic signals for all spatial orientations and frequencies),
the size of R can be calculated for speech and noise by approximating
the propagation of speech and noise waves as spherically symmetric
sources. For these the energy of the propagating wave decreases as
1/r2:

/Mi(z)2d2di+d
R M2Czi)2 d1 d1

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The distance d1 is the distance from the acoustic source to M1i d2 is
the distance from the acoustic source to M2, and d = d2-d1 (see Figure
34). It is assumed that 01 is closer to the speech source (the user's
mouth) so that d is always positive. If the microphones and the user's
mouth are all on a line, then d = 2do, the distance between the
microphones. For matched omnidirectional microphones, the magnitude
of R, depends only on the relative distance between the microphones and
the acoustic source. For noise sources, the distances are typically a
meter or more, and for speech sources, the distances are on the order of
10 cm, but the distances are not so limited. Therefore for a 2-cm array
typical values of R are:

_d212cm
RS _
1.2
d110cm

d2102cm
R _1.02
d1100cm

where the "S" subscript denotes the ratio for speech sources and "N" the
ratio for noise sources. There is not a significant amount of separation
between noise and speech sources in this case, and therefore it would be
difficult to implement a robust solution using simple omnidirectional
microphones.
A better implementation is to use directional microphones where
the second microphone has minimal speech response. As described
herein, such microphones can be constructed using omnidirectional
microphones 01 and 02:

V1(Z)=_Y(Z)a(Z)02(Z) + 01(z)z-Y
V2(z)=a(Z)02(Z) -Y(Z)01(Z)Z Y
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where a(z) is a calibration filter used to compensate 02's response so
that it is the same as 01r p(z) is a filter that describes the relationship
between 01 and calibrated 02 for speech, and y is a fixed delay that
depends on the size of the array. There is no loss of generality in
defining a(z) as above, as either microphone may be compensated to
match the other. For this configuration V1 and V2 have very similar noise
response magnitudes and very dissimilar speech response magnitudes if

d
Y _ c

where again d = 2do and c is the speed of sound in air, which is
temperature dependent and approximately

T m
c=331.3 J1 -
273.15 sec
where T is the temperature of the air in Celsius.
The filter J3(z) can be calculated using wave theory to be

P W = d1 - d1 [ 2 ]
dZ d1 + d

where again dk is the distance from the user's mouth to Ok. Figure 35
is a block diagram of V2 construction using a fixed 13(z), under an
embodiment. This fixed (or static) p works sufficiently well if the
calibration filter a(z) is accurate and d1 and d2 are accurate for the user.
This fixed-p algorithm, however, neglects important effects such as

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reflection, diffraction, poor array orientation (i.e. the microphones and
the mouth of the user are not all on a line), and the possibility of
different d1 and d2 values for different users.
The filter 13(z) can also be determined experimentally using an
adaptive filter. Figure 36 is a block diagram of V2 construction using an
adaptive J3(z), under an embodiment, where:

(z) a z02 z [ 3
- Z-Y01(Z)

The adaptive process varies /3(z) to minimize the output of V2 when only
speech is being received by 01 and 02. A small amount of noise may be
tolerated with little ill effect, but it is preferred that only speech is
being
received when the coefficients of Q(z) are calculated. Any adaptive
process may be used; a normalized least-mean squares (NLMS)
algorithm was used in the examples below.
The V1 can be constructed using the current value for /3(z) or the
fixed filter a(z) can be used for simplicity. Figure 37 is a block diagram
of V1 construction, under an embodiment.
Now the ratio R is

IIV1(z)II (-g(z)a(z)O2(z) + o1(z)z-Y)z
R II V (z)II (a(z)Oz(z)-R(Z)61(Z)Z-Y)Z

where double bar indicates norm and again any size window may be
used. If /(z) has been accurately calculated, the ratio for speech should
be relatively high (e.g., greater than approximately 2) and the ratio for
noise should be relatively low (e.g., less than approximately 1.1). The
ratio calculated will depend on both the relative energies of the speech
and noise as well as the orientation of the noise and the reverberance of



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the environment. In practice, either the adapted filter $(z) or the static
filter b(z) may be used for V1(z) with little effect on R - but it is
important to use the adapted filter #(z) in V2(z) for best performance.
Many techniques known to those skilled in the art (e.g., smoothing, etc.)
can be used to make R more amenable to use in generating a VAD and
the embodiments herein are not so limited.
The ratio R can be calculated for the entire frequency band of
interest, or can be calculated in frequency subbands. One effective
subband discovered was 250 Hz to 1250 Hz, another was 200 Hz to 3000
Hz, but many others are possible and useful.
Once generated, the vector of the ratio R versus time (or the
matrix of R versus time if multiple subbands are used) can be used with
any detection system (such as one that uses fixed and/or adaptive
thresholds) to determine when speech is occurring. While many
detection systems and methods are known to exist by those skilled in the
art and may be used, the method described herein for generating an R
so that the speech is easily discernable is novel. It is important to note
that the R does not depend on the type of noise or its orientation or
frequency content; R simply depends on the V1 and V2 spatial response
similarity for noise and spatial response dissimilarity for speech. In this
way it is very robust and can operate smoothly in a variety of noisy
acoustic environments.
Figure 38 is a flow diagram of acoustic voice activity detection
3800, under an embodiment. The detection comprises forming a first
virtual microphone by combining a first signal of a first physical
microphone and a second signal of a second physical microphone 3802.
The detection comprises forming a filter that describes a relationship for
speech between the first physical microphone and the second physical
microphone 3804. The detection comprises forming a second virtual

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microphone by applying the filter to the first signal to generate a first
intermediate signal, and summing the first intermediate signal and the
second signal 3806. The detection comprises generating an energy ratio
of energies of the first virtual microphone and the second virtual
microphone 3808. The detection comprises detecting acoustic voice
activity of a speaker when the energy ratio is greater than a threshold
value 3810.
The accuracy of the adaptation to the R(z) of the system is a factor
in determining the effectiveness of the AVAD. A more accurate
adaptation to the actual R(z) of the system leads to lower energy of the
speech response in V2i and a higher ratio R. The noise (far-field)
magnitude response is largely unchanged by the adaptation process, so
the ratio R will be near unity for accurately adapted beta. For purposes
of accuracy, the system can be trained on speech alone, or the noise
should be low enough in energy so as not to affect or to have a minimal
affect the training.
To make the training as accurate as possible, the coefficients of the
filter j3(z) of an embodiment are generally updated under the following
conditions, but the embodiment is not so limited: speech is being
produced (requires a relatively high SNR or other method of detection
such as an Aliph Skin Surface Microphone (SSM) as described in United
States Patent Application number 10/769,302, filed January 30, 2004,
which is incorporated by reference herein in its entirety); no wind is
detected (wind can be detected using many different methods known in
the art, such as examining the microphones for uncorrelated low-
frequency noise); and the current value of R is much larger than a
smoothed history of R values (this ensures that training occurs only
when strong speech is present). These procedures are flexible and
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others may be used without significantly affecting the performance of the
system. These restrictions can make the system relatively more robust.
Even with these precautions, it is possible that the system
accidentally trains on noise (e.g., there may be a higher likelihood of this
without use of a non-acoustic VAD device such as the SSM used in the
Jawbone headset produced by Aliph, San Francisco, California). Thus, an
embodiment includes a further failsafe system to preclude accidental
training from significantly disrupting the system. The adaptive R is
limited to certain values expected for speech. For example, values for di
for an ear-mounted headset will normally fall between 9 and 14
centimeters, so using an array length of 2do = 2.0 cm and Equation 2
above,

d, d,
d2 d1 + 2do
which means that

0.82 < IJ3(z)I < 0.88.
The magnitude of the 3 filter can therefore be limited to between
approximately 0.82 and 0.88 to preclude problems if noise is present
during training. Looser limits can be used to compensate for inaccurate
calibrations (the response of omnidirectional microphones is usually
calibrated to one another so that their frequency response is the same to
the same acoustic source - if the calibration is not completely accurate
the virtual microphones may not form properly).
Similarly, the phase of the p filter can be limited to be what is
expected from a speech source within +- 30 degrees from the axis of the
array. As described herein, and with reference to Figure 34,

(seconds)
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d2 - dl
r c

d1 = Jds - 2dsdo cos(8) + do
d2 = Jds2 + 2dsdo cos(O) + do

where ds is the distance from the midpoint of the array to the speech
source. Varying ds from 10 to 15 cm and allowing 0 to vary between 0
and +- 30 degrees, the maximum difference in y results from the
difference of y at 0 degrees (58.8 sec) and y at +-30 degrees for ds = 10
cm (50.8 sec). This means that the maximum expected phase
difference is 58.8 - 50.8 = 8.0 sec, or 0.064 samples at an 8 kHz
sampling rate. Since

cp(f) = 27rft = 21Tf(8.0x10-6) rad
the maximum phase difference realized at 4 kHz is only 0.2 rad or about
11.4 degrees, a small amount, but not a negligible one. Therefore the R
filter should almost linear phase, but some allowance made for
differences in position and angle. In practice a slightly larger amount
was used (0.071 samples at 8 kHz) in order to compensate for poor
calibration and diffraction effects, and this worked well. The limit on the
phase in the example below was implemented as the ratio of the central
tap energy to the combined energy of the other taps:

(center tap)2
phase limit ratio = III3II

where p is the current estimate. This limits the phase by restricting the
effects of the non-center taps. Other ways of limiting the phase of the
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beta filter are known to those skilled in the art and the algorithm
presented here is not so limited.
Embodiments are presented herein that use both a fixed 13(z) and an
adaptive 3(z), as described in detail above. In both cases, R was calculated
using frequencies between 250 and 3000 Hz using a window size of 200
samples at 8 kHz. The results for V1 (top plot), V2 (middle plot), R
(bottom plot, solid line, windowed using a 200 sample rectangular
window at 8 kHz) and the VAD (bottom plot, dashed line) are shown in
Figures 39-44. Figures 39-44 demonstrate the use of a fixed beta
filter (3(z) in conditions of only noise (street and bus noise, approximately
70 dB SPL at the ear), only speech (normalized to 94 dB SPL at the
mouth reference point (MRP)), and mixed noise and speech,
respectively. A Bruel & Kjaer Head and Torso Simulator (HATS) was
used for the tests and the omnidirectional microphones mounted on
HATS' ear with the midline of the array approximately 11 cm from the
MRP. The fixed beta filter used was 13F(z)=0.82, where the "F" subscript
indicates a fixed filter. The VAD was calculated using a fixed threshold of
1.5.
Figure 39 shows experimental results of the algorithm using a
fixed beta when only noise is present, under an embodiment. The top
plot is V1i the middle plot is V2, and the bottom plot is R (solid line) and
the VAD result (dashed line) versus time. Examining Figure 39, the
response of both V1 and V2 are very similar, and the ratio R is very near
unity for the entire sample. The VAD response has occasional false
positives denoted by spikes in the R plot (windows that are identified by
the algorithm as containing speech when they do not), but these are
easily removed using standard pulse removal algorithms and/or
smoothing of the R results.
Figure 40 shows experimental results of the algorithm using a
fixed beta when only speech is present, under an embodiment. The top


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plot is V1, the middle plot is V21 and the bottom plot is R (solid line) and
the VAD result (dashed line) versus time. The R ratio is between
approximately 2 and approximately 7 on average, and the speech is
easily discernable using the fixed threshold. These results show that the
response of the two virtual microphones to speech are very different,
and indeed that ratio R varies from 2-7 during speech. There are very
few false positives and very few false negatives (windows that contain
speech but are not identified as speech windows). The speech is easily
and accurately detected.
Figure 41 shows experimental results of the algorithm using a
fixed beta when speech and noise is present, under an embodiment. The
top plot is V1, the middle plot is V2, and the bottom plot is R (solid line)
and the VAD result (dashed line) versus time. The R ratio is lower than
when no noise is present, but the VAD remains accurate with only a few
false positives. There are more false negatives than with no noise, but
the speech remains easily detectable using standard thresholding
algorithms. Even in a moderately loud noise environment (Figure 41)
the R ratio remains significantly above unity, and the VAD once again
returns few false positives. More false negatives are observed, but these
may be reduced using standard methods such as smoothing of R and
allowing the VAD to continue reporting voiced windows for a few
windows after R is under the threshold.
Results using the adaptive beta filter are shown in Figures 42-44.
The adaptive filter used was a five-tap NLMS FIR filter using the
frequency band from 100 Hz to 3500 Hz. A fixed filter of z-0.43 is used to
filter 01 so that 01 and 02 are aligned for speech before the adaptive
filter is calculated. The adaptive filter was constrained using the
methods above using a low p limit of 0.73, a high p limit of 0.98, and a
phase limit ratio of 0.98. Again a fixed threshold was used to generate

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the VAD result from the ratio R, but in this case a threshold value of 2.5
was used since the R values using the adaptive beta filter are normally
greater than when the fixed filter is used. This allows for a reduction of
false positives without significantly increasing false negatives.
Figure 42 shows experimental results of the algorithm using an
adaptive beta when only noise is present, under an embodiment. The
top plot is V11 the middle plot is V2, and the bottom plot is R (solid line)
and the VAD result (dashed line) versus time, with the y-axis expanded
to 0-50. Again, V1 and V2 are very close in energy and the R ratio is
near unity. Only a single false positive was generated.
Figure 43 shows experimental results of the algorithm using an
adaptive beta when only speech is present, under an embodiment. The
top plot is V11 the middle plot is V2, and the bottom plot is (solid line)
and the VAD result (dashed line) versus time, expanded to 0-50. The V2
response is greatly reduced using the adaptive beta, and the R ratio has
increased from the range of approximately 2-7 to the range of
approximately 5-30 on average, making the speech even simpler to
detect using standard thresholding algorithms. There are almost no
false positives or false negatives. Therefore, the response of V2 to
speech is minimal, R is very high, and all of the speech is easily detected
with almost no false positives.
Figure 44 shows experimental results of the algorithm using an
adaptive beta when speech and noise is present, under an embodiment.
The top plot is V1, the middle plot is V2, and the bottom plot is R (solid
line) and the VAD result (dashed line) versus time, with the y-axis
expanded to 0-50. The R ratio is again lower than when no noise is
present, but this R with significant noise present results in a VAD signal
that is about the same as the case using the fixed beta with no noise
present. This shows that use of the adaptive beta allows the system to

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perform well in higher noise environments than the fixed beta.
Therefore, with mixed noise and speech, there are again very few false
positives and fewer false negatives than in the results of Figure 41,
demonstrating that the adaptive filter can outperform the fixed filter in
the same noise environment. In practice, the adaptive filter has proven
to be significantly more sensitive to speech and less sensitive to noise.
DETECTING VOICED AND UNVOICED SPEECH USING BOTH ACOUSTIC AND
NONACOUSTIC SENSORS
Systems and methods for discriminating voiced and unvoiced speech
from background noise are provided below including a Non-Acoustic Sensor
Voiced Speech Activity Detection (NAVSAD) system and a Pathfinder Speech
Activity Detection (PSAD) system. The noise removal and reduction methods
provided herein, while allowing for the separation and classification of
unvoiced
and voiced human speech from background noise, address the shortcomings of
typical systems known in the art by cleaning acoustic signals of interest
without
distortion.
Figure 45 is a block diagram of a NAVSAD system 4500, under an
embodiment. The NAVSAD system couples microphones 10 and sensors 20 to
at least one processor 30. The sensors 20 of an embodiment include voicing
activity detectors or non-acoustic sensors. The processor 30 controls
subsystems including a detection subsystem 50, referred to herein as a
detection algorithm, and a denoising subsystem 40. Operation of the denoising
subsystem 40 is described in detail in the Related Applications. The NAVSAD
system works extremely well in any background acoustic noise environment.
Figure 46 is a block diagram of a PSAD system 4600, under an
embodiment. The PSAD system couples microphones 10 to at least one
processor 30. The processor 30 includes a detection subsystem 50, referred to
herein as a detection algorithm, and a denoising subsystem 40. The PSAD
system is highly sensitive in low acoustic noise environments and relatively
insensitive in high acoustic noise environments. The PSAD can operate

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independently or as a backup to the NAVSAD, detecting voiced speech if the
NAVSAD fails.
Note that the detection subsystems 50 and denoising subsystems 40 of
both the NAVSAD and PSAD systems of an embodiment are algorithms
controlled by the processor 30, but are not so limited. Alternative
embodiments of the NAVSAD and PSAD systems can include detection
subsystems 50 and/or denoising subsystems 40 that comprise additional
hardware, firmware, software, and/or combinations of hardware, firmware, and
software. Furthermore, functions of the detection subsystems 50 and denoising
subsystems 40 may be distributed across numerous components of the
NAVSAD and PSAD systems.
Figure 47 is a block diagram of a denoising subsystem 4700, referred to
herein as the Pathfinder system, under an embodiment. The Pathfinder system
is briefly described below, and is described in detail in the Related
Applications.
Two microphones Mic 1 and Mic 2 are used in the Pathfinder system, and Mic 1
is considered the "signal" microphone. With reference to Figure 45, the
Pathfinder system 4700 is equivalent to the NAVSAD system 4500 when the
voicing activity detector (VAD) 4720 is a non-acoustic voicing sensor 20 and
the
noise removal subsystem 4740 includes the detection subsystem 50 and the
denoising subsystem 40. With reference to Figure 46, the Pathfinder system
4700 is equivalent to the PSAD system 4600 in the absence of the VAD 4720,
and when the noise removal subsystem 4740 includes the detection subsystem
50 and the denoising subsystem 40.
The NAVSAD and PSAD systems support a two-level commercial
approach in which (i) a relatively less expensive PSAD system supports an
acoustic approach that functions in most low- to medium-noise environments,
and (ii) a NAVSAD system adds a non-acoustic sensor to enable detection of
voiced speech in any environment. Unvoiced speech is normally not detected
using the sensor, as it normally does not sufficiently vibrate human tissue.
However, in high noise situations detecting the unvoiced speech is not as
important, as it is normally very low in energy and easily washed out by the
noise. Therefore in high noise environments the unvoiced speech is unlikely to

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affect the voiced speech denoising. Unvoiced speech information is most
important in the presence of little to no noise and, therefore, the unvoiced
detection should be highly sensitive in low noise situations, and insensitive
in
high noise situations. This is not easily accomplished, and comparable
acoustic
unvoiced detectors known in the art are incapable of operating under these
environmental constraints.
The NAVSAD and PSAD systems include an array algorithm for speech
detection that uses the difference in frequency content between two
microphones to calculate a relationship between the signals of the two
microphones. This is in contrast to conventional arrays that attempt to use
the
time/phase difference of each microphone to remove the noise outside of an
"area of sensitivity". The methods described herein provide a significant
advantage, as they do not require a specific orientation of the array with
respect to the signal.
Further, the systems described herein are sensitive to noise of every type
and every orientation, unlike conventional arrays that depend on specific
noise
orientations. Consequently, the frequency-based arrays presented herein are
unique as they depend only on the relative orientation of the two microphones
themselves with no dependence on the orientation of the noise and signal with
respect to the microphones. This results in a robust signal processing system
with respect to the type of noise, microphones, and orientation between the
noise/signal source and the microphones.
The systems described herein use the information derived from the
Pathfinder noise suppression system and/or a non-acoustic sensor described in
the Related Applications to determine the voicing state of an input signal, as
described in detail below. The voicing state includes silent, voiced, and
unvoiced states. The NAVSAD system, for example, includes a non-acoustic
sensor to detect the vibration of human tissue associated with speech. The
non-acoustic sensor of an embodiment is a General Electromagnetic Movement
Sensor (GEMS) as described briefly below and in detail in the Related
Applications, but is not so limited. Alternative embodiments, however, may use


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any sensor that is able to detect human tissue motion associated with speech
and is unaffected by environmental acoustic noise.
The GEMS is a radio frequency device (2.4 GHz) that allows the detection
of moving human tissue dielectric interfaces. The GEMS includes an RF
interferometer that uses homodyne mixing to detect small phase shifts
associated with target motion. In essence, the sensor sends out weak
electromagnetic waves (less than 1 milliwatt) that reflect off of whatever is
around the sensor. The reflected waves are mixed with the original transmitted
waves and the results analyzed for any change in position of the targets.
Anything that moves near the sensor will cause a change in phase of the
reflected wave that will be amplified and displayed as a change in voltage
output from the sensor. A similar sensor is described by Gregory C. Burnett
(1999) in "The physiological basis of glottal electromagnetic micropower
sensors (GEMS) and their use in defining an excitation function for the human
vocal tract"; Ph.D. Thesis, University of California at Davis.
Figure 48 is a flow diagram of a detection algorithm 50 for use in
detecting voiced and unvoiced speech, under an embodiment. With reference
to Figures 45 and 46, both the NAVSADand PSAD systems of an embodiment
include the detection algorithm 50 as the detection subsystem 50. This
detection algorithm 50 operates in real-time and, in an embodiment, operates
on 20 millisecond windows and steps 10 milliseconds at a time, but is not so
limited. The voice activity determination is recorded for the first 10
milliseconds, and the second 10 milliseconds functions as a "look-ahead"
buffer.
While an embodiment uses the 20/10 windows, alternative embodiments may
use numerous other combinations of window values.
Consideration was given to a number of multi-dimensional factors in
developing the detection algorithm 50. The biggest consideration was to
maintaining the effectiveness of the Pathfinder denoising technique, described
in detail in the Related Applications and reviewed herein. Pathfinder
performance can be compromised if the adaptive filter training is conducted on
speech rather than on noise. It is therefore important not to exclude any

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significant amount of speech from the VAD to keep such disturbances to a
minimum.
Consideration was also given to the accuracy of the characterization
between voiced and unvoiced speech signals, and distinguishing each of these
speech signals from noise signals. This type of characterization can be useful
in
such applications as speech recognition and speaker verification.
Furthermore, the systems using the detection algorithm of an
embodiment function in environments containing varying amounts of
background acoustic noise. If the non-acoustic sensor is available, this
external
noise is not a problem for voiced speech. However, for unvoiced speech (and
voiced if the non-acoustic sensor is not available or has malfunctioned)
reliance
is placed on acoustic data alone to separate noise from unvoiced speech. An
advantage inheres in the use of two microphones in an embodiment of the
Pathfinder noise suppression system, and the spatial relationship between the
microphones is exploited to assist in the detection of unvoiced speech.
However, there may occasionally be noise levels high enough that the speech
will be nearly undetectable and the acoustic-only method will fail. In these
situations, the non-acoustic sensor (or hereafter just the sensor) will be
required to ensure good performance.
In the two-microphone system, the speech source should be relatively
louder in one designated microphone when compared to the other microphone.
Tests have shown that this requirement is easily met with conventional
microphones when the microphones are placed on the head, as any noise
should result in an H1 with a gain near unity.
Regarding the NAVSAD system, and with reference to Figure 45 and
Figure 47, the NAVSAD relies on two parameters to detect voiced speech.
These two parameters include the energy of the sensor in the window of
interest, determined in an embodiment by the standard deviation (SD), and
optionally the cross-correlation (XCORR) between the acoustic signal from
microphone 1 and the sensor data. The energy of the sensor can be
determined in any one of a number of ways, and the SD is just one convenient
way to determine the energy.

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For the sensor, the SD is akin to the energy of the signal, which normally
corresponds quite accurately to the voicing state, but may be susceptible to
movement noise (relative motion of the sensor with respect to the human user)
and/or electromagnetic noise. To further differentiate sensor noise from
tissue
motion, the XCORR can be used. The XCORR is only calculated to 15 delays,
which corresponds to just under 2 milliseconds at 8000 Hz.
The XCORR can also be useful when the sensor signal is distorted or
modulated in some fashion. For example, there are sensor locations (such as
the jaw or back of the neck) where speech production can be detected but
where the signal may have incorrect or distorted time-based information. That
is, they may not have well defined features in time that will match with the
acoustic waveform. However, XCORR is more susceptible to errors from
acoustic noise, and in high (<0 dB SNR) environments is almost useless.
Therefore it should not be the sole source of voicing information.
The sensor detects human tissue motion associated with the closure of
the vocal folds, so the acoustic signal produced by the closure of the folds
is
highly correlated with the closures. Therefore, sensor data that correlates
highly with the acoustic signal is declared as speech, and sensor data that
does
not correlate well is termed noise. The acoustic data is expected to lag
behind
the sensor data by about 0.1 to 0.8 milliseconds (or about 1-7 samples) as a
result of the delay time due to the relatively slower speed of sound (around
330
m/s). However, an embodiment uses a 15-sample correlation, as the acoustic
wave shape varies significantly depending on the sound produced, and a larger
correlation width is needed to ensure detection.
The SD and XCORR signals are related, but are sufficiently different so
that the voiced speech detection is more reliable. For simplicity, though,
either
parameter may be used. The values for the SD and XCORR are compared to
empirical thresholds, and if both are above their threshold, voiced speech is
declared. Example data is presented and described below.
Figures 49A, 49B, and 50 show data plots for an example in which a
subject twice speaks the phrase "pop pan", under an embodiment. Figure 49A
plots the received GEMS signal 4902 for this utterance along with the mean

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correlation 4904 between the GEMS signal and the Mic 1 signal and the
threshold T1 used for voiced speech detection. Figure 49B plots the received
GEMS signal 4902 for this utterance along with the standard deviation 4906 of
the GEMS signal and the threshold T2 used for voiced speech detection. Figure
50 plots voiced speech 5002 detected from the acoustic or audio signal 5008,
along with the GEMS signal 5004 and the acoustic noise 5006; no unvoiced
speech is detected in this example because of the heavy background babble
noise 5006. The thresholds have been set so that there are virtually no false
negatives, and only occasional false positives. A voiced speech activity
detection accuracy of greater than 99% has been attained under any acoustic
background noise conditions.
The NAVSAD can determine when voiced speech is occurring with high
degrees of accuracy due to the non-acoustic sensor data. However, the sensor
offers little assistance in separating unvoiced speech from noise, as unvoiced
speech normally causes no detectable signal in most non-acoustic sensors. If
there is a detectable signal, the NAVSAD can be used, although use of the SD
method is dictated as unvoiced speech is normally poorly correlated. In the
absence of a detectable signal use is made of the system and methods of the
Pathfinder noise removal algorithm in determining when unvoiced speech is
occurring. A brief review of the Pathfinder algorithm is described below,
while a
detailed description is provided in the Related Applications.
With reference to Figure 47, the acoustic information coming into
Microphone 1 is denoted by mi(n), the information coming into Microphone 2 is
similarly labeled m2(n), and the GEMS sensor is assumed available to determine
voiced speech areas. In the z (digital frequency) domain, these signals are
represented as M1(z) and M2(z). Then
M, (z) = S(z) + N2 (z)
M2(z)=N(z)+S2(z)
with
N2 (z) = N(z)H, (z)
S2 (Z) = S(z)H2 (z)
so that

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Ml (z) = S(z)+ N(z)HI (z)
(1)
M2(Z)=N(z)+S(Z)H2(Z)
This is the general case for all two microphone systems. There is always
going to be some leakage of noise into Mic 1, and some leakage of signal into
Mic 2. Equation 1 has four unknowns and only two relationships and cannot be
solved explicitly.
However, there is another way to solve for some of the unknowns in
Equation 1. Examine the case where the signal is not being generated - that
is, where the GEMS signal indicates voicing is not occurring. In this case,
s(n)
= S(z) = 0, and Equation 1 reduces to

M,,, (z) = N(z)H1(z)
Men (z) = N(z)
A(Z)

where the n subscript on the M variables indicate that only noise is being
received. This leads to

Mln (z) = MZn (z)H1(z)
HI (z) = Mln (z) (2)
Mtn (z)

H1(z) can be calculated using any of the available system identification
algorithms and the microphone outputs when only noise is being received. The
calculation can be done adaptively, so that if the noise changes significantly
H1(z) can be recalculated quickly.
With a solution for one of the unknowns in Equation 1, solutions can be
found for another, H2(z), by using the amplitude of the GEMS or similar device
along with the amplitude of the two microphones. When the GEMS indicates
voicing, but the recent (less than 1 second) history of the microphones
indicate
low levels of noise, assume that n(s) = N(z) - 0. Then Equation 1 reduces to
MIS (z) S(z)
M2S (z) = S(z)H2 (z)
which in turn leads to



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M25 W = M1s (z)H2 W
H2 (Z) M2,(Z)
1 is(

which is the inverse of the H1(z) calculation, but note that different inputs
are
being used.
After calculating H1(z) and H2(z) above, they are used to remove the
noise from the signal. Rewrite Equation 1 as

S(z) = M, (z) - N(z)H, (z)
N(z) = M2 (z) - S(z)H2 (z)
1 ~,
S(z) = M1(z)- [M2 (z) - S(z)H2 (z)]H1(z)
S(z){1- Hz (z)H1(z)1 = M1 (z) - M2 (z)H1(z)
and solve for S(z) as:

S(z) = M1(z)-M2(z)H1(z) (3)
1- Hz (z)H1 (z)

In practice H2(z) is usually quite small, so that H2(z)H1(z) << 1, and
S(z} MI(Z) -M2(z)H,(z),

obviating the need for the H2(z) calculation.
With reference to Figure 46 and Figure 47, the PSAD system is described.
As sound waves propagate, they normally lose energy as they travel due to
diffraction and dispersion. Assuming the sound waves originate from a point
source
and radiate isotropically, their amplitude will decrease as a function of 1/r,
where r is
the distance from the originating point. This function of 1/r proportional to
amplitude
is the worst case, if confined to a smaller area the reduction will be less.
However it
is an adequate model for the configurations of interest, specifically the
propagation
of noise and speech to microphones located somewhere on the user's head.
Figure 51 is a microphone array for use under an embodiment of the PSAD
system. Placing the microphones Mic 1 and Mic 2 in a linear array with the
mouth
on the array midline, the difference in signal strength in Mic 1 and Mic 2
(assuming
the microphones have identical frequency responses) will be proportional to
both d1
and Ad. Assuming a 1/r (or in this case 1/d) relationship, it is seen that

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AM= Mic1l = AH, (z) oc dl + Ad
IMic2l d,

where AM is the difference in gain between Mic 1 and Mic 2 and therefore
Hi(z), as
above in Equation 2. The variable d1 is the distance from Mic 1 to the speech
or
noise source.
Figure 52 is a plot 5200 of AM versus d, for several Ad values, under an
embodiment. It is clear that as Ad becomes larger and the noise source is
closer,
AM becomes larger. The variable Ad will change depending on the orientation to
the speech/noise source, from the maximum value on the array midline to zero
perpendicular to the array midline. From the plot 5200 it is clear that for
small Ad
and for distances over approximately 30 centimeters (cm), AM is close to
unity.
Since most noise sources are farther away than 30 cm and are unlikely to be on
the
midline on the array, it is probable that when calculating Hi(z) as above in
Equation
2, AM (or equivalently the gain of Hi(z)) will be close to unity. Conversely,
for noise
sources that are close (within a few centimeters), there could be a
substantial
difference in gain depending on which microphone is closer to the noise.
If the "noise" is the user speaking, and Mic 1 is closer to the mouth than Mic
2, the gain increases. Since environmental noise normally originates much
farther
away from the user's head than speech, noise will be found during the time
when
the gain of Hi(z) is near unity or some fixed value, and speech can be found
after a
sharp rise in gain. The speech can be unvoiced or voiced, as long as it is of
sufficient volume compared to the surrounding noise. The gain will stay
somewhat
high during the speech portions, then descend quickly after speech ceases. The
rapid increase and decrease in the gain of Hi(z) should be sufficient to allow
the
detection of speech under almost any circumstances. The gain in this example
is
calculated by the sum of the absolute value of the filter coefficients. This
sum is not
equivalent to the gain, but the two are related in that a rise in the sum of
the
absolute value reflects a rise in the gain.
As an example of this behavior, Figure 53 shows a plot 5300 of the gain
parameter 5302 as the sum of the absolute values of Hi(z) and the acoustic
data
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5304 or audio from microphone 1. The speech signal was an utterance of the
phrase "pop pan", repeated twice. The evaluated bandwidth included the
frequency
range from 2500 Hz to 3500 Hz, although 1500Hz to 2500 Hz was additionally
used
in practice. Note the rapid increase in the gain when the unvoiced speech is
first
encountered, then the rapid return to normal when the speech ends. The large
changes in gain that result from transitions between noise and speech can be
detected by any standard signal processing techniques. The standard deviation
of
the last few gain calculations is used, with thresholds being defined by a
running
average of the standard deviations and the standard deviation noise floor. The
later
changes in gain for the voiced speech are suppressed in this plot 5300 for
clarity.
Figure 54 is an alternative plot 5400 of acoustic data presented in Figure 53.
The data used to form plot 5300 is presented again in this plot 5400, along
with
audio data 5404 and GEMS data 5406 without noise to make the unvoiced speech
apparent. The voiced signal 5402 has three possible values: 0 for noise, 1 for
unvoiced, and 2 for voiced. Denoising is only accomplished when V = 0. It is
clear
that the unvoiced speech is captured very well, aside from two single dropouts
in
the unvoiced detection near the end of each "pop". However, these single-
window
dropouts are not common and do not significantly affect the denoising
algorithm.
They can easily be removed using standard smoothing techniques.
What is not clear from this plot 5400 is that the PSAD system functions as an
automatic backup to the NAVSAD. This is because the voiced speech (since it
has
the same spatial relationship to the mics as the unvoiced) will be detected as
unvoiced if the sensor or NAVSAD system fail for any reason. The voiced speech
will be misclassified as unvoiced, but the denoising will still not take
place,
preserving the quality of the speech signal.
However, this automatic backup of the NAVSAD system functions best in an
environment with low noise (approximately 10+ dB SNR), as high amounts (10 dB
of SNR or less) of acoustic noise can quickly overwhelm any acoustic-only
unvoiced
detector, including the PSAD. This is evident in the difference in the voiced
signal
data 5002 and 5402 shown in plots 5000 and 5400 of Figures 50 and 54,
respectively, where the same utterance is spoken, but the data of plot 5000
shows
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no unvoiced speech because the unvoiced speech is undetectable. This is the
desired behavior when performing denoising, since if the unvoiced speech is
not
detectable then it will not significantly affect the denoising process. Using
the
Pathfinder system to detect unvoiced speech ensures detection of any unvoiced
speech loud enough to distort the denoising.
Regarding hardware considerations, and with reference to Figure 51, the
configuration of the microphones can have an effect on the change in gain
associated with speech and the thresholds needed to detect speech. In general,
each configuration will require testing to determine the proper thresholds,
but tests
with two very different microphone configurations showed the same thresholds
and
other parameters to work well. The first microphone set had the signal
microphone
near the mouth and the noise microphone several centimeters away at the ear,
while the second configuration placed the noise and signal microphones back-to-

back within a few centimeters of the mouth. The results presented herein were
derived using the first microphone configuration, but the results using the
other set
are virtually identical, so the detection algorithm is relatively robust with
respect to
microphone placement.
A number of configurations are possible using the NAVSAD and PSAD
systems to detect voiced and unvoiced speech. One configuration uses the
NAVSAD system (non-acoustic only) to detect voiced speech along with the PSAD
system to detect unvoiced speech; the PSAD also functions as a backup to the
NAVSAD system for detecting voiced speech. An alternative configuration uses
the
NAVSAD system (non-acoustic correlated with acoustic) to detect voiced speech
along with the PSAD system to detect unvoiced speech; the PSAD also functions
as a backup to the NAVSAD system for detecting voiced speech. Another
alternative configuration uses the PSAD system to detect both voiced and
unvoiced
speech.
While the systems described above have been described with reference to
separating voiced and unvoiced speech from background acoustic noise, there
are
no reasons more complex classifications can not be made. For more in-depth
characterization of speech, the system can bandpass the information from Mic 1
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and Mic 2 so that it is possible to see which bands in the Mic 1 data are more
heavily composed of noise and which are more weighted with speech. Using this
knowledge, it is possible to group the utterances by their spectral
characteristics
similar to conventional acoustic methods; this method would work better in
noisy
environments.
As an example, the "k" in "kick" has significant frequency content form 500
Hz to 4000 Hz, but a "sh" in "she" only contains significant energy from 1700-
4000
Hz. Voiced speech could be classified in a similar manner. For instance, an
/i/
("ee") has significant energy around 300 Hz and 2500 Hz, and an /a/ ("ah") has
energy at around 900 Hz and 1200 Hz. This ability to discriminate unvoiced and
voiced speech in the presence of noise is, thus, very useful.

ACOUSTIC VIBRATION SENSOR
An acoustic vibration sensor, also referred to as a speech sensing device,
is described below. The acoustic vibration sensor is similar to a microphone
in
that it captures speech information from the head area of a human talker or
talker in noisy environments. Previous solutions to this problem have either
been vulnerable to noise, physically too large for certain applications, or
cost
prohibitive. In contrast, the acoustic vibration sensor described herein
accurately detects and captures speech vibrations in the presence of
substantial
airborne acoustic noise, yet within a smaller and cheaper physical package.
The noise-immune speech information provided by the acoustic vibration sensor
can subsequently be used in downstream speech processing applications
(speech enhancement and noise suppression, speech encoding, speech
recognition, talker verification, etc.) to improve the performance of those
applications.
Figure 55 is a cross section view of an acoustic vibration sensor 5500,
also referred to herein as the sensor 5500, under an embodiment. Figure 56A
is an exploded view of an acoustic vibration sensor 5500, under the
embodiment of Figure 55. Figure 56B is perspective view of an acoustic
vibration sensor 5500, under the embodiment of Figure 55. The sensor 5500
includes an enclosure 5502 having a first port 5504 on a first side and at
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one second port 5506 on a second side of the enclosure 5502. A diaphragm
5508, also referred to as a sensing diaphragm 5508, is positioned between the
first and second ports. A coupler 5510, also referred to as the shroud 5510 or
cap 5510, forms an acoustic seal around the enclosure 5502 so that the first
port 5504 and the side of the diaphragm facing the first port 5504 are
isolated
from the airborne acoustic environment of the human talker. The coupler 5510
of an embodiment is contiguous, but is not so limited. The second port 5506
couples a second side of the diaphragm to the external environment.
The sensor also includes electret material 5520 and the associated
components and electronics coupled to receive acoustic signals from the talker
via the coupler 5510 and the diaphragm 5508 and convert the acoustic signals
to electrical signals representative of human speech. Electrical contacts 5530
provide the electrical signals as an output. Alternative embodiments can use
any type/combination of materials and/or electronics to convert the acoustic
signals to electrical signals representative of human speech and output the
electrical signals.
The coupler 5510 of an embodiment is formed using materials having
acoustic impedances matched to the impedance of human skin (characteristic
acoustic impedance of skin is approximately 1.5x106 Pax s/m). The coupler
5510 therefore, is formed using a material that includes at least one of
silicone
gel, dielectric gel, thermoplastic elastomers (TPE), and rubber compounds, but
is not so limited. As an example, the coupler 5510 of an embodiment is formed
using Kraiburg TPE products. As another example, the coupler 5510 of an
embodiment is formed using Sylgard Silicone products.
The coupler 5510 of an embodiment includes a contact device 5512 that
includes, for example, a nipple or protrusion that protrudes from either or
both
sides of the coupler 5510. In operation, a contact device 5512 that protrudes
from both sides of the coupler 5510 includes one side of the contact device
5512 that is in contact with the skin surface of the talker and another side
of
the contact device 5512 that is in contact with the diaphragm, but the
embodiment is not so limited. The coupler 5510 and the contact device 5512
can be formed from the same or different materials.

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The coupler 5510 transfers acoustic energy efficiently from skin/flesh of a
talker to the diaphragm, and seals the diaphragm from ambient airborne
acoustic signals. Consequently, the coupler 5510 with the contact device 5512
efficiently transfers acoustic signals directly from the talker's body (speech
vibrations) to the diaphragm while isolating the diaphragm from acoustic
signals in the airborne environment of the talker (characteristic acoustic
impedance of air is approximately 415 Pa x s/m). The diaphragm is isolated
from acoustic signals in the airborne environment of the talker by the coupler
5510 because the coupler 5510 prevents the signals from reaching the
diaphragm, thereby reflecting and/or dissipating much of the energy of the
acoustic signals in the airborne environment. Consequently, the sensor 5500
responds primarily to acoustic energy transferred from the skin of the talker,
not air. When placed against the head of the talker, the sensor 5500 picks up
speech-induced acoustic signals on the surface of the skin while airborne
acoustic noise signals are largely rejected, thereby increasing the signal-to-
noise ratio and providing a very reliable source of speech information.
Performance of the sensor 5500 is enhanced through the use of the seal
provided between the diaphragm and the airborne environment of the talker.
The seal is provided by the coupler 5510. A modified gradient microphone is
used in an embodiment because it has pressure ports on both ends. Thus,
when the first port 5504 is sealed by the coupler 5510, the second port 5506
provides a vent for air movement through the sensor 5500.
Figure 57 is a schematic diagram of a coupler 5510 of an acoustic
vibration sensor, under the embodiment of Figure 55. The dimensions shown
are in millimeters and are only intended to serve as an example for one
embodiment. Alternative embodiments of the coupler can have different
configurations and/or dimensions. The dimensions of the coupler 5510 show
that the acoustic vibration sensor 5500 is small in that the sensor 5500 of an
embodiment is approximately the same size as typical microphone capsules
found in mobile communication devices. This small form factor allows for use
of the sensor 5510 in highly mobile miniaturized applications, where some
example applications include at least one of cellular telephones, satellite

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telephones, portable telephones, wireline telephones, Internet telephones,
wireless transceivers, wireless communication radios, personal digital
assistants
(PDAs), personal computers (PCs), headset devices, head-worn devices, and
earpieces.
The acoustic vibration sensor provides very accurate Voice Activity
Detection (VAD) in high noise environments, where high noise environments
include airborne acoustic environments in which the noise amplitude is as
large
if not larger than the speech amplitude as would be measured by conventional
omnidirectional microphones. Accurate VAD information provides significant
performance and efficiency benefits in a number of important speech
processing applications including but not limited to: noise suppression
algorithms such as the Pathfinder algorithm available from Aliph, Brisbane,
California and described in the Related Applications; speech compression
algorithms such as the Enhanced Variable Rate Coder (EVRC) deployed in many
commercial systems; and speech recognition systems.
In addition to providing signals having an improved signal-to-noise ratio,
the acoustic vibration sensor uses only minimal power to operate (on the order
of 200 micro Amps, for example). In contrast to alternative solutions that
require power, filtering, and/or significant amplification, the acoustic
vibration
sensor uses a standard microphone interface to connect with signal processing
devices. The use of the standard microphone interface avoids the additional
expense and size of interface circuitry in a host device and supports for of
the
sensor in highly mobile applications where power usage is an issue.
Figure 58 is an exploded view of an acoustic vibration sensor 5800,
under an alternative embodiment. The sensor 5800 includes an enclosure 5802
having a first port 5804 on a first side and at least one second port (not
shown)
on a second side of the enclosure 5802. A diaphragm 5808 is positioned
between the first and second ports. A layer of silicone gel 5809 or other
similar
substance is formed in contact with at least a portion of the diaphragm 5808.
A
coupler 5810 or shroud 5810 is formed around the enclosure 5802 and the
silicon gel 5809 where a portion of the coupler 5810 is in contact with the
silicon gel 5809. The coupler 5810 and silicon gel 5809 in combination form an

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acoustic seal around the enclosure 5802 so that the first port 5804 and the
side
of the diaphragm facing the first port 5804 are isolated from the acoustic
environment of the human talker. The second port couples a second side of the
diaphragm to the acoustic environment.
As described above, the sensor includes additional electronic materials as
appropriate that couple to receive acoustic signals from the talker via the
coupler 5810, the silicon gel 5809, and the diaphragm 5808 and convert the
acoustic signals to electrical signals representative of human speech.
Alternative embodiments can use any type/combination of materials and/or
electronics to convert the acoustic signals to electrical signals
representative of
human speech.
The coupler 5810 and/or gel 5809 of an embodiment are formed using
materials having impedances matched to the impedance of human skin. As
such, the coupler 5810 is formed using a material that includes at least one
of
silicone gel, dielectric gel, thermoplastic elastomers (TPE), and rubber
compounds, but is not so limited. The coupler 5810 transfers acoustic energy
efficiently from skin/flesh of a talker to the diaphragm, and seals the
diaphragm
from ambient airborne acoustic signals. Consequently, the coupler 5810
efficiently transfers acoustic signals directly from the talker's body (speech
vibrations) to the diaphragm while isolating the diaphragm from acoustic
signals in the airborne environment of the talker. The diaphragm is isolated
from acoustic signals in the airborne environment of the talker by the silicon
gel
5809/coupler 5810 because the silicon gel 5809/coupler 5810 prevents the
signals from reaching the diaphragm, thereby reflecting and/or dissipating
much of the energy of the acoustic signals in the airborne environment.
Consequently, the sensor 5800 responds primarily to acoustic energy
transferred from the skin of the talker, not air. When placed again the head
of
the talker, the sensor 5800 picks up speech-induced acoustic signals on the
surface of the skin while airborne acoustic noise signals are largely
rejected,
thereby increasing the signal-to-noise ratio and providing a very reliable
source
of speech information.

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There are many locations outside the ear from which the acoustic
vibration sensor'can detect skin vibrations associated with the production of
speech. The sensor can be mounted in a device, handset, or earpiece in any
manner, the only restriction being that reliable skin contact is used to
detect
the skin-borne vibrations associated with the production of speech. Figure 59
shows representative areas of sensitivity 5900-5920 on the human head
appropriate for placement of the acoustic vibration sensor 5500/5800, under an
embodiment. The areas of sensitivity 5900-5920 include numerous locations
5902-5908 in an area behind the ear 5900, at least one location 5912 in an
area in front of the ear 5910, and in numerous locations 5922-5928 in the ear
canal area 5920. The areas of sensitivity 5900-5920 are the same for both
sides of the human head. These representative areas of sensitivity 5900-5920
are provided as examples only and do not limit the embodiments described
herein to use in these areas.
Figure 60 is a generic headset device 6000 that includes an acoustic
vibration sensor 5500/5800 placed at any of a number of locations 6002-6010,
under an embodiment. Generally, placement of the acoustic vibration sensor
5500/5800 can be on any part of the device 6000 that corresponds to the areas
of sensitivity 5900-5920 (Figure 59) on the human head. While a headset
device is shown as an example, any number of communication devices known
in the art can carry and/or couple to an acoustic vibration sensor 5500/5800.
Figure 61 is a diagram of a manufacturing method 6100 for an acoustic
vibration sensor, under an embodiment. Operation begins with, for example, a
uni-directional microphone 6120, at block 6102. Silicon gel 6122 is formed
over/on the diaphragm (not shown) and the associated port, at block 6104. A
material 6124, for example polyurethane film, is formed or placed over the
microphone 6120/silicone gel 6122 combination, at block 6106, to form a
coupler or shroud. A snug fit collar or other device is placed on the
microphone
to secure the material of the coupler during curing, at block 6108.
Note that the silicon gel (block 6102) is an optional component that
depends on the embodiment of the sensor being manufactured, as described
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that includes a contact device 5512 (referring to Figure 55) will not include
the
formation of silicon gel 6122 over/on the diaphragm. Further, the coupler
formed over the microphone for this sensor 5500 will include the contact
device
5512 or formation of the contact device 5512.
The embodiments described herein include a system comprising a first
detector that receives a first signal and a second detector that receives a
second signal. The system of an embodiment comprises a voice activity
detector (VAD) coupled to the first detector. The VAD generates a VAD signal
when the first signal corresponds to voiced speech. The system of an
embodiment comprises a wind detector coupled to the second detector. The
wind detector correlates signals received at the second detector and derives
from the correlation a plurality of wind metrics that characterize wind noise
that
is acoustic disturbance corresponding to at least one of air flow and air
pressure
in the second detector. The wind detector controls a configuration of the
second detector according to the plurality of wind metrics. The wind detector
uses the plurality of wind metrics to dynamically control mixing of the first
signal and the second signal to generate an output signal for transmission.
The embodiments described herein include a system comprising: a first
detector that receives a first signal and a second detector that receives a
second signal; a voice activity detector (VAD) coupled to the first detector,
the
VAD generating a VAD signal when the first signal corresponds to voiced
speech; and a wind detector coupled to the second detector, wherein the wind
detector correlates signals received at the second detector and derives from
the
correlation a plurality of wind metrics that characterize wind noise that is
acoustic disturbance corresponding to at least one of air flow and air
pressure
in the second detector, wherein the wind detector controls a configuration of
the second detector according to the plurality of wind metrics, wherein the
wind
detector uses the plurality of wind metrics to dynamically control mixing of
the
first signal and the second signal to generate an output signal for
transmission.
The first detector of an embodiment is a vibration sensor.
The first detector of an embodiment is a skin surface microphone (SSM).
The second detector of an embodiment is an acoustic sensor.

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The second detector of an embodiment comprises two omnidirectional
microphones.
The two omnidirectional microphones of an embodiment are positioned
adjacent one another and are separated by a distance approximately in a range
of 10 millimeters (mm) to 40 mm.
The wind detector of an embodiment comprises an adaptive filter coupled
to the second detector, wherein the wind detector correlates signals by
calculating energy of an adaptive filter error.
The wind detector of an embodiment comprises a first exponential
averaging filter and a second exponential averaging filter coupled to the
adaptive filter, wherein the wind detector applies the energy to the first
exponential averaging filter and the second exponential averaging filter.
The wind detector of an embodiment generates an instantaneous wind
level from the energy, wherein the instantaneous wind level represents an
instant wind level of the wind noise.
The plurality of wind metrics of an embodiment comprise a wind present
metric that characterizes the instantaneous wind level relative to a present
wind threshold over which the wind noise negatively affects electronic
operations in a host electronic system.
The plurality of wind metrics of an embodiment comprise a wind mode
metric that characterizes the instantaneous wind level relative to a wind high
threshold over which the wind noise is considered to have a relatively high
impact on audio intelligibility in a host electronic system.
The wind detector of an embodiment generates a current wind level from
the energy, wherein the current wind level represents an average current wind
level of the wind noise.
The plurality of wind metrics of an embodiment comprise a wind index
metric that characterizes the current wind level relative to a minimum wind
threshold under which the wind noise is considered to have a negligible impact
on noise suppression and audio intelligibility in a host electronic system.
The plurality of wind metrics of an embodiment comprise a wind mode
metric that the wind detector generates to control the configuration of the
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second detector, wherein the wind mode metric characterizes instantaneous
wind level relative to a wind high threshold over which the wind noise is
considered to have a relatively high impact on audio intelligibility in a host
electronic system.
In response to the wind mode metric indicating that instantaneous wind
level exceeds the wind high threshold, the wind detector of an embodiment
controls the configuration of the second detector by controlling generation of
a
summed detector signal by summing signals from each of two microphones of
the second detector.
The wind detector of an embodiment controls the configuration of the
second detector by controlling application of single-microphone noise
suppression to the summed detector signal.
The wind detector of an embodiment controls the configuration of the
second detector by controlling separate processing of signals from each of two
microphones of the second detector when the wind mode metric indicates
instantaneous wind level is below the wind high threshold.
The wind detector of an embodiment controls the configuration of the
second detector by controlling application of dual-microphone noise
suppression
to the signals from the two microphones.
The system of an embodiment comprises a gain controller coupled to the
first detector and the wind detector.
The gain controller of an embodiment controls gain applied to the first
signal in response to the plurality of wind metrics and the VAD signal.
The gain controller of an embodiment adjusts a gain applied to the first
signal when the plurality of wind metrics indicates no wind is present.
The plurality of wind metrics of an embodiment includes a wind present
metric that characterizes an instantaneous wind level derived from the second
signal relative to a present wind threshold over which the wind noise
negatively
affects electronic operations in a host electronic system.
The system of an embodiment comprises adjusting the gain to match a
first root mean square (RMS) of the first signal to a second RMS of a noise-
suppressed speech signal.

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The system of an embodiment comprises generating a VAD signal when
the first signal corresponds to voiced speech, and using the VAD signal to
noise
gate the first signal.
The gain controller of an embodiment adjusts a gain applied to the first
signal when the VAD signal indicates the first signal corresponds to voiced
speech.
The system of an embodiment comprises a first filter coupled to the first
detector and a second filter coupled to the second detector.
The first filter of an embodiment is a low-pass filter and the second filter
is a high-pass filter.
The plurality of wind metrics of an embodiment dynamically control
mixing of the first signal and the second signal.
The plurality of wind metrics of an embodiment dynamically adjust a
response of the first filter to which the first signal is applied and
dynamically
adjust a response of the second filter to which the second signal is applied.
The plurality of wind metrics of an embodiment comprise a wind index
metric that characterizes a current wind level relative to a minimum wind
threshold under which the wind noise is considered to have a negligible impact
on noise suppression and audio intelligibility in a host electronic system,
wherein the current wind level represents an average current wind level of the
wind noise.
The system of an embodiment comprises estimating a wind frequency
response of the wind noise from the wind index metric.
The system of an embodiment comprises a comfort equalizer coupled to
the second detector.
The comfort equalizer of an embodiment generates a comfort wind
component and adds the comfort wind component to audio signals, wherein the
comfort wind component provides listener awareness of wind presence.
The comfort equalizer of an embodiment is coupled to a transmitter, and
adds the comfort wind component to audio signals processed for transmission.
The comfort equalizer of an embodiment is coupled to a receiver, and
adds the comfort wind component to audio signals processed for reception.
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The comfort equalizer of an embodiment generates the comfort wind
component by subtracting signals from each of two microphones of the second
detector to generate a difference signal.
The system of an embodiment comprises modulating the difference
signal by a gain to generate a modulated signal.
The gain of an embodiment comprises a static gain that provides an
appropriate level of wind noise feedback in a loudspeaker.
The gain of an embodiment comprises a gating factor derived from a
wind present metric output by the wind detector, wherein the wind present
metric characterizes an instantaneous wind level derived from the second
signal
relative to a present wind threshold over which the wind noise negatively
affects electronic operations in a host electronic system.
The system of an embodiment comprises filtering the modulated signal to
provide the comfort wind component, the filtering comprising limiting an
amount of low-frequency wind noise and high-frequency wind noise.
The embodiments described herein include a system comprising a first
detector that receives a first signal and a second detector that receives a
second signal. The system of an embodiment comprises a voice activity
detector (VAD) coupled to the first detector. The VAD generates a VAD signal
when the first signal corresponds to voiced speech. The system of an
embodiment comprises a wind detector coupled to the second detector. The
wind detector correlates signals -received at the second detector and derives
from the correlation.a plurality of wind metrics that characterize wind noise
that
is acoustic disturbance corresponding to at least one of air flow and air
pressure
in the second detector. The wind detector uses the plurality of wind metrics
to
dynamically control mixing of the first signal and the second signal to
generate
an output signal for transmission.
The embodiments described herein include a system comprising: a first
detector that receives a first signal and a second detector that receives a
second signal; a voice activity detector (VAD) coupled to the first detector,
the
VAD generating a VAD signal when the first signal corresponds to voiced
speech; and a wind detector coupled to the second detector, wherein the wind



CA 02798282 2012-11-02
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detector correlates signals received at the second detector and derives from
the
correlation a plurality of wind metrics that characterize wind noise that is
acoustic disturbance corresponding to at least one of air flow and air
pressure
in the second detector, wherein the wind detector uses the plurality of wind
metrics to dynamically control mixing of the first signal and the second
signal to
generate an output signal for transmission.
The embodiments described herein include a system comprising a first
detector that receives a first signal and a second detector that receives a
second signal. The system of an embodiment comprises a voice activity
detector (VAD) coupled to the first detector. The VAD generates a VAD signal
when the first signal corresponds to voiced speech. The system of an
embodiment comprises a wind detector coupled to the second detector. The
wind detector correlates signals received at the second detector and derives
from the correlation a plurality of wind metrics that characterize wind noise
that
is acoustic disturbance corresponding to at least one of air flow and air
pressure
in the second detector. The wind detector controls a configuration of the
second detector according to the plurality of wind metrics.
The embodiments described herein include a system comprising: a first
detector that receives a first signal and a second detector that receives a
second signal; a voice activity detector (VAD) coupled to the first detector,
the
VAD generating a VAD signal when the first signal corresponds to voiced
speech; and a wind detector coupled to the second detector, wherein the wind
detector correlates signals received at the second detector and derives from
the
correlation a plurality of wind metrics that characterize wind noise that is
acoustic disturbance corresponding to at least one of air flow and air
pressure
in the second detector, wherein the wind detector controls a configuration of
the second detector according to the plurality of wind metrics.
The embodiments described herein include a method comprising
receiving a first signal at a first detector and a second signal at a second
detector. The method of an embodiment comprises determining a correlation
between signals received at the second detector and deriving from the
correlation a plurality of wind metrics that characterize wind noise that is

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acoustic disturbance corresponding to at least one of air flow and air
pressure
in the second detector. The method of an embodiment comprises controlling a
configuration of the second detector according to the plurality of wind
metrics.
The method of an embodiment comprises generating an output signal for
transmission by dynamically mixing the first signal and the second signal
according to the plurality of wind metrics.
The embodiments described herein include a method comprising:
receiving a first signal at a first detector and a second signal at a second
detector; determining a correlation between signals received at the second
detector and deriving from the correlation a plurality of wind metrics that
characterize wind noise that is acoustic disturbance corresponding to at least
one of air flow and air pressure in the second detector; controlling a
configuration of the second detector according to the plurality of wind
metrics;
and generating an output signal for transmission by dynamically mixing the
first
signal and the second signal according to the plurality of wind metrics.
The first detector of an embodiment is a vibration sensor.
The first detector of an embodiment is a skin surface microphone (SSM).
The second detector of an embodiment is an acoustic sensor.
The second detector of an embodiment comprises two omnidirectional
microphones.
The method of an embodiment comprises positioning the two
omnidirectional microphones adjacent one another and separating the two
omnidirectional microphones by a distance approximately in a range of 10
millimeters (mm) to 40 mm.
Determining the correlation of an embodiment comprises calculating
energy of an adaptive filter error.
The method of an embodiment comprises applying the energy to a first
exponential averaging filter and a second exponential averaging filter.
The method of an embodiment comprises deriving an instantaneous wind
level from the energy, wherein the instantaneous wind level represents an
instant wind level of the wind noise.

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The plurality of wind metrics of an embodiment comprise a wind present
metric that characterizes the instantaneous wind level relative to a present
wind threshold over which the wind noise negatively affects electronic
operations in a host electronic system.
The plurality of wind metrics of an embodiment comprise a wind mode
metric that characterizes the instantaneous wind level relative to a wind high
threshold over which the wind noise is considered to have a relatively high
impact on audio intelligibility in a host electronic system.
The method of an embodiment comprises deriving a current wind level
from the energy, wherein the current wind level represents an average current
wind level of the wind noise.
The plurality of wind metrics of an embodiment comprise a wind index
metric that characterizes the current wind level relative to a minimum wind
threshold under which the wind noise is considered to have a negligible impact
on noise suppression and audio intelligibility in a host electronic system.
The method of an embodiment comprises controlling a gain applied to
the first signal in response to the plurality of wind metrics and a voice
activity
detection (VAD) signal.
The method of an embodiment comprises adjusting the gain when the
plurality of wind metrics indicates no wind is present.
The plurality of wind metrics of an embodiment is a wind present metric
that characterizes an instantaneous wind level derived from the second signal
relative to a present wind threshold over which the wind noise negatively
affects electronic operations in a host electronic system.
The method of an embodiment comprises adjusting the gain when the
VAD signal indicates the first signal corresponds to voiced speech.
The method of an embodiment comprises adjusting the gain to match a
first root mean square (RMS) of the first signal to a second RMS of a noise-
suppressed speech signal.
The method of an embodiment comprises generating a VAD signal when
the first signal corresponds to voiced speech, and using the VAD signal to
noise
gate the first signal.

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The controlling of the configuration of the second detector of an
embodiment according to the plurality of wind metrics comprises use of a wind
mode metric that characterizes instantaneous wind level relative to a wind
high
threshold over which the wind noise is considered to have a relatively high
impact on audio intelligibility in a host electronic system.
The controlling of the configuration of the second detector of an
embodiment comprises, when the wind mode metric indicates instantaneous
wind level exceeds the wind high threshold, generating a summed detector
signal by summing signals from each of two microphones of the second
detector.
The controlling of the configuration of the second detector of an
embodiment comprises applying single-microphone noise suppression to the
summed detector signal.
The controlling of the configuration of the second detector of an
embodiment comprises separately processing signals from each of two
microphones of the second detector when the wind mode metric indicates
instantaneous wind level is below the wind high threshold.
The controlling of the configuration of the second detector of an
embodiment comprises applying dual-microphone noise suppression to the
signals from the two microphones.
Dynamically mixing the first signal and the second signal of an
embodiment according to the plurality of wind metrics comprises dynamically
adjusting a response of a first filter to which the first signal is applied
and
dynamically adjusting a response of a second filter to which the second signal
is
applied.
The first filter of an embodiment is a low-pass filter and the second filter
of an embodiment is a high-pass filter.
The plurality of wind metrics of an embodiment is a wind index metric
that characterizes a current wind level relative to a minimum wind threshold
under which the wind noise is considered to have a negligible impact on noise
suppression and audio intelligibility in a host electronic system, wherein the
current wind level represents an average current wind level of the wind noise.
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The method of an embodiment comprises estimating a wind frequency
response of the wind noise from the wind index metric.
The method of an embodiment comprises generating a comfort wind
component and adding the comfort wind component to receive and transmit
audio, wherein the comfort wind component provides listener awareness of
wind presence.
The method of an embodiment comprises generating the comfort wind
component by subtracting signals from each of two microphones of the second
detector to generate a difference signal.
The method of an embodiment comprises modulating the difference
signal by a gain to generate a modulated signal.
The gain of an embodiment comprises a static gain that provides an
appropriate level of wind noise feedback in a loudspeaker.
The gain of an embodiment comprises a gating factor derived from a
wind present metric that characterizes an instantaneous wind level derived
from the second signal relative to a present wind threshold over which the
wind
noise negatively affects electronic operations in a host electronic system.
The method of an embodiment comprises filtering the modulated signal
to provide the comfort wind component, the filtering comprising limiting an
amount of low-frequency wind noise and high-frequency wind noise reaching a
receiver.
The embodiments described herein include a method comprising
receiving a first signal at a first detector and a second signal at a second
detector. The method of an embodiment comprises determining a correlation
between signals received at the second detector and deriving from the
correlation a plurality of wind metrics that characterize wind noise that is
acoustic disturbance corresponding to at least one of air flow and air
pressure
in the second detector. The method of an embodiment comprises controlling a
configuration of the second detector according to the plurality of wind
metrics.
The embodiments described herein include a method comprising:
receiving a first signal at a first detector and a second signal at a second
detector; determining a correlation between signals received at the second



CA 02798282 2012-11-02
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detector and deriving from the correlation a plurality of wind metrics that
characterize wind noise that is acoustic disturbance corresponding to at least
one of air flow and air pressure in the second detector; and controlling a
configuration of the second detector according to the plurality of wind
metrics.
The embodiments described herein include a method comprising
receiving a first signal at a first detector and a second signal at a second
detector. The method of an embodiment comprises determining a correlation
between signals received at the second detector and deriving from the
correlation a plurality of wind metrics that characterize wind noise that is
acoustic disturbance corresponding to at least one of air flow and air
pressure
in the second detector. The method of an embodiment comprises generating
an output signal for transmission by dynamically mixing the first signal and
the
second signal according to the plurality of wind metrics.
The embodiments described herein include a method comprising:
receiving a first signal at a first detector and a second signal at a second
detector; determining a correlation between signals received at the second
detector and deriving from the correlation a plurality of wind metrics that
characterize wind noise that is acoustic disturbance corresponding tout least
one of air flow and air pressure in the second detector; and generating an
output signal for transmission by dynamically mixing the first signal and the
second signal according to the plurality of wind metrics.
The systems and methods described herein include and/or run under
and/or in association with a processing system. The processing system
includes any collection of processor-based devices or computing devices
operating together, or components of processing systems or devices, as is
known in the art. For example, the processing system can include one or more
of a portable computer, portable communication device operating in a
communication network, and/or a network server. The portable computer can
be any of a number and/or combination of devices selected from among
personal computers, cellular telephones, personal digital assistants, portable
computing devices, and portable communication devices, but is not so limited.
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The processing system can include components within a larger computer
system.
The processing system of an embodiment includes at least one processor
and at least one memory device or subsystem. The processing system can also
include or be coupled to at least one database. The term "processor" as
generally used herein refers to any logic processing unit, such as one or more
central processing units (CPUs), digital signal processors (DSPs), application-

specific integrated circuits (ASIC), etc. The processor and memory can be
monolithically integrated onto a single chip, distributed among a number of
chips or components of a host system, and/or provided by some combination of
algorithms. The methods described herein can be implemented in one or more
of software algorithm(s), programs, firmware, hardware, components, circuitry,
in any combination.
System components embodying the systems and methods described
herein can be located together or in separate locations. Consequently, system
components embodying the systems and methods described herein can be
components of a single system, multiple systems, and/or geographically
separate systems. These components can also be subcomponents or
subsystems of a single system, multiple systems, and/or geographically
separate systems. These components can be coupled to one or more other
components of a host system or a system coupled to the host system.
Communication paths couple the system components and include any
medium for communicating or transferring files among the components. The
communication paths include wireless connections, wired connections, and
hybrid wireless/wired connections. The communication paths also include
couplings or connections to networks including local area networks (LANs),
metropolitan area networks (MANs), wide area networks (WANs), proprietary
networks, interoffice or backend networks, and the Internet. Furthermore, the
communication paths include removable fixed mediums like floppy disks, hard
disk drives, and CD-ROM disks, as well as flash RAM, Universal Serial Bus
(USB)
connections, RS-232 connections, telephone lines, buses, and electronic mail
messages.

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Unless the context clearly requires otherwise, throughout the description,
the words "comprise," "comprising," and the like are to be construed in an
inclusive sense as opposed to an exclusive or exhaustive sense; that is to
say,
in a sense of "including, but not limited to." Additionally, the words
"herein,"
"hereunder," "above," "below," and words of similar import refer to this
application as a whole and not to any particular portions of this application.
When the word "or" is used in reference to a list of two or more items, that
word covers all of the following interpretations of the word: any of the items
in
the list, all of the items in the list and any combination of the items in the
list.
The above description of embodiments is not intended to be exhaustive
or to limit the systems and methods described to the precise form disclosed.
While specific embodiments and examples are described herein for illustrative
purposes, various equivalent modifications are possible within the scope of
other systems and methods, as those skilled in the relevant art will
recognize.
The teachings provided herein can be applied to other processing systems and
methods, not only for the systems and methods described above.
The elements and acts of the various embodiments described above can
be combined to provide further embodiments. These and other changes can be
made to the embodiments in light of the above detailed description.
In general, in the following claims, the terms used should not be
construed to limit the embodiments described herein and corresponding
systems and methods to the specific embodiments disclosed in the specification
and the claims, but should be construed to include all systems and methods
that operate under the claims. Accordingly, the embodiments described herein
are not limited by the disclosure, but instead the scope is to be determined
entirely by the claims.
While certain aspects of the embodiments described herein are presented
below in certain claim forms, the inventors contemplate the various aspects of
the embodiments and corresponding systems and methods in any number of
claim forms. Accordingly, the inventors reserve the right to add additional
claims after filing the application to pursue such additional claim forms for
other
aspects of the embodiments described herein.

83

A single figure which represents the drawing illustrating the invention.

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Title Date
Forecasted Issue Date Unavailable
(86) PCT Filing Date 2011-05-03
(87) PCT Publication Date 2011-11-10
(85) National Entry 2012-11-02
Examination Requested 2016-05-03
Dead Application 2018-10-02

Abandonment History

Abandonment Date Reason Reinstatement Date
2017-10-02 R30(2) - Failure to Respond
2018-05-03 FAILURE TO PAY APPLICATION MAINTENANCE FEE

Payment History

Fee Type Anniversary Year Due Date Amount Paid Paid Date
Filing $400.00 2012-11-02
Maintenance Fee - Application - New Act 2 2013-05-03 $100.00 2013-05-03
Maintenance Fee - Application - New Act 3 2014-05-05 $100.00 2014-05-01
Maintenance Fee - Application - New Act 4 2015-05-04 $100.00 2015-04-08
Registration of Documents $100.00 2015-08-26
Maintenance Fee - Application - New Act 5 2016-05-03 $200.00 2016-04-13
Request for Examination $800.00 2016-05-03
Maintenance Fee - Application - New Act 6 2017-05-03 $200.00 2017-04-05
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Current Owners on Record
ALIPHCOM
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None
Past Owners that do not appear in the "Owners on Record" listing will appear in other documentation within the application.

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Description 2012-11-02 83 3,636
Representative Drawing 2012-11-02 1 24
Cover Page 2013-01-08 1 52
PCT 2012-11-02 13 966
Assignment 2012-11-02 2 65
Correspondence 2013-03-25 2 53
Correspondence 2013-04-03 1 16
Correspondence 2013-04-03 1 15
Correspondence 2013-04-15 1 15
Correspondence 2013-04-15 1 17
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Correspondence 2015-12-21 1 49
Prosecution-Amendment 2016-05-03 2 82
Prosecution-Amendment 2017-03-31 4 225
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